



MMUSIC Working Group                                      H. Schulzrinne
Internet-Draft                                       Columbia University
Obsoletes: 2326 (if approved)                                     A. Rao
Intended status: Standards Track                                   Cisco
Expires: August 14, 2014                                     R. Lanphier

                                                           M. Westerlund
                                                             Ericsson AB
                                                    M. Stiemerling (Ed.)
                                                                     NEC
                                                       February 10, 2014


                Real Time Streaming Protocol 2.0 (RTSP)
                    draft-ietf-mmusic-rfc2326bis-40

Abstract

   This memorandum defines RTSP version 2.0 which obsoletes RTSP version
   1.0 defined in RFC 2326.

   The Real Time Streaming Protocol, or RTSP, is an application-layer
   protocol for setup and control of the delivery of data with real-time
   properties.  RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video.  Sources of data can include both live data feeds and stored
   clips.  This protocol is intended to control multiple data delivery
   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and provide a means for choosing delivery
   mechanisms based upon RTP (RFC 3550).

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on August 14, 2014.




Schulzrinne, et al.      Expires August 14, 2014                [Page 1]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

   This document may contain material from IETF Documents or IETF
   Contributions published or made publicly available before November
   10, 2008.  The person(s) controlling the copyright in some of this
   material may not have granted the IETF Trust the right to allow
   modifications of such material outside the IETF Standards Process.
   Without obtaining an adequate license from the person(s) controlling
   the copyright in such materials, this document may not be modified
   outside the IETF Standards Process, and derivative works of it may
   not be created outside the IETF Standards Process, except to format
   it for publication as an RFC or to translate it into languages other
   than English.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .  10
   2.  Protocol Overview . . . . . . . . . . . . . . . . . . . . . .  11
     2.1.  Presentation Description  . . . . . . . . . . . . . . . .  11
     2.2.  Session Establishment . . . . . . . . . . . . . . . . . .  12
     2.3.  Media Delivery Control  . . . . . . . . . . . . . . . . .  13
     2.4.  Session Parameter Manipulations . . . . . . . . . . . . .  15
     2.5.  Media Delivery  . . . . . . . . . . . . . . . . . . . . .  16
       2.5.1.  Media Delivery Manipulations  . . . . . . . . . . . .  16
     2.6.  Session Maintenance and Termination . . . . . . . . . . .  19
     2.7.  Extending RTSP  . . . . . . . . . . . . . . . . . . . . .  20
   3.  Document Conventions  . . . . . . . . . . . . . . . . . . . .  21
     3.1.  Notational Conventions  . . . . . . . . . . . . . . . . .  21
     3.2.  Terminology . . . . . . . . . . . . . . . . . . . . . . .  21
   4.  Protocol Parameters . . . . . . . . . . . . . . . . . . . . .  24
     4.1.  RTSP Version  . . . . . . . . . . . . . . . . . . . . . .  24
     4.2.  RTSP IRI and URI  . . . . . . . . . . . . . . . . . . . .  25
     4.3.  Session Identifiers . . . . . . . . . . . . . . . . . . .  27
     4.4.  Media Time Formats  . . . . . . . . . . . . . . . . . . .  27
       4.4.1.  SMPTE Relative Timestamps . . . . . . . . . . . . . .  28



Schulzrinne, et al.      Expires August 14, 2014                [Page 2]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


       4.4.2.  Normal Play Time  . . . . . . . . . . . . . . . . . .  28
       4.4.3.  Absolute Time . . . . . . . . . . . . . . . . . . . .  30
     4.5.  Feature-Tags  . . . . . . . . . . . . . . . . . . . . . .  30
     4.6.  Message Body Tags . . . . . . . . . . . . . . . . . . . .  31
     4.7.  Media Properties  . . . . . . . . . . . . . . . . . . . .  31
       4.7.1.  Random Access and Seeking . . . . . . . . . . . . . .  32
       4.7.2.  Retention . . . . . . . . . . . . . . . . . . . . . .  33
       4.7.3.  Content Modifications . . . . . . . . . . . . . . . .  33
       4.7.4.  Supported Scale Factors . . . . . . . . . . . . . . .  33
       4.7.5.  Mapping to the Attributes . . . . . . . . . . . . . .  34
   5.  RTSP Message  . . . . . . . . . . . . . . . . . . . . . . . .  34
     5.1.  Message Types . . . . . . . . . . . . . . . . . . . . . .  34
     5.2.  Message Headers . . . . . . . . . . . . . . . . . . . . .  35
     5.3.  Message Body  . . . . . . . . . . . . . . . . . . . . . .  36
     5.4.  Message Length  . . . . . . . . . . . . . . . . . . . . .  36
   6.  General Header Fields . . . . . . . . . . . . . . . . . . . .  36
   7.  Request . . . . . . . . . . . . . . . . . . . . . . . . . . .  38
     7.1.  Request Line  . . . . . . . . . . . . . . . . . . . . . .  38
     7.2.  Request Header Fields . . . . . . . . . . . . . . . . . .  40
   8.  Response  . . . . . . . . . . . . . . . . . . . . . . . . . .  42
     8.1.  Status-Line . . . . . . . . . . . . . . . . . . . . . . .  42
       8.1.1.  Status Code and Reason Phrase . . . . . . . . . . . .  42
     8.2.  Response Headers  . . . . . . . . . . . . . . . . . . . .  46
   9.  Message Body  . . . . . . . . . . . . . . . . . . . . . . . .  46
     9.1.  Message-Body Header Fields  . . . . . . . . . . . . . . .  47
     9.2.  Message Body  . . . . . . . . . . . . . . . . . . . . . .  48
     9.3.  Message Body Format Negotiation . . . . . . . . . . . . .  48
   10. Connections . . . . . . . . . . . . . . . . . . . . . . . . .  49
     10.1.  Reliability and Acknowledgements . . . . . . . . . . . .  49
     10.2.  Using Connections  . . . . . . . . . . . . . . . . . . .  50
     10.3.  Closing Connections  . . . . . . . . . . . . . . . . . .  53
     10.4.  Timing Out Connections and RTSP Messages . . . . . . . .  54
     10.5.  Showing Liveness . . . . . . . . . . . . . . . . . . . .  55
     10.6.  Use of IPv6  . . . . . . . . . . . . . . . . . . . . . .  57
     10.7.  Overload Control . . . . . . . . . . . . . . . . . . . .  57
   11. Capability Handling . . . . . . . . . . . . . . . . . . . . .  58
     11.1.  Feature Tag: play.basic  . . . . . . . . . . . . . . . .  60
   12. Pipelining Support  . . . . . . . . . . . . . . . . . . . . .  61
   13. Method Definitions  . . . . . . . . . . . . . . . . . . . . .  61
     13.1.  OPTIONS  . . . . . . . . . . . . . . . . . . . . . . . .  63
     13.2.  DESCRIBE . . . . . . . . . . . . . . . . . . . . . . . .  64
     13.3.  SETUP  . . . . . . . . . . . . . . . . . . . . . . . . .  66
       13.3.1.  Changing Transport Parameters  . . . . . . . . . . .  69
     13.4.  PLAY . . . . . . . . . . . . . . . . . . . . . . . . . .  70
       13.4.1.  General Usage  . . . . . . . . . . . . . . . . . . .  70
       13.4.2.  Aggregated Sessions  . . . . . . . . . . . . . . . .  75
       13.4.3.  Updating current PLAY Requests . . . . . . . . . . .  76
       13.4.4.  Playing On-Demand Media  . . . . . . . . . . . . . .  79



Schulzrinne, et al.      Expires August 14, 2014                [Page 3]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


       13.4.5.  Playing Dynamic On-Demand Media  . . . . . . . . . .  79
       13.4.6.  Playing Live Media . . . . . . . . . . . . . . . . .  79
       13.4.7.  Playing Live with Recording  . . . . . . . . . . . .  80
       13.4.8.  Playing Live with Time-Shift . . . . . . . . . . . .  81
     13.5.  PLAY_NOTIFY  . . . . . . . . . . . . . . . . . . . . . .  81
       13.5.1.  End-of-Stream  . . . . . . . . . . . . . . . . . . .  82
       13.5.2.  Media-Properties-Update  . . . . . . . . . . . . . .  84
       13.5.3.  Scale-Change . . . . . . . . . . . . . . . . . . . .  85
     13.6.  PAUSE  . . . . . . . . . . . . . . . . . . . . . . . . .  86
     13.7.  TEARDOWN . . . . . . . . . . . . . . . . . . . . . . . .  89
       13.7.1.  Client to Server . . . . . . . . . . . . . . . . . .  89
       13.7.2.  Server to Client . . . . . . . . . . . . . . . . . .  90
     13.8.  GET_PARAMETER  . . . . . . . . . . . . . . . . . . . . .  91
     13.9.  SET_PARAMETER  . . . . . . . . . . . . . . . . . . . . .  93
     13.10. REDIRECT . . . . . . . . . . . . . . . . . . . . . . . .  95
   14. Embedded (Interleaved) Binary Data  . . . . . . . . . . . . .  97
   15. Proxies . . . . . . . . . . . . . . . . . . . . . . . . . . .  99
     15.1.  Proxies and Protocol Extensions  . . . . . . . . . . . . 101
     15.2.  Multiplexing and Demultiplexing of Messages  . . . . . . 102
   16. Caching . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
     16.1.   Validation Model  . . . . . . . . . . . . . . . . . . . 103
       16.1.1.  Last-Modified Dates  . . . . . . . . . . . . . . . . 104
       16.1.2.  Message Body Tag Cache Validators  . . . . . . . . . 104
       16.1.3.  Weak and Strong Validators . . . . . . . . . . . . . 104
       16.1.4.  Rules for When to Use Message Body Tags and Last-
                Modified Dates . . . . . . . . . . . . . . . . . . . 107
       16.1.5.  Non-validating Conditionals  . . . . . . . . . . . . 108
     16.2.  Invalidation After Updates or Deletions  . . . . . . . . 108
   17. Status Code Definitions . . . . . . . . . . . . . . . . . . . 109
     17.1.  Informational 1xx  . . . . . . . . . . . . . . . . . . . 109
       17.1.1.  100 Continue . . . . . . . . . . . . . . . . . . . . 109
     17.2.  Success 2xx  . . . . . . . . . . . . . . . . . . . . . . 110
       17.2.1.  200 OK . . . . . . . . . . . . . . . . . . . . . . . 110
     17.3.  Redirection 3xx  . . . . . . . . . . . . . . . . . . . . 110
       17.3.1.  300  . . . . . . . . . . . . . . . . . . . . . . . . 111
       17.3.2.  301 Moved Permanently  . . . . . . . . . . . . . . . 111
       17.3.3.  302 Found  . . . . . . . . . . . . . . . . . . . . . 111
       17.3.4.  303 See Other  . . . . . . . . . . . . . . . . . . . 111
       17.3.5.  304 Not Modified . . . . . . . . . . . . . . . . . . 111
       17.3.6.  305 Use Proxy  . . . . . . . . . . . . . . . . . . . 112
     17.4.  Client Error 4xx . . . . . . . . . . . . . . . . . . . . 112
       17.4.1.  400 Bad Request  . . . . . . . . . . . . . . . . . . 112
       17.4.2.  401 Unauthorized . . . . . . . . . . . . . . . . . . 112
       17.4.3.  402 Payment Required . . . . . . . . . . . . . . . . 113
       17.4.4.  403 Forbidden  . . . . . . . . . . . . . . . . . . . 113
       17.4.5.  404 Not Found  . . . . . . . . . . . . . . . . . . . 113
       17.4.6.  405 Method Not Allowed . . . . . . . . . . . . . . . 113
       17.4.7.  406 Not Acceptable . . . . . . . . . . . . . . . . . 113



Schulzrinne, et al.      Expires August 14, 2014                [Page 4]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


       17.4.8.  407 Proxy Authentication Required  . . . . . . . . . 114
       17.4.9.  408 Request Timeout  . . . . . . . . . . . . . . . . 114
       17.4.10. 410 Gone . . . . . . . . . . . . . . . . . . . . . . 114
       17.4.11. 411 Length Required  . . . . . . . . . . . . . . . . 114
       17.4.12. 412 Precondition Failed  . . . . . . . . . . . . . . 114
       17.4.13. 413 Request Message Body Too Large . . . . . . . . . 115
       17.4.14. 414 Request-URI Too Long . . . . . . . . . . . . . . 115
       17.4.15. 415 Unsupported Media Type . . . . . . . . . . . . . 115
       17.4.16. 451 Parameter Not Understood . . . . . . . . . . . . 115
       17.4.17. 452 reserved . . . . . . . . . . . . . . . . . . . . 115
       17.4.18. 453 Not Enough Bandwidth . . . . . . . . . . . . . . 116
       17.4.19. 454 Session Not Found  . . . . . . . . . . . . . . . 116
       17.4.20. 455 Method Not Valid in This State . . . . . . . . . 116
       17.4.21. 456 Header Field Not Valid for Resource  . . . . . . 116
       17.4.22. 457 Invalid Range  . . . . . . . . . . . . . . . . . 116
       17.4.23. 458 Parameter Is Read-Only . . . . . . . . . . . . . 116
       17.4.24. 459 Aggregate Operation Not Allowed  . . . . . . . . 116
       17.4.25. 460 Only Aggregate Operation Allowed . . . . . . . . 116
       17.4.26. 461 Unsupported Transport  . . . . . . . . . . . . . 117
       17.4.27. 462 Destination Unreachable  . . . . . . . . . . . . 117
       17.4.28. 463 Destination Prohibited . . . . . . . . . . . . . 117
       17.4.29. 464 Data Transport Not Ready Yet . . . . . . . . . . 117
       17.4.30. 465 Notification Reason Unknown  . . . . . . . . . . 117
       17.4.31. 466 Key Management Error . . . . . . . . . . . . . . 117
       17.4.32. 470 Connection Authorization Required  . . . . . . . 118
       17.4.33. 471 Connection Credentials not accepted  . . . . . . 118
       17.4.34. 472 Failure to establish secure connection . . . . . 118
     17.5.  Server Error 5xx . . . . . . . . . . . . . . . . . . . . 118
       17.5.1.  500 Internal Server Error  . . . . . . . . . . . . . 118
       17.5.2.  501 Not Implemented  . . . . . . . . . . . . . . . . 118
       17.5.3.  502 Bad Gateway  . . . . . . . . . . . . . . . . . . 118
       17.5.4.  503 Service Unavailable  . . . . . . . . . . . . . . 119
       17.5.5.  504 Gateway Timeout  . . . . . . . . . . . . . . . . 119
       17.5.6.  505 RTSP Version Not Supported . . . . . . . . . . . 119
       17.5.7.  551 Option not supported . . . . . . . . . . . . . . 119
       17.5.8.  553 Proxy Unavailable  . . . . . . . . . . . . . . . 119
   18. Header Field Definitions  . . . . . . . . . . . . . . . . . . 120
     18.1.  Accept . . . . . . . . . . . . . . . . . . . . . . . . . 131
     18.2.  Accept-Credentials . . . . . . . . . . . . . . . . . . . 131
     18.3.  Accept-Encoding  . . . . . . . . . . . . . . . . . . . . 132
     18.4.  Accept-Language  . . . . . . . . . . . . . . . . . . . . 133
     18.5.  Accept-Ranges  . . . . . . . . . . . . . . . . . . . . . 134
     18.6.  Allow  . . . . . . . . . . . . . . . . . . . . . . . . . 134
     18.7.  Authentication-Info  . . . . . . . . . . . . . . . . . . 135
     18.8.  Authorization  . . . . . . . . . . . . . . . . . . . . . 135
     18.9.  Bandwidth  . . . . . . . . . . . . . . . . . . . . . . . 136
     18.10. Blocksize  . . . . . . . . . . . . . . . . . . . . . . . 136
     18.11. Cache-Control  . . . . . . . . . . . . . . . . . . . . . 137



Schulzrinne, et al.      Expires August 14, 2014                [Page 5]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     18.12. Connection . . . . . . . . . . . . . . . . . . . . . . . 139
     18.13. Connection-Credentials . . . . . . . . . . . . . . . . . 140
     18.14. Content-Base . . . . . . . . . . . . . . . . . . . . . . 141
     18.15. Content-Encoding . . . . . . . . . . . . . . . . . . . . 141
     18.16. Content-Language . . . . . . . . . . . . . . . . . . . . 142
     18.17. Content-Length . . . . . . . . . . . . . . . . . . . . . 143
     18.18. Content-Location . . . . . . . . . . . . . . . . . . . . 143
     18.19. Content-Type . . . . . . . . . . . . . . . . . . . . . . 144
     18.20. CSeq . . . . . . . . . . . . . . . . . . . . . . . . . . 144
     18.21. Date . . . . . . . . . . . . . . . . . . . . . . . . . . 146
     18.22. Expires  . . . . . . . . . . . . . . . . . . . . . . . . 147
     18.23. From . . . . . . . . . . . . . . . . . . . . . . . . . . 148
     18.24. If-Match . . . . . . . . . . . . . . . . . . . . . . . . 148
     18.25. If-Modified-Since  . . . . . . . . . . . . . . . . . . . 149
     18.26. If-None-Match  . . . . . . . . . . . . . . . . . . . . . 149
     18.27. Last-Modified  . . . . . . . . . . . . . . . . . . . . . 150
     18.28. Location . . . . . . . . . . . . . . . . . . . . . . . . 150
     18.29. Media-Properties . . . . . . . . . . . . . . . . . . . . 151
     18.30. Media-Range  . . . . . . . . . . . . . . . . . . . . . . 153
     18.31. MTag . . . . . . . . . . . . . . . . . . . . . . . . . . 153
     18.32. Notify-Reason  . . . . . . . . . . . . . . . . . . . . . 154
     18.33. Pipelined-Requests . . . . . . . . . . . . . . . . . . . 154
     18.34. Proxy-Authenticate . . . . . . . . . . . . . . . . . . . 155
     18.35. Proxy-Authentication-Info  . . . . . . . . . . . . . . . 155
     18.36. Proxy-Authorization  . . . . . . . . . . . . . . . . . . 156
     18.37. Proxy-Require  . . . . . . . . . . . . . . . . . . . . . 156
     18.38. Proxy-Supported  . . . . . . . . . . . . . . . . . . . . 156
     18.39. Public . . . . . . . . . . . . . . . . . . . . . . . . . 157
     18.40. Range  . . . . . . . . . . . . . . . . . . . . . . . . . 158
     18.41. Referrer . . . . . . . . . . . . . . . . . . . . . . . . 160
     18.42. Request-Status . . . . . . . . . . . . . . . . . . . . . 160
     18.43. Require  . . . . . . . . . . . . . . . . . . . . . . . . 161
     18.44. Retry-After  . . . . . . . . . . . . . . . . . . . . . . 162
     18.45. RTP-Info . . . . . . . . . . . . . . . . . . . . . . . . 162
     18.46. Scale  . . . . . . . . . . . . . . . . . . . . . . . . . 164
     18.47. Seek-Style . . . . . . . . . . . . . . . . . . . . . . . 165
     18.48. Server . . . . . . . . . . . . . . . . . . . . . . . . . 167
     18.49. Session  . . . . . . . . . . . . . . . . . . . . . . . . 167
     18.50. Speed  . . . . . . . . . . . . . . . . . . . . . . . . . 168
     18.51. Supported  . . . . . . . . . . . . . . . . . . . . . . . 169
     18.52. Terminate-Reason . . . . . . . . . . . . . . . . . . . . 170
     18.53. Timestamp  . . . . . . . . . . . . . . . . . . . . . . . 170
     18.54. Transport  . . . . . . . . . . . . . . . . . . . . . . . 171
     18.55. Unsupported  . . . . . . . . . . . . . . . . . . . . . . 178
     18.56. User-Agent . . . . . . . . . . . . . . . . . . . . . . . 178
     18.57. Via  . . . . . . . . . . . . . . . . . . . . . . . . . . 179
     18.58. WWW-Authenticate . . . . . . . . . . . . . . . . . . . . 179
   19. Security Framework  . . . . . . . . . . . . . . . . . . . . . 180



Schulzrinne, et al.      Expires August 14, 2014                [Page 6]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     19.1.  RTSP and HTTP Authentication . . . . . . . . . . . . . . 180
       19.1.1.  Digest Authentication  . . . . . . . . . . . . . . . 181
     19.2.  RTSP over TLS  . . . . . . . . . . . . . . . . . . . . . 182
     19.3.  Security and Proxies . . . . . . . . . . . . . . . . . . 183
       19.3.1.  Accept-Credentials . . . . . . . . . . . . . . . . . 184
       19.3.2.  User approved TLS procedure  . . . . . . . . . . . . 185
   20. Syntax  . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
     20.1.  Base Syntax  . . . . . . . . . . . . . . . . . . . . . . 187
     20.2.  RTSP Protocol Definition . . . . . . . . . . . . . . . . 189
       20.2.1.  Generic Protocol elements  . . . . . . . . . . . . . 190
       20.2.2.  Message Syntax . . . . . . . . . . . . . . . . . . . 192
       20.2.3.  Header Syntax  . . . . . . . . . . . . . . . . . . . 196
     20.3.  SDP extension Syntax . . . . . . . . . . . . . . . . . . 205
   21. Security Considerations . . . . . . . . . . . . . . . . . . . 205
     21.1.  Signaling Protocol Threats . . . . . . . . . . . . . . . 206
     21.2.  Media Stream Delivery Threats  . . . . . . . . . . . . . 209
       21.2.1.  Remote Denial of Service Attack  . . . . . . . . . . 210
       21.2.2.  RTP Security analysis  . . . . . . . . . . . . . . . 211
   22. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 212
     22.1.  Feature-tags . . . . . . . . . . . . . . . . . . . . . . 213
       22.1.1.  Description  . . . . . . . . . . . . . . . . . . . . 214
       22.1.2.  Registering New Feature-tags with IANA . . . . . . . 214
       22.1.3.  Registered entries . . . . . . . . . . . . . . . . . 214
     22.2.  RTSP Methods . . . . . . . . . . . . . . . . . . . . . . 215
       22.2.1.  Description  . . . . . . . . . . . . . . . . . . . . 215
       22.2.2.  Registering New Methods with IANA  . . . . . . . . . 215
       22.2.3.  Registered Entries . . . . . . . . . . . . . . . . . 216
     22.3.  RTSP Status Codes  . . . . . . . . . . . . . . . . . . . 216
       22.3.1.  Description  . . . . . . . . . . . . . . . . . . . . 216
       22.3.2.  Registering New Status Codes with IANA . . . . . . . 216
       22.3.3.  Registered Entries . . . . . . . . . . . . . . . . . 217
     22.4.  RTSP Headers . . . . . . . . . . . . . . . . . . . . . . 217
       22.4.1.  Description  . . . . . . . . . . . . . . . . . . . . 217
       22.4.2.  Registering New Headers with IANA  . . . . . . . . . 217
       22.4.3.  Registered entries . . . . . . . . . . . . . . . . . 217
     22.5.  Accept-Credentials . . . . . . . . . . . . . . . . . . . 219
       22.5.1.  Accept-Credentials policies  . . . . . . . . . . . . 219
       22.5.2.  Accept-Credentials hash algorithms . . . . . . . . . 219
     22.6.  Cache-Control Cache Directive Extensions . . . . . . . . 220
     22.7.  Media Properties . . . . . . . . . . . . . . . . . . . . 221
       22.7.1.  Description  . . . . . . . . . . . . . . . . . . . . 221
       22.7.2.  Registration Rules . . . . . . . . . . . . . . . . . 221
       22.7.3.  Registered Values  . . . . . . . . . . . . . . . . . 221
     22.8.  Notify-Reason header . . . . . . . . . . . . . . . . . . 222
       22.8.1.  Description  . . . . . . . . . . . . . . . . . . . . 222
       22.8.2.  Registration Rules . . . . . . . . . . . . . . . . . 222
       22.8.3.  Registered Values  . . . . . . . . . . . . . . . . . 222
     22.9.  Range Header Formats . . . . . . . . . . . . . . . . . . 223



Schulzrinne, et al.      Expires August 14, 2014                [Page 7]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


       22.9.1.  Description  . . . . . . . . . . . . . . . . . . . . 223
       22.9.2.  Registration Rules . . . . . . . . . . . . . . . . . 223
       22.9.3.  Registered Values  . . . . . . . . . . . . . . . . . 223
     22.10. Terminate-Reason Header  . . . . . . . . . . . . . . . . 223
       22.10.1.  Redirect Reasons  . . . . . . . . . . . . . . . . . 224
       22.10.2.  Terminate-Reason Header Parameters  . . . . . . . . 224
     22.11. RTP-Info header parameters . . . . . . . . . . . . . . . 225
       22.11.1.  Description . . . . . . . . . . . . . . . . . . . . 225
       22.11.2.  Registration Rules  . . . . . . . . . . . . . . . . 225
       22.11.3.  Registered Values . . . . . . . . . . . . . . . . . 225
     22.12. Seek-Style Policies  . . . . . . . . . . . . . . . . . . 225
       22.12.1.  Description . . . . . . . . . . . . . . . . . . . . 226
       22.12.2.  Registration Rules  . . . . . . . . . . . . . . . . 226
       22.12.3.  Registered Values . . . . . . . . . . . . . . . . . 226
     22.13. Transport Header Registries  . . . . . . . . . . . . . . 227
       22.13.1.  Transport Protocol Identifier . . . . . . . . . . . 227
       22.13.2.  Transport modes . . . . . . . . . . . . . . . . . . 228
       22.13.3.  Transport Parameters  . . . . . . . . . . . . . . . 229
     22.14. URI Schemes  . . . . . . . . . . . . . . . . . . . . . . 230
       22.14.1.  The rtsp URI Scheme . . . . . . . . . . . . . . . . 230
       22.14.2.  The rtsps URI Scheme  . . . . . . . . . . . . . . . 231
       22.14.3.  The rtspu URI Scheme  . . . . . . . . . . . . . . . 232
     22.15. SDP attributes . . . . . . . . . . . . . . . . . . . . . 233
     22.16. Media Type Registration for text/parameters  . . . . . . 234
   23. References  . . . . . . . . . . . . . . . . . . . . . . . . . 235
     23.1.  Normative References . . . . . . . . . . . . . . . . . . 235
     23.2.  Informative References . . . . . . . . . . . . . . . . . 239
   Appendix A.  Examples . . . . . . . . . . . . . . . . . . . . . . 241
     A.1.  Media on Demand (Unicast) . . . . . . . . . . . . . . . . 241
     A.2.  Media on Demand using Pipelining  . . . . . . . . . . . . 245
     A.3.  Secured Media Session for on Demand Content . . . . . . . 247
     A.4.  Media on Demand (Unicast) . . . . . . . . . . . . . . . . 250
     A.5.  Single Stream Container Files . . . . . . . . . . . . . . 254
     A.6.  Live Media Presentation Using Multicast . . . . . . . . . 256
     A.7.  Capability Negotiation  . . . . . . . . . . . . . . . . . 257
   Appendix B.  RTSP Protocol State Machine  . . . . . . . . . . . . 258
     B.1.  States  . . . . . . . . . . . . . . . . . . . . . . . . . 259
     B.2.  State variables . . . . . . . . . . . . . . . . . . . . . 259
     B.3.  Abbreviations . . . . . . . . . . . . . . . . . . . . . . 259
     B.4.  State Tables  . . . . . . . . . . . . . . . . . . . . . . 260
   Appendix C.  Media Transport Alternatives . . . . . . . . . . . . 264
     C.1.  RTP . . . . . . . . . . . . . . . . . . . . . . . . . . . 264
       C.1.1.  AVP . . . . . . . . . . . . . . . . . . . . . . . . . 265
       C.1.2.  AVP/UDP . . . . . . . . . . . . . . . . . . . . . . . 265
       C.1.3.  AVPF/UDP  . . . . . . . . . . . . . . . . . . . . . . 266
       C.1.4.  SAVP/UDP  . . . . . . . . . . . . . . . . . . . . . . 267
       C.1.5.  SAVPF/UDP . . . . . . . . . . . . . . . . . . . . . . 269
       C.1.6.  RTCP usage with RTSP  . . . . . . . . . . . . . . . . 269



Schulzrinne, et al.      Expires August 14, 2014                [Page 8]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     C.2.  RTP over TCP  . . . . . . . . . . . . . . . . . . . . . . 271
       C.2.1.  Interleaved RTP over TCP  . . . . . . . . . . . . . . 271
       C.2.2.  RTP over independent TCP  . . . . . . . . . . . . . . 272
     C.3.  Handling Media Clock Time Jumps in the RTP Media Layer  . 276
     C.4.  Handling RTP Timestamps after PAUSE . . . . . . . . . . . 280
     C.5.  RTSP / RTP Integration  . . . . . . . . . . . . . . . . . 282
     C.6.  Scaling with RTP  . . . . . . . . . . . . . . . . . . . . 282
     C.7.  Maintaining NPT synchronization with RTP timestamps . . . 282
     C.8.  Continuous Audio  . . . . . . . . . . . . . . . . . . . . 282
     C.9.  Multiple Sources in an RTP Session  . . . . . . . . . . . 282
     C.10. Usage of SSRCs and the RTCP BYE Message During an RTSP
           Session . . . . . . . . . . . . . . . . . . . . . . . . . 283
     C.11. Future Additions  . . . . . . . . . . . . . . . . . . . . 283
   Appendix D.  Use of SDP for RTSP Session Descriptions . . . . . . 284
     D.1.  Definitions . . . . . . . . . . . . . . . . . . . . . . . 284
       D.1.1.  Control URI . . . . . . . . . . . . . . . . . . . . . 284
       D.1.2.  Media Streams . . . . . . . . . . . . . . . . . . . . 286
       D.1.3.  Payload Type(s) . . . . . . . . . . . . . . . . . . . 286
       D.1.4.  Format-Specific Parameters  . . . . . . . . . . . . . 286
       D.1.5.  Directionality of media stream  . . . . . . . . . . . 287
       D.1.6.  Range of Presentation . . . . . . . . . . . . . . . . 287
       D.1.7.  Time of Availability  . . . . . . . . . . . . . . . . 288
       D.1.8.  Connection Information  . . . . . . . . . . . . . . . 288
       D.1.9.  Message Body Tag  . . . . . . . . . . . . . . . . . . 289
     D.2.  Aggregate Control Not Available . . . . . . . . . . . . . 289
     D.3.  Aggregate Control Available . . . . . . . . . . . . . . . 290
     D.4.  Grouping of Media Lines in SDP  . . . . . . . . . . . . . 291
     D.5.  RTSP external SDP delivery  . . . . . . . . . . . . . . . 292
   Appendix E.  RTSP Use Cases . . . . . . . . . . . . . . . . . . . 292
     E.1.  On-demand Playback of Stored Content  . . . . . . . . . . 292
     E.2.  Unicast Distribution of Live Content  . . . . . . . . . . 294
     E.3.  On-demand Playback using Multicast  . . . . . . . . . . . 294
     E.4.  Inviting an RTSP server into a conference . . . . . . . . 295
     E.5.  Live Content using Multicast  . . . . . . . . . . . . . . 296
   Appendix F.  Text format for Parameters . . . . . . . . . . . . . 296
   Appendix G.  Requirements for Unreliable Transport of RTSP  . . . 297
   Appendix H.  Backwards Compatibility Considerations . . . . . . . 298
     H.1.  Play Request in Play State  . . . . . . . . . . . . . . . 299
     H.2.  Using Persistent Connections  . . . . . . . . . . . . . . 299
   Appendix I.  Changes  . . . . . . . . . . . . . . . . . . . . . . 299
     I.1.  Brief Overview  . . . . . . . . . . . . . . . . . . . . . 299
     I.2.  Detailed List of Changes  . . . . . . . . . . . . . . . . 300
   Appendix J.  Acknowledgements . . . . . . . . . . . . . . . . . . 307
     J.1.  Contributors  . . . . . . . . . . . . . . . . . . . . . . 308
   Appendix K.  RFC Editor Consideration . . . . . . . . . . . . . . 308
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . . 308





Schulzrinne, et al.      Expires August 14, 2014                [Page 9]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


1.  Introduction

   This memo defines version 2.0 of the Real Time Streaming Protocol
   (RTSP 2.0).  RTSP 2.0 is an application-layer protocol for setup and
   control over the delivery of data with real-time properties,
   typically streaming media.  Streaming media is, for instance, video
   on demand or audio live streaming.  Put simply, RTSP acts as a
   "network remote control" for multimedia servers.

   The protocol operates between RTSP 2.0 clients and servers, but also
   supports the usage of proxies placed between clients and servers.
   Clients can request information about streaming media from servers by
   asking for a description of the media or use media description
   provided externally.  The media delivery protocol is used to
   establish the media streams described by the media description.
   Clients can then request to play out the media, pause it, or stop it
   completely.  The requested media can consist of multiple audio and
   video streams that are delivered as time-synchronized streams from
   servers to clients.

   RTSP 2.0 is a replacement of RTSP 1.0 [RFC2326] and obsoletes that
   specification.  This protocol is based on RTSP 1.0 but is not
   backwards compatible other than in the basic version negotiation
   mechanism.  The changes are documented in Appendix I.  There are many
   reasons why RTSP 2.0 can't be backwards compatible with RTSP 1.0 but
   some of the main ones are:

   o  Most headers that needed to be extensible did not define the
      allowed syntax, preventing safe deployment of extensions;

   o  The changed behavior of the PLAY method when received in Play
      state;

   o  Changed behavior of the extensibility model and its mechanism;

   o  The change of syntax for some headers.

   In summary, there are so many small details that changing version
   became necessary to enable clarification and consistent behavior.
   Anyone implementing RTSP for a new usage where they have no installed
   based of RTSP 1.0 should only implement RTSP 2.0 to avoid having to
   deal with RTSP 1.0 inconsistencies.

   This document is structured as follows.  It begins with an overview
   of the protocol operations and its functions in an informal way.
   Then a set of definitions of terms used and document conventions is
   introduced.  It is followed by the actual RTSP 2.0 core protocol
   specification.  The appendixes describe and define some



Schulzrinne, et al.      Expires August 14, 2014               [Page 10]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   functionalities that are not part of the core RTSP specification, but
   which are still important to enable some usages.  Among them, the RTP
   usage is defined in Appendix C, the Session Description Protocol
   (SDP) usage with RTSP is defined in Appendix D, and the text/
   parameters file format Appendix F, are three normative specification
   appendixes.  Others include a number of informational parts
   discussing the changes, use cases, different considerations or
   motivations.

2.  Protocol Overview

   This section provides an informative overview of the different
   mechanisms in the RTSP 2.0 protocol, to give the reader a high level
   understanding before getting into all the different details.  In case
   of conflict with this description and the later sections, the later
   sections take precedence.  For more information about use cases
   considered for RTSP see Appendix E.

   RTSP 2.0 is a bi-directional request and response protocol that first
   establishes a context including content resources (the media) and
   then controls the delivery of these content resources from the
   provider to the consumer.  RTSP has three fundamental parts: Session
   Establishment, Media Delivery Control, and an extensibility model
   described below.  The protocol is based on some assumptions about
   existing functionality to provide a complete solution for client
   controlled real-time media delivery.

   RTSP uses text-based messages, requests and responses, that may
   contain a binary message body.  An RTSP request starts with a method
   line that identifies the method, the protocol and version and the
   resource to act on.  The resource is identified by a URI and the
   hostname part of the URI is used by RTSP client to resolve the IPv4
   or IPv6 address of the RTSP server.  Following the method line are a
   number of RTSP headers.  This part is ended by two consecutive
   carriage return line feed (CRLF) character pairs.  The message body
   if present follows the two CRLF and the body's length is described by
   a message header.  RTSP responses are similar, but start with a
   response line with the protocol and version, followed by a status
   code and a reason phrase.  RTSP messages are sent over a reliable
   transport protocol between the client and server.  RTSP 2.0 requires
   clients and servers to implement TCP, and TLS over TCP, as mandatory
   transports for RTSP messages.

2.1.  Presentation Description

   RTSP exists to provide access to multi-media presentations and
   content, but tries to be agnostic about the media type or the actual
   media delivery protocol that is used.  To enable a client to



Schulzrinne, et al.      Expires August 14, 2014               [Page 11]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   implement a complete system, an RTSP-external mechanism for
   describing the presentation and the delivery protocol(s) is used.
   RTSP assumes that this description is either delivered completely out
   of band or as a data object in the response to a client's request
   using the DESCRIBE method (Section 13.2).

   Parameters that commonly have to be included in the Presentation
   Description are the following:

   o  Number of media streams;

   o  The resource identifier for each media stream/resource that is to
      be controlled by RTSP;

   o  The protocol that each media stream is to be delivered over;

   o  Transport protocol parameters that are not negotiated or vary with
      each client;

   o  Media encoding information enabling a client to correctly decode
      the media upon reception;

   o  An aggregate control resource identifier.

   RTSP uses its own URI schemes ("rtsp" and "rtsps") to reference media
   resources and aggregates under common control (See Section 4.2).

   This specification describes in Appendix D how one uses SDP [RFC4566]
   for Presentation Description

2.2.  Session Establishment

   The RTSP client can request the establishment of an RTSP session
   after having used the presentation description to determine which
   media streams are available, which media delivery protocol is used
   and the resource identifiers of the media streams.  The RTSP session
   is a common context between the client and the server that consists
   of one or more media resources that are to be under common media
   delivery control.

   The client creates an RTSP session by sending a request using the
   SETUP method (Section 13.3) to the server.  In the "Transport" header
   (Section 18.54) of the SETUP request, the client also includes all
   the transport parameters necessary to enable the media delivery
   protocol to function.  This includes parameters that are pre-
   established by the presentation description but necessary for any
   middlebox to correctly handle the media delivery protocols.  The
   Transport header in a request may contain multiple alternatives for



Schulzrinne, et al.      Expires August 14, 2014               [Page 12]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   media delivery in a prioritized list, which the server can select
   from.  These alternatives are typically based on information in the
   presentation description.

   The server determines if the media resource is available upon
   receiving a SETUP request and if any of the transport parameter
   specifications are acceptable.  If that is successful, an RTSP
   session context is created and the relevant parameters and state is
   stored.  An identifier is created for the RTSP session and included
   in the response in the Session header (Section 18.49).  The SETUP
   response includes a Transport header that specifies which of the
   alternatives has been selected and relevant parameters.

   A SETUP request that references an existing RTSP session but
   identifies a new media resource is a request to add that media
   resource under common control with the already present media
   resources in an aggregated session.  A client can expect this to work
   for all media resources under RTSP control within a multi-media
   content.  However, aggregating resources from different content are
   likely to be refused by the server.  Even if a RTSP session contains
   only a single media, the RTSP session can be referenced by the
   aggregate control URI.

   To avoid an extra round trip in the session establishment of
   aggregated RTSP sessions, RTSP 2.0 supports pipelined requests; i.e.,
   the client can send multiple requests back-to-back without waiting
   first for the completion of any of them.  The client uses a client-
   selected identifier in the Pipelined-Requests header (Section 18.33)
   to instruct the server to bind multiple requests together as if they
   included the session identifier.

   The SETUP response also provides additional information about the
   established sessions in a couple of different headers.  The Media-
   Properties header (Section 18.29) includes a number of properties
   that apply for the aggregate that is valuable when doing media
   delivery control and configuring user interface.  The Accept-Ranges
   header (Section 18.5) informs the client about which range formats
   that the server supports with these media resources.  The Media-Range
   header (Section 18.30) informs the client about the time range of the
   media currently available.

2.3.  Media Delivery Control

   After having established an RTSP session, the client can start
   controlling the media delivery.  The basic operations are Start by
   using the PLAY method (Section 13.4) and Halt by using the PAUSE
   method (Section 13.6).  PLAY also allows for choosing the starting
   media position from which the server should deliver the media.  The



Schulzrinne, et al.      Expires August 14, 2014               [Page 13]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   positioning is done by using the Range header (Section 18.40) that
   supports several different time formats: Normal Play Time (NPT)
   (Section 4.4.2), Society of Motion Picture and Television Engineers
   (SMPTE) Timestamps (Section 4.4.1) and absolute time (Section 4.4.3).
   The Range header does further allow the client to specify a position
   where delivery should end, thus allowing a specific interval to be
   delivered.

   The support for positioning/searching within a media content depends
   on the content's media properties.  Content exists in a number of
   different types, such as: on-demand, live, and live with simultaneous
   recording.  Even within these categories there are differences in how
   the content is generated and distributed, which affect how it can be
   accessed for playback.  The properties applicable for the RTSP
   session are provided by the server in the SETUP response using the
   Media-Properties header (Section 18.29).  These are expressed using
   one or several independent attributes.  A first attribute is Random
   Access, which expresses if positioning can be done, and with what
   granularity.  Another aspect is whether the content will change
   during the lifetime of the session.  While on-demand content will be
   provided in full from the beginning, a live stream being recorded
   results in the length of the accessible content growing as the
   session goes on.  There also exists content that is dynamically built
   by another protocol than RTSP and thus also changes in steps during
   the session, but maybe not continuously.  Furthermore, when content
   is recorded, there are cases where not the complete content is
   maintained, but, for example, only the last hour.  All these
   properties result in the need for mechanisms that will be discussed
   below.

   When the client accesses on-demand content that allows random access,
   the client can issue the PLAY request for any point in the content
   between the start and the end.  The server will deliver media from
   the closest random access point prior to the requested point and
   indicate that in its PLAY response.  If the client issues a PAUSE,
   the delivery will be halted and the point at which the server stopped
   will be reported back in the response.  The client can later resume
   by sending a PLAY request without a range header.  When the server is
   about to complete the PLAY request by delivering the end of the
   content or the requested range, the server will send a PLAY_NOTIFY
   request (Section 13.5) indicating this.

   When playing live content with no extra functions, such as recording,
   the client will receive the live media from the server after having
   sent a PLAY request.  Seeking in such content is not possible as the
   server does not store it, but only forwards it from the source of the
   session.  Thus delivery continues until the client sends a PAUSE
   request, tears down the session, or the content ends.



Schulzrinne, et al.      Expires August 14, 2014               [Page 14]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   For live sessions that are being recorded the client will need to
   keep track of how the recording progresses.  Upon session
   establishment the client will learn the current duration of the
   recording from the Media-Range header.  As the recording is ongoing
   the content grows in direct relation to the passed time.  Therefore,
   each server's response to a PLAY request will contain the current
   Media-Range header.  The server should also regularly send
   approximately every 5 minutes the current media range in a
   PLAY_NOTIFY request (Section 13.5.2).  If the live transmission ends,
   the server must send a PLAY_NOTIFY request with the updated Media-
   Properties indicating that the content stopped being a recorded live
   session and instead became on-demand content; the request also
   contains the final media range.  While the live delivery continues
   the client can request to play the current live point by using the
   NPT timescale symbol "now", or it can request a specific point in the
   available content by an explicit range request for that point.  If
   the requested point is outside of the available interval the server
   will adjust the position to the closest available point, i.e., either
   at the beginning or the end.

   A special case of recording is that where the recording is not
   retained longer than a specific time period, thus as the live
   delivery continues the client can access any media within a moving
   window that covers, for example, "now" to "now" minus 1 hour.  A
   client that pauses on a specific point within the content may not be
   able to retrieve the content anymore.  If the client waits too long
   before resuming the pause point, the content may no longer be
   available.  In this case the pause point will be adjusted to the
   closest point in the available media.

2.4.  Session Parameter Manipulations

   A session may have additional state or functionality that affects how
   the server or client treats the session, content, how it functions,
   or feedback on how well the session works.  Such extensions are not
   defined in this specification, but may be done in various extensions.
   RTSP has two methods for retrieving and setting parameter values on
   either the client or the server: GET_PARAMETER (Section 13.8) and
   SET_PARAMETER (Section 13.9).  These methods carry the parameters in
   a message body of the appropriate format.  One can also use headers
   to query state with the GET_PARAMETER method.  As an example, clients
   needing to know the current media-range for a time-progressing
   session can use the GET_PARAMETER method and include the media-range.
   Furthermore, synchronization information can be requested by using a
   combination of RTP-Info (Section 18.45) and Range (Section 18.40).

   RTSP 2.0 does not have a strong mechanism for providing negotiation
   of the headers, or parameters and their formats, that can be used.



Schulzrinne, et al.      Expires August 14, 2014               [Page 15]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   However, responses will indicate request-headers or parameters that
   are not supported.  A priori determination of what features are
   available needs to be done through out-of-band mechanisms, like the
   session description, or through the usage of feature tags
   (Section 4.5).

2.5.  Media Delivery

   This document specifies how media is delivered with RTP [RFC3550]
   over UDP [RFC0768], TCP [RFC0793] or the RTSP connection.  Additional
   protocols may be specified in the future based on demand.

   The usage of RTP as media delivery protocol requires some additional
   information to function well.  The PLAY response contains information
   to enable reliable and timely delivery of how a client should
   synchronize different sources in the different RTP sessions.  It also
   provides a mapping between RTP timestamps and the content time scale.
   When the server wants to notify the client about the completion of
   the media delivery, it sends a PLAY_NOTIFY request to the client.
   The PLAY_NOTIFY request includes information about the stream end,
   including the last RTP sequence number for each stream, thus enabling
   the client to empty the buffer smoothly.

2.5.1.  Media Delivery Manipulations

   The basic playback functionality of RTSP enables delivery of a range
   of requested content to the client at the pace intended by the
   content's creator.  However, RTSP can also manipulate the delivery to
   the client in two ways.

   Scale:  The ratio of media content time delivered per unit playback
      time.

   Speed:  The ratio of playback time delivered per unit of wallclock
      time.

   Both affect the media delivery per time unit.  However, they
   manipulate two independent time scales and the effects are possible
   to combine.

   Scale (Section 18.46) is used for fast forward or slow motion control
   as it changes the amount of content timescale that should be played
   back per time unit.  Scale > 1.0, means fast forward, e.g., Scale=2.0
   results in that 2 seconds of content is played back every second of
   playback.  Scale = 1.0 is the default value that is used if no Scale
   is specified, i.e., playback at the content's original rate.  Scale
   values between 0 and 1.0 is providing for slow motion.  Scale can be
   negative to allow for reverse playback in either regular pace (Scale



Schulzrinne, et al.      Expires August 14, 2014               [Page 16]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   = -1.0) or fast backwards (Scale < -1.0) or slow motion backwards
   (-1.0 < Scale < 0).  Scale = 0 would be equal to pause and is not
   allowed.

   In most cases the realization of scale means server side manipulation
   of the media to ensure that the client can actually play it back.
   The nature of these media manipulations and when they are needed is
   highly media-type dependent.  Let's consider an example with two
   common media types audio and video.

   It is very difficult to modify the playback rate of audio.  A maximum
   of 10-30% is possible by changing the pitch-rate of speech.  Music
   goes out of tune if one tries to manipulate the playback rate by
   resampling it.  This is a well known problem and audio is commonly
   muted or played back in short segments with skips to keep up with the
   current playback point.

   For video it is possible to manipulate the frame rate, although the
   rendering capabilities are often limited to certain frame rates.
   Also the allowed bitrates in decoding, the structure used in the
   encoding and the dependency between frames and other capabilities of
   the rendering device limits the possible manipulations.  Therefore,
   the basic fast forward capabilities often are implemented by
   selecting certain subsets of frames.

   Due to the media restrictions, the possible scale values are commonly
   restricted to the set of realizable scale ratios.  To enable the
   clients to select from the possible scale values, RTSP can signal the
   supported Scale ratios for the content.  To support aggregated or
   dynamic content, where this may change during the ongoing session and
   dependent on the location within the content, a mechanism for
   updating the media properties and the scale factor currently in use,
   exists.

   Speed (Section 18.50) affects how much of the playback timeline is
   delivered in a given wallclock period.  The default is Speed = 1
   which means to deliver at the same rate the media is consumed.  Speed
   > 1 means that the receiver will get content faster than it regularly
   would consume it.  Speed < 1 means that delivery is slower than the
   regular media rate.  Speed values of 0 or lower have no meaning and
   are not allowed.  This mechanism enables two general functionalities.
   One is client side scale operations, i.e., the client receives all
   the frames and makes the adjustment to the playback locally.  The
   second is delivery control for buffering of media.  By specifying a
   speed over 1.0 the client can build up the amount of playback time it
   has present in its buffers to a level that is sufficient for its
   needs.




Schulzrinne, et al.      Expires August 14, 2014               [Page 17]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   A naive implementation of Speed would only affect the transmission
   schedule of the media and has a clear impact on the needed bandwidth.
   This would result in the data rate being proportional to the speed
   factor.  Speed = 1.5, i.e., 50% faster than normal delivery, would
   result in a 50% increase in the data transport rate.  If that can be
   supported or not depends solely on the underlying network path.
   Scale may also have some impact on the required bandwidth due to the
   manipulation of the content in the new playback schedule.  An example
   is fast forward where only the independently decodable intra frames
   are included in the media stream.  This usage of solely intra frames
   increases the data rate significantly compared to a normal sequence
   with the same number of frames, where most frames are encoded using
   prediction.

   This potential increase of the data rate needs to be handled by the
   media sender.  The client has requested that the media will be
   delivered in a specific way, which should be honored.  However, the
   media sender cannot ignore if the network path between the sender and
   the receiver can't handle the resulting media stream.  In that case
   the media stream needs to be adapted to fit the available resources
   of the path.  This can result in a reduced media quality.

   The need for bitrate adaptation becomes especially problematic in
   connection with the Speed semantics.  If the goal is to fill up the
   buffer, the client may not want to do that at the cost of reduced
   quality.  If the client wants to make local playout changes then it
   may actually require that the requested speed be honored.  To resolve
   this issue, Speed uses a range so that both cases can be supported.
   The server is requested to use the highest possible speed value
   within the range which is compatible with the available bandwidth.
   As long as the server can maintain a speed value within the range it
   shall not change the media quality, but instead modify the actual
   delivery rate in response to available bandwidth and reflect this in
   the Speed value in the response.  However, if this is not possible,
   the server should instead modify the media quality to respect the
   lowest speed value and the available bandwidth.

   This functionality enables the local scaling implementation to use a
   tight range, or even a range where the lower bound equals the upper
   bound, to identify that it requires the server to deliver the
   requested amount of media time per delivery time independent of how
   much it needs to adapt the media quality to fit within the available
   path bandwidth.  For buffer filling, it is suitable to use a range
   with a reasonable span and with a lower bound at the nominal media
   rate 1.0, such as 1.0 - 2.5.  If the client wants to reduce the
   buffer, it can specify an upper bound that is below 1.0 to force the
   server to deliver slower than the nominal media rate.




Schulzrinne, et al.      Expires August 14, 2014               [Page 18]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


2.6.  Session Maintenance and Termination

   The session context that has been established is kept alive by having
   the client show liveness.  This is done in two main ways:

   o  Media transport protocol keep-alive.  RTP Control Protocol (RTCP)
      may be used when using RTP.

   o  Any RTSP request referencing the session context.

   Section 10.5 discusses the methods for showing liveness in more
   depth.  If the client fails to show liveness for more than the
   established session timeout value (normally 60 seconds), the server
   may terminate the context.  Other values may be selected by the
   server through the inclusion of the timeout parameter in the session
   header.

   The session context is normally terminated by the client sending a
   TEARDOWN request (Section 13.7) to the server referencing the
   aggregated control URI.  An individual media resource can be removed
   from a session context by a TEARDOWN request referencing that
   particular media resource.  If all media resources are removed from a
   session context, the session context is terminated.

   A client may keep the session alive indefinitely if allowed by the
   server; however, a client is recommended to release the session
   context when an extended period of time without media delivery
   activity has passed.  The client can re-establish the session context
   if required later.  What constitutes an extended period of time is
   dependent on the client, server and their usage.  It is recommended
   that the client terminates the session before ten times the session
   timeout value has passed.  A server may terminate the session after
   one session timeout period without any client activity beyond keep-
   alive.  When a server terminates the session context, it does that by
   sending a TEARDOWN request indicating the reason.

   A server can also request that the client tear down the session and
   re-establish it at an alternative server, as may be needed for
   maintenance.  This is done by using the REDIRECT method
   (Section 13.10).  The Terminate-Reason header (Section 18.52) is used
   to indicate when and why.  The Location header indicates where it
   should connect if there is an alternative server available.  When the
   deadline expires, the server simply stops providing the service.  To
   achieve a clean closure, the client needs to initiate session
   termination prior to the deadline.  In case the server has no other
   server to redirect to, and wants to close the session for
   maintenance, it shall use the TEARDOWN method with a Terminate-Reason
   header.



Schulzrinne, et al.      Expires August 14, 2014               [Page 19]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


2.7.  Extending RTSP

   RTSP is quite a versatile protocol which supports extensions in many
   different directions.  Even this core specification contains several
   blocks of functionality that are optional to implement.  The use case
   and need for the protocol deployment should determine what parts are
   implemented.  Allowing for extensions makes it possible for RTSP to
   reach out to additional use cases.  However, extensions will affect
   the interoperability of the protocol and therefore it is important
   that they can be added in a structured way.

   The client can learn the capability of a server by using the OPTIONS
   method (Section 13.1) and the Supported header (Section 18.51).  It
   can also try and possibly fail using new methods, or require that
   particular features are supported using the Require (Section 18.43)
   or Proxy-Require (Section 18.37) header.

   The RTSP protocol in itself can be extended in three ways, listed
   here in increasing order of the magnitude of changes supported:

   o  Existing methods can be extended with new parameters, for example,
      headers, as long as these parameters can be safely ignored by the
      recipient.  If the client needs negative acknowledgment when a
      method extension is not supported, a tag corresponding to the
      extension may be added in the field of the Require or Proxy-
      Require headers.

   o  New methods can be added.  If the recipient of the message does
      not understand the request, it must respond with error code 501
      (Not Implemented) so that the sender can avoid using this method
      again.  A client may also use the OPTIONS method to inquire about
      methods supported by the server.  The server must list the methods
      it supports using the Public response-header.

   o  A new version of the protocol can be defined, allowing almost all
      aspects (except the position of the protocol version number) to
      change.  A new version of the protocol must be registered through
      an IETF standards track document.

   The basic capability discovery mechanism can be used to both discover
   support for a certain feature and to ensure that a feature is
   available when performing a request.  For a detailed explanation of
   this see Section 11.

   New media delivery protocols may be added and negotiated at session
   establishment, in addition to extensions to the core protocol.
   Certain types of protocol manipulations can be done through parameter
   formats using SET_PARAMETER and GET_PARAMETER.



Schulzrinne, et al.      Expires August 14, 2014               [Page 20]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


3.  Document Conventions

3.1.  Notational Conventions

   Since a few of the definitions are identical to HTTP/1.1, this
   specification only points to the section where they are defined
   rather than copying it.  For brevity, [HX.Y] is to be taken to refer
   to Section X.Y of the HTTP/1.1 specification ([RFC2616]).

   All the mechanisms specified in this document are described in both
   prose and the Augmented Backus-Naur form (ABNF) described in detail
   in [RFC5234].

   Indented paragraphs are used to provide informative background and
   motivation.  This is intended to give readers who were not involved
   with the formulation of the specification an understanding of why
   things are the way they are in RTSP.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in
   [RFC2119].

   The word, "unspecified" is used to indicate functionality or features
   that are not defined in this specification.  Such functionality
   cannot be used in a standardized manner without further definition in
   an extension specification to RTSP.

3.2.  Terminology

   Aggregate control:  The concept of controlling multiple streams using
      a single timeline, generally maintained by the server.  A client,
      for example, uses aggregate control when it issues a single play
      or pause message to simultaneously control both the audio and
      video in a movie.  A session which is under aggregate control is
      referred to as an aggregated session.

   Aggregate control URI:  The URI used in an RTSP request to refer to
      and control an aggregated session.  It normally, but not always,
      corresponds to the presentation URI specified in the session
      description.  See Section 13.3 for more information.

   Client:  The client requests media service from the media server.

   Connection:  A transport layer virtual circuit established between
      two programs for the purpose of communication.





Schulzrinne, et al.      Expires August 14, 2014               [Page 21]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Container file:  A file which may contain multiple media streams
      which often constitutes a presentation when played together.  The
      concept of a container file is not embedded in the protocol.
      However, RTSP servers may offer aggregate control on the media
      streams within these files.

   Continuous media:  Data where there is a timing relationship between
      source and sink; that is, the sink needs to reproduce the timing
      relationship that existed at the source.  The most common examples
      of continuous media are audio and motion video.  Continuous media
      can be real-time (interactive or conversational), where there is a
      "tight" timing relationship between source and sink, or streaming
      where the relationship is less strict.

   Feature-tag:  A tag representing a certain set of functionality,
      i.e., a feature.

   IRI:  Internationalized Resource Identifier, is similar to a URI, but
      allows characters from the whole Universal Character Set (Unicode/
      ISO 10646), rather than the US-ASCII only.  See [RFC3987] for more
      information.

   Live:  Normally used to describe a presentation or session with media
      coming from an ongoing event.  This generally results in the
      session having an unbound or only loosely defined duration, and
      sometimes no seek operations are possible.

   Media initialization:  Datatype/codec specific initialization.  This
      includes such things as clock rates, color tables, etc.  Any
      transport-independent information which is required by a client
      for playback of a media stream occurs in the media initialization
      phase of stream setup.

   Media parameter:  Parameter specific to a media type that may be
      changed before or during stream delivery.

   Media server:  The server providing media delivery services for one
      or more media streams.  Different media streams within a
      presentation may originate from different media servers.  A media
      server may reside on the same host or on a different host from
      which the presentation is invoked.

   (Media) stream:  A single media instance, e.g., an audio stream or a
      video stream as well as a single whiteboard or shared application
      group.  When using RTP, a stream consists of all RTP and RTCP
      packets created by a source within an RTP session.





Schulzrinne, et al.      Expires August 14, 2014               [Page 22]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Message:  The basic unit of RTSP communication, consisting of a
      structured sequence of octets matching the syntax defined in
      Section 20 and transmitted over a connection-based transport.  A
      message is either a Request or a Response.

   Message Body:  The information transferred as the payload of a
      message (Request or response).  A message body consists of meta-
      information in the form of message-body headers and content in the
      form of a message-body, as described in Section 9.

   Non-Aggregated Control:  Control of a single media stream.

   Presentation:  A set of one or more streams presented to the client
      as a complete media feed and described by a presentation
      description as defined below.  Presentations with more than one
      media stream are often handled in RTSP under aggregate control.

   Presentation description:  A presentation description contains
      information about one or more media streams within a presentation,
      such as the set of encodings, network addresses and information
      about the content.  Other IETF protocols such as SDP ([RFC4566])
      use the term "session" for a presentation.  The presentation
      description may take several different formats, including but not
      limited to the session description protocol format, SDP.

   Response:  An RTSP response to a Request.  One type of RTSP message.
      If an HTTP response is meant, it is indicated explicitly.

   Request:  An RTSP request.  One type of RTSP message.  If an HTTP
      request is meant, it is indicated explicitly.

   Request-URI:  The URI used in a request to indicate the resource on
      which the request is to be performed.

   RTSP agent:  Refers to either an RTSP client, an RTSP server, or an
      RTSP proxy.  In this specification, there are many capabilities
      that are common to these three entities such as the capability to
      send requests or receive responses.  This term will be used when
      describing functionality that is applicable to all three of these
      entities.

   RTSP session:  A stateful abstraction upon which the main control
      methods of RTSP operate.  An RTSP session is a common context; it
      is created and maintained on client's request and can be destroyed
      by either the client or server.  It is established by an RTSP
      server upon the completion of a successful SETUP request (when a
      200 OK response is sent) and is labeled with a session identifier
      at that time.  The session exists until timed out by the server or



Schulzrinne, et al.      Expires August 14, 2014               [Page 23]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      explicitly removed by a TEARDOWN request.  An RTSP session is a
      stateful entity; an RTSP server maintains an explicit session
      state machine (see Appendix B) where most state transitions are
      triggered by client requests.  The existence of a session implies
      the existence of state about the session's media streams and their
      respective transport mechanisms.  A given session can have one or
      more media streams associated with it.  An RTSP server uses the
      session to aggregate control over multiple media streams.

   Origin Server:  The server on which a given resource resides.

   Transport initialization:  The negotiation of transport information
      (e.g., port numbers, transport protocols) between the client and
      the server.

   URI:  Universal Resource Identifier, see [RFC3986].  The URIs used in
      RTSP are generally URLs as they give a location for the resource.
      As URLs are a subset of URIs, they will be referred to as URIs to
      cover also the cases when an RTSP URI would not be an URL.

   URL:  Universal Resource Locator, is a URI which identifies the
      resource through its primary access mechanism, rather than
      identifying the resource by name or by some other attribute(s) of
      that resource.

4.  Protocol Parameters

4.1.  RTSP Version

   This specification defines version 2.0 of RTSP.

   RTSP uses a "<major>.<minor>" numbering scheme to indicate versions
   of the protocol.  The protocol versioning policy is intended to allow
   the sender to indicate the format of a message and its capacity for
   understanding further RTSP communication, rather than the features
   obtained via that communication.  No change is made to the version
   number for the addition of message components which do not affect
   communication behavior or which only add to extensible field values.

   The <minor> number is incremented when the changes made to the
   protocol add features which do not change the general message parsing
   algorithm, but which may add to the message semantics and imply
   additional capabilities of the sender.  The <major> number is
   incremented when the format of a message within the protocol is
   changed.  The version of an RTSP message is indicated by an RTSP-
   Version field in the first line of the message.  Note that the major
   and minor numbers MUST be treated as separate integers and that each
   MAY be incremented higher than a single digit.  Thus, RTSP/2.4 is a



Schulzrinne, et al.      Expires August 14, 2014               [Page 24]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   lower version than RTSP/2.13, which in turn is lower than RTSP/12.3.
   Leading zeros SHALL NOT be sent and MUST be ignored by recipients.

4.2.  RTSP IRI and URI

   RTSP 2.0 defines and registers or updates three URI schemes "rtsp",
   "rtsps" and "rtspu".  The usage of the last, "rtspu", is unspecified
   in RTSP 2.0, and is defined here to register the URI scheme that was
   defined in RTSP 1.0.  The "rtspu" scheme indicates unspecified
   transport of the RTSP messages over unreliable transport (UDP in RTSP
   1.0).  An RTSP server MUST respond with an error code indicating the
   "rtspu" scheme is not implemented (501) to a request that carries a
   "rtspu" URI scheme.

   The details of the syntax of "rtsp" and "rtsps" URIs has been changed
   from RTSP 1.0.  These changes are:

   o  Support for IPV6 literal in host part and future IP literals
      through RFC 3986 defined mechanism.

   o  A new relative format to use in the RTSP protocol elements that is
      not required to start with "/".

   Neither should have any significant impact on interoperability.  If
   one is required to use IPv6 literals to reach an RTSP server, then
   that RTSP server must be IPv6 capable, and RTSP 1.0 is not a fully
   IPv6 capable protocol.  If an RTSP 1.0 client attempts to process the
   URI it will not match the allowed syntax and be considered invalid
   and processing will be stopped.  This is clearly a failure to reach
   the resource, however it is not a signification issue as RTSP 2.0
   support was needed anyway in both server and client.  Thus failure
   will only occur in a later step when there is a RTSP version mismatch
   between client and server.  The second change will only occur inside
   RTSP message headers, as the request URI must be an absolute URI.
   Thus such usages will only occur after an agent has accepted and
   started processing RTSP 2.0 messages, and an RTSP 1.0 only agent will
   not be required to parse such types of relative URIs.

   This specification also defines the format of the RTSP IRI [RFC3987]
   that can be used as RTSP resource identifiers and locators, in web
   pages, user interfaces, on paper, etc.  However, the RTSP request
   message format only allows usage of the absolute URI format.  The
   RTSP IRI format MUST use the rules and transformation for IRIs to
   URIs, as defined in [RFC3987].  This allows a URI that matches the
   RTSP 2.0 specification, and so is suitable for use in a request, to
   be created from an RTSP IRI.





Schulzrinne, et al.      Expires August 14, 2014               [Page 25]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   The RTSP IRI and URI are both syntax restricted compared to the
   generic syntax defined in [RFC3986] and [RFC3987]:

   o  An absolute URI requires the authority part; i.e., a host identity
      MUST be provided.

   o  Parameters in the path element are prefixed with the reserved
      separator ";".

   The "scheme" and "host" parts of all URIs [RFC3986] and IRIs
   [RFC3987] are case-insensitive.  All other parts of RTSP URIs and
   IRIs are case- sensitive, and MUST NOT be case-mapped.

   The fragment identifier is used as defined in sections 3.5 and 4.3 of
   [RFC3986], i.e., the fragment is to be stripped from the IRI by the
   requester and not included in the request URI.  The user agent needs
   to interpret the value of the fragment based on the media type the
   request relates to; i.e., the media type indicated in Content-Type
   header in the response to DESCRIBE.

   The syntax of any URI query string is unspecified and responder
   (usually the server) specific.  The query is, from the requester's
   perspective, an opaque string and needs to be handled as such.
   Please note that relative URI with queries are difficult to handle
   due to the RFC 3986 relative URI handling rules.  Any change of the
   path element using a relative URI results in the stripping of the
   query, which means the relative part needs to contain the query.

   The URI scheme "rtsp" requires that commands are issued via a
   reliable protocol (within the Internet, TCP), while the scheme
   "rtsps" identifies a reliable transport using secure transport (TLS
   [RFC5246], see (Section 19).

   For the scheme "rtsp", if no port number is provided in the authority
   part of the URI, the port number 554 MUST be used.  For the scheme
   "rtsps", if no port number is provided in the authority part of the
   URI port number, the TCP port 322 MUST be used.

   A presentation or a stream is identified by a textual media
   identifier, using the character set and escape conventions of URIs
   [RFC3986].  URIs may refer to a stream or an aggregate of streams;
   i.e., a presentation.  Accordingly, requests described in
   (Section 13) can apply to either the whole presentation or an
   individual stream within the presentation.  Note that some request
   methods can only be applied to streams, not presentations, and vice
   versa.

   For example, the RTSP URI:



Schulzrinne, et al.      Expires August 14, 2014               [Page 26]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      rtsp://media.example.com:554/twister/audiotrack

   may identify the audio stream within the presentation "twister",
   which can be controlled via RTSP requests issued over a TCP
   connection to port 554 of host media.example.com.

   Also, the RTSP URI:

      rtsp://media.example.com:554/twister

   identifies the presentation "twister", which may be composed of audio
   and video streams, but could also be something else like a random
   media redirector.

      This does not imply a standard way to reference streams in URIs.
      The presentation description defines the hierarchical
      relationships in the presentation and the URIs for the individual
      streams.  A presentation description may name a stream "a.mov" and
      the whole presentation "b.mov".

   The path components of the RTSP URI are opaque to the client and do
   not imply any particular file system structure for the server.

      This decoupling also allows presentation descriptions to be used
      with non-RTSP media control protocols simply by replacing the
      scheme in the URI.

4.3.  Session Identifiers

   Session identifiers are strings of length 8-128 characters.  A
   session identifier MUST be generated cryptographically random (see
   [RFC4086]).  It is RECOMMENDED that it contains 128 bits of entropy,
   i.e., approximately 22 characters from a high quality generator (see
   Section 21).  However, note that the session identifier does not
   provide any security against session hijacking unless it is kept
   confidential by the client, server and trusted proxies.

4.4.  Media Time Formats

   RTSP currently supports three different media time formats defined
   below.  Additional time formats may be specified in the future.
   These time formats can be used with the Range header (Section 18.40)
   to request playback and specify at which media position protocol
   requests actually will or have taken place.  They are also used in
   description of the media's properties using the Media-Range header
   (Section 18.30).  The unqualified format identifier is used on its
   own in Accept-Ranges header (Section 18.5) to declare supported time




Schulzrinne, et al.      Expires August 14, 2014               [Page 27]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   formats and also in the Range header (Section 18.40) to request the
   time format used in the response.

4.4.1.  SMPTE Relative Timestamps

   A Society of Motion Picture and Television Engineers (SMPTE) relative
   timestamp expresses time relative to the start of the clip.  Relative
   timestamps are expressed as SMPTE time codes [SMPTE_TC] for frame-
   level access accuracy.  The time code has the format

      hours:minutes:seconds:frames.subframes,

   with the origin at the start of the clip.  The default SMPTE format
   is "SMPTE 30 drop" format, with frame rate is 29.97 frames per
   second.  Other SMPTE codes MAY be supported (such as "SMPTE 25")
   through the use of "smpte-type".  For SMPTE 30, the "frames" field in
   the time value can assume the values 0 through 29.  The difference
   between 30 and 29.97 frames per second is handled by dropping the
   first two frame indices (values 00 and 01) of every minute, except
   every tenth minute.  If the frame and the subframe values are zero,
   they may be omitted.  Subframes are measured in one-hundredth of a
   frame.

   Examples:

     smpte=10:12:33:20-
     smpte=10:07:33-
     smpte=10:07:00-10:07:33:05.01
     smpte-25=10:07:00-10:07:33:05.01

4.4.2.  Normal Play Time

   Normal play time (NPT) indicates the stream absolute position
   relative to the beginning of the presentation.  The timestamp
   consists of two parts: the mandatory first part may be expressed in
   either seconds or hours, minutes, and seconds.  The optional second
   part consists of a decimal point and decimal figures and indicates
   fractions of a second.

   The beginning of a presentation corresponds to 0.0 seconds.  Negative
   values are not defined.

   The special constant "now" is defined as the current instant of a
   live event.  It MAY only be used for live events, and MUST NOT be
   used for on-demand (i.e., non-live) content.

   NPT is defined as in DSM-CC [ISO.13818-6.1995]: "Intuitively, NPT is
   the clock the viewer associates with a program.  It is often



Schulzrinne, et al.      Expires August 14, 2014               [Page 28]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   digitally displayed on a VCR.  NPT advances normally when in normal
   play mode (scale = 1), advances at a faster rate when in fast scan
   forward (high positive scale ratio), decrements when in scan reverse
   (negative scale ratio) and is fixed in pause mode.  NPT is
   (logically) equivalent to SMPTE time codes."

   Examples:

     npt=123.45-125
     npt=12:05:35.3-
     npt=now-

   The syntax is based on ISO 8601 [ISO.8601.2000] and expresses the
   time elapsed since presentation start, with two different notations
   allowed:

   o  The npt-hhmmss notation uses an ISO 8601 extended complete
      representation of the time of the day format (Section 5.3.1.1 of
      [ISO.8601.2000] ) using colon (":") as separators between hours,
      minutes and seconds (hh:mm:ss).  The hour counter is not limited
      to 0-24 hours; up to nineteen (19) digits of hours are allowed.

   o  In accordance with the requirements of the ISO 8601 time format,
      the hours, minutes, and seconds MUST all be present, with two
      digits used for minutes and for seconds, and with a least two
      digits for hours.  An NPT of 7 minutes and 0 seconds is
      represented as "00:07:00", and an NPT of 392 hours, 0 minutes, and
      6 seconds is represented as "392:00:06".

   o  RTSP 1.0 allowed NPT in the npt-hhmmss notation without any
      leading zeros, to ensure that implementations doesn't fail if any
      implementation follows the RTSP 1.0 format, all implementations
      are REQUIRED to support receiving NPT values, hours, minutes or
      seconds, without leading zeros.

   o  The npt-sec notation expresses the time in seconds, using between
      one and nineteen (19) digits.

   Both notations allow decimal fractions of seconds as specified in
   Section 5.3.1.3 of [ISO.8601.2000], using at most 9 digits, and
   allowing only "." (full stop) as the decimal separator.

   The npt-sec notation is optimized for automatic generation, the npt-
   hhmmss notation for consumption by human readers.  The "now" constant
   allows clients to request to receive the live feed rather than the
   stored or time-delayed version.  This is needed since neither
   absolute time nor zero time are appropriate for this case.




Schulzrinne, et al.      Expires August 14, 2014               [Page 29]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


4.4.3.  Absolute Time

   Absolute time is expressed following a specific types of ISO 8601
   [ISO.8601.2000] based timestamps.  The date is complete
   representation calendar date in basic format (YYYYMMDD) without
   separators (per Section 5.2.1.1 of [ISO.8601.2000]).  The time of day
   is provided in the complete representation basic format (hhmmss) as
   specified in Section 5.3.1.1 of [ISO.8601.2000], allowing decimal
   fractions of seconds following Section 5.3.1.3 requiring "." (full
   stop) as decimal separator and limiting the number of digits to no
   more than nine (9).  The time expressed MUST be using UTC (GMT), i.e.
   no timezone offsets allowed.  The full date and time specification is
   the eight digit date followed by a "T" followed by the six digits
   time value, optionally followed by a full stop followed by one to
   nine fractions of a second and ended by "Z", e.g.
   YYYYMMDDThhmmss.ssZ.

      The reason for this time format rather than using "Date and Time
      on the Internet: Timestamps" [RFC3339] are historic and using the
      format specified in RTSP 1.0.  The motivations raised in RFC 3339
      applies to why a selection from ISO 8601 was done, but a different
      and even more restrictive selection was applied in this case.

   Example for clock format range request for a starting time of
   November 8, 1996 at 14h 37 min and 20 and a quarter seconds UTC
   playing for 10 min and 5 seconds, a Media-Properties header's "Time-
   Limited UTC property for 24th of December 2014 at 15 hours and 00
   mins, and a Terminate-Readon headers "time" property for 18th of June
   2013 at 16 hours, 12 minutes and 56 seconds:

     clock=19961108T143720.25Z-19961108T144725.25Z
     Time-Limited=20141224T1500Z
     time=20130618T161256Z

4.5.  Feature-Tags

   Feature-tags are unique identifiers used to designate features in
   RTSP.  These tags are used in Require (Section 18.43), Proxy-Require
   (Section 18.37), Proxy-Supported (Section 18.38), Supported
   (Section 18.51) and Unsupported (Section 18.55) header fields.

   A feature-tag definition MUST indicate which combination of clients,
   servers or proxies it applies to.

   The creator of a new RTSP feature-tag should either prefix the
   feature-tag with a reverse domain name (e.g.,
   "com.example.mynewfeature" is an apt name for a feature whose
   inventor can be reached at "example.com"), or register the new



Schulzrinne, et al.      Expires August 14, 2014               [Page 30]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   feature-tag with the Internet Assigned Numbers Authority (IANA) (see
   IANA Section 22).

   The usage of feature-tags is further described in Section 11 that
   deals with capability handling.

4.6.  Message Body Tags

   Message body tags are opaque strings that are used to compare two
   message bodies from the same resource, for example in caches or to
   optimize setup after a redirect.  Message body tags can be carried in
   the MTag header (see Section 18.31) or in SDP (see Appendix D.1.9).
   MTag is similar to ETag in HTTP/1.1 (see Section 3.11 of [RFC2068]).

   A message body tag MUST be unique across all versions of all message
   bodies associated with a particular resource.  A given message body
   tag value MAY be used for message bodies obtained by requests on
   different URIs.  The use of the same message body tag value in
   conjunction with message bodies obtained by requests on different
   URIs does not imply the equivalence of those message bodies

   Message body tags are used in RTSP to make some methods conditional.
   The methods are made conditional through the inclusion of headers;
   see "If-Match" (Section 18.24) and "If-None-Match" (Section 18.26).
   Note that RTSP message body tags apply to the complete presentation;
   i.e., both the presentation description and the individual media
   streams.  Thus message body tags can be used to verify at setup time
   after a redirect that the same session description applies to the
   media at the new location using the If-Match header.

4.7.  Media Properties

   When an RTSP server handles media, it is important to consider the
   different properties a media instance for delivery and playback can
   have.  This specification considers the media properties listed below
   in its protocol operations.  They are derived from the differences
   between a number of supported usages.

   On-demand:  Media that has a fixed (given) duration that doesn't
      change during the life time of the RTSP session and is known at
      the time of the creation of the session.  It is expected that the
      content of the media will not change, even if the representation,
      i.e., encoding, quality, etc, may change.  Generally one can seek,
      i.e., request any range, within the media.

   Dynamic On-demand:  This is a variation of the on-demand case where
      external methods are used to manipulate the actual content of the




Schulzrinne, et al.      Expires August 14, 2014               [Page 31]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      media setup for the RTSP session.  The main example is a content
      defined by a playlist.

   Live:  Live media represents a progressing content stream (such as
      broadcast TV) where the duration may or may not be known.  It is
      not seekable, only the content presently being delivered can be
      accessed.

   Live with Recording:  A Live stream that is combined with a server-
      side capability to store and retain the content of the live
      session, and allow for random access delivery within the part of
      the already recorded content.  The actual behavior of the media
      stream is very much dependent on the retention policy for the
      media stream; either the server will be able to capture the
      complete media stream, or it will have a limitation in how much
      will be retained.  The media range will dynamically change as the
      session progress.  For servers with a limited amount of storage
      available for recording, there will typically be a sliding window
      that moves forward while new data is made available and older data
      is discarded.

   To cover the above usages, the following media properties with
   appropriate values are specified:

4.7.1.  Random Access and Seeking

   Random Access is the ability to specify and get media delivered
   starting from any time instant within the content, an operation
   called seeking.  The Media-Properties header will indicate the
   general capability for a media resource to perform random access:

   Random-Access:  The media is seekable to any out of a large number of
      points within the media.  Due to media encoding limitations, a
      particular point may not be reachable, but seeking to a point
      close by is enabled.  A floating point number of seconds may be
      provided to express the worst case distance between random access
      points.

   Beginning-Only:  Seeking is only possible to the beginning of the
      content.

   No-seeking:  Seeking is not possible at all.

   If random access is possible, as indicated by the Media-Properties
   header, the actual behavior policy when seeking can be controlled
   using the Seek-Style header (Section 18.47).





Schulzrinne, et al.      Expires August 14, 2014               [Page 32]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


4.7.2.  Retention

   Media may have different retention policies in place that affect the
   operation on media.  The following different media retention policies
   are defined:

   Unlimited:  The media will not be removed as long as the RTSP session
      is in existence.

   Time-Limited:  The media will not be removed before the given
      wallclock time.  After that time it may or may not be available
      any more.

   Time-Duration:  The media (on fragment or unit basis) will be
      retained for the specified duration.

4.7.3.  Content Modifications

   There is also the question of how the content may change over time
   for a given media resource:

   Immutable:  The content of the media will not change, even if the
      representation, i.e., encoding, quality, etc., may change.

   Dynamic:  The content can change due to external methods or triggers,
      such as playlists, but this will be announced by explicit updates.

   Time-Progressing:  As time progresses new content will become
      available.  If the content also is retained it will become longer
      as everything between the start point and the point currently
      being made available can be accessed.  If the media server uses a
      sliding window policy for retention, the start point will also
      change as time progresses.

4.7.4.  Supported Scale Factors

   Content often supports only a limited set or range of scales when
   delivering the media.  To enable the client to know what values or
   ranges of scale operations that the whole content or the current
   position supports, a media properties attribute for this is defined
   which contains a list with the values and/or ranges that are
   supported.  The attribute is named "Scales".  The "Scales" attribute
   may be updated at any point in the content due to content consisting
   of spliced pieces or content being dynamically updated by out-of-band
   mechanisms.






Schulzrinne, et al.      Expires August 14, 2014               [Page 33]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


4.7.5.  Mapping to the Attributes

   This section shows examples of how one would map the above usages to
   the properties and their values.

   Example of On-demand:
      Random Access: Random-Access=5.0, Content Modifications:
      Immutable, Retention: Unlimited or Time-Limited.

   Example of Dynamic On-demand:
      Random Access: Random-Access=3.0, Content Modifications: Dynamic,
      Retention: Unlimited or Time-Limited.

   Example of Live:
      Random Access: No-seeking, Content Modifications: Time-
      Progressing, Retention: Time-Duration=0.0

   Example of Live with Recording:
      Random Access: Random-Access=3.0, Content Modifications: Time-
      Progressing, Retention: Time-Duration=7200.0

5.  RTSP Message

   RTSP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding RFC 3629 [RFC3629].  Lines MUST be terminated by CRLF.

      Text-based protocols make it easier to add optional parameters in
      a self-describing manner.  Since the number of parameters and the
      frequency of commands is low, processing efficiency is not a
      concern.  Text-based protocols, if done carefully, also allow easy
      implementation of research prototypes in scripting languages such
      as TCL, Visual Basic and Perl.

   The ISO 10646 character set avoids character set switching, but is
   invisible to the application as long as US-ASCII is being used.  This
   is also the encoding used for RTCP [RFC3550].

   A request contains a method, the object the method is operating upon,
   and parameters to further describe the method.  Methods are
   idempotent unless otherwise noted.  Methods are also designed to
   require little or no state maintenance at the media server.

5.1.  Message Types

   RTSP messages are either requests from client to server, or server to
   client, and responses in the reverse direction.  Request (Section 7)
   and Response (Section 8) messages use a format based on the generic
   message format of RFC 5322 [RFC5322] for transferring bodies (the



Schulzrinne, et al.      Expires August 14, 2014               [Page 34]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   payload of the message).  Both types of messages consist of a start-
   line, zero or more header fields (also known as "headers"), an empty
   line (i.e., a line with nothing preceding the CRLF) indicating the
   end of the headers, and possibly the data of the message body.  The
   below ABNF [RFC5234] is for information and the formal message
   specification is present in Section 20.2.2.

   generic-message = start-line
                   *(rtsp-header CRLF)
                     CRLF
                   [ message-body-data ]
   start-line = Request-Line | Status-Line

   In the interest of robustness, agents MUST ignore any empty line(s)
   received where a Request-Line or Status-Line is expected.  In other
   words, if the agent is reading the protocol stream at the beginning
   of a message and receives any number of CRLFs first, it MUST ignore
   any of the CRLFs.

5.2.  Message Headers

   RTSP header fields (see Section 18) include general-header, request-
   header, response-header, and message-body header fields.

   The order in which header fields with differing field names are
   received is not significant.  However, it is "good practice" to send
   general-header fields first, followed by request-header or response-
   header fields, and ending with the Message-body header fields.

   Multiple header fields with the same field-name MAY be present in a
   message if and only if the entire field-value for that header field
   is defined as a comma-separated list.  It MUST be possible to combine
   the multiple header fields into one "field-name: field-value" pair,
   without changing the semantics of the message, by appending each
   subsequent field-value to the first, each separated by a comma.  The
   order in which header fields with the same field-name are received is
   therefore significant to the interpretation of the combined field
   value, and thus a proxy MUST NOT change the order of these field
   values when a message is forwarded.

   Unknown message headers MUST be ignored (skipping over the header to
   the next protocol element, and not causing an error) by a RTSP server
   or client.  An RTSP Proxy MUST forward unknown message headers.
   Message headers defined outside of this specification that are
   required to be interpreted by the RTSP agent will need to use feature
   tags (Section 4.5) and include them in the appropriate Require
   (Section 18.43) or Proxy-Require (Section 18.37) header.




Schulzrinne, et al.      Expires August 14, 2014               [Page 35]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


5.3.  Message Body

   The message body (if any) of an RTSP message is used to carry further
   information for a particular resource associated with the request or
   response.  An example of a message body is a Session Description
   Protocol (SDP) message.

   The presence of a message body in either a request or a response MUST
   be signaled by the inclusion of a Content-Length header (see
   Section 18.17) and Content-Type (see Section 18.19).  A message body
   MUST NOT be included in a request or response if the specification of
   the particular method (see Method Definitions (Section 13)) does not
   allow sending a message body.  In case a message body is received in
   a message when not expected the message body data SHOULD be
   discarded.  This is to allow future extensions to define optional use
   of a message body.

5.4.  Message Length

   An RTSP Message that does not contain any message body is terminated
   by the first empty line after the header fields (Note: An empty line
   is a line with nothing preceding the CRLF.).  In RTSP messages that
   contain message bodies the empty line is followed by the message
   body.  The length of that body is determined by the value of the
   Content-Length header (Section 18.17).  The value in the header
   represents the length of the message-body in octets.  If this header
   field is not present, a value of zero is assumed, i.e., no message
   body present in the message.  Unlike an HTTP message, an RTSP message
   MUST contain a Content-Length header whenever it contains a message
   body.  Note that RTSP does not support the HTTP/1.1 "chunked"
   transfer coding (see [H3.6.1]).

      Given the moderate length of presentation descriptions returned,
      the server should always be able to determine its length, even if
      it is generated dynamically, making the chunked transfer encoding
      unnecessary.

6.  General Header Fields

   General headers are headers that may be used in both requests and
   responses.  The general-headers are listed in Table 1:










Schulzrinne, et al.      Expires August 14, 2014               [Page 36]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


                +--------------------+--------------------+
                | Header Name        | Defined in Section |
                +--------------------+--------------------+
                | Accept-Ranges      | Section 18.5       |
                |                    |                    |
                | Cache-Control      | Section 18.11      |
                |                    |                    |
                | Connection         | Section 18.12      |
                |                    |                    |
                | CSeq               | Section 18.20      |
                |                    |                    |
                | Date               | Section 18.21      |
                |                    |                    |
                | Media-Properties   | Section 18.29      |
                |                    |                    |
                | Media-Range        | Section 18.30      |
                |                    |                    |
                | Pipelined-Requests | Section 18.33      |
                |                    |                    |
                | Proxy-Supported    | Section 18.38      |
                |                    |                    |
                | Range              | Section 18.40      |
                |                    |                    |
                | RTP-Info           | Section 18.45      |
                |                    |                    |
                | Scale              | Section 18.46      |
                |                    |                    |
                | Seek-Style         | Section 18.47      |
                |                    |                    |
                | Server             | Section 18.48      |
                |                    |                    |
                | Session            | Section 18.49      |
                |                    |                    |
                | Speed              | Section 18.50      |
                |                    |                    |
                | Supported          | Section 18.51      |
                |                    |                    |
                | Timestamp          | Section 18.53      |
                |                    |                    |
                | Transport          | Section 18.54      |
                |                    |                    |
                | User-Agent         | Section 18.56      |
                |                    |                    |
                | Via                | Section 18.57      |
                +--------------------+--------------------+

                 Table 1: The general headers used in RTSP




Schulzrinne, et al.      Expires August 14, 2014               [Page 37]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


7.  Request

   A request message uses the format outlined below regardless of the
   direction of a request, client to server or server to client:

   o  Request line, containing the method to be applied to the resource,
      the identifier of the resource, and the protocol version in use;

   o  Zero or more Header lines, that can be of the following types:
      general-headers (Section 6), request-headers (Section 7.2), or
      message body headers (Section 9.1);

   o  One empty line (CRLF) to indicate the end of the header section;

   o  Optionally a message-body, consisting of one or more lines.  The
      length of the message body in octets is indicated by the Content-
      Length message header.

7.1.  Request Line

   The request line provides the key information about the request: what
   method, on what resources and using which RTSP version.  The methods
   that are defined by this specification are listed in Table 2.




























Schulzrinne, et al.      Expires August 14, 2014               [Page 38]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


                  +---------------+--------------------+
                  | Method        | Defined in Section |
                  +---------------+--------------------+
                  | DESCRIBE      | Section 13.2       |
                  |               |                    |
                  | GET_PARAMETER | Section 13.8       |
                  |               |                    |
                  | OPTIONS       | Section 13.1       |
                  |               |                    |
                  | PAUSE         | Section 13.6       |
                  |               |                    |
                  | PLAY          | Section 13.4       |
                  |               |                    |
                  | PLAY_NOTIFY   | Section 13.5       |
                  |               |                    |
                  | REDIRECT      | Section 13.10      |
                  |               |                    |
                  | SETUP         | Section 13.3       |
                  |               |                    |
                  | SET_PARAMETER | Section 13.9       |
                  |               |                    |
                  | TEARDOWN      | Section 13.7       |
                  +---------------+--------------------+

                         Table 2: The RTSP Methods

   The syntax of the RTSP request line is the following:

      <Method> SP <Request-URI> SP <RTSP-Version> CRLF

   Note: This syntax cannot be freely changed in future versions of
   RTSP.  This line needs to remain parsable by older RTSP
   implementations since it indicates the RTSP version of the message.

   In contrast to HTTP/1.1 [RFC2616], RTSP requests identify the
   resource through an absolute RTSP URI (including scheme, host, and
   port) (see Section 4.2) rather than just the absolute path.

      HTTP/1.1 requires servers to understand the absolute URI, but
      clients are supposed to use the Host request-header.  This is
      purely needed for backward-compatibility with HTTP/1.0 servers, a
      consideration that does not apply to RTSP.

   An asterisk "*" can be used instead of an absolute URI in the
   Request-URI part to indicate that the request does not apply to a
   particular resource, but to the server or proxy itself, and is only
   allowed when the request method does not necessarily apply to a
   resource.



Schulzrinne, et al.      Expires August 14, 2014               [Page 39]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   For example:

      OPTIONS * RTSP/2.0

   An OPTIONS in this form will determine the capabilities of the server
   or the proxy that first receives the request.  If the capability of
   the specific server needs to be determined, without regard to the
   capability of an intervening proxy, the server should be addressed
   explicitly with an absolute URI that contains the server's address.

   For example:

      OPTIONS rtsp://example.com RTSP/2.0

7.2.  Request Header Fields

   The RTSP headers in Table 3 can be included in a request, as request-
   headers, to modify the specifics of the request.

































Schulzrinne, et al.      Expires August 14, 2014               [Page 40]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


               +---------------------+--------------------+
               | Header              | Defined in Section |
               +---------------------+--------------------+
               | Accept              | Section 18.1       |
               |                     |                    |
               | Accept-Credentials  | Section 18.2       |
               |                     |                    |
               | Accept-Encoding     | Section 18.3       |
               |                     |                    |
               | Accept-Language     | Section 18.4       |
               |                     |                    |
               | Authorization       | Section 18.8       |
               |                     |                    |
               | Bandwidth           | Section 18.9       |
               |                     |                    |
               | Blocksize           | Section 18.10      |
               |                     |                    |
               | From                | Section 18.23      |
               |                     |                    |
               | If-Match            | Section 18.24      |
               |                     |                    |
               | If-Modified-Since   | Section 18.25      |
               |                     |                    |
               | If-None-Match       | Section 18.26      |
               |                     |                    |
               | Notify-Reason       | Section 18.32      |
               |                     |                    |
               | Proxy-Authorization | Section 18.36      |
               |                     |                    |
               | Proxy-Require       | Section 18.37      |
               |                     |                    |
               | Referrer            | Section 18.41      |
               |                     |                    |
               | Request-Status      | Section 18.42      |
               |                     |                    |
               | Require             | Section 18.43      |
               |                     |                    |
               | Terminate-Reason    | Section 18.52      |
               +---------------------+--------------------+

                     Table 3: The RTSP request headers

   Detailed header definitions are provided in Section 18.

   New request-headers may be defined.  If the receiver of the request
   is required to understand the request-header, the request MUST
   include a corresponding feature tag in a Require or Proxy-Require
   header to ensure the processing of the header.



Schulzrinne, et al.      Expires August 14, 2014               [Page 41]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


8.  Response

   After receiving and interpreting a request message, the recipient
   responds with an RTSP response message.  Normally, there is only one,
   final, response.  Only responses using the response code class 1xx,
   are allowed to send one or more 1xx response messages prior to the
   final response message.

   The valid response codes and the methods they can be used with are
   listed in Table 4.

8.1.  Status-Line

   The first line of a Response message is the Status-Line, consisting
   of the protocol version followed by a numeric status code and the
   textual phrase associated with the status code, with each element
   separated by SP characters.  No CR or LF is allowed except in the
   final CRLF sequence.

   <RTSP-Version> SP <Status-Code> SP <Reason-Phrase> CRLF

8.1.1.  Status Code and Reason Phrase

   The Status-Code element is a 3-digit integer result code of the
   attempt to understand and satisfy the request.  These codes are fully
   defined in Section 17.  The Reason-Phrase is intended to give a short
   textual description of the Status-Code.  The Status-Code is intended
   for use by automata and the Reason-Phrase is intended for the human
   user.  The client is not required to examine or display the Reason-
   Phrase.

   The first digit of the Status-Code defines the class of response.
   The last two digits do not have any categorization role.  There are 5
   values for the first digit:

   1xx:  Informational - Request received, continuing process

   2xx:  Success - The action was successfully received, understood, and
         accepted

   3rr:  Redirection - Further action needs to be taken in order to
         complete the request (3rr rather than 3xx is used as 304 is
         excluded, see Section 17.3)

   4xx:  Client Error - The request contains bad syntax or cannot be
         fulfilled





Schulzrinne, et al.      Expires August 14, 2014               [Page 42]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   5xx:  Server Error - The server failed to fulfill an apparently valid
         request

   The individual values of the numeric status codes defined for RTSP/
   2.0, and an example set of corresponding Reason-Phrases, are
   presented in Table 4.  The reason phrases listed here are only
   recommended; they may be replaced by local equivalents without
   affecting the protocol.  Note that RTSP adopts most HTTP/1.1
   [RFC2616] status codes and adds RTSP-specific status codes starting
   at x50 to avoid conflicts with future HTTP status codes that are
   desirable to import into RTSP.  All these codes are RTSP specific and
   RTSP has its own registry separate from HTTP for status codes.

   RTSP status codes are extensible.  RTSP applications are not required
   to understand the meaning of all registered status codes, though such
   understanding is obviously desirable.  However, applications MUST
   understand the class of any status code, as indicated by the first
   digit, and treat any unrecognized response as being equivalent to the
   x00 status code of that class, with an exception for unknown 3xx
   codes, which MUST be treated as a 302 (Found).  The reason being that
   no 300 (Multiple Choices in HTTP) is defined for RTSP.  An response
   with an unrecognized status code MUST NOT be cached.  For example, if
   an unrecognized status code of 431 is received by the client, it can
   safely assume that there was something wrong with its request and
   treat the response as if it had received a 400 status code.  In such
   cases, user agents SHOULD present to the user the message body
   returned with the response, since that message body is likely to
   include human-readable information which will explain the unusual
   status.

   +------+---------------------------------+--------------------------+
   | Code | Reason                          | Method                   |
   +------+---------------------------------+--------------------------+
   | 100  | Continue                        | all                      |
   |      |                                 |                          |
   |      |                                 |                          |
   |      |                                 |                          |
   | 200  | OK                              | all                      |
   |      |                                 |                          |
   |      |                                 |                          |
   |      |                                 |                          |
   | 301  | Moved Permanently               | all                      |
   |      |                                 |                          |
   | 302  | Found                           | all                      |
   |      |                                 |                          |
   | 303  | reserved                        | n/a                      |
   |      |                                 |                          |
   | 304  | Not Modified                    | all                      |



Schulzrinne, et al.      Expires August 14, 2014               [Page 43]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   |      |                                 |                          |
   | 305  | Use Proxy                       | all                      |
   |      |                                 |                          |
   |      |                                 |                          |
   |      |                                 |                          |
   | 400  | Bad Request                     | all                      |
   |      |                                 |                          |
   | 401  | Unauthorized                    | all                      |
   |      |                                 |                          |
   | 402  | Payment Required                | all                      |
   |      |                                 |                          |
   | 403  | Forbidden                       | all                      |
   |      |                                 |                          |
   | 404  | Not Found                       | all                      |
   |      |                                 |                          |
   | 405  | Method Not Allowed              | all                      |
   |      |                                 |                          |
   | 406  | Not Acceptable                  | all                      |
   |      |                                 |                          |
   | 407  | Proxy Authentication Required   | all                      |
   |      |                                 |                          |
   | 408  | Request Timeout                 | all                      |
   |      |                                 |                          |
   | 410  | Gone                            | all                      |
   |      |                                 |                          |
   | 412  | Precondition Failed             | DESCRIBE, SETUP          |
   |      |                                 |                          |
   | 413  | Request Message Body Too Large  | all                      |
   |      |                                 |                          |
   | 414  | Request-URI Too Long            | all                      |
   |      |                                 |                          |
   | 415  | Unsupported Media Type          | all                      |
   |      |                                 |                          |
   | 451  | Parameter Not Understood        | SET_PARAMETER,           |
   |      |                                 | GET_PARAMETER            |
   |      |                                 |                          |
   | 452  | reserved                        | n/a                      |
   |      |                                 |                          |
   | 453  | Not Enough Bandwidth            | SETUP                    |
   |      |                                 |                          |
   | 454  | Session Not Found               | all                      |
   |      |                                 |                          |
   | 455  | Method Not Valid In This State  | all                      |
   |      |                                 |                          |
   | 456  | Header Field Not Valid          | all                      |
   |      |                                 |                          |
   | 457  | Invalid Range                   | PLAY, PAUSE              |
   |      |                                 |                          |



Schulzrinne, et al.      Expires August 14, 2014               [Page 44]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   | 458  | Parameter Is Read-Only          | SET_PARAMETER            |
   |      |                                 |                          |
   | 459  | Aggregate Operation Not Allowed | all                      |
   |      |                                 |                          |
   | 460  | Only Aggregate Operation        | all                      |
   |      | Allowed                         |                          |
   |      |                                 |                          |
   | 461  | Unsupported Transport           | all                      |
   |      |                                 |                          |
   | 462  | Destination Unreachable         | all                      |
   |      |                                 |                          |
   | 463  | Destination Prohibited          | SETUP                    |
   |      |                                 |                          |
   | 464  | Data Transport Not Ready Yet    | PLAY                     |
   |      |                                 |                          |
   | 465  | Notification Reason Unknown     | PLAY_NOTIFY              |
   |      |                                 |                          |
   | 466  | Key Management Error            | all                      |
   |      |                                 |                          |
   | 470  | Connection Authorization        | all                      |
   |      | Required                        |                          |
   |      |                                 |                          |
   | 471  | Connection Credentials not      | all                      |
   |      | accepted                        |                          |
   |      |                                 |                          |
   | 472  | Failure to establish secure     | all                      |
   |      | connection                      |                          |
   |      |                                 |                          |
   |      |                                 |                          |
   |      |                                 |                          |
   | 500  | Internal Server Error           | all                      |
   |      |                                 |                          |
   | 501  | Not Implemented                 | all                      |
   |      |                                 |                          |
   | 502  | Bad Gateway                     | all                      |
   |      |                                 |                          |
   | 503  | Service Unavailable             | all                      |
   |      |                                 |                          |
   | 504  | Gateway Timeout                 | all                      |
   |      |                                 |                          |
   | 505  | RTSP Version Not Supported      | all                      |
   |      |                                 |                          |
   | 551  | Option Not Supported            | all                      |
   |      |                                 |                          |
   | 553  | Proxy Unavailable               | all                      |
   +------+---------------------------------+--------------------------+

          Table 4: Status codes and their usage with RTSP methods



Schulzrinne, et al.      Expires August 14, 2014               [Page 45]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


8.2.  Response Headers

   The response-headers allow the request recipient to pass additional
   information about the response which cannot be placed in the Status-
   Line.  This header gives information about the server and about
   further access to the resource identified by the Request-URI.  All
   headers currently classified as response-headers are listed in
   Table 5.

              +------------------------+--------------------+
              | Header                 | Defined in Section |
              +------------------------+--------------------+
              | Authentication-Info    | Section 18.7       |
              |                        |                    |
              | Connection-Credentials | Section 18.13      |
              |                        |                    |
              | Location               | Section 18.28      |
              |                        |                    |
              | MTag                   | Section 18.31      |
              |                        |                    |
              | Proxy-Authenticate     | Section 18.34      |
              |                        |                    |
              | Public                 | Section 18.39      |
              |                        |                    |
              | Retry-After            | Section 18.44      |
              |                        |                    |
              | Unsupported            | Section 18.55      |
              |                        |                    |
              | WWW-Authenticate       | Section 18.58      |
              +------------------------+--------------------+

                    Table 5: The RTSP response headers

   Response-header names can be extended reliably only in combination
   with a change in the protocol version.  However, the usage of
   feature-tags in the request allows the responding party to learn the
   capability of the receiver of the response.  A new or experimental
   header can be given the semantics of response-header if all parties
   in the communication recognize them to be a response-header.
   Unrecognized headers in responses MUST be ignored.

9.  Message Body

   Some Request and Response messages include a message body, if not
   otherwise restricted by the request method or response status code.
   The message body consists of the content data itself (see also
   Section 5.3).




Schulzrinne, et al.      Expires August 14, 2014               [Page 46]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   The SET_PARAMETER and GET_PARAMETER requests and responses, and the
   DESCRIBE response as defined by this specification can have a message
   body; the purpose of the message body is defined in each case.  All
   4xx and 5xx responses MAY also have a message body to carry
   additional response information.  Generally a message body MAY be
   attached to any RTSP 2.0 request or response, but the content of the
   message body MAY be ignored by the receiver.  Extensions to this
   specification can specify the purpose and content of message bodies,
   including requiring their inclusion.

   In this section, both sender and recipient refer to either the client
   or the server, depending on who sends and who receives the message
   body.

9.1.  Message-Body Header Fields

   Message-body header fields define meta-information about the content
   data in the message body.  The message-body header fields are listed
   in Table 6.

                 +------------------+--------------------+
                 | Header           | Defined in Section |
                 +------------------+--------------------+
                 | Allow            | Section 18.6       |
                 |                  |                    |
                 | Content-Base     | Section 18.14      |
                 |                  |                    |
                 | Content-Encoding | Section 18.15      |
                 |                  |                    |
                 | Content-Language | Section 18.16      |
                 |                  |                    |
                 | Content-Length   | Section 18.17      |
                 |                  |                    |
                 | Content-Location | Section 18.18      |
                 |                  |                    |
                 | Content-Type     | Section 18.19      |
                 |                  |                    |
                 | Expires          | Section 18.22      |
                 |                  |                    |
                 | Last-Modified    | Section 18.27      |
                 +------------------+--------------------+

                  Table 6: The RTSP message-body headers

   The extension-header mechanism allows additional message-body header
   fields to be defined without changing the protocol, but these fields
   cannot be assumed to be recognizable by the recipient.  Unrecognized




Schulzrinne, et al.      Expires August 14, 2014               [Page 47]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   header fields MUST be ignored by the recipient and forwarded by
   proxies.

9.2.  Message Body

   An RTSP message with a message body MUST include the Content-Type and
   Content-Length headers.  When a message body is included with a
   message, the data type of that content data is determined via the
   header fields Content-Type and Content-Encoding.

   Content-Type specifies the media type of the underlying data.  There
   is no default media format and the actual format used in the body is
   required to be explicitly stated in the Content-Type header.  By
   being explicit and always require inclusion of the Content-Type
   header with accurate information one avoids the many pitfalls in
   heuristic based interpretation of the body content.  These are issue
   HTTP and email have suffered from.

   Content-Encoding may be used to indicate any additional content
   codings applied to the data, usually for the purpose of data
   compression, that are a property of the requested resource.  The
   default encoding is 'identity', i.e. no transformation of the message
   body.

   The Content-Length of a message is the length of the content,
   measured in octets.

9.3.  Message Body Format Negotiation

   The content format of the message body is provided using the Content-
   Type header (Section 18.19).  To enable the responder of a request to
   determine which media type it should use, the requestor may include
   the Accept header (Section 18.1) in a request to identify supported
   media types or media type ranges suitable to the response.  In case
   the responder is not supporting any of the specified formats, then
   the request response will be a 406 (Not Acceptable) error code.

   The media types that may be used on requests with message bodies need
   to be determined through the use of feature-tags, specification
   requirement or trial and error.  Trial and error works because when
   the responder does not support the media type of the message body it
   will respond with a 415 (Unsupported Media Type).

   The formats supported and their negotiation is done individually on a
   per method and direction (request or response body) direction.
   Requirements on supporting particular media types for use as message
   bodies in requests and response SHALL also be specified on per method
   and direction basis.



Schulzrinne, et al.      Expires August 14, 2014               [Page 48]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


10.  Connections

   RTSP Messages are transferred between RTSP agents and proxies using a
   transport connection.  This transport connection uses TCP or TCP/TLS.
   This transport connection is referred to as the 'connection' or 'RTSP
   connection' within this document.

   RTSP requests can be transmitted using the two different connection
   scenarios listed below:

   o  persistent - a transport connection is used for several request/
      response transactions;

   o  transient - a transport connection is used for each single request
      /response transaction.

   RFC 2326 attempted to specify an optional mechanism for transmitting
   RTSP messages in connectionless mode over a transport protocol such
   as UDP.  However, it was not specified in sufficient detail to allow
   for interoperable implementations.  In an attempt to reduce
   complexity and scope, and due to lack of interest, RTSP 2.0 does not
   attempt to define a mechanism for supporting RTSP over UDP or other
   connectionless transport protocols.  A side-effect of this is that
   RTSP requests MUST NOT be sent to multicast groups since no
   connection can be established with a specific receiver in multicast
   environments.

   Certain RTSP headers, such as the CSeq header (Section 18.20), which
   may appear to be relevant only to connectionless transport scenarios
   are still retained and MUST be implemented according to the
   specification.  In the case of CSeq, it is quite useful for matching
   responses to requests if the requests are pipelined (see Section 12).
   It is also useful in proxies for keeping track of the different
   requests when aggregating several client requests on a single TCP
   connection.

10.1.  Reliability and Acknowledgements

   Since RTSP messages are transmitted using reliable transport
   protocols, they MUST NOT be retransmitted at the RTSP protocol level.
   Instead, the implementation must rely on the underlying transport to
   provide reliability.  The RTSP implementation may use any indication
   of reception acknowledgment of the message from the underlying
   transport protocols to optimize the RTSP behavior.

      If both the underlying reliable transport such as TCP and the RTSP
      application retransmit requests, each packet loss or message loss
      may result in two retransmissions.  The receiver typically cannot



Schulzrinne, et al.      Expires August 14, 2014               [Page 49]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      take advantage of the application-layer retransmission since the
      transport stack will not deliver the application-layer
      retransmission before the first attempt has reached the receiver.
      If the packet loss is caused by congestion, multiple
      retransmissions at different layers will exacerbate the
      congestion.

   Lack of acknowledgment of an RTSP request should be handled within
   the constraints of the connection timeout considerations described
   below (Section 10.4).

10.2.  Using Connections

   A TCP transport can be used for both persistent connections (for
   several message exchanges) and transient connections (for a single
   message exchange).  Implementations of this specification MUST
   support RTSP over TCP.  The scheme of the RTSP URI (Section 4.2)
   allows the client to specify the port it will contact the server on,
   and defines the default port to use if one is not explicitly given.

   In addition to the registered default ports, i.e., 554 (rtsp) and 322
   (rtsps), there is an alternative port 8554 registered.  This port may
   provide some benefits over non-registered ports if a RTSP server is
   unable to use the default ports.  The benefits may include pre-
   configured security policies as well as classifiers in network
   monitoring tools.

   A RTSP client opening a TCP connection for accessing a particular
   resource as identified by a URI uses the IP address and port derived
   from the host and port parts of the URI.  The IP address is either
   the explicit address provided in the URI or any of the addresses
   provided when performing A and AAAA record DNS lookups of the host
   name in the URI.

   A server MUST handle both persistent and transient connections.

      Transient connections facilitate mechanisms for fault tolerance.
      They also allow for application layer mobility.  A server and
      client pair that support transient connections can survive the
      loss of a TCP connection; e.g., due to a NAT timeout.  When the
      client has discovered that the TCP connection has been lost, it
      can set up a new one when there is need to communicate again.

   A persistent connection is RECOMMENDED to be used for all
   transactions between the server and client, including messages for
   multiple RTSP sessions.  However, a persistent connection MAY be
   closed after a few message exchanges.  For example, a client may use
   a persistent connection for the initial SETUP and PLAY message



Schulzrinne, et al.      Expires August 14, 2014               [Page 50]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   exchanges in a session and then close the connection.  Later, when
   the client wishes to send a new request, such as a PAUSE for the
   session, a new connection would be opened.  This connection may
   either be transient or persistent.

   An RTSP agent MAY use one connection to handle multiple RTSP sessions
   on the same server.  The RTSP agent SHALL NOT use more than one
   connection per RTSP session at any given point.

      Having only one connection in use at any time avoids confusion on
      which connection any server to client requests shall be sent on.
      Using a single connection for multiple RTSP session also saves
      complexity by enabling the server to maintain less state about its
      connection resources on the server.  Not using more than one
      connection at a time for a particular RTSP session avoids wasting
      connection resources and allows the server to track only the most
      recently used client to server connection for each RTSP session as
      being the currently valid server to client connection.

   RTSP allows a server to send requests to a client.  However, this can
   be supported only if a client establishes a persistent connection
   with the server.  In cases where a persistent connection does not
   exist between a server and its client, due to the lack of a signaling
   channel the server may be forced to silently discard RTSP messages,
   and may even drop an RTSP session without notifying the client.  An
   example of such a case is when the server desires to send a REDIRECT
   request for an RTSP session to the client but is not able to do so
   because it cannot reach the client.  A server that attempts to send a
   request to a client that has no connection currently to the server
   SHALL discard the request.

      Without a persistent connection between the client and the server,
      the media server has no reliable way of reaching the client.
      Because the likely failure of server to client established
      connections the server will not even attempt establishing any
      connection.

      Queuing of server to client requests has been considered.  However
      a security issue exists as to how it might be possible to
      authorize a client establishing a new connection as being a
      legitimate receiver of a request related to a particular RTSP
      session, without the client first issuing requests related to the
      pending request.  Thus, it would be likely to make any such
      requests even more delayed and less useful.

   The sending of client and server requests can be asynchronous events.
   To avoid deadlock situations both client and server MUST be able to
   send and receive requests simultaneously.  As an RTSP response may be



Schulzrinne, et al.      Expires August 14, 2014               [Page 51]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   queued up for transmission, reception or processing behind the peer
   RTSP agent's own requests, all RTSP agents are required to have a
   certain capability of handling outstanding messages.  A potential
   issue is that outstanding requests may timeout despite them being
   processed by the peer due to the response being caught in the queue
   behind a number of requests that the RTSP agent is processing but
   that take some time to complete.  To avoid this problem an RTSP agent
   is recommended to buffer incoming messages locally so that any
   response messages can be processed immediately upon reception.  If
   responses are separated from requests and directly forwarded for
   processing, not only can the result be used immediately, the state
   associated with that outstanding request can also be released.
   However, buffering a number of requests on the receiving RTSP agent
   consumes resources and enables a resource exhaustion attack on the
   agent.  Therefore this buffer should be limited so that an
   unreasonable number of requests or total message size is not allowed
   to consume the receiving agent's resources.  In most APIs, having the
   receiving agent stop reading from the TCP socket will result in TCP's
   window being clamped.  Thus forcing the buffering onto the sending
   agent when the load is larger than expected.  However, as both RTSP
   message sizes and frequency may be changed in the future by protocol
   extensions, an agent should be careful against taking harsher
   measurements against a potential attack.  When under attack an RTSP
   agent can close TCP connections and release state associated with
   that TCP connection.

   To provide some guidance on what is reasonable the following
   guidelines are given.  It is RECOMMENDED that:

   o  an RTSP agent should not have more than 10 outstanding requests
      per RTSP session;

   o  an RTSP agent should not have more than 10 outstanding requests
      that are not related to an RTSP session or that are requesting to
      create an RTSP session.

   In light of the above, it is RECOMMENDED that clients use persistent
   connections whenever possible.  A client that supports persistent
   connections MAY "pipeline" its requests (see Section 12).

   RTSP Agents can send requests to multiple different destinations,
   either servers or client contexts over the same connection to a
   proxy.  Then the proxy forks the message to the different
   destinations over proxy to agent connections.  In these cases when
   multiple requests are outstanding the requesting agent MUST be ready
   to receive the responses out of order compared to the order they
   where sent on the connection.  The order between multiple messages




Schulzrinne, et al.      Expires August 14, 2014               [Page 52]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   for each destination will be maintained, however, the order between
   response from different destinations can be different.

      The reason for this is to avoid a head-of-line blocking
      sitauation.  In a sequence of requests an early outstanding
      request may take time to be processed at one destination.
      Simultaneously, a response from any other destination that was
      later in the sequence of requests, may have arrived at the proxy.
      Thus allowing out-of-order responses avoids forcing the proxy to
      buffer this response and instead deliver it as soon as possible.
      Note, this will not affect the order in which the messages sent to
      each separate destination were processed at request destination.

   This scenario can occur in two cases involving proxies.  The first is
   a client issuing requests for sessions on different servers using a
   common client to proxy connection.  The second is for server to
   client requests, like REDIRECT being sent by the server over a common
   transport connection the proxy created for its different connecting
   clients.

10.3.  Closing Connections

   The client MAY close a connection at any point when no outstanding
   request/response transactions exist for any RTSP session being
   managed through the connection.  The server, however, SHOULD NOT
   close a connection until all RTSP sessions being managed through the
   connection have been timed out (Section 18.49).  A server SHOULD NOT
   close a connection immediately after responding to a session-level
   TEARDOWN request for the last RTSP session being controlled through
   the connection.  Instead, the server should wait for a reasonable
   amount of time for the client to receive and act upon the TEARDOWN
   response, and initiate the connection closing.  The server SHOULD
   wait at least 10 seconds after sending the TEARDOWN response before
   closing the connection.

      This is to ensure that the client has time to issue a SETUP for a
      new session on the existing connection after having torn the last
      one down. 10 seconds should give the client ample opportunity to
      get its message to the server.

   A server SHOULD NOT close the connection directly as a result of
   responding to a request with an error code.

      Certain error responses such as "460 Only Aggregate Operation
      Allowed" (Section 17.4.25) are used for negotiating capabilities
      of a server with respect to content or other factors.  In such
      cases, it is inefficient for the server to close a connection on
      an error response.  Also, such behavior would prevent



Schulzrinne, et al.      Expires August 14, 2014               [Page 53]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      implementation of advanced/special types of requests or result in
      extra overhead for the client when testing for new features.  On
      the other hand, keeping connections open after sending an error
      response poses a Denial of Service security risk (Section 21).

   The server MAY close a connection if it receives an incomplete
   message and if the message is not completed within a reasonable
   amount of time.  It is RECOMMENDED that the server waits at least 10
   seconds for the completion of a message or for the next part of the
   message to arrive (which is an indication that the transport and the
   client are still alive).  Servers believing they are under attack or
   otherwise starved for resources during that event MAY consider using
   a shorter timeout.

   If a server closes a connection while the client is attempting to
   send a new request, the client will have to close its current
   connection, establish a new connection and send its request over the
   new connection.

   An RTSP message SHOULD NOT be terminated by closing the connection.
   Such a message MAY be considered to be incomplete by the receiver and
   discarded.  An RTSP message is properly terminated as defined in
   Section 5.

10.4.  Timing Out Connections and RTSP Messages

   Receivers of a request (responder) SHOULD respond to requests in a
   timely manner even when a reliable transport such as TCP is used.
   Similarly, the sender of a request (requester) SHOULD wait for a
   sufficient time for a response before concluding that the responder
   will not be acting upon its request.

   A responder SHOULD respond to all requests within 5 seconds.  If the
   responder recognizes that processing of a request will take longer
   than 5 seconds, it SHOULD send a 100 (Continue) response as soon as
   possible.  It SHOULD continue sending a 100 response every 5 seconds
   thereafter until it is ready to send the final response to the
   requester.  After sending a 100 response, the responder MUST send a
   final response indicating the success or failure of the request.

   A requester SHOULD wait at least 10 seconds for a response before
   concluding that the responder will not be responding to its request.
   After receiving a 100 response, the requester SHOULD continue waiting
   for further responses.  If more than 10 seconds elapses without
   receiving any response, the requester MAY assume that the responder
   is unresponsive and abort the connection by closing the TCP
   connection.




Schulzrinne, et al.      Expires August 14, 2014               [Page 54]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   In cases multiple RTSP sessions share the same transport connection,
   abandoning a request and closing the connection may have significant
   impact on those other sessions.  First of all, other RTSP requests
   may have become queued up due to the request taking long time to
   process.  Secondly also those sessions loose the possibility to
   receive server to client requests.  To mitigate that situation the
   RTSP client or server SHOULD establish a new connection and send any
   queued up and non-responded requests on this new connection.
   Secondly, to ensure that the RTSP server knows which connection that
   is valid for a particular RTSP session, the RTSP agent SHOULD send a
   keep-alive request, if no other request will be sent immediately for
   that RTSP session, for each RTSP session on the old connection.  The
   keep-alive request will normally be a SET_PARAMETER with a session
   header to inform the server that this agent cares about this RTSP
   session.

   A requester SHOULD wait longer than 10 seconds for a response if it
   is experiencing significant transport delays on its connection to the
   responder.  The requester is capable of determining the round trip
   time (RTT) of the request/response cycle using the Timestamp header
   (Section 18.53) in any RTSP request.

      10 seconds was chosen for the following reasons.  It gives TCP
      time to perform a couple of retransmissions, even if operating on
      default values.  It is short enough that users may not abandon the
      process themselves.  However, it should be noted that 10 seconds
      can be aggressive on certain type of networks.  The 5 seconds
      value for 1xx messages is half the timeout giving a reasonable
      chance of successful delivery before timeout happens on the
      requester side.

10.5.  Showing Liveness

   RTSP requires the client to periodically show its liveness to the
   server or the server may terminate any session state.  Several
   different protocol mechanism includes in their usage a liveness proof
   from the client.  These mechanisms are; RTSP requests with a Session
   header to the server; if RTP & RTCP is used for media data transport
   and the transport is established the RTCP message proves liveness; or
   through any other used media transport protocol capable of indicating
   liveness of the RTSP client.  It is RECOMMENDED that a client does
   not wait to the last second of the timeout before trying to send a
   liveness message.  The RTSP message may take some time to arrive
   safely at the receiver, due to packet loss and TCP retransmissions.
   To show liveness between RTSP requests being issued to accomplish
   other things, the following mechanisms can be used, in descending
   order of preference:




Schulzrinne, et al.      Expires August 14, 2014               [Page 55]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   RTCP: If RTP is used for media transport RTCP SHOULD be used.  If
         RTCP is used to report transport statistics, it will
         necessarily also function as a keep-alive.  The server can
         determine the client by network address and port together with
         the fact that the client is reporting on the server's RTP
         sender sources (SSRCs).  A downside of using RTCP is that it
         only gives statistical guarantees of reaching the server.
         However, the probability of a false client timeout is so low
         that it can be ignored in most cases.  For example, assume a
         session with 60 seconds timeout and enough bitrate assigned to
         RTCP messages to send a message from client to server on
         average every 5 seconds.  That client has, for a network with
         5% packet loss, a probability of failing to confirm liveness
         within the timeout interval for that session of 2.4*E-16.
         Sessions with shorter timeouts, or much higher packet loss, or
         small RTCP bandwidths SHOULD also implement one or more of the
         mechanisms below.

   SET_PARAMETER:  When using SET_PARAMETER for keep-alive, a body
         SHOULD NOT be included.  This method is the RECOMMENDED RTSP
         method to use for a request intended only to perform keep-
         alive.  Support of SET_PARAMETER is mandatory for RTSP Servers
         to ensure clients can use this method.

   GET_PARAMETER:  When using GET_PARAMETER for keep-alive, a body
         SHOULD NOT be included.  Dependent on implementation support in
         server.  Use OPTIONS method to determine if there are method
         support or simply try.

   OPTIONS:  This method is also usable, but it causes the server to
         perform more unnecessary processing and results in bigger
         responses than necessary for the task.  The reason is that the
         server needs to determine the capabilities associated with the
         media resource to correctly populate the Public and Allow
         headers.

   The timeout parameter of the Session header (Section 18.49) MAY be
   included in a SETUP response, and MUST NOT be included in requests.
   The server uses it to indicate to the client how long the server is
   prepared to wait between RTSP commands or other signs of life before
   closing the session due to lack of activity (see Appendix B).  The
   timeout is measured in seconds, with a default of 60 seconds.  The
   length of the session timeout MUST NOT be changed in an established
   session.







Schulzrinne, et al.      Expires August 14, 2014               [Page 56]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


10.6.  Use of IPv6

   Explicit IPv6 [RFC2460] support was not present in RTSP 1.0 (RFC
   2326).  RTSP 2.0 has been updated for explicit IPv6 support.
   Implementations of RTSP 2.0 MUST understand literal IPv6 addresses in
   URIs and RTSP headers.  Although the general URI format envisages
   potential future new versions of the literal IP address, usage of any
   such new version would require other modifications to the RTSP
   specification (e.g. address fields in the Transport header
   (Section 18.54)).

10.7.  Overload Control

   Overload in RTSP can occur when servers and proxies have insufficient
   resources to complete the processing of a request.  An improper
   handling of such an overload situation at proxies and servers can
   impact the operation of the RTSP deployment, and probably worsen the
   situation.  RTSP defines the 503 (Service Unavailable) response
   (Section 17.5.4) to let servers and proxies notify requesting proxies
   and RTSP clients about an overload situation.  In conjunction with
   the Retry-After header (Section 18.44) the server or proxy can
   indicate the time after which the requesting entity can send another
   request to the proxy or server.

   There are two scopes of such 503 answers, one for established RTSP
   sessions, where the request resulting in the 503 response as well as
   the response carries a Session header identifying the session which
   is suffering overload.  This response only applies to this particular
   session.  The other scope is the general RTSP server as identified by
   the host in the request URL.  Such a 503 answer with any Retry-After
   header applies to all non-session specific requests to that server,
   including SETUP request intended to create a new RTSP session.

   Another scope for overload situation exists, which is the RTSP proxy.
   To enable an RTSP proxy to signal that it is overloaded, or otherwise
   unavailable and can't handle the request, a 553 response code has
   been defined with the meaning "Proxy Unavailable".  As with servers,
   there is a separation in response scopes between requests associated
   with existing RTSP sessions, and requests to create new sessions or
   general proxy requests.

   Simply implementing and using the 503 (Service Unavailable) and 553
   (Proxy Unavailable) is not sufficient for properly handling overload
   situations.  For instance, a simplistic approach would be to send the
   503 response with a Retry-After header set to a fixed value.
   However, this can cause the situation where multiple RTSP clients
   again send requests to a proxy or server at roughly the same time
   which may again cause an overload situation, or if the "old" overload



Schulzrinne, et al.      Expires August 14, 2014               [Page 57]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   situation is not yet solved, i.e., the length indicated in the Retry-
   After header was too short.

   An RTSP server or proxy in an overload situation must select the
   value of the Retry-After header carefully and bearing in mind its
   current load situation.  It is REQUIRED to increase the timeout
   period in proportion to the current load on the server, i.e., an
   increasing workload should result in an increased length of the
   indicated unavailability.  It is REQUIRED to not send the same value
   in the Retry-After header to all requesting proxies and clients, but
   to add a variation to the mean value of the Retry-After header.

   A more complex case may arise when a load balancing RTSP proxy is in
   use.  This is the case when an RTSP proxy is used to select amongst a
   set of RTSP servers to handle the requests or when multiple server
   addresses are available for a given server name.  The proxy or client
   may receive a 503 (Service Unavailable) or 553 (Proxy Unavailable)
   from one of its RTSP servers or proxies, or a TCP timeout (if the
   server is even unable to handle the request message).  The proxy or
   client simply retries the other addresses or configured proxies, but
   may also receive a 503 (Service Unavailable) or 553 (Proxy
   Unavailable) response or TCP timeouts from those addresses.  In such
   a situation, where none of the RTSP servers/proxies/addresses can
   handle the request, the RTSP agent has to wait before it can send any
   new requests to the RTSP server.  Any additional request to a
   specific address MUST be delayed according to the Retry-After headers
   received.  For addresses where no response was received or TCP
   timeout occurred, an initial wait timer SHOULD be set to 5 seconds.
   That timer MUST be doubled for each additional failure to connect or
   receive response until the value exceeds 30 minutes when the timers
   mean value may be set to 30 minutes.  It is REQUIRED to not set the
   same value in the timer for each scheduling, but instead to add a
   variation to the mean value, resulting in picking a random value
   within the range from 0.5 to 1.5 times the mean value.

11.  Capability Handling

   This section describes the available capability handling mechanism
   which allows RTSP to be extended.  Extensions to this version of the
   protocol are basically done in two ways.  First, new headers can be
   added.  Secondly, new methods can be added.  The capability handling
   mechanism is designed to handle both cases.

   When a method is added, the involved parties can use the OPTIONS
   method to discover whether it is supported.  This is done by issuing
   an OPTIONS request to the other party.  Depending on the URI it will
   either apply in regards to a certain media resource, the whole server
   in general, or simply the next hop.  The OPTIONS response MUST



Schulzrinne, et al.      Expires August 14, 2014               [Page 58]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   contain a Public header which declares all methods supported for the
   indicated resource.

   It is not necessary to use OPTIONS to discover support of a method,
   as the client could simply try the method.  If the receiver of the
   request does not support the method it will respond with an error
   code indicating the method is either not implemented (501) or does
   not apply for the resource (405).  The choice between the two
   discovery methods depends on the requirements of the service.

   Feature-tags are defined to handle functionality additions that are
   not new methods.  Each feature-tag represents a certain block of
   functionality.  The amount of functionality that a feature-tag
   represents can vary significantly.  A feature-tag can for example
   represent the functionality a single RTSP header provides.  Another
   feature-tag can represent much more functionality, such as the
   "play.basic" feature-tag (Section 11.1) which represents the minimal
   media delivery for playback implementation.

   Feature-tags are used to determine whether the client, server or
   proxy supports the functionality that is necessary to achieve the
   desired service.  To determine support of a feature-tag, several
   different headers can be used, each explained below:

   Supported:  This header is used to determine the complete set of
         functionality that both client and server have in general and
         is not dependent on a specific resource.  The intended usage is
         to determine before one needs to use a functionality that it is
         supported.  It can be used in any method, but OPTIONS is the
         most suitable one as it at the same time determines all methods
         that are implemented.  When sending a request the requester
         declares all its capabilities by including all supported
         feature-tags.  This results in the receiver learning the
         requester's feature support.  The receiver then includes its
         set of features in the response.

   Proxy-Supported:  This header is used similarly to the Supported
         header, but instead of giving the supported functionality of
         the client or server it provides both the requester and the
         responder a view of the common functionality supported in
         general by all members of the proxy chain between the two
         supports and not dependent on the resource.  Proxies are
         required to add this header whenever the Supported header is
         present, but proxies may also add it independently of the
         requester.

   Require:  This header can be included in any request where the end-
         point, i.e., the client or server, is required to understand



Schulzrinne, et al.      Expires August 14, 2014               [Page 59]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         the feature to correctly perform the request.  This can, for
         example, be a SETUP request where the server is required to
         understand a certain parameter to be able to set up the media
         delivery correctly.  Ignoring this parameter would not have the
         desired effect and is not acceptable.  Therefore the end-point
         receiving a request containing a Require MUST negatively
         acknowledge any feature that it does not understand and not
         perform the request.  The response in cases where features are
         not supported are 551 (Option Not Supported).  Also the
         features that are not supported are given in the Unsupported
         header in the response.

   Proxy-Require:  This header has the same purpose and workings as
         Require except that it only applies to proxies and not the end-
         point.  Features that need to be supported by both proxies and
         end-points need to be included in both the Require and Proxy-
         Require header.

   Unsupported:  This header is used in a 551 error response, to
         indicate which features were not supported.  Such a response is
         only the result of the usage of the Require and/or Proxy-
         Require header where one or more features where not supported.
         This information allows the requester to make the best of
         situations as it knows which features are not supported.

11.1.  Feature Tag: play.basic

   An implementation supporting all normative parts of this
   specification for the setup and control of playback of media uses the
   feature tag "play.basic" to indicate this support.  The appendices
   (starting with letters), are not part of the functionality include in
   the feature tag unless the appendix is explicitly specified in a main
   section as being a required appendix.

      Note: This feature tag does not mandate any media delivery
      protocol, such as RTP.

      In RTSP 1.0 there was a minimal implementation section.  However,
      that was not consistent with the rest of the specification.  So
      rather than making an attempt to explicitly enumerate the features
      for play.basic this specification has to be taken as a whole and
      the necessary features normatively defined as being required are
      included.








Schulzrinne, et al.      Expires August 14, 2014               [Page 60]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


12.  Pipelining Support

   Pipelining is a general method to improve performance of request
   response protocols by allowing the requesting agent to have more than
   one request outstanding and send them over the same persistent
   connection.  For RTSP, where the relative order of requests will
   matter, it is important to maintain the order of the requests.
   Because of this, the responding agent MUST process the incoming
   requests in their sending order.  The sending order can be determined
   by the CSeq header and its sequence number.  For TCP the delivery
   order will be the same between two agents, as the sending order.  The
   processing of the request MUST also have been finished before
   processing the next request from the same agent.  The responses MUST
   be sent in the order the requests were processed.

   RTSP 2.0 has extended support for pipelining compared to RTSP 1.0.
   The major improvement is to allow all requests involved in setting up
   and initiating media delivery to be pipelined after each other.  This
   is accomplished by the utilization of the Pipelined-Requests header
   (see Section 18.33).  This header allows a client to request that two
   or more requests are processed in the same RTSP session context which
   the first request creates.  In other words, a client can request that
   two or more media streams are set-up and then played without needing
   to wait for a single response.  This speeds up the initial startup
   time for an RTSP session by at least one RTT.

   If a pipelined request builds on the successful completion of one or
   more prior requests the requester must verify that all requests were
   executed as expected.  A common example will be two SETUP requests
   and a PLAY request.  In case one of the SETUP fails unexpectedly, the
   PLAY request can still be successfully executed.  However, the
   resulting presentation will not be as expected by the requesting
   client, as only a single media instead of two will be played.  In
   this case the client can send a PAUSE request, correct the failing
   SETUP request and then request it to be played.

13.  Method Definitions

   The method indicates what is to be performed on the resource
   identified by the Request-URI.  The method name is case-sensitive.
   New methods may be defined in the future.  Method names MUST NOT
   start with a $ character (decimal 36) and MUST be a token as defined
   by the ABNF [RFC5234] in the syntax chapter Section 20.  The methods
   are summarized in Table 7.







Schulzrinne, et al.      Expires August 14, 2014               [Page 61]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


    +---------------+-----------+--------+-------------+-------------+
    | method        | direction | object | Server req. | Client req. |
    +---------------+-----------+--------+-------------+-------------+
    | DESCRIBE      | C -> S    | P,S    | recommended | recommended |
    |               |           |        |             |             |
    | GET_PARAMETER | C -> S    | P,S    | optional    | optional    |
    |               |           |        |             |             |
    |               | S -> C    | P,S    | optional    | optional    |
    |               |           |        |             |             |
    | OPTIONS       | C -> S    | P,S    | required    | required    |
    |               |           |        |             |             |
    |               | S -> C    | P,S    | optional    | optional    |
    |               |           |        |             |             |
    | PAUSE         | C -> S    | P,S    | required    | required    |
    |               |           |        |             |             |
    | PLAY          | C -> S    | P,S    | required    | required    |
    |               |           |        |             |             |
    | PLAY_NOTIFY   | S -> C    | P,S    | required    | required    |
    |               |           |        |             |             |
    | REDIRECT      | S -> C    | P,S    | optional    | required    |
    |               |           |        |             |             |
    | SETUP         | C -> S    | S      | required    | required    |
    |               |           |        |             |             |
    | SET_PARAMETER | C -> S    | P,S    | required    | optional    |
    |               |           |        |             |             |
    |               | S -> C    | P,S    | optional    | optional    |
    |               |           |        |             |             |
    | TEARDOWN      | C -> S    | P,S    | required    | required    |
    |               |           |        |             |             |
    |               | S -> C    | P      | required    | required    |
    +---------------+-----------+--------+-------------+-------------+

   Table 7: Overview of RTSP methods, their direction, and what objects
    (P: presentation, S: stream) they operate on.  Further it indicates
     if a server or a client implementation are required (mandatory),
         recommended or if it is optional to implement the method.

      Note on Table 7: GET_PARAMETER is optional.  For example, a fully
      functional server can be built to deliver media without any
      parameters.  However, SET_PARAMETER is required, i.e., mandatory
      to implement for the server, this is due to its usage for keep-
      alive.  PAUSE is required because it is the only way of leaving
      the Play state without terminating the whole session.

   If an RTSP agent does not support a particular method, it MUST return
   501 (Not Implemented) and the requesting RTSP agent, in turn, SHOULD
   NOT try this method again for the given agent / resource combination.
   An RTSP proxy whose main function is to log or audit and not modify



Schulzrinne, et al.      Expires August 14, 2014               [Page 62]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   transport or media handling in any way MAY forward RTSP messages with
   unknown methods.  Note that the proxy still needs to perform the
   minimal required processing, like adding the Via header.

13.1.  OPTIONS

   The semantics of the RTSP OPTIONS method is similar to that of the
   HTTP OPTIONS method described in [H9.2].  In RTSP however, OPTIONS is
   bi-directional, in that a client can send the request to a server and
   vice versa.  A client MUST implement the capability to send an
   OPTIONS request and a server or a proxy MUST implement the capability
   to respond to an OPTIONS request.  In addition to this "MUST
   implement" functionality, clients, servers and proxies MAY provide
   support both for sending OPTIONS requests and generating responses to
   the requests.

   An OPTIONS request may be issued at any time.  Such a request does
   not modify the session state.  However, it may prolong the session
   lifespan (see below).  The URI in an OPTIONS request determines the
   scope of the request and the corresponding response.  If the Request-
   URI refers to a specific media resource on a given host, the scope is
   limited to the set of methods supported for that media resource by
   the indicated RTSP agent.  A Request-URI with only the host address
   limits the scope to the specified RTSP agent's general capabilities
   without regard to any specific media.  If the Request-URI is an
   asterisk ("*"), the scope is limited to the general capabilities of
   the next hop (i.e., the RTSP agent in direct communication with the
   request sender).

   Regardless of the scope of the request, the Public header MUST always
   be included in the OPTIONS response listing the methods that are
   supported by the responding RTSP agent.  In addition, if the scope of
   the request is limited to a media resource, the Allow header MUST be
   included in the response to enumerate the set of methods that are
   allowed for that resource unless the set of methods completely
   matches the set in the Public header.  If the given resource is not
   available, the RTSP agent SHOULD return an appropriate response code
   such as 3rr or 4xx.  The Supported header MAY be included in the
   request to query the set of features that are supported by the
   responding RTSP agent.

   The OPTIONS method can be used to keep an RTSP session alive.
   However, this is not the preferred way of session keep-alive
   signaling, see Section 18.49.  An OPTIONS request intended for
   keeping alive an RTSP session MUST include the Session header with
   the associated session identifier.  Such a request SHOULD also use
   the media or the aggregated control URI as the Request-URI.




Schulzrinne, et al.      Expires August 14, 2014               [Page 63]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Example:

     C->S:  OPTIONS rtsp://server.example.com RTSP/2.0
            CSeq: 1
            User-Agent: PhonyClient/1.2
            Proxy-Require: gzipped-messages
            Supported: play.basic

     S->C:  RTSP/2.0 200 OK
            CSeq: 1
            Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS
            Supported: play.basic, setup.rtp.rtcp.mux, play.scale
            Server: PhonyServer/1.1


   Note that some of the feature-tags in Supported and Proxy-Require are
   fictitious features.

13.2.  DESCRIBE

   The DESCRIBE method is used to retrieve the description of a
   presentation or media object from a server.  The Request-URI of the
   DESCRIBE request identifies the media resource of interest.  The
   client MAY include the Accept header in the request to list the
   description formats that it understands.  The server MUST respond
   with a description of the requested resource and return the
   description in the message body of the response, if the DESCRIBE
   method request can be successfully fulfilled.  The DESCRIBE reply-
   response pair constitutes the media initialization phase of RTSP.

   The DESCRIBE response SHOULD contain all media initialization
   information for the resource(s) that it describes.  Servers SHOULD
   NOT use the DESCRIBE response as a means of media indirection by
   having the description point at another server; instead, using the
   3rr responses is RECOMMENDED.

      By forcing a DESCRIBE response to contain all media initialization
      information for the set of streams that it describes, and
      discouraging the use of DESCRIBE for media indirection, any
      looping problems can be avoided that might have resulted from
      other approaches.

   Example:








Schulzrinne, et al.      Expires August 14, 2014               [Page 64]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/2.0
           CSeq: 312
           User-Agent: PhonyClient/1.2
           Accept: application/sdp, application/example

     S->C: RTSP/2.0 200 OK
           CSeq: 312
           Date: Thu, 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer/1.1
           Content-Base: rtsp://server.example.com/fizzle/foo/
           Content-Type: application/sdp
           Content-Length: 358

           v=0
           o=MNobody 2890844526 2890842807 IN IP4 192.0.2.46
           s=SDP Seminar
           i=A Seminar on the session description protocol
           u=http://www.example.com/lectures/sdp.ps
           e=seminar@example.com (Seminar Management)
           c=IN IP4 0.0.0.0
           a=control:*
           t=2873397496 2873404696
           m=audio 3456 RTP/AVP 0
           a=control:audio
           m=video 2232 RTP/AVP 31
           a=control:video

   Media initialization is a requirement for any RTSP-based system, but
   the RTSP specification does not dictate that this is required to be
   done via the DESCRIBE method.  There are three ways that an RTSP
   client may receive initialization information:

   o  via an RTSP DESCRIBE request

   o  via some other protocol (HTTP, email attachment, etc.)

   o  via some form of user interface

   If a client obtains a valid description from an alternate source, the
   client MAY use this description for initialization purposes without
   issuing a DESCRIBE request for the same media.  The client should use
   any MTag to either validate the presentation description or make the
   session establishment conditional on being valid.

   It is RECOMMENDED that minimal servers support the DESCRIBE method,
   and highly recommended that minimal clients support the ability to
   act as "helper applications" that accept a media initialization file




Schulzrinne, et al.      Expires August 14, 2014               [Page 65]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   from a user interface, and/or other means that are appropriate to the
   operating environment of the clients.

13.3.  SETUP

   The description below uses the following states in a protocol state
   machine that is related to a specific session when that session has
   been created.  The state transitions are driven by protocol
   interactions.  For additional information about the state machine see
   Appendix B.

   Init: Initial state: no session exists.

   Ready:  Session is ready to start playing.

   Play: Session is playing, i.e., sending media stream data in the
         direction S->C.

   The SETUP request for a URI specifies the transport mechanism to be
   used for the streamed media.  The SETUP method may be used in two
   different cases; Create an RTSP session and change the transport
   parameters of already set up media stream(s).  SETUP can be used in
   all three states; Init, and Ready, for both purposes and in PLAY to
   change the transport parameters.  The usage of SETUP method in the
   Play state to add a media resource to the session is unspecified
   (Section 3.1).

   The Transport header, see Section 18.54, specifies the media
   transport parameters acceptable to the client for data transmission;
   the response will contain the transport parameters selected by the
   server.  This allows the client to enumerate in descending order of
   preference the transport mechanisms and parameters acceptable to it,
   while the server can select the most appropriate.  It is expected
   that the session description format used will enable the client to
   select a limited number of possible configurations that are offered
   to the server to choose from.  All transport related parameters SHALL
   be included in the Transport header; the use of other headers for
   this purpose is NOT RECOMMENDED due to middleboxes, such as firewalls
   or NATs.

   For the benefit of any intervening firewalls, a client MUST indicate
   the known transport parameters, even if it has no influence over
   these parameters, for example, where the server advertises a fixed
   multicast address as destination.

      Since SETUP includes all transport initialization information,
      firewalls and other intermediate network devices (which need this
      information) are spared the more arduous task of parsing the



Schulzrinne, et al.      Expires August 14, 2014               [Page 66]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      DESCRIBE response, which has been reserved for media
      initialization.

   The client MUST include the Accept-Ranges header in the request
   indicating all supported unit formats in the Range header.  This
   allows the server to know which formats it may use in future session
   related responses, such as a PLAY response without any range in the
   request.  If the client does not support a time format necessary for
   the presentation, the server MUST respond using 456 (Header Field Not
   Valid for Resource) and include the Accept-Ranges header with the
   range unit formats supported for the resource.

   In a SETUP response the server MUST include the Accept-Ranges header
   (see Section 18.5) to indicate which time formats are acceptable to
   use for this media resource.

   The SETUP response 200 OK MUST include the Media-Properties header
   (see Section 18.29 ).  The combination of the parameters of the
   Media-Properties header indicates the nature of the content present
   in the session (see also Section 4.7).  For example, a live stream
   with time shifting is indicated by

   o  Random Access set to Random-Access,

   o  Content Modifications set to Time Progressing,

   o  Retention set to Time-Duration (with specific recording window
      time value).

   The SETUP response 200 OK MUST include the Media-Range header (see
   Section 18.30) if the media is Time-Progressing.

   A basic example for SETUP:


















Schulzrinne, et al.      Expires August 14, 2014               [Page 67]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
           CSeq: 302
           Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",
                      RTP/AVP/TCP;unicast;interleaved=0-1
           Accept-Ranges: npt, clock
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 302
           Date: Thu, 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer/1.1
           Session: 47112344;timeout=60
           Transport: RTP/AVP;unicast;dest_addr="192.0.2.53:4588"/
                      "192.0.2.53:4589"; src_addr="198.51.100.241:6256"/
                      "198.51.100.241:6257"; ssrc=2A3F93ED
           Accept-Ranges: npt
           Media-Properties: Random-Access=3.2, Time-Progressing,
                             Time-Duration=3600.0
           Media-Range: npt=0-2893.23

   In the above example the client wants to create an RTSP session
   containing the media resource "rtsp://example.com/foo/bar/baz.rm".
   The transport parameters acceptable to the client are either RTP/AVP/
   UDP (UDP per default) to be received on client port 4588 and 4589 at
   the address the RTSP setup connection comes from or RTP/AVP
   interleaved on the RTSP control channel.  The server selects the RTP/
   AVP/UDP transport and adds the address and ports it will send and
   receive RTP and RTCP from, and the RTP SSRC that will be used by the
   server.

   The server MUST generate a session identifier in response to a
   successful SETUP request, unless a SETUP request to a server includes
   a session identifier or a Pipelined-Requests header referencing an
   existing session context, in which case the server MUST bundle this
   SETUP request into the existing session (aggregated session) or
   return error 459 (Aggregate Operation Not Allowed) (see
   Section 17.4.24).  An Aggregate control URI MUST be used to control
   an aggregated session.  This URI MUST be different from the stream
   control URIs of the individual media streams included in the
   aggregate (see Section 13.4.2 for aggregated sessions and for the
   particular URIs see Appendix D.1.1).  The Aggregate control URI is to
   be specified by the session description if the server supports
   aggregated control and aggregated control is desired for the session.
   However, even if aggregated control is offered the client MAY chose
   to not set up the session in aggregated control.  If an Aggregate
   control URI is not specified in the session description, it is
   normally an indication that non-aggregated control should be used.




Schulzrinne, et al.      Expires August 14, 2014               [Page 68]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   The SETUP of media streams in an aggregate which has not been given
   an aggregated control URI is unspecified.

      While the session ID sometimes carries enough information for
      aggregate control of a session, the Aggregate control URI is still
      important for some methods such as SET_PARAMETER where the control
      URI enables the resource in question to be easily identified.  The
      Aggregate control URI is also useful for proxies, enabling them to
      route the request to the appropriate server, and for logging,
      where it is useful to note the actual resource that a request was
      operating on.

   A session will exist until it is either removed by a TEARDOWN request
   or is timed-out by the server.  The server MAY remove a session that
   has not demonstrated liveness signs from the client(s) within a
   certain timeout period.  The default timeout value is 60 seconds; the
   server MAY set this to a different value and indicate so in the
   timeout field of the Session header in the SETUP response.  For
   further discussion see Section 18.49.  Signs of liveness for an RTSP
   session are:

   o  Any RTSP request from a client which includes a Session header
      with that session's ID.

   o  If RTP is used as a transport for the underlying media streams, an
      RTCP sender or receiver report from the client(s) for any of the
      media streams in that RTSP session.  RTCP Sender Reports may for
      example be received in sessions where the server is invited into a
      conference session and is valid for keep-alive.

   If a SETUP request on a session fails for any reason, the session
   state, as well as transport and other parameters for associated
   streams MUST remain unchanged from their values as if the SETUP
   request had never been received by the server.

13.3.1.  Changing Transport Parameters

   A client MAY issue a SETUP request for a stream that is already set
   up or playing in the session to change transport parameters, which a
   server MAY allow.  If it does not allow changing of parameters, it
   MUST respond with error 455 (Method Not Valid In This State).  The
   reasons to support changing transport parameters include allowing
   application layer mobility and flexibility to utilize the best
   available transport as it becomes available.  If a client receives a
   455 when trying to change transport parameters while the server is in
   Play state, it MAY try to put the server in Ready state using PAUSE,
   before issuing the SETUP request again.  If that also fails the
   changing of transport parameters will require that the client



Schulzrinne, et al.      Expires August 14, 2014               [Page 69]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   performs a TEARDOWN of the affected media and then to set it up
   again.  For an aggregated session avoiding tearing down all the media
   at the same time will avoid the creation of a new session.

   All transport parameters MAY be changed.  However, the primary usage
   expected is to either change the transport protocol completely, like
   switching from Interleaved TCP mode to UDP or vice versa, or to
   change the delivery address.

   In a SETUP response for a request to change the transport parameters
   while in Play state, the server MUST include the Range to indicate at
   what point the new transport parameters will be used.  Further, if
   RTP is used for delivery, the server MUST also include the RTP-Info
   header to indicate at what timestamp and RTP sequence number the
   change will take place.  If both RTP-Info and Range are included in
   the response the "rtp_time" parameter and start point in the Range
   header MUST be for the corresponding time, i.e., be used in the same
   way as for PLAY to ensure the correct synchronization information is
   available.

   If the transport parameters change while in Play state results in a
   change of synchronization related information, for example changing
   RTP SSRC, the server MUST provide in the SETUP response the necessary
   synchronization information.  However, the server is RECOMMENDED to
   avoid changing the synchronization information if possible.

13.4.  PLAY

   This section describes the usage of the PLAY method in general, for
   aggregated sessions, and in different usage scenarios.

13.4.1.  General Usage

   The PLAY method tells the server to start sending data via the
   mechanism specified in SETUP and which part of the media should be
   played out.  PLAY requests are valid when the session is in Ready or
   Play states.  A PLAY request MUST include a Session header to
   indicate which session the request applies to.

   Upon receipt of the PLAY request, the server MUST position the normal
   play time to the beginning of the range specified in the received
   Range header, within the limits of the media resource and in
   accordance with the Seek-Style header (Section 18.47) and deliver
   stream data until the end of the range if given, until a new PLAY
   request is received, or until the end of the media is reached.  If no
   Range header is present in the PLAY request the server SHALL play
   from current pause point until the end of media.  The pause point
   defaults at session start to the beginning of the media.  For media



Schulzrinne, et al.      Expires August 14, 2014               [Page 70]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   that is time-progressing and has no retention, the pause point will
   always be set equal to NPT "now", i.e., the current delivery point.
   The pause point may also be set to a particular point in the media by
   the PAUSE method, see Section 13.6.  The pause point for media that
   is currently playing is equal to the current media position.  For
   time-progressing media with time-limited retention, if the pause
   point represents a position that is older than what is retained by
   the server, the pause point will be moved to the oldest retained.

   What range values are valid depends on the type of content.  For
   content that isn't time progressing the range value is valid if the
   given range is part of any media within the aggregate.  In other
   words the valid media range for the aggregate is the union of all of
   the media components in the aggregate.  If a given range value points
   outside of the media, the response MUST be the 457 (Invalid Range)
   error code and include the Media-Range header (Section 18.30) with
   the valid range for the media.  Except for time progressing content
   where the client requests a start point prior to what is retained,
   the start point is adjusted to the oldest retained content.  For a
   start point that is beyond the media front edge, i.e., beyond the
   current value for "now", the server SHALL adjust the start value to
   the current front edge.  The Range header's stop point value may
   point beyond the current media edge.  In that case, the server SHALL
   deliver media from the requested (and possibly adjusted) start point
   until the provided stop point, or the end of the media is reached
   prior to the specified stop point.  Please note that if one simply
   wants to play from a particular start point until the end of media
   using a Range header with an implicit stop point is RECOMMENDED.

   If a client requests to start playing at the end of media, either
   explicitly with a Range header or implicitly with a pause point that
   is at the end of media, a 457 (Invalid Range) error MUST be sent and
   include the Media-Range header (Section 18.30).  It is specified
   below that the Range header also must be included in the response and
   that it will carry the pause point in the media, in the case of the
   session being in Ready State.  Note that this also applies if the
   pause point or requested start point is at the beginning of the media
   and a Scale header (Section 18.46) is included with a negative value
   (playing backwards).

   For media with random access properties a client may express its
   preference on which policy for start point selection the server shall
   use.  This is done by including the Seek-Style header (Section 18.47)
   in the PLAY request.  The Seek-Style applied will affect the content
   of the Range header as it will be adjusted to indicate from what
   point the media actually is delivered.





Schulzrinne, et al.      Expires August 14, 2014               [Page 71]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   A client desiring to play the media from the beginning MUST send a
   PLAY request with a Range header pointing at the beginning, e.g.,
   "npt=0-".  If a PLAY request is received without a Range header and
   media delivery has stopped at the end, the server SHOULD respond with
   a 457 "Invalid Range" error response.  In that response, the current
   pause point MUST be included in a Range header.

   All range specifiers in this specification allow for ranges with an
   implicit start point (e.g., "npt=-30").  When used in a PLAY request,
   the server treats this as a request to start or resume delivery from
   the current pause point, ending at the end time specified in the
   Range header.  If the pause point is located later than the given end
   value, a 457 (Invalid Range) response MUST be given.

   The example below will play seconds 10 through 25.  It also requests
   the server to deliver media from the first Random Access Point prior
   to the indicated start point.

     C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0
           CSeq: 835
           Session: 12345678
           Range: npt=10-25
           Seek-Style: RAP
           User-Agent: PhonyClient/1.2

   Servers MUST include a "Range" header in any PLAY response, even if
   no Range header was present in the request.  The response MUST use
   the same format as the request's range header contained.  If no Range
   header was in the request, the format used in any previous PLAY
   request within the session SHOULD be used.  If no format has been
   indicated in a previous request the server MAY use any time format
   supported by the media and indicated in the Accept-Ranges header in
   the SETUP request.  It is RECOMMENDED that NPT is used if supported
   by the media.

   For any error response to a PLAY request, the server's response
   depends on the current session state.  If the session is in Ready
   state, the current pause-point is returned using Range header with
   the pause point as the explicit start-point and an implicit stop-
   point.  For time-progressing content where the pause-point moves with
   real-time due to limited retention, the current pause point is
   returned.  For sessions in Play state, the current playout point and
   the remaining parts of the range request is returned.  For any media
   with retention longer than 0 seconds the currently valid Media-Range
   header SHALL also be included in the response.

   A PLAY response MAY include a header carrying synchronization
   information.  As the information necessary is dependent on the media



Schulzrinne, et al.      Expires August 14, 2014               [Page 72]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   transport format, further rules specifying the header and its usage
   are needed.  For RTP the RTP-Info header is specified, see
   Section 18.45, and used in the following example.

   Here is a simple example for a single audio stream where the client
   requests the media starting from 3.52 seconds and to the end.  The
   server sends a 200 OK response with the actual play time which is 10
   ms prior (3.51) and the RTP-Info header that contains the necessary
   parameters for the RTP stack.

   C->S: PLAY rtsp://example.com/audio RTSP/2.0
         CSeq: 836
         Session: 12345678
         Range: npt=3.52-
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 836
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer/1.0
         Range: npt=3.51-324.39
         Seek-Style: First-Prior
         RTP-Info:url="rtsp://example.com/audio"
            ssrc=0D12F123:seq=14783;rtptime=2345962545

   S->C: RTP Packet TS=2345962545 => NPT=3.51
         Media duration=0.16 seconds

   The server replies with the actual start point that will be
   delivered.  This may differ from the requested range if alignment of
   the requested range to valid frame boundaries is required for the
   media source.  Note that some media streams in an aggregate may need
   to be delivered from even earlier points.  Also, some media formats
   have a very long duration per individual data unit, therefore it
   might be necessary for the client to parse the data unit, and select
   where to start.  The server SHALL also indicate which policy it uses
   for selecting the actual start point by including a Seek-Style
   header.

   In the following example the client receives the first media packet
   that stretches all the way up and past the requested playtime.  Thus,
   it is the client's decision whether to render to the user the time
   between 3.52 and 7.05, or to skip it.  In most cases it is probably
   most suitable not to render that time period.







Schulzrinne, et al.      Expires August 14, 2014               [Page 73]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->S: PLAY rtsp://example.com/audio RTSP/2.0
         CSeq: 836
         Session: 12345678
         Range: npt=7.05-
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 836
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer/1.0
         Range: npt=3.52-
         Seek-Style: First-Prior
         RTP-Info:url="rtsp://example.com/audio"
            ssrc=0D12F123:seq=14783;rtptime=2345962545

   S->C: RTP Packet TS=2345962545 => NPT=3.52
         Duration=4.15 seconds

   After playing the desired range, the presentation does NOT change to
   the Ready state, media delivery simply stops.  If it is necessary to
   put the stream into the Ready state, a PAUSE request MUST be issued
   to do that.  A PLAY request while the stream is still in the Play
   state is legal, and can be issued without an intervening PAUSE
   request.  Such a request MUST replace the current PLAY action with
   the new one requested, i.e., being handled in the same way as if as
   the request was received in Ready state.  In the case that the range
   in Range header has an implicit start time ("-endtime"), the server
   MUST continue to play from where it currently was until the specified
   end point.  This is useful to change the end to at another point than
   in the previous request.

   The following example plays the whole presentation starting at SMPTE
   time code 0:10:20 until the end of the clip.  Note: The RTP-Info
   headers has been broken into several lines, where following lines
   start with whitespace as allowed by the syntax.
















Schulzrinne, et al.      Expires August 14, 2014               [Page 74]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->S: PLAY rtsp://audio.example.com/twister.en RTSP/2.0
         CSeq: 833
         Session: 12345678
         Range: smpte=0:10:20-
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 833
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Session: 12345678
         Server: PhonyServer/1.0
         Range: smpte=0:10:22-0:15:45
         Seek-Style: Next
         RTP-Info:url="rtsp://example.com/twister.en"
            ssrc=0D12F123:seq=14783;rtptime=2345962545

   For playing back a recording of a live presentation, it may be
   desirable to use clock units:

   C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/2.0
         CSeq: 835
         Session: 12345678
         Range: clock=19961108T142300Z-19961108T143520Z
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 835
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Session: 12345678
         Server: PhonyServer/1.0
         Range: clock=19961108T142300Z-19961108T143520Z
         Seek-Style: Next
         RTP-Info:url="rtsp://example.com/meeting.en"
            ssrc=0D12F123:seq=53745;rtptime=484589019

13.4.2.  Aggregated Sessions

   PLAY requests can operate on sessions controlling a single media and
   on aggregated sessions controlling multiple media.

   In an aggregated session the PLAY request MUST contain an aggregated
   control URI.  A server MUST respond with error 460 (Only Aggregate
   Operation Allowed) if the client PLAY Request-URI is for a single
   media.  The media in an aggregate MUST be played in sync.  If a
   client wants individual control of the media, it needs to use
   separate RTSP sessions for each media.





Schulzrinne, et al.      Expires August 14, 2014               [Page 75]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   For aggregated sessions where the initial SETUP request (creating a
   session) is followed by one or more additional SETUP requests, a PLAY
   request MAY be pipelined after those additional SETUP requests
   without awaiting their responses.  This procedure can reduce the
   delay from start of session establishment until media play-out has
   started with one round trip time.  However, a client needs to be
   aware that using this procedure will result in the playout of the
   server state established at the time of processing the PLAY, i.e.,
   after the processing of all the requests prior to the PLAY request in
   the pipeline.  This state may not be the intended one due to failure
   of any of the prior requests.  A client can easily determine this
   based on the responses from those requests.  In case of failure, the
   client can halt the media playout using PAUSE and try to establish
   the intended state again before issuing another PLAY request.

13.4.3.  Updating current PLAY Requests

   Clients can issue PLAY requests while the stream is in Play state and
   thus updating their request.

   The important difference compared to a PLAY request in Ready state is
   the handling of the current play point and how the Range header in
   the request is constructed.  The session is actively playing media
   and the play point will be moving, making the exact time a request
   will take effect hard to predict.  Depending on how the PLAY header
   appears two different cases exist: total replacement or continuation.
   A total replacement is signaled by having the first range
   specification have an explicit start value, e.g., "npt=45-" or
   "npt=45-60", in which case the server stops playout at the current
   playout point and then starts delivering media according to the Range
   header.  This is equivalent to having the client first send a PAUSE
   and then a new PLAY request that isn't based on the pause point.  In
   the case of continuation the first range specifier has an implicit
   start point and an explicit stop value (Z), e.g., "npt=-60", which
   indicate that it MUST convert the range specifier being played prior
   to this PLAY request (X to Y) into (X to Z) and continue as this was
   the request originally played.  If the current delivery point is
   beyond the stop point, the server SHALL immediately pause delivery.
   As the request has been completed successfully it shall be responded
   with 200 OK.  A PLAY_NOTIFY with end-of-stream is also sent to
   indicate the actual stop point.  The pause point is set to the
   requested stop point.

   Following is an example of this behavior: The server has received
   requests to play ranges 10 to 15.  If the new PLAY request arrives at
   the server 4 seconds after the previous one, it will take effect
   while the server still plays the first range (10-15).  The server




Schulzrinne, et al.      Expires August 14, 2014               [Page 76]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   changes the current play to continue to 25 seconds, i.e., the
   equivalent single request would be PLAY with "range: npt=10-25".

     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-15
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 834
           Date: Thu, 23 Jan 1997 15:35:06 GMT
           Session: 12345678
           Server: PhonyServer/1.0
           Range: npt=10-15
           Seek-Style: Next
           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934207921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=789DAF12:seq=57654;rtptime=2792482193
           Session: 12345678

     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 835
           Session: 12345678
           Range: npt=-25
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Date: Thu, 23 Jan 1997 15:35:09 GMT
           Session: 12345678
           Server: PhonyServer/1.0
           Range: npt=14-25
           Seek-Style: Next
           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934239921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=789DAF12:seq=57654;rtptime=2792842193

   A common use of a PLAY request while in Play state is changing the
   scale of the media, i.e., entering or leaving fast forward or fast
   rewind.  The client can issue an updating PLAY request that is either
   a continuation or a complete replacement, as discussed above this
   section.  Below is an example of a client that is requesting a fast
   forward (scale=2) without giving a stop point and then change from
   fast forward to regular playout (scale = 1).  In the second PLAY
   request the time is set explicitly to be where ever the server



Schulzrinne, et al.      Expires August 14, 2014               [Page 77]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   currently plays out (npt=now-) and the server responds with the
   actual playback point where the new scale actually takes effect
   (npt=02:17:27.144-).

     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 2034
           Session: 12345678
           Range: npt=now-
           Scale: 2.0
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 2034
           Date: Thu, 23 Jan 1997 15:35:06 GMT
           Session: 12345678
           Server: PhonyServer/1.0
           Range: npt=02:17:21.394-
           Seek-Style: Next
           Scale: 2.0
           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934207921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=789DAF12:seq=57654;rtptime=2792482193


   [playing in fast forward and now returning to scale = 1]

     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 2035
           Session: 12345678
           Range: npt=now-
           Scale: 1.0
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 2035
           Date: Thu, 23 Jan 1997 15:35:09 GMT
           Session: 12345678
           Server: PhonyServer/1.0
           Range: npt=02:17:27.144-
           Seek-Style: Next
           Scale: 1.0
           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934239921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=789DAF12:seq=57654;rtptime=2792842193





Schulzrinne, et al.      Expires August 14, 2014               [Page 78]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


13.4.4.  Playing On-Demand Media

   On-demand media is indicated by the content of the Media-Properties
   header in the SETUP response by (see also Section 18.29):

   o  Random Access property is set to Random-Access;

   o  Content Modifications set to Immutable;

   o  Retention set to Unlimited or Time-Limited.

   Playing on-demand media follows the general usage as described in
   Section 13.4.1.

13.4.5.  Playing Dynamic On-Demand Media

   Dynamic on-demand media is indicated by the content of the Media-
   Properties header in the SETUP response by (see also Section 18.29):

   o  Random Access set to Random-Access;

   o  Content Modifications set to Dynamic;

   o  Retention set to Unlimited or Time-Limited.

   Playing on-demand media follows the general usage as described in
   Section 13.4.1 as long as the media has not been changed.

   There are two ways for the client to be informed about changes of
   media resources in Play state.  The client will receive a PLAY_NOTIFY
   request with Notify-Reason header set to media-properties-update (see
   Section 13.5.2.  The client can use the value of the Media-Range to
   decide further actions, if the Media-Range header is present in the
   PLAY_NOTIFY request.  The second way is that the client issues a
   GET_PARAMETER request without a body but including a Media-Range
   header.  The 200 OK response MUST include the current Media-Range
   header (see Section 18.30).

13.4.6.  Playing Live Media

   Live media is indicated by the content of the Media-Properties header
   in the SETUP response by (see also Section 18.29):

   o  Random-Access set to No-Seeking;

   o  Content Modifications set to Time-Progressing;

   o  Retention with Time-Duration set to 0.0.



Schulzrinne, et al.      Expires August 14, 2014               [Page 79]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   For live media, the SETUP response 200 OK MUST include the Media-
   Range header (see Section 18.30).

   A client MAY send PLAY requests without the Range header.  If the
   request includes the Range header it MUST use a symbolic value
   representing "now".  For NPT that range specification is "npt=now-".
   The server MUST include the Range header in the response and it MUST
   indicate an explicit time value and not a symbolic value.  In other
   words, "npt=now-" is not valid to be used in the response.  Instead
   the time since session start is recommended expressed as an open
   interval, e.g., "npt=96.23-".  An absolute time value (clock) for the
   corresponding time MAY be given, i.e., "clock=20030213T143205Z-".
   The Absolute Time format can only be used if client has shown support
   for it using the Accept-Ranges header.

13.4.7.  Playing Live with Recording

   Certain media servers may offer recording services of live sessions
   to their clients.  This recording would normally be from the
   beginning of the media session.  Clients can randomly access the
   media between now and the beginning of the media session.  This live
   media with recording is indicated by the content of the Media-
   Properties header in the SETUP response by (see also Section 18.29):

   o  Random Access set to Random-Access;

   o  Content Modifications set to Time-Progressing;

   o  Retention set to Time-Limited or Unlimited

   The SETUP response 200 OK MUST include the Media-Range header (see
   Section 18.30) for this type of media.  For live media with
   recording, the Range header indicates the current delivery point in
   the media and the Media-Range header indicates the currently
   available media window around the current time.  This window can
   cover recorded content in the past (seen from current time in the
   media) or recorded content in the future (seen from current time in
   the media).  The server adjusts the delivery point to the requested
   border of the window.  If the client requests a delivery point that
   is located outside the recording window, e.g., if the requested point
   is too far in the past, the server selects the oldest point in the
   recording.  The considerations in Section 13.5.3 apply if a client
   requests delivery with Scale (Section 18.46) values other than 1.0
   (Normal playback rate) while delivering live media with recording.







Schulzrinne, et al.      Expires August 14, 2014               [Page 80]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


13.4.8.  Playing Live with Time-Shift

   Certain media servers may offer time-shift services to their clients.
   This time shift records a fixed interval in the past, i.e., a sliding
   window recording mechanism, but not past this interval.  Clients can
   randomly access the media between now and the interval.  This live
   media with recording is indicated by the content of the Media-
   Properties header in the SETUP response by (see also Section 18.29):

   o  Random Access set to Random-Access;

   o  Content Modifications set to Time-Progressing;

   o  Retention set to Time-Duration and a value indicating the
      recording interval (>0).

   The SETUP response 200 OK MUST include the Media-Range header (see
   Section 18.30) for this type of media.  For live media with recording
   the Range header indicates the current time in the media and the
   Media Range indicates a window around the current time.  This window
   can cover recorded content in the past (seen from current time in the
   media) or recorded content in the future (seen from current time in
   the media).  The server adjusts the play point to the requested
   border of the window, if the client requests a play point that is
   located outside the recording windows, e.g., if requested too far in
   the past, the server selects the oldest range in the recording.  The
   considerations in Section 13.5.3 apply, if a client requests delivery
   using a Scale (Section 18.46) value other than 1.0 (Normal playback
   rate) while delivering live media with time-shift.

13.5.  PLAY_NOTIFY

   The PLAY_NOTIFY method is issued by a server to inform a client about
   an asynchronous event for a session in Play state.  The Session
   header MUST be presented in a PLAY_NOTIFY request and indicates the
   scope of the request.  Sending of PLAY_NOTIFY requests requires a
   persistent connection between server and client, otherwise there is
   no way for the server to send this request method to the client.

   PLAY_NOTIFY requests have an end-to-end (i.e., server to client)
   scope, as they carry the Session header, and apply only to the given
   session.  The client SHOULD immediately return a response to the
   server.

   PLAY_NOTIFY requests MAY use both aggregate control URI and
   individual media resource URIs depending on the scope of the
   notification.  This scope may have important distinctions for
   aggregated sessions, and each reason for a PLAY_NOTIFY request needs



Schulzrinne, et al.      Expires August 14, 2014               [Page 81]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   to specify the interpretation and if aggregated control URIs or
   individual URIs may be used in requests.

   PLAY_NOTIFY requests can be used with a message body, depending on
   the value of the Notify-Reason header.  It is described in the
   particular section for each Notify-Reason if a message body is used.
   However, currently there is no Notify-Reason that allows using a
   message body.  In this case, there is a need to obey some limitations
   when adding new Notify-Reasons that intend to use a message body: the
   server can send any type of message body, but it is not ensured that
   the client can understand the received message body.  This is related
   to DESCRIBE (see Section 13.2 ), but in this particular case the
   client can state its acceptable message bodies by using the Accept
   header.  In the case of PLAY_NOTIFY, the server does not know which
   message bodies are understood by the client.

   The Notify-Reason header (see Section 18.32) specifies the reason why
   the server sends the PLAY_NOTIFY request.  This is extensible and new
   reasons can be added in the future (see Section 22.8).  In case the
   client does not understand the reason for the notification it MUST
   respond with a 465 (Notification Reason Unknown) (Section 17.4.30)
   error code.  Servers can send PLAY_NOTIFY with these types:

   o  end-of-stream (see Section 13.5.1);

   o  media-properties-update (see Section 13.5.2);

   o  scale-change (see Section 13.5.3).

13.5.1.  End-of-Stream

   A PLAY_NOTIFY request with Notify-Reason header set to end-of-stream
   indicates the completion or near completion of the PLAY request and
   the ending delivery of the media stream(s).  The request MUST NOT be
   issued unless the server is in the Play state.  The end of the media
   stream delivery notification may be used to indicate either a
   successful completion of the PLAY request currently being served, or
   to indicate some error resulting in failure to complete the request.
   The Request-Status header (Section 18.42) MUST be included to
   indicate which request the notification is for and its completion
   status.  The message response status codes (Section 8.1.1) are used
   to indicate how the PLAY request concluded.  The sender of a
   PLAY_NOTIFY can issue an updated PLAY_NOTIFY, in the case of a
   PLAY_NOTIFY sent with wrong information.  For instance, a PLAY_NOTIFY
   was issued before reaching the end-of-stream, but some error occurred
   resulting in that the previously sent PLAY_NOTIFY contained a wrong
   time when the stream will end.  In this case a new PLAY_NOTIFY MUST




Schulzrinne, et al.      Expires August 14, 2014               [Page 82]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   be sent including the correct status for the completion and all
   additional information.

   PLAY_NOTIFY requests with Notify-Reason header set to end-of-stream
   MUST include a Range header and the Scale header if the scale value
   is not 1.  The Range header indicates the point in the stream or
   streams where delivery is ending with the timescale that was used by
   the server in the PLAY response for the request being fulfilled.  The
   server MUST NOT use the "now" constant in the Range header; it MUST
   use the actual numeric end position in the proper timescale.  When
   end-of-stream notifications are issued prior to having sent the last
   media packets, this is evident as the end time in the Range header is
   beyond the current time in the media being received by the client,
   e.g., "npt=-15", if npt is currently at 14.2 seconds.  The Scale
   header is to be included so that it is evident if the media time
   scale is moving backwards and/or have a non-default pace.  The end-
   of-stream notification does not prevent the client from sending a new
   PLAY request.

   If RTP is used as media transport, a RTP-Info header MUST be
   included, and the RTP-Info header MUST indicate the last sequence
   number in the seq parameter.

   For an RTSP Session where media resources are under aggregated
   control the media resources will normally end at approximately the
   same time, although some small differences may exist, on the scale of
   a few hundred of milliseconds.  In those cases a RTSP session under
   aggregated control SHOULD send only a single PLAY_NOTIFY request.  By
   using the aggregate control URI in the PLAY_NOTIFY request the RTSP
   server indicates that this applies to all media resources within the
   session.  In cases RTP is used for media delivery corresponding RTP-
   Info needs to be included for all media resources.  In cases where
   one or more media resource has significantly shorter duration than
   some other resources in the aggregated session the server MAY send
   end-of-stream notifications using individual media resource URIs to
   indicate to agents that there will be no more media for this
   particular media resource related to the current active PLAY request.
   In such cases when the remaining media resources comes to end-of-
   stream they MUST send a PLAY_NOTIFY request using the aggregate
   control URI to indicate that no more resources remain.

   A PLAY_NOTIFY request with Notify-Reason header set to end-of-stream
   MUST NOT carry a message body.

   This example request notifies the client about a future end-of-stream
   event:





Schulzrinne, et al.      Expires August 14, 2014               [Page 83]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 854
           Notify-Reason: end-of-stream
           Request-Status: cseq=853 status=200 reason="OK"
           Range: npt=-145
           RTP-Info:url="rtsp://example.com/fizzle/foo/audio"
              ssrc=0D12F123:seq=14783;rtptime=2345962545,
              url="rtsp://example.com/fizzle/video"
              ssrc=789DAF12:seq=57654;rtptime=2792482193

           Session: uZ3ci0K+Ld-M
           Date: Mon, 08 Mar 2010 13:37:16 GMT

     C->S: RTSP/2.0 200 OK
           CSeq: 854
           User-Agent: PhonyClient/1.2
           Session: uZ3ci0K+Ld-M

13.5.2.  Media-Properties-Update

   A PLAY_NOTIFY request with Notify-Reason header set to media-
   properties-update indicates an update of the media properties for the
   given session (see Section 18.29) and/or the available media range
   that can be played as indicated by Media-Range (Section 18.30).
   PLAY_NOTIFY requests with Notify-Reason header set to media-
   properties-update MUST include a Media-Properties and Date header and
   SHOULD include a Media-Range header.  The Media-Properties header has
   session scope, thus for aggregated sessions the PLAY_NOTIFY request
   MUST be using the aggregated control URI.

   This notification MUST be sent for media that are Time-Progressing
   every time an event happens that changes the basis for making
   estimates on how the available for play-back media range will
   progress with wall clock time.  In addition it is RECOMMENDED that
   the server sends these notifications approximately every 5 minutes
   for time-progressing content to ensure the long-term stability of the
   client estimation and allowing for clock skew detection by the
   client.  The time between notifications should be greater than 1
   minute and less than 2 hours.  For the reasons just explained,
   requests MUST include a Media-Range header to provide current Media
   duration and a Range header to indicate the current playing point and
   any remaining parts of the requested range.

      The recommendation for sending updates every 5 minutes is due to
      any clock skew issues.  In 5 minutes the clock skew should not
      become too significant as this is not used for media playback and
      synchronization, only for determining which content is available
      to the user.



Schulzrinne, et al.      Expires August 14, 2014               [Page 84]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   A PLAY_NOTIFY request with Notify-Reason header set to media-
   properties-update MUST NOT carry a message body.

    S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
           Date: Tue, 14 Apr 2008 15:48:06 GMT
           CSeq: 854
           Notify-Reason: media-properties-update
           Session: uZ3ci0K+Ld-M
           Media-Properties: Time-Progressing,
                 Time-Limited=20080415T153919.36Z, Random-Access=5.0
           Media-Range: npt=00:00:00-01:37:21.394
           Range: npt=01:15:49.873-

     C->S: RTSP/2.0 200 OK
           CSeq: 854
           User-Agent: PhonyClient/1.2
           Session: uZ3ci0K+Ld-M

13.5.3.  Scale-Change

   The server may be forced to change the rate of media time per
   playback time when a client requests delivery using a Scale
   (Section 18.46) value other than 1.0 (normal playback rate).  For
   time progressing media with some retention, i.e., the server stores
   already sent content, a client requesting to play with Scale values
   larger than 1 may catch up with the front end of the media.  The
   server will then be unable to continue to provide content at Scale
   larger than 1 as content is only made available by the server at
   Scale=1.  Another case is when Scale < 1 and the media retention is
   time-duration limited.  In this case the delivery point can reach the
   oldest media unit available, and further playback at this scale
   becomes impossible as there will be no media available.  To avoid
   having the client lose any media, the scale will need to be adjusted
   to the same rate at which the media is removed from the storage
   buffer, commonly Scale = 1.0.

   Another case is when the content itself consists of spliced pieces or
   is dynamically updated.  In these cases the server may be required to
   change from one supported scale value (different than Scale=1.0) to
   another.  In this case the server will pick the closest value and
   inform the client of what it has picked.  In these cases the media
   properties will also be sent updating the supported Scale values.
   This enables a client to adjust the Scale value used.

   To minimize impact on playback in any of the above cases the server
   MUST modify the playback properties and set Scale to a supportable
   value and continue delivery of the media.  When doing this
   modification it MUST send a PLAY_NOTIFY message with the Notify-



Schulzrinne, et al.      Expires August 14, 2014               [Page 85]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Reason header set to "scale-change".  The request MUST contain a
   Range header with the media time when the change took effect, a Scale
   header with the new value in use, Session header with the identifier
   for the session it applies to and a Date header with the server
   wallclock time of the change.  For time progressing content also the
   Media-Range and the Media-Properties at this point in time MUST be
   included.  The Media-Properties header MUST be included if the scale
   change was due to the content changing what scale values that is
   supported.

   For media streams being delivered using RTP also a RTP-Info header
   MUST be included.  It MUST contain the rtptime parameter with a value
   corresponding to the point of change in that media and optionally
   also the sequence number.

   PLAY_NOTIFY requests for aggregated sessions MUST use the aggregated
   control URI in the request.  The scale change for any aggregated
   session applies to all media streams part of the aggregate.

   A PLAY_NOTIFY request with Notify-Reason header set to "Scale-Change"
   MUST NOT carry a message body.

     S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
           Date: Tue, 14 Apr 2008 15:48:06 GMT
           CSeq: 854
           Notify-Reason: scale-change
           Session: uZ3ci0K+Ld-M
           Media-Properties: Time-Progressing,
                 Time-Limited=20080415T153919.36Z, Random-Access=5.0
           Media-Range: npt=00:00:00-01:37:21.394
           Range: npt=01:37:21.394-
           Scale: 1
           RTP-Info: url="rtsp://example.com/fizzle/foo/audio"
               ssrc=0D12F123:rtptime=2345962545,
               url="rtsp://example.com/fizzle/videotrack"
               ssrc=789DAF12:seq=57654;rtptime=2792482193

     C->S: RTSP/2.0 200 OK
           CSeq: 854
           User-Agent: PhonyClient/1.2
           Session: uZ3ci0K+Ld-M

13.6.  PAUSE

   The PAUSE request causes the stream delivery to immediately be
   interrupted (halted).  A PAUSE request MUST be done either with the
   aggregated control URI for aggregated sessions, resulting in all
   media being halted, or the media URI for non-aggregated sessions.



Schulzrinne, et al.      Expires August 14, 2014               [Page 86]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Any attempt to do muting of a single media with a PAUSE request in an
   aggregated session MUST be responded to with error 460 (Only
   Aggregate Operation Allowed).  After resuming playback,
   synchronization of the tracks MUST be maintained.  Any server
   resources are kept, though servers MAY close the session and free
   resources after being paused for the duration specified with the
   timeout parameter of the Session header in the SETUP message.

   Example:

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 834
           Date: Thu, 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-75.00

   The PAUSE request causes stream delivery to be interrupted
   immediately on receipt of the message and the pause point is set to
   the current point in the presentation.  That pause point in the media
   stream needs to be maintained.  A subsequent PLAY request without
   Range header resumes from the pause point and plays until media end.

   The pause point after any PAUSE request MUST be returned to the
   client by adding a Range header with what remains unplayed of the
   PLAY request's range.  For media with random access properties, if
   one desires to resume playing a ranged request, one simply includes
   the Range header from the PAUSE response and includes the Seek-Style
   header with the Next policy in the PLAY request.  For media that is
   time-progressing and has retention duration=0 the follow-up PLAY
   request to start media delivery again, MUST use "npt=now-" and not
   the answer given in the response to PAUSE.
















Schulzrinne, et al.      Expires August 14, 2014               [Page 87]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-30
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 834
           Date: Thu, 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer/1.0
           Range: npt=10-30
           Seek-Style: First-Prior
           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934207921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=4FAD8726:seq=57654;rtptime=2792482193
           Session: 12345678

   After 11 seconds, i.e., at 21 seconds into the presentation:
     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 835
           Session: 12345678
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:17 GMT
           Server: PhonyServer/1.0
           Range: npt=21-30
           Session: 12345678

   If a client issues a PAUSE request and the server acknowledges and
   enters the Ready state, the proper server response, if the player
   issues another PAUSE, is still 200 OK.  The 200 OK response MUST
   include the Range header with the current pause point.  See examples
   below:















Schulzrinne, et al.      Expires August 14, 2014               [Page 88]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 834
           Session: 12345678
           Date: Thu, 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-98.36

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 835
           Session: 12345678
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Session: 12345678
           Date: 23 Jan 1997 15:35:07 GMT
           Range: npt=45.76-98.36

13.7.  TEARDOWN

13.7.1.  Client to Server

   The TEARDOWN client to server request stops the stream delivery for
   the given URI, freeing the resources associated with it.  A TEARDOWN
   request can be performed on either an aggregated or a media control
   URI.  However, some restrictions apply depending on the current
   state.  The TEARDOWN request MUST contain a Session header indicating
   what session the request applies to.  The TEARDOWN request MUST NOT
   include a Terminate-Reason header.

   A TEARDOWN using the aggregated control URI or the media URI in a
   session under non-aggregated control (single media session) MAY be
   done in any state (Ready and Play).  A successful request MUST result
   in that media delivery being immediately halted and the session state
   being destroyed.  This MUST be indicated through the lack of a
   Session header in the response.

   A TEARDOWN using a media URI in an aggregated session can only be
   done in Ready state.  Such a request only removes the indicated media
   stream and associated resources from the session.  This may result in
   a session returning to non-aggregated control, because it only
   contains a single media after the request's completion.  A session
   that will exist after the processing of the TEARDOWN request MUST in
   the response to that TEARDOWN request contain a Session header.  Thus



Schulzrinne, et al.      Expires August 14, 2014               [Page 89]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   the presence of the Session header indicates to the receiver of the
   response if the session is still extant or has been removed.

   Example:

     C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 892
           Session: 12345678
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 892
           Server: PhonyServer/1.0

13.7.2.  Server to Client

   The server can send TEARDOWN requests in the server to client
   direction to indicate that the server has been forced to terminate
   the ongoing session.  This may happen for several reasons, such as
   server maintenance without available backup, or that the session has
   been inactive for extended periods of time.  The reason is provided
   in the Terminate-Reason header (Section 18.52).

   When a RTSP client has maintained a RTSP session that otherwise is
   inactive for an extended period of time the server may reclaim the
   resources.  That is done by issuing a TEARDOWN request with the
   Terminate-Reason set to "Session-Timeout".  This MAY be done when the
   client has been inactive in the RTSP session for more than one
   Session Timeout period (Section 18.49).  However, the server is
   RECOMMENDED to not perform this operation until an extended period of
   inactivity of 10 times the Session Timeout period has passed.  It is
   up to the operator of the RTSP server to actually configure how long
   this extended period of inactivity is.  An operator should take into
   account when doing this configuration what the served content is and
   what this means for the extended period of inactivity.

   In case the server needs to stop providing service to the established
   sessions and there is no server to point at in a REDIRECT request,
   then TEARDOWN SHALL be used to terminate the session.  This method
   can also be used when non-recoverable internal errors have happened
   and the server has no other option then to terminate the sessions.

   The TEARDOWN request MUST be done only on the session aggregate
   control URI (i.e., it is not allowed to terminate individual media
   streams, if it is a session aggregate) and MUST include the following
   headers; Session and Terminate-Reason headers.  The request only
   applies to the session identified in the Session header.  The server




Schulzrinne, et al.      Expires August 14, 2014               [Page 90]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   may include a message to the client's user with the "user-msg"
   parameter.

   The TEARDOWN request may alternatively be done on the wild card URI *
   and without any session header.  The scope of such a request is
   limited to the next-hop (i.e., the RTSP agent in direct communication
   with the server) and applies, as well, to the RTSP connection between
   the next-hop RTSP agent and the server.  This request indicates that
   all sessions and pending requests being managed via the connection
   are terminated.  Any intervening proxies SHOULD do all of the
   following in the order listed:

   1.  respond to the TEARDOWN request

   2.  disconnect the control channel from the requesting server

   3.  pass the TEARDOWN request to each applicable client (typically
       those clients with an active session or an unanswered request)

      Note: The proxy is responsible for accepting TEARDOWN responses
      from its clients; these responses MUST NOT be passed on to either
      the original server or the target server in the redirect.

13.8.  GET_PARAMETER

   The GET_PARAMETER request retrieves the value of any specified
   parameter or parameters for a presentation or stream specified in the
   URI.  If the Session header is present in a request, the value of a
   parameter MUST be retrieved in the specified session context.  There
   are two ways of specifying the parameters to be retrieved.

   The first is by including headers which have been defined such that
   you can use them for this purpose.  Headers for this purpose should
   allow empty, or stripped value parts to avoid having to specify bogus
   data when indicating the desire to retrieve a value.  The successful
   completion of the request should also be evident from any filled out
   values in the response.  The headers in this specification that MAY
   be used for retrieving their current value using GET_PARAMETER are
   listed below; additional headers MAY be specified in the future:

   o  Accept-Ranges

   o  Media-Range

   o  Media-Properties

   o  Range




Schulzrinne, et al.      Expires August 14, 2014               [Page 91]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  RTP-Info

   The other way is to specify a message body that lists the
   parameter(s) that are desired to be retrieved.  The Content-Type
   header (Section 18.19) is used to specify which format the message
   body has.  If the receiver of the request does not support the media
   type used for the message body, it SHALL respond using the error code
   415 (Unsupported Media Type).  The responder to a GET_PARAMETER
   request MUST use the media type of the request for the response.  For
   additional considerations regarding message body negotiation see
   Section 9.3.

   RTSP Agents implementing support for responding to GET_PARAMETER
   requests SHALL implement the text/parameters format (Appendix F).
   This to ensure that at least one known format for parameters is
   implemented and thus prevent parameter format negotiation failure.

   Parameters specified within the body of the message must all be
   understood by the request receiving agent.  If one or more parameters
   are not understood a 451 (Parameter Not Understood) MUST be sent
   including a body listing these parameters that weren't understood.
   If all parameters are understood their values are filled in and
   returned in the response message body.

   The method can also be used without a message body or any header that
   requests parameters for keep-alive purpose.  The keep-alive timer has
   been updated for any request that is successful, i.e., a 200 OK
   response is received.  Any non-required header present in such a
   request may or may not have been processed.  Normally the presence of
   filled out values in the header will be indication that the header
   has been processed.  However, for cases when this is difficult to
   determine, it is recommended to use a feature-tag and the Require
   header.  For this reason it is usually easier if any parameters to be
   retrieved are sent in the body, rather than using any header.

   Example:















Schulzrinne, et al.      Expires August 14, 2014               [Page 92]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 431
           User-Agent: PhonyClient/1.2
           Session: 12345678
           Content-Length: 26
           Content-Type: text/parameters

           packets_received
           jitter

     C->S: RTSP/2.0 200 OK
           CSeq: 431
           Session: 12345678
           Server: PhonyServer/1.1
           Date: Mon, 08 Mar 2010 13:43:23 GMT
           Content-Length: 38
           Content-Type: text/parameters

           packets_received: 10
           jitter: 0.3838

13.9.  SET_PARAMETER

   This method requests to set the value of a parameter or a set of
   parameters for a presentation or stream specified by the URI.  The
   method MAY also be used without a message body.  It is the
   RECOMMENDED method to be used in a request sent for the sole purpose
   of updating the keep-alive timer.  If this request is successful,
   i.e., a 200 OK response is received, then the keep-alive timer has
   been updated.  Any non-required header present in such a request may
   or may not have been processed.  To allow a client to determine if
   any such header has been processed, it is necessary to use a feature
   tag and the Require header.  Due to this reason it is RECOMMENDED
   that any parameters are sent in the body, rather than using any
   header.

   When using a message body to list the parameter(s) that are desired
   to be set the Content-Type header (Section 18.19) is used to specify
   which format the message body has.  If the receiver of the request is
   not supporting the media type used for the message body, it SHALL
   respond using the error code 415 (Unsupported Media Type).  For
   additional considerations regarding message body negotiation see
   Section 9.3.

   RTSP Agents implementing support for responding to SET_PARAMETER
   requests SHALL implement the text/parameters format (Appendix F).
   This to ensure that at least one known format for parameters is
   implemented and thus prevent parameter format negotiation failure.



Schulzrinne, et al.      Expires August 14, 2014               [Page 93]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   A request is RECOMMENDED to only contain a single parameter to allow
   the client to determine why a particular request failed.  If the
   request contains several parameters, the server MUST only act on the
   request if all of the parameters can be set successfully.  A server
   MUST allow a parameter to be set repeatedly to the same value, but it
   MAY disallow changing parameter values.  If the receiver of the
   request does not understand or cannot locate a parameter, error 451
   (Parameter Not Understood) MUST be used.  When a parameter is not
   allowed to change, the error code is 458 (Parameter Is Read-Only).
   The response body MUST contain only the parameters that have errors.
   Otherwise, a body MUST NOT be returned.  The response body MUST use
   the media type of the request for the response.

   Note: transport parameters for the media stream MUST only be set with
   the SETUP command.

      Restricting setting transport parameters to SETUP is for the
      benefit of firewalls.

      The parameters are split in a fine-grained fashion so that there
      can be more meaningful error indications.  However, it may make
      sense to allow the setting of several parameters if an atomic
      setting is desirable.  Imagine device control where the client
      does not want the camera to pan unless it can also tilt to the
      right angle at the same time.

   Example:

     C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 421
           User-Agent: PhonyClient/1.2
           Session: iixT43KLc
           Date: Mon, 08 Mar 2010 14:45:04 GMT
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff

     S->C: RTSP/2.0 451 Parameter Not Understood
           CSeq: 421
           Session: iixT43KLc
           Server: PhonyServer/1.0
           Date: Mon, 08 Mar 2010 14:44:56 GMT
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff




Schulzrinne, et al.      Expires August 14, 2014               [Page 94]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


13.10.  REDIRECT

   The REDIRECT method is issued by a server to inform a client that the
   service provided will be terminated and where a corresponding service
   can be provided instead.  This may happen for different reasons.  One
   is that the server is being administered such that it must stop
   providing service.  Thus the client is required to connect to another
   server location to access the resource indicated by the Request-URI.

   The REDIRECT request SHALL contain a Terminate-Reason header
   (Section 18.52) to inform the client of the reason for the request.
   Additional parameters related to the reason may also be included.
   The intention here is to allow a server administrator to do a
   controlled shutdown of the RTSP server.  That requires sufficient
   time to inform all entities having associated state with the server
   and for them to perform a controlled migration from this server to a
   fall back server.

   A REDIRECT request with a Session header has end-to-end (i.e., server
   to client) scope and applies only to the given session.  Any
   intervening proxies SHOULD NOT disconnect the control channel while
   there are other remaining end-to-end sessions.  The REQUIRED Location
   header MUST contain a complete absolute URI pointing to the resource
   to which the client SHOULD reconnect.  Specifically, the Location
   MUST NOT contain just the host and port.  A client may receive a
   REDIRECT request with a Session header, if and only if, an end-to-end
   session has been established.

   A client may receive a REDIRECT request without a Session header at
   any time when it has communication or a connection established with a
   server.  The scope of such a request is limited to the next-hop
   (i.e., the RTSP agent in direct communication with the server) and
   applies to all sessions controlled, as well as the connection between
   the next-hop RTSP agent and the server.  A REDIRECT request without a
   Session header indicates that all sessions and pending requests being
   managed via the connection MUST be redirected.  The Location header,
   if included in such a request, SHOULD contain an absolute URI with
   only the host address and the OPTIONAL port number of the server to
   which the RTSP agent SHOULD reconnect.  Any intervening proxies
   SHOULD do all of the following in the order listed:

   1.  respond to the REDIRECT request

   2.  disconnect the control channel from the requesting server

   3.  connect to the server at the given host address





Schulzrinne, et al.      Expires August 14, 2014               [Page 95]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   4.  pass the REDIRECT request to each applicable client (typically
       those clients with an active session or an unanswered request)

      Note: The proxy is responsible for accepting REDIRECT responses
      from its clients; these responses MUST NOT be passed on to either
      the original server or the redirected server.

   When the server lacks any alternative server and needs to terminate a
   session or all sessions the TEARDOWN request SHALL be used instead.

   When no Terminate-Reason "time" parameter is included in a REDIRECT
   request, the client SHALL perform the redirection immediately and
   return a response to the server.  The server shall consider the
   session as terminated and can free any associated state after it
   receives the successful (2xx) response.  The server MAY close the
   signaling connection upon receiving the response and the client
   SHOULD close the signaling connection after sending the 2xx response.
   The exception to this is when the client has several sessions on the
   server being managed by the given signaling connection.  In this
   case, the client SHOULD close the connection when it has received and
   responded to REDIRECT requests for all the sessions managed by the
   signaling connection.

   The Terminate-Reason header "time" parameter MAY be used to indicate
   the wallclock time by when the redirection MUST have taken place.  To
   allow a client to determine that redirect time without being time
   synchronized with the server, the server MUST include a Date header
   in the request.  The client should have terminated the session and
   closed the connection before the redirection time-line terminated.
   The server MAY simply cease to provide service when the deadline time
   has been reached, or it may issue TEARDOWN requests to the remaining
   sessions.

   If the REDIRECT request times out following the rules in Section 10.4
   the server MAY terminate the session or transport connection that
   would be redirected by the request.  This is a safeguard against
   misbehaving clients that refuse to respond to a REDIRECT request.
   Thus, removing any incentive to not acknowledge the reception of a
   REDIRECT request.

   After a REDIRECT request has been processed, a client that wants to
   continue to receive media for the resource identified by the Request-
   URI will have to establish a new session with the designated host.
   If the URI given in the Location header is a valid resource URI, a
   client SHOULD issue a DESCRIBE request for the URI.






Schulzrinne, et al.      Expires August 14, 2014               [Page 96]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      Note: The media resource indicated by the Location header can be
      identical, slightly different or totally different.  This is the
      reason why a new DESCRIBE request SHOULD be issued.

   If the Location header contains only a host address, the client may
   assume that the media on the new server is identical to the media on
   the old server, i.e., all media configuration information from the
   old session is still valid except for the host address.  However, the
   usage of conditional SETUP using MTag identifiers is RECOMMENDED as a
   means to verify the assumption.

   This example request redirects traffic for this session to the new
   server at the given absolute time:

     S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 732
           Location: rtsp://s2.example.com:8001
           Terminate-Reason: Server-Admin ;time=19960213T143205Z
           Session: uZ3ci0K+Ld-M
           Date: Thu, 13 Feb 1996 14:30:43 GMT

     C->S: RTSP/2.0 200 OK
           CSeq: 732
           User-Agent: PhonyClient/1.2
           Session: uZ3ci0K+Ld-M

14.  Embedded (Interleaved) Binary Data

   In order to fulfill certain requirements on the network side, e.g.,
   in conjunction with network address translators that block RTP
   traffic over UDP, it may be necessary to interleave RTSP messages and
   media stream data.  This interleaving should generally be avoided
   unless necessary since it complicates client and server operation and
   imposes additional overhead.  Also, head-of-line blocking may cause
   problems.  Interleaved binary data SHOULD only be used if RTSP is
   carried over TCP.  Interleaved data is not allowed inside RTSP
   messages.

   Stream data such as RTP packets is encapsulated by an ASCII dollar
   sign (36 decimal), followed by a one-octet channel identifier,
   followed by the length of the encapsulated binary data as a binary,
   two-octet unsigned integer in network octet order (Appendix B of
   [RFC0791]).  The stream data follows immediately afterwards, without
   a CRLF, but including the upper-layer protocol headers.  Each $ block
   MUST contain exactly one upper-layer protocol data unit, e.g., one
   RTP packet.





Schulzrinne, et al.      Expires August 14, 2014               [Page 97]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      Note that this mechanism does not support PDUs larger than 65535
      octets, which matches the maximum payload size of regular, non-
      jumbo IPv4 and IPv6 packets.  If the media delivery protocol
      intended to be used has larger PDUs than that, definition of a PDU
      fragmentation mechanism will be required to support embedded
      binary data.

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      | "$" = 36      | Channel ID    | Length in octets              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      : Binary data (Length according to Length field)                :
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

             Figure 1: Embedded Interleaved Binary Data Format

   The channel identifier is defined in the Transport header with the
   interleaved parameter (Section 18.54).

   When the transport choice is RTP, RTCP messages are also interleaved
   by the server over the TCP connection.  The usage of RTCP messages is
   indicated by including an interval containing a second channel in the
   interleaved parameter of the Transport header, see Section 18.54.  If
   RTCP is used, packets MUST be sent on the first available channel
   higher than the RTP channel.  The channels are bi-directional, using
   the same Channel ID in both directions, and therefore RTCP traffic is
   sent on the second channel in both directions.

      RTCP is sometimes needed for synchronization when two or more
      streams are interleaved in such a fashion.  Also, this provides a
      convenient way to tunnel RTP/RTCP packets through the RTSP
      connection (TCP or TCP/TLS) when required by the network
      configuration and transfer them onto UDP when possible.

















Schulzrinne, et al.      Expires August 14, 2014               [Page 98]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


     C->S: SETUP rtsp://example.com/bar.file RTSP/2.0
           CSeq: 2
           Transport: RTP/AVP/TCP;unicast;interleaved=0-1
           Accept-Ranges: npt, smpte, clock
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 2
           Date: Thu, 05 Jun 1997 18:57:18 GMT
           Transport: RTP/AVP/TCP;unicast;interleaved=5-6
           Session: 12345678
           Accept-Ranges: npt
           Media-Properties: Random-Access=0.2, Immutable, Unlimited

     C->S: PLAY rtsp://example.com/bar.file RTSP/2.0
           CSeq: 3
           Session: 12345678
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 3
           Session: 12345678
           Date: Thu, 05 Jun 1997 18:57:19 GMT
           RTP-Info: url="rtsp://example.com/bar.file"
             ssrc=0D12F123:seq=232433;rtptime=972948234
           Range: npt=0-56.8
           Seek-Style: RAP

     S->C: $005{2 octet length}{"length" octets data, w/RTP header}
     S->C: $005{2 octet length}{"length" octets data, w/RTP header}
     S->C: $006{2 octet length}{"length" octets  RTCP packet}

15.  Proxies

   RTSP Proxies are RTSP agents that are located in between a client and
   a server.  A proxy can take on both the role as a client and as
   server depending on what it tries to accomplish.  RTSP proxies use
   two transport layer connections, one from the RTSP client to the RTSP
   proxy and a second from the RTSP proxy to the RTSP server.  Proxies
   are introduced for several different reasons and those listed below
   are often combined.

   Caching Proxy:  This type of proxy is used to reduce the workload on
         servers and connections.  By caching the description and media
         streams, i.e., the presentation, the proxy can serve a client
         with content, but without requesting it from the server once it
         has been cached and has not become stale.  See the caching
         Section 16.  This type of proxy is also expected to understand



Schulzrinne, et al.      Expires August 14, 2014               [Page 99]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         RTSP end-point functionality, i.e., functionality identified in
         the Require header in addition to what Proxy-Require demands.

   Translator Proxy:  This type of proxy is used to ensure that an RTSP
         client gets access to servers and content on an external
         network or using content encodings not supported by the client.
         The proxy performs the necessary translation of addresses,
         protocols or encodings.  This type of proxy is expected to also
         understand RTSP end-point functionality, i.e., functionality
         identified in the Require header in addition to what Proxy-
         Require demands.

   Access Proxy:  This type of proxy is used to ensure that an RTSP
         client gets access to servers on an external network.  Thus
         this proxy is placed on the border between two domains, e.g., a
         private address space and the public Internet.  The proxy
         performs the necessary translation, usually addresses.  This
         type of proxy is required to redirect the media to itself or a
         controlled gateway that performs the translation before the
         media can reach the client.

   Security Proxy:  This type of proxy is used to help facilitate
         security functions around RTSP.  For example when having a
         firewalled network, the security proxy requests that the
         necessary pinholes in the firewall are opened when a client in
         the protected network wants to access media streams on the
         external side.  This proxy can perform its function without
         redirecting the media between the server and client.  However,
         in deployments with private address spaces this proxy is likely
         to be combined with the access proxy.  Anyway, the
         functionality of this proxy is usually closely tied into
         understanding all aspects of the media transport.

   Auditing Proxy:  RTSP proxies can also provide network owners with a
         logging and audit point for RTSP sessions, e.g., for
         corporations that track their employees usage of the network.
         This type of proxy can perform its function without inserting
         itself or any other node in the media transport.  This proxy
         type can also accept unknown methods as it doesn't interfere
         with the clients' requests.

   All types of proxies can also be used when using secured
   communication with TLS as RTSP 2.0 allows the client to approve
   certificate chains used for connection establishment from a proxy,
   see Section 19.3.2.  However, that trust model may not be suitable
   for all types of deployment.  In those cases, the secured sessions do
   by-pass the proxies.




Schulzrinne, et al.      Expires August 14, 2014              [Page 100]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Access proxies SHOULD NOT be used in equipment like NATs and
   firewalls that aren't expected to be regularly maintained, like home
   or small office equipment.  In these cases it is better to use the
   NAT traversal procedures defined for RTSP 2.0
   [I-D.ietf-mmusic-rtsp-nat].  The reason for these recommendations is
   that any extensions of RTSP resulting in new media transport
   protocols or profiles, new parameters, etc. may fail in a proxy that
   isn't maintained.  This would impede RTSP's future development and
   usage.

15.1.  Proxies and Protocol Extensions

   The existence of proxies must always be considered when developing
   new RTSP extensions.  Most types of proxies will need to implement
   any new method to operate correctly in the presence of that
   extension.  New headers can be introduced and will not be blocked by
   older proxies.  However, it is important to consider if this header
   and its function is required to be understood by the proxy or can be
   simply forwarded.  If the header needs to be understood, a feature-
   tag representing the functionality MUST be included in the Proxy-
   Require header.  Below are guidelines for analysis whether the header
   needs to be understood.  The transport header and its parameters are
   extensible which on the other hand requires handling rules for a
   proxy in order to ensure a correct interpretation.

   Whether a proxy needs to understand a header is not easy to
   determine, as they serve a broad variety of functions.  When
   evaluating if a header needs to be understood, one can divide the
   functionality into three main categories:

   Media modifying:  The caching and translator proxies are modifying
      the actual media and therefore need to understand also the request
      directed to the server that affects how the media is rendered.
      Thus, this type of proxy needs to also understand the server side
      functionality.

   Transport modifying:  The access and the security proxy both need to
      understand how the media transport is performed, either for
      opening pinholes or to translate the outer headers, e.g., IP and
      UDP or TCP.

   Non-modifying:  The audit proxy is special in that it does not modify
      the messages in other ways than to insert the Via header.  That
      makes it possible for this type to forward RTSP messages that
      contain different types of unknown methods, headers or header
      parameters.





Schulzrinne, et al.      Expires August 14, 2014              [Page 101]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Based on the above classification, one should evaluate if the new
   functionality requires the Transport modifying type of proxies to
   understand it or not.

15.2.  Multiplexing and Demultiplexing of Messages

   RTSP proxies may have to multiplex multiple RTSP sessions from their
   clients towards RTSP servers.  This requires that RTSP requests from
   multiple clients are multiplexed onto a common connection for
   requests outgoing to an RTSP server and on the way back the responses
   are demultiplexed from the server to per client responses.  On the
   protocol level this requires that request and response messages are
   handled in both ways, requiring that there is a mechanism to
   correlate what request/response pair exchanged between proxy and
   server is mapped to what client (or client request).

   This multiplexing of requests and demultiplexing of responses is done
   by using the CSeq header field.  The proxy has to rewrite the CSeq in
   requests to the server and responses from the server and remember
   what CSeq is mapped to what client.  The proxy also needs to ensure
   that the order of the message related to each client is maintained.
   Section 18.20 is defining the handling of how requests and responses
   are rewritten.

16.  Caching

   In HTTP, request-response pairs are cached.  RTSP differs
   significantly in that respect.  Responses are not cacheable, with the
   exception of the presentation description returned by DESCRIBE.
   (Since the responses for anything but DESCRIBE and GET_PARAMETER do
   not return any data, caching is not really an issue for these
   requests.)  However, it is desirable for the continuous media data,
   typically delivered out-of-band with respect to RTSP, to be cached,
   as well as the session description.

   On receiving a SETUP or PLAY request, a proxy ascertains whether it
   has an up-to-date copy of the continuous media content and its
   description.  It can determine whether the copy is up-to-date by
   issuing a SETUP or DESCRIBE request, respectively, and comparing the
   Last-Modified header with that of the cached copy.  If the copy is
   not up-to-date, it modifies the SETUP transport parameters as
   appropriate and forwards the request to the origin server.
   Subsequent control commands such as PLAY or PAUSE then pass the proxy
   unmodified.  The proxy delivers the continuous media data to the
   client, while possibly making a local copy for later reuse.  The
   exact allowed behavior of the cache is given by the cache-response
   directives described in Section 18.11.  A cache MUST answer any
   DESCRIBE requests if it is currently serving the stream to the



Schulzrinne, et al.      Expires August 14, 2014              [Page 102]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   requester, as it is possible that low-level details of the stream
   description may have changed on the origin-server.

   Note that an RTSP cache, is of the "cut-through" variety.  Rather
   than retrieving the whole resource from the origin server, the cache
   simply copies the streaming data as it passes by on its way to the
   client.  Thus, it does not introduce additional latency.

   To the client, an RTSP proxy cache appears like a regular media
   server.  To the media origin server an RTSP proxy cache appears like
   a client.  Just as an HTTP cache has to store the content type,
   content language, and so on for the objects it caches, a media cache
   has to store the presentation description.  Typically, a cache
   eliminates all transport-references (e.g., multicast information)
   from the presentation description, since these are independent of the
   data delivery from the cache to the client.  Information on the
   encodings remains the same.  If the cache is able to translate the
   cached media data, it would create a new presentation description
   with all the encoding possibilities it can offer.

16.1.  Validation Model

   When a cache has a stale entry that it would like to use as a
   response to a client's request, it first has to check with the origin
   server (or possibly an intermediate cache with a fresh response) to
   see if its cached entry is still usable.  This is called "validating"
   the cache entry.  To avoid having to pay the overhead of
   retransmitting the full response if the cached entry is good, and at
   the same time avoiding to pay the overhead of an extra round trip if
   the cached entry is invalid, the RTSP protocol supports the use of
   conditional methods.

   The key protocol features for supporting conditional methods are
   those concerned with "cache validators."  When an origin server
   generates a full response, it attaches some sort of validator to it,
   which is kept with the cache entry.  When a client (user agent or
   proxy cache) makes a conditional request for a resource for which it
   has a cache entry, it includes the associated validator in the
   request.

   The server then checks that validator against the current validator
   for the requested resource, and, if they match (see Section 16.1.3),
   it responds with a special status code (usually, 304 (Not Modified))
   and no message body.  Otherwise, it returns a full response
   (including message body).  Thus, avoiding transmitting the full
   response if the validator matches, and avoiding an extra round trip
   if it does not match.




Schulzrinne, et al.      Expires August 14, 2014              [Page 103]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   In RTSP, a conditional request looks exactly the same as a normal
   request for the same resource, except that it carries a special
   header (which includes the validator) that implicitly turns the
   method (usually DESCRIBE or SETUP) into a conditional.

   The protocol includes both positive and negative senses of cache-
   validating conditions.  That is, it is possible to request either
   that a method be performed if and only if a validator matches or if
   and only if no validators match.

      Note: a response that lacks a validator may still be cached, and
      served from cache until it expires, unless this is explicitly
      prohibited by a cache-control directive (see Section 18.11).
      However, a cache cannot do a conditional retrieval if it does not
      have a validator for the resource, which means it will not be
      refreshable after it expires.

   Media streams that are being adapted based on the transport capacity
   between the server and the cache makes caching more difficult.  A
   server needs to consider how it views caching of media streams that
   it adapts and potentially instruct any caches to not cache such
   streams.

16.1.1.  Last-Modified Dates

   The Last-Modified header (Section 18.27) value is often used as a
   cache validator.  In simple terms, a cache entry is considered to be
   valid if the cache entry was created after the Last-Modified time.

16.1.2.  Message Body Tag Cache Validators

   The MTag response-header field value, a message body tag, provides
   for an "opaque" cache validator.  This might allow more reliable
   validation in situations where it is inconvenient to store
   modification dates, where the one-second resolution of RTSP-date
   values is not sufficient, or where the origin server wishes to avoid
   certain paradoxes that might arise from the use of modification
   dates.

   Message body tags are described in Section 4.6

16.1.3.  Weak and Strong Validators

   Since both origin servers and caches will compare two validators to
   decide if they represent the same or different entities, one normally
   would expect that if the message body (i.e., the presentation
   description) or any associated message body headers changes in any
   way, then the associated validator would change as well.  If this is



Schulzrinne, et al.      Expires August 14, 2014              [Page 104]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   true, then this validator is a "strong validator."  The Message body
   (i.e., the presentation description) or any associated message body
   headers is named an entity for a better understanding.

   However, there might be cases when a server prefers to change the
   validator only on semantically significant changes, and not when
   insignificant aspects of the entity change.  A validator that does
   not always change when the resource changes is a "weak validator."

   Message body tags are normally "strong validators," but the protocol
   provides a mechanism to tag a message body tag as "weak."  One can
   think of a strong validator as one that changes whenever the bits of
   an entity changes, while a weak value changes whenever the meaning of
   an entity changes.  Alternatively, one can think of a strong
   validator as part of an identifier for a specific entity, while a
   weak validator is part of an identifier for a set of semantically
   equivalent entities.

      Note: One example of a strong validator is an integer that is
      incremented in stable storage every time an entity is changed.

      An entity's modification time, if represented with one-second
      resolution, could be a weak validator, since it is possible that
      the resource might be modified twice during a single second.

      Support for weak validators is optional.  However, weak validators
      allow for more efficient caching of equivalent objects.

   A "use" of a validator is either when a client generates a request
   and includes the validator in a validating header field, or when a
   server compares two validators.

   Strong validators are usable in any context.  Weak validators are
   only usable in contexts that do not depend on exact equality of an
   entity.  For example, either kind is usable for a conditional
   DESCRIBE of a full entity.  However, only a strong validator is
   usable for a sub-range retrieval, since otherwise the client might
   end up with an internally inconsistent entity.

   Clients MAY issue DESCRIBE requests with either weak validators or
   strong validators.  Clients MUST NOT use weak validators in other
   forms of requests.

   The only function that the RTSP protocol defines on validators is
   comparison.  There are two validator comparison functions, depending
   on whether the comparison context allows the use of weak validators
   or not:




Schulzrinne, et al.      Expires August 14, 2014              [Page 105]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  The strong comparison function: in order to be considered equal,
      both validators MUST be identical in every way, and both MUST NOT
      be weak.

   o  The weak comparison function: in order to be considered equal,
      both validators MUST be identical in every way, but either or both
      of them MAY be tagged as "weak" without affecting the result.

   A message body tag is strong unless it is explicitly tagged as weak.

   A Last-Modified time, when used as a validator in a request, is
   implicitly weak unless it is possible to deduce that it is strong,
   using the following rules:

   o  The validator is being compared by an origin server to the actual
      current validator for the entity and,

   o  That origin server reliably knows that the associated entity did
      not change more than once during the second covered by the
      presented validator.

   OR

   o  The validator is about to be used by a client in an If-Modified-
      Since, because the client has a cache entry for the associated
      entity, and

   o  That cache entry includes a Date value, which gives the time when
      the origin server sent the original response, and

   o  The presented Last-Modified time is at least 60 seconds before the
      Date value.

   OR

   o  The validator is being compared by an intermediate cache to the
      validator stored in its cache entry for the entity, and

   o  That cache entry includes a Date value, which gives the time when
      the origin server sent the original response, and

   o  The presented Last-Modified time is at least 60 seconds before the
      Date value.

   This method relies on the fact that if two different responses were
   sent by the origin server during the same second, but both had the
   same Last-Modified time, then at least one of those responses would
   have a Date value equal to its Last-Modified time.  The arbitrary 60-



Schulzrinne, et al.      Expires August 14, 2014              [Page 106]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   second limit guards against the possibility that the Date and Last-
   Modified values are generated from different clocks, or at somewhat
   different times during the preparation of the response.  An
   implementation MAY use a value larger than 60 seconds, if it is
   believed that 60 seconds is too short.

   If a client wishes to perform a sub-range retrieval on a value for
   which it has only a Last-Modified time and no opaque validator, it
   MAY do this only if the Last-Modified time is strong in the sense
   described here.

16.1.4.  Rules for When to Use Message Body Tags and Last-Modified Dates

   This document adopt a set of rules and recommendations for origin
   servers, clients, and caches regarding when various validator types
   ought to be used, and for what purposes.

   RTSP origin servers:

   o  SHOULD send a message body tag validator unless it is not feasible
      to generate one.

   o  MAY send a weak message body tag instead of a strong message body
      tag, if performance considerations support the use of weak message
      body tags, or if it is unfeasible to send a strong message body
      tag.

   o  SHOULD send a Last-Modified value if it is feasible to send one,
      unless the risk of a breakdown in semantic transparency that could
      result from using this date in an If-Modified-Since header would
      lead to serious problems.

   In other words, the preferred behavior for an RTSP origin server is
   to send both a strong message body tag and a Last-Modified value.

   In order to be legal, a strong message body tag MUST change whenever
   the associated entity value changes in any way.  A weak message body
   tag SHOULD change whenever the associated entity changes in a
   semantically significant way.

      Note: in order to provide semantically transparent caching, an
      origin server MUST avoid reusing a specific strong message body
      tag value for two different entities, or reusing a specific weak
      message body tag value for two semantically different entities.
      Cache entries might persist for arbitrarily long periods,
      regardless of expiration times, so it might be inappropriate to
      expect that a cache will never again attempt to validate an entry
      using a validator that it obtained at some point in the past.



Schulzrinne, et al.      Expires August 14, 2014              [Page 107]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   RTSP clients:

   o  If a message body tag has been provided by the origin server, MUST
      use that message body tag in any cache-conditional request (using
      If-Match or If-None-Match).

   o  If only a Last-Modified value has been provided by the origin
      server, SHOULD use that value in non-subrange cache-conditional
      requests (using If-Modified-Since).

   o  If both a message body tag and a Last-Modified value have been
      provided by the origin server, SHOULD use both validators in
      cache-conditional requests.

   An RTSP origin server, upon receiving a conditional request that
   includes both a Last-Modified date (e.g., in an If-Modified-Since
   header) and one or more message body tags (e.g., in an If-Match, If-
   None-Match, or If-Range header field) as cache validators, MUST NOT
   return a response status of 304 (Not Modified) unless doing so is
   consistent with all of the conditional header fields in the request.

      Note: The general principle behind these rules is that RTSP
      servers and clients should transmit as much non-redundant
      information as is available in their responses and requests.  RTSP
      systems receiving this information will make the most conservative
      assumptions about the validators they receive.

16.1.5.  Non-validating Conditionals

   The principle behind message body tags is that only the service
   author knows the semantics of a resource well enough to select an
   appropriate cache validation mechanism, and the specification of any
   validator comparison function more complex than octet-equality would
   open up a can of worms.  Thus, comparisons of any other headers are
   never used for purposes of validating a cache entry.

16.2.  Invalidation After Updates or Deletions

   The effect of certain methods performed on a resource at the origin
   server might cause one or more existing cache entries to become non-
   transparently invalid.  That is, although they might continue to be
   "fresh," they do not accurately reflect what the origin server would
   return for a new request on that resource.

   There is no way for the RTSP protocol to guarantee that all such
   cache entries are marked invalid.  For example, the request that
   caused the change at the origin server might not have gone through




Schulzrinne, et al.      Expires August 14, 2014              [Page 108]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   the proxy where a cache entry is stored.  However, several rules help
   reduce the likelihood of erroneous behavior.

   In this section, the phrase "invalidate an entity" means that the
   cache will either remove all instances of that entity from its
   storage, or will mark these as "invalid" and in need of a mandatory
   revalidation before they can be returned in response to a subsequent
   request.

   Some RTSP methods MUST cause a cache to invalidate an entity.  This
   is either the entity referred to by the Request-URI, or by the
   Location or Content-Location headers (if present).  These methods
   are:

   o  DESCRIBE

   o  SETUP

   In order to prevent denial of service attacks, an invalidation based
   on the URI in a Location or Content-Location header MUST only be
   performed if the host part is the same as in the Request-URI.

   A cache that passes through requests for methods it does not
   understand SHOULD invalidate any entities referred to by the Request-
   URI.

17.  Status Code Definitions

   Where applicable, HTTP status [H10] codes are reused.  See Table 4 in
   Section 8.1 for a listing of which status codes may be returned by
   which requests.  All error messages, 4xx and 5xx MAY return a body
   containing further information about the error.

17.1.  Informational 1xx

17.1.1.  100 Continue

   The client SHOULD continue with its request.  This interim response
   is used to inform the client that the initial part of the request has
   been received and has not yet been rejected by the server.  The
   client SHOULD continue by sending the remainder of the request or, if
   the request has already been completed, ignore this response.  The
   server MUST send a final response after the request has been
   completed.







Schulzrinne, et al.      Expires August 14, 2014              [Page 109]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


17.2.  Success 2xx

   This class of status code indicates that the client's request was
   successfully received, understood, and accepted.

17.2.1.  200 OK

   The request has succeeded.  The information returned with the
   response is dependent on the method used in the request.

17.3.  Redirection 3xx

   The notation "3xx" indicates response codes from 300 to 399 inclusive
   which are meant for redirection.  The response code 304 is excluded,
   as it is not used for redirection and instead the "3rr" notation is
   used.  The 304 response code appears here, rather than a 2xx response
   code which would have been appropriate, this as 304 has been used
   also in RTSP 1.0 [RFC2326].

   Within RTSP, redirection may be used for load balancing or
   redirecting stream requests to a server topologically closer to the
   client.  Mechanisms to determine topological proximity are beyond the
   scope of this specification.

   A 3rr code MAY be used to respond to any request.  The Location
   header MUST be included in any 3rr response.  It is RECOMMENDED that
   they are used if necessary before a session is established, i.e., in
   response to DESCRIBE or SETUP.  However, in cases where a server is
   not able to send a REDIRECT request to the client, the server MAY
   need to resort to using 3rr responses to inform a client with an
   established session about the need for redirecting the session.  If a
   3rr response is received for a request in relation to an established
   session, the client SHOULD send a TEARDOWN request for the session,
   and MAY reestablish the session using the resource indicated by the
   Location.

   If the Location header is used in a response it MUST contain an
   absolute URI pointing out the media resource the client is redirected
   to, the URI MUST NOT only contain the host name.

   In the event that an unknown 3rr status code is received, the agent
   SHOULD behave as if a 302 response code had been received
   (Section 17.3.3).








Schulzrinne, et al.      Expires August 14, 2014              [Page 110]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


17.3.1.  300

   This response code is not used in RTSP 2.0.

17.3.2.  301 Moved Permanently

   The requested resource is moved permanently and resides now at the
   URI given by the Location header.  The user client SHOULD redirect
   automatically to the given URI.  This response MUST NOT contain a
   message-body.  The Location header MUST be included in the response.

17.3.3.  302 Found

   The requested resource resides temporarily at the URI given by the
   Location header.  This response is intended to be used for many types
   of temporary redirects; e.g., load balancing.  It is RECOMMENDED that
   the server set the reason phrase to something more meaningful than
   "Found" in these cases.  The user client SHOULD redirect
   automatically to the given URI.  This response MUST NOT contain a
   message-body.

   This example shows a client being redirected to a different server:

     C->S: SETUP rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 2
           Transport: RTP/AVP/TCP;unicast;interleaved=0-1
           Accept-Ranges: npt, smpte, clock
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 302 Try Other Server
           CSeq: 2
           Location: rtsp://s2.example.com:8001/fizzle/foo

17.3.4.  303 See Other

   This status code MUST NOT be used in RTSP 2.0.  However, it was
   allowed in RTSP 1.0 [RFC2326].

17.3.5.  304 Not Modified

   If the client has performed a conditional DESCRIBE or SETUP (see
   Section 18.25) and the requested resource has not been modified, the
   server SHOULD send a 304 response.  This response MUST NOT contain a
   message-body.

   The response MUST include the following header fields:

   o  Date



Schulzrinne, et al.      Expires August 14, 2014              [Page 111]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  MTag and/or Content-Location, if the header(s) would have been
      sent in a 200 response to the same request.

   o  Expires and Cache-Control if the field-value might differ from
      that sent in any previous response for the same variant.

   This response is independent for the DESCRIBE and SETUP requests.
   That is, a 304 response to DESCRIBE does NOT imply that the resource
   content is unchanged (only the session description) and a 304
   response to SETUP does NOT imply that the resource description is
   unchanged.  The MTag and If-Match headers may be used to link the
   DESCRIBE and SETUP in this manner.

17.3.6.  305 Use Proxy

   The requested resource MUST be accessed through the proxy given by
   the Location field.  The Location field gives the URI of the proxy.
   The recipient is expected to repeat this single request via the
   proxy. 305 responses MUST only be generated by origin servers.

17.4.  Client Error 4xx

17.4.1.  400 Bad Request

   The request could not be understood by the server due to malformed
   syntax.  The client SHOULD NOT repeat the request without
   modifications.  If the request does not have a CSeq header, the
   server MUST NOT include a CSeq in the response.

17.4.2.  401 Unauthorized

   The request requires user authentication.  The response MUST include
   a WWW-Authenticate header (Section 18.58) field containing a
   challenge applicable to the requested resource.  The client MAY
   repeat the request with a suitable Authorization header field.  If
   the request already included Authorization credentials, then the 401
   response indicates that authorization has been refused for those
   credentials.  If the 401 response contains the same challenge as the
   prior response, and the user agent has already attempted
   authentication at least once, then the user SHOULD be presented the
   message body that was given in the response, since that message body
   might include relevant diagnostic information.  HTTP access
   authentication is explained in [RFC2617].








Schulzrinne, et al.      Expires August 14, 2014              [Page 112]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


17.4.3.  402 Payment Required

   This code is reserved for future use.

17.4.4.  403 Forbidden

   The server understood the request, but is refusing to fulfill it.
   Authorization will not help and the request SHOULD NOT be repeated.
   If the server wishes to make public why the request has not been
   fulfilled, it SHOULD describe the reason for the refusal in the
   message body.  If the server does not wish to make this information
   available to the client, the status code 404 (Not Found) can be used
   instead.

17.4.5.  404 Not Found

   The server has not found anything matching the Request-URI.  No
   indication is given of whether the condition is temporary or
   permanent.  The 410 (Gone) status code SHOULD be used if the server
   knows, through some internally configurable mechanism, that an old
   resource is permanently unavailable and has no forwarding address.
   This status code is commonly used when the server does not wish to
   reveal exactly why the request has been refused, or when no other
   response is applicable.

17.4.6.  405 Method Not Allowed

   The method specified in the request is not allowed for the resource
   identified by the Request-URI.  The response MUST include an Allow
   header containing a list of valid methods for the requested resource.
   This status code is also to be used if a request attempts to use a
   method not indicated during SETUP.

17.4.7.  406 Not Acceptable

   The resource identified by the request is only capable of generating
   response message bodies which have content characteristics not
   acceptable according to the Accept headers sent in the request.

   The response SHOULD include a message body containing a list of
   available message body characteristics and location(s) from which the
   user or user agent can choose the one most appropriate.  The message
   body format is specified by the media type given in the Content-Type
   header field.  Depending upon the format and the capabilities of the
   user agent, selection of the most appropriate choice MAY be performed
   automatically.  However, this specification does not define any
   standard for such automatic selection.




Schulzrinne, et al.      Expires August 14, 2014              [Page 113]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   If the response could be unacceptable, a user agent SHOULD
   temporarily stop receipt of more data and query the user for a
   decision on further actions.

17.4.8.  407 Proxy Authentication Required

   This code is similar to 401 (Unauthorized) (Section 17.4.2), but
   indicates that the client must first authenticate itself with the
   proxy.  The proxy MUST return a Proxy-Authenticate header field
   (Section 18.34) containing a challenge applicable to the proxy for
   the requested resource.

17.4.9.  408 Request Timeout

   The client did not produce a request within the time that the server
   was prepared to wait.  The client MAY repeat the request without
   modifications at any later time.

17.4.10.  410 Gone

   The requested resource is no longer available at the server and the
   forwarding address is not known.  This condition is expected to be
   considered permanent.  If the server does not know, or has no
   facility to determine, whether or not the condition is permanent, the
   status code 404 (Not Found) SHOULD be used instead.  This response is
   cacheable unless indicated otherwise.

   The 410 response is primarily intended to assist the task of
   repository maintenance by notifying the recipient that the resource
   is intentionally unavailable and that the server owners desire that
   remote links to that resource be removed.  Such an event is common
   for limited-time, promotional services and for resources belonging to
   individuals no longer working at the server's site.  It is not
   necessary to mark all permanently unavailable resources as "gone" or
   to keep the mark for any length of time -- that is left to the
   discretion of the owner of the server.

17.4.11.  411 Length Required

   This error code is not defined for RTSP.  This as the use of Content-
   Length (Section 18.17) is always required when message bodies are
   included in RTSP messages.

17.4.12.  412 Precondition Failed

   The precondition given in one or more of the 'if-' request-header
   fields evaluated to false when it was tested on the server.  See
   these sections for the 'if-' headers: If-Match Section 18.24, If-



Schulzrinne, et al.      Expires August 14, 2014              [Page 114]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Modified-Since Section 18.25, and If-None-Match Section 18.26.  This
   response code allows the client to place preconditions on the current
   resource meta information (header field data) and thus prevent the
   requested method from being applied to a resource other than the one
   intended.

17.4.13.  413 Request Message Body Too Large

   The server is refusing to process a request because the request
   message body is larger than the server is willing or able to process.
   The server MAY close the connection to prevent the client from
   continuing the request.

   If the condition is temporary, the server SHOULD include a Retry-
   After header field to indicate that it is temporary and after what
   time the client MAY try again.

17.4.14.  414 Request-URI Too Long

   The server is refusing to service the request because the Request-URI
   is longer than the server is willing to interpret.  This rare
   condition is only likely to occur when a client has used a request
   with long query information, when the client has descended into a URI
   "black hole" of redirection (e.g., a redirected URI prefix that
   points to a suffix of itself), or when the server is under attack by
   a client attempting to exploit security holes present in some servers
   using fixed-length buffers for reading or manipulating the Request-
   URI.

17.4.15.  415 Unsupported Media Type

   The server is refusing to service the request because the message
   body of the request is in a format not supported by the requested
   resource for the requested method.

17.4.16.  451 Parameter Not Understood

   The recipient of the request does not support one or more parameters
   contained in the request.  When returning this error message the
   sender SHOULD return a message body containing the offending
   parameter(s).

17.4.17.  452 reserved

   This status code MUST NOT be used in RTSP 2.0.  However, it was
   allowed in RTSP 1.0 [RFC2326].





Schulzrinne, et al.      Expires August 14, 2014              [Page 115]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


17.4.18.  453 Not Enough Bandwidth

   The request was refused because there was insufficient bandwidth.
   This may, for example, be the result of a resource reservation
   failure.

17.4.19.  454 Session Not Found

   The RTSP session identifier in the Session header is missing,
   invalid, or has timed out.

17.4.20.  455 Method Not Valid in This State

   The client or server cannot process this request in its current
   state.  The response MUST contain an Allow header to make error
   recovery possible.

17.4.21.  456 Header Field Not Valid for Resource

   The server could not act on a required request-header.  For example,
   if PLAY contains the Range header field but the stream does not allow
   seeking.  This error message may also be used for specifying when the
   time format in Range is impossible for the resource.  In that case
   the Accept-Ranges header MUST be returned to inform the client of
   which format(s) that are allowed.

17.4.22.  457 Invalid Range

   The Range value given is out of bounds, e.g., beyond the end of the
   presentation.

17.4.23.  458 Parameter Is Read-Only

   The parameter to be set by SET_PARAMETER can be read but not
   modified.  When returning this error message the sender SHOULD return
   a message body containing the offending parameter(s).

17.4.24.  459 Aggregate Operation Not Allowed

   The requested method may not be applied on the URI in question since
   it is an aggregate (presentation) URI.  The method may be applied on
   a media URI.

17.4.25.  460 Only Aggregate Operation Allowed

   The requested method may not be applied on the URI in question since
   it is not an aggregate control (presentation) URI.  The method may be
   applied on the aggregate control URI.



Schulzrinne, et al.      Expires August 14, 2014              [Page 116]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


17.4.26.  461 Unsupported Transport

   The Transport field did not contain a supported transport
   specification.

17.4.27.  462 Destination Unreachable

   The data transmission channel could not be established because the
   client address could not be reached.  This error will most likely be
   the result of a client attempt to place an invalid dest_addr
   parameter in the Transport field.

17.4.28.  463 Destination Prohibited

   The data transmission channel was not established because the server
   prohibited access to the client address.  This error is most likely
   the result of a client attempt to redirect media traffic to another
   destination with a dest_addr parameter in the Transport header.

17.4.29.  464 Data Transport Not Ready Yet

   The data transmission channel to the media destination is not yet
   ready for carrying data.  However, the responding agent still expects
   that the data transmission channel will be established at some point
   in time.  Note, however, that this may result in a permanent failure
   like 462 "Destination Unreachable".

   An example when this error may occur is in the case a client sends a
   PLAY request to a server prior to ensuring that the TCP connections
   negotiated for carrying media data was successfully established (In
   violation of this specification).  The server would use this error
   code to indicate that the requested action could not be performed due
   to the failure of completing the connection establishment.

17.4.30.  465 Notification Reason Unknown

   This indicates that the client has received a PLAY_NOTIFY
   (Section 13.5) with a Notify-Reason header (Section 18.32) unknown to
   the client.

17.4.31.  466 Key Management Error

   This indicates that there has been an error in a Key Management
   function used in conjunction with a request.  For example usage of
   MIKEY [RFC3830] according to Appendix C.1.4.1 may result in this
   error.





Schulzrinne, et al.      Expires August 14, 2014              [Page 117]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


17.4.32.  470 Connection Authorization Required

   The secured connection attempt needs user or client authorization
   before proceeding.  The next hop's certificate is included in this
   response in the Accept-Credentials header.

17.4.33.  471 Connection Credentials not accepted

   When performing a secure connection over multiple connections, an
   intermediary has refused to connect to the next hop and carry out the
   request due to unacceptable credentials for the used policy.

17.4.34.  472 Failure to establish secure connection

   A proxy fails to establish a secure connection to the next hop RTSP
   agent.  This is primarily caused by a fatal failure at the TLS
   handshake, for example due to server not accepting any cipher suites.

17.5.  Server Error 5xx

   Response status codes beginning with the digit "5" indicate cases in
   which the server is aware that it has erred or is incapable of
   performing the request The server SHOULD include a message body
   containing an explanation of the error situation, and whether it is a
   temporary or permanent condition.  User agents SHOULD display any
   included message body to the user.  These response codes are
   applicable to any request method.

17.5.1.  500 Internal Server Error

   The server encountered an unexpected condition which prevented it
   from fulfilling the request.

17.5.2.  501 Not Implemented

   The server does not support the functionality required to fulfill the
   request.  This is the appropriate response when the server does not
   recognize the request method and is not capable of supporting it for
   any resource.

17.5.3.  502 Bad Gateway

   The server, while acting as a gateway or proxy, received an invalid
   response from the upstream server it accessed in attempting to
   fulfill the request.






Schulzrinne, et al.      Expires August 14, 2014              [Page 118]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


17.5.4.  503 Service Unavailable

   The server is currently unable to handle the request due to a
   temporary overloading or maintenance of the server.  The implication
   is that this is a temporary condition which will be alleviated after
   some delay.  If known, the length of the delay MAY be indicated in a
   Retry-After header.  If no Retry-After is given, the client SHOULD
   handle the response as it would for a 500 response.  The client MUST
   honor the length, if given in the Retry-After header.

         Note: The existence of the 503 status code does not imply that
         a server must use it when becoming overloaded.  Some servers
         may wish to simply refuse the connection.

   The response scope is dependent on the Request.  If the request was
   in relation to an existing RTSP session, the scope of the overload
   response is to this individual RTSP session.  If the request was non-
   session specific or intended to form a RTSP session it applies to the
   RTSP server identified by the host name in the request URI.

17.5.5.  504 Gateway Timeout

   The server, while acting as a proxy, did not receive a timely
   response from the upstream server specified by the URI or some other
   auxiliary server (e.g., DNS) it needed to access in attempting to
   complete the request.

17.5.6.  505 RTSP Version Not Supported

   The server does not support, or refuses to support, the RTSP protocol
   version that was used in the request message.  The server is
   indicating that it is unable or unwilling to complete the request
   using the same major version as the client other than with this error
   message.  The response SHOULD contain a message body describing why
   that version is not supported and what other protocols are supported
   by that server.

17.5.7.  551 Option not supported

   A feature-tag given in the Require or the Proxy-Require fields was
   not supported.  The Unsupported header MUST be returned stating the
   feature for which there is no support.

17.5.8.  553 Proxy Unavailable

   The proxy is currently unable to handle the request due to a
   temporary overloading or maintenance of the proxy.  The implication
   is that this is a temporary condition which will be alleviated after



Schulzrinne, et al.      Expires August 14, 2014              [Page 119]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   some delay.  If known, the length of the delay MAY be indicated in a
   Retry-After header.  If no Retry-After is given, the client SHOULD
   handle the response as it would for a 500 response.  The client MUST
   honor the length, if given in the Retry-After header.

         Note: The existence of the 553 status code does not imply that
         a proxy must use it when becoming overloaded.  Some proxies may
         wish to simply refuse the connection.

   The response scope is dependent on the Request.  If the request was
   in relation to an existing RTSP session, the scope of the overload
   response is to this individual RTSP session.  If the request was non-
   session specific or intended to form a RTSP session it applies to all
   such requests to this proxy.

18.  Header Field Definitions

       +---------------+----------------+--------+---------+------+
       | method        | direction      | object | acronym | Body |
       +---------------+----------------+--------+---------+------+
       | DESCRIBE      | C -> S         | P,S    | DES     | r    |
       |               |                |        |         |      |
       | GET_PARAMETER | C -> S, S -> C | P,S    | GPR     | R,r  |
       |               |                |        |         |      |
       | OPTIONS       | C -> S, S -> C | P,S    | OPT     |      |
       |               |                |        |         |      |
       | PAUSE         | C -> S         | P,S    | PSE     |      |
       |               |                |        |         |      |
       | PLAY          | C -> S         | P,S    | PLY     |      |
       |               |                |        |         |      |
       | PLAY_NOTIFY   | S -> C         | P,S    | PNY     | R    |
       |               |                |        |         |      |
       | REDIRECT      | S -> C         | P,S    | RDR     |      |
       |               |                |        |         |      |
       | SETUP         | C -> S         | S      | STP     |      |
       |               |                |        |         |      |
       | SET_PARAMETER | C -> S, S -> C | P,S    | SPR     | R,r  |
       |               |                |        |         |      |
       | TEARDOWN      | C -> S         | P,S    | TRD     |      |
       |               |                |        |         |      |
       |               | S -> C         | P      | TRD     |      |
       +---------------+----------------+--------+---------+------+

   Table 8: Overview of RTSP methods, their direction, and what objects
   (P: presentation, S: stream) they operate on.  Body notes if a method
       is allowed to carry body and in which direction, R = Request,
    r=response.  Note: All error messages for statuses 4xx and 5xx are
                          allowed to carry a body



Schulzrinne, et al.      Expires August 14, 2014              [Page 120]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   The general syntax for header fields is covered in Section 5.2.  This
   section lists the full set of header fields along with notes on
   meaning, and usage.  The syntax definition for header fields are
   present in Section 20.2.3.  Throughout this section, [HX.Y] is used
   to reference Section X.Y of the HTTP/1.1 specification RFC 2616
   [RFC2616].  Examples of each header field are given.

   Information about header fields in relation to methods and proxy
   processing is summarized in Table 9, Table 10, Table 11, and
   Table 12.

   The "where" column describes the request and response types in which
   the header field can be used.  Values in this column are:

   R:    header field may only appear in requests;

   r:    header field may only appear in responses;

   2xx, 4xx, etc.:  A numerical value or range indicates response codes
         with which the header field can be used;

   c:    header field is copied from the request to the response.

   G:    header field is a general-header and may be present in both
         requests and responses.

   Note: General headers does not always use the "G" value in the where
   column.  This is due to differencies when the header may be applied
   in requests compared to responses.  When such differencies exist they
   are expressed using two differet rows, one with where being "R" and
   one with it being "r".

   The "proxy" column describes the operations a proxy may perform on a
   header field.  An empty proxy column indicates that the proxy MUST
   NOT do any changes to that header, all allowed operations are
   explicitly stated:

   a:    A proxy can add or concatenate the header field if not present.

   m:    A proxy can modify an existing header field value.

   d:    A proxy can delete a header field value.

   r:    A proxy needs to be able to read the header field, and thus
         this header field cannot be encrypted.

   The rest of the columns relate to the presence of a header field in a
   method.  The method names when abbreviated, are according to Table 8:



Schulzrinne, et al.      Expires August 14, 2014              [Page 121]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   c:    Conditional; requirements on the header field depend on the
         context of the message.

   m:    The header field is mandatory.

   m*:   The header field SHOULD be sent, but clients/servers need to be
         prepared to receive messages without that header field.

   o:    The header field is optional.

   *:    The header field MUST be present if the message body is not
         empty.  See Section 18.17, Section 18.19 and Section 5.3 for
         details.

   -:    The header field is not applicable.

   "Optional" means that a Client/Server MAY include the header field in
   a request or response.  The Client/Server behavior when receiving
   such headers varies, for some it may ignore the header field, in
   other cases it is a request to process the header.  This is regulated
   by the method and header descriptions.  Example of headers that
   require processing are the Require and Proxy-Require header fields
   discussed in Section 18.43 and Section 18.37.  A "mandatory" header
   field MUST be present in a request, and MUST be understood by the
   Client/Server receiving the request.  A mandatory response-header
   field MUST be present in the response, and the header field MUST be
   understood by the Client/Server processing the response.  "Not
   applicable" means that the header field MUST NOT be present in a
   request.  If one is placed in a request by mistake, it MUST be
   ignored by the Client/Server receiving the request.  Similarly, a
   header field labeled "not applicable" for a response means that the
   Client/Server MUST NOT place the header field in the response, and
   the Client/Server MUST ignore the header field in the response.

   An RTSP agent MUST ignore extension headers that are not understood.

   The From and Location header fields contain a URI.  If the URI
   contains a comma, or semicolon, the URI MUST be enclosed in double
   quotes (").  Any URI parameters are contained within these quotes.
   If the URI is not enclosed in double quote, any semicolon-delimited
   parameters are header-parameters, not URI parameters.

   +-------------------+------+------+----+----+-----+-----+-----+-----+
   | Header            | Wher | Prox | DE | OP | STP | PLY | PSE | TRD |
   |                   | e    | y    | S  | T  |     |     |     |     |
   +-------------------+------+------+----+----+-----+-----+-----+-----+
   | Accept            | R    |      | o  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |



Schulzrinne, et al.      Expires August 14, 2014              [Page 122]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   | Accept-           | R    | rm   | o  | o  | o   | o   | o   | o   |
   | Credentials       |      |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Accept-Encoding   | R    | r    | o  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Accept-Language   | R    | r    | o  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Accept-Ranges     | G    | r    | -  | -  | m   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Accept-Ranges     | 456  | r    | -  | -  | -   | m   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Allow             | r    | am   | c  | c  | c   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Allow             | 405  | am   | m  | m  | m   | m   | m   | m   |
   |                   |      |      |    |    |     |     |     |     |
   | Authentication-   | r    |      | o  | o  | o   | o   | o   | o/- |
   | Info              |      |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Authorization     | R    |      | o  | o  | o   | o   | o   | o   |
   |                   |      |      |    |    |     |     |     |     |
   | Bandwidth         | R    |      | o  | o  | o   | o   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Blocksize         | R    |      | o  | -  | o   | o   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Cache-Control     | G    | r    | o  | -  | o   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Connection        | G    | ad   | o  | o  | o   | o   | o   | o   |
   |                   |      |      |    |    |     |     |     |     |
   | Connection-       | 470, | ar   | o  | o  | o   | o   | o   | o   |
   | Credentials       | 407  |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Base      | r    |      | o  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Base      | 4xx, |      | o  | o  | o   | o   | o   | o   |
   |                   | 5xx  |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Encoding  | R    | r    | -  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Encoding  | r    | r    | o  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Encoding  | 4xx, | r    | o  | o  | o   | o   | o   | o   |
   |                   | 5xx  |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Language  | R    | r    | -  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Language  | r    | r    | o  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Language  | 4xx, | r    | o  | o  | o   | o   | o   | o   |



Schulzrinne, et al.      Expires August 14, 2014              [Page 123]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   |                   | 5xx  |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Length    | r    | r    | *  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Length    | 4xx, | r    | *  | *  | *   | *   | *   | *   |
   |                   | 5xx  |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Location  | r    | r    | o  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Location  | 4xx, | r    | o  | o  | o   | o   | o   | o   |
   |                   | 5xx  |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Type      | r    | r    | *  | -  | -   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Content-Type      | 4xx, | ar   | *  | *  | *   | *   | *   | *   |
   |                   | 5xx  |      |    |    |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | CSeq              | Gc   | rm   | m  | m  | m   | m   | m   | m   |
   |                   |      |      |    |    |     |     |     |     |
   | Date              | G    | am   | o/ | o/ | o/* | o/* | o/* | o/* |
   |                   |      |      | *  | *  |     |     |     |     |
   |                   |      |      |    |    |     |     |     |     |
   | Expires           | r    | r    | o  | -  | o   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | From              | R    | r    | o  | o  | o   | o   | o   | o   |
   |                   |      |      |    |    |     |     |     |     |
   | If-Match          | R    | r    | -  | -  | o   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | If-Modified-Since | R    | r    | o  | -  | o   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | If-None-Match     | R    | r    | o  | -  | o   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Last-Modified     | r    | r    | o  | -  | o   | -   | -   | -   |
   |                   |      |      |    |    |     |     |     |     |
   | Location          | 3rr  |      | o  | o  | o   | o   | o   | o   |
   +-------------------+------+------+----+----+-----+-----+-----+-----+

     Table 9: Overview of RTSP header fields (A-L) related to methods
           DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.

   +------------------+---------+-----+----+----+----+-----+-----+-----+
   | Header           | Where   | Pro | DE | OP | ST | PLY | PSE | TRD |
   |                  |         | xy  | S  | T  | P  |     |     |     |
   +------------------+---------+-----+----+----+----+-----+-----+-----+
   | Media-           | G       |     | -  | -  | m  | m   | m   | -   |
   | Properties       |         |     |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Media-Range      | G       |     | -  | -  | m  | m   | m   | -   |



Schulzrinne, et al.      Expires August 14, 2014              [Page 124]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   |                  |         |     |    |    |    |     |     |     |
   | MTag             | r       | r   | o  | -  | o  | -   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Pipelined-       | G       | amd | -  | o  | o  | o   | o   | o   |
   | Requests         |         | r   |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy-           | 407     | amr | m  | m  | m  | m   | m   | m   |
   | Authenticate     |         |     |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy-           | r       | amd | o  | o  | o  | o   | o   | o/- |
   | Authentication-  |         | r   |    |    |    |     |     |     |
   | Info             |         |     |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy-           | R       | rd  | o  | o  | o  | o   | o   | o   |
   | Authorization    |         |     |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy- Require   | R       | ar  | o  | o  | o  | o   | o   | o   |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy- Require   | r       | r   | c  | c  | c  | c   | c   | c   |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy- Supported | R       | amr | c  | c  | c  | c   | c   | c   |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy- Supported | r       |     | c  | c  | c  | c   | c   | c   |
   |                  |         |     |    |    |    |     |     |     |
   | Public           | r       | amr | -  | m  | -  | -   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Public           | 501     | amr | m  | m  | m  | m   | m   | m   |
   |                  |         |     |    |    |    |     |     |     |
   | Range            | R       |     | -  | -  | -  | o   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Range            | r       |     | -  | -  | c  | m   | m   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Referrer         | R       |     | o  | o  | o  | o   | o   | o   |
   |                  |         |     |    |    |    |     |     |     |
   | Request- Status  | R       |     | -  | -  | -  | -   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Require          | R       |     | o  | o  | o  | o   | o   | o   |
   |                  |         |     |    |    |    |     |     |     |
   | Retry-After      | 3rr,503 |     | o  | o  | o  | o   | o   | -   |
   |                  | ,553    |     |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Retry-After      | 413     |     | o  | -  | -  | -   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | RTP-Info         | r       |     | -  | -  | c  | c   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Scale            | R       | r   | -  | -  | -  | o   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Scale            | r       | amr | -  | -  | -  | c   | -   | -   |



Schulzrinne, et al.      Expires August 14, 2014              [Page 125]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   |                  |         |     |    |    |    |     |     |     |
   | Seek-Style       | R       |     | -  | -  | -  | o   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Seek-Style       | r       |     | -  | -  | -  | m   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Server           | R       | r   | -  | o  | -  | -   | -   | o   |
   |                  |         |     |    |    |    |     |     |     |
   | Server           | r       | r   | o  | o  | o  | o   | o   | o   |
   |                  |         |     |    |    |    |     |     |     |
   | Session          | R       | r   | -  | o  | o  | m   | m   | m   |
   |                  |         |     |    |    |    |     |     |     |
   | Session          | r       | r   | -  | c  | m  | m   | m   | o   |
   |                  |         |     |    |    |    |     |     |     |
   | Speed            | R       | adm | -  | -  | -  | o   | -   | -   |
   |                  |         | r   |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Speed            | r       | adm | -  | -  | -  | c   | -   | -   |
   |                  |         | r   |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Supported        | R       | amr | o  | o  | o  | o   | o   | o   |
   |                  |         |     |    |    |    |     |     |     |
   | Supported        | r       | amr | c  | c  | c  | c   | c   | c   |
   |                  |         |     |    |    |    |     |     |     |
   | Terminate-Reason | R       | r   | -  | -  | -  | -   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Timestamp        | R       | adm | o  | o  | o  | o   | o   | o   |
   |                  |         | r   |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Timestamp        | c       | adm | m  | m  | m  | m   | m   | m   |
   |                  |         | r   |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Transport        | G       | mr  | -  | -  | m  | -   | -   | -   |
   |                  |         |     |    |    |    |     |     |     |
   | Unsupported      | r       |     | c  | c  | c  | c   | c   | c   |
   |                  |         |     |    |    |    |     |     |     |
   | User-Agent       | R       |     | m* | m* | m* | m*  | m*  | m*  |
   |                  |         |     |    |    |    |     |     |     |
   | Via              | R       | amr | o  | o  | o  | o   | o   | o   |
   |                  |         |     |    |    |    |     |     |     |
   | Via              | c       | dr  | m  | m  | m  | m   | m   | m   |
   |                  |         |     |    |    |    |     |     |     |
   | WWW-             | 401     |     | m  | m  | m  | m   | m   | m   |
   | Authenticate     |         |     |    |    |    |     |     |     |
   +------------------+---------+-----+----+----+----+-----+-----+-----+

     Table 10: Overview of RTSP header fields (M-W) related to methods
           DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.




Schulzrinne, et al.      Expires August 14, 2014              [Page 126]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   +---------------------------+-------+-------+-----+-----+-----+-----+
   | Header                    | Where | Proxy | GPR | SPR | RDR | PNY |
   +---------------------------+-------+-------+-----+-----+-----+-----+
   | Accept                    | R     | arm   | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Accept-Credentials        | R     | rm    | o   | o   | o   | -   |
   |                           |       |       |     |     |     |     |
   | Accept-Encoding           | R     | r     | o   | o   | o   | -   |
   |                           |       |       |     |     |     |     |
   | Accept-Language           | R     | r     | o   | o   | o   | -   |
   |                           |       |       |     |     |     |     |
   | Accept-Ranges             | G     | rm    | o   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Allow                     | 405   | amr   | m   | m   | m   | -   |
   |                           |       |       |     |     |     |     |
   | Authentication-Info       | r     |       | o/- | o/- | -   | -   |
   |                           |       |       |     |     |     |     |
   | Authorization             | R     |       | o   | o   | o   | -   |
   |                           |       |       |     |     |     |     |
   | Bandwidth                 | R     |       | -   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Blocksize                 | R     |       | -   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Cache-Control             | G     | r     | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Connection                | G     |       | o   | o   | o   | o   |
   |                           |       |       |     |     |     |     |
   | Connection-Credentials    | 470,  | ar    | o   | o   | o   | -   |
   |                           | 407   |       |     |     |     |     |
   |                           |       |       |     |     |     |     |
   | Content-Base              | R     |       | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Base              | r     |       | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Base              | 4xx,  |       | o   | o   | o   | o   |
   |                           | 5xx   |       |     |     |     |     |
   |                           |       |       |     |     |     |     |
   | Content-Encoding          | R     | r     | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Encoding          | r     | r     | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Encoding          | 4xx,  | r     | o   | o   | o   | o   |
   |                           | 5xx   |       |     |     |     |     |
   |                           |       |       |     |     |     |     |
   | Content-Language          | R     | r     | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Language          | r     | r     | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |



Schulzrinne, et al.      Expires August 14, 2014              [Page 127]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   | Content-Language          | 4xx,  | r     | o   | o   | o   | o   |
   |                           | 5xx   |       |     |     |     |     |
   |                           |       |       |     |     |     |     |
   | Content-Length            | R     | r     | *   | *   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Length            | r     | r     | *   | *   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Length            | 4xx,  | r     | *   | *   | *   | *   |
   |                           | 5xx   |       |     |     |     |     |
   |                           |       |       |     |     |     |     |
   | Content-Location          | R     |       | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Location          | r     |       | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Location          | 4xx,  |       | o   | o   | o   | o   |
   |                           | 5xx   |       |     |     |     |     |
   |                           |       |       |     |     |     |     |
   | Content-Type              | R     |       | *   | *   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Type              | r     |       | *   | *   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Content-Type              | 4xx,  |       | *   | *   | *   | *   |
   |                           | 5xx   |       |     |     |     |     |
   |                           |       |       |     |     |     |     |
   | CSeq                      | R,c   | mr    | m   | m   | m   | m   |
   |                           |       |       |     |     |     |     |
   | Date                      | R     | a     | o   | o   | m   | o   |
   |                           |       |       |     |     |     |     |
   | Date                      | r     | am    | o   | o   | o   | o   |
   |                           |       |       |     |     |     |     |
   | Expires                   | r     | r     | -   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | From                      | R     | r     | o   | o   | o   | -   |
   |                           |       |       |     |     |     |     |
   | If-Match                  | R     | r     | -   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | If-Modified-Since         | R     | am    | o   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | If-None-Match             | R     | am    | o   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Last-Modified             | R     | r     | -   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Last-Modified             | r     | r     | o   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Location                  | 3rr   |       | o   | o   | o   | -   |
   |                           |       |       |     |     |     |     |
   | Location                  | R     |       | -   | -   | m   | -   |
   |                           |       |       |     |     |     |     |



Schulzrinne, et al.      Expires August 14, 2014              [Page 128]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   | Media-Properties          | R     | amr   | o   | -   | -   | c   |
   |                           |       |       |     |     |     |     |
   | Media-Properties          | r     | mr    | c   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Media-Range               | R     |       | o   | -   | -   | c   |
   |                           |       |       |     |     |     |     |
   | Media-Range               | r     |       | c   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | MTag                      | r     | r     | o   | -   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Notify-Reason             | R     |       | -   | -   | -   | m   |
   |                           |       |       |     |     |     |     |
   | Pipelined-Requests        | R     | amdr  | o   | o   | -   | -   |
   |                           |       |       |     |     |     |     |
   | Proxy-Authenticate        | 407   | amdr  | m   | m   | m   | -   |
   |                           |       |       |     |     |     |     |
   | Proxy-Authentication-Info | r     | amdr  | o/- | o/- | -   | -   |
   |                           |       |       |     |     |     |     |
   | Proxy-Authorization       | R     | amdr  | o   | o   | o   | -   |
   |                           |       |       |     |     |     |     |
   | Proxy-Require             | R     | ar    | o   | o   | o   | -   |
   |                           |       |       |     |     |     |     |
   | Proxy-Supported           | R     | amr   | c   | c   | c   | -   |
   |                           |       |       |     |     |     |     |
   | Proxy-Supported           | r     |       | c   | c   | c   | -   |
   |                           |       |       |     |     |     |     |
   | Public                    | 501   | admr  | m   | m   | m   | -   |
   +---------------------------+-------+-------+-----+-----+-----+-----+

     Table 11: Overview of RTSP header fields (A-P) related to methods
         GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY.

      +------------------+---------+-------+-----+-----+-----+-----+
      | Header           | Where   | Proxy | GPR | SPR | RDR | PNY |
      +------------------+---------+-------+-----+-----+-----+-----+
      | Range            | R       |       | o   | -   | o   | m   |
      |                  |         |       |     |     |     |     |
      | Referrer         | R       |       | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Request-Status   | R       |       | -   | -   | -   | c   |
      |                  |         |       |     |     |     |     |
      | Require          | R       | r     | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Retry-After      | 3rr,503 |       | o   | o   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Retry-After      | 413     |       | o   | o   | -   | -   |
      |                  |         |       |     |     |     |     |
      | RTP-Info         | R       | r     | o   | -   | -   | C   |



Schulzrinne, et al.      Expires August 14, 2014              [Page 129]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      |                  |         |       |     |     |     |     |
      | RTP-Info         | r       | r     | c   | -   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Scale            | G       |       | -   | -   | -   | c   |
      |                  |         |       |     |     |     |     |
      | Seek-Style       | G       |       | -   | -   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Server           | R       | r     | o   | o   | o   | o   |
      |                  |         |       |     |     |     |     |
      | Server           | r       | r     | o   | o   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Session          | R       | r     | o   | o   | o   | m   |
      |                  |         |       |     |     |     |     |
      | Session          | r       | r     | c   | c   | o   | m   |
      |                  |         |       |     |     |     |     |
      | Speed            | G       |       | -   | -   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Supported        | R       | adrm  | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Supported        | r       | adrm  | c   | c   | c   | -   |
      |                  |         |       |     |     |     |     |
      | Terminate-Reason | R       | r     | -   | -   | m   | -   |
      |                  |         |       |     |     |     |     |
      | Timestamp        | R       | adrm  | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Timestamp        | c       | adrm  | m   | m   | m   | -   |
      |                  |         |       |     |     |     |     |
      | Transport        | G       | mr    | -   | -   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Unsupported      | r       | arm   | c   | c   | c   | -   |
      |                  |         |       |     |     |     |     |
      | User-Agent       | R       | r     | m*  | m*  | -   | -   |
      |                  |         |       |     |     |     |     |
      | User-Agent       | r       | r     | m*  | m*  | m*  | m*  |
      |                  |         |       |     |     |     |     |
      | Via              | R       | amr   | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Via              | c       | dr    | m   | m   | m   | -   |
      |                  |         |       |     |     |     |     |
      | WWW-Authenticate | 401     |       | m   | m   | m   | -   |
      +------------------+---------+-------+-----+-----+-----+-----+

     Table 12: Overview of RTSP header fields (R-W) related to methods
         GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY.







Schulzrinne, et al.      Expires August 14, 2014              [Page 130]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


18.1.  Accept

   The Accept request-header field can be used to specify certain
   presentation description and parameter media types [RFC6838] which
   are acceptable for the response to DESCRIBE and GET_PARAMETER
   requests.

   See Section 20.2.3 for the syntax.

   The asterisk "*" character is used to group media types into ranges,
   with "*/*" indicating all media types and "type/*" indicating all
   subtypes of that type.  The media-range MAY include media type
   parameters that are applicable to that range.

   Each media-range MAY be followed by one or more accept-params,
   beginning with the "q" parameter for indicating a relative quality
   factor.  The first "q" parameter (if any) separates the media-range
   parameter(s) from the accept-params.  Quality factors allow the user
   or user agent to indicate the relative degree of preference for that
   media-range, using the qvalue scale from 0 to 1 (section 3.9).  The
   default value is q=1.

   Example of use:

     Accept: application/example ;q=0.7, application/sdp

   Indicates that the requesting agent prefers the media type
   application/sdp through the default 1.0 rating but also accepts the
   application/example media type with a 0.7 quality rating.

   If no Accept header field is present, then it is assumed that the
   client accepts all media types.  If an Accept header field is
   present, and if the server cannot send a response which is acceptable
   according to the combined Accept field value, then the server SHOULD
   send a 406 (not acceptable) response.

18.2.  Accept-Credentials

   The Accept-Credentials header is a request-header used to indicate to
   any trusted intermediary how to handle further secured connections to
   proxies or servers.  See Section 19 for the usage of this header.  It
   MUST NOT be included in server to client requests.

   In a request the header MUST contain the method (User, Proxy, or Any)
   for approving credentials selected by the requester.  The method MUST
   NOT be changed by any proxy, unless it is "Proxy" when a proxy MAY
   change it to "user" to take the role of user approving each further
   hop.  If the method is "User" the header contains zero or more of



Schulzrinne, et al.      Expires August 14, 2014              [Page 131]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   credentials that the client accepts.  The header may contain zero
   credentials in the first RTSP request to a RTSP server when using the
   "User" method.  This is because the client has not yet received any
   credentials to accept.  Each credential MUST consist of one URI
   identifying the proxy or server, the hash algorithm identifier, and
   the hash over that agent's ASN.1 distinguished encoding rules (DER)
   encoded certificate [RFC5280] in BASE64, according to Section 4 of
   [RFC4648] and where the padding bits are set to zero.  All RTSP
   clients and proxies MUST implement the SHA-256[FIPS-pub-180-2]
   algorithm for computation of the hash of the DER encoded certificate.
   The SHA-256 algorithm is identified by the token "sha-256".

   The intention with allowing for other hash algorithms is to enable
   the future retirement of algorithms that are not implemented
   somewhere else than here.  Thus the definition of future algorithms
   for this purpose is intended to be extremely limited.  A feature tag
   can be used to ensure that support for the replacement algorithm
   exists.

   Example:

   Accept-Credentials:User
     "rtsps://proxy2.example.com/";sha-256;exaIl9VMbQMOFGClx5rXnPJKVNI=,
     "rtsps://server.example.com/";sha-256;lurbjj5khhB0NhIuOXtt4bBRH1M=

18.3.  Accept-Encoding

   The Accept-Encoding request-header field is similar to Accept, but
   restricts the content-codings (see Section 18.15),i.e.,
   transformation codings of the message body, such as gzip compression,
   that are acceptable in the response.

   A server tests whether a content-coding is acceptable, according to
   an Accept-Encoding field, using these rules:

   1.  If the content-coding is one of the content-codings listed in the
       Accept-Encoding field, then it is acceptable, unless it is
       accompanied by a qvalue of 0.  (As defined in [H3.9], a qvalue of
       0 means "not acceptable.")

   2.  The special "*" symbol in an Accept-Encoding field matches any
       available content-coding not explicitly listed in the header
       field.

   3.  If multiple content-codings are acceptable, then the acceptable
       content-coding with the highest non-zero qvalue is preferred.





Schulzrinne, et al.      Expires August 14, 2014              [Page 132]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   4.  The "identity" content-coding is always acceptable, i.e., no
       transformation at all, unless specifically refused because the
       Accept-Encoding field includes "identity;q=0", or because the
       field includes "*;q=0" and does not explicitly include the
       "identity" content-coding.  If the Accept-Encoding field-value is
       empty, then only the "identity" encoding is acceptable.

   If an Accept-Encoding field is present in a request, and if the
   server cannot send a response which is acceptable according to the
   Accept-Encoding header, then the server SHOULD send an error response
   with the 406 (Not Acceptable) status code.

   If no Accept-Encoding field is present in a request, the server MAY
   assume that the client will accept any content coding.  In this case,
   if "identity" is one of the available content-codings, then the
   server SHOULD use the "identity" content-coding, unless it has
   additional information that a different content-coding is meaningful
   to the client.

18.4.  Accept-Language

   The Accept-Language request-header field is similar to Accept, but
   restricts the set of natural languages that are preferred as a
   response to the request.  Note that the language specified applies to
   the presentation description and any reason phrases, but not the
   media content.

   A language tag identifies a natural language spoken, written, or
   otherwise conveyed by human beings for communication of information
   to other human beings.  Computer languages are explicitly excluded.
   The syntax and registry of RTSP 2.0 language tags is the same as that
   defined by [RFC5646].

   Each language-range MAY be given an associated quality value which
   represents an estimate of the user's preference for the languages
   specified by that range.  The quality value defaults to "q=1".  For
   example:

      Accept-Language: da, en-gb;q=0.8, en;q=0.7

   would mean: "I prefer Danish, but will accept British English and
   other types of English."  A language-range matches a language-tag if
   it exactly equals the full tag, or if it exactly equals a prefix of
   the tag, i.e., the primary-tag in the ABNF, such that the character
   following primary-tag is "-".  The special range "*", if present in
   the Accept-Language field, matches every tag not matched by any other
   range present in the Accept-Language field.




Schulzrinne, et al.      Expires August 14, 2014              [Page 133]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      Note: This use of a prefix matching rule does not imply that
      language tags are assigned to languages in such a way that it is
      always true that if a user understands a language with a certain
      tag, then this user will also understand all languages with tags
      for which this tag is a prefix.  The prefix rule simply allows the
      use of prefix tags if this is the case.

   In the process of selecting a language, each language-tag is assigned
   a qualification factor, i.e., if a language being supported by the
   client is actually supported by the server and what "preference"
   level the language achieves.  The quality value (q-value) of the
   longest language-range in the field that matches the language-tag is
   assigned as the qualification factor for a particular language-tag.
   If no language-range in the field matches the tag, the language
   qualification factor assigned is 0.  If no Accept-Language header is
   present in the request, the server SHOULD assume that all languages
   are equally acceptable.  If an Accept-Language header is present,
   then all languages which are assigned a qualification factor greater
   than 0 are acceptable.

18.5.  Accept-Ranges

   The Accept-Ranges general-header field allows indication of the
   format supported in the Range header.  The client MUST include the
   header in SETUP requests to indicate which formats are acceptable
   when received in PLAY and PAUSE responses, and REDIRECT requests.
   The server MUST include the header in SETUP and 456 error responses
   to indicate the formats supported for the resource indicated by the
   request URI.  The header MAY be included in GET_PARAMETER request and
   response pairs.  The GET_PARAMETER request MUST contain a Session
   header to identify the session context the request is related to.
   The requester and responder will indicate their capabilities
   regarding Range formats respectively.

      Accept-Ranges: npt, smpte, clock

   The syntax is defined in Section 20.2.3.

18.6.  Allow

   The Allow message-body header field lists the methods supported by
   the resource identified by the Request-URI.  The purpose of this
   field is to inform the recipient of the complete set of valid methods
   associated with the resource.  An Allow header field MUST be present
   in a 405 (Method Not Allowed) response.  The Allow header MUST also
   be present in all OPTIONS responses where the content of the header
   will not include exactly the same methods as listed in the Public
   header.



Schulzrinne, et al.      Expires August 14, 2014              [Page 134]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   The Allow message-body header MUST also be included in SETUP and
   DESCRIBE responses, if the methods allowed for the resource are
   different from the complete set of methods defined in this memo.

   Example of use:

      Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE

18.7.  Authentication-Info

   The Authentication-Info response-header is used by the server to
   communicate some information regarding the successful authentication
   in the response message.  This usage of this header is specified in
   [RFC2617] with some RTSP clarification in Section 19.1.  This header
   MUST only be used in response messages related to client to server
   requests.

18.8.  Authorization

   An RTSP client that wishes to authenticate itself with a server using
   authentication mechanism from HTTP [RFC2617] , usually, but not
   necessarily, after receiving a 401 response, does so by including an
   Authorization request-header field with the request.  The
   Authorization field value consists of credentials containing the
   authentication information of the user agent for the realm of the
   resource being requested.  This header MUST only be used in client to
   server requests.

   If a request is authenticated and a realm specified, the same
   credentials SHOULD be valid for all other requests within this realm
   (assuming that the authentication scheme itself does not require
   otherwise, such as credentials that vary according to a challenge
   value or using synchronized clocks).  Each client to server request
   MUST be individually authorized by including the Authorization header
   with the information.

   When a shared cache (see Section 16) receives a request containing an
   Authorization field, it MUST NOT return the corresponding response as
   a reply to any other request, unless one of the following specific
   exceptions holds:

   1.  If the response includes the "max-age" cache-control directive,
       the cache MAY use that response in replying to a subsequent
       request.  But (if the specified maximum age has passed) a proxy
       cache MUST first revalidate it with the origin server, using the
       request-headers from the new request to allow the origin server
       to authenticate the new request.  (This is the defined behavior




Schulzrinne, et al.      Expires August 14, 2014              [Page 135]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


       for max-age.)  If the response includes "max-age=0", the proxy
       MUST always revalidate it before re-using it.

   2.  If the response includes the "must-revalidate" cache-control
       directive, the cache MAY use that response in replying to a
       subsequent request.  But if the response is stale, all caches
       MUST first revalidate it with the origin server, using the
       request-headers from the new request to allow the origin server
       to authenticate the new request.

   3.  If the response includes the "public" cache-control directive, it
       MAY be returned in reply to any subsequent request.

18.9.  Bandwidth

   The Bandwidth request-header field describes the estimated bandwidth
   available to the client, expressed as a positive integer and measured
   in kilobits per second.  The bandwidth available to the client may
   change during an RTSP session, e.g., due to mobility, congestion,
   etc.

   Clients may not be able to accurately determine the available
   bandwidth, for example because the first hop is not a bottleneck.
   For example most local area networks (LAN) will not be a bottleneck
   if the server is not in the same LAN.  Thus link speeds of WLAN or
   Ethernet networks are normally not a basis for estimating the
   available bandwidth.  Cellular devices or other devices directly
   connected to a modem or connection enabling device may more
   accurately estimate the bottleneck bandwidth and what is a reasonable
   share of it for RTSP controlled media.  The client will also need to
   take into account other traffic sharing the bottleneck.  For example
   by only assigning a certain fraction to RTSP and its media streams.
   It is RECOMMENDED that only clients that have accurate and explicit
   information about bandwidth bottlenecks uses this header.

   This header is not a substitute for proper congestion control.  It is
   only a method providing an initial estimate and coarsely determines
   if the selected content can be delivered at all.

   Example:

     Bandwidth: 62360

18.10.  Blocksize

   The Blocksize request-header field is sent from the client to the
   media server asking the server for a particular media packet size.
   This packet size does not include lower-layer headers such as IP,



Schulzrinne, et al.      Expires August 14, 2014              [Page 136]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   UDP, or RTP.  The server is free to use a blocksize which is lower
   than the one requested.  The server MAY truncate this packet size to
   the closest multiple of the minimum, media-specific block size, or
   override it with the media-specific size if necessary.  The block
   size MUST be a positive decimal number, measured in octets.  The
   server only returns an error (4xx) if the value is syntactically
   invalid.

18.11.  Cache-Control

   The Cache-Control general-header field is used to specify directives
   that MUST be obeyed by all caching mechanisms along the request/
   response chain.

   Cache directives MUST be passed through by a proxy or gateway
   application, regardless of their significance to that application,
   since the directives may be applicable to all recipients along the
   request/response chain.  It is not possible to specify a cache-
   directive for a specific cache.

   Cache-Control should only be specified in a DESCRIBE, GET_PARAMETER,
   SET_PARAMETER and SETUP request and its response.  Note: Cache-
   Control does not govern just the caching of responses as for HTTP,
   instead it also applies to the media stream identified by the SETUP
   request.  The RTSP requests are generally not cacheable, for further
   information see Section 16.  Below are the descriptions of the cache
   directives that can be included in the Cache-Control header.

   no-cache:  Indicates that the media stream or RTSP response MUST NOT
         be cached anywhere.  This allows an origin server to prevent
         caching even by caches that have been configured to return
         stale responses to client requests.  Note, there is no security
         function preventing the caching of content.

   public:  Indicates that the media stream or RTSP response is
         cacheable by any cache.

   private:  Indicates that the media stream or RTSP response is
         intended for a single user and MUST NOT be cached by a shared
         cache.  A private (non-shared) cache may cache the media
         streams.

   no-transform:  An intermediate cache (proxy) may find it useful to
         convert the media type of a certain stream.  A proxy might, for
         example, convert between video formats to save cache space or
         to reduce the amount of traffic on a slow link.  Serious
         operational problems may occur, however, when these
         transformations have been applied to streams intended for



Schulzrinne, et al.      Expires August 14, 2014              [Page 137]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         certain kinds of applications.  For example, applications for
         medical imaging, scientific data analysis and those using end-
         to-end authentication all depend on receiving a stream that is
         bit-for-bit identical to the original media stream or RTSP
         response.  Therefore, if a response includes the no-transform
         directive, an intermediate cache or proxy MUST NOT change the
         encoding of the stream or response.  Unlike HTTP, RTSP does not
         provide for partial transformation at this point, e.g.,
         allowing translation into a different language.

   only-if-cached:  In some cases, such as times of extremely poor
         network connectivity, a client may want a cache to return only
         those media streams or RTSP responses that it currently has
         stored, and not to receive these from the origin server.  To do
         this, the client may include the only-if-cached directive in a
         request.  If it receives this directive, a cache SHOULD either
         respond using a cached media stream or response that is
         consistent with the other constraints of the request, or
         respond with a 504 (Gateway Timeout) status.  However, if a
         group of caches is being operated as a unified system with good
         internal connectivity, such a request MAY be forwarded within
         that group of caches.

   max-stale:  Indicates that the client is willing to accept a media
         stream or RTSP response that has exceeded its expiration time.
         If max-stale is assigned a value, then the client is willing to
         accept a response that has exceeded its expiration time by no
         more than the specified number of seconds.  If no value is
         assigned to max-stale, then the client is willing to accept a
         stale response of any age.

   min-fresh:  Indicates that the client is willing to accept a media
         stream or RTSP response whose freshness lifetime is no less
         than its current age plus the specified time in seconds.  That
         is, the client wants a response that will still be fresh for at
         least the specified number of seconds.

   must-revalidate:  When the must-revalidate directive is present in a
         SETUP response received by a cache, that cache MUST NOT use the
         cache entry after it becomes stale to respond to a subsequent
         request without first revalidating it with the origin server.
         That is, the cache is required to do an end-to-end revalidation
         every time, if, based solely on the origin server's Expires,
         the cached response is stale.

   proxy-revalidate:  The proxy-revalidate directive has the same
         meaning as the must-revalidate directive, except that it does
         not apply to non-shared user agent caches.  It can be used on a



Schulzrinne, et al.      Expires August 14, 2014              [Page 138]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         response to an authenticated request to permit the user's cache
         to store and later return the response without needing to
         revalidate it (since it has already been authenticated once by
         that user), while still requiring proxies that service many
         users to revalidate each time (in order to make sure that each
         user has been authenticated).  Note that such authenticated
         responses also need the public cache control directive in order
         to allow them to be cached at all.

   max-age:  When an intermediate cache is forced, by means of a max-
         age=0 directive, to revalidate its own cache entry, and the
         client has supplied its own validator in the request, the
         supplied validator might differ from the validator currently
         stored with the cache entry.  In this case, the cache MAY use
         either validator in making its own request without affecting
         semantic transparency.

         However, the choice of validator might affect performance.  The
         best approach is for the intermediate cache to use its own
         validator when making its request.  If the server replies with
         304 (Not Modified), then the cache can return its now validated
         copy to the client with a 200 (OK) response.  If the server
         replies with a new message body and cache validator, however,
         the intermediate cache can compare the returned validator with
         the one provided in the client's request, using the strong
         comparison function.  If the client's validator is equal to the
         origin server's, then the intermediate cache simply returns 304
         (Not Modified).  Otherwise, it returns the new message body
         with a 200 (OK) response.

18.12.  Connection

   The Connection general-header field allows the sender to specify
   options that are desired for that particular connection.  It MUST NOT
   be communicated by proxies over further connections.

   RTSP 2.0 proxies MUST parse the Connection header field before a
   message is forwarded and, for each connection-token in this field,
   remove any header field(s) from the message with the same name as the
   connection-token.  Connection options are signaled by the presence of
   a connection-token in the Connection header field, not by any
   corresponding additional header field(s), since the additional header
   field may not be sent if there are no parameters associated with that
   connection option.

   Message headers listed in the Connection header MUST NOT include end-
   to-end headers, such as Cache-Control.




Schulzrinne, et al.      Expires August 14, 2014              [Page 139]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   RTSP 2.0 defines the "close" connection option for the sender to
   signal that the connection will be closed after completion of the
   response.  For example, Connection: close in either the request or
   the response-header fields indicates that the connection SHOULD NOT
   be considered `persistent' (Section 10.2) after the current request/
   response is complete.

   The use of the connection option "close" in RTSP messages SHOULD be
   limited to error messages when the server is unable to recover and
   therefore sees it necessary to close the connection.  The reason is
   that the client has the choice of continuing using a connection
   indefinitely, as long as it sends valid messages.

18.13.  Connection-Credentials

   The Connection-Credentials response-header is used to carry the chain
   of credentials for any next hop that needs to be approved by the
   requester.  It MUST only be used in server to client responses.

   The Connection-Credentials header in an RTSP response MUST, if
   included, contain the credential information (in form of a list of
   certificates providing the chain of certification) of the next hop
   that an intermediary needs to securely connect to.  The header MUST
   include the URI of the next hop (proxy or server) and a BASE64
   (according to Section 4 of [RFC4648] and where the padding bits are
   set to zero) encoded binary structure containing a sequence of DER
   encoded X.509v3 certificates [RFC5280].

   The binary structure starts with the number of certificates
   (NR_CERTS) included as a 16 bit unsigned integer.  This is followed
   by NR_CERTS number of 16 bit unsigned integers providing the size in
   octets of each DER encoded certificate.  This is followed by NR_CERTS
   number of DER encoded X.509v3 certificates in a sequence (chain).
   This format is exemplified in Figure 2.  The proxy or server's
   certificate must come first in the structure.  Each following
   certificate must directly certify the one preceding it.  Because
   certificate validation requires that root keys be distributed
   independently, the self-signed certificate which specifies the root
   certificate authority may optionally be omitted from the chain, under
   the assumption that the remote end must already possess it in order
   to validate it in any case.

   Example:

   Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC...


   Where MIIDNTCC... is a Base64 encoding of the following structure:



Schulzrinne, et al.      Expires August 14, 2014              [Page 140]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


        0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |  Number of certificates       | Size of certificate #1        |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       | Size of certificate #2        | Size of certificate #3        |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       : DER Encoding of Certificate #1                                :
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       : DER Encoding of Certificate #2                                :
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       : DER Encoding of Certificate #3                                :
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Figure 2: Connection-Credentials header's Certificate Format Example

18.14.  Content-Base

   The Content-Base message-body header field may be used to specify the
   base URI for resolving relative URIs within the message body.

   Content-Base: rtsp://media.example.com/movie/twister/

   If no Content-Base field is present, the base URI of an message body
   is defined either by its Content-Location (if that Content-Location
   URI is an absolute URI) or the URI used to initiate the request, in
   that order of precedence.  Note, however, that the base URI of the
   contents within the message-body may be redefined within that
   message-body.

18.15.  Content-Encoding

   The Content-Encoding message-body header field is used as a modifier
   to the media-type.  When present, its value indicates what additional
   content codings have been applied to the message body, and thus what
   decoding mechanisms must be applied in order to obtain the media-type
   referenced by the Content-Type header field.  Content-Encoding is
   primarily used to allow a document to be compressed without losing
   the identity of its underlying media type.

   The content-coding is a characteristic of the message body identified
   by the Request-URI.  Typically, the message body is stored with this
   encoding and is only decoded before rendering or analogous usage.
   However, an RTSP proxy MAY modify the content-coding if the new
   coding is known to be acceptable to the recipient, unless the "no-
   transform" cache-control directive is present in the message.





Schulzrinne, et al.      Expires August 14, 2014              [Page 141]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   If the content-coding of a message body is not "identity", then the
   message MUST include a Content-Encoding Message-body header that
   lists the non-identity content-coding(s) used.

   If the content-coding of a message body in a request message is not
   acceptable to the origin server, the server SHOULD respond with a
   status code of 415 (Unsupported Media Type).

   If multiple encodings have been applied to a message body, the
   content codings MUST be listed in the order in which they were
   applied, first to last from left to right.  Additional information
   about the encoding parameters MAY be provided by other header fields
   not defined by this specification.

18.16.  Content-Language

   The Content-Language message-body header field describes the natural
   language(s) of the intended audience for the enclosed message body.
   Note that this might not be equivalent to all the languages used
   within the message body.

   Language tags are mentioned in Section 18.4.  The primary purpose of
   Content-Language is to allow a user to identify and differentiate
   entities according to the user's own preferred language.  Thus, if
   the body content is intended only for a Danish-literate audience, the
   appropriate field is

      Content-Language: da

   If no Content-Language is specified, the default is that the content
   is intended for all language audiences.  This might mean that the
   sender does not consider it to be specific to any natural language,
   or that the sender does not know for which language it is intended.

   Multiple languages MAY be listed for content that is intended for
   multiple audiences.  For example, a rendition of the "Treaty of
   Waitangi," presented simultaneously in the original Maori and English
   versions, would call for

      Content-Language: mi, en

   However, just because multiple languages are present within a message
   body does not mean that it is intended for multiple linguistic
   audiences.  An example would be a beginner's language primer, such as
   "A First Lesson in Latin," which is clearly intended to be used by an
   English-literate audience.  In this case, the Content-Language would
   properly only include "en".




Schulzrinne, et al.      Expires August 14, 2014              [Page 142]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Content-Language MAY be applied to any media type -- it is not
   limited to textual documents.

18.17.  Content-Length

   The Content-Length message-body header field contains the length of
   the message body of the RTSP message (i.e., after the double CRLF
   following the last header).  Unlike HTTP, it MUST be included in all
   messages that carry a message body beyond the header portion of the
   RTSP message.  If it is missing, a default value of zero is assumed.
   Any Content-Length greater than or equal to zero is a valid value.

18.18.  Content-Location

   The Content-Location message-body header field MAY be used to supply
   the resource location for the message body enclosed in the message
   when that body is accessible from a location separate from the
   requested resource's URI.  A server SHOULD provide a Content-Location
   for the variant corresponding to the response message body;
   especially in the case where a resource has multiple variants
   associated with it, and those entities actually have separate
   locations by which they might be individually accessed, the server
   SHOULD provide a Content-Location for the particular variant which is
   returned.

   As example, if an RTSP client performs a DESCRIBE request on a given
   resource, e.g., "rtsp://a.example.com/movie/Plan9FromOuterSpace",
   then the server may use additional information, such as the User-
   Agent header, to determine the capabilities of the agent.  The server
   will then return a media description tailored to that class of RTSP
   agents.  To indicate which specific description the agent receives
   the resource identifier ("rtsp://a.example.com/movie/
   Plan9FromOuterSpace/FullHD.sdp") is provided in Content-Location,
   while the description is still a valid response for the generic
   resource identifier.  Thus enabling both debugging and cache
   operation as discussed below.

   The Content-Location value is not a replacement for the original
   requested URI; it is only a statement of the location of the resource
   corresponding to this particular variant at the time of the request.
   Future requests MAY specify the Content-Location URI as the request
   URI if the desire is to identify the source of that particular
   variant.  This is useful if the RTSP agent desires to verify if the
   resource variant is current through a conditional request.

   A cache cannot assume that a message body with a Content-Location
   different from the URI used to retrieve it can be used to respond to
   later requests on that Content-Location URI.  However, the Content-



Schulzrinne, et al.      Expires August 14, 2014              [Page 143]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Location can be used to differentiate between multiple variants
   retrieved from a single requested resource.

   If the Content-Location is a relative URI, the relative URI is
   interpreted relative to the Request-URI.

   Note, that Content-Location can be used in some cases to derive the
   base-URI for relative URI(s) present in session description formats.
   This needs to be taken into account when Content-Location is used.
   The easiest way to avoid needing to consider that issue is to include
   the Content-Base whenever the Content-Location is included.

   Note also, when using Media Tags in conjunction with Content-Location
   it is important that the different versions have different MTags,
   even if provided under different Content-Location URIs.  This as they
   have still been provided under the same request URI.

   Note also, as in most cases the URI used in the DESCRIBE and the
   SETUP requests are different, the URI provided in a DESCRIBE Content-
   Location response can't directly be used in a SETUP request.  Instead
   the extra step of resolving URIs combined with the media descriptions
   indication, like with SDP's a=control attribute.

18.19.  Content-Type

   The Content-Type message-body header indicates the media type of the
   message body sent to the recipient.  Note that the content types
   suitable for RTSP are likely to be restricted in practice to
   presentation descriptions and parameter-value types.

18.20.  CSeq

   The CSeq general-header field specifies the sequence number (integer)
   for an RTSP request-response pair.  This field MUST be present in all
   requests and responses.  RTSP agents maintain a sequence number
   series for each responder to which they have an open message
   transport channel.  For each new RTSP request an agent originates on
   a particular RTSP message transport the CSeq value MUST be
   incremented by one.  The initial sequence number can be any number,
   however, it is RECOMMENDED to start at 0.  Each sequence number
   series is unique between each requester and responder, i.e., the
   client has one series for its requests to a server and the server has
   another when sending requests to the client.  Each requester and
   responder is identified by its socket address (IP address and port
   number), i.e., per direction of a TCP connection.  Any retransmitted
   request MUST contain the same sequence number as the original, i.e.,
   the sequence number is not incremented for retransmissions of the
   same request.  The RTSP agent receiving requests MUST process the



Schulzrinne, et al.      Expires August 14, 2014              [Page 144]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   requests arriving on a particular transport in the order of the
   sequence numbers.  Responses are sent in the order that they are
   generated.  The RTSP response MUST have the same sequence number as
   was present in the corresponding request.  A RTSP Agent receiving a
   response MAY receive the responses out of order compared to the order
   of the requests it sent.  Thus, the agent MUST use the sequence
   number in the response to pair it with the corresponding request.

      The main purpose of the sequence number is to map responses to
      requests.

      The requirement to use a sequence number increment of one for each
      new request is to support any future specification of RTSP message
      transport over a protocol that does not provide in order delivery
      or is unreliable.

      The above rules relating to the initial sequence number may appear
      unnecessarily loose.  The reason is to cater for some common
      behavior of existing implementations: When using multiple reliable
      connections in sequence it may still be easiest to use a single
      sequence number series for a client connecting with a particular
      server.  Thus, the initial sequence number may be arbitrary
      depending on the number of previous requests.  For any unreliable
      transport a stricter definition or other solution will be required
      to enable detection of any loss of the first request.

      When using multiple sequential transport connections, there is no
      protocol mechanism to ensure in order processing as the sequence
      number is scoped on the individual transport connection and its
      five tuple.  Thus, there are potential issues with opening a new
      transport connection to the same host for which there already
      exists a transport connection with outstanding requests and
      previously despatched requests related to the same RTSP session.

   RTSP Proxies also need to follow the above rules.  This implies that
   proxies that aggregate requests from multiple clients onto a single
   transport towards a server or a next hop proxy need to renumber these
   requests to form a unified sequence on that transport, fulfilling the
   above rules.  A proxy capable of fulfilling some agent's request
   without emitting its own request (e.g., a caching proxy that fulfils
   a request from its cache), also causes a need to renumber as the
   number of received requests with a particular target, may not be the
   same as the number of emitted requests towards that target agent.  A
   proxy that needs to renumber, needs to perform the corresponding
   renumbering back to the original sequence number for any received
   response before forwarding it back to the originator of the request.





Schulzrinne, et al.      Expires August 14, 2014              [Page 145]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      A client connected to a proxy, and using that transport to send
      requests to multiple servers creates a situation where it is quite
      likely to receive the responses out of order.  This is because the
      proxy will establish separate transports from the proxy to the
      servers on which to forward the client's requests.  When the
      responses arrive from the different servers they will be forwarded
      to the client in the order they arrive at the proxy and can be
      processed, not the order of the client's original sequence
      numbers.  This is intentional to avoid some session's requests
      being blocked by another server's slow processing of requests.

18.21.  Date

   The Date general-header field represents the date and time at which
   the message was originated.  The inclusion of the Date header in RTSP
   message follows these rules:

   o  An RTSP message, sent either by the client or the server,
      containing a body MUST include a Date header, if the sending host
      has a clock;

   o  Clients and servers are RECOMMENDED to include a Date header in
      all other RTSP messages, if the sending host has a clock;

   o  If the server does not have a clock that can provide a reasonable
      approximation of the current time, its responses MUST NOT include
      a Date header field.  In this case, this rule MUST be followed:
      Some origin server implementations might not have a clock
      available.  An origin server without a clock MUST NOT assign
      Expires or Last-Modified values to a response, unless these values
      were associated with the resource by a system or user with a
      reliable clock.  It MAY assign an Expires value that is known, at
      or before server configuration time, to be in the past (this
      allows "pre-expiration" of responses without storing separate
      Expires values for each resource).

   A received message that does not have a Date header field MUST be
   assigned one by the recipient if the message will be cached by that
   recipient.  An RTSP implementation without a clock MUST NOT cache
   responses without revalidating them on every use.  An RTSP cache,
   especially a shared cache, SHOULD use a mechanism, such as Network
   Time Protocol (NTP) [RFC5905], to synchronize its clock with a
   reliable external standard.

   The RTSP-date, a full date as specified by Section 3.3 of [RFC5322],
   sent in a Date header SHOULD NOT represent a date and time subsequent
   to the generation of the message.  It SHOULD represent the best
   available approximation of the date and time of message generation,



Schulzrinne, et al.      Expires August 14, 2014              [Page 146]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   unless the implementation has no means of generating a reasonably
   accurate date and time.  In theory, the date ought to represent the
   moment just before the message body is generated.  In practice, the
   date can be generated at any time during the message origination
   without affecting its semantic value.

      Note: The RTSP 2.0 date format is defined to be the RFC 5322 full
      date format.  This format is more flexible than the RFC 1123 date
      format used by RTSP 1.0.  Thus implementations should use single
      spaces as recommended by RFC 5322 as separators and support
      receiving the obsolete format.

      Further Note that the syntax allow for a comment to be added at
      the end of the date.

      [RFC Editor please remove this note in brackets: Prior to version
      37 of the draft, rfc2326bis envisaged sticking with the RFC 1123
      format.]

18.22.  Expires

   The Expires message-body header field gives a date and time after
   which the description or media-stream should be considered stale.
   The interpretation depends on the method:

   DESCRIBE response:  The Expires header indicates a date and time
         after which the presentation description (body) SHOULD be
         considered stale.

   SETUP response:  The Expires header indicate a date and time after
         which the media stream SHOULD be considered stale.

   A stale cache entry may not normally be returned by a cache (either a
   proxy cache or an user agent cache) unless it is first validated with
   the origin server (or with an intermediate cache that has a fresh
   copy of the message body).  See Section 16 for further discussion of
   the expiration model.

   The presence of an Expires field does not imply that the original
   resource will change or cease to exist at, before, or after that
   time.

   The format is an absolute date and time as defined by RTSP-date.  An
   example of its use is

     Expires: Wed, 23 Jan 2013 15:36:52 +0000





Schulzrinne, et al.      Expires August 14, 2014              [Page 147]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   RTSP/2.0 clients and caches MUST treat other invalid date formats,
   especially including the value "0", as having occurred in the past
   (i.e., already expired).

   To mark a response as "already expired," an origin server should use
   an Expires date that is equal to the Date header value.  To mark a
   response as "never expires," an origin server SHOULD use an Expires
   date approximately one year from the time the response is sent.  RTSP
   /2.0 servers SHOULD NOT send Expires dates more than one year in the
   future.

18.23.  From

   The From request-header field, if given, SHOULD contain an Internet
   e-mail address for the human user who controls the requesting user
   agent.  The address SHOULD be machine-usable, as defined by "mailbox"
   in [RFC1123].

   This header field MAY be used for logging purposes and as a means for
   identifying the source of invalid or unwanted requests.  It SHOULD
   NOT be used as an insecure form of access protection.  The
   interpretation of this field is that the request is being performed
   on behalf of the person given, who accepts responsibility for the
   method performed.  In particular, robot agents SHOULD include this
   header so that the person responsible for running the robot can be
   contacted if problems occur on the receiving end.

   The Internet e-mail address in this field MAY be separate from the
   Internet host which issued the request.  For example, when a request
   is passed through a proxy the original issuer's address SHOULD be
   used.

   The client SHOULD NOT send the From header field without the user's
   approval, as it might conflict with the user's privacy interests or
   their site's security policy.  It is strongly recommended that the
   user be able to disable, enable, and modify the value of this field
   at any time prior to a request.

18.24.  If-Match

   The If-Match request-header field is especially useful for ensuring
   the integrity of the presentation description, independent of how the
   presentation description was received.  The presentation description
   can be fetched via means external to RTSP (such as HTTP) or via the
   DESCRIBE message.  In the case of retrieving the presentation
   description via RTSP, the server implementation is guaranteeing the
   integrity of the description between the time of the DESCRIBE message
   and the SETUP message.  By including the MTag given in or with the



Schulzrinne, et al.      Expires August 14, 2014              [Page 148]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   session description in an If-Match header part of the SETUP request,
   the client ensures that resources set up are matching the
   description.  A SETUP request with the If-Match header for which the
   MTag validation check fails, MUST generate a response using 412
   (Precondition Failed).

   This validation check is also very useful if a session has been
   redirected from one server to another.

18.25.  If-Modified-Since

   The If-Modified-Since request-header field is used with the DESCRIBE
   and SETUP methods to make them conditional.  If the requested variant
   has not been modified since the time specified in this field, a
   description will not be returned from the server (DESCRIBE) or a
   stream will not be set up (SETUP).  Instead, a 304 (Not Modified)
   response MUST be returned without any message-body.

   An example of the field is:

     If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT

18.26.  If-None-Match

   This request-header can be used with one or several message body tags
   to make DESCRIBE requests conditional.  A client that has one or more
   message bodies previously obtained from the resource, can verify that
   none of those entities is current by including a list of their
   associated message body tags in the If-None-Match header field.  The
   purpose of this feature is to allow efficient updates of cached
   information with a minimum amount of transaction overhead.  As a
   special case, the value "*" matches any current entity of the
   resource.

   If any of the message body tags match the message body tag of the
   message body that would have been returned in the response to a
   similar DESCRIBE request (without the If-None-Match header) on that
   resource, or if "*" is given and any current entity exists for that
   resource, then the server MUST NOT perform the requested method,
   unless required to do so because the resource's modification date
   fails to match that supplied in an If-Modified-Since header field in
   the request.  Instead, if the request method was DESCRIBE, the server
   SHOULD respond with a 304 (Not Modified) response, including the
   cache-related header fields (particularly MTag) of one of the message
   bodies that matched.  For all other request methods, the server MUST
   respond with a status of 412 (Precondition Failed).





Schulzrinne, et al.      Expires August 14, 2014              [Page 149]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   See Section 16.1.3 for rules on how to determine if two message body
   tags match.

   If none of the message body tags match, then the server MAY perform
   the requested method as if the If-None-Match header field did not
   exist, but MUST also ignore any If-Modified-Since header field(s) in
   the request.  That is, if no message body tags match, then the server
   MUST NOT return a 304 (Not Modified) response.

   If the request would, without the If-None-Match header field, result
   in anything other than a 2xx or 304 status, then the If-None-Match
   header MUST be ignored.  (See Section 16.1.4 for a discussion of
   server behavior when both If-Modified-Since and If-None-Match appear
   in the same request.)

   The result of a request having both an If-None-Match header field and
   an If-Match header field is unspecified and MUST be considered an
   illegal request.

18.27.  Last-Modified

   The Last-Modified message-body header field indicates the date and
   time at which the origin server believes the presentation description
   or media stream was last modified.  For the method DESCRIBE, the
   header field indicates the last modification date and time of the
   description, for SETUP that of the media stream.

   An origin server MUST NOT send a Last-Modified date which is later
   than the server's time of message origination.  In such cases, where
   the resource's last modification would indicate some time in the
   future, the server MUST replace that date with the message
   origination date.

   An origin server SHOULD obtain the Last-Modified value of the message
   body as close as possible to the time that it generates the Date
   value of its response.  This allows a recipient to make an accurate
   assessment of the message body's modification time, especially if the
   message body changes near the time that the response is generated.

   RTSP servers SHOULD send Last-Modified whenever feasible.

18.28.  Location

   The Location response-header field is used to redirect the recipient
   to a location other than the Request-URI for completion of the
   request or identification of a new resource.  For 3rr responses, the
   location SHOULD indicate the server's preferred URI for automatic




Schulzrinne, et al.      Expires August 14, 2014              [Page 150]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   redirection to the resource.  The field value consists of a single
   absolute URI.

   Note: The Content-Location header field (Section 18.18) differs from
   Location in that the Content-Location identifies the original
   location of the message body enclosed in the request.  It is
   therefore possible for a response to contain header fields for both
   Location and Content-Location.  Also, see Section 16.2 for cache
   requirements of some methods.

18.29.  Media-Properties

   This general-header is used in SETUP response or PLAY_NOTIFY requests
   to indicate the media's properties that currently are applicable to
   the RTSP session.  PLAY_NOTIFY MAY be used to modify these properties
   at any point.  However, the client SHOULD have received the update
   prior to any action related to the new media properties taking
   effect.  For aggregated sessions, the Media-Properties header will be
   returned in each SETUP response.  The header received in the latest
   response is the one that applies on the whole session from this point
   until any future update.  The header MAY be included without value in
   GET_PARAMETER requests to the server with a Session header included
   to query the current Media-Properties for the session.  The responder
   MUST include the current session's media properties.

   The media properties expressed by this header is the one applicable
   to all media in the RTSP session.  For aggregated sessions, the
   header expressed the combined media-properties.  As a result,
   aggregation of media MAY result in a change of the media properties,
   and thus the content of the Media-Properties header contained in
   subsequent SETUP responses.

   The header contains a list of property values that are applicable to
   the currently setup media or aggregate of media as indicated by the
   RTSP URI in the request.  No ordering is enforced within the header.
   Property values should be grouped into a single group that handles a
   particular orthogonal property.  Values or groups that express
   multiple properties SHOULD NOT be used.  The list of properties that
   can be expressed MAY be extended at any time.  Unknown property
   values MUST be ignored.

   This specification defines the following 4 groups and their property
   values:

   Random Access:

      Random-Access:  Indicates that random access is possible.  May
         optionally include a floating point value in seconds indicating



Schulzrinne, et al.      Expires August 14, 2014              [Page 151]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         the longest duration between any two random access points in
         the media.

      Beginning-Only:  Seeking is limited to the beginning only.

      No-Seeking:  No seeking is possible.

   Content Modifications:

      Immutable:  The content will not be changed during the life-time
         of the RTSP session.

      Dynamic:  The content may be changed based on external methods or
         triggers

      Time-Progressing:  The media accessible progresses as wallclock
         time progresses.

   Retention:

      Unlimited:  Content will be retained for the duration of the life-
         time of the RTSP session.

      Time-Limited:  Content will be retained at least until the
         specified wallclock time.  The time must be provided in the
         absolute time format specified in Section 4.4.3.

      Time-Duration:  Each individual media unit is retained for at
         least the specified time duration.  This definition allows for
         retaining data with a time based sliding window.  The time
         duration is expressed as floating point number in seconds. 0.0
         is a valid value as this indicates that no data is retained in
         a time-progressing session.

   Supported Scale:

      Scales:  A quoted comma separated list of one or more decimal
         values or ranges of scale values supported by the content in
         arbitrary order.  A range has a start and stop value separated
         by a colon.  A range indicates that the content supports fine
         grained selection of scale values.  Fine grained allows for
         steps at least as small as one tenth of a scale value.  A
         content is considered to support fine grained selection when
         the server in response to a given scale value can produce
         content with an actual scale that is less than 1 tenth of scale
         unit, i.e., 0.1, from the requested value.  Negative values are
         supported.  The value 0 has no meaning and MUST NOT be used.




Schulzrinne, et al.      Expires August 14, 2014              [Page 152]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Examples of this header for on-demand content and a live stream
   without recording are:

   On-demand:
   Media-Properties: Random-Access=2.5, Unlimited, Immutable,
        Scales="-20, -10, -4, 0.5:1.5, 4, 8, 10, 15, 20"

   Live stream without recording/timeshifting:
   Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0.0

18.30.  Media-Range

   The Media-Range general-header is used to give the range of the media
   at the time of sending the RTSP message.  This header MUST be
   included in SETUP response, and PLAY and PAUSE response for media
   that are Time-Progressing, and PLAY and PAUSE response after any
   change for media that are Dynamic, and in PLAY_NOTIFY request that
   are sent due to Media-Property-Update.  Media-Range header without
   any range specifications MAY be included in GET_PARAMETER requests to
   the server to request the current range.  The server MUST in this
   case include the current range at the time of sending the response.

   The header MUST include range specifications for all time formats
   supported for the media, as indicated in Accept-Ranges header
   (Section 18.5) when setting up the media.  The server MAY include
   more than one range specification of any given time format to
   indicate media that has non-continuous range.  The range
   specifications SHALL be ordered with the range with the lowest value
   or earliest start time first, followed by ranges with increasingly
   higher values or later start time.

   For media that has the Time-Progressing property, the Media-Range
   values will only be valid for the particular point in time when it
   was issued.  As wallclock progresses so will also the media range.
   However, it shall be assumed that media time progresses in direct
   relationship to wallclock time (with the exception of clock skew) so
   that a reasonably accurate estimation of the media range can be
   calculated.

18.31.  MTag

   The MTag response-header MAY be included in DESCRIBE, GET_PARAMETER
   or SETUP responses.  The message body tags (Section 4.6) returned in
   a DESCRIBE response, and the one in SETUP refers to the presentation,
   i.e., both the returned session description and the media stream.
   This allows for verification that one has the right session
   description to a media resource at the time of the SETUP request.




Schulzrinne, et al.      Expires August 14, 2014              [Page 153]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   However, it has the disadvantage that a change in any of the parts
   results in invalidation of all the parts.

   If the MTag is provided both inside the message body, e.g., within
   the "a=mtag" attribute in SDP, and in the response message, then both
   tags MUST be identical.  It is RECOMMENDED that the MTag is primarily
   given in the RTSP response message, to ensure that caches can use the
   MTag without requiring content inspection.  However, for session
   descriptions that are distributed outside of RTSP, for example using
   HTTP, etc. it will be necessary to include the message body tag in
   the session description as specified in Appendix D.1.9.

   SETUP and DESCRIBE requests can be made conditional upon the MTag
   using the headers If-Match (Section 18.24) and If-None-Match (
   Section 18.26).

18.32.  Notify-Reason

   The Notify-Reason response-header is solely used in the PLAY_NOTIFY
   method.  It indicates the reason why the server has sent the
   asynchronous PLAY_NOTIFY request (see Section 13.5).

18.33.  Pipelined-Requests

   The Pipelined-Requests general-header is used to indicate that a
   request is to be executed in the context created by a previous
   request(s).  The primary usage of this header is to allow pipelining
   of SETUP requests so that any additional SETUP request after the
   first one does not need to wait for the session ID to be sent back to
   the requesting agent.  The header contains a unique identifier that
   is scoped by the persistent connection used to send the requests.

   Upon receiving a request with the Pipelined-Requests the responding
   agent MUST look up if there exists a binding between this Pipelined-
   Requests identifier for the current persistent connection and an RTSP
   session ID.  If that exists then the received request is processed
   the same way as if it contained the Session header with the found
   session ID.  If there does not exist a mapping and no Session header
   is included in the request, the responding agent MUST create a
   binding upon the successful completion of a session creating request,
   i.e., SETUP.  A binding MUST NOT be created, if the request failed to
   create an RTSP session.  In case the request contains both a Session
   header and the Pipelined-Requests header the Pipelined-Requests MUST
   be ignored.

   Note: Based on the above definition at least the first request
   containing a new unique Pipelined-Requests will be required to be a
   SETUP request (unless the protocol is extended with new methods of



Schulzrinne, et al.      Expires August 14, 2014              [Page 154]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   creating a session).  After that first one, additional SETUP requests
   or requests of any type using the RTSP session context may include
   the Pipelined-Requests header.

   When responding to any request that contained the Pipelined-Requests
   header the server MUST also include the Session header when a binding
   to a session context exists.  An RTSP agent that knows the session
   identifier SHOULD NOT use the Pipelined-Requests header in any
   request and only use the Session header.  This as the Session
   identifier is persistent across transport contexts, like TCP
   connections, which the Pipelined-Requests identifier is not.

   The RTSP agent sending the request with a Pipelined-Requests header
   has the responsibility for using a unique and previously unused
   identifier within the transport context.  Currently only a TCP
   connection is defined as such transport context.  A server MUST
   delete the Pipelined-Requests identifier and its binding to a session
   upon the termination of that session.  Despite the previous mandate,
   RTSP agents are RECOMMENDED to not reuse identifiers to allow for
   better error handling and logging.

   RTSP Proxies may need to translate Pipelined-Requests identifier
   values from incoming requests to outgoing to allow for aggregation of
   requests onto a persistent connection.

18.34.  Proxy-Authenticate

   The Proxy-Authenticate response-header field MUST be included as part
   of a 407 (Proxy Authentication Required) response.  The field value
   consists of a challenge that indicates the authentication scheme and
   parameters applicable to the proxy for this Request-URI.

   The HTTP access authentication process is described in [RFC2617].
   Unlike WWW-Authenticate, the Proxy-Authenticate header field applies
   only to the current connection and SHOULD NOT be passed on to
   downstream agents.  This header MUST only be used in response
   messages related to client to server requests.

18.35.  Proxy-Authentication-Info

   The Proxy-Authentication-Info response-header is used by the proxy to
   communicate some information regarding the successful authentication
   to the proxy in the message response.  The content and usage of this
   header is described in the HTTP access authentication [RFC2617] that
   is also used by RTSP and clarified in Section 19.1.  This header MUST
   only be used in response messages related to client to server
   requests.  This header has hop by hop scope.




Schulzrinne, et al.      Expires August 14, 2014              [Page 155]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


18.36.  Proxy-Authorization

   The Proxy-Authorization request-header field allows the client to
   identify itself (or its user) to a proxy which requires
   authentication.  The Proxy-Authorization field value consists of
   credentials containing the authentication information of the user
   agent for the proxy and/or realm of the resource being requested.

   The HTTP access authentication process is described in [RFC2617].
   Unlike Authorization, the Proxy-Authorization header field applies
   only to the next hop proxy.  This header MUST only be used in client
   to server requests.

18.37.  Proxy-Require

   The Proxy-Require request-header field is used to indicate proxy-
   sensitive features that MUST be supported by the proxy.  Any Proxy-
   Require header features that are not supported by the proxy MUST be
   negatively acknowledged by the proxy to the client using the
   Unsupported header.  The proxy MUST use the 551 (Option Not
   Supported) status code in the response.  Any feature-tag included in
   the Proxy-Require does not apply to the end-point (server or client).
   To ensure that a feature is supported by both proxies and servers the
   tag needs to be included in also a Require header.

   See Section 18.43 for more details on the mechanics of this message
   and a usage example.  See discussion in the proxies section
   (Section 15.1) about when to consider that a feature requires proxy
   support.

   Example of use:

      Proxy-Require: play.basic

18.38.  Proxy-Supported

   The Proxy-Supported general-header field enumerates all the
   extensions supported by the proxy using feature-tags.  The header
   carries the intersection of extensions supported by the forwarding
   proxies.  The Proxy-Supported header MAY be included in any request
   by a proxy.  It MUST be added by any proxy if the Supported header is
   present in a request.  When present in a request, the receiver MUST
   in the response copy the received Proxy-Supported header.

   The Proxy-Supported header field contains a list of feature-tags
   applicable to proxies, as described in Section 4.5.  The list is the
   intersection of all feature-tags understood by the proxies.  To
   achieve an intersection, the proxy adding the Proxy-Supported header



Schulzrinne, et al.      Expires August 14, 2014              [Page 156]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   includes all proxy feature-tags it understands.  Any proxy receiving
   a request with the header, MUST check the list and removes any
   feature-tag(s) it does not support.  A Proxy-Supported header present
   in the response MUST NOT be modified by the proxies.  These feature
   tags are the ones the proxy chain support in general, and is not
   specific to the request resource.

   Example:

     C->P1: OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech
            User-Agent: PhonyClient/1.2

    P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech
            Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
            Via: 2.0 pro.example.com

    P2->S:  OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech
            Proxy-Supported: proxy-foo, proxy-blech
            Via: 2.0 pro.example.com, 2.0 prox2.example.com

     S->C:  RTSP/2.0 200 OK
            Supported: foo, bar, baz
            Proxy-Supported: proxy-foo, proxy-blech
            Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
            Via: 2.0 pro.example.com, 2.0 prox2.example.com

18.39.  Public

   The Public response-header field lists the set of methods supported
   by the response sender.  This header applies to the general
   capabilities of the sender and its only purpose is to indicate the
   sender's capabilities to the recipient.  The methods listed may or
   may not be applicable to the Request-URI; the Allow header field
   (Section 18.6) MAY be used to indicate methods allowed for a
   particular URI.

   Example of use:

      Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN

   In the event that there are proxies between the sender and the
   recipient of a response, each intervening proxy MUST modify the
   Public header field to remove any methods that are not supported via
   that proxy.  The resulting Public header field will contain an




Schulzrinne, et al.      Expires August 14, 2014              [Page 157]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   intersection of the sender's methods and the methods allowed through
   by the intervening proxies.

      In general, proxies should allow all methods to transparently pass
      through from the sending RTSP agent to the receiving RTSP agent,
      but there may be cases where this is not desirable for a given
      proxy.  Modification of the Public response-header field by the
      intervening proxies ensures that the request sender gets an
      accurate response indicating the methods that can be used on the
      target agent via the proxy chain.

18.40.  Range

   The Range general-header specifies a time range in PLAY
   (Section 13.4), PAUSE (Section 13.6), SETUP (Section 13.3), REDIRECT
   (Section 13.10), and PLAY_NOTIFY (Section 13.5) requests and
   responses.  It MAY be included in GET_PARAMETER requests from the
   client to the server with only a Range format and no value to request
   the current media position, whether the session is in Play or Ready
   state in the included format.  The server SHALL, if supporting the
   range format, respond with the current playing point or pause point
   as the start of the range.  If an explicit stop point was used in the
   previous PLAY request, then that value shall be included as stop
   point.  Note that if the server is currently under any type of media
   playback manipulation affecting the interpretation of Range, like
   Scale, that is also required to be included in any GET_PARAMETER
   response to provide complete information.

   The range can be specified in a number of units.  This specification
   defines smpte (Section 4.4.1), npt (Section 4.4.2), and clock
   (Section 4.4.3) range units.  While octet ranges (Byte Ranges)
   [H14.35.1] and other extended units MAY be used, their behavior is
   unspecified since they are not normally meaningful in RTSP.  Servers
   supporting the Range header MUST understand the NPT range format and
   SHOULD understand the SMPTE range format.  If the Range header is
   sent in a time format that is not understood, the recipient SHOULD
   return 456 (Header Field Not Valid for Resource) and include an
   Accept-Ranges header indicating the supported time formats for the
   given resource.

   Example:

     Range: clock=19960213T143205Z-

   The Range header contains a range of one single range format.  A
   range is a half-open interval with a start and an end point,
   including the start point, but excluding the end point.  A range may
   either be fully specified with explicit values for start point and



Schulzrinne, et al.      Expires August 14, 2014              [Page 158]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   end point, or have either start or end point be implicit.  An
   implicit start point indicates the session's pause point, and if no
   pause point is set the start of the content.  An implicit end point
   indicates the end of the content.  The usage of both implicit start
   and end point is not allowed in the same range header, however, the
   exclusion of the range header has that meaning, i.e., from pause
   point (or start) until end of content.

      Regarding the half-open intervals; a range of A-B starts exactly
      at time A, but ends just before B. Only the start time of a media
      unit such as a video or audio frame is relevant.  For example,
      assume that video frames are generated every 40 ms.  A range of
      10.0-10.1 would include a video frame starting at 10.0 or later
      time and would include a video frame starting at 10.08, even
      though it lasted beyond the interval.  A range of 10.0-10.08, on
      the other hand, would exclude the frame at 10.08.

      Please note the difference between NPT time scales' "now" and an
      implicit start value.  Implicit value reference the current pause-
      point.  While "now" is the currently ongoing time.  In a time-
      progressing session with recording (retention for some or full
      time) the pause point may be 2 min into the session while now
      could be 1 hour into the session.

   By default, range intervals increase, where the second point is
   larger than the first point.

   Example:

       Range: npt=10-15

   However, range intervals can also decrease if the Scale header (see
   Section 18.46) indicates a negative scale value.  For example, this
   would be the case when a playback in reverse is desired.

   Example:

       Scale: -1
       Range: npt=15-10

   Decreasing ranges are still half open intervals as described above.
   Thus, for range A-B, A is closed and B is open.  In the above
   example, 15 is closed and 10 is open.  An exception to this rule is
   the case when B=0 in a decreasing range.  In this case, the range is
   closed on both ends, as otherwise there would be no way to reach 0 on
   a reverse playback for formats that have such a notion, like NPT and
   SMPTE.




Schulzrinne, et al.      Expires August 14, 2014              [Page 159]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Example:

       Scale: -1
       Range: npt=15-0

   In this range both 15 and 0 are closed.

   A decreasing range interval without a corresponding negative Scale
   header is not valid.

18.41.  Referrer

   The Referrer request-header field allows the client to specify, for
   the server's benefit, the address (URI) of the resource from which
   the Request-URI was obtained.  The URI refers to that of the
   presentation description, typically retrieved via HTTP.  The Referrer
   request-header allows a server to generate lists of back-links to
   resources for interest, logging, optimized caching, etc.  It also
   allows obsolete or mistyped links to be traced for maintenance.  The
   Referrer field MUST NOT be sent if the Request-URI was obtained from
   a source that does not have its own URI, such as input from the user
   keyboard.

   If the field value is a relative URI, it SHOULD be interpreted
   relative to the Request-URI.  The URI MUST NOT include a fragment
   identifier.

   Because the source of a link might be private information or might
   reveal an otherwise private information source, it is strongly
   recommended that the user be able to select whether or not the
   Referrer field is sent.  For example, a streaming client could have a
   toggle switch for openly/anonymously, which would respectively enable
   /disable the sending of Referrer and From information.

   Clients SHOULD NOT include a Referrer header field in a (non-secure)
   RTSP request if the referring page was transferred with a secure
   protocol.

18.42.  Request-Status

   This request-header is used to indicate the end result for requests
   that take time to complete, such as PLAY (Section 13.4).  It is sent
   in PLAY_NOTIFY (Section 13.5) with the end-of-stream reason to report
   how the PLAY request concluded, either in success or in failure.  The
   header carries a reference to the request it reports on using the
   CSeq number for the session indicated by the Session header in the
   request.  It provides both a numerical status code (according to
   Section 8.1.1) and a human readable reason phrase.



Schulzrinne, et al.      Expires August 14, 2014              [Page 160]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Example:
   Request-Status: cseq=63 status=500 reason="Media data unavailable"

18.43.  Require

   The Require request-header field is used by agents to ensure that the
   other end-point supports features that are required in respect to
   this request.  It can also be used to query if the other end-point
   supports certain features, however, the use of the Supported general-
   header (Section 18.51) is much more effective in this purpose.  In
   case any of the feature-tags listed by the Require header are not
   supported by the server or client receiving the request, it MUST
   respond to the request using the error code 551 (Option Not
   Supported) and include the Unsupported header listing those feature-
   tags which are NOT supported.  This header does not apply to proxies,
   for the same functionality in respect to proxies see Proxy-Require
   header (Section 18.37) with the exception of media modifying proxies.
   Media modifying proxies, due to their nature of handling media in a
   way that is very similar to a server, do need to understand also the
   server's features to correctly serve the client.

      This is to make sure that the client-server interaction will
      proceed without delay when all features are understood by both
      sides, and only slow down if features are not understood (as in
      the example below).  For a well-matched client-server pair, the
      interaction proceeds quickly, saving a round-trip often required
      by negotiation mechanisms.  In addition, it also removes state
      ambiguity when the client requires features that the server does
      not understand.

   Example (Not complete):

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0
           CSeq: 302
           Require: funky-feature
           Funky-Parameter: funkystuff

   S->C:   RTSP/2.0 551 Option not supported
           CSeq: 302
           Unsupported: funky-feature

   In this example, "funky-feature" is the feature-tag which indicates
   to the client that the fictional Funky-Parameter field is required.
   The relationship between "funky-feature" and Funky-Parameter is not
   communicated via the RTSP exchange, since that relationship is an
   immutable property of "funky-feature" and thus should not be
   transmitted with every exchange.




Schulzrinne, et al.      Expires August 14, 2014              [Page 161]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Proxies and other intermediary devices MUST ignore this header.  If a
   particular extension requires that intermediate devices support it,
   the extension should be tagged in the Proxy-Require field instead
   (see Section 18.37).  See discussion in the proxies section
   (Section 15.1) about when to consider that a feature requires proxy
   support.

18.44.  Retry-After

   The Retry-After response-header field can be used with a 503 (Service
   Unavailable) or 553 (Proxy Unavailable) response to indicate how long
   the service is expected to be unavailable to the requesting client.
   This field MAY also be used with any 3rr (Redirection) response to
   indicate the minimum time the user-agent is asked to wait before
   issuing the redirected request.  The value of this field can be
   either an RTSP-date or an integer number of seconds (in decimal)
   after the time of the response.

   Example:

   Retry-After: Fri, 31 Dec 1999 23:59:59 GMT
   Retry-After: 120

   In the latter example, the delay is 2 minutes.

18.45.  RTP-Info

   The RTP-Info general-header field is used to set RTP-specific
   parameters in the PLAY and GET_PARAMETER responses or a PLAY_NOTIFY
   and GET_PARAMETER requests.  For streams using RTP as transport
   protocol the RTP-Info header SHOULD be part of a 200 response to
   PLAY.

      The exclusion of the RTP-Info in a PLAY response for RTP
      transported media will result in a client needing to synchronize
      the media streams using RTCP.  This may have negative impact as
      the RTCP can be lost, and does not need to be particularly timely
      in its arrival.  Also functionality that informs the client from
      which packet a seek has occurred is affected.

   The RTP-Info MAY be included in SETUP responses to provide
   synchronization information when changing transport parameters, see
   Section 13.3.  The RTP-Info header and the Range header MAY be
   included in a GET_PARAMETER request from client to server without any
   values to request the current playback point and corresponding RTP
   synchronization information.  When the RTP-Info header is included in
   a Request the Range header MUST also be included (Note, Range header
   only MAY be used).  The server response SHALL include both the Range



Schulzrinne, et al.      Expires August 14, 2014              [Page 162]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   header and the RTP-Info header.  If the session is in Play state,
   then the value of the Range header SHALL be filled in with the
   current playback point and with the corresponding RTP-Info values.
   If the server is another state, no values are included in the RTP-
   Info header.  The header is included in PLAY_NOTIFY requests with the
   Notify-Reason of end-of-stream to provide RTP information about the
   end of the stream.

   The header can carry the following parameters:

   url:  Indicates the stream URI for which the following RTP parameters
         correspond, this URI MUST be the same as used in the SETUP
         request for this media stream.  Any relative URI MUST use the
         Request-URI as base URI.  This parameter MUST be present.

   ssrc: The Synchronization source (SSRC) that the RTP timestamp and
         sequence number provided applies to.  This parameter MUST be
         present.

   seq:  Indicates the sequence number of the first packet of the stream
         that is direct result of the request.  This allows clients to
         gracefully deal with packets when seeking.  The client uses
         this value to differentiate packets that originated before the
         seek from packets that originated after the seek.  Note that a
         client may not receive the packet with the expressed sequence
         number, and instead packets with a higher sequence number, due
         to packet loss or reordering.  This parameter is RECOMMENDED to
         be present.

   rtptime:  MUST indicate the RTP timestamp value corresponding to the
         start time value in the Range response-header, or if not
         explicitly given the implied start point.  The client uses this
         value to calculate the mapping of RTP time to NPT or other
         media timescale.  This parameter SHOULD be present to ensure
         inter-media synchronization is achieved.  There exists no
         requirement that any received RTP packet will have the same RTP
         timestamp value as the one in the parameter used to establish
         synchronization.

      A mapping from RTP timestamps to Network Time Protocol (NTP)
      format timestamps (wallclock) is available via RTCP.  However,
      this information is not sufficient to generate a mapping from RTP
      timestamps to media clock time (NPT, etc.).  Furthermore, in order
      to ensure that this information is available at the necessary time
      (immediately at startup or after a seek), and that it is delivered
      reliably, this mapping is placed in the RTSP control channel.





Schulzrinne, et al.      Expires August 14, 2014              [Page 163]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      In order to compensate for drift for long, uninterrupted
      presentations, RTSP clients should additionally map NPT to NTP,
      using initial RTCP sender reports to do the mapping, and later
      reports to check drift against the mapping.

   Example:

   Range:npt=3.25-15
   RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;
            rtptime=12345678,url="rtsp://example.com/foo/video"
            ssrc=9A9DE123:seq=30211;rtptime=29567112

   Lets assume that Audio uses a 16kHz RTP timestamp clock and Video
   a 90kHz RTP timestamp clock. Then the media synchronization is
   depicted in the following way.

   NPT    3.0---3.1---3.2-X-3.3---3.4---3.5---3.6
   Audio               PA A
   Video                  V    PV

   X: NPT time value = 3.25, from Range header.
   A: RTP timestamp value for Audio from RTP-Info header (12345678).
   V: RTP timestamp value for Video from RTP-Info header (29567112).
   PA: RTP audio packet carrying an RTP timestamp of 12344878. Which
       corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2
   PV: RTP video packet carrying an RTP timestamp of 29573412. Which
       corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32

18.46.  Scale

   The Scale general-header indicates the requested or used view rate
   for the media resource being played back.  A scale value of 1
   indicates normal play at the normal forward viewing rate.  If not 1,
   the value corresponds to the rate with respect to normal viewing
   rate.  For example, a ratio of 2 indicates twice the normal viewing
   rate ("fast forward") and a ratio of 0.5 indicates half the normal
   viewing rate.  In other words, a ratio of 2 has content time increase
   at twice the playback time.  For every second of elapsed (wallclock)
   time, 2 seconds of content time will be delivered.  A negative value
   indicates reverse direction.  For certain media transports this may
   require certain considerations to work consistent, see Appendix C.1
   for description on how RTP handles this.

   The transmitted data rate SHOULD NOT be changed by selection of a
   different scale value.  The resulting bit-rate should be reasonably
   close to the nominal bit-rate of the content for Scale = 1.  The
   server has to actively manipulate the data when needed to meet the
   bitrate constraints.  Implementation of scale changes depends on the



Schulzrinne, et al.      Expires August 14, 2014              [Page 164]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   server and media type.  For video, a server may, for example, deliver
   only key frames or selected frames.  For audio, it may time-scale the
   audio while preserving pitch or, less desirably, deliver fragments of
   audio, or completely mute the audio.

   The server and content may restrict the range of scale values that it
   supports.  The supported values are indicated by the Media-Properties
   header (Section 18.29).  The client SHOULD only indicate request
   values to be supported.  However, as the values may change as the
   content progresses a requested value may no longer be valid when the
   request arrives.  Thus, a non-supported value in a request does not
   generate an error, only forces the server to choose the closest
   value.  The response MUST always contain the actual scale value
   chosen by the server.

   If the server does not implement the possibility to scale, it will
   not return a Scale header.  A server supporting Scale operations for
   PLAY MUST indicate this with the use of the "play.scale" feature-tag.

   When indicating a negative scale for a reverse playback, the Range
   header MUST indicate a decreasing range as described in
   Section 18.40.

   Example of playing in reverse at 3.5 times normal rate:

     Scale: -3.5
     Range: npt=15-10

18.47.  Seek-Style

   When a client sends a PLAY request with a Range header to perform a
   random access to the media, the client does not know if the server
   will pick the first media samples or the first random access point
   prior to the request range.  Depending on use case, the client may
   have a strong preference.  To express this preference and provide the
   client with information on how the server actually acted on that
   preference the Seek-Style general-header is defined.

   Seek-Style is a general-header that MAY be included in any PLAY
   request to indicate the client's preference for any media stream that
   has random access properties.  The server MUST always include the
   header in any PLAY response for media with random access properties
   to indicate what policy was applied.  A server that receives an
   unknown Seek-Style policy MUST ignore it and select the server
   default policy.  A client receiving an unknown policy MUST ignore it
   and use the Range header and any media synchronization information as
   basis to determine what the server did.




Schulzrinne, et al.      Expires August 14, 2014              [Page 165]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   This specification defines the following seek policies that may be
   requested (see also Section 4.7.1):

   RAP:  Random Access Point (RAP) is the behavior of requesting the
      server to locate the closest previous random access point that
      exists in the media aggregate and deliver from that.  By
      requesting a RAP, media quality will be the best possible as all
      media will be delivered from a point where full media state can be
      established in the media decoder.

   CoRAP:  Conditional Random Access Point (CoRAP) is a variant of the
      above RAP behavior.  This policy is primarily intended for cases
      where there is larger distance between the random access points in
      the media.  CoRAP is conditioned on that there is a Random Access
      Point closer to the requested start point than to the current
      pause point.  This policy assumes that the media state existing
      prior to the pause is usable if delivery is continued.  If the
      client or server knows that this is not the fact the RAP policy
      should be used.  In other words: in most cases when the client
      requests a start point prior to the current pause point, a valid
      decoding dependency chain from the media delivered prior to the
      pause and to the requested media unit will not exist.  If the
      server searched to a random access point the server MUST return
      the CoRAP policy in the Seek-Style header and adjust the Range
      header to reflect the position of the picked RAP.  In case the
      random access point is further away and the server selects to
      continue from the current pause point it MUST include the "Next"
      policy in the Seek-Style header and adjust the Range header start
      point to the current pause point.

   First-Prior:  The first-prior policy will start delivery with the
      media unit that has a playout time first prior to the requested
      time.  For discrete media that would only include media units that
      would still be rendered at the request time.  For continuous media
      that is media that will be rendered during the requested start
      time of the range.

   Next:  The next media units after the provided start time of the
      range.  For continuous framed media that would mean the first next
      frame after the provided time.  For discrete media the first unit
      that is to be rendered after the provided time.  The main usage
      for this case is when the client knows it has all media up to a
      certain point and would like to continue delivery so that a
      complete non-interrupted media playback can be achieved.  Example
      of such scenarios include switching from a broadcast/multicast
      delivery to a unicast based delivery.  This policy MUST only be
      used on the client's explicit request.




Schulzrinne, et al.      Expires August 14, 2014              [Page 166]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Please note that these expressed preferences exist for optimizing the
   startup time or the media quality.  The "Next" policy breaks the
   normal definition of the Range header to enable a client to request
   media with minimal overlap, although some may still occur for
   aggregated sessions.  RAP and First-Prior both fulfill the
   requirement of providing media from the requested range and forward.
   However, unless RAP is used, the media quality for many media codecs
   using predictive methods can be severely degraded unless additional
   data is available as, for example, already buffered, or through other
   side channels.

18.48.  Server

   The Server general-header field contains information about the
   software used by the origin server to create or handle the request.
   The field can contain multiple product tokens and comments
   identifying the server and any significant subproducts.  The product
   tokens are listed in order of their significance for identifying the
   application.

   Example:

   Server: PhonyServer/1.0

   If the response is being forwarded through a proxy, the proxy
   application MUST NOT modify the Server response-header.  Instead, it
   SHOULD include a Via field (Section 18.57).  If the response is
   generated by the proxy, the proxy application MUST return the Server
   response-header as previously returned by the server.

18.49.  Session

   The Session general-header field identifies an RTSP session.  An RTSP
   session is created by the server as a result of a successful SETUP
   request and in the response the session identifier is given to the
   client.  The RTSP session exists until destroyed by a TEARDOWN,
   REDIRECT or timed out by the server.

   The session identifier is chosen by the server (see Section 4.3) and
   MUST be returned in the SETUP response.  Once a client receives a
   session identifier, it MUST be included in any request related to
   that session.  This means that the Session header MUST be included in
   a request, using the following methods: PLAY, PAUSE, and TEARDOWN,
   and MAY be included in SETUP, OPTIONS, SET_PARAMETER, GET_PARAMETER,
   and REDIRECT, and MUST NOT be included in DESCRIBE.  The Session
   header MUST NOT be included in the following methods, if these
   requests are pipelined and if the session identifier is not yet




Schulzrinne, et al.      Expires August 14, 2014              [Page 167]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   known: PLAY, PAUSE, TEARDOWN, SETUP, OPTIONS SET_PARAMETER, and
   GET_PARAMETER.

   In an RTSP response the session header MUST be included in methods,
   SETUP, PLAY, and PAUSE, and MAY be included in methods, TEARDOWN, and
   REDIRECT, and if included in the request of the following methods it
   MUST also be included in the response, OPTIONS, GET_PARAMETER, and
   SET_PARAMETER, and MUST NOT be included in DESCRIBE responses.

   Note that a session identifier identifies an RTSP session across
   transport sessions or connections.  RTSP requests for a given session
   can use different URIs (Presentation and media URIs).  Note, that
   there are restrictions depending on the session which URIs that are
   acceptable for a given method.  However, multiple "user" sessions for
   the same URI from the same client will require use of different
   session identifiers.

      The session identifier is needed to distinguish several delivery
      requests for the same URI coming from the same client.

   The response 454 (Session Not Found) MUST be returned if the session
   identifier is invalid.

   The header MAY include a parameter for session timeout period.  If
   not explicitly provided this value is set to 60 seconds.  As this
   affects how often session keep-alives are needed values smaller than
   30 seconds are not recommended.  However, larger than default values
   can be useful in applications of RTSP that have inactive but
   established sessions for longer time periods.

      60 seconds was chosen as session timeout value due to: Resulting
      in not too frequent keep-alive messages and having low sensitivity
      to variations in request response timing.  If one reduces the
      timeout value to below 30 seconds the corresponding request
      response timeout becomes a significant part of the session
      timeout. 60 seconds also allows for reasonably rapid recovery of
      committed server resources in case of client failure.

18.50.  Speed

   The Speed general-header field requests the server to deliver
   specific amounts of nominal media time per unit of delivery time,
   contingent on the server's ability and desire to serve the media
   stream at the given speed.  The client requests the delivery speed to
   be within a given range with a lower and upper bound.  The server
   SHALL deliver at the highest possible speed within the range, but not
   faster than the upper-bound, for which the underlying network path
   can support the resulting transport data rates.  As long as any speed



Schulzrinne, et al.      Expires August 14, 2014              [Page 168]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   value within the given range can be provided the server SHALL NOT
   modify the media quality.  Only if the server is unable to deliver
   media at the speed value provided by the lower bound shall it reduce
   the media quality.

   Implementation of the Speed functionality by the server is OPTIONAL.
   The server can indicate its support through a feature-tag,
   play.speed.  The lack of a Speed header in the response is an
   indication of lack of support of this functionality.

   The speed parameter values are expressed as a positive decimal value,
   e.g., a value of 2.0 indicates that data is to be delivered twice as
   fast as normal.  A speed value of zero is invalid.  The range is
   specified in the form "lower bound - upper bound".  The lower bound
   value may be smaller or equal to the upper bound.  All speeds may not
   be possible to support.  Therefore the server MAY modify the
   requested values to the closest supported.  The actual supported
   speed MUST be included in the response.  Note, however, that the use
   cases may vary and that Speed value ranges such as 0.7 - 0.8,
   0.3-2.0, 1.0-2.5, 2.5-2.5 all have their usage.

   Example:

     Speed: 1.0-2.5

   Use of this header changes the bandwidth used for data delivery.  It
   is meant for use in specific circumstances where delivery of the
   presentation at a higher or lower rate is desired.  The main use
   cases are buffer operations or local scale operations.  Implementors
   should keep in mind that bandwidth for the session may be negotiated
   beforehand (by means other than RTSP), and therefore re-negotiation
   may be necessary.  To perform Speed operations the server needs to
   ensure that the network path can support the resulting bit-rate.
   Thus the media transport needs to support feedback so that the server
   can react and adapt to the available bitrate.

18.51.  Supported

   The Supported general-header enumerates all the extensions supported
   by the client or server using feature tags.  The header carries the
   extensions supported by the message sending client or server.  The
   Supported header MAY be included in any request.  When present in a
   request, the receiver MUST respond with its corresponding Supported
   header.  Note that the Supported header is also included in 4xx and
   5xx responses.

   The Supported header contains a list of feature-tags, described in
   Section 4.5, that are understood by the client or server.  These



Schulzrinne, et al.      Expires August 14, 2014              [Page 169]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   feature tags are the ones the server or client support in general,
   and is not specific to the request resource.

   Example:

     C->S:  OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech
            User-Agent: PhonyClient/1.2

     S->C:  RTSP/2.0 200 OK
            Supported: bar, blech, baz

18.52.  Terminate-Reason

   The Terminate-Reason request-header allows the server when sending a
   REDIRECT or TEARDOWN request to provide a reason for the session
   termination and any additional information.  This specification
   identifies three reasons for Redirections and may be extended in the
   future:

   Server-Admin:  The server needs to be shutdown for some
      administrative reason.

   Session-Timeout:  A client's session has been kept alive for extended
      periods of time and the server has determined that it needs to
      reclaim the resources associated with this session.

   Internal-Error  An internal error that is impossible to recover from
      has occurred forcing the server to terminate the session.

   The Server may provide additional parameters containing information
   around the redirect.  This specification defines the following ones.

   time:  Provides a wallclock time when the server will stop providing
      any service.

   user-msg:  An UTF-8 text string with a message from the server to the
      user.  This message SHOULD be displayed to the user.

18.53.  Timestamp

   The Timestamp general-header describes when the agent sent the
   request.  The value of the timestamp is of significance only to the
   agent and may use any timescale.  The responding agent MUST echo the
   exact same value and MAY, if it has accurate information about this,
   add a floating point number indicating the number of seconds that has
   elapsed since it has received the request.  The timestamp can be used
   by the agent to compute the round-trip time to the responding agent



Schulzrinne, et al.      Expires August 14, 2014              [Page 170]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   so that it can adjust the timeout value for retransmissions when
   running over an unreliable protocol.  It also resolves retransmission
   ambiguities for unreliable transport of RTSP.

   Note that the present specification provides only for reliable
   transport of RTSP messages.  The Timestamp general-header is
   specified in case the protocol is extended in the future to use
   unreliable transport.

18.54.  Transport

   The Transport general-header indicates which transport protocol is to
   be used and configures its parameters such as destination address,
   compression, multicast time-to-live and destination port for a single
   stream.  It sets those values not already determined by a
   presentation description.

   A Transport request-header MAY contain a list of transport options
   acceptable to the client, in the form of multiple transport
   specification entries.  Transport specifications are comma separated,
   listed in decreasing order of preference.  Each transport
   specification consists of a transport protocol identifier, followed
   by any number of parameters, each parameter separated by a semicolon.
   A Transport request-header MAY contain multiple transport
   specifications using the same transport protocol Identifier.  The
   server MUST return a Transport response-header in the response to
   indicate the values actually chosen if any.  If no transport
   specification is supported, no transport header is returned and the
   response MUST use the status code 461 (Unsupported Transport)
   (Section 17.4.26).  In case more than one transport specification was
   present in the request, the server MUST return the single transport
   specification (transport-spec) which was actually chosen, if any.
   The number of transport-spec entries is expected to be limited as the
   client will receive guidance on what configurations that are possible
   from the presentation description.

   The Transport header MAY also be used in subsequent SETUP requests to
   change transport parameters.  A server MAY refuse to change
   parameters of an existing stream.

   The transport protocol identifier defines for each transport
   specification which transport protocol to use and any related rules.
   Each transport protocol identifier defines the parameters that are
   required to occur; additional optional parameters MAY occur.  This
   flexibility is provided as parameters may be different and provide
   different options to the RTSP Agent.  A transport specification may
   only contain one of any given parameter within it.  A parameter
   consists of a name and optionally a value string.  Parameters MAY be



Schulzrinne, et al.      Expires August 14, 2014              [Page 171]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   given in any order.  Additionally, a transport specification may only
   contain either of the unicast or the multicast transport type
   parameter.  The transport protocol identifier and all parameters need
   to be understood in a transport specification; if not, the transport
   specification MUST be ignored.  An RTSP proxy of any type that uses
   or modifies the transport specification, e.g., access proxy or
   security proxy, MUST remove specifications with unknown parameters
   before forwarding the RTSP message.  If that results in no remaining
   transport specification the proxy SHALL send a 461 (Unsupported
   Transport) (Section 17.4.26) response without any Transport header.

      The Transport header is restricted to describing a single media
      stream.  (RTSP can also control multiple streams as a single
      entity.)  Making it part of RTSP rather than relying on a
      multitude of session description formats greatly simplifies
      designs of firewalls.

   The general syntax for the transport protocol identifier is a list of
   slash separated tokens:

   Value1/Value2/Value3...

   Which for RTP transports take the form:

   RTP/profile/lower-transport.

   The default value for the "lower-transport" parameters is specific to
   the profile.  For RTP/AVP, the default is UDP.

   There are two different methods for how to specify where the media
   should be delivered for unicast transport:

   dest_addr:  The presence of this parameter and its values indicates
         the destination address or addresses (host address and port
         pairs for IP flows) necessary for the media transport.

   No dest_addr:  The lack of the dest_addr parameter indicates that the
         server MUST send media to the same address from which the RTSP
         messages originates.

   The choice of method for indicating where the media is to be
   delivered depends on the use case.  In some cases the only allowed
   method will be to use no explicit address indication and have the
   server deliver media to the source of the RTSP messages.

   For Multicast there is several methods for specifying addresses but
   they are different in how they work compared with unicast:




Schulzrinne, et al.      Expires August 14, 2014              [Page 172]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   dest_addr with client picked address:  The address and relevant
         parameters, like TTL (scope), for the actual multicast group to
         deliver the media to.  There are security implications
         (Section 21) with this method that need to be addressed if
         using this method because a RTSP server can be used as a Denial
         of Service (DoS) attacker on an existing multicast group.

   dest_addr using Session Description Information:  The information
         included in the transport header can all be coming from the
         session description, e.g., the SDP c= and m= line.  This
         mitigates some of the security issues of the previous methods
         as it is the session provider that picks the multicast group
         and scope.  The client MUST include the information if it is
         available in the session description.

   No dest_addr:  The behavior when no explicit multicast group is
         present in a request is not defined.

   An RTSP proxy will need to take care.  If the media is not desired to
   be routed through the proxy, the proxy will need to introduce the
   destination indication.

   Below are the configuration parameters associated with transport:

   General parameters:

   unicast / multicast:  This parameter is a mutually exclusive
         indication of whether unicast or multicast delivery will be
         attempted.  One of the two values MUST be specified.  Clients
         that are capable of handling both unicast and multicast
         transmission need to indicate such capability by including two
         full transport-specs with separate parameters for each.

   layers:  The number of multicast layers to be used for this media
         stream.  The layers are sent to consecutive addresses starting
         at the dest_addr address.  If the parameter is not included, it
         defaults to a single layer.

   dest_addr:  A general destination address parameter that can contain
         one or more address specifications.  Each combination of
         protocol/profile/lower transport needs to have the format and
         interpretation of its address specification defined.  For RTP/
         AVP/UDP and RTP/AVP/TCP, the address specification is a tuple
         containing a host address and port.  Note, only a single
         destination parameter per transport spec is intended.  The
         usage of multiple destinations to distribute a single media to
         multiple entities is unspecified.




Schulzrinne, et al.      Expires August 14, 2014              [Page 173]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         The client originating the RTSP request MAY specify the
         destination address of the stream recipient with the host
         address part of the tuple.  When the destination address is
         specified, the recipient may be a different party than the
         originator of the request.  To avoid becoming the unwitting
         perpetrator of a remote-controlled denial-of-service attack, a
         server MUST perform security checks (see Section 21.2.1) and
         SHOULD log such attempts before allowing the client to direct a
         media stream to a recipient address not chosen by the server.
         Implementations cannot rely on TCP as reliable means of client
         identification.  If the server does not allow the host address
         part of the tuple to be set, it MUST return 463 (Destination
         Prohibited).

         The host address part of the tuple MAY be empty, for example
         ":58044", in cases when it is desired to specify only the
         destination port.  Responses to requests including the
         Transport header with a dest_addr parameter SHOULD include the
         full destination address that is actually used by the server.
         The server MUST NOT remove address information present already
         in the request when responding unless the protocol requires it.

   src_addr:  A general source address parameter that can contain one or
         more address specifications.  Each combination of protocol/
         profile/lower transport needs to have the format and
         interpretation of its address specification defined.  For RTP/
         AVP/UDP and RTP/AVP/TCP, the address specification is a tuple
         containing a host address and port.

         This parameter MUST be specified by the server if it transmits
         media packets from another address than the one RTSP messages
         are sent to.  This will allow the client to verify source
         address and give it a destination address for its RTCP feedback
         packets, if RTP is used.  The address or addresses indicated in
         the src_addr parameter SHOULD be used both for sending and
         receiving of the media stream's data packets.  The main reasons
         are threefold: First, indicating the port and source address(s)
         lets the receiver know where from the packets is expected to
         originate.  Secondly, traversal of NATs is greatly simplified
         when traffic is flowing symmetrically over a NAT binding.
         Thirdly, certain NAT traversal mechanisms, needs to know to
         which address and port to send so called "binding packets" from
         the receiver to the sender, thus creating an address binding in
         the NAT that the sender to receiver packet flow can use.







Schulzrinne, et al.      Expires August 14, 2014              [Page 174]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


            This information may also be available through SDP.
            However, since this is more a feature of transport than
            media initialization, the authoritative source for this
            information should be in the SETUP response.

   mode: The mode parameter indicates the methods to be supported for
         this session.  Currently defined valid values are "PLAY".  If
         not provided, the default is "PLAY".  The "RECORD" value was
         defined in RFC 2326 and is in this specification unspecified
         but reserved.  RECORD and other values may be specified in the
         future.

   interleaved:  The interleaved parameter implies mixing the media
         stream with the control stream in whatever protocol is being
         used by the control stream, using the mechanism defined in
         Section 14.  The argument provides the channel number to be
         used in the $ block (see Section 14) and MUST be present.  This
         parameter MAY be specified as an interval, e.g.,
         interleaved=4-5 in cases where the transport choice for the
         media stream requires it, e.g., for RTP with RTCP.  The channel
         number given in the request is only a guidance from the client
         to the server on what channel number(s) to use.  The server MAY
         set any valid channel number in the response.  The declared
         channel(s) are bi-directional, so both end-parties MAY send
         data on the given channel.  One example of such usage is the
         second channel used for RTCP, where both server and client send
         RTCP packets on the same channel.



            This allows RTP/RTCP to be handled similarly to the way that
            it is done with UDP, i.e., one channel for RTP and the other
            for RTCP.

   MIKEY:  This parameter is used in conjunction with transport
         specifications that can utilize MIKEY [RFC3830] for security
         context establishment.  So far only the SRTP based RTP profiles
         SAVP and SAVPF can utilize MIKEY and this is defined in
         Appendix C.1.4.1.  This parameter can be included both in
         request and response messages.  The binary MIKEY message SHALL
         be BASE64 [RFC4648] encoded before being included in the value
         part of the parameter, where the encoding adheres to the
         definition in Section 4 of RFC 4648 and where the padding bits
         are set to zero.

   Multicast-specific:





Schulzrinne, et al.      Expires August 14, 2014              [Page 175]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   ttl:  multicast time-to-live for IPv4.  When included in requests the
         value indicate the TTL value that the client requests the
         server to use.  In a response, the value actually being used by
         the server is returned.  A server will need to consider what
         values that are reasonable and also the authority of the user
         to set this value.  Corresponding functions are not needed for
         IPv6 as the scoping is part of the IPv6 multicast address
         [RFC4291].

   RTP-specific:

   These parameters MAY only be used if the media transport protocol is
   RTP.

   ssrc: The ssrc parameter, if included in a SETUP response, indicates
         the RTP SSRC [RFC3550] value(s) that will be used by the media
         server for RTP packets within the stream.  It is expressed as
         an eight digit hexadecimal value.

         The ssrc parameter MUST NOT be specified in requests.  The
         functionality of specifying the ssrc parameter in a SETUP
         request is deprecated as it is incompatible with the
         specification of RTP in RFC 3550[RFC3550].  If the parameter is
         included in the Transport header of a SETUP request, the server
         SHOULD ignore it, and choose appropriate SSRCs for the stream.
         The server SHOULD set the ssrc parameter in the Transport
         header of the response.

   RTCP-mux:  Use to negotiate the usage of RTP and RTCP multiplexing
         [RFC5761] on a single underlying transport stream / flow.  The
         presence of this parameter in a SETUP request indicates the
         client's support and requires the server to use RTP and RTCP
         multiplexing.  The client SHALL only include one transport
         stream in the Transport header specification.  To provide the
         server with a choice between using RTP/RTCP multiplexing or
         not, two different transport header specifications must be
         included.

   The parameters setup and connection defined below MAY only be used if
   the media transport protocol of the lower-level transport is
   connection-oriented (such as TCP).  However, these parameters MUST
   NOT be used when interleaving data over the RTSP connection.

   setup:  Clients use the setup parameter on the Transport line in a
         SETUP request, to indicate the roles it wishes to play in a TCP
         connection.  This parameter is adapted from [RFC4145].  The use
         of this parameter in RTP/AVP/TCP non-interleaved transport is
         discussed in Appendix C.2.2; the discussion below is limited to



Schulzrinne, et al.      Expires August 14, 2014              [Page 176]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         syntactic issues.  Clients may specify the following values for
         the setup parameter:

         active:  The client will initiate an outgoing connection.

         passive:  The client will accept an incoming connection.

         actpass:  The client is willing to accept an incoming
            connection or to initiate an outgoing connection.

         If a client does not specify a setup value, the "active" value
         is assumed.

         In response to a client SETUP request where the setup parameter
         is set to "active", a server's 2xx reply MUST assign the setup
         parameter to "passive" on the Transport header line.

         In response to a client SETUP request where the setup parameter
         is set to "passive", a server's 2xx reply MUST assign the setup
         parameter to "active" on the Transport header line.

         In response to a client SETUP request where the setup parameter
         is set to "actpass", a server's 2xx reply MUST assign the setup
         parameter to "active" or "passive" on the Transport header
         line.

         Note that the "holdconn" value for setup is not defined for
         RTSP use, and MUST NOT appear on a Transport line.

   connection:  Clients use the connection parameter in a transport
         specification part of the Transport header in a SETUP request,
         to indicate the client's preference for either reusing an
         existing connection between client and server (in which case
         the client sets the "connection" parameter to "existing"), or
         requesting the creation of a new connection between client and
         server (in which cast the client sets the "connection"
         parameter to "new").  Typically, clients use the "new" value
         for the first SETUP request for a URL, and "existing" for
         subsequent SETUP requests for a URL.

         If a client SETUP request assigns the "new" value to
         "connection", the server response MUST also assign the "new"
         value to "connection" on the Transport line.

         If a client SETUP request assigns the "existing" value to
         "connection", the server response MUST assign a value of
         "existing" or "new" to "connection" on the Transport line, at
         its discretion.



Schulzrinne, et al.      Expires August 14, 2014              [Page 177]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         The default value of "connection" is "existing", for all SETUP
         requests (initial and subsequent).

   The combination of transport protocol, profile and lower transport
   needs to be defined.  A number of combinations are defined in the
   Appendix C.

   Below is a usage example, showing a client advertising the capability
   to handle multicast or unicast, preferring multicast.  Since this is
   a unicast-only stream, the server responds with the proper transport
   parameters for unicast.

     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
           CSeq: 302
           Transport: RTP/AVP;multicast;mode="PLAY",
               RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
               "192.0.2.5:3457";mode="PLAY"
           Accept-Ranges: npt, smpte, clock
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 302
           Date: Fri, 20 Dec 2013 10:20:32 +0000
           Session: 47112344
           Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
              "192.0.2.5:3457";src_addr="192.0.2.224:6256"/
              "192.0.2.224:6257";mode="PLAY"
           Accept-Ranges: npt
           Media-Properties: Random-Access=0.6, Dynamic,
                             Time-Limited=20081128T165900

18.55.  Unsupported

   The Unsupported response-header lists the features not supported by
   the responding RTSP agent.  In the case where the feature was
   specified via the Proxy-Require field (Section 18.37), if there is a
   proxy on the path between the client and the server, the proxy MUST
   send a response message with a status code of 551 (Option Not
   Supported).  The request MUST NOT be forwarded.

   See Section 18.43 for a usage example.

18.56.  User-Agent

   The User-Agent general-header field contains information about the
   user agent originating the request or producing a response.  This is
   for statistical purposes, the tracing of protocol violations, and
   automated recognition of user agents for the sake of tailoring



Schulzrinne, et al.      Expires August 14, 2014              [Page 178]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   responses to avoid particular user agent limitations.  User agents
   SHOULD include this field with requests.  The field can contain
   multiple product tokens and comments identifying the agent and any
   subproducts which form a significant part of the user agent.  By
   convention, the product tokens are listed in order of their
   significance for identifying the application.

   Example:

   User-Agent: PhonyClient/1.2

18.57.  Via

   The Via general-header field MUST be used by proxies to indicate the
   intermediate protocols and recipients between the user agent and the
   server on requests, and between the origin server and the client on
   responses.  The field is intended to be used for tracking message
   forwards, avoiding request loops, and identifying the protocol
   capabilities of all senders along the request/response chain.

   Multiple Via field values represents each proxy that has forwarded
   the message.  Each recipient MUST append its information such that
   the end result is ordered according to the sequence of forwarding
   applications.

   Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by
   default, forward the names and ports of hosts within the private/
   protected region.  This information SHOULD only be propagated if
   explicitly enabled.  If not enabled, the via-received of any host
   behind the firewall/NAT SHOULD be replaced by an appropriate
   pseudonym for that host.

   For organizations that have strong privacy requirements for hiding
   internal structures, a proxy MAY combine an ordered subsequence of
   Via header field entries with identical sent-protocol values into a
   single such entry.  Applications MUST NOT combine entries which have
   different received-protocol values.

18.58.  WWW-Authenticate

   The WWW-Authenticate response-header field MUST be included in 401
   (Unauthorized) response messages.  The field value consists of at
   least one challenge that indicates the authentication scheme(s) and
   parameters applicable to the Request-URI.  This header MUST only be
   used in response messages related to client to server requests.

   The HTTP access authentication process is described in [RFC2617] with
   some clarification in Section 19.1.  User agents are advised to take



Schulzrinne, et al.      Expires August 14, 2014              [Page 179]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   special care in parsing the WWW-Authenticate field value as it might
   contain more than one challenge, or if more than one WWW-Authenticate
   header field is provided, the contents of a challenge itself can
   contain a comma-separated list of authentication parameters.

19.  Security Framework

   The RTSP security framework consists of two high level components:
   the pure authentication mechanisms based on HTTP authentication, and
   the message transport protection based on TLS, which is independent
   of RTSP.  Because of the similarity in syntax and usage between RTSP
   servers and HTTP servers, the security for HTTP is re-used to a large
   extent.

19.1.  RTSP and HTTP Authentication

   RTSP and HTTP share common authentication schemes, and thus follow
   the same usage guidelines as specified in [RFC2617] with the
   additions for digest authentication specified below in
   Section 19.1.1.  Servers SHOULD implement both basic and digest
   [RFC2617] authentication.  Clients MUST implement both basic and
   digest authentication [RFC2617] so that a server that requires the
   client to authenticate can trust that the capability is present.

   It should be stressed that using the HTTP authentication alone does
   not provide full control message security.  Therefore, in
   environments requiring tighter security for the control messages, TLS
   SHOULD be used, see Section 19.2.  Any RTSP message containing an
   Authorization header using basic authorization MUST be using a TLS
   connection with confidentiality protection enabled, i.e., no NULL
   encryption.

   In cases where there is a chain of proxies between the client and the
   server, each proxy may individually request the client or previous
   proxy to authenticate itself.  This is done using the Proxy-
   Authenticate (Section 18.34), the Proxy-Authorization (Section 18.36)
   and the Proxy-Authentication-Info (Section 18.35) headers.  These
   headers are hop-by-hop headers and are only scoped to the current
   connection and hop.  Thus if a proxy chain exists, a proxy connecting
   to another proxy will have to act as a client to authorize itself
   towards the next proxy.  The WWW-Authenticate (Section 18.58),
   Authorization (Section 18.8) and Authentication-Info (Section 18.7)
   headers are end-to-end and must not be modified by proxies.

   This authentication mechanism works only for client to server
   requests as currently defined.  This leaves server to client request
   outside of the context of TLS based communication more vulnerable to
   message injection attacks on the client.  Based on the server to



Schulzrinne, et al.      Expires August 14, 2014              [Page 180]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   client methods that exist, the potential risks are various; hijacking
   (REDIRECT), denial of service (TEARDOWN and PLAY_NOTIFY) or attacks
   with uncertain results (SET_PARAMETER).

19.1.1.  Digest Authentication

   This section describes the modifications and clarifications required
   to apply the HTTP Digest authentication scheme to RTSP.  The RTSP
   scheme usage is almost completely identical to that for HTTP
   [RFC2617].  These are based on the procedures defined for SIP 2.0
   [RFC3261].

   The rules for Digest authentication follow those defined in
   [RFC2617], with "HTTP/1.1" replaced by "RTSP/2.0" in addition to the
   following differences:

   1.  Use the ABNF specified in this document, rather than the one in
       [RFC2617].  Consequently the following is assured:

       *  Using the right RTSP URIs allowed in the challenge as well as
          in the digest.

       *  Resolved the error in the "uri" parameter of the Authorization
          header in [RFC2617].

   2.  If MTags are used then the example procedure for choosing a nonce
       based on Etag can work based on replacing ETag with the MTag.

   3.  As a clarification to the calculation of the A2 value for message
       integrity assurance in the Digest authentication scheme,
       implementers should assume, when the entity-body is empty (that
       is, when the RTSP messages have no message body) that the hash of
       the message-body resolves to the MD5 hash of an empty string, or:
       H(entity-body) = MD5("") = "d41d8cd98f00b204e9800998ecf8427e".

   4.  RFC 2617 notes that a cnonce value MUST NOT be sent in an
       Authorization (and by extension Proxy-Authorization) header field
       if no qop directive has been sent.  Therefore, any algorithms
       that have a dependency on the cnonce (including "MD5-Sess")
       require that the qop directive be sent.  Use of the "qop"
       parameter is optional in RFC 2617 for the purposes of backwards
       compatibility with RFC 2069; since this specification defines
       RTSP 2.0 there is no backwards compatibility issue with
       mandating.  Thus, all RTSP agents MUST implement qop-options.







Schulzrinne, et al.      Expires August 14, 2014              [Page 181]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


19.2.  RTSP over TLS

   RTSP agents MUST implement RTSP over TLS as defined in this section
   and the next Section 19.3.  RTSP MUST follow the same guidelines with
   regards to TLS [RFC5246] usage as specified for HTTP, see [RFC2818].
   RTSP over TLS is separated from unsecured RTSP both on the URI level
   and the port level.  Instead of using the "rtsp" scheme identifier in
   the URI, the "rtsps" scheme identifier MUST be used to signal RTSP
   over TLS.  If no port is given in a URI with the "rtsps" scheme, port
   322 MUST be used for TLS over TCP/IP.

   When a client tries to setup an insecure channel to the server (using
   the "rtsp" URI), and the policy for the resource requires a secure
   channel, the server MUST redirect the client to the secure service by
   sending a 301 redirect response code together with the correct
   Location URI (using the "rtsps" scheme).  A user or client MAY
   upgrade a non secured URI to a secured by changing the scheme from
   "rtsp" to "rtsps".  A server implementing support for "rtsps" MUST
   allow this.

   It should be noted that TLS allows for mutual authentication (when
   using both server and client certificates).  Still, one of the more
   common ways TLS is used is to only provide server side authentication
   (often to avoid client certificates).  TLS is then used in addition
   to HTTP authentication, providing transport security and server
   authentication, while HTTP Authentication is used to authenticate the
   client.

   RTSP includes the possibility to keep a TCP session up between the
   client and server, throughout the RTSP session lifetime.  It may be
   convenient to keep the TCP session, not only to save the extra setup
   time for TCP, but also the extra setup time for TLS (even if TLS uses
   the resume function, there will be almost two extra round trips).
   Still, when TLS is used, such behavior introduces extra active state
   in the server, not only for TCP and RTSP, but also for TLS.  This may
   increase the vulnerability to DoS attacks.

   There exists a potential security vulnerability when reusing TCP and
   TLS state for different resources (URIs).  If two different host
   names point at the same IP address it can be desirable to re-use the
   TCP/TLS connection to that server.  In that case the RTSP agent
   having the TCP/TLS connection MUST verify that the server certificate
   associated with the connection has a SubjectAltName matching the host
   name present in the URI for the resource an RTSP request is to be
   issued for.

   In addition to these recommendations, Section 19.3 gives further
   recommendations of TLS usage with proxies.



Schulzrinne, et al.      Expires August 14, 2014              [Page 182]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


19.3.  Security and Proxies

   The nature of a proxy is often to act as a "man-in-the-middle", while
   security is often about preventing the existence of a "man-in-the-
   middle".  This section provides clients with the possibility to use
   proxies even when applying secure transports (TLS) between the RTSP
   agents.  The TLS proxy mechanism allows for server and proxy
   identification using certificates.  However, the client cannot be
   identified based on certificates.  The client needs to select between
   using the procedure specified below or using a TLS connection
   directly (by-passing any proxies) to the server.  The choice may be
   dependent on policies.

   There are in general two categories of proxies, the transparent
   proxies (of which the client is not aware) and the non-transparent
   proxies (of which the client is aware).  This memo specifies only
   non-transparent RTSP proxies, i.e., proxies visible to the RTSP
   client and RTSP server.  An infrastructure based on proxies requires
   that the trust model is such that both client and servers can trust
   the proxies to handle the RTSP messages correctly.  To be able to
   trust a proxy, the client and server also need to be aware of the
   proxy.  Hence, transparent proxies cannot generally be seen as
   trusted and will not work well with security (unless they work only
   at the transport layer).  In the rest of this section any reference
   to proxy will be to a non-transparent proxy, which inspects or
   manipulates the RTSP messages.

   HTTP Authentication is built on the assumption of proxies and can
   provide user-proxy authentication and proxy-proxy/server
   authentication in addition to the client-server authentication.

   When TLS is applied and a proxy is used, the client will connect to
   the proxy's address when connecting to any RTSP server.  This implies
   that for TLS, the client will authenticate the proxy server and not
   the end server.  Note that when the client checks the server
   certificate in TLS, it MUST check the proxy's identity (URI or
   possibly other known identity) against the proxy's identity as
   presented in the proxy's Certificate message.

   The problem is that for a proxy accepted by the client, the proxy
   needs to be provided information on which grounds it should accept
   the next-hop certificate.  Both the proxy and the user may have rules
   for this, and the user should have the possibility to select the
   desired behavior.  To handle this case, the Accept-Credentials header
   (See Section 18.2) is used, where the client can request the proxy/
   proxies to relay back the chain of certificates used to authenticate
   any intermediate proxies as well as the server.  The assumption that
   the proxies are viewed as trusted, gives the user a possibility to



Schulzrinne, et al.      Expires August 14, 2014              [Page 183]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   enforce policies to each trusted proxy of whether it should accept
   the next agent in the chain.  However, it should be noted that not
   all deployments will return the chain of certificates used to
   authenticate any intermediate proxies as well as the server.  An
   operator of such a deployment may want to hide its topology from the
   client.  It should be noted well that the client does not have any
   insight into the proxy's operation.  Even if the proxy is trusted, it
   can still return an incomplete chain of certificates.

   A proxy MUST use TLS for the next hop if the RTSP request includes a
   "rtsps" URI.  TLS MAY be applied on intermediate links (e.g., between
   client and proxy, or between proxy and proxy), even if the resource
   and the end server are not required to use it.  The chain of proxies
   used by a client to reach a server and their TLS sessions MUST have
   commensurate security.  Therefore a proxy MUST, when initiating the
   next hop TLS connection, use the incoming TLS connections cipher
   suite list, only modified by removing any cipher suites that the
   proxy does not support.  In case a proxy fails to establish a TLS
   connection due to cipher suite mismatch between proxy and next hop
   proxy or server, this is indicated using error code 472 (Failure to
   establish secure connection).

19.3.1.  Accept-Credentials

   The Accept-Credentials header can be used by the client to distribute
   simple authorization policies to intermediate proxies.  The client
   includes the Accept-Credentials header to dictate how the proxy
   treats the server/next proxy certificate.  There are currently three
   methods defined:

   Any:  which means that the proxy (or proxies) MUST accept whatever
         certificate is presented.  This is of course not a recommended
         option to use, but may be useful in certain circumstances (such
         as testing).

   Proxy:  which means that the proxy (or proxies) MUST use its own
         policies to validate the certificate and decide whether to
         accept it or not.  This is convenient in cases where the user
         has a strong trust relation with the proxy.  Reasons why a
         strong trust relation may exist are: personal/company proxy,
         proxy has a out-of-band policy configuration mechanism.

   User: which means that the proxy (or proxies) MUST send credential
         information about the next hop to the client for authorization.
         The client can then decide whether the proxy should accept the
         certificate or not.  See Section 19.3.2 for further details.





Schulzrinne, et al.      Expires August 14, 2014              [Page 184]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   If the Accept-Credentials header is not included in the RTSP request
   from the client, then the "Proxy" method MUST be used as default.  If
   another method than the "Proxy" is to be used, then the Accept-
   Credentials header MUST be included in all of the RTSP requests from
   the client.  This is because it cannot be assumed that the proxy
   always keeps the TLS state or the user's previous preference between
   different RTSP messages (in particular if the time interval between
   the messages is long).

   With the "Any" and "Proxy" methods the proxy will apply the policy as
   defined for each method.  If the policy does not accept the
   credentials of the next hop, the proxy MUST respond with a message
   using status code 471 (Connection Credentials not accepted).

   An RTSP request in the direction server to client MUST NOT include
   the Accept-Credentials header.  As for the non-secured communication,
   the possibility for these requests depends on the presence of a
   client established connection.  However, if the server to client
   request is in relation to a session established over a TLS secured
   channel, it MUST be sent in a TLS secured connection.  That secured
   connection MUST also be the one used by the last client to server
   request.  If no such transport connection exists at the time when the
   server desires to send the request, the server MUST discard the
   message.

   Further policies MAY be defined and registered, but should be done so
   with caution.

19.3.2.  User approved TLS procedure

   For the "User" method, each proxy MUST perform the following
   procedure for each RTSP request:

   o  Setup the TLS session to the next hop if not already present
      (i.e., run the TLS handshake, but do not send the RTSP request).

   o  Extract the peer certificate chain for the TLS session.

   o  Check if a matching identity and hash of the peer certificate is
      present in the Accept-Credentials header.  If present, send the
      message to the next hop, and conclude these procedures.  If not,
      go to the next step.

   o  The proxy responds to the RTSP request with a 470 or 407 response
      code.  The 407 response code MAY be used when the proxy requires
      both user and connection authorization from user or client.  In
      this message the proxy MUST include a Connection-Credentials




Schulzrinne, et al.      Expires August 14, 2014              [Page 185]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      header, see Section 18.13 with the next hop's identity and
      certificate.

   The client MUST upon receiving a 470 or 407 response with Connection-
   Credentials header take the decision on whether to accept the
   certificate or not (if it cannot do so, the user SHOULD be
   consulted).  Using IP addresses in the next hop URI and certificates
   rather than domain names makes it very difficult for a user to
   determine if it should approve the next hop or not.  Proxies are
   RECOMMENDED to use domain names to identify themselves in URIs and in
   the certificates.  If the certificate is accepted, the client has to
   again send the RTSP request.  In that request the client has to
   include the Accept-Credentials header including the hash over the DER
   encoded certificate for all trusted proxies in the chain.

   Example:

   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
         CSeq: 2
         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
                    "192.0.2.5:4589"
         Accept-Ranges: npt, smpte, clock
         Accept-Credentials: User

   P->C: RTSP/2.0 470 Connection Authorization Required
         CSeq: 2
         Connection-Credentials: "rtsps://test.example.org";
         MIIDNTCCAp...

   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
         CSeq: 3
         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
                    "192.0.2.5:4589"
         Accept-Credentials: User "rtsps://test.example.org";sha-256;
         dPYD7txpoGTbAqZZQJ+vaeOkyH4=
         Accept-Ranges: npt, smpte, clock

   P->S: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
         CSeq: 3
         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
                    "192.0.2.5:4589"
         Via: RTSP/2.0 proxy.example.org
         Accept-Credentials: User "rtsps://test.example.org";sha-256;
         dPYD7txpoGTbAqZZQJ+vaeOkyH4=
         Accept-Ranges: npt, smpte, clock

   One implication of this process is that the connection for secured
   RTSP messages may take significantly more round-trip times for the



Schulzrinne, et al.      Expires August 14, 2014              [Page 186]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   first message.  A complete extra message exchange between the proxy
   connecting to the next hop and the client results because of the
   process for approval for each hop.  However, if each message contains
   the chain of proxies that the requester accepts, the remaining
   message exchange should not be delayed.  The procedure of including
   the credentials in each request rather than building state in each
   proxy, avoids the need for revocation procedures.

20.  Syntax

   The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)
   as defined in RFC 5234 [RFC5234].  It uses the basic definitions
   present in RFC 5234.

   Please note that ABNF strings, e.g., "Accept", are case insensitive
   as specified in section 2.3 of RFC 5234.

   The RTSP syntax makes use of the ISO 10646 character set in UTF-8
   encoding RFC 3629 [RFC3629].

20.1.  Base Syntax

   RTSP header values can be folded onto multiple lines if the
   continuation line begins with a space or horizontal tab.  All linear
   white space, including folding, has the same semantics as SP.  A
   recipient MAY replace any linear white space with a single SP before
   interpreting the field value or forwarding the message downstream.
   This is intended to behave exactly as HTTP/1.1 as described in RFC
   2616 [RFC2616].  The SWS construct is used when linear white space is
   optional, generally between tokens and separators.

   To separate the header name from the rest of value, a colon is used,
   which, by the above rule, allows whitespace before, but no line
   break, and whitespace after, including a line break.  The HCOLON
   defines this construct.
















Schulzrinne, et al.      Expires August 14, 2014              [Page 187]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   OCTET           =  %x00-FF ; any 8-bit sequence of data
   CHAR            =  %x01-7F ; any US-ASCII character (octets 1 - 127)
   UPALPHA         =  %x41-5A ; any US-ASCII uppercase letter "A".."Z"
   LOALPHA         =  %x61-7A ;any US-ASCII lowercase letter "a".."z"
   ALPHA           =  UPALPHA / LOALPHA
   DIGIT           =  %x30-39 ; any US-ASCII digit "0".."9"
   CTL             =  %x00-1F / %x7F  ; any US-ASCII control character
                      ; (octets 0 - 31) and DEL (127)
   CR              =  %x0D ; US-ASCII CR, carriage return (13)
   LF              =  %x0A  ; US-ASCII LF, linefeed (10)
   SP              =  %x20  ; US-ASCII SP, space (32)
   HT              =  %x09  ; US-ASCII HT, horizontal-tab (9)
   BACKSLASH       =  %x5C  ; US-ASCII backslash (92)
   CRLF            =  CR LF

   LWS             =  [CRLF] 1*( SP / HT ) ; Line-breaking White Space
   SWS             =  [LWS] ; Separating White Space
   HCOLON          =  *( SP / HT ) ":" SWS
   TEXT            =  %x20-7E / %x80-FF  ; any OCTET except CTLs
   tspecials       =  "(" / ")" / "<" / ">" / "@"
                   /  "," / ";" / ":" / BACKSLASH / DQUOTE
                   /  "/" / "[" / "]" / "?" / "="
                   /  "{" / "}" / SP / HT
   token           =  1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
                   /  %x41-5A / %x5E-7A / %x7C / %x7E)
                      ; 1*<any CHAR except CTLs or tspecials>
   quoted-string   =  ( DQUOTE *qdtext DQUOTE )
   qdtext          = %x20-21 / %x23-5B / %x5D-7E / quoted-pair
                   / UTF8-NONASCII
                   ; No DQUOTE and no "\"
   quoted-pair     = "\\" / ( "\" DQUOTE )
   ctext           =  %x20-27 / %x2A-7E
                   /  %x80-FF  ; any OCTET except CTLs, "(" and ")"
   generic-param   =  token [ EQUAL gen-value ]
   gen-value       =  token / host / quoted-string
















Schulzrinne, et al.      Expires August 14, 2014              [Page 188]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   safe            =  "$" / "-" / "_" / "." / "+"
   extra           =  "!" / "*" / "'" / "(" / ")" / ","
   rtsp-extra      =  "!" / "*" / "'" / "(" / ")"

   HEX             =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F"
                   /  "a" / "b" / "c" / "d" / "e" / "f"
   LHEX            =  DIGIT /  "a" / "b" / "c" / "d" / "e" / "f"
                      ; lowercase "a-f" Hex
   reserved        =  ";" / "/" / "?" / ":" / "@" / "&" / "="

   unreserved      =  ALPHA / DIGIT / safe / extra
   rtsp-unreserved  =  ALPHA / DIGIT / safe / rtsp-extra

   base64          =  *base64-unit [base64-pad]
   base64-unit     =  4base64-char
   base64-pad      =  (2base64-char "==") / (3base64-char "=")
   base64-char     =  ALPHA / DIGIT / "+" / "/"

   SLASH    =  SWS "/" SWS ; slash
   EQUAL    =  SWS "=" SWS ; equal
   LPAREN   =  SWS "(" SWS ; left parenthesis
   RPAREN   =  SWS ")" SWS ; right parenthesis
   COMMA    =  SWS "," SWS ; comma
   SEMI     =  SWS ";" SWS ; semicolon
   COLON    =  SWS ":" SWS ; colon
   MINUS    =  SWS "-" SWS ; minus/dash
   LDQUOT   =  SWS DQUOTE ; open double quotation mark
   RDQUOT   =  DQUOTE SWS ; close double quotation mark
   RAQUOT   =  ">" SWS ; right angle quote
   LAQUOT   =  SWS "<" ; left angle quote

   TEXT-UTF8char    =  %x21-7E / UTF8-NONASCII
   UTF8-NONASCII    = UTF8-2 / UTF8-3 / UTF8-4
   UTF8-1           = <As defined in RFC 3629>
   UTF8-2           = <As defined in RFC 3629>
   UTF8-3           = <As defined in RFC 3629>
   UTF8-4           = <As defined in RFC 3629>
   UTF8-tail        = <As defined in RFC 3629>

   POS-FLOAT        = 1*12DIGIT ["." 1*9DIGIT]
   FLOAT            = ["-"] POS-FLOAT

20.2.  RTSP Protocol Definition








Schulzrinne, et al.      Expires August 14, 2014              [Page 189]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


20.2.1.  Generic Protocol elements

   RTSP-IRI       =  schemes ":" IRI-rest
   IRI-rest       =  ihier-part [ "?" iquery ]
   ihier-part     =  "//" iauthority ipath-abempty
   RTSP-IRI-ref   =  RTSP-IRI / irelative-ref
   irelative-ref  =  irelative-part [ "?" iquery ]
   irelative-part =  "//" iauthority ipath-abempty
                     / ipath-absolute
                     / ipath-noscheme
                     / ipath-empty

   iauthority     =  < As defined in RFC 3987>
   ipath          =  ipath-abempty   ; begins with "/" or is empty
                     / ipath-absolute  ; begins with "/" but not "//"
                     / ipath-noscheme  ; begins with a non-colon segment
                     / ipath-rootless  ; begins with a segment
                     / ipath-empty     ; zero characters

   ipath-abempty   =  *( "/" isegment )
   ipath-absolute  =  "/" [ isegment-nz *( "/" isegment ) ]
   ipath-noscheme  =  isegment-nz-nc *( "/" isegment )
   ipath-rootless  =  isegment-nz *( "/" isegment )
   ipath-empty     =  0<ipchar>

   isegment        =  *ipchar [";" *ipchar]
   isegment-nz     =  1*ipchar [";" *ipchar]
                      / ";" *ipchar
   isegment-nz-nc  =  (1*ipchar-nc [";" *ipchar-nc])
                      / ";" *ipchar-nc
                      ; non-zero-length segment without any colon ":"
                      ; No parameter (; delimited) inside path.

   ipchar         =  iunreserved / pct-encoded / sub-delims / ":" / "@"
   ipchar-nc      =  iunreserved / pct-encoded / sub-delims / "@"
                     ; sub-delims is different from RFC 3987
                     ; not including ";"

   iquery         =  < As defined in RFC 3987>
   iunreserved    =  < As defined in RFC 3987>
   pct-encoded    =  < As defined in RFC 3987>










Schulzrinne, et al.      Expires August 14, 2014              [Page 190]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   RTSP-URI       =  schemes ":" URI-rest
   RTSP-REQ-URI   =  schemes ":" URI-req-rest
   RTSP-URI-Ref   =  RTSP-URI / RTSP-Relative
   RTSP-REQ-Ref   =  RTSP-REQ-URI / RTSP-REQ-Rel
   schemes        =  "rtsp" / "rtsps" / scheme
   scheme         =  < As defined in RFC 3986>
   URI-rest       =  hier-part [ "?" query ]
   URI-req-rest   =  hier-part [ "?" query ]
                     ; Note fragment part not allowed in requests
   hier-part      =  "//" authority path-abempty

   RTSP-Relative  =  relative-part [ "?" query ]
   RTSP-REQ-Rel   =  relative-part [ "?" query ]
   relative-part  =  "//" authority path-abempty
                     / path-absolute
                     / path-noscheme
                     / path-empty

   authority      =  < As defined in RFC 3986>
   query          =  < As defined in RFC 3986>

   path           =  path-abempty    ; begins with "/" or is empty
                     / path-absolute ; begins with "/" but not "//"
                     / path-noscheme ; begins with a non-colon segment
                     / path-rootless ; begins with a segment
                     / path-empty    ; zero characters

   path-abempty   =  *( "/" segment )
   path-absolute  =  "/" [ segment-nz *( "/" segment ) ]
   path-noscheme  =  segment-nz-nc *( "/" segment )
   path-rootless  =  segment-nz *( "/" segment )
   path-empty     =  0<pchar>

   segment        =  *pchar [";" *pchar]
   segment-nz     =  ( 1*pchar [";" *pchar]) / (";" *pchar)
   segment-nz-nc  =  ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc)
                     ; non-zero-length segment without any colon ":"
                     ; No parameter (; delimited) inside path.

   pchar          =  unreserved / pct-encoded / sub-delims / ":" / "@"
   pchar-nc       =  unreserved / pct-encoded / sub-delims / "@"

   sub-delims     =  "!" / "$" / "&" / "'" / "(" / ")"
                     / "*" / "+" / "," / "="
                     ; sub-delims is different from RFC 3986/3987
                     ; not including ";"





Schulzrinne, et al.      Expires August 14, 2014              [Page 191]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   smpte-range        =  smpte-type [EQUAL smpte-range-spec]
                         ; See section 4.4
   smpte-range-spec   =  ( smpte-time "-" [ smpte-time ] )
                      /  ( "-" smpte-time )
   smpte-type         =  "smpte" / "smpte-30-drop"
                      /  "smpte-25" / smpte-type-extension
                      ; other timecodes may be added
   smpte-type-extension  =  "smpte" token
   smpte-time         =  1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
                         [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]

   npt-range        =  "npt" [EQUAL npt-range-spec]
   npt-range-spec   =  ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
   npt-time         =  "now" / npt-sec / npt-hhmmss / npt-hhmmss-comp
   npt-sec          =  1*19DIGIT [ "." 1*9DIGIT ]
   npt-hhmmss       =  npt-hh ":" npt-mm ":" npt-ss [ "." 1*9DIGIT ]
   npt-hh           =  2*19DIGIT   ; any positive number
   npt-mm           =  2*2DIGIT  ; 0-59
   npt-ss           =  2*2DIGIT  ; 0-59
   npt-hhmmss-comp  =  npt-hh-comp ":" npt-mm-comp ":" npt-ss-comp
                       [ "." 1*9DIGIT ] # Compatibility format
   npt-hh-comp      =  1*19DIGIT   ; any positive number
   npt-mm-comp      =  1*2DIGIT  ; 0-59
   npt-ss-comp      =  1*2DIGIT  ; 0-59

   utc-range        =  "clock" [EQUAL utc-range-spec]
   utc-range-spec   =  ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
   utc-time         =  utc-date "T" utc-clock "Z"
   utc-date         =  8DIGIT
   utc-clock        =  6DIGIT [ "." 1*9DIGIT ]

   feature-tag       =  token

   session-id        =  1*256( ALPHA / DIGIT / safe )

   extension-header  =  header-name HCOLON header-value
   header-name       =  token
   header-value      =  *(TEXT-UTF8char / LWS)

20.2.2.  Message Syntax











Schulzrinne, et al.      Expires August 14, 2014              [Page 192]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   RTSP-message  = Request / Response  ; RTSP/2.0 messages

   Request       = Request-Line
                   *((general-header
                   /  request-header
                   /  message-body-header) CRLF)
                   CRLF
                   [ message-body-data ]

   Response     = Status-Line
                  *((general-header
                  /  response-header
                  /  message-body-header) CRLF)
                  CRLF
                  [ message-body-data ]

   Request-Line = Method SP Request-URI SP RTSP-Version CRLF

   Status-Line  = RTSP-Version SP Status-Code SP Reason-Phrase CRLF

   Method  =  "DESCRIBE"
           /  "GET_PARAMETER"
           /  "OPTIONS"
           /  "PAUSE"
           /  "PLAY"
           /  "PLAY_NOTIFY"
           /  "REDIRECT"
           /  "SETUP"
           /  "SET_PARAMETER"
           /  "TEARDOWN"
           /  extension-method

   extension-method  =  token

   Request-URI  =  "*" / RTSP-REQ-URI
   RTSP-Version =  "RTSP/" 1*DIGIT "." 1*DIGIT

   message-body-data = 1*OCTET

   Status-Code  =  "100"  ; Continue
                /  "200"  ; OK
                /  "301"  ; Moved Permanently
                /  "302"  ; Found
                /  "303"  ; See Other
                /  "304"  ; Not Modified
                /  "305"  ; Use Proxy
                /  "400"  ; Bad Request
                /  "401"  ; Unauthorized



Schulzrinne, et al.      Expires August 14, 2014              [Page 193]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


                /  "402"  ; Payment Required
                /  "403"  ; Forbidden
                /  "404"  ; Not Found
                /  "405"  ; Method Not Allowed
                /  "406"  ; Not Acceptable
                /  "407"  ; Proxy Authentication Required
                /  "408"  ; Request Time-out
                /  "410"  ; Gone
                /  "412"  ; Precondition Failed
                /  "413"  ; Request Message Body Too Large
                /  "414"  ; Request-URI Too Large
                /  "415"  ; Unsupported Media Type
                /  "451"  ; Parameter Not Understood
                /  "452"  ; reserved
                /  "453"  ; Not Enough Bandwidth
                /  "454"  ; Session Not Found
                /  "455"  ; Method Not Valid in This State
                /  "456"  ; Header Field Not Valid for Resource
                /  "457"  ; Invalid Range
                /  "458"  ; Parameter Is Read-Only
                /  "459"  ; Aggregate operation not allowed
                /  "460"  ; Only aggregate operation allowed
                /  "461"  ; Unsupported Transport
                /  "462"  ; Destination Unreachable
                /  "463"  ; Destination Prohibited
                /  "464"  ; Data Transport Not Ready Yet
                /  "465"  ; Notification Reason Unknown
                /  "466"  ; Key Management Error
                /  "470"  ; Connection Authorization Required
                /  "471"  ; Connection Credentials not accepted
                /  "472"  ; Failure to establish secure connection
                /  "500"  ; Internal Server Error
                /  "501"  ; Not Implemented
                /  "502"  ; Bad Gateway
                /  "503"  ; Service Unavailable
                /  "504"  ; Gateway Time-out
                /  "505"  ; RTSP Version not supported
                /  "551"  ; Option not supported
                /  extension-code

   extension-code  =  3DIGIT

   Reason-Phrase   =  1*(TEXT-UTF8char / HT / SP)








Schulzrinne, et al.      Expires August 14, 2014              [Page 194]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   rtsp-header     = general-header
                   / request-header
                   / response-header
                   / message-body-header

   general-header  =  Accept-Ranges
                   /  Cache-Control
                   /  Connection
                   /  CSeq
                   /  Date
                   /  Media-Properties
                   /  Media-Range
                   /  Pipelined-Requests
                   /  Proxy-Supported
                   /  Range
                   /  RTP-Info
                   /  Scale
                   /  Seek-Style
                   /  Server
                   /  Session
                   /  Speed
                   /  Supported
                   /  Timestamp
                   /  Transport
                   /  User-Agent
                   /  Via
                   /  extension-header

   request-header  =  Accept
                   /  Accept-Credentials
                   /  Accept-Encoding
                   /  Accept-Language
                   /  Authorization
                   /  Bandwidth
                   /  Blocksize
                   /  From
                   /  If-Match
                   /  If-Modified-Since
                   /  If-None-Match
                   /  Notify-Reason
                   /  Proxy-Authorization
                   /  Proxy-Require
                   /  Referrer
                   /  Request-Status
                   /  Require
                   /  Terminate-Reason
                   /  extension-header




Schulzrinne, et al.      Expires August 14, 2014              [Page 195]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   response-header  =  Authentication-Info
                    /  Connection-Credentials
                    /  Location
                    /  MTag
                    /  Proxy-Authenticate
                    /  Proxy-Authentication-Info
                    /  Public
                    /  Retry-After
                    /  Unsupported
                    /  WWW-Authenticate
                    /  extension-header

   message-body-header    =  Allow
                    /  Content-Base
                    /  Content-Encoding
                    /  Content-Language
                    /  Content-Length
                    /  Content-Location
                    /  Content-Type
                    /  Expires
                    /  Last-Modified
                    /  extension-header

20.2.3.  Header Syntax

   Accept            =  "Accept" HCOLON
                        [ accept-range *(COMMA accept-range) ]
   accept-range      =  media-type-range [SEMI accept-params]
   media-type-range  =  ( "*/*"
                        / ( m-type SLASH "*" )
                        / ( m-type SLASH m-subtype )
                       ) *( SEMI m-parameter )
   accept-params     =  "q" EQUAL qvalue *(SEMI generic-param )
   qvalue            =  ( "0" [ "." *3DIGIT ] )
                     /  ( "1" [ "." *3("0") ] )
   Accept-Credentials =  "Accept-Credentials" HCOLON cred-decision
   cred-decision     =  ("User" [LWS cred-info])
                     /  "Proxy"
                     /  "Any"
                     /  (token [LWS 1*header-value])
                     ; For future extensions
   cred-info         =  cred-info-data *(COMMA cred-info-data)

   cred-info-data    =  DQUOTE RTSP-REQ-URI DQUOTE SEMI hash-alg
                        SEMI base64
   hash-alg          =  "sha-256" / extension-alg
   extension-alg     =  token
   Accept-Encoding   =  "Accept-Encoding" HCOLON



Schulzrinne, et al.      Expires August 14, 2014              [Page 196]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


                        [ encoding *(COMMA encoding) ]
   encoding          =  codings [SEMI accept-params]
   codings           =  content-coding / "*"
   content-coding    =  "identity" / token
   Accept-Language   =  "Accept-Language" HCOLON
                        language *(COMMA language)
   language          =  language-range [SEMI accept-params]
   language-range    =  language-tag / "*"
   language-tag      =  primary-tag *( "-" subtag )
   primary-tag       =  1*8ALPHA
   subtag            =  1*8ALPHA
   Accept-Ranges     =  "Accept-Ranges" HCOLON acceptable-ranges
   acceptable-ranges =  (range-unit *(COMMA range-unit))
   range-unit        =  "npt" / "smpte" / "smpte-30-drop" / "smpte-25"
                        / "clock" / extension-format
   extension-format  =  token
   Allow             =  "Allow" HCOLON Method *(COMMA Method)
   Authentication-Info = "Authentication-Info" HCOLON auth-info
   auth-info         = auth-info-entry *(COMMA auth-info-entry)
   auth-info-entry   = nextnonce
                     / message-qop
                     / response-auth
                     / cnonce
                     / nonce-count
   nextnonce         = "nextnonce" EQUAL nonce-value
   response-auth     = "rspauth" EQUAL response-digest
   response-digest   = DQUOTE *LHEX DQUOTE
   Authorization     =  "Authorization" HCOLON credentials
   credentials       =  basic-credential
                     /  digest-credential
                     /  other-response
   basic-credential  = "Basic" LWS basic-credentials
   basic-credentials = base64 ; Base64 encoding of user-password
   user-password     = basic-username ":" password
   basic-username    = *CF-TEXT
   CF-TEXT           = %x20-39 / %x3B-7E / %x80-FF ; TEXT without :
   password          = *TEXT
   digest-credential = ("Digest" LWS digest-response)
   digest-response   =  dig-resp *(COMMA dig-resp)
   dig-resp          =  username / realm / nonce / digest-uri
                     /  dresponse / algorithm / cnonce
                     /  opaque / message-qop
                     /  nonce-count / auth-param
   username          =  "username" EQUAL username-value
   username-value    =  quoted-string
   digest-uri        =  "uri" EQUAL LDQUOT digest-uri-value RDQUOT
   digest-uri-value  =  RTSP-REQ-URI
   message-qop       =  "qop" EQUAL qop-value



Schulzrinne, et al.      Expires August 14, 2014              [Page 197]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   cnonce            =  "cnonce" EQUAL cnonce-value
   cnonce-value      =  nonce-value
   nonce-count       =  "nc" EQUAL nc-value
   nc-value          =  8LHEX
   dresponse         =  "response" EQUAL request-digest
   request-digest    =  LDQUOT 32LHEX RDQUOT
   auth-param        =  auth-param-name EQUAL
                        ( token / quoted-string )
   auth-param-name   =  token
   other-response    =  auth-scheme LWS auth-param
                        *(COMMA auth-param)
   auth-scheme       =  token

   Bandwidth         =  "Bandwidth" HCOLON 1*19DIGIT

   Blocksize         =  "Blocksize" HCOLON 1*9DIGIT

   Cache-Control     =  "Cache-Control" HCOLON cache-directive
                        *(COMMA cache-directive)
   cache-directive   =  cache-rqst-directive
                     /  cache-rspns-directive

   cache-rqst-directive =  "no-cache"
                        /  "max-stale" [EQUAL delta-seconds]
                        /  "min-fresh" EQUAL delta-seconds
                        /  "only-if-cached"
                        /  cache-extension

   cache-rspns-directive =  "public"
                            /  "private"
                            /  "no-cache"
                            /  "no-transform"
                            /  "must-revalidate"
                            /  "proxy-revalidate"
                            /  "max-age" EQUAL delta-seconds
                            /  cache-extension

   cache-extension   =  token [EQUAL (token / quoted-string)]
   delta-seconds     =  1*19DIGIT

   Connection         =  "Connection" HCOLON connection-token
                         *(COMMA connection-token)
   connection-token   =  "close" / token

   Connection-Credentials = "Connection-Credentials" HCOLON cred-chain
   cred-chain         =  DQUOTE RTSP-REQ-URI DQUOTE SEMI base64

   Content-Base       =  "Content-Base" HCOLON RTSP-URI



Schulzrinne, et al.      Expires August 14, 2014              [Page 198]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Content-Encoding   =  "Content-Encoding" HCOLON
                         content-coding *(COMMA content-coding)
   Content-Language   =  "Content-Language" HCOLON
                         language-tag *(COMMA language-tag)
   Content-Length     =  "Content-Length" HCOLON 1*19DIGIT
   Content-Location   =  "Content-Location" HCOLON RTSP-REQ-Ref
   Content-Type       =  "Content-Type" HCOLON media-type
   media-type         =  m-type SLASH m-subtype *(SEMI m-parameter)
   m-type             =  discrete-type / composite-type
   discrete-type      =  "text" / "image" / "audio" / "video"
                      /  "application" / extension-token
   composite-type   =  "message" / "multipart" / extension-token
   extension-token  =  ietf-token / x-token
   ietf-token       =  token
   x-token          =  "x-" token
   m-subtype        =  extension-token / iana-token
   iana-token       =  token
   m-parameter      =  m-attribute EQUAL m-value
   m-attribute      =  token
   m-value          =  token / quoted-string

   CSeq           =  "CSeq" HCOLON cseq-nr
   cseq-nr        =  1*9DIGIT
   Date           =  "Date" HCOLON RTSP-date
   RTSP-date      =  date-time ;
   date-time      =  <As defined in RFC 5322>
   Expires        =  "Expires" HCOLON RTSP-date
   From           =  "From" HCOLON from-spec
   from-spec      =  ( name-addr / addr-spec ) *( SEMI from-param )
   name-addr      =  [ display-name ] LAQUOT addr-spec RAQUOT
   addr-spec      =  RTSP-REQ-URI / absolute-URI
   absolute-URI   =  < As defined in RFC 3986>
   display-name   =  *(token LWS) / quoted-string
   from-param     =  tag-param / generic-param
   tag-param      =  "tag" EQUAL token
   If-Match       =  "If-Match" HCOLON ("*" / message-tag-list)
   message-tag-list =  message-tag *(COMMA message-tag)
   message-tag      =  [ weak ] opaque-tag
   weak             =  "W/"
   opaque-tag       =  quoted-string
   If-Modified-Since  =  "If-Modified-Since" HCOLON RTSP-date
   If-None-Match    =  "If-None-Match" HCOLON ("*" / message-tag-list)
   Last-Modified    =  "Last-Modified" HCOLON RTSP-date
   Location         =  "Location" HCOLON RTSP-REQ-URI
   Media-Properties = "Media-Properties" HCOLON [media-prop-list]
   media-prop-list  = media-prop-value *(COMMA media-prop-value)
   media-prop-value = ("Random-Access" [EQUAL POS-FLOAT])
                    / "Beginning-Only"



Schulzrinne, et al.      Expires August 14, 2014              [Page 199]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


                    / "No-Seeking"
                    / "Immutable"
                    / "Dynamic"
                    / "Time-Progressing"
                    / "Unlimited"
                    / ("Time-Limited" EQUAL utc-time)
                    / ("Time-Duration" EQUAL POS-FLOAT)
                    / ("Scales" EQUAL scale-value-list)
                    / media-prop-ext
   media-prop-ext   = token [EQUAL (1*rtsp-unreserved / quoted-string)]
   scale-value-list = DQUOTE scale-entry *(COMMA scale-entry) DQUOTE
   scale-entry      = scale-value / (scale-value COLON scale-value)
   scale-value      = FLOAT
   Media-Range      = "Media-Range" HCOLON [ranges-list]
   ranges-list      =  ranges-spec *(COMMA ranges-spec)
   MTag             =  "MTag" HCOLON message-tag
   Notify-Reason    = "Notify-Reason" HCOLON Notify-Reas-val
   Notify-Reas-val  = "end-of-stream"
                    / "media-properties-update"
                    / "scale-change"
                    / Notify-Reason-extension
   Notify-Reason-extension  = token
   Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id
   startup-id  = 1*8DIGIT



























Schulzrinne, et al.      Expires August 14, 2014              [Page 200]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


Proxy-Authenticate =  "Proxy-Authenticate" HCOLON challenge-list
challenge-list     =  challenge *(COMMA challenge)
challenge          =  ("Digest" LWS digest-cln *(COMMA digest-cln))
                   /  ("Basic" LWS realm)
                   /  other-challenge
other-challenge    =  auth-scheme LWS auth-param
                      *(COMMA auth-param)
digest-cln         =  realm / domain / nonce
                   /  opaque / stale / algorithm
                   /  qop-options / auth-param
realm              =  "realm" EQUAL realm-value
realm-value        =  quoted-string
domain             =  "domain" EQUAL LDQUOT RTSP-REQ-Ref
                      *(1*SP RTSP-REQ-Ref ) RDQUOT
nonce              =  "nonce" EQUAL nonce-value
nonce-value        =  quoted-string
opaque             =  "opaque" EQUAL quoted-string
stale              =  "stale" EQUAL ( "true" / "false" )
algorithm          =  "algorithm" EQUAL ("MD5" / "MD5-sess" / token)
qop-options        =  "qop" EQUAL LDQUOT qop-value
                      *("," qop-value) RDQUOT
qop-value          =  "auth" / "auth-int" / token
Proxy-Authentication-Info = "Proxy-Authentication-Info" HCOLON auth-info
Proxy-Authorization = "Proxy-Authorization" HCOLON credentials
Proxy-Require      =  "Proxy-Require" HCOLON feature-tag-list
feature-tag-list   =  feature-tag *(COMMA feature-tag)
Proxy-Supported    =  "Proxy-Supported" HCOLON [feature-tag-list]

Public             =  "Public" HCOLON Method *(COMMA Method)

Range              =  "Range" HCOLON ranges-spec

ranges-spec        =  npt-range / utc-range / smpte-range
                   /  range-ext
range-ext          =  extension-format [EQUAL range-value]
range-value        =  1*(rtsp-unreserved / quoted-string / ":" )

Referrer           =  "Referrer" HCOLON (absolute-URI / RTSP-URI-Ref)
Request-Status     =  "Request-Status" HCOLON req-status-info
req-status-info    =  cseq-info LWS status-info LWS reason-info
cseq-info          =  "cseq" EQUAL cseq-nr
status-info        =  "status" EQUAL Status-Code
reason-info        =  "reason" EQUAL DQUOTE Reason-Phrase DQUOTE
Require            =  "Require" HCOLON feature-tag-list







Schulzrinne, et al.      Expires August 14, 2014              [Page 201]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   RTP-Info         =  "RTP-Info" HCOLON [rtsp-info-spec
                       *(COMMA rtsp-info-spec)]
   rtsp-info-spec   =  stream-url 1*ssrc-parameter
   stream-url       =  "url" EQUAL DQUOTE RTSP-REQ-Ref DQUOTE
   ssrc-parameter   =  LWS "ssrc" EQUAL ssrc HCOLON
                       ri-parameter *(SEMI ri-parameter)
   ri-parameter     =  ("seq" EQUAL 1*5DIGIT)
                    /  ("rtptime" EQUAL 1*10DIGIT)
                    /  generic-param

   Retry-After      =  "Retry-After" HCOLON (RTSP-date / delta-seconds)
   Scale            =  "Scale" HCOLON scale-value
   Seek-Style       =  "Seek-Style" HCOLON Seek-S-values
   Seek-S-values    =  "RAP"
                    /  "CoRAP"
                    /  "First-Prior"
                    /  "Next"
                    /  Seek-S-value-ext
   Seek-S-value-ext =  token

   Server           =  "Server" HCOLON ( product / comment )
                       *(LWS (product / comment))
   product          =  token [SLASH product-version]
   product-version  =  token
   comment          =  LPAREN *( ctext / quoted-pair) RPAREN

   Session          =  "Session" HCOLON session-id
                       [ SEMI "timeout" EQUAL delta-seconds ]

   Speed            =  "Speed" HCOLON lower-bound MINUS upper-bound
   lower-bound      =  POS-FLOAT
   upper-bound      =  POS-FLOAT

   Supported        =  "Supported" HCOLON [feature-tag-list]

















Schulzrinne, et al.      Expires August 14, 2014              [Page 202]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Terminate-Reason      =  "Terminate-Reason" HCOLON TR-Info
   TR-Info              =  TR-Reason *(SEMI TR-Parameter)
   TR-Reason            =  "Session-Timeout"
                        /  "Server-Admin"
                        /  "Internal-Error"
                        /  token
   TR-Parameter         =  TR-time / TR-user-msg / generic-param
   TR-time              =  "time" EQUAL utc-time
   TR-user-msg          =  "user-msg" EQUAL quoted-string

   Timestamp        =  "Timestamp" HCOLON timestamp-value [LWS delay]
   timestamp-value  =  *19DIGIT [ "." *9DIGIT ]
   delay            =  *9DIGIT [ "." *9DIGIT ]

   Transport        =  "Transport" HCOLON transport-spec
                       *(COMMA transport-spec)
   transport-spec   =  transport-id *trns-parameter
   transport-id     =  trans-id-rtp / other-trans
   trans-id-rtp     =  "RTP/" profile ["/" lower-transport]
                       ; no LWS is allowed inside transport-id
   other-trans      =  token *("/" token)






























Schulzrinne, et al.      Expires August 14, 2014              [Page 203]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   profile           = "AVP" / "SAVP" / "AVPF" / "SAVPF" / token
   lower-transport   = "TCP" / "UDP" / token
   trns-parameter    = (SEMI ( "unicast" / "multicast" ))
                     / (SEMI "interleaved" EQUAL channel ["-" channel])
                     / (SEMI "ttl" EQUAL ttl)
                     / (SEMI "layers" EQUAL 1*DIGIT)
                     / (SEMI "ssrc" EQUAL ssrc *(SLASH ssrc))
                     / (SEMI "mode" EQUAL mode-spec)
                     / (SEMI "dest_addr" EQUAL addr-list)
                     / (SEMI "src_addr" EQUAL addr-list)
                     / (SEMI "setup" EQUAL contrans-setup)
                     / (SEMI "connection" EQUAL contrans-con)
                     / (SEMI "RTCP-mux")
                     / (SEMI "MIKEY" EQUAL MIKEY-Value)
                     / (SEMI trn-param-ext)
   contrans-setup    = "active" / "passive" / "actpass"
   contrans-con      = "new" / "existing"
   trn-param-ext     = par-name [EQUAL trn-par-value]
   par-name          = token
   trn-par-value     = *(rtsp-unreserved / quoted-string)
   ttl               = 1*3DIGIT ; 0 to 255
   ssrc              = 8HEX
   channel           = 1*3DIGIT ; 0 to 255
   MIKEY-Value       = base64
   mode-spec         = ( DQUOTE mode *(COMMA mode) DQUOTE )
   mode              = "PLAY" / token
   addr-list         = quoted-addr *(SLASH quoted-addr)
   quoted-addr       = DQUOTE (host-port / extension-addr) DQUOTE
   host-port         = ( host [":" port] )
                     / ( ":" port )
   extension-addr    = 1*qdtext
   host              = < As defined in RFC 3986>
   port              = < As defined in RFC 3986>


















Schulzrinne, et al.      Expires August 14, 2014              [Page 204]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Unsupported     = "Unsupported" HCOLON feature-tag-list
   User-Agent      = "User-Agent" HCOLON ( product / comment )
                     *(LWS (product / comment))
   Via             = "Via" HCOLON via-parm *(COMMA via-parm)
   via-parm        = sent-protocol LWS sent-by *( SEMI via-params )
   via-params      = via-ttl / via-maddr
                   / via-received / via-extension
   via-ttl         = "ttl" EQUAL ttl
   via-maddr       = "maddr" EQUAL host
   via-received    = "received" EQUAL (IPv4address / IPv6address)
   IPv4address     = < As defined in RFC 3986>
   IPv6address     = < As defined in RFC 3986>
   via-extension   = generic-param
   sent-protocol   = protocol-name SLASH protocol-version
                     SLASH transport-prot
   protocol-name   = "RTSP" / token
   protocol-version = token
   transport-prot  = "UDP" / "TCP" / "TLS" / other-transport
   other-transport = token
   sent-by         = host [ COLON port ]

   WWW-Authenticate = "WWW-Authenticate" HCOLON challenge-list

20.3.  SDP extension Syntax

   This section defines in ABNF the SDP extensions defined for RTSP.
   See Appendix D for the definition of the extensions in text.

   control-attribute   =  "a=control:" *SP RTSP-REQ-Ref CRLF

   a-range-def         =  "a=range:" ranges-spec CRLF

   a-mtag-def          =  "a=mtag:" message-tag CRLF

21.  Security Considerations

   The security considerations and threats around RTSP and its usage can
   be divided into considerations around the signaling protocol itself
   and the issues related to the media stream delivery.  However, when
   it comes to mitigations of security threats, a threat depending on
   the media stream delivery may in fact be mitigated by a mechanism in
   the signaling protocol.

   There are several chapters and an appendix in this document that
   define security solutions for the protocol.  These sections will be
   referenced when discussing the threats below.  But the reader should
   take special notice of the Security Framework (Section 19) and the




Schulzrinne, et al.      Expires August 14, 2014              [Page 205]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   specification of how to use SRTP and its key-mangement
   (Appendix C.1.4) to achieve certain aspects of the media security.

21.1.  Signaling Protocol Threats

   This section focuses on issues related to the signaling protocol.
   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security considerations outlined in [H15] apply
   also.

   Specifically, please note the following:

   Abuse of Server Log Information:  A server is in the position to save
         personal data about a user's requests which might identify
         their media consumption patterns or subjects of interest.  This
         information is clearly confidential in nature and its handling
         can be constrained by law in certain countries.  RTSP servers
         will presumably have similar logging mechanisms to HTTP, and
         thus should be equally guarded in protecting the contents of
         those logs, thus protecting the privacy of the users of the
         servers.  People using the RTSP protocol to provide media are
         responsible for ensuring that logging material is not
         distributed without the permission of any individuals that are
         identifiable by the published results.

   Transfer of Sensitive Information:  There is no reason to believe
         that information transferred in RTSP message, such as the URI
         and the content of headers, especially the Server, Via,
         Referrer and From headers, may be any less sensitive than when
         used in HTTP.  Therefore, all of the precautions regarding the
         protection of data privacy and user privacy apply to
         implementors of RTSP clients, servers, and proxies.  See
         [H15.1.2] for further details.

         The RTSP methods defined in this document is primarily used to
         establish and control the delivery of the media data
         represented by the URI, thus the RTSP message bodies are
         generally less sensitive than the ones in HTTP.  Where HTTP
         bodies could contain for example your medical records, in RTSP
         the sensitive video of your medical operation would be in the
         media stream over the media transport protocol, not in the RTSP
         message.  Still one have to take note of what potential
         sensitive informative are included in the RTSP protocol.  The
         protection of the media data is separate, can be applied
         directly between client and server, and is dependent on the
         media transport protocol in use.  See Section 21.2 for further
         discussion.  This possibility for separation of security
         between media resource content and the signalling protocol



Schulzrinne, et al.      Expires August 14, 2014              [Page 206]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         mitigates the risk of exposing the media content when using
         hop-by-hop security for RTSP signaling using proxies
         (Section 19.3).

   Attacks Based On File and Path Names:  Though RTSP URIs are opaque
         handles that do not necessarily have file system semantics, it
         is anticipated that many implementations will translate
         portions of the Request-URIs directly to file system calls.  In
         such cases, file systems SHOULD follow the precautions outlined
         in [H15.2], such as checking for ".." in path components.

   Personal Information:  RTSP clients are often privy to the same
         information that HTTP clients are (user name, location, etc.)
         and thus should be equally sensitive.  See [H15.1] for further
         recommendations.

   Privacy Issues Connected to Accept Headers:  Since similar usages of
         the "Accept" headers exist in RTSP as in HTTP, the same caveats
         outlined in [H15.1.4] with regards to their use should be
         followed.

   DNS Spoofing:  Presumably, given the longer connection times
         typically associated to RTSP sessions relative to HTTP
         sessions, RTSP client DNS optimizations should be less
         prevalent.  Nonetheless, the recommendations provided in
         [H15.3] are still relevant to any implementation which attempts
         to rely on a DNS-to-IP mapping to hold beyond a single use of
         the mapping.

   Location Headers and Spoofing:  If a single server supports multiple
         organizations that do not trust each another, then it MUST
         check the values of the Content-Location header fields in
         responses that are generated under control of said
         organizations to make sure that they do not attempt to
         invalidate resources over which they have no authority.
         ([H15.4])

   In addition to the recommendations in the current HTTP specification
   (RFC 2616 [RFC2616], as of this writing) and also of the previous RFC
   2068 [RFC2068], future HTTP specifications may provide additional
   guidance on security issues.

   The following are added considerations for RTSP implementations.

   Session hijacking:  Since there is no or little relation between a
         transport layer connection and an RTSP session, it is possible
         for a malicious client to issue requests with random session
         identifiers which could affect other clients of an unsuspecting



Schulzrinne, et al.      Expires August 14, 2014              [Page 207]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         server.  To mitigate this the server SHALL use a large, random
         and non-sequential session identifier to minimize the
         possibility of this kind of attack.  However, unless the RTSP
         signaling is always confidentiality protected, e.g., using TLS,
         an on-path attacker will be able to hijack a session.  Another
         choice for preventing session hijacking is to use client
         authentication and only allow the authenticated client creating
         the session to access that session.

   Authentication:  Servers SHOULD implement both basic and digest
         [RFC2617] authentication.  In environments requiring tighter
         security for the control messages, the transport layer
         mechanism TLS [RFC5246] SHOULD be used.

   Suspicious behavior:  RTSP servers upon detecting instances of
         behavior which is deemed a security risk SHOULD return error
         code 403 (Forbidden).  RTSP servers SHOULD also be aware of
         attempts to probe the server for weaknesses and entry points
         and MAY arbitrarily disconnect and ignore further requests from
         clients which are deemed to be in violation of local security
         policy.

   TLS through proxies:  If one uses the possibility to connect TLS in
         multiple legs (Section 19.3) one really needs to be aware of
         the trust model.  That procedure requires full faith and trust
         in all proxies, which will be identified, that one allows to
         connect through.  They are men in the middle and have access to
         all that goes on over the TLS connection.  Thus it is important
         to consider if that trust model is acceptable in the actual
         application.  Further discussion of the actual trust model is
         in Section 19.3.  It is important to note what difference in
         security properties, if any, that may exist with the used media
         transport protocol and its security mechanism.  Using SRTP and
         the MIKEY based key-establishment defined in Appendix C.1.4.1,
         enables to media key-establishment to done end-to-end without
         revealing the keys to the proxies.

   Resource Exhaustion:  As RTSP is a stateful protocol and establishes
         resource usage on the server there is a clear possibility to
         attack the server by trying to overbook these resources to
         perform a denial of service attack.  This attack can be both
         against ongoing sessions and to prevent others from
         establishing sessions.  RTSP agents will need to have
         mechanisms to prevent single peers from consuming extensive
         amounts of resources.  The methods for guarding against this
         are varied and depends on the agent's role and capabilities and
         policies.  Each implementation has to carefully consider their
         methods and policies to mitigate this threat.  For example



Schulzrinne, et al.      Expires August 14, 2014              [Page 208]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         regarding handling of connections there are recommendations in
         Section 10.7.

   The above threats and considerations have resulted in a set of
   security functions and mechanisms built into or used by the protocol.
   The signaling protocol relies on two security features defined in the
   Security Framework (Section 19) namely client authentication using
   HTTP authentication and TLS based transport protection of the
   signaling messages.  Both of these mechanisms are required to be
   implemented by any RTSP agent.

   A number of different security mitigations have been designed into
   the protocol and will be instantiated if the specification is
   implemented as written, for example by ensuring sufficient amount of
   entropy in the randomly generated session identifiers when not using
   client authentication to minimize the risk of session hijacking.
   When client authentication is used the protection against hijacking
   will be greatly improved by scoping the accessible sessions to the
   one this client identity has created.  Some of the above threats are
   such that the implementation of the RTSP functionality itself needs
   to consider which policy and strategy it uses to mitigate them.

21.2.  Media Stream Delivery Threats

   The fact that RTSP establishes and controls a media stream delivery
   results in a set of security issues related to the media streams.
   This section will attempt to analyze general threats, however the
   choice of media stream transport protocol, such as RTP will result in
   some differences in threats and what mechanisms exist to mitigate
   them.  Thus it becomes important that each specification of a new
   media stream transport and delivery protocol usable by RTSP requires
   its own security analysis.  This section includes one for RTP.

   The set of general threats from or by the media stream delivery
   itself are:

   Concentrated denial-of-service attack:  The protocol offers the
      opportunity for a remote-controlled denial-of-service (DoS)
      attack, where the media stream is the hammer in that DoS attack.
      See Section 21.2.1.

   Media Confidentiality:  The media delivery may contain content of any
      type and it is not possible in general to determine how sensitive
      this content is from a confidentiality point.  Thus it is a strong
      requirement that any media delivery protocol provides a method for
      providing confidentiality of the actual media content.  In
      addition to the media level confidentiality it becomes critical
      that no resource identifiers used in the signaling are exposed to



Schulzrinne, et al.      Expires August 14, 2014              [Page 209]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      an attacker as they may have human understandable names, or may be
      also available to the attacker so they can determine the content
      the user was delivered.  Thus the signaling protocol must also
      provide confidentiality protection of any information related to
      the media resource.

   Media Integrity and Authentication:  There are several reasons, such
      as discrediting the target, misinformation of the target, why an
      attacker will be interested in substituting the media stream sent
      out from the RTSP server with one of the attacker's creation or
      selection.  Therefore it is important that the media protocol
      provides mechanisms to verify the source authentication, integrity
      and prevent replay attacks on the media stream.

   Scope of Multicast:  If RTSP is used to control the transmission of
      media onto a multicast network the scope of the delivery must be
      considered.  RTSP supports the TTL Transport header parameter to
      indicate this scope for IPv4.  IPv6 has a different mechanism for
      scope boundary.  However, such scope control has risks, as it may
      be set too large and distribute media beyond the intended scope.

   Below (Section 21.2.2) a protocol specific analysis of security
   considerations for RTP based media transport is done.  In that
   section it is also made clear the requirements on implementing
   security functions for RTSP agents supporting media delivery over
   RTP.

21.2.1.  Remote Denial of Service Attack

   The attacker may initiate traffic flows to one or more IP addresses
   by specifying them as the destination in SETUP requests.  While the
   attacker's IP address may be known in this case, this is not always
   useful in prevention of more attacks or ascertaining the attacker's
   identity.  Thus, an RTSP server MUST only allow client-specified
   destinations for RTSP-initiated traffic flows if the server has
   ensured that the specified destination address accepts receiving
   media through different security mechanisms.  Security mechanisms
   that are acceptable in order of increasing generality are:

   o  Verification of the client's identity against a database of known
      users using RTSP authentication mechanisms (preferably digest
      authentication or stronger)

   o  A list of addresses that have consented to be media destinations,
      especially considering user identity

   o  Media path based verification




Schulzrinne, et al.      Expires August 14, 2014              [Page 210]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   The server SHOULD NOT allow the destination field to be set unless a
   mechanism exists in the system to authorize the request originator to
   direct streams to the recipient.  It is preferred that this
   authorization be performed by the media recipient (destination)
   itself and the credentials passed along to the server.  However, in
   certain cases, such as when the recipient address is a multicast
   group, or when the recipient is unable to communicate with the server
   in an out-of-band manner, this may not be possible.  In these cases
   the server may chose another method such as a server-resident
   authorization list to ensure that the request originator has the
   proper credentials to request stream delivery to the recipient.

   One solution that performs the necessary verification of acceptance
   of media suitable for unicast based delivery is the Interactive
   Connectivity Establishment (ICE) [RFC5245] based NAT traversal method
   described in [I-D.ietf-mmusic-rtsp-nat].  This mechanism uses random
   passwords and a username so that the probability of unintended
   indication as a valid media destination is very low.  In addition the
   server includes in its Session Traversal Utilities for NAT (STUN)
   [RFC5389] requests a cookie (consisting of random material) that the
   destination echoes back, thus the solution also safe-guards against
   having an off-path attacker being able to spoof the STUN checks.
   This leaves this solution vulnerable only to on-path attackers that
   can see the STUN requests go to the target of attack and thus forge a
   response.

   For delivery to multicast addresses there is a need for another
   solution which is not specified in this memo.

21.2.2.  RTP Security analysis

   RTP is a commonly used media transport protocol and has been the most
   common choice for RTSP 1.0 implementations.  The core RTP protocol
   has been in use for a long time and it has well-known security
   properties and the RTP security consideration (Section 9 of
   [RFC3550]) needs to be reviewed.  In perspective of the usage of RTP
   in context of RTSP the following properties should be noted:

   Stream Additions:  RTP has support for multiple simultaneous media
      streams in each RTP session.  As some use cases require support
      for non-synchronized adding and removal of media streams and their
      identifiers an attacker can easily insert additional media streams
      into a session context that according to protocol design is
      intended to be played out.  Another threat vector is one of denial
      of service by exhausting the resources of the RTP session
      receiver, for example by using a large number of SSRC identifiers
      simultaneously.  The strong mitigation of this is to ensure that
      one cryptographically authenticates any incoming packet flow to



Schulzrinne, et al.      Expires August 14, 2014              [Page 211]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      the RTP session.  Weak mitigations like blocking additional media
      streams in session contexts easily lead to a denial of service
      vulnerability in addition to preventing certain RTP extensions or
      use cases which rely on multiple media streams, such as RTP
      retransmission [RFC4588] to function.

   Forged Feedback:  The built in RTP control Protocol (RTCP) also
      offers a large attack surface for a couple of different types of
      attacks.  One venue is to send RTCP feedback to the media sender
      indicating large amounts of packet loss and thus trigger a media
      bit-rate adaptation response from the sender resulting in lowered
      media quality and potentially shut down of the media stream.
      Another attack is to perform a resource exhaustion attack on the
      receiver by using many SSRC identifiers to create large state
      tables and increase the RTCP related processing demands.

   RTP/RTCP Extensions:  RTP and RTCP extensions generally provide
      additional and sometimes extremely powerful tools to do denial of
      service or service disruption.  For example the Code Control
      Message [RFC5104] RTCP extensions enables both locking down the
      bit-rate to low values and disruption of video quality by
      requesting Intra frames.

   Taking into account the above general discussion in Section 21.2 and
   the RTP specific discussion in this section it is clear that it is
   necessary that a strong security mechanism is supported to protect
   RTP.  Therefore this specification has the following requirements on
   RTP security functions for all RTSP agents that handles media streams
   and where the media stream transport is done using RTP.

   RTSP agents supporting RTP MUST implement Secure RTP (SRTP) [RFC3711]
   and thus the SAVP profile.  In addition the secure AVP profile
   (SAVPF) [RFC5124] MUST also be supported if the AVPF profile is
   implemented.  This specification requires no additional cryptographic
   transforms or configuration values beyond those specified as
   mandatory to implement in RFC3711, i.e., AES-CM and HMAC-SHA1.  The
   default key-management mechanism which MUST be implemented is the one
   defined in the MIKEY Key Establishment (Appendix C.1.4.1).  The MIKEY
   implementation MUST implement the necessary functions for MIKEY-RSA-R
   mode [RFC4738] and in addition the SRTP parameter negotiation
   necessary to negotiate the supported SRTP transforms and parameters.

22.  IANA Considerations

   This section sets up a number of registries for RTSP 2.0 that should
   be maintained by IANA.  These registries are separate from any
   registries existing for RTSP 1.0.  For each registry there is a
   description of what it is required to contain, what specification is



Schulzrinne, et al.      Expires August 14, 2014              [Page 212]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   needed when adding an entry with IANA, and finally the entries that
   this document needs to register.  See also the Section 2.7 "Extending
   RTSP".  There is also an IANA registration of three SDP attributes.

   Registries or entries in registries which have been made for RTSP 1.0
   are not moved to RTSP 2.0.  The registries and entries in registries
   of RTSP 1.0 and RTSP 2.0 are independent.  If any registry or entry
   in a registry is also required in RTSP 2.0, it MUST follow the
   procedure defined below to allocate the registry or entry in a
   registry.

   The sections describing how to register an item uses some of the
   registration policies described in RFC 5226 [RFC5226], namely "First
   Come, First Served", "Expert Review, "Specification Required", and
   "Standards Action".

      RFC-Editor Note: Please replace all occurrences of RFCXXXX with
      the RFC number this specification receives when published.

   In case a registry requires a contact person, the authors, with
   Magnus Westerlund (magnus.westerlund@ericsson.com) as primary, are
   the contact persons for any entries created by this document.

   IANA will request the following information for any registration
   request:

   o  A name of the item to register according to the rules specified by
      the intended registry.

   o  Indication of who has change control over the feature (for
      example, IETF, ISO, ITU-T, other international standardization
      bodies, a consortium, a particular company or group of companies,
      or an individual);

   o  A reference to a further description, if available, for example
      (in decreasing order of preference) an RFC, a published standard,
      a published paper, a patent filing, a technical report, documented
      source code or a computer manual;

   o  For proprietary features, contact information (postal and email
      address);

22.1.  Feature-tags








Schulzrinne, et al.      Expires August 14, 2014              [Page 213]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


22.1.1.  Description

   When a client and server try to determine what part and functionality
   of the RTSP specification and any future extensions that its counter
   part implements there is need for a namespace.  This registry
   contains named entries representing certain functionality.

   The usage of feature-tags is explained in Section 11 and
   Section 13.1.

22.1.2.  Registering New Feature-tags with IANA

   The registering of feature-tags is done on a First Come, First Served
   [RFC5226] basis.

   The registry entry for a feature-tag has the following information:

   o  The name of the feature-tag

      *  If the registrant indicates that the feature is proprietary,
         IANA should request a vendor "prefix" portion of the name.  The
         name will then be the vendor prefix followed by a "." followed
         by the rest of the provided feature name.

      *  If the feature is not proprietary, then IANA need not collect a
         prefix for the name.

   o  A one paragraph description of what the feature-tag represents

   o  The applicability (server, client, proxy, or some combination)

   o  A reference to a specification, if applicable

   Feature-tag names (including the vendor prefix) may contain any non-
   space and non-control characters.  There is no length limit on
   feature-tags.

   Examples for a vendor tag describing a proprietary feature are:

         vendorA.specfeat01

         vendorA.specfeat02

22.1.3.  Registered entries

   The following feature-tags are defined in this specification and
   hereby registered.  The change control belongs to the IETF.




Schulzrinne, et al.      Expires August 14, 2014              [Page 214]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   play.basic:  The implementation for delivery and playback operations
         according to the core RTSP specification, as defined in this
         memo.  Applies for both clients, servers and proxies.  See
         Section 11.1.

   play.scale:  Support of scale operations for media playback.  Applies
         only for servers.  See Section 18.46.

   play.speed:  Support of the speed functionality for media delivery.
         Applies only for servers.  See Section 18.50.

   setup.rtp.rtcp.mux  Support of the RTP and RTCP multiplexing as
         discussed in Appendix C.1.6.4.  Applies for both client and
         servers and any media caching proxy.

   This should be represented by IANA as a table with the feature tags,
   contact person and their references.

22.2.  RTSP Methods

22.2.1.  Description

   Methods are described in Section 13.  Extending the protocol with new
   methods allow for totally new functionality.

22.2.2.  Registering New Methods with IANA

   A new method is registered through an IETF Standards Action
   [RFC5226].  The reason is that new methods may radically change the
   protocol's behavior and purpose.

   A specification for a new RTSP method consist of the following items:

   o  A method name which follows the ABNF rules for methods.

   o  A clear specification what a request using the method does and
      what responses are expected.  Which directions the method is used,
      C->S or S->C or both.  How the use of headers, if any, modifies
      the behavior and effect of the method.

   o  A list or table specifying which of the IANA registered headers
      that are allowed to be used with the method in request or/and
      response.  The list or table SHOULD follow the format of tables in
      Section 18.

   o  Describe how the method relates to network proxies.





Schulzrinne, et al.      Expires August 14, 2014              [Page 215]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


22.2.3.  Registered Entries

   This specification, RFCXXXX, registers 10 methods: DESCRIBE,
   GET_PARAMETER, OPTIONS, PAUSE, PLAY, PLAY_NOTIFY, REDIRECT, SETUP,
   SET_PARAMETER, and TEARDOWN.  The initial table of the registry is
   provided below.

   Method         Directionality           Reference
   -----------------------------------------------------
   DESCRIBE       C->S                     [RFCXXXX]
   GET_PARAMETER  C->S, S->C               [RFCXXXX]
   OPTIONS        C->S, S->C               [RFCXXXX]
   PAUSE          C->S                     [RFCXXXX]
   PLAY           C->S                     [RFCXXXX]
   PLAY_NOTIFY    S->C                     [RFCXXXX]
   REDIRECT       S->C                     [RFCXXXX]
   SETUP          C->S                     [RFCXXXX]
   SET_PARAMETER  C->S, S->C               [RFCXXXX]
   TEARDOWN       C->S, S->C               [RFCXXXX]

22.3.  RTSP Status Codes

22.3.1.  Description

   A status code is the three digit number used to convey information in
   RTSP response messages, see Section 8.  The number space is limited
   and care should be taken not to fill the space.

22.3.2.  Registering New Status Codes with IANA

   A new status code registration follows the policy of IETF Review
   [RFC5226].  New RTSP functionality requiring Status Codes should
   first be registered in the range x50-x99.  Only when the range is
   full should registrations be done in the x00-x49 range, unless it is
   to adopt an HTTP extension also to RTSP.  The reason is to enable any
   HTTP extension to be adopted to RTSP without needing to renumber any
   related status codes.  A specification for a new status code specify
   the following:

   o  The registered number.

   o  A description of what the status code means and the expected
      behavior of the sender and receiver of the code.








Schulzrinne, et al.      Expires August 14, 2014              [Page 216]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


22.3.3.  Registered Entries

   RFCXXXX, registers the numbered status code defined in the ABNF entry
   "Status-Code" except "extension-code" (that defines the syntax
   allowed for future extensions) in Section 20.2.2.

22.4.  RTSP Headers

22.4.1.  Description

   By specifying new headers a method(s) can be enhanced in many
   different ways.  An unknown header will be ignored by the receiving
   agent.  If the new header is vital for a certain functionality, a
   feature-tag for the functionality can be created and demanded to be
   used by the counter-part with the inclusion of a Require header
   carrying the feature-tag.

22.4.2.  Registering New Headers with IANA

   Registrations in the registry can be done following the Expert Review
   policy [RFC5226].  A specification is recommended to be provided,
   preferably an IETF RFC or other Standards Developing Organization
   specification.  The minimal information in a registration request is
   the header name and the contact information.

   The expert reviewer verifies that the registration request contain
   the following information:

   o  The name of the header.

   o  An ABNF specification of the header syntax.

   o  A list or table specifying when the header may be used,
      encompassing all methods, their request or response, the direction
      (C->S or S->C).

   o  How the header is to be handled by proxies.

   o  A description of the purpose of the header.

22.4.3.  Registered entries

   All headers specified in Section 18 in RFCXXXX are to be registered.
   The Registry is to include header name and reference.

   Furthermore the following legacy RTSP headers defined in other
   specifications are registered with header name, reference and
   description according to below list.  Note: These references may not



Schulzrinne, et al.      Expires August 14, 2014              [Page 217]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   fulfill all of the above rules for registrations due to their legacy
   status.

   o  x-wap-profile defined in [TS-26234].  The x-wap-profile request-
      header contains one or more absolute URLs to the requesting
      agent's device capability profile.

   o  x-wap-profile-diff defined in [TS-26234].  The x-wap-profile-diff
      request-header contains a subset of a device capability profile.

   o  x-wap-profile-warning defined in [TS-26234].  The x-wap-profile-
      warning is a response-header that contains error codes explaining
      to what extent the server has been able to match the terminal
      request in regards to device capability profile as described using
      x-wap-profile and x-wap-profile-diff headers.

   o  x-predecbufsize defined in [TS-26234].  This response-header
      provides an RTSP agent with the TS 26.234 Annex G hypothetical
      pre-decoder buffer size.

   o  x-initpredecbufperiod defined in [TS-26234].  This response-header
      provides an RTSP agent with the TS 26.234 Annex G hypothetical
      pre-decoder buffering period.

   o  x-initpostdecbufperiod defined in [TS-26234].  This response-
      header provides an RTSP agent with the TS 26.234 Annex G post-
      decoder buffering period.

   o  3gpp-videopostdecbufsize defined in [TS-26234].  This response-
      header provides an RTSP agent with the TS 26.234 defined post-
      decoder buffer size usable for H.264 (AVC) video streams.

   o  3GPP-Link-Char defined in [TS-26234].  This request-header
      provides the RTSP server with the RTSP client's link
      characteristics as determined from the radio interface.  The
      information that can be provided are guaranteed bit-rate, maximum
      bit-rate and maximum transfer delay.

   o  3GPP-Adaptation defined in [TS-26234].  This general-header is
      part of the bit-rate adaptation solution specified for PSS.  It
      provides the RTSP client's buffer sizes and target buffer levels
      to the server and responses are used to acknowledge the support
      and values.

   o  3GPP-QoE-Metrics defined in [TS-26234].  This general-header is
      used by PSS RTSP agents to negotiate the quality of experience
      metrics that a client should gather and report to the server.




Schulzrinne, et al.      Expires August 14, 2014              [Page 218]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  3GPP-QoE-Feedback defined in [TS-26234].  This request-header is
      used by RTSP clients supporting PSS to report the actual values of
      the metrics gathered in its quality of experience metering.

   The use of "x-" is NOT RECOMMENDED but the above headers in the
   register list were defined prior to the clarification.

22.5.  Accept-Credentials

   The security framework's TLS connection mechanism has two
   registerable entities.

22.5.1.  Accept-Credentials policies

   This registry are for polices for a RTSP proxy's handling and
   verification of TLS certificates when establishing outbound TLS
   connection on clients behalf.  In Section 19.3.1 three policies for
   how to handle certificates are specified.  Further policies may be
   defined and registration is done through an IETF Standards Action
   [RFC5226].  The registration is required to contain the following
   information:

   o  Name of the policy.

   o  A describing text that explains how the policy works for handling
      the certificates.

   o  A contact person.

   This specification registers the following values:

   Any:  A policy requiring the proxy to accept any received
         certificate.

   Proxy:  A policy where the proxy applies its own policies to
         determine which certificates are accepted or not.

   User: A policy where the certificate is required to be forwarded down
         the proxy chain to the client, thus allowing the user to
         decided to accept or refuse a certificate.

22.5.2.  Accept-Credentials hash algorithms

   The Accept-Credentials header (See Section 18.2) allows for the usage
   of other algorithms for hashing the DER records of accepted entities.
   The registration of any future algorithm is expected to be extremely
   rare and could also cause interoperability problems.  Therefore the
   bar for registering new algorithms is intentionally placed high.



Schulzrinne, et al.      Expires August 14, 2014              [Page 219]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Any registration of a new hash algorithm requires an IETF Standards
   Action [RFC5226].  The registration needs to fulfill the following
   requirement:

   o  The algorithms identifier meeting the "token" ABNF requirement.

   o  Provide a definition of the algorithm.

   The registered value is:

   Hash Alg. Id   Reference
   ------------------------
   sha-256        [RFCXXXX]

22.6.  Cache-Control Cache Directive Extensions

   There exists a number of cache directives which can be sent in the
   Cache-Control header.  A registry for these cache directives is
   established by IANA.  New registrations in this registry requires an
   IETF Standards Action or IESG Approval [RFC5226].  The registration
   needs to contain the following information.

   o  Name of the directive

   o  A definition of the parameter value, if any is allowed.

   o  Specification if it is a request or response directive.

   o  A describing text that explains how the cache directive is used
      for RTSP controlled media streams.

   o  A contact person.

   This specification registers the following values:

      no-cache:

      public:

      private:

      no-transform:

      only-if-cached:

      max-stale:

      min-fresh:



Schulzrinne, et al.      Expires August 14, 2014              [Page 220]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      must-revalidate:

      proxy-revalidate:

      max-age:

   The registry should be represented as: Name of the directive, contact
   person and reference.

22.7.  Media Properties

22.7.1.  Description

   The media streams being controlled by RTSP can have many different
   properties.  The media properties required to cover the use cases
   that were in mind when writing the specification are defined.
   However, it can be expected that further innovation will result in
   new use cases or media streams with properties not covered by the
   ones specified here.  Thus new media properties can be specified.  As
   new media properties may need a substantial amount of new definitions
   to correctly specify behavior for this property the bar is intended
   to be high.

22.7.2.  Registration Rules

   Registering a new media property is done following the Specification
   Required policy [RFC5226].  The Expert reviewer verifies that a
   registration request fulfill the following requirements.

   o  Have an ABNF definition of the media property value name that
      meets "media-prop-ext" definition.

   o  Define which media property group it belongs to or define a new
      group.

   o  Description of all changes to the behavior of the RTSP protocol as
      result of these changes.

   o  A Contact Person for the Registration.

22.7.3.  Registered Values

   This specification registers the ten values listed in Section 18.29.
   The registry should be represented as: Name of the media property,
   property group, contact person and reference.






Schulzrinne, et al.      Expires August 14, 2014              [Page 221]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


22.8.  Notify-Reason header

22.8.1.  Description

   Notify-Reason values are used for indicating the reason the
   notification was sent.  Each reason has its associated rules on what
   headers and information that may or must be included in the
   notification.  New notification behaviors need to be specified to
   enable interoperable usage, thus a specification of each new value is
   required.

22.8.2.  Registration Rules

   Registrations for new Notify-Reason value follows the Specification
   Required policy [RFC5226].  The Expert Reviewer verifies that the
   request fulfills the following requirements:

   o  An ABNF definition of the Notify reason value name that meets
      "Notify-Reason-extension" definition

   o  Description of which headers shall be included in the request and
      response, when it should be sent, and any effect it has on the
      server client state.

   o  A Contact Person for the Registration

22.8.3.  Registered Values

   This specification registers 3 values defined in the Notify-Reas-val
   ABNF, Section 20.2.3:

   end-of-stream:  This Notify-Reason value indicates the end of a media
      stream.

   media-properties-update:  This Notify-Reason value allows the server
      to indicate that the properties of the media has changed during
      the playout.

   scale-change:  This Notify-Reason value allows the server to notify
      the client about a change in the Scale of the media.

   The registry entries should be represented in the registry as: Name,
   short description, contact and reference.








Schulzrinne, et al.      Expires August 14, 2014              [Page 222]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


22.9.  Range Header Formats

22.9.1.  Description

   The Range header (Section 18.40) allows for different range formats.
   These range formats also needs an identifier to be used the Accept-
   Ranges header (Section 18.5).  New range formats may be registered,
   but moderation should be applied as it makes interoperability more
   difficult.

22.9.2.  Registration Rules

   A registration follows the Specification Required policy [RFC5226].
   The Expert Reviewer verifies that the request fulfills the following
   requirements:

   o  An ABNF definition of the range format that fulfills the "range-
      ext" definition.

   o  Define the range format identifier used in Accept-Ranges header
      according to the "extension-format" definition.

   o  Rules for how one handles the range when using a negative Scale.

   o  A Contact person for the registration.

22.9.3.  Registered Values

   The registry should be represented as: Range header format
   identifier, Name of the range format, contact person and reference.
   This specification registers the following values.

   npt:  Normal Play Time

   clock:  UTC Absolute Time format

   smpte:  SMPTE Timestamps

   smpte-30-drop:  SMPTE Timestamps 29.97 Frames/sec (30 Hz with Drop)

   smpte-25:  SMPTE Timestamps 25 Frames/sec

22.10.  Terminate-Reason Header

   The Terminate-Reason header (Section 18.52) has two registries for
   extensions.





Schulzrinne, et al.      Expires August 14, 2014              [Page 223]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


22.10.1.  Redirect Reasons

   This registry contains reasons for session Termination that can be
   included in a Terminate-Reason header (Section 18.52).  Registrations
   are following the policy of Expert Review [RFC5226].  The Expert
   Reviewer verifies that the registration contains the following
   information:

   o  The value follows the Terminate-Reason ABNF, i.e., be a token.

   o  That the specification provide a definition of what procedures are
      to be followed when a client receives this redirect reason.

   o  A Contact Person

   This specification registers three values:

   o  Session-Timeout

   o  Server-Admin

   o  Internal-Error

   The registry should be represented as: Name of the Redirect Reason,
   contact person and reference.

22.10.2.  Terminate-Reason Header Parameters

   This registry contains parameters that may be included in the
   Terminate-Reason header (Section 18.52) in addition to a reason.
   Registrations are done under the policy of Specification Required
   [RFC5226].  The Expert Reviewer verifies that the registration or the
   reference specification contains the following:

   o  A Parameter Name.

   o  A Parameter following the syntax allowed by the RTSP 2.0
      specification.

   o  A Reference to the specification.

   o  A contact person.

   This specification registers:

   o  time

   o  user-msg



Schulzrinne, et al.      Expires August 14, 2014              [Page 224]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   The registry should be represented as: Name of the Terminate Reason,
   contact person and reference.

22.11.  RTP-Info header parameters

22.11.1.  Description

   The RTP-Info header (Section 18.45) carries one or more parameter
   value pairs with information about a particular point in the RTP
   stream.  RTP extensions or new usages may need new types of
   information.  As RTP information that could be needed is likely to be
   generic enough and to maximize the interoperability, new registration
   requires Specification Required.

22.11.2.  Registration Rules

   Registrations for new RTP-Info values follows the policy of
   Specification Required [RFC5226].  The Expert Reviewer verifies that
   the registration and its reference contains the following
   information.

   o  Have an ABNF definition that meets the "generic-param" definition.

   o  A Reference to the specification.

   o  A Contact Person for the Registration.

22.11.3.  Registered Values

   This specification registers the following parameter value pairs:

   o  url

   o  ssrc

   o  seq

   o  rtptime

   The registry should be represented as: Name of the parameter, contact
   person and reference.

22.12.  Seek-Style Policies








Schulzrinne, et al.      Expires August 14, 2014              [Page 225]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


22.12.1.  Description

   Seek-Style Policies defines how the RTSP agent seeks in media content
   when given a position within the media content.  New seek policies
   may be registered, however, a large number of these will complicate
   implementation substantially.  The impact of unknown policies is that
   the server will not honor the unknown and use the server default
   policy instead.

22.12.2.  Registration Rules

   Registrations of new Seek-Style polices follows the policy of
   Specification Required [RFC5226].  The Expert Reviewer verifies that
   the registration fulfill the following requirements:

   o  Have an ABNF definition of the Seek-Style policy name that meets
      "Seek-S-value-ext" definition

   o  Short Description

   o  A Contact Person for the Registration

   o  Description of which headers shall be included in the request and
      response, when it should be sent, and any affect it has on the
      server client state.

22.12.3.  Registered Values

   This specification registers 4 values (Name - Short Description):

   o  RAP - Using the closest Random Access Point prior or at the
      requested start position.

   o  CoRAP - Conditional Random Access Point is like RAP, but only if
      the RAP is closer prior to the requested start position than
      current pause point.

   o  First-Prior - The first-prior policy will start delivery with the
      media unit that has a playout time first prior to the requested
      start position.

   o  Next - The next media units after the provided start position.

   The registry should be represented as: Name of the Seek-Style Policy,
   short description, contact person and reference.






Schulzrinne, et al.      Expires August 14, 2014              [Page 226]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


22.13.  Transport Header Registries

   The transport header (Section 18.54) contains a number of parameters
   which have possibilities for future extensions.  Therefore registries
   for these need to be defined.

22.13.1.  Transport Protocol Identifier

   A Transport Protocol Specification consists of a Transport Protocol
   Identifier, representing some combination of transport protocols, and
   any number of transport header parameters required or optional to use
   with the identified protocol specification.  This registry contains
   the identifiers used by registered Transport Protocol Identifiers.

   A registration for the parameter transport protocol identifier
   follows the policy of Specification Required [RFC5226].  The expert
   reviewer verifies that the registration fulfill the following
   requirements:

   o  A contact person or organization with address and email.

   o  A value definition that are following the ABNF syntax definition
      of "transport-id" Section 20.2.3.

   o  A descriptive text that explains how the registered value are used
      in RTSP, which underlying transport protocols that are used, and
      any required Transport header parameters.

   The registry should be represented as: The protocol ID string,
   contact person and reference.

   This specification registers the following values:

   RTP/AVP:  Use of the RTP [RFC3550] protocol for media transport in
         combination with the "RTP profile for audio and video
         conferences with minimal control" [RFC3551] over UDP.  The
         usage is explained in RFC XXXX, Appendix C.1.

   RTP/AVP/UDP:  the same as RTP/AVP.

   RTP/AVPF:  Use of the RTP [RFC3550] protocol for media transport in
         combination with the "Extended RTP Profile for RTCP-based
         Feedback (RTP/AVPF)" [RFC4585] over UDP.  The usage is
         explained in RFC XXXX, Appendix C.1.

   RTP/AVPF/UDP:  the same as RTP/AVPF.





Schulzrinne, et al.      Expires August 14, 2014              [Page 227]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   RTP/SAVP:  Use of the RTP [RFC3550] protocol for media transport in
         combination with the "The Secure Real-time Transport Protocol
         (SRTP)" [RFC3711] over UDP.  The usage is explained in RFC
         XXXX, Appendix C.1.

   RTP/SAVP/UDP:  the same as RTP/SAVP.

   RTP/SAVPF:  Use of the RTP[RFC3550] protocol for media transport in
         combination with the Extended Secure RTP Profile for Real-time
         Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)
         [RFC5124] over UDP.  The usage is explained in RFC XXXX,
         Appendix C.1.

   RTP/SAVPF/UDP:  the same as RTP/SAVPF.

   RTP/AVP/TCP:  Use of the RTP [RFC3550] protocol for media transport
         in combination with the "RTP profile for audio and video
         conferences with minimal control" [RFC3551] over TCP.  The
         usage is explained in RFC XXXX, Appendix C.2.2.

   RTP/AVPF/TCP:  Use of the RTP [RFC3550] protocol for media transport
         in combination with the "Extended RTP Profile for RTCP-based
         Feedback (RTP/AVPF)" [RFC4585] over TCP.  The usage is
         explained in RFC XXXX, Appendix C.2.2.

   RTP/SAVP/TCP:  Use of the RTP [RFC3550] protocol for media transport
         in combination with the "The Secure Real-time Transport
         Protocol (SRTP)" [RFC3711] over TCP.  The usage is explained in
         RFC XXXX, Appendix C.2.2.

   RTP/SAVPF/TCP:  Use of the RTP [RFC3550] protocol for media transport
         in combination with the "Extended Secure RTP Profile for Real-
         time Transport Control Protocol (RTCP)-Based Feedback (RTP/
         SAVPF)" [RFC5124] over TCP.  The usage is explained in RFC
         XXXX, Appendix C.2.2.

22.13.2.  Transport modes

   The Transport Mode is a Transport header (Section 18.54) parameter,
   it is used to identify the general mode of media transport.  The PLAY
   value registered defines a PLAYBACK mode, where media flows from
   Server to Client.

   A registration for the transport parameter mode follows the policy of
   IETF Standards Action [RFC5226].  The registration needs to meet the
   following requirements:





Schulzrinne, et al.      Expires August 14, 2014              [Page 228]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  A value definition that are following the ABNF "token" definition
      Section 20.2.3.

   o  A describing text that explains how the registered value are used
      in RTSP.

   This specification registers 1 value:

   PLAY: See RFC XXXX.

   The registry should be represented as: The Transport Mode value,
   contact person and reference.

22.13.3.  Transport Parameters

   Transport Parameters are different parameters used in a Transport
   Header's transport specification (Section 18.54) to provide
   additional information required beyond the transport protocol
   identifier to establish a functioning transport.

   A registration for parameters that may be included in the Transport
   header follows the policy of Specification Required [RFC5226].  The
   expert reviewer verifies that the registration fulfill the following
   requirements:

   o  A Transport Parameter Name following the "token" ABNF definition.

   o  A value definition, if the parameter takes a value, that are
      following the ABNF definition "trn-par-value" Section 20.2.3.

   o  A describing text that explains how the registered value are used
      in RTSP.

   This specification registers all the transport parameters defined in
   Section 18.54.  This is a copy of this list:

   o  unicast

   o  multicast

   o  interleaved

   o  ttl

   o  layers

   o  ssrc




Schulzrinne, et al.      Expires August 14, 2014              [Page 229]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  mode

   o  dest_addr

   o  src_addr

   o  setup

   o  connection

   o  RTCP-mux

   o  MIKEY

   The registry should be represented as: The transport parameter name,
   contact person and reference.

22.14.  URI Schemes

   This specification updates two URI schemes, one previously registered
   "rtsp", and one missing in the registry "rtspu", previously only
   defined in the RTSP 1.0 [RFC2326], one new URI scheme "rtsps" is also
   registered.  These URI schemes are registered in an existing registry
   (Uniform Resource Identifier (URI) Schemes) which is not created by
   this memo.  Registrations are following RFC 4395[RFC4395].

22.14.1.  The rtsp URI Scheme

   URI scheme name:  rtsp

   Status:  Permanent

   URI scheme syntax:  See Section 20.2.1 of RFC XXXX.

   URI scheme semantics:  The rtsp scheme is used to indicate resources
         accessible through the usage of the Real-time Streaming
         Protocol (RTSP).  RTSP allows different operations on the
         resource identified by the URI, but the primary purpose is the
         streaming delivery of the resource to a client.  However, the
         operations that are currently defined are: DESCRIBE,
         GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
         SETUP, SET_PARAMETER, and TEARDOWN.

   Encoding considerations:  IRIs in this scheme are defined and needs
         to be encoded as RTSP URIs when used within the RTSP protocol.
         That encoding is done according to RFC 3987.





Schulzrinne, et al.      Expires August 14, 2014              [Page 230]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC
         2326), RTSP 2.0 (RFC XXXX)

   Interoperability considerations:  The extensions in the URI syntax
         performed between RTSP 1.0 and 2.0 can create interoperability
         issues.  The changes are:

            Support for IPV6 literal in host part and future IP literals
            through RFC 3986 defined mechanism.

            A new relative format to use in the RTSP protocol elements
            that is not required to start with "/".

         The above changes should have no impact on interoperability as
         in detail discussed in Section 4.2 of RFCXXXX.

   Security considerations:  All the security threats identified in
         Section 7 of RFC 3986 apply also to this scheme.  They need to
         be reviewed and considered in any implementation utilizing this
         scheme.

   Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com

   Author/Change controller:  IETF

   References:  RFC 2326, RFC 3986, RFC 3987, RFC XXXX

22.14.2.  The rtsps URI Scheme

   URI scheme name:  rtsps

   Status:  Permanent

   URI scheme syntax:  See Section 20.2.1 of RFC XXXX.

   URI scheme semantics:  The rtsps scheme is used to indicate resources
         accessible through the usage of the Real-time Streaming
         Protocol (RTSP) over TLS.  RTSP allows different operations on
         the resource identified by the URI, but the primary purpose is
         the streaming delivery of the resource to a client.  However,
         the operations that are currently defined are: DESCRIBE,
         GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
         SETUP, SET_PARAMETER, and TEARDOWN.

   Encoding considerations:  IRIs in this scheme are defined and needs
         to be encoded as RTSP URIs when used within the RTSP protocol.
         That encoding is done according to RFC 3987.




Schulzrinne, et al.      Expires August 14, 2014              [Page 231]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC
         2326), RTSP 2.0 (RFC XXXX)

   Interoperability considerations:  The "rtsps" scheme was never
         officially defined for RTSP 1.0, however it has seen widespread
         use in actual deployments of RTSP 1.0.  Therefore this section
         discusses the believed changes between the unspecified RTSP 1.0
         "rtsps" scheme and RTSP 2.0 definition.  The extensions in the
         URI syntax performed between RTSP 1.0 and 2.0 can create
         interoperability issues.  The changes are:

            Support for IPV6 literal in host part and future IP literals
            through RFC 3986 defined mechanism.

            A new relative format to use in the RTSP protocol elements
            that is not required to start with "/".

         The above changes should have no impact on interoperability as
         in detail discussed in Section 4.2 of RFCXXXX.

   Security considerations:  All the security threats identified in
         Section 7 of RFC 3986 apply also to this scheme.  They need to
         be reviewed and considered in any implementation utilizing this
         scheme.

   Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com

   Author/Change controller:  IETF

   References:  RFC 2326, RFC 3986, RFC 3987, RFC XXXX

22.14.3.  The rtspu URI Scheme

   URI scheme name:  rtspu

   Status:  Permanent

   URI scheme syntax:  See Section 3.2 of RFC 2326.

   URI scheme semantics:  The rtspu scheme is used to indicate resources
         accessible through the usage of the Real-time Streaming
         Protocol (RTSP) over unreliable datagram transport.  RTSP
         allows different operations on the resource identified by the
         URI, but the primary purpose is the streaming delivery of the
         resource to a client.  However, the operations that are
         currently defined are: DESCRIBE, GET_PARAMETER, OPTIONS,
         REDIRECT,PLAY, PLAY_NOTIFY, PAUSE, SETUP, SET_PARAMETER, and
         TEARDOWN.



Schulzrinne, et al.      Expires August 14, 2014              [Page 232]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Encoding considerations:  This scheme is not intended to be used with
         characters outside the US-ASCII repertoire.

   Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC
         2326)

   Interoperability considerations:  The definition of the transport
         mechanism of RTSP over UDP has interoperability issues.  That
         makes the usage of this scheme problematic.

   Security considerations:  All the security threats identified in
         Section 7 of RFC 3986 apply also to this scheme.  They needs to
         be reviewed and considered in any implementation utilizing this
         scheme.

   Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com

   Author/Change controller:  IETF

   References:  RFC 2326

22.15.  SDP attributes

   This specification defines three SDP [RFC4566] attributes that it is
   requested that IANA register.


























Schulzrinne, et al.      Expires August 14, 2014              [Page 233]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   SDP Attribute ("att-field"):

        Attribute name:     range
        Long form:          Media Range Attribute
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX, RFC 2326
        Values:             See ABNF definition.

        Attribute name:     control
        Long form:          RTSP control URI
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX, RFC 2326
        Values:             Absolute or Relative URIs.

        Attribute name:     mtag
        Long form:          Message Tag
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             See ABNF definition


22.16.  Media Type Registration for text/parameters

   Type name:  text

   Subtype name:  parameters

   Required parameters:

   Optional parameters:  charset: The charset parameter is applicable to
      the encoding of the parameter values.  The default charset is
      UTF-8, if the 'charset' parameter is not present.

   Encoding considerations:  8bit

   Security considerations:  This format may carry any type of
      parameters.  Some can have security requirements, like privacy,
      confidentiality or integrity requirements.  The format has no
      built in security protection.  For the usage it was defined the



Schulzrinne, et al.      Expires August 14, 2014              [Page 234]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      transport can be protected between server and client using TLS.
      However, care must be taken to consider if also the proxies are
      trusted with the parameters in case hop-by-hop security is used.
      If stored as a file in file system, the necessary precautions need
      to be taken in relation to the parameters requirements including
      object security such as S/MIME [RFC5751].

   Interoperability considerations:  This media type was mentioned as a
      fictional example in [RFC2326], but was not formally specified.
      This has resulted in usage of this media type which may not match
      its formal definition.

   Published specification:  RFC XXXX, Appendix F.

   Applications that use this media type:  Applications that use RTSP
      and have additional parameters they like to read and set using the
      RTSP GET_PARAMETER and SET_PARAMETER methods.

   Additional information:

   Magic number(s):

   File extension(s):

   Macintosh file type code(s):

   Person & email address to contact for further information:  Magnus We
      sterlund (magnus.westerlund@ericsson.com)

   Intended usage:   Common

   Restrictions on usage:   None

   Author:  Magnus Westerlund (magnus.westerlund@ericsson.com)

   Change controller:  IETF

   Addition Notes:

23.  References

23.1.  Normative References

   [FIPS-pub-180-2]
              National Institute of Standards and Technology (NIST),
              "Federal Information Processing Standards Publications
              (FIPS PUBS) 180-2: Secure Hash Standard", August 2002.




Schulzrinne, et al.      Expires August 14, 2014              [Page 235]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", draft-ietf-
              avtcore-rtp-circuit-breakers-04 (work in progress),
              January 2014.

   [I-D.ietf-mmusic-rtsp-nat]
              Goldberg, J., Westerlund, M., and T. Zeng, "A Network
              Address Translator (NAT) Traversal Mechanism for Media
              Controlled by Real-Time Streaming Protocol (RTSP)", draft-
              ietf-mmusic-rtsp-nat-20 (work in progress), February 2014.

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              August 1980.

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7, RFC
              793, September 1981.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", RFC 2460, December 1998.

   [RFC2616]  Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
              Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
              Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.

   [RFC2617]  Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
              Leach, P., Luotonen, A., and L. Stewart, "HTTP
              Authentication: Basic and Digest Access Authentication",
              RFC 2617, June 1999.

   [RFC2818]  Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3629]  Yergeau, F., "UTF-8, a transformation format of ISO
              10646", STD 63, RFC 3629, November 2003.






Schulzrinne, et al.      Expires August 14, 2014              [Page 236]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              August 2004.

   [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
              Resource Identifier (URI): Generic Syntax", STD 66, RFC
              3986, January 2005.

   [RFC3987]  Duerst, M. and M. Suignard, "Internationalized Resource
              Identifiers (IRIs)", RFC 3987, January 2005.

   [RFC4086]  Eastlake, D., Schiller, J., and S. Crocker, "Randomness
              Requirements for Security", BCP 106, RFC 4086, June 2005.

   [RFC4291]  Hinden, R. and S. Deering, "IP Version 6 Addressing
              Architecture", RFC 4291, February 2006.

   [RFC4395]  Hansen, T., Hardie, T., and L. Masinter, "Guidelines and
              Registration Procedures for New URI Schemes", BCP 35, RFC
              4395, February 2006.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
              and RTP Control Protocol (RTCP) Packets over Connection-
              Oriented Transport", RFC 4571, July 2006.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
              2006.

   [RFC4648]  Josefsson, S., "The Base16, Base32, and Base64 Data
              Encodings", RFC 4648, October 2006.

   [RFC4738]  Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-
              RSA-R: An Additional Mode of Key Distribution in
              Multimedia Internet KEYing (MIKEY)", RFC 4738, November
              2006.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.



Schulzrinne, et al.      Expires August 14, 2014              [Page 237]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   [RFC5226]  Narten, T. and H. Alvestrand, "Guidelines for Writing an
              IANA Considerations Section in RFCs", BCP 26, RFC 5226,
              May 2008.

   [RFC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234, January 2008.

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246, August 2008.

   [RFC5280]  Cooper, D., Santesson, S., Farrell, S., Boeyen, S.,
              Housley, R., and W. Polk, "Internet X.509 Public Key
              Infrastructure Certificate and Certificate Revocation List
              (CRL) Profile", RFC 5280, May 2008.

   [RFC5322]  Resnick, P., Ed., "Internet Message Format", RFC 5322,
              October 2008.

   [RFC5646]  Phillips, A. and M. Davis, "Tags for Identifying
              Languages", BCP 47, RFC 5646, September 2009.

   [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
              Mail Extensions (S/MIME) Version 3.2 Message
              Specification", RFC 5751, January 2010.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888, June 2010.

   [RFC6838]  Freed, N., Klensin, J., and T. Hansen, "Media Type
              Specifications and Registration Procedures", BCP 13, RFC
              6838, January 2013.

   [SMPTE_TC]
              Society of Motion Picture and Television Engineers, "SMPTE
              Standard for Television -- Time and Control Code, ST
              12M-1-2008", .

   [TS-26234]
              Third Generation Partnership Project (3GPP), "Transparent
              end-to-end Packet-switched Streaming Service (PSS);
              Protocols and codecs; Technical Specification 26.234",
              December 2002.






Schulzrinne, et al.      Expires August 14, 2014              [Page 238]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


23.2.  Informative References

   [ISO.13818-6.1995]
              International Organization for Standardization,
              "Information technology - Generic coding of moving
              pictures and associated audio information - part 6:
              Extension for digital storage media and control", ISO
              Draft Standard 13818-6, November 1995.

   [ISO.8601.2000]
              International Organization for Standardization, "Data
              elements and interchange formats - Information interchange
              - Representation of dates and times", ISO/IEC Standard
              8601, December 2000.

   [RFC0791]  Postel, J., "Internet Protocol", STD 5, RFC 791, September
              1981.

   [RFC1123]  Braden, R., "Requirements for Internet Hosts - Application
              and Support", STD 3, RFC 1123, October 1989.

   [RFC2068]  Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
              Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1",
              RFC 2068, January 1997.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC2663]  Srisuresh, P. and M. Holdrege, "IP Network Address
              Translator (NAT) Terminology and Considerations", RFC
              2663, August 1999.

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, October 2000.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264, June
              2002.

   [RFC3339]  Klyne, G., Ed. and C. Newman, "Date and Time on the
              Internet: Timestamps", RFC 3339, July 2002.





Schulzrinne, et al.      Expires August 14, 2014              [Page 239]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in
              the Session Description Protocol (SDP)", RFC 4145,
              September 2005.

   [RFC4567]  Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
              Carrara, "Key Management Extensions for Session
              Description Protocol (SDP) and Real Time Streaming
              Protocol (RTSP)", RFC 4567, July 2006.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

   [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
              Formats", RFC 4855, February 2007.

   [RFC4856]  Casner, S., "Media Type Registration of Payload Formats in
              the RTP Profile for Audio and Video Conferences", RFC
              4856, February 2007.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, February 2008.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              2010.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [RFC5583]  Schierl, T. and S. Wenger, "Signaling Media Decoding
              Dependency in the Session Description Protocol (SDP)", RFC
              5583, July 2009.

   [RFC5905]  Mills, D., Martin, J., Burbank, J., and W. Kasch, "Network
              Time Protocol Version 4: Protocol and Algorithms
              Specification", RFC 5905, June 2010.

   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298, June
              2011.

   [Stevens98]
              Stevens, W., "Unix Networking Programming - Volume 1,
              second edition", 1998.



Schulzrinne, et al.      Expires August 14, 2014              [Page 240]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


Appendix A.  Examples

   This section contains several different examples trying to illustrate
   possible ways of using RTSP.  The examples can also help with the
   understanding of how functions of RTSP work.  However, remember that
   these are examples and the normative and syntax description in the
   other sections take precedence.  Please also note that many of the
   examples contain syntax illegal line breaks to accommodate the
   formatting restriction that the RFC series impose.

A.1.  Media on Demand (Unicast)

   This is an example of media on demand streaming of a media stored in
   a container file.  For purposes of this example, a container file is
   a storage entity in which multiple continuous media types pertaining
   to the same end-user presentation are present.  In effect, the
   container file represents an RTSP presentation, with each of its
   components being RTSP controlled media streams.  Container files are
   a widely used means to store such presentations.  While the
   components are transported as independent streams, it is desirable to
   maintain a common context for those streams at the server end.

      This enables the server to keep a single storage handle open
      easily.  It also allows treating all the streams equally in case
      of any prioritization of streams by the server.

   It is also possible that the presentation author may wish to prevent
   selective retrieval of the streams by the client in order to preserve
   the artistic effect of the combined media presentation.  Similarly,
   in such a tightly bound presentation, it is desirable to be able to
   control all the streams via a single control message using an
   aggregate URI.

   The following is an example of using a single RTSP session to control
   multiple streams.  It also illustrates the use of aggregate URIs.  In
   a container file it is also desirable to not write any URI parts
   which are not kept, when the container is distributed, like the host
   and most of the path element.  Therefore this example also uses the
   "*" and relative URI in the delivered SDP.

   Also this presentation description (SDP) is not cacheble, as the
   Expires header is set to an equal value with date indicating
   immediate expiration of its validity.

   Client C requests a presentation from media server M. The movie is
   stored in a container file.  The client has obtained an RTSP URI to
   the container file.




Schulzrinne, et al.      Expires August 14, 2014              [Page 241]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 1
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:20:32 +0000
         Content-Type: application/sdp
         Content-Length: 271
         Content-Base: rtsp://example.com/twister.3gp/
         Expires: Fri, 20 Dec 2013 12:20:32 +0000

         v=0
         o=- 2890844256 2890842807 IN IP4 198.51.100.5
         s=RTSP Session
         i=An Example of RTSP Session Usage
         e=adm@example.com
         c=IN IP4 0.0.0.0
         a=control: *
         a=range:npt=00:00:00-00:10:34.10
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control: trackID=1
         m=video 0 RTP/AVP 26
         a=control: trackID=4

























Schulzrinne, et al.      Expires August 14, 2014              [Page 242]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
         Accept-Ranges: npt, smpte, clock

   M->C: RTSP/2.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast; ssrc=93CB001E;
                    dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
                    src_addr="198.51.100.5:9000"/"198.51.100.5:9001"
         Session: 12345678
         Expires: Fri, 20 Dec 2013 12:20:33 +0000
         Date: Fri, 20 Dec 2013 10:20:33 +0000
         Accept-Ranges: npt
         Media-Properties: Random-Access=0.02, Immutable, Unlimited

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
         Session: 12345678
         Accept-Ranges: npt, smpte, clock


   M->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast; ssrc=A813FC13;
                    dest_addr="192.0.2.53:8002"/"192.0.2.53:8003";
                    src_addr="198.51.100.5:9002"/"198.51.100.5:9003";

         Session: 12345678
         Expires: Fri, 20 Dec 2013 12:20:33 +0000
         Date: Fri, 20 Dec 2013 10:20:33 +0000
         Accept-Range: NPT
         Media-Properties: Random-Access=0.8, Immutable, Unlimited

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 4
         User-Agent: PhonyClient/1.2
         Range: npt=30-
         Seek-Style: RAP
         Session: 12345678




Schulzrinne, et al.      Expires August 14, 2014              [Page 243]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   M->C: RTSP/2.0 200 OK
         CSeq: 4
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:20:34 +0000
         Session: 12345678
         Range: npt=30-634.10
         Seek-Style: RAP
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12345;rtptime=3450012,
           url="rtsp://example.com/twister.3gp/trackID=1"
            ssrc=4F312DD8:seq=54321;rtptime=2876889

   C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 5
         User-Agent: PhonyClient/1.2
         Session: 12345678

   # Pause happens 0.87 seconds after starting to play

   M->C: RTSP/2.0 200 OK
         CSeq: 5
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:20:35 +0000
         Session: 12345678
         Range: npt=30.87-634.10

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 6
         User-Agent: PhonyClient/1.2
         Range: npt=30.87-634.10
         Seek-Style: Next
         Session: 12345678

   M->C: RTSP/2.0 200 OK
         CSeq: 6
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:22:13 +0000
         Session: 12345678
         Range: npt=30.87-634.10
         Seek-Style: Next
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12555;rtptime=6330012,
           url="rtsp://example.com/twister.3gp/trackID=1"
            ssrc=4F312DD8:seq=55021;rtptime=3132889


   C->M: TEARDOWN rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 7



Schulzrinne, et al.      Expires August 14, 2014              [Page 244]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         User-Agent: PhonyClient/1.2
         Session: 12345678

   M->C: RTSP/2.0 200 OK
         CSeq: 7
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:31:53 +0000

A.2.  Media on Demand using Pipelining

   This example is basically the example above (Appendix A.1), but now
   utilizing pipelining to speed up the setup.  It requires only two
   round trip times until the media starts flowing.  First of all, the
   session description is retrieved to determine what media resources
   need to be setup.  In the second step, one sends the necessary SETUP
   requests and the PLAY request to initiate media delivery.

   Client C requests a presentation from media server M. The movie is
   stored in a container file.  The client has obtained an RTSP URI to
   the container file.

   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 1
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:20:32 +0000
         Content-Type: application/sdp
         Content-Length: 271
         Content-Base: rtsp://example.com/twister.3gp/
         Expires: Fri, 20 Dec 2013 12:20:32 +0000

         v=0
         o=- 2890844256 2890842807 IN IP4 192.0.2.5
         s=RTSP Session
         i=An Example of RTSP Session Usage
         e=adm@example.com
         c=IN IP4 0.0.0.0
         a=control: *
         a=range:npt=00:00:00-00:10:34.10
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control: trackID=1
         m=video 0 RTP/AVP 26
         a=control: trackID=4




Schulzrinne, et al.      Expires August 14, 2014              [Page 245]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
         Accept-Ranges: npt, smpte, clock
         Pipelined-Requests: 7654

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
         Accept-Ranges: npt, smpte, clock
         Pipelined-Requests: 7654

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 4
         User-Agent: PhonyClient/1.2
         Range: npt=0-
         Seek-Style: RAP
         Pipelined-Requests: 7654

   M->C: RTSP/2.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;
                    dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
                    src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
                    ssrc=93CB001E
         Session: 12345678
         Expires: Fri, 20 Dec 2013 12:20:32 +0000
         Date: Fri, 20 Dec 2013 10:20:32 +0000
         Accept-Ranges: npt
         Pipelined-Requests: 7654
         Media-Properties: Random-Access=0.2, Immutable, Unlimited

   M->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;
                    dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
                    src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
                    ssrc=A813FC13
         Session: 12345678
         Expires: Sat, 21 Dec 2013 10:20:32 +0000
         Date: Fri, 20 Dec 2013 10:20:32 +0000
         Accept-Range: NPT



Schulzrinne, et al.      Expires August 14, 2014              [Page 246]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         Pipelined-Requests: 7654
         Media-Properties: Random-Access=0.8, Immutable, Unlimited

   M->C: RTSP/2.0 200 OK
         CSeq: 4
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:20:32 +0000
         Session: 12345678
         Range: npt=0-623.10
         Seek-Style: RAP
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12345;rtptime=3450012,
           url="rtsp://example.com/twister.3gp/trackID=1"
            ssrc=4F312DD8:seq=54321;rtptime=2876889
         Pipelined-Requests: 7654

A.3.  Secured Media Session for on Demand Content

   This example is basically the above example (Appendix A.2), but now
   including establishment of SRTP crypto contexts to get a secured
   media delivery.  First of all, the client attempts to fetch this
   insecurely, but the server redirects to a URI indicating a
   requirement on using a secure connection for the RTSP messages.  The
   client establishes a TCP/TLS connections and the session description
   is retrieved to determine what media resources need to be setup.  In
   the this session description secure media (SRTP) is indicated.  In
   the next step, the client sends the necessary SETUP requests
   including MIKEY messages.  This is pipeline with a PLAY request to
   initiate media delivery.

   Client C requests a presentation from media server M. The movie is
   stored in a container file.  The client has obtained an RTSP URI to
   the container file.

   Note: The MIKEY messages below are not valid MIKEY message and are
   BASE64 encoded random data to represent where the MIKEY messages
   would go.

   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 301 Moved Permanently
         CSeq: 1
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:25:32 +0000
         Location: rtsps://example.com/twister.3gp




Schulzrinne, et al.      Expires August 14, 2014              [Page 247]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->M: Establish TCP/TLS connection and verify server's
         certificate that is represents example.com.
         Used for all below RTSP messages.

   C->M: DESCRIBE rtsps://example.com/twister.3gp RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:25:33 +0000
         Content-Type: application/sdp
         Content-Length: 271
         Content-Base: rtsps://example.com/twister.3gp/
         Expires: Fri, 20 Dec 2013 12:25:33 +0000

         v=0
         o=- 2890844256 2890842807 IN IP4 192.0.2.5
         s=RTSP Session
         i=An Example of RTSP Session Usage
         e=adm@example.com
         c=IN IP4 0.0.0.0
         a=control: *
         a=range:npt=00:00:00-00:10:34.10
         t=0 0
         m=audio 0 RTP/SAVP 0
         a=control: trackID=1
         m=video 0 RTP/SAVP 26
         a=control: trackID=4

   C->M: SETUP rtsps://example.com/twister.3gp/trackID=1 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/SAVP;unicast;dest_addr=":8000"/":8001";
            MIKEY=VGhpcyBpcyB0aGUgZmlyc3Qgc3RyZWFtcyBNSUtFWSBtZXNzYWdl
         Accept-Ranges: npt, smpte, clock
         Pipelined-Requests: 7654

   C->M: SETUP rtsps://example.com/twister.3gp/trackID=4 RTSP/2.0
         CSeq: 4
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/SAVP;unicast;dest_addr=":8002"/":8003";
            MIKEY=TUlLRVkgZm9yIHN0cmVhbSB0d2lzdGVyLjNncC90cmFja0lEPTQ=
         Accept-Ranges: npt, smpte, clock
         Pipelined-Requests: 7654



Schulzrinne, et al.      Expires August 14, 2014              [Page 248]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->M: PLAY rtsps://example.com/twister.3gp/ RTSP/2.0
         CSeq: 5
         User-Agent: PhonyClient/1.2
         Range: npt=0-
         Seek-Style: RAP
         Pipelined-Requests: 7654

   M->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Transport: RTP/SAVP;unicast;
            dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
            src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
            ssrc=93CB001E;
            MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD0x
         Session: 12345678
         Expires: Fri, 20 Dec 2013 12:25:34 +0000
         Date: Fri, 20 Dec 2013 10:25:34 +0000
         Accept-Ranges: npt
         Pipelined-Requests: 7654
         Media-Properties: Random-Access=0.2, Immutable, Unlimited

   M->C: RTSP/2.0 200 OK
         CSeq: 4
         Server: PhonyServer/1.0
         Transport: RTP/SAVP;unicast;
            dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
            src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
            ssrc=A813FC13;
            MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD00
         Session: 12345678
         Expires: Fri, 20 Dec 2013 12:25:34 +0000
         Date: Fri, 20 Dec 2013 10:25:34 +0000
         Accept-Range: NPT
         Pipelined-Requests: 7654
         Media-Properties: Random-Access=0.8, Immutable, Unlimited

   M->C: RTSP/2.0 200 OK
         CSeq: 5
         Server: PhonyServer/1.0
         Date: Fri, 20 Dec 2013 10:25:34 +0000
         Session: 12345678
         Range: npt=0-623.10
         Seek-Style: RAP
         RTP-Info: url="rtsps://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12345;rtptime=3450012,
           url="rtsps://example.com/twister.3gp/trackID=1"
            ssrc=4F312DD8:seq=54321;rtptime=2876889;



Schulzrinne, et al.      Expires August 14, 2014              [Page 249]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         Pipelined-Requests: 7654

A.4.  Media on Demand (Unicast)

   An alternative example of media on demand with a bit more tweaks is
   the following.  Client C requests a movie distributed from two
   different media servers A (audio.example.com) and V (
   video.example.com).  The media description is stored on a web server
   W.  The media description contains descriptions of the presentation
   and all its streams, including the codecs that are available, dynamic
   RTP payload types, the protocol stack, and content information such
   as language or copyright restrictions.  It may also give an
   indication about the timeline of the movie.

   In this example, the client is only interested in the last part of
   the movie.



































Schulzrinne, et al.      Expires August 14, 2014              [Page 250]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->W: GET /twister.sdp HTTP/1.1
         Host: www.example.com
         Accept: application/sdp

   W->C: HTTP/1.1 200 OK
         Date: Wed, 23 Jan 2013 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 278
         Expires: Thu, 24 Jan 2013 15:35:06 GMT

         v=0
         o=- 2890844526 2890842807 IN IP4 198.51.100.5
         s=RTSP Session
         e=adm@example.com
         c=IN IP4 0.0.0.0
         a=range:npt=00:00:00-01:49:34
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control:rtsp://audio.example.com/twister/audio.en
         m=video 0 RTP/AVP 31
         a=control:rtsp://video.example.com/twister/video

   C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
                    RTP/AVP/TCP;unicast;interleaved=0-1
         Accept-Ranges: npt, smpte, clock

   A->C: RTSP/2.0 200 OK
         CSeq: 1
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;
                    dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
                    src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
         Date: Wed, 23 Jan 2013 15:35:12 +0000
         Server: PhonyServer/1.0
         Expires: Thu, 24 Jan 2013 15:35:12 +0000
         Cache-Control: public
         Accept-Ranges: npt, smpte
         Media-Properties: Random-Access=0.02, Immutable, Unlimited










Schulzrinne, et al.      Expires August 14, 2014              [Page 251]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->V: SETUP rtsp://video.example.com/twister/video RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;
                    dest_addr="192.0.2.53:3058"/"192.0.2.53:3059",
                    RTP/AVP/TCP;unicast;interleaved=0-1
         Accept-Ranges: npt, smpte, clock

   V->C: RTSP/2.0 200 OK
         CSeq: 1
         Session: 23456789
         Transport: RTP/AVP/UDP;unicast;
            dest_addr="192.0.2.53:3058"/"192.0.2.53:3059";
            src_addr="198.51.100.5:5002"/"198.51.100.5:5003"
         Date: Wed, 23 Jan 2013 15:35:12 +0000
         Server: PhonyServer/1.0
         Cache-Control: public
         Expires: Thu, 24 Jan 2013 15:35:12 +0000
         Accept-Ranges: npt, smpte
         Media-Properties: Random-Access=1.2, Immutable, Unlimited

   C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 23456789
         Range: smpte=0:10:00-

   V->C: RTSP/2.0 200 OK
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-1:49:23
         Seek-Style: First-Prior
         RTP-Info: url="rtsp://video.example.com/twister/video"
                   ssrc=A17E189D:seq=12312232;rtptime=78712811
         Server: PhonyServer/2.0
         Date: Wed, 23 Jan 2013 15:35:13 +0000















Schulzrinne, et al.      Expires August 14, 2014              [Page 252]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 12345678
         Range: smpte=0:10:00-

   A->C: RTSP/2.0 200 OK
         CSeq: 2
         Session: 12345678
         Range: smpte=0:10:00-1:49:23
         Seek-Style: First-Prior
         RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
                   ssrc=3D124F01:seq=876655;rtptime=1032181
         Server: PhonyServer/1.0
         Date: Wed, 23 Jan 2013 15:35:13 +0000



   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 12345678

   A->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: Wed, 23 Jan 2013 15:36:52 +0000

   C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 23456789

   V->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/2.0
         Date: Wed, 23 Jan 2013 15:36:52 +0000


   Even though the audio and video track are on two different servers
   that may start at slightly different times and may drift with respect
   to each other over time, the client can perform initial
   synchronization of the two media using RTP-Info and Range received in
   the PLAY responses.  If the two servers are time synchronized the
   RTCP packets can also be used to maintain synchronization.






Schulzrinne, et al.      Expires August 14, 2014              [Page 253]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


A.5.  Single Stream Container Files

   Some RTSP servers may treat all files as though they are "container
   files", yet other servers may not support such a concept.  Because of
   this, clients needs to use the rules set forth in the session
   description for Request-URIs, rather than assuming that a consistent
   URI may always be used throughout.  Below is an example of how a
   multi-stream server might expect a single-stream file to be served:

   C->S: DESCRIBE rtsp://foo.example.com/test.wav RTSP/2.0
         Accept: application/x-rtsp-mh, application/sdp
         CSeq: 1
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 1
         Content-base: rtsp://foo.example.com/test.wav/
         Content-type: application/sdp
         Content-length: 163
         Server: PhonyServer/1.0
         Date: Wed, 23 Jan 2013 15:36:52 +0000
         Expires: Thu, 24 Jan 2013 15:36:52 +0000

         v=0
         o=- 872653257 872653257 IN IP4 192.0.2.5
         s=mu-law wave file
         i=audio test
         c=IN IP4 0.0.0.0
         t=0 0
         a=control: *
         m=audio 0 RTP/AVP 0
         a=control:streamid=0



















Schulzrinne, et al.      Expires August 14, 2014              [Page 254]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->S: SETUP rtsp://foo.example.com/test.wav/streamid=0 RTSP/2.0
         Transport: RTP/AVP/UDP;unicast;
            dest_addr=":6970"/":6971";mode="PLAY"
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Accept-Ranges: npt, smpte, clock

   S->C: RTSP/2.0 200 OK
         Transport: RTP/AVP/UDP;unicast;
             dest_addr="192.0.2.53:6970"/"192.0.2.53:6971";
             src_addr="198.51.100.5:6970"/"198.51.100.5:6971";
             mode="PLAY";ssrc=EAB98712
         CSeq: 2
         Session: 2034820394
         Expires: Thu, 24 Jan 2013 15:36:52 +0000
         Server: PhonyServer/1.0
         Date: Wed, 23 Jan 2013 15:36:52 +0000
         Accept-Ranges: npt
         Media-Properties: Random-Acces=0.5, Immutable, Unlimited


   C->S: PLAY rtsp://foo.example.com/test.wav/ RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 2034820394

   S->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: Wed, 23 Jan 2013 15:36:52 +0000
         Session: 2034820394
         Range: npt=0-600
         Seek-Style: RAP
         RTP-Info: url="rtsp://foo.example.com/test.wav/streamid=0"
            ssrc=0D12F123:seq=981888;rtptime=3781123

   Note the different URI in the SETUP command, and then the switch back
   to the aggregate URI in the PLAY command.  This makes complete sense
   when there are multiple streams with aggregate control, but is less
   than intuitive in the special case where the number of streams is
   one.  However, the server has declared the aggregated control URI in
   the SDP and therefore this is legal.

   In this case, it is also required that servers accept implementations
   that use the non-aggregated interpretation and use the individual
   media URI, like this:





Schulzrinne, et al.      Expires August 14, 2014              [Page 255]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 2034820394

A.6.  Live Media Presentation Using Multicast

   The media server M chooses the multicast address and port.  Here, it
   is assumed that the web server only contains a pointer to the full
   description, while the media server M maintains the full description.

   C->W: GET /sessions.html HTTP/1.1
         Host: www.example.com

   W->C: HTTP/1.1 200 OK
         Content-Type: text/html

         <html>
           ...
           <a href "rtsp://live.example.com/concert/audio">
              Streamed Live Music performance </a>
           ...
         </html>


   C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/2.0
         CSeq: 1
         Supported: play.basic, play.scale
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 183
         Server: PhonyServer/1.0
         Date: Wed, 23 Jan 2013 15:36:52 +0000
         Supported: play.basic

         v=0
         o=- 2890844526 2890842807 IN IP4 192.0.2.5
         s=RTSP Session
         t=0 0
         m=audio 3456 RTP/AVP 0
         c=IN IP4 233.252.0.54/16
         a=control: rtsp://live.example.com/concert/audio
         a=range:npt=0-





Schulzrinne, et al.      Expires August 14, 2014              [Page 256]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0
         CSeq: 2
         Transport: RTP/AVP;multicast;
              dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
         Accept-Ranges: npt, smpte, clock
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Date: Wed, 23 Jan 2013 15:36:52 +0000
         Transport: RTP/AVP;multicast;
              dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
              ;ssrc=4D12AB92/0DF876A3
         Session: 0456804596
         Accept-Ranges: npt, clock
         Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0


   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0
         CSeq: 3
         Session: 0456804596
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: Wed, 23 Jan 2013 15:36:52 +0000
         Session: 0456804596
         Seek-Style: Next
         Range:npt=1256-
         RTP-Info: url="rtsp://live.example.com/concert/audio"
                   ssrc=0D12F123:seq=1473; rtptime=80000

A.7.  Capability Negotiation

   This example illustrates how the client and server determine their
   capability to support a special feature, in this case "play.scale".
   The server, through the clients request and the included Supported
   header, learns the client supports RTSP 2.0, and also supports the
   playback time scaling feature of RTSP.  The server's response
   contains the following feature related information to the client; it
   supports the basic media delivery functions (play.basic), the
   extended functionality of time scaling of content (play.scale), and
   one "example.com" proprietary feature (com.example.flight).  The
   client also learns the methods supported (Public header) by the
   server for the indicated resource.




Schulzrinne, et al.      Expires August 14, 2014              [Page 257]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0
         CSeq: 1
         Supported: play.basic, play.scale
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 1
         Public:OPTIONS,SETUP,PLAY,PAUSE,TEARDOWN,DESCRIBE,GET_PARAMETER
         Allow: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN, DESCRIBE
         Server: PhonyServer/2.0
         Supported: play.basic, play.scale, com.example.flight

   When the client sends its SETUP request it tells the server that it
   requires support of the play.scale feature for this session by
   including the Require header.

   C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;
                    dest_addr="192.0.2.53:3056"/"192.0.2.53:3057",
                    RTP/AVP/TCP;unicast;interleaved=0-1
         Require: play.scale
         Accept-Ranges: npt, smpte, clock
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 3
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;
            dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
            src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
         Server: PhonyServer/2.0
         Accept-Ranges: npt, smpte
         Media-Properties: Random-Access=0.8, Immutable, Unlimited

Appendix B.  RTSP Protocol State Machine

   The RTSP session state machine describes the behavior of the protocol
   from RTSP session initialization through RTSP session termination.
   It is probably easiest to think of this as the server's state and
   then view the the client as needing to track what it believes the
   server's state will be based on sent or received RTSP messages.  Thus
   in most cases the state tables below can be read as: If the client
   does X, and assuming it fulfills any pre-requisite(s), the (server)
   state will move to the new state and the indicated response will
   returned.  However, there are also server to client notifications or
   requests, where the action describes what notification or request



Schulzrinne, et al.      Expires August 14, 2014              [Page 258]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   will occur, its requisites and what new state will result after the
   server has received the response, as well as describing the client's
   response to the action.

   The State machine is defined on a per session basis which is uniquely
   identified by the RTSP session identifier.  The session may contain
   one or more media streams depending on state.  If a single media
   stream is part of the session it is in non-aggregated control.  If
   two or more is part of the session it is in aggregated control.

   The below state machine is an informative description of the
   protocols behavior.  In case of ambiguity with the earlier parts of
   this specification, the description in the earlier parts take
   precedence.

B.1.  States

   The state machine contains three states, described below.  For each
   state there exists a table which shows which requests and events are
   allowed and whether they will result in a state change.

   Init: Initial state no session exists.

   Ready:  Session is ready to start playing.

   Play: Session is playing, i.e., sending media stream data in the
         direction S->C.

B.2.  State variables

   This representation of the state machine needs more than its state to
   work.  A small number of variables are also needed and they are
   explained below.

   NRM:  The number of media streams part of this session.

   RP:   Resume point, the point in the presentation time line at which
         a request to continue playing will resume from.  A time format
         for the variable is not mandated.

B.3.  Abbreviations

   To make the state tables more compact a number of abbreviations are
   used, which are explained below.

   IFI:  IF Implemented.

   md:   Media



Schulzrinne, et al.      Expires August 14, 2014              [Page 259]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   PP:   Pause Point, the point in the presentation time line at which
         the presentation was paused.

   Prs:  Presentation, the complete multimedia presentation.

   RedP: Redirect Point, the point in the presentation time line at
         which a REDIRECT was specified to occur.

   SES:  Session.

B.4.  State Tables

   This section contains a table for each state.  The table contains all
   the requests and events that this state is allowed to act on.  The
   events which are method names are, unless noted, requests with the
   given method in the direction client to server (C->S).  In some cases
   there exist one or more requisite.  The response column tells what
   type of response actions should be performed.  Possible actions that
   are requested for an event include: response codes, e.g., 200,
   headers that need to be included in the response, setting of state
   variables, or setting of other session related parameters.  The new
   state column tells which state the state machine changes to.

   The response to a valid request meeting the requisites is normally a
   2xx (SUCCESS) unless otherwise noted in the response column.  The
   exceptions need to be given a response according to the response
   column.  If the request does not meet the requisite, is erroneous or
   some other type of error occurs, the appropriate response code is to
   be sent.  If the response code is a 4xx the session state is
   unchanged.  A response code of 3rr will result in that the session is
   ended and its state is changed to Init.  A response code of 304
   results in no state change.  However, there are restrictions to when
   a 3rr response may be used.  A 5xx response does not result in any
   change of the session state, except if the error is not possible to
   recover from.  A unrecoverable error results in the ending of the
   session.  As it in the general case can't be determined if it was a
   unrecoverable error or not the client will be required to test.  In
   the case that the next request after a 5xx is responded with 454
   (Session Not Found) the client knows that the session has ended.  For
   any request message that cannot be responded to within the time
   defined in Section 10.4, a 100 response must be sent.

   The server will timeout the session after the period of time
   specified in the SETUP response, if no activity from the client is
   detected.  Therefore there exists a timeout event for all states
   except Init.





Schulzrinne, et al.      Expires August 14, 2014              [Page 260]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   In the case that NRM = 1 the presentation URI is equal to the media
   URI or a specified presentation URI.  For NRM > 1 the presentation
   URI needs to be other than any of the medias that are part of the
   session.  This applies to all states.

   +---------------+-----------------+---------------------------------+
   | Event         | Prerequisite    | Response                        |
   +---------------+-----------------+---------------------------------+
   | DESCRIBE      | Needs REDIRECT  | 3rr, Redirect                   |
   |               |                 |                                 |
   | DESCRIBE      |                 | 200, Session description        |
   |               |                 |                                 |
   | OPTIONS       | Session ID      | 200, Reset session timeout      |
   |               |                 | timer                           |
   |               |                 |                                 |
   | OPTIONS       |                 | 200                             |
   |               |                 |                                 |
   | SET_PARAMETER | Valid parameter | 200, change value of parameter  |
   |               |                 |                                 |
   | GET_PARAMETER | Valid parameter | 200, return value of parameter  |
   +---------------+-----------------+---------------------------------+

               Table 13: None state-machine changing events

   The methods in Table 13 do not have any effect on the state machine
   or the state variables.  However, some methods do change other
   session related parameters, for example SET_PARAMETER which will set
   the parameter(s) specified in its body.  Also all of these methods
   that allow Session header will also update the keep-alive timer for
   the session.

   +------------------+----------------+-----------+-------------------+
   | Action           | Requisite      | New State | Response          |
   +------------------+----------------+-----------+-------------------+
   | SETUP            |                | Ready     | NRM=1, RP=0.0     |
   |                  |                |           |                   |
   | SETUP            | Needs Redirect | Init      | 3rr Redirect      |
   |                  |                |           |                   |
   | S -> C: REDIRECT | No Session hdr | Init      | Terminate all SES |
   +------------------+----------------+-----------+-------------------+

                           Table 14: State: Init

   The initial state of the state machine, see Table 14 can only be left
   by processing a correct SETUP request.  As seen in the table the two
   state variables are also set by a correct request.  This table also
   shows that a correct SETUP can in some cases be redirected to another
   URI and/or server by a 3rr response.



Schulzrinne, et al.      Expires August 14, 2014              [Page 261]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   +-------------+------------------------+---------+------------------+
   | Action      | Requisite              | New     | Response         |
   |             |                        | State   |                  |
   +-------------+------------------------+---------+------------------+
   | SETUP       | New URI                | Ready   | NRM +=1          |
   |             |                        |         |                  |
   | SETUP       | URI Setup prior        | Ready   | Change transport |
   |             |                        |         | param            |
   |             |                        |         |                  |
   | TEARDOWN    | Prs URI,               | Init    | No session hdr,  |
   |             |                        |         | NRM = 0          |
   |             |                        |         |                  |
   | TEARDOWN    | md URI,NRM=1           | Init    | No Session hdr,  |
   |             |                        |         | NRM = 0          |
   |             |                        |         |                  |
   | TEARDOWN    | md URI,NRM>1           | Ready   | Session hdr, NRM |
   |             |                        |         | -= 1             |
   |             |                        |         |                  |
   | PLAY        | Prs URI, No range      | Play    | Play from RP     |
   |             |                        |         |                  |
   | PLAY        | Prs URI, Range         | Play    | According to     |
   |             |                        |         | range            |
   |             |                        |         |                  |
   | PLAY        | md URI, NRM=1, Range   | Play    | According to     |
   |             |                        |         | range            |
   |             |                        |         |                  |
   | PLAY        | md URI, NRM=1          | Play    | Play from RP     |
   |             |                        |         |                  |
   | PAUSE       | Prs URI                | Ready   | Return PP        |
   |             |                        |         |                  |
   | SC:REDIRECT | Terminate-Reason       | Ready   | Set RedP         |
   |             |                        |         |                  |
   | SC:REDIRECT | No Terminate-Reason    | Init    | Session is       |
   |             | time parameter         |         | removed          |
   |             |                        |         |                  |
   | Timeout     |                        | Init    |                  |
   |             |                        |         |                  |
   | RedP        |                        | Init    | TEARDOWN of      |
   | reached     |                        |         | session          |
   +-------------+------------------------+---------+------------------+

                          Table 15: State: Ready

   In the Ready state, see Table 15, some of the actions are depending
   on the number of media streams (NRM) in the session, i.e., aggregated
   or non-aggregated control.  A SETUP request in the Ready state can
   either add one more media stream to the session or, if the media
   stream (same URI) already is part of the session, change the



Schulzrinne, et al.      Expires August 14, 2014              [Page 262]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   transport parameters.  TEARDOWN is depending on both the Request-URI
   and the number of media streams within the session.  If the Request-
   URI is the presentations URI the whole session is torn down.  If a
   media URI is used in the TEARDOWN request and more than one media
   exists in the session, the session will remain and a session header
   is returned in the response.  If only a single media stream remains
   in the session when performing a TEARDOWN with a media URI the
   session is removed.  The number of media streams remaining after
   tearing down a media stream determines the new state.

   +----------------+-----------------------+--------+-----------------+
   | Action         | Requisite             | New    | Response        |
   |                |                       | State  |                 |
   +----------------+-----------------------+--------+-----------------+
   | PAUSE          | Prs URI               | Ready  | Set RP to       |
   |                |                       |        | present point   |
   |                |                       |        |                 |
   | End of media   | All media             | Play   | Set RP = End of |
   |                |                       |        | media           |
   |                |                       |        |                 |
   | End of range   |                       | Play   | Set RP = End of |
   |                |                       |        | range           |
   |                |                       |        |                 |
   | PLAY           | Prs URI, No range     | Play   | Play from       |
   |                |                       |        | present point   |
   |                |                       |        |                 |
   | PLAY           | Prs URI, Range        | Play   | According to    |
   |                |                       |        | range           |
   |                |                       |        |                 |
   | SC:PLAY_NOTIFY |                       | Play   | 200             |
   |                |                       |        |                 |
   | SETUP          | New URI               | Play   | 455             |
   |                |                       |        |                 |
   | SETUP          | Setuped URI           | Play   | 455             |
   |                |                       |        |                 |
   | SETUP          | Setuped URI, IFI      | Play   | Change          |
   |                |                       |        | transport       |
   |                |                       |        | param.          |
   |                |                       |        |                 |
   | TEARDOWN       | Prs URI               | Init   | No session hdr  |
   |                |                       |        |                 |
   | TEARDOWN       | md URI,NRM=1          | Init   | No Session hdr, |
   |                |                       |        | NRM=0           |
   |                |                       |        |                 |
   | TEARDOWN       | md URI                | Play   | 455             |
   |                |                       |        |                 |
   | SC:REDIRECT    | Terminate Reason with | Play   | Set RedP        |
   |                | Time parameter        |        |                 |



Schulzrinne, et al.      Expires August 14, 2014              [Page 263]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   |                |                       |        |                 |
   | SC:REDIRECT    |                       | Init   | Session is      |
   |                |                       |        | removed         |
   |                |                       |        |                 |
   | RedP reached   |                       | Init   | TEARDOWN of     |
   |                |                       |        | session         |
   |                |                       |        |                 |
   | Timeout        |                       | Init   | Stop Media      |
   |                |                       |        | playout         |
   +----------------+-----------------------+--------+-----------------+

                           Table 16: State: Play

   The Play state table, see Table 16, contains a number of requests
   that need a presentation URI (labeled as Prs URI) to work on (i.e.,
   the presentation URI has to be used as the Request-URI).  This is due
   to the exclusion of non-aggregated stream control in sessions with
   more than one media stream.

   To avoid inconsistencies between the client and server, automatic
   state transitions are avoided.  This can be seen at for example "End
   of media" event when all media has finished playing, the session
   still remains in Play state.  An explicit PAUSE request needs to be
   sent to change the state to Ready.  It may appear that there exist
   automatic transitions in "RedP reached" and "PP reached".  However,
   they are requested and acknowledged before they take place.  The time
   at which the transition will happen is known by looking at the range
   header.  If the client sends a request close in time to these
   transitions it needs to be prepared for receiving error messages, as
   the state may or may not have changed.

Appendix C.  Media Transport Alternatives

   This section defines how certain combinations of protocols, profiles
   and lower transports are used.  This includes the usage of the
   Transport header's source and destination address parameters
   "src_addr" and "dest_addr".

C.1.  RTP

   This section defines the interaction of RTSP with respect to the RTP
   protocol [RFC3550].  It also defines any necessary media transport
   signaling with regards to RTP.

   The available RTP profiles and lower layer transports are described
   below along with rules on signaling the available combinations.





Schulzrinne, et al.      Expires August 14, 2014              [Page 264]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


C.1.1.  AVP

   The usage of the "RTP Profile for Audio and Video Conferences with
   Minimal Control" [RFC3551] when using RTP for media transport over
   different lower layer transport protocols is defined below in regards
   to RTSP.

   One such case is defined within this document: the use of embedded
   (interleaved) binary data as defined in Section 14.  The usage of
   this method is indicated by including the "interleaved" parameter.

   When using embedded binary data the "src_addr" and "dest_addr" MUST
   NOT be used.  This addressing and multiplexing is used as defined
   with use of channel numbers and the interleaved parameter.

C.1.2.  AVP/UDP

   This part describes sending of RTP [RFC3550] over lower transport
   layer UDP [RFC0768] according to the profile "RTP Profile for Audio
   and Video Conferences with Minimal Control" defined in RFC 3551
   [RFC3551].  Implementations of RTP/AVP/UDP MUST implement RTCP
   (Appendix C.1.6).  This profile requires one or two uni- or bi-
   directional UDP flows per media stream.  The first UDP flow is for
   RTP and the second is for RTCP.  Multiplexing of RTP and RTCP
   (Appendix C.1.6.4) MAY be used, in which case a single UDP flow is
   used for both parts.  Embedding of RTP data with the RTSP messages,
   in accordance with Section 14, SHOULD NOT be performed when RTSP
   messages are transported over unreliable transport protocols, like
   UDP [RFC0768].

   The RTP/UDP and RTCP/UDP flows can be established using the Transport
   header's "src_addr", and "dest_addr" parameters.

   In RTSP PLAY mode, the transmission of RTP packets from client to
   server is unspecified.  The behavior in regards to such RTP packets
   MAY be defined in future.

   The "src_addr" and "dest_addr" parameters are used in the following
   way for media delivery and playback mode, i.e., Mode=PLAY:

   o  The "src_addr" and "dest_addr" parameters MUST contain either 1 or
      2 address specifications.  Note that two address specifications
      MAY be provided even if RTP and RTCP multiplexing is negotiated.

   o  Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST
      contain either:

      *  both an address and a port number, or



Schulzrinne, et al.      Expires August 14, 2014              [Page 265]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      *  a port number without an address.

   o  The first address specification given in either of the parameters
      applies to the RTP stream.  The second specification if present
      applies to the RTCP stream, unless in case RTP and RTCP
      multiplexing is negotiated where both RTP and RTCP will use the
      first specification.

   o  The RTP/UDP packets from the server to the client MUST be sent to
      the address and port given by the first address specification of
      the "dest_addr" parameter.

   o  The RTCP/UDP packets from the server to the client MUST be sent to
      the address and port given by the second address specification of
      the "dest_addr" parameter, unless RTP and RTCP multiplexing has
      been negotiated, in which case RTCP MUST be sent to the first
      address specification.  If no second pair is specified and RTP and
      RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.

   o  The RTCP/UDP packets from the client to the server MUST be sent to
      the address and port given by the second address specification of
      the "src_addr" parameter, unless RTP and RTCP multiplexing has
      been negotiated, in which case RTCP MUST be sent to the first
      address specification.  If no second pair is specified and RTP and
      RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.

   o  The RTP/UDP packets from the client to the server MUST be sent to
      the address and port given by the first address specification of
      the "src_addr" parameter.

   o  RTP and RTCP Packets SHOULD be sent from the corresponding
      receiver port, i.e., RTCP packets from the server should be sent
      from the "src_addr" parameters second address port pair, unless
      RTP and RTCP multiplexing has been negotiated in which case the
      first address port pair is used.

C.1.3.  AVPF/UDP

   The RTP profile "Extended RTP Profile for RTCP-based Feedback (RTP/
   AVPF)" [RFC4585] MAY be used as RTP profiles in sessions using RTP.
   All that is defined for AVP MUST also apply for AVPF.

   The usage of AVPF is indicated by the media initialization protocol
   used.  In the case of SDP it is indicated by media lines (m=)
   containing the profile RTP/AVPF.  That SDP MAY also contain further
   AVPF related SDP attributes configuring the AVPF session regarding
   reporting interval and feedback messages to be used [RFC4585].  This
   configuration MUST be followed.



Schulzrinne, et al.      Expires August 14, 2014              [Page 266]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


C.1.4.  SAVP/UDP

   The RTP profile "The Secure Real-time Transport Protocol (SRTP)"
   [RFC3711] is an RTP profile (SAVP) that MAY be used in RTSP sessions
   using RTP.  All that is defined for AVP MUST also apply for SAVP.

   The usage of SRTP requires that a security context is established.
   The default key-management unless otherwise signalled SHALL be MIKEY
   in RSA-R mode as defined in Appendix C.1.4.1, and not according to
   the procedure defined in "Key Management Extensions for Session
   Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)"
   [RFC4567].  The reason is that RFC 4567 sends the initial MIKEY
   message in SDP, thus both requiring the usage of the DESCRIBE method
   and forcing the server to keep state for clients performing DESCRIBE
   in anticipation that they might require key management.

   MIKEY is selected as default method for establishing SRTP
   cryptographic context within an RTSP session as it can be embedded in
   the RTSP messages, while still ensuring confidentiality of content of
   the keying material, even when using hop-by-hop TLS security for the
   RTSP messages.  This method does also support pipelining of the RTSP
   messages.

C.1.4.1.  MIKEY Key Establishment

   This method for using MIKEY [RFC3830] to establish the SRTP
   cryptographic context is initiated in the client's SETUP request, and
   the server's response to the SETUP carries the MIKEY response.  This
   ensures that the crypto context establishment happens simultaneously
   with the establishment of the media stream being protected.  By using
   MIKEY's RSA-R mode [RFC4738] the client can be the initiator and
   still allow the server to set the parameters in accordance with the
   actual media stream.

   The SRTP cryptographic context establishment is done according to the
   following process:

   1.   The client determines that SAVP or SAVPF shall be used from
        media description format, e.g., SDP.  If no other key management
        method is explicitly signalled, then MIKEY SHALL be used as
        defined herein.  The use of SRTP with RTSP is only defined with
        MIKEY with keys established as defined in this Section.  Future
        documents may define how an RTSP implementation treats SDP that
        indicates some other key mechanism to be used.  The need for
        such specification include [RFC4567] that is not defined for use
        in RTSP 2.0 within this document.





Schulzrinne, et al.      Expires August 14, 2014              [Page 267]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   2.   The client SHALL establish a TLS connection for RTSP messages,
        directly or hop by hop with the server.  If hop-by-hop TLS
        security is used, the User method SHALL be indicated in the
        Accept-Credentials header.  Note that using hop-by-hop does
        allow the proxy to insert itself as a man in the middle also in
        the MIKEY exchange by providing one of its certificates, rather
        than the server's in the Connection-Credentials header.  The
        client SHALL therefore validate the server certificate.

   3.   The client retrieves the server's certificate from a direct TLS
        connection, or if hop by hop from Connection-Credentials header.
        The client then checks that the server certificate is valid and
        belongs to the server.

   4.   The client forms the MIKEY Initiator message using RSA-R mode in
        unicast mode as specified in [RFC4738].  The client SHOULD use
        the same certificate for TLS and in MIKEY to enable the server
        to bind the two together.  The client's certificate SHALL be
        included in the MIKEY message.  The client SHALL indicate its
        SRTP capabilities in the message.

   5.   The MIKEY message from the previous step is base64 [RFC4648]
        encoded and becomes the value of the MIKEY parameter that is
        included in the transport specification(s) that specifies a SRTP
        based profile (SAVP, SAVPF) in the SETUP request.

   6.   Any proxy encountering the MIKEY parameter SHALL forward it
        without modification.  A proxy requiring to understand transport
        specification which doesn't support SAVP/SAVPF with MIKEY will
        discard the whole transport specification.  Most types of
        proxies can easily support SAVP and SAVPF with MIKEY.  If
        possible bypassing the proxy should be tried.

   7.   The server upon receiving the SETUP request, will need to decide
        upon the transport specification to use, if multiple are
        included by the client.  In the determination of which transport
        specifications that are supported and preferred, the server
        SHOULD decode the MIKEY message to take the embedded SRTP
        parameters into account.  If all transport specs require SRTP
        but no MIKEY parameter or other supported keying method is
        included, the server SHALL respond with 403.

   8.   Upon generating a response the following outcomes can occur:

        *  A transport spec not using SRTP and MIKEY is selected.  Thus
           the response will not contain any MIKEY parameter.





Schulzrinne, et al.      Expires August 14, 2014              [Page 268]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


        *  A transport spec using SRTP and MIKEY is selected but an
           error is encountered in the MIKEY processing.  In that case
           an RTSP error response code of 466 "Key Management Error"
           SHALL be used.  A MIKEY message describing the error MAY be
           included.

        *  A transport spec using SRTP and MIKEY is selected and a MIKEY
           response message can be created.  The server SHOULD use the
           same certificate for TLS and in MIKEY to enable client to
           bind the two together.  If a different certificate is used it
           SHALL be included in the MIKEY message.  It is RECOMMENDED
           that the envelope key cache type is set to 'Cache' and that a
           single envelope key is reused for all MIKEY messages to the
           client.  That message is included in the MIKEY parameter part
           of the single selected transport specification in the SETUP
           response.  The server will set the SRTP parameters as
           preferred for this media stream within the supported range by
           the client.

   9.   The server transmits the SETUP response back to the client.

   10.  The client receives the SETUP response and if the response code
        indicates a successful request it decodes the MIKEY message and
        establishes the SRTP cryptographic context from the parameters
        in the MIKEY response.

   In the above method the client's certificate may be self-signed in
   cases where the client's identity is not necessary to authenticate
   and the security goal is only to ensure that the RTSP signaling
   client is the same as the one receiving the SRTP security context.

C.1.5.  SAVPF/UDP

   The RTP profile "Extended Secure RTP Profile for RTCP-based Feedback
   (RTP/SAVPF)" [RFC5124] is an RTP profile (SAVPF) that MAY be used in
   RTSP sessions using RTP.  All that is defined for AVPF MUST also
   apply for SAVPF.

   The usage of SRTP requires that a cryptographic context is
   established.  The default mechanism for establishing that security
   association is to use MIKEY[RFC3830] with RTSP as defined in
   Appendix C.1.4.1.

C.1.6.  RTCP usage with RTSP

   RTCP has several usages when RTP is used for media transport as
   explained below.  Due to that RTCP MUST be supported if an RTSP agent
   handles RTP.



Schulzrinne, et al.      Expires August 14, 2014              [Page 269]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


C.1.6.1.  Media synchronization

   RTCP provides media synchronization and clock drift compensation.
   The initial media synchronization is available from RTP-Info header.
   However, to be able to handle any clock drift between the media
   streams, RTCP is needed.

C.1.6.2.  RTSP Session keep-alive

   RTCP traffic from the RTSP client to the RTSP server MUST function as
   keep-alive.  This requires an RTSP server supporting RTP to use the
   received RTCP packets as indications that the client desires the
   related RTSP session to be kept alive.

C.1.6.3.  Bit-rate adaption

   RTCP Receiver reports and any additional feedback from the client
   MUST be used to adapt the bit-rate used over the transport for all
   cases when RTP is sent over UDP.  An RTP sender without reserved
   resources MUST NOT use more than its fair share of the available
   resources.  This can be determined by comparing on short to medium
   term (some seconds) the used bit-rate and adapt it so that the RTP
   sender sends at a bit-rate comparable to what a TCP sender would
   achieve on average over the same path.

   To ensure that the implementation's adaptation mechanism has a well
   defined outer envelope, all implementations using a non-congestion
   controlled unicast transport protocol, like UDP, MUST implement
   Multimedia Congestion Control: Circuit Breakers for Unicast RTP
   Sessions [I-D.ietf-avtcore-rtp-circuit-breakers].

C.1.6.4.  RTP and RTCP Multiplexing

   RTSP can be used to negotiate the usage of RTP and RTCP multiplexing
   as described in [RFC5761].  This allows servers and client to reduce
   the amount of resources required for the session by only requiring
   one underlying transport stream per media stream instead of two when
   using RTP and RTCP.  This lessens the server port consumption and
   also the necessary state and keep-alive work when operating across
   Network and Address Translators [RFC2663].

   Content must be prepared with some consideration for RTP and RTCP
   multiplexing, mainly ensuring that the RTP payload types used do not
   collide with the ones used for RTCP packet types.  This option likely
   needs explicit support from the content unless the RTP payload types
   can be remapped by the server and that is correctly reflected in the
   session description.  Beyond that support of this feature should come
   at little cost and much gain.



Schulzrinne, et al.      Expires August 14, 2014              [Page 270]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   It is recommended that if the content and server support RTP and RTCP
   multiplexing that this is indicated in the session description, for
   example using the SDP attribute "a=rtcp-mux".  If the SDP message
   contains the a=rtcp-mux attribute for a media stream, the server MUST
   support RTP and RTCP multiplexing.  If indicated or otherwise desired
   by the client it can include the Transport parameter "RTCP-mux" in
   any transport specification where it desires to use RTCP-mux.  The
   server will indicate if it supports RTCP-mux.  Servers and Clients
   SHOULD support RTP and RTCP multiplexing.

   For capability exchange, an RTSP feature tag for RTP and RTCP
   multiplexing is defined: "setup.rtp.rtcp.mux".

   To minimize the risk of negotiation failure while using RTP and RTCP
   multiplexing some recommendations are here provided.  If the session
   description includes explicit indication of support (a=rtcp-mux in
   SDP), then a RTSP agent can safely create a SETUP request with a
   transport specification with only a single dest_addr parameter
   address specification.  If no such explicit indication is provided,
   then even if the feature tag "setup.rtp.rtcp.mux" is provided in a
   Supported header by the RTSP server or the feature tag included in
   the Required header in the SETUP request, the media resource may not
   support RTP and RTCP multiplexing.  Thus, to maximize the probability
   of successful negotiation the RTSP agent is recommended to include
   two dest_addr parameter address specifications in the first or first
   set (if pipelining is used) of SETUP request(s) for any media
   resource aggregate.  That way the RTSP server can either accept RTP
   and RTCP multiplexing and only use the first address specification,
   and if not use both specifications.  The RTSP agent after having
   received the response for a successful negotiation of the usage of
   RTP and RTCP multiplexing, can then release the resources associated
   with the second address specification.

C.2.  RTP over TCP

   Transport of RTP over TCP can be done in two ways: over independent
   TCP connections using RFC 4571 [RFC4571] or interleaved in the RTSP
   connection.  In both cases the protocol MUST be "rtp" and the lower
   layer MUST be TCP.  The profile may be any of the above specified
   ones; AVP, AVPF, SAVP or SAVPF.

C.2.1.  Interleaved RTP over TCP

   The use of embedded (interleaved) binary data transported on the RTSP
   connection is possible as specified in Section 14.  When using this
   declared combination of interleaved binary data the RTSP messages
   MUST be transported over TCP.  TLS may or may not be used.  If TLS is
   used both RTSP messages and the binary data will be protected by TLS.



Schulzrinne, et al.      Expires August 14, 2014              [Page 271]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   One should, however, consider that this will result in all media
   streams go through any proxy.  Using independent TCP connections can
   avoid that issue.

C.2.2.  RTP over independent TCP

   In this Appendix, it is described the sending of RTP [RFC3550] over
   lower transport layer TCP [RFC0793] according to "Framing Real-time
   Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over
   Connection-Oriented Transport" [RFC4571].  This Appendix adapts the
   guidelines for using RTP over TCP within SIP/SDP [RFC4145] to work
   with RTSP.

   A client codes the support of RTP over independent TCP by specifying
   an RTP/AVP/TCP transport option without an interleaved parameter in
   the Transport line of a SETUP request.  This transport option MUST
   include the "unicast" parameter.

   If the client wishes to use RTP with RTCP, two address specifications
   needs to be included in the dest_addr parameter.  If the client
   wishes to use RTP without RTCP, one address specification is included
   in the dest_addr parameter.  If the client wishes to multiplex RTP
   and RTCP on a single transport flow (see Appendix C.1.6.4), one or
   two address specifications are included in the dest_addr parameter in
   addition to the RTCP-mux transport parameter.  Two address
   specifications are allowed to allow successful negotiation when
   server or content can't support RTP and RTCP multiplexing.  Ordering
   rules of dest_addr ports follow the rules for RTP/AVP/UDP.

   If the client wishes to play the active role in initiating the TCP
   connection, it MAY set the "setup" parameter (See Section 18.54) on
   the Transport line to be "active", or it MAY omit the setup
   parameter, as active is the default.  If the client signals the
   active role, the ports in the address specifications in the dest_addr
   parameter MUST be set to 9 (the discard port).

   If the client wishes to play the passive role in TCP connection
   initiation, it MUST set the "setup" parameter on the Transport line
   to be "passive".  If the client is able to assume the active or the
   passive role, it MUST set the "setup" parameter on the Transport line
   to be "actpass".  In either case, the dest_addr parameter's address
   specification port value for RTP MUST be set to the TCP port number
   on which the client is expecting to receive the TCP connection for
   RTP, and the dest_addr's address specification port value for RTCP
   MUST be set to the TCP port number on which the client is expecting
   to receive the TCP connection for RTCP.  In the case that the client
   wishes to multiplex RTP and RTCP on a single transport flow, the
   RTCP-mux parameter is included and one or two dest_addr parameter



Schulzrinne, et al.      Expires August 14, 2014              [Page 272]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   address specifications are included, as mentioned earlier in this
   section.

   If upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, a
   server decides to accept this requested option, the 2xx reply MUST
   contain a Transport option that specifies RTP/AVP/TCP (without using
   the interleaved parameter, and with using the unicast parameter).
   The dest_addr parameter value MUST be echoed from the parameter value
   in the client request unless the destination address (only port) was
   not provided in which case the server MAY include the source address
   of the RTSP TCP connection with the port number unchanged.

   In addition, the server reply MUST set the setup parameter on the
   Transport line, to indicate the role the server will play in the
   connection setup.  Permissible values are "active" (if a client set
   "setup" to "passive" or "actpass") and "passive" (if a client set
   "setup" to "active" or "actpass").

   If a server sets "setup" to "passive", the "src_addr" in the reply
   MUST indicate the ports the server is willing to receive an TCP
   connection for RTP and (if the client requested an TCP connection for
   RTCP by specifying two dest_addr address specifications) an TCP/RTCP
   connection.  If a server sets "setup" to "active", the ports
   specified in "src_addr" address specifications MUST be set to 9.  The
   server MAY use the "ssrc" parameter, following the guidance in
   Section 18.54.  The server sets only one address specification in the
   case that the client has indicated only a single address
   specification or in case RTP and RTCP multiplexing was requested and
   accepted by server.  Port ordering for src_addr follows the rules for
   RTP/AVP/UDP.

   Servers MUST support taking the passive role and MAY support taking
   the active role.  Servers with a public IP address take the passive
   role, thus enabling clients behind NATs and Firewalls a better chance
   of successful connect to the server by actively connecting outwards.
   Therefore the clients are RECOMMENDED to take the active role.

   After sending (receiving) a 2xx reply for a SETUP method for a non-
   interleaved RTP/AVP/TCP media stream, the active party SHOULD
   initiate the TCP connection as soon as possible.  The client MUST NOT
   send a PLAY request prior to the establishment of all the TCP
   connections negotiated using SETUP for the session.  In case the
   server receives a PLAY request in a session that has not yet
   established all the TCP connections, it MUST respond using the 464
   "Data Transport Not Ready Yet" (Section 17.4.29) error code.

   Once the PLAY request for a media resource transported over non-
   interleaved RTP/AVP/TCP occurs, media begins to flow from server to



Schulzrinne, et al.      Expires August 14, 2014              [Page 273]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   client over the RTP TCP connection, and RTCP packets flow
   bidirectionally over the RTCP TCP connection.  Unless RTP and RTCP
   multiplexing has been negotiated in which case RTP and RTCP will flow
   over a common TCP connection.  As in the RTP/UDP case, client to
   server traffic on a RTP only TCP session is unspecified by this memo.
   The packets that travel on these connections MUST be framed using the
   protocol defined in [RFC4571], not by the framing defined for
   interleaving RTP over the RTSP connection defined in Section 14.

   A successful PAUSE request for a media being transported over RTP/AVP
   /TCP pauses the flow of packets over the connections, without closing
   the connections.  A successful TEARDOWN request signals that the TCP
   connections for RTP and RTCP are to be closed by the RTSP client as
   soon as possible.

   Subsequent SETUP requests on an already-SETUP RTP/AVP/TCP URI may be
   ambiguous in the following way: does the client wish to open up new
   TCP connection for RTP or RTCP for the URI, or does the client wish
   to continue using the existing TCP connections?  The client SHOULD
   use the "connection" parameter (defined in Section 18.54) on the
   Transport line to make its intention clear (by setting "connection"
   to "new" if new connections are needed, and by setting "connection"
   to "existing" if the existing connections are to be used).  After a
   2xx reply for a SETUP request for a new connection, parties should
   close the pre-existing connections, after waiting a suitable period
   for any stray RTP or RTCP packets to arrive.

   The usage of SRTP, i.e., either SAVP or SAVPF profiles, requires that
   a security association is established.  The default mechanism for
   establishing that security association is to use MIKEY[RFC3830] with
   RTSP as defined Appendix C.1.4.1.

   Below, a rewriten version of the example "media on demand"
   (Appendix A.1) shows the use of RTP/AVP/TCP non-interleaved:

















Schulzrinne, et al.      Expires August 14, 2014              [Page 274]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
            CSeq: 1
            User-Agent: PhonyClient/1.2

      M->C: RTSP/2.0 200 OK
            CSeq: 1
            Server: PhonyServer/1.0
            Date: Wed, 23 Jan 2013 15:36:52 +0000
            Content-Type: application/sdp
            Content-Length: 227
            Content-Base: rtsp://example.com/twister.3gp/
            Expires: Thu, 24 Jan 2013 15:36:52 +0000

            v=0
            o=- 2890844256 2890842807 IN IP4 198.51.100.34
            s=RTSP Session
            i=An Example of RTSP Session Usage
            e=adm@example.com
            c=IN IP4 0.0.0.0
            a=control: *
            a=range:npt=00:00:00-00:10:34.10
            t=0 0
            m=audio 0 RTP/AVP 0
            a=control: trackID=1

      C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
            CSeq: 2
            User-Agent: PhonyClient/1.2
            Require: play.basic
            Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9";
                       setup=active;connection=new
            Accept-Ranges: npt, smpte, clock



















Schulzrinne, et al.      Expires August 14, 2014              [Page 275]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      M->C: RTSP/2.0 200 OK
            CSeq: 2
            Server: PhonyServer/1.0
            Transport: RTP/AVP/TCP;unicast;
                       dest_addr=":9"/":9";
                       src_addr="198.51.100.5:53478"/"198.51.100:54091";
                       setup=passive;connection=new;ssrc=93CB001E
            Session: 12345678
            Expires: Thu, 24 Jan 2013 15:36:52 +0000
            Date: Wed, 23 Jan 2013 15:36:52 +0000
            Accept-Ranges: npt
            Media-Properties: Random-Access=0.8, Immutable, Unlimited

      C->M: TCP Connection Establishment x2

      C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
            CSeq: 4
            User-Agent: PhonyClient/1.2
            Range: npt=30-
            Session: 12345678

      M->C: RTSP/2.0 200 OK
            CSeq: 4
            Server: PhonyServer/1.0
            Date: Wed, 23 Jan 2013 15:36:54 +0000
            Session: 12345678
            Range: npt=30-623.10
            Seek-Style: First-Prior
            RTP-Info:  url="rtsp://example.com/twister.3gp/trackID=1"
               ssrc=4F312DD8:seq=54321;rtptime=2876889

C.3.  Handling Media Clock Time Jumps in the RTP Media Layer

   RTSP allows media clients to control selected, non-contiguous
   sections of media presentations, rendering those streams with an RTP
   media layer [RFC3550].  Two cases occur, the first is when a new PLAY
   request replaces an old ongoing request and the new request results
   in a jump in the media.  This should produce in the RTP layer a
   continuous media stream.  A client may also directly following a
   completed PLAY request perform a new PLAY request.  This will result
   in some gap in the media layer.  The below text will look into both
   cases.

   A PLAY request that replaces an ongoing request allows the media
   layer rendering the RTP stream without being affected by jumps in
   media clock time.  The RTP timestamps for the new media range is set
   so that they become continuous with the previous media range in the
   previous request.  The RTP sequence number for the first packet in



Schulzrinne, et al.      Expires August 14, 2014              [Page 276]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   the new range will be the next following the last packet in the
   previous range, i.e., monotonically increasing.  The goal is to allow
   the media rendering layer to work without interruption or
   reconfiguration across the jumps in media clock.  This should be
   possible in all cases of replaced PLAY requests for media that has
   random-access properties.  In this case care is needed to align
   frames or similar media dependent structures.

   In cases where jumps in media clock time are a result of RTSP
   signaling operations arriving after a completed PLAY operation, the
   request timing will result in that media becomes non-continuous.  The
   server becomes unable to send the media so that it arrives timely and
   still carry timestamps to make the media stream continuous.  In these
   cases the server will produce RTP streams where there are gaps in the
   RTP timeline for the media.  In such cases, if the media has frame
   structure, aligning the timestamp for the next frame with the
   previous structure reduces the burden to render this media.  The gap
   should represent the time the server hasn't been serving media, e.g.,
   the time between the end of the media stream or a PAUSE request and
   the new PLAY request.  In these cases the RTP sequence number would
   normally be monotonically increasing across the gap.

   For RTSP sessions with media that lacks random access properties,
   such as live streams, any media clock jump is commonly the result of
   a correspondingly long pause of delivery.  The RTP timestamp will
   have increased in direct proportion to the duration of the paused
   delivery.  Note also that in this case the RTP sequence number should
   be the next packet number.  If not, the RTCP packet loss reporting
   will indicate as loss all packets not received between the point of
   pausing and later resuming.  This may trigger congestion avoidance
   mechanisms.  An allowed exception from the above recommendation on
   monotonically increasing RTP sequence number is live media streams,
   likely being relayed.  In this case, when the client resumes
   delivery, it will get the media that is currently being delivered to
   the server itself.  For this type of basic delivery of live streams
   to multiple users over unicast, individual rewriting of RTP sequence
   numbers becomes quite a burden.  For solutions that anyway caches
   media, timeshifts, etc, the rewriting should be a minor issue.

   The goal when handling jumps in media clock time is that the provided
   stream is continuous without gaps in RTP timestamp or sequence
   number.  However, when delivery has been halted for some reason the
   RTP timestamp when resuming MUST represent the duration the delivery
   was halted.  RTP sequence number MUST generally be the next number,
   i.e., monotonically increasing modulo 65536.  For media resources
   with the properties Time-Progressing and Time-Duration=0.0 the server
   MAY create RTP media streams with RTP sequence number jumps in them
   due to the client first halting delivery and later resuming it (PAUSE



Schulzrinne, et al.      Expires August 14, 2014              [Page 277]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   and then later PLAY).  However, servers utilizing this exception must
   take into consideration the resulting RTCP receiver reports that
   likely contain loss reports for all the packets part of the
   discontinuity.  A client cannot rely on that a server will align when
   resuming playing even if it is RECOMMENDED.  The RTP-Info header will
   provide information on how the server acts in each case.

      One cannot assume that the RTSP client can communicate with the
      RTP media agent, as the two may be independent processes.  If the
      RTP timestamp shows the same gap as the NPT, the media agent will
      assume that there is a pause in the presentation.  If the jump in
      NPT is large enough, the RTP timestamp may roll over and the media
      agent may believe later packets to be duplicates of packets just
      played out.  Having the RTP timestamp jump will also affect the
      RTCP measurements based on this.

   As an example, assume an RTP timestamp frequency of 8000 Hz, a
   packetization interval of 100 ms and an initial sequence number and
   timestamp of zero.

      C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
        CSeq: 4
        Session: abcdefgh
        Range: npt=10-15
        User-Agent: PhonyClient/1.2

      S->C: RTSP/2.0 200 OK
        CSeq: 4
        Session: abcdefgh
        Range: npt=10-15
        RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                  ssrc=0D12F123:seq=0;rtptime=0

   The ensuing RTP data stream is depicted below:


      S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s
      S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s
       . . .
      S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s


   Upon the completion of the requested delivery the server sends a
   PLAY_NOTIFY







Schulzrinne, et al.      Expires August 14, 2014              [Page 278]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


        S->C: PLAY_NOTIFY rtsp://example.com/fizzle RTSP/2.0
              CSeq: 5
              Notify-Reason: end-of-stream
              Request-Status: cseq=4 status=200 reason="OK"
              Range: npt=-15
              RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=49;rtptime=39200
              Session: abcdefgh

        C->S: RTSP/2.0 200 OK
              CSeq: 5
              User-Agent: PhonyClient/1.2

   Upon the completion of the play range, the client follows up with a
   request to PLAY from a new NPT.

   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
         CSeq: 6
         Session: abcdefg
         Range: npt=18-20
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 6
         Session: abcdefg
         Range: npt=18-20
         RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=50;rtptime=40100

   The ensuing RTP data stream is depicted below:

      S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
      S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
       . . .
      S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s

   In this example, first, NPT 10 through 15 is played, then the client
   requests the server to skip ahead and play NPT 18 through 20.  The
   first segment is presented as RTP packets with sequence numbers 0
   through 49 and timestamp 0 through 39,200.  The second segment
   consists of RTP packets with sequence number 50 through 69, with
   timestamps 40,100 through 55,200.  While there is a gap in the NPT,
   there is no gap in the sequence number space of the RTP data stream.

   The RTP timestamp gap is present in the above example due to the time
   it takes to perform the second play request, in this case 12.5 ms
   (100/8000).




Schulzrinne, et al.      Expires August 14, 2014              [Page 279]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


C.4.  Handling RTP Timestamps after PAUSE

   During a PAUSE / PLAY interaction in an RTSP session, the duration of
   time for which the RTP transmission was halted MUST be reflected in
   the RTP timestamp of each RTP stream.  The duration can be calculated
   for each RTP stream as the time elapsed from when the last RTP packet
   was sent before the PAUSE request was received and when the first RTP
   packet was sent after the subsequent PLAY request was received.  The
   duration includes all latency incurred and processing time required
   to complete the request.

      The RTP RFC [RFC3550] states that: The RTP timestamp for each unit
      [packet] would be related to the wallclock time at which the unit
      becomes current on the virtual presentation timeline.

      In order to satisfy the requirements of [RFC3550], the RTP
      timestamp space needs to increase continuously with real time.
      While this is not optimal for stored media, it is required for RTP
      and RTCP to function as intended.  Using a continuous RTP
      timestamp space allows the same timestamp model for both stored
      and live media and allows better opportunity to integrate both
      types of media under a single control.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero.

   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15

         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15
         RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=0;rtptime=0

   The ensuing RTP data stream is depicted below:

      S -> C: RTP packet - seq = 0, rtptime = 0,    NPT time = 10s
      S -> C: RTP packet - seq = 1, rtptime = 800,  NPT time = 10.1s
      S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
      S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s




Schulzrinne, et al.      Expires August 14, 2014              [Page 280]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   The client then sends a PAUSE request:

   C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0
         CSeq: 5
         Session: abcdefg
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 5
         Session: abcdefg
         Range: npt=10.4-15

   20 seconds elapse and then the client sends a PLAY request.  In
   addition the server requires 15 ms to process the request:

   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
         CSeq: 6
         Session: abcdefg
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 6
         Session: abcdefg
         Range: npt=10.4-15
         RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=4;rtptime=164400

   The ensuing RTP data stream is depicted below:

      S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
      S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
      S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s

   First, NPT 10 through 10.3 is played, then a PAUSE is received by the
   server.  After 20 seconds a PLAY is received by the server which
   takes 15 ms to process.  The duration of time for which the session
   was paused is reflected in the RTP timestamp of the RTP packets sent
   after this PLAY request.

   A client can use the RTSP range header and RTP-Info header to map NPT
   time of a presentation with the RTP timestamp.

   Note: In RFC 2326 [RFC2326], this matter was not clearly defined and
   was misunderstood commonly.  However, for RTSP 2.0 it is expected
   that this will be handled correctly and no exception handling will be
   required.





Schulzrinne, et al.      Expires August 14, 2014              [Page 281]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Note further: It may be required to reset some of the state to ensure
   the correct media decoding and the usual jitter-buffer handling when
   issuing a PLAY request.

C.5.  RTSP / RTP Integration

   For certain data types, tight integration between the RTSP layer and
   the RTP layer will be necessary.  This by no means precludes the
   above restrictions.  Combined RTSP/RTP media clients should use the
   RTP-Info field to determine whether incoming RTP packets were sent
   before or after a seek or before or after a PAUSE.

C.6.  Scaling with RTP

   For scaling (see Section 18.46), RTP timestamps should correspond to
   the rendering timing.  For example, when playing video recorded at 30
   frames/second at a scale of two and speed (Section 18.50) of one, the
   server would drop every second frame to maintain and deliver video
   packets with the normal timestamp spacing of 3,000 per frame, but NPT
   would increase by 1/15 second for each video frame.

      Note: The above scaling puts requirements on the media codec or a
      media stream to support it.  For example motion JPEG or other non-
      predictive video coding can easier handle the above example.

C.7.  Maintaining NPT synchronization with RTP timestamps

   The client can maintain a correct display of NPT (Normal Play Time)
   by noting the RTP timestamp value of the first packet arriving after
   repositioning.  The sequence parameter of the RTP-Info
   (Section 18.45) header provides the first sequence number of the next
   segment.

C.8.  Continuous Audio

   For continuous audio, the server SHOULD set the RTP marker bit at the
   beginning of serving a new PLAY request or at jumps in timeline.
   This allows the client to perform playout delay adaptation.

C.9.  Multiple Sources in an RTP Session

   Note that more than one SSRC MAY be sent in the media stream.  If it
   happens all sources are expected to be rendered simultaneously.








Schulzrinne, et al.      Expires August 14, 2014              [Page 282]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


C.10.  Usage of SSRCs and the RTCP BYE Message During an RTSP Session

   The RTCP BYE message indicates the end of use of a given SSRC.  If
   all sources leave an RTP session, it can, in most cases, be assumed
   to have ended.  Therefore, a client or server MUST NOT send an RTCP
   BYE message until it has finished using a SSRC.  A server SHOULD keep
   using a SSRC until the RTP session is terminated.  Prolonging the use
   of a SSRC allows the established synchronization context associated
   with that SSRC to be used to synchronize subsequent PLAY requests
   even if the PLAY response is late.

   An SSRC collision with the SSRC that transmits media does also have
   consequences, as it will normally force the media sender to change
   its SSRC in accordance with the RTP specification [RFC3550].
   However, an RTSP server may wait and see if the client changes and
   thus resolve the conflict to minimize the impact.  As media sender
   SSRC change will result in a loss of synchronization context, and
   require any receiver to wait for RTCP sender reports for all media
   requiring synchronization before being able to play out synchronized.
   Due to these reasons a client joining a session should take care to
   not select the same SSRC(s) as the server indicates in the ssrc
   Transport header parameter.  Any SSRC signalled in the Transport
   header MUST be avoided.  A client detecting a collision prior to
   sending any RTP or RTCP messages SHALL also select a new SSRC.

C.11.  Future Additions

   It is the intention that any future protocol or profile regarding
   media delivery and lower transport should be easy to add to RTSP.
   This section provides the necessary steps that needs to be meet.

   The following things needs to be considered when adding a new
   protocol or profile for use with RTSP:

   o  The protocol or profile needs to define a name tag representing
      it.  This tag is required to be an ABNF "token" to be possible to
      use in the Transport header specification.

   o  The useful combinations of protocol, profiles and lower layer
      transport for this extension needs to be defined.  For each
      combination declare the necessary parameters to use in the
      Transport header.

   o  For new media protocols the interaction with RTSP needs to be
      addressed.  One important factor will be the media
      synchronization.  It may be necessary to have new headers similar
      to RTP info to carry this information.




Schulzrinne, et al.      Expires August 14, 2014              [Page 283]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  Discuss congestion control for media, especially if transport
      without built in congestion control is used.

   See the IANA section (Section 22) for information how to register new
   attributes.

Appendix D.  Use of SDP for RTSP Session Descriptions

   The Session Description Protocol (SDP, [RFC4566]) may be used to
   describe streams or presentations in RTSP.  This description is
   typically returned in reply to a DESCRIBE request on a URI from a
   server to a client, or received via HTTP from a server to a client.

   This appendix describes how an SDP file determines the operation of
   an RTSP session.  Thus, it is worth pointing out that the
   interpretation of the SDP is done in the context of the SDP receiver,
   which is the one being configured.  This is the same as in SAP
   [RFC2974]; this differs from SDP Offer/Answer [RFC3264] where each
   SDP is interpreted in the context of the agent providing it.

   SDP as is provides no mechanism by which a client can distinguish,
   without human guidance, between several media streams to be rendered
   simultaneously and a set of alternatives (e.g., two audio streams
   spoken in different languages).  The SDP extension "Grouping of Media
   Lines in the Session Description Protocol (SDP)" [RFC5888] provides
   such functionality to some degree.  Appendix D.4 describes the usage
   of SDP media line grouping for RTSP.

D.1.  Definitions

   The terms "session-level", "media-level" and other key/attribute
   names and values used in this appendix are to be used as defined in
   SDP[RFC4566]:

D.1.1.  Control URI

   The "a=control:" attribute is used to convey the control URI.  This
   attribute is used both for the session and media descriptions.  If
   used for individual media, it indicates the URI to be used for
   controlling that particular media stream.  If found at the session
   level, the attribute indicates the URI for aggregate control
   (presentation URI).  The session level URI MUST be different from any
   media level URI.  The presence of a session level control attribute
   MUST be interpreted as support for aggregated control.  The control
   attribute MUST be present on media level unless the presentation only
   contains a single media stream, in which case the attribute MAY be
   present on the session level only and then also apply to that single
   media stream.



Schulzrinne, et al.      Expires August 14, 2014              [Page 284]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   ABNF for the attribute is defined in Section 20.3.

   Example:

     a=control:rtsp://example.com/foo

   This attribute MAY contain either relative or absolute URIs,
   following the rules and conventions set out in RFC 3986 [RFC3986].
   Implementations MUST look for a base URI in the following order:

   1.  the RTSP Content-Base field;

   2.  the RTSP Content-Location field;

   3.  the RTSP Request-URI.

   If this attribute contains only an asterisk (*), then the URI MUST be
   treated as if it were an empty embedded URI, and thus inherit the
   entire base URI.

      Note, RFC 2326 was very unclear on the processing of relative URI
      and several RTSP 1.0 implementations at the point of publishing
      this document did not perform RFC 3986 processing to determine the
      resulting URI, instead simple concatenation is common.  To avoid
      this issue completely it is recommended to use absolute URI in the
      SDP.

   The URI handling for SDPs from container files need special
   consideration.  For example let's assume that a container file has
   the URI: "rtsp://example.com/container.mp4".  Let's further assume
   this URI is the base URI, and that there is an absolute media level
   URI: "rtsp://example.com/container.mp4/trackID=2".  A relative media
   level URI that resolves in accordance with RFC 3986 [RFC3986] to the
   above given media URI is: "container.mp4/trackID=2".  It is usually
   not desirable to need to include in or modify the SDP stored within
   the container file with the server local name of the container file.
   To avoid this, one can modify the base URI used to include a trailing
   slash, e.g., "rtsp://example.com/container.mp4/".  In this case the
   relative URI for the media will only need to be: "trackID=2".
   However, this will also mean that using "*" in the SDP will result in
   control URI including the trailing slash, i.e., "rtsp://example.com/
   container.mp4/".

      Note: The usage of TrackID in the above is not a standardized
      form, but one example out of several similar strings such as
      TrackID, Track_ID, StreamID that is used by different server
      vendors to indicate a particular piece of media inside a container
      file.



Schulzrinne, et al.      Expires August 14, 2014              [Page 285]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


D.1.2.  Media Streams

   The "m=" field is used to enumerate the streams.  It is expected that
   all the specified streams will be rendered with appropriate
   synchronization.  If the session is over multicast, the port number
   indicated SHOULD be used for reception.  The client MAY try to
   override the destination port, through the Transport header.  The
   servers MAY allow this, the response will indicate if allowed or not.
   If the session is unicast, the port numbers are the ones RECOMMENDED
   by the server to the client, about which receiver ports to use; the
   client MUST still include its receiver ports in its SETUP request.
   The client MAY ignore this recommendation.  If the server has no
   preference, it SHOULD set the port number value to zero.

   The "m=" lines contain information about which transport protocol,
   profile, and possibly lower-layer is to be used for the media stream.
   The combination of transport, profile and lower layer, like RTP/AVP/
   UDP needs to be defined for how to be used with RTSP.  The currently
   defined combinations are defined in Appendix C, further combinations
   MAY be specified.

   Example:

     m=audio 0 RTP/AVP 31

D.1.3.  Payload Type(s)

   The payload type(s) are specified in the "m=" line.  In case the
   payload type is a static payload type from RFC 3551 [RFC3551], no
   other information may be required.  In case it is a dynamic payload
   type, the media attribute "rtpmap" is used to specify what the media
   is.  The "encoding name" within the "rtpmap" attribute may be one of
   those specified in [RFC4856], or a media type registered with IANA
   according to [RFC4855], or an experimental encoding as specified in
   SDP [RFC4566]).  Codec-specific parameters are not specified in this
   field, but rather in the "fmtp" attribute described below.

   The selection of the RTP payload type numbers used may be required to
   consider RTP and RTCP Multiplexing [RFC5761] if that is to be
   supported by the server.

D.1.4.  Format-Specific Parameters

   Format-specific parameters are conveyed using the "fmtp" media
   attribute.  The syntax of the "fmtp" attribute is specific to the
   encoding(s) that the attribute refers to.  Note that some of the
   format specific parameters may be specified outside of the fmtp




Schulzrinne, et al.      Expires August 14, 2014              [Page 286]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   parameters, like for example the "ptime" attribute for most audio
   encodings.

D.1.5.  Directionality of media stream

   The SDP attributes "a=sendrecv", "a=recvonly" and "a=sendonly"
   provide instructions about the direction the media streams flow
   within a session.  When using RTSP the SDP can be delivered to a
   client using either RTSP DESCRIBE or a number of RTSP external
   methods, like HTTP, FTP, and email.  Based on this the SDP applies to
   how the RTSP client will see the complete session.  Thus media
   streams delivered from the RTSP server to the client, would be given
   the "a=recvonly" attribute.

   "a=recvonly" in a SDP provided to the RTSP client indicates that
   media delivery will only occur in the direction from the RTSP server
   to the client.  SDP provided to the RTSP client that lacks any of the
   directionality attributes (a=recvonly, a=sendonly, a=sendrecv) would
   be interpreted as having a=sendrecv.  At the time of writing there
   exist no RTSP mode suitable for media traffic in the direction from
   the RTSP client to the server.  Thus all RTSP SDP SHOULD have
   a=recvonly attribute when using the PLAY mode defined in this
   document.  If future modes are defined for media in client to server
   direction, then usage of a=sendonly, or a=sendrecv may become
   suitable to indicate intended media directions.

D.1.6.  Range of Presentation

   The "a=range" attribute defines the total time range of the stored
   session or an individual media.  Non-seekable live sessions can be
   indicated as specified below, while the length of live sessions can
   be deduced from the "t=" and "r=" SDP parameters.

   The attribute is both a session and a media level attribute.  For
   presentations that contain media streams of the same duration, the
   range attribute SHOULD only be used at session-level.  In case of
   different lengths the range attribute MUST be given at media level
   for all media, and SHOULD NOT be given at session level.  If the
   attribute is present at both media level and session level the media
   level values MUST be used.

   Note: Usually one will specify the same length for all media, even if
   there isn't media available for the full duration on all media.
   However, that requires that the server accepts PLAY requests within
   that range.

   Servers MUST take care to provide RTSP Range (see Section 18.40)
   values that are consistent with what is presented in the SDP for the



Schulzrinne, et al.      Expires August 14, 2014              [Page 287]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   content.  There is no reason for non dynamic content, like media
   clips provided on demand to have inconsistent values.  Inconsistent
   values between the SDP and the actual values for the content handled
   by the server is likely to generate some failure, like 457 "Invalid
   Range", in case the client uses PLAY requests with a Range header.
   In case the content is dynamic in length and it is infeasible to
   provide a correct value in the SDP the server is recommended to
   describe this as non-seekable content (see below).  The server MAY
   override that property in the response to a PLAY request using the
   correct values in the Range header.

   The unit is specified first, followed by the value range.  The units
   and their values are as defined in Section 4.4.1, Section 4.4.2 and
   Section 4.4.3 and MAY be extended with further formats.  Any open
   ended range (start-), i.e., without stop range, is of unspecified
   duration and MUST be considered as non-seekable content unless this
   property is overridden.  Multiple instances carrying different clock
   formats MAY be included at either session or media level.

   ABNF for the attribute is defined in Section 20.3.

   Examples:

     a=range:npt=0-34.4368
     a=range:clock=19971113T211503Z-19971113T220300Z
     Non seekable stream of unknown duration:
     a=range:npt=0-

D.1.7.  Time of Availability

   The "t=" field defines when the SDP is valid.  For on-demand content
   the server SHOULD indicate a stop time value for which it guarantees
   the description to be valid, and a start time that is equal to or
   before the time at which the DESCRIBE request was received.  It MAY
   also indicate start and stop times of 0, meaning that the session is
   always available.

   For sessions that are of live type, i.e., specific start time,
   unknown stop time, likely unseekable, the "t=" and "r=" field SHOULD
   be used to indicate the start time of the event.  The stop time
   SHOULD be given so that the live event will have ended at that time,
   while still not be unnecessary long into the future.

D.1.8.  Connection Information

   In SDP used with RTSP, the "c=" field contains the destination
   address for the media stream.  If a multicast address is specified
   the client SHOULD use this address in any SETUP request as



Schulzrinne, et al.      Expires August 14, 2014              [Page 288]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   destination address, including any additional parameters, such as
   TTL.  For on-demand unicast streams and some multicast streams, the
   destination address MAY be specified by the client via the SETUP
   request, thus overriding any specified address.  To identify streams
   without a fixed destination address, where the client is required to
   specify a destination address, the "c=" field SHOULD be set to a null
   value.  For addresses of type "IP4", this value MUST be "0.0.0.0",
   and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" (can also be
   written as "::"), i.e., the unspecified address according to RFC 4291
   [RFC4291].

D.1.9.  Message Body Tag

   The optional "a=mtag" attribute identifies a version of the session
   description.  It is opaque to the client.  SETUP requests may include
   this identifier in the If-Match field (see Section 18.24) to only
   allow session establishment if this attribute value still corresponds
   to that of the current description.  The attribute value is opaque
   and may contain any character allowed within SDP attribute values.

   ABNF for the attribute is defined in Section 20.3.

   Example:

     a=mtag:"158bb3e7c7fd62ce67f12b533f06b83a"

      One could argue that the "o=" field provides identical
      functionality.  However, it does so in a manner that would put
      constraints on servers that need to support multiple session
      description types other than SDP for the same piece of media
      content.

D.2.  Aggregate Control Not Available

   If a presentation does not support aggregate control no session level
   "a=control:" attribute is specified.  For a SDP with multiple media
   sections specified, each section will have its own control URI
   specified via the "a=control:" attribute.

   Example:











Schulzrinne, et al.      Expires August 14, 2014              [Page 289]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   v=0
   o=- 2890844256 2890842807 IN IP4 192.0.2.56
   s=I came from a web page
   e=adm@example.com
   c=IN IP4 0.0.0.0
   t=0 0
   m=video 8002 RTP/AVP 31
   a=control:rtsp://audio.example.com/movie.aud
   m=audio 8004 RTP/AVP 3
   a=control:rtsp://video.example.com/movie.vid

   Note that the position of the control URI in the description implies
   that the client establishes separate RTSP control sessions to the
   servers audio.example.com and video.example.com.

   It is recommended that an SDP file contains the complete media
   initialization information even if it is delivered to the media
   client through non-RTSP means.  This is necessary as there is no
   mechanism to indicate that the client should request more detailed
   media stream information via DESCRIBE.

D.3.  Aggregate Control Available

   In this scenario, the server has multiple streams that can be
   controlled as a whole.  In this case, there are both a media-level
   "a=control:" attributes, which are used to specify the stream URIs,
   and a session-level "a=control:" attribute which is used as the
   Request-URI for aggregate control.  If the media-level URI is
   relative, it is resolved to absolute URIs according to Appendix D.1.1
   above.

   Example:



















Schulzrinne, et al.      Expires August 14, 2014              [Page 290]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   C->M: DESCRIBE rtsp://example.com/movie RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 1
         Date: Wed, 23 Jan 2013 15:36:52 +0000
         Expires: Wed, 23 Jan 2013 16:36:52 +0000
         Content-Type: application/sdp
         Content-Base: rtsp://example.com/movie/
         Content-Length: 227

         v=0
         o=- 2890844256 2890842807 IN IP4 192.0.2.211
         s=I contain
         i=<more info>
         e=adm@example.com
         c=IN IP4 0.0.0.0
         a=control:*
         t=0 0
         m=video 8002 RTP/AVP 31
         a=control:trackID=1
         m=audio 8004 RTP/AVP 3
         a=control:trackID=2

   In this example, the client is recommended to establish a single RTSP
   session to the server, and uses the URIs rtsp://example.com/movie/
   trackID=1 and rtsp://example.com/movie/trackID=2 to set up the video
   and audio streams, respectively.  The URI rtsp://example.com/movie/,
   which is resolved from the "*", controls the whole presentation
   (movie).

   A client is not required to issue SETUP requests for all streams
   within an aggregate object.  Servers should allow the client to ask
   for only a subset of the streams.

D.4.  Grouping of Media Lines in SDP

   For some types of media it is desirable to express a relationship
   between various media components, for instance, for lip
   synchronization or Scalable Video Codec (SVC) [RFC5583].  This
   relationship is expressed on the SDP level by grouping of media
   lines, as described in [RFC5888] and can be exposed to RTSP.

   For RTSP it is mainly important to know how to handle grouped medias
   received by means of SDP, i.e., if the media are under aggregate
   control (see Appendix D.3) or if aggregate control is not available
   (see Appendix D.2).



Schulzrinne, et al.      Expires August 14, 2014              [Page 291]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   It is RECOMMENDED that grouped medias are handled by aggregate
   control, to give the client the ability to control either the whole
   presentation or single medias.

D.5.  RTSP external SDP delivery

   There are some considerations that need to be made when the session
   description is delivered to the client outside of RTSP, for example
   via HTTP or email.

   First of all, the SDP needs to contain absolute URIs, since relative
   will in most cases not work as the delivery will not correctly
   forward the base URI.

   The writing of the SDP session availability information, i.e., "t="
   and "r=", needs to be carefully considered.  When the SDP is fetched
   by the DESCRIBE method, the probability that it is valid is very
   high.  However, the same is much less certain for SDPs distributed
   using other methods.  Therefore the publisher of the SDP should take
   care to follow the recommendations about availability in the SDP
   specification [RFC4566] in Section 4.2.

Appendix E.  RTSP Use Cases

   This Appendix describes the most important and considered use cases
   for RTSP.  They are listed in descending order of importance in
   regards to ensuring that all necessary functionality is present.
   This specification only fully supports usage of the two first.  Also
   in these first two cases, there are special cases or exceptions that
   are not supported without extensions, e.g., the redirection of media
   delivery to another address than the controlling agent's (client's).

E.1.  On-demand Playback of Stored Content

   An RTSP capable server stores content suitable for being streamed to
   a client.  A client desiring playback of any of the stored content
   uses RTSP to set up the media transport required to deliver the
   desired content.  RTSP is then used to initiate, halt and manipulate
   the actual transmission (playout) of the content.  RTSP is also
   required to provide necessary description and synchronization
   information for the content.

   The above high level description can be broken down into a number of
   functions that RTSP needs to be capable of.

   Presentation Description:  Provide initialization information about
         the presentation (content); for example, which media codecs are
         needed for the content.  Other information that is important



Schulzrinne, et al.      Expires August 14, 2014              [Page 292]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         includes the number of media streams the presentation contains,
         the transport protocols used for the media streams, and
         identifiers for these media streams.  This information is
         required before setup of the content is possible and to
         determine if the client is even capable of using the content.

         This information need not be sent using RTSP; other external
         protocols can be used to transmit the transport presentation
         descriptions.  Two good examples are the use of HTTP [RFC2616]
         or email to fetch or receive presentation descriptions like SDP
         [RFC4566]

   Setup:  Set up some or all of the media streams in a presentation.
         The setup itself consists of selecting the protocol for media
         transport and the necessary parameters for the protocol, like
         addresses and ports.

   Control of Transmission:  After the necessary media streams have been
         established the client can request the server to start
         transmitting the content.  The client must be allowed to start
         or stop the transmission of the content at arbitrary times.
         The client must also be able to start the transmission at any
         point in the timeline of the presentation.

   Synchronization:  For media transport protocols like RTP [RFC3550] it
         might be beneficial to carry synchronization information within
         RTSP.  This may be due to either the lack of inter-media
         synchronization within the protocol itself, or the potential
         delay before the synchronization is established (which is the
         case for RTP when using RTCP).

   Termination:  Terminate the established contexts.

   For this use case there are a number of assumptions about how it
   works.  These are:

   On-Demand content:  The content is stored at the server and can be
         accessed at any time during a time period when it is intended
         to be available.

   Independent sessions:  A server is capable of serving a number of
         clients simultaneously, including from the same piece of
         content at different points in that presentations time-line.

   Unicast Transport:  Content for each individual client is transmitted
         to them using unicast traffic.





Schulzrinne, et al.      Expires August 14, 2014              [Page 293]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   It is also possible to redirect the media traffic to a different
   destination than that of the agent controlling the traffic.  However,
   allowing this without appropriate mechanisms for checking that the
   destination approves of this allows for distributed denial of service
   attacks (DDoS).

E.2.  Unicast Distribution of Live Content

   This use case is similar to the above on-demand content case (see
   Appendix E.1) the difference is the nature of the content itself.
   Live content is continuously distributed as it becomes available from
   a source; i.e., the main difference from on-demand is that one starts
   distributing content before the end of it has become available to the
   server.

   In many cases the consumer of live content is only interested in
   consuming what actually happens "now"; i.e., very similar to
   broadcast TV.  However, in this case it is assumed that there exists
   no broadcast or multicast channel to the users, and instead the
   server functions as a distribution node, sending the same content to
   multiple receivers, using unicast traffic between server and client.
   This unicast traffic and the transport parameters are individually
   negotiated for each receiving client.

   Another aspect of live content is that it often has a very limited
   time of availability, as it is only available for the duration of the
   event the content covers.  An example of such a live content could be
   a music concert which lasts 2 hour and starts at a predetermined
   time.  Thus there is a need to announce when and for how long the
   live content is available.

   In some cases, the server providing live content may be saving some
   or all of the content to allow clients to pause the stream and resume
   it from the paused point, or to "rewind" and play continuously from a
   point earlier than the live point.  Hence, this use case does not
   necessarily exclude playing from other than the live point of the
   stream, playing with scales other than 1.0, etc.

E.3.  On-demand Playback using Multicast

   It is possible to use RTSP to request that media be delivered to a
   multicast group.  The entity setting up the session (the controller)
   will then control when and what media is delivered to the group.
   This use case has some potential for denial of service attacks by
   flooding a multicast group.  Therefore, a mechanism is needed to
   indicate that the group actually accepts the traffic from the RTSP
   server.




Schulzrinne, et al.      Expires August 14, 2014              [Page 294]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   An open issue in this use case is how one ensures that all receivers
   listening to the multicast or broadcast receives the session
   presentation configuring the receivers.  This specification has to
   rely on an external solution to solve this issue.

E.4.  Inviting an RTSP server into a conference

   If one has an established conference or group session, it is possible
   to have an RTSP server distribute media to the whole group.
   Transmission to the group is simplest when controlled by a single
   participant or leader of the conference.  Shared control might be
   possible, but would require further investigation and possibly
   extensions.

   This use case assumes that there exists either multicast or a
   conference focus that redistribute media to all participants.

   This use case is intended to be able to handle the following
   scenario: A conference leader or participant (hereafter called the
   controller) has some pre-stored content on an RTSP server that he
   wants to share with the group.  The controller sets up an RTSP
   session at the streaming server for this content and retrieves the
   session description for the content.  The destination for the media
   content is set to the shared multicast group or conference focus.
   When desired by the controller, he/she can start and stop the
   transmission of the media to the conference group.

   There are several issues with this use case that are not solved by
   this core specification for RTSP:

   Denial of service:  To avoid an RTSP server from being an unknowing
         participant in a denial of service attack the server needs to
         be able to verify the destination's acceptance of the media.
         Such a mechanism to verify the approval of received media does
         not yet exist; instead, only policies can be used, which can be
         made to work in controlled environments.

   Distributing the presentation description to all participants in the
   group:
            To enable a media receiver to correctly decode the content
            the media configuration information needs to be distributed
            reliably to all participants.  This will most likely require
            support from an external protocol.

      Passing control of the session:  If it is desired to pass control
            of the RTSP session between the participants, some support
            will be required by an external protocol to exchange state




Schulzrinne, et al.      Expires August 14, 2014              [Page 295]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


            information and possibly floor control of who is controlling
            the RTSP session.

E.5.  Live Content using Multicast

   This use case in its simplest form does not require any use of RTSP
   at all; this is what multicast conferences being announced with SAP
   [RFC2974] and SDP are intended to handle.  However, in use cases
   where more advanced features like access control to the multicast
   session are desired, RTSP could be used for session establishment.

   A client desiring to join a live multicasted media session with
   cryptographic (encryption) access control could use RTSP in the
   following way.  The source of the session announces the session and
   gives all interested an RTSP URI.  The client connects to the server
   and requests the presentation description, allowing configuration for
   reception of the media.  In this step it is possible for the client
   to use secured transport and any desired level of authentication; for
   example, for billing or access control.  An RTSP link also allows for
   load balancing between multiple servers.

   If these were the only goals, they could be achieved by simply using
   HTTP.  However, for cases where the sender likes to keep track of
   each individual receiver of a session, and possibly use the session
   as a side channel for distributing key-updates or other information
   on a per-receiver basis, and the full set of receivers is not known
   prior to the session start, the state establishment that RTSP
   provides can be beneficial.  In this case a client would establish an
   RTSP session for this multicast group with the RTSP server.  The RTSP
   server will not transmit any media, but instead will point to the
   multicast group.  The client and server will be able to keep the
   session alive for as long as the receiver participates in the session
   thus enabling, for example, the server to push updates to the client.

   This use case will most likely not be able to be implemented without
   some extensions to the server-to-client push mechanism.  Here the
   PLAY_NOTIFY method (see Section 13.5) with a suitable extension could
   provide clear benefits.

Appendix F.  Text format for Parameters

   A resource of type "text/parameters" consists of either 1) a list of
   parameters (for a query) or 2) a list of parameters and associated
   values (for an response or setting of the parameter).  Each entry of
   the list is a single line of text.  Parameters are separated from
   values by a colon.  The parameter name MUST only use US-ASCII visible
   characters while the values are UTF-8 text strings.  The media type
   registration form is in Section 22.16.



Schulzrinne, et al.      Expires August 14, 2014              [Page 296]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   There is a potential interoperability issue for this format.  It was
   named in RFC 2326 but never defined, even if used in examples that
   hint at the syntax.  This format matches the purpose and its syntax
   supports the examples provided.  However, it goes further by allowing
   UTF-8 in the value part, thus usage of UTF-8 strings may not be
   supported.  However, as individual parameters are not defined, the
   using application anyway needs to have out-of-band agreement or using
   feature-tag to determine if the end-point supports the parameters.

   The ABNF [RFC5234] grammar for "text/parameters" content is:

   file             = *((parameter / parameter-value) CRLF)
   parameter        = 1*visible-except-colon
   parameter-value  = parameter *WSP ":" value
   visible-except-colon = %x21-39 / %x3B-7E    ; VCHAR - ":"
   value            = *(TEXT-UTF8char / WSP)
   TEXT-UTF8char    = <as defined in Section 20.1>
   WSP              = <See RFC 5234> ; Space or HTAB
   VCHAR            = <See RFC 5234>
   CRLF             = <See RFC 5234>

Appendix G.  Requirements for Unreliable Transport of RTSP

   This appendix provides guidance for those who want to implement RTSP
   messages over unreliable transports as has been defined in RTSP 1.0
   [RFC2326].  RFC 2326 defined the "rtspu" URI scheme and provided some
   basic information for the transport of RTSP messages over UDP.  The
   information is being provided here as there has been at at least one
   commercial implementation and compatibility with that should be
   maintained.

   The following points should be considered for an interoperable
   implementation:

   o  Requests shall be acknowledged by the receiver.  If there is no
      acknowledgement, the sender may resend the same message after a
      timeout of one round-trip time (RTT).  Any retransmissions due to
      lack of acknowledgement must carry the same sequence number as the
      original request.

   o  The round-trip time can be estimated as in TCP (RFC 6298)
      [RFC6298], with an initial round-trip value of 500 ms.  An
      implementation may cache the last RTT measurement as the initial
      value for future connections.

   o  The Timestamp header (Section 18.53) is used to avoid the
      retransmission ambiguity problem [Stevens98].




Schulzrinne, et al.      Expires August 14, 2014              [Page 297]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  The registered default port for RTSP over UDP for the server is
      554.

   o  RTSP messages can be carried over any lower-layer transport
      protocol that is 8-bit clean.

   o  RTSP messages are vulnerable to bit errors and should not be
      subjected to them.

   o  Source authentication, or at least validation that RTSP messages
      comes from the same entity becomes extremely important, as session
      hijacking may be substantially easier for RTSP message transport
      using an unreliable protocol like UDP than for TCP.

   There are two RTSP headers that are primarily intended for being used
   by the unreliable handling of RTSP messages and which will be
   maintained:

   o  CSeq: See Section 18.20.  It should be noted that the CSeq header
      is also required to match requests and responses independent
      whether a reliable or unreliable transport is used.

   o  Timestamp: See Section 18.53

Appendix H.  Backwards Compatibility Considerations

   This section contains notes on issues about backwards compatibility
   with clients or servers being implemented according to RFC 2326
   [RFC2326].  Note that there exists no requirement to implement RTSP
   1.0; in fact this document recommend against it as it is difficult to
   do in an interoperable way.

   A server implementing RTSP/2.0 MUST include an RTSP-Version of RTSP/
   2.0 in all responses to requests containing RTSP-Version RTSP/2.0.
   If a server receives an RTSP/1.0 request, it MAY respond with an RTSP
   /1.0 response if it chooses to support RFC 2326.  If the server
   chooses not to support RFC 2326, it MUST respond with a 505 (RTSP
   Version not supported) status code.  A server MUST NOT respond to an
   RTSP-Version RTSP/1.0 request with an RTSP-Version RTSP/2.0 response.

   Clients implementing RTSP/2.0 MAY use an OPTIONS request with a RTSP-
   Version of 2.0 to determine whether a server supports RTSP/2.0.  If
   the server responds with either an RTSP-Version of 1.0 or a status
   code of 505 (RTSP Version not supported), the client will have to use
   RTSP/1.0 requests if it chooses to support RFC 2326.






Schulzrinne, et al.      Expires August 14, 2014              [Page 298]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


H.1.  Play Request in Play State

   The behavior in the server when a Play is received in Play state has
   changed (Section 13.4).  In RFC 2326, the new PLAY request would be
   queued until the current Play completed.  Any new PLAY request now
   takes effect immediately replacing the previous request.

H.2.  Using Persistent Connections

   Some server implementations of RFC 2326 maintain a one-to-one
   relationship between a connection and an RTSP session.  Such
   implementations require clients to use a persistent connection to
   communicate with the server and when a client closes its connection,
   the server may remove the RTSP session.  This is worth noting if a
   RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.

Appendix I.  Changes

   This appendix briefly lists the differences between RTSP 1.0
   [RFC2326] and RTSP 2.0 for an informational purpose.  For
   implementers of RTSP 2.0 it is recommended to read carefully through
   this memo and not to rely on the list of changes below to adapt from
   RTSP 1.0 to RTSP 2.0, as RTSP 2.0 is not intended to be backwards
   compatible with RTSP 1.0 [RFC2326] other than the version negotiation
   mechanism.

I.1.  Brief Overview

   The following protocol elements were removed in RTSP 2.0 compared to
   RTSP 1.0:

   o  there is no section on minimal implementation anymore, but more
      the definition of RTSP 2.0 core;

   o  the RECORD and ANNOUNCE methods and all related functionality
      (including 201 (Created) and 250 (Low On Storage Space) status
      codes);

   o  the use of UDP for RTSP message transport was removed due to
      missing interest and to broken specification;

   o  the use of PLAY method for keep-alive in Play state.

   The following protocol elements were added or changed in RTSP 2.0
   compared to RTSP 1.0:

   o  RTSP session TEARDOWN from the server to the client;




Schulzrinne, et al.      Expires August 14, 2014              [Page 299]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   o  IPv6 support;

   o  extended IANA registries (e.g., transport headers parameters,
      transport-protocol, profile, lower-transport, and mode);

   o  request pipelining for quick session start-up;

   o  fully reworked state-machine;

   o  RTSP messages now use URIs rather then URLs;

   o  incorporated much of related HTTP text ([RFC2616]) in this memo,
      compared to just referencing the sections in HTTP, to avoid
      ambiguities;

   o  the REDIRECT method was expanded and diversified for different
      situations;

   o  Includes a new section about how to setup different media
      transport alternatives and their profiles, and lower layer
      protocols.  This caused the appendix on RTP interaction to be
      moved there instead of being in the part which describes RTP.  The
      section also includes guidelines what to consider when writing
      usage guidelines for new protocols and profiles;

   o  Added an asynchronous notification method PLAY_NOTIFY.  This
      method is used by the RTSP server to asynchronously notify clients
      about session changes while in Play state.  To a limited extent
      this is comparable with some implementations of ANNOUNCE in RTSP
      1.0 not intended for Recording.

I.2.  Detailed List of Changes

   Compared to RTSP 1.0 (RFC 2326), the below changes has been made when
   defining RTSP 2.0.  Note that this list does not reflect minor
   changes in wording or correction of typographical errors.

   o  The section on minimal implementation was deleted without
      substitution.

   o  The Transport header has been changed in the following way:

      *  The ABNF has been changed to define that extensions are
         possible, and that unknown parameters result in that servers
         ignore the transport specification.






Schulzrinne, et al.      Expires August 14, 2014              [Page 300]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      *  To prevent backwards compatibility issues, any extension or new
         parameter requires the usage of a feature-tag combined with the
         Require header.

      *  Syntax unclarities with the Mode parameter have been resolved.

      *  Syntax error with ";" for multicast and unicast has been
         resolved.

      *  Two new addressing parameters have been defined, src_addr and
         dest_addr.  These replace the parameters "port", "client_port",
         "server_port", "destination", "source".

      *  Support for IPv6 explicit addresses in all address fields has
         been included.

      *  To handle URI definitions that contain ";" or "," a quoted URI
         format has been introduced and is required.

      *  Defined IANA registries for the transport headers parameters,
         transport-protocol, profile, lower-transport, and mode.

      *  The transport headers interleaved parameter's text was made
         more strict and uses formal requirements levels.  It was also
         clarified that the interleaved channels are symmetric and that
         it is the server that sets the channel numbers.

      *  It has been clarified that the client can't request of the
         server to use a certain RTP SSRC, using a request with the
         transport parameter SSRC.

      *  Syntax definition for SSRC has been clarified to require 8HEX.
         It has also been extended to allow multiple values for clients
         supporting this version.

      *  Clarified the text on the transport headers "dest_addr"
         parameters regarding what security precautions the server is
         required to perform.

   o  The Range formats has been changed in the following way:

      *  The NPT format has been given an initial NPT identifier that
         must now be used.

      *  All formats now support initial open ended formats of type
         "npt=-10" and also format only "Range: smpte" ranges for usage
         with GET_PARAMETER requests.




Schulzrinne, et al.      Expires August 14, 2014              [Page 301]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      *  The npt-hhmmss notation now follows ISO 8601 more stricter.

   o  RTSP message handling has been changed in the following way:

      *  RTSP messages now use URIs rather then URLs.

      *  It has been clarified that a 4xx message due to missing CSeq
         header shall be returned without a CSeq header.

      *  The 300 (Multiple Choices) response code has been removed.

      *  Rules for how to handle timing out RTSP messages has been
         added.

      *  Extended Pipelining rules allowing for quick session startup.

      *  Sequence numbering and proxy handling of sequence number
         defined, including case when response arrive out of order.

   o  The HTTP references have been updated to RFC 2616 and RFC 2617.
      Most of the text has been copied and then altered to fit RTSP into
      this specification.  Public, and the Content-Base header has also
      been imported from RFC 2068 so that they are defined in the RTSP
      specification.  Known effects on RTSP due to HTTP clarifications:

      *  Content-Encoding header can include encoding of type
         "identity".

   o  The state machine section has been completely rewritten.  It now
      includes more details and is also more clear about the model used.

   o  An IANA section has been included which contains a number of
      registries and their rules.  This will allow us to use IANA to
      keep track of RTSP extensions.

   o  The transport of RTSP messages has seen the following changes:

      *  The use of UDP for RTSP message transport has been deprecated
         due to missing interest and to broken specification.

      *  The rules for how TCP connections are to be handled has been
         clarified.  Now it is made clear that servers should not close
         the TCP connection unless they have been unused for significant
         time.

      *  Strong recommendations why server and clients should use
         persistent connections have also been added.




Schulzrinne, et al.      Expires August 14, 2014              [Page 302]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      *  There is now a requirement on the servers to handle non-
         persistent connections as this provides fault tolerance.

      *  Added wording on the usage of Connection:Close for RTSP.

      *  Specified usage of TLS for RTSP messages, including a scheme to
         approve a proxy's TLS connection to the next hop.

   o  The following header related changes have been made:

      *  Accept-Ranges response-header is added.  This header clarifies
         which range formats that can be used for a resource.

      *  Fixed the missing definitions for the Cache-Control header.
         Also added to the syntax definition the missing delta-seconds
         for max-stale and min-fresh parameters.

      *  Put requirement on CSeq header that the value is increased by
         one for each new RTSP request.  A Recommendation to start at 0
         has also been added.

      *  Added requirement that the Date header must be used for all
         messages with message body and the Server should always include
         it.

      *  Removed possibility of using Range header with Scale header to
         indicate when it is to be activated, since it can't work as
         defined.  Also added rule that lack of Scale header in response
         indicates lack of support for the header.  Feature-tags for
         scaled playback has been defined.

      *  The Speed header must now be responded to indicate support and
         the actual speed going to be used.  A feature-tag is defined.
         Notes on congestion control were also added.

      *  The Supported header was borrowed from SIP [RFC3261] to help
         with the feature negotiation in RTSP.

      *  Clarified that the Timestamp header can be used to resolve
         retransmission ambiguities.

      *  The Session header text has been expanded with an explanation
         on keep-alive and which methods to use.  SET_PARAMETER is now
         recommended to use if only keep-alive within RTSP is desired.

      *  It has been clarified how the Range header formats are used to
         indicate pause points in the PAUSE response.




Schulzrinne, et al.      Expires August 14, 2014              [Page 303]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      *  Clarified that RTP-Info URIs that are relative, use the
         Request-URI as base URI.  Also clarified that the used URI must
         be the one that was used in the SETUP request.  The URIs are
         now also required to be quoted.  The header also expresses the
         SSRC for the provided RTP timestamp and sequence number values.

      *  Added text that requires the Range to always be present in PLAY
         responses.  Clarified what should be sent in case of live
         streams.

      *  The headers table has been updated using a structure borrowed
         from SIP.  Those tables convey much more information and should
         provide a good overview of the available headers.

      *  It has been clarified that any message with a message body is
         required to have a Content-Length header.  This was the case in
         RFC 2326, but could be misinterpreted.

      *  ETag has changed name to MTag.

      *  To resolve functionality around MTag.  The MTag and If-None-
         Match header have been added from HTTP with necessary
         clarification in regards to RTSP operation.

      *  Imported the Public header from HTTP RFC 2068 [RFC2068] since
         it has been removed from HTTP due to lack of use.  Public is
         used quite frequently in RTSP.

      *  Clarified rules for populating the Public header so that it is
         an intersection of the capabilities of all the RTSP agents in a
         chain.

      *  Added the Media-Range header for listing the current
         availability of the media range.

      *  Added the Notify-Reason header for giving the reason when
         sending PLAY_NOTIFY requests.

      *  A new header Seek-Style has been defined to direct and inform
         how any seek operation should/have been performed.

   o  The Protocol Syntax has been changed in the following way:

      *  All ABNF definitions are updated according to the rules defined
         in RFC 5234 [RFC5234] and have been gathered in a separate
         Section 20.





Schulzrinne, et al.      Expires August 14, 2014              [Page 304]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      *  The ABNF for the User-Agent and Server headers have been
         corrected.

      *  Some definitions in the introduction regarding the RTSP session
         have been changed.

      *  The protocol has been made fully IPv6 capable.

      *  The CHAR rule has been changed to exclude NULL.

   o  The Status codes have been changed in the following way:

      *  The use of status code 303 "See Other" has been deprecated as
         it does not make sense to use in RTSP.

      *  When sending response 451 and 458 the response body should
         contain the offending parameters.

      *  Clarification on when a 3rr redirect status code can be
         received has been added.  This includes receiving 3rr as a
         result of a request within a established session.  This
         provides clarification to a previous unspecified behavior.

      *  Removed the 201 (Created) and 250 (Low On Storage Space) status
         codes as they are only relevant to recording, which is
         deprecated.

      *  Several new Status codes have been defined: 464 "Data Transport
         Not Ready Yet", 465 "Notification Reason Unknown", 470
         "Connection Authorization Required", 471 "Connection
         Credentials not accepted", 472 "Failure to establish secure
         connection".

   o  The following functionality has been deprecated from the protocol:

      *  The use of Queued Play.

      *  The use of PLAY method for keep-alive in Play state.

      *  The RECORD and ANNOUNCE methods and all related functionality.
         Some of the syntax has been removed.

      *  The possibility to use timed execution of methods with the time
         parameter in the Range header.

      *  The description on how rtspu works is not part of the core
         specification and will require external description.  Only that




Schulzrinne, et al.      Expires August 14, 2014              [Page 305]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


         it exists is defined here and some requirements for the
         transport is provided.

   o  The following changes have been made in relation to methods:

      *  The OPTIONS method has been clarified with regards to the use
         of the Public and Allow headers.

      *  Added text clarifying the usage of SET_PARAMETER for keep-alive
         and usage without any body.

      *  PLAY method is now allowed to be pipelined with the pipelining
         of one or more SETUP requests following the initial that
         generates the session for aggregated control.

      *  REDIRECT has been expanded and diversified for different
         situations.

      *  Added a new method PLAY_NOTIFY.  This method is used by the
         RTSP server to asynchronously notify clients about session
         changes.

   o  Wrote a new section about how to setup different media transport
      alternatives and their profiles, and lower layer protocols.  This
      caused the appendix on RTP interaction to be moved there instead
      of being in the part which describes RTP.  The section also
      includes guidelines what to consider when writing usage guidelines
      for new protocols and profiles.

   o  Setup and usage of independent TCP connections for transport of
      RTP has been specified.

   o  Added a new section describing the available mechanisms to
      determine if functionality is supported, called "Capability
      Handling".  Renamed option-tags to feature-tags.

   o  Added a contributors section with people who have contributed
      actual text to the specification.

   o  Added a section Use Cases that describes the major use cases for
      RTSP.

   o  Clarified the usage of a=range and how to indicate live content
      that are not seekable with this header.

   o  Text specifying the special behavior of PLAY for live content.

   o  Security features of RTSP have been clarified:



Schulzrinne, et al.      Expires August 14, 2014              [Page 306]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


      *  HTTP based authorization has been clarified requring both Basic
         and DIGEST support

      *  TLS support mandated

      *  IF one implements RTP then SRTP and defined MIKEY based key-
         exchange must be supported

      *  Various minor mitigations discussed or resulted in protocol
         changes.

Appendix J.  Acknowledgements

   This memorandum defines RTSP version 2.0 which is a revision of the
   Proposed Standard RTSP version 1.0 which is defined in [RFC2326].
   The authors of RFC 2326 are Henning Schulzrinne, Anup Rao, and Robert
   Lanphier.

   Both RTSP version 1.0 and RTSP version 2.0 borrow format and
   descriptions from HTTP/1.1.

   Robert Sparks and especially Elwyn Davies provided very valuable and
   detailed reviews in the IETF last call that greately improved the
   document and resolved many issues, especially regarding consistency.

   This document has benefited greatly from the comments of all those
   participating in the MMUSIC-WG.  In addition to those already
   mentioned, the following individuals have contributed to this
   specification:

   Rahul Agarwal, Claudio Allocchio, Jeff Ayars, Milko Boic, Torsten
   Braun, Brent Browning, Bruce Butterfield, Steve Casner, Maureen
   Chesire, Jinhang Choi, Francisco Cortes, Elwyn Davies, Spencer
   Dawkins, Kelly Djahandari, Martin Dunsmuir, Adrian Farrel, Stephen
   Farrell, Ross Finlayson, Eric Fleischman, Jay Geagan, Andy Grignon,
   Christian Groves, V. Guruprasad, Peter Haight, Mark Handley, Brad
   Hefta-Gaub, Volker Hilt, John K. Ho, Patrick Hoffman, Go Hori,
   Philipp Hoschka, Anne Jones, Ingemar Johansson, Jae-Hwan Kim, Anders
   Klemets, Ruth Lang, Barry Leiba, Stephanie Leif, Jonathan Lennox,
   Eduardo F. Llach, Chris Lonvick, Xavier Marjou, Thomas Marshall, Rob
   McCool, Martti Mela, David Oran, Joerg Ott, Joe Pallas, Maria
   Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Colin Perkins,
   Pekka Pessi, Igor Plotnikov, Pete Resnick, Peter Saint-Andre, Holger
   Schmidt, Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff
   Smith, Alexander Sokolsky, Dale Stammen, John Francis Stracke, Geetha
   Srikantan, Scott Taylor, David Walker, Stephan Wenger, Dale R.
   Worley, and Byungjo Yoon , and especially to Flemming Andreasen.




Schulzrinne, et al.      Expires August 14, 2014              [Page 307]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


J.1.  Contributors

   The following people have made written contributions that were
   included in the specification:

   o  Tom Marshall contributed text on the usage of 3rr status codes.

   o  Thomas Zheng contributed text on the usage of the Range in PLAY
      responses and proposed an earlier version of the PLAY_NOTIFY
      method.

   o  Sean Sheedy contributed text on the timeout behavior of RTSP
      messages and connections, the 463 status code, and proposed an
      earlier version of the PLAY_NOTIFY method.

   o  Greg Sherwood proposed an earlier version of the PLAY_NOTIFY
      method.

   o  Fredrik Lindholm contributed text about the RTSP security
      framework.

   o  John Lazzaro contributed the text for RTP over Independent TCP.

   o  Aravind Narasimhan contributed by rewriting Media Transport
      Alternatives (Appendix C) and editorial improvements on a number
      of places in the specification.

   o  Torbjorn Einarsson has done some editorial improvements of the
      text.

Appendix K.  RFC Editor Consideration

   Please replace RFC XXXX with the RFC number this specification
   receives.

Authors' Addresses

   Henning Schulzrinne
   Columbia University
   1214 Amsterdam Avenue
   New York, NY  10027
   USA

   Email: schulzrinne@cs.columbia.edu







Schulzrinne, et al.      Expires August 14, 2014              [Page 308]

Internet-Draft   Real Time Streaming Protocol 2.0 (RTSP)   February 2014


   Anup Rao
   Cisco
   USA

   Email: anrao@cisco.com


   Rob Lanphier
   Seattle, WA
   USA

   Email: robla@robla.net


   Magnus Westerlund
   Ericsson AB
   Faeroegatan 6
   STOCKHOLM  SE-164 80
   SWEDEN

   Email: magnus.westerlund@ericsson.com


   Martin Stiemerling
   NEC Laboratories Europe, NEC Europe Ltd.
   Kurfuersten-Anlage 36
   Heidelberg  69115
   Germany

   Phone: +49 (0) 6221 4342 113
   Email: mls.ietf@gmail.com
   URI:   http://www.stiemerling.org



















Schulzrinne, et al.      Expires August 14, 2014              [Page 309]
