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Versions: (RFC 2326) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 34 35 36 37 38 39 40 RFC 7826

MMUSIC Working Group                                      H. Schulzrinne
Internet-Draft                                       Columbia University
Intended status: Standards Track                                  A. Rao
Expires: May 7, 2009                                               Cisco
                                                             R. Lanphier

                                                           M. Westerlund
                                                             Ericsson AB
                                                    M. Stiemerling (Ed.)
                                                                     NEC
                                                        November 3, 2008


                Real Time Streaming Protocol 2.0 (RTSP)
                    draft-ietf-mmusic-rfc2326bis-19

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
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   This Internet-Draft will expire on May 7, 2009.











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Abstract

   This memorandum defines RTSP version 2.0 which is a revision of the
   Proposed Standard RTSP version 1.0 which is defined in RFC 2326.

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for control over 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).


Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   9
     1.1.   Scope and Background . . . . . . . . . . . . . . . . . .   9
     1.2.   RTSP Specificication Update  . . . . . . . . . . . . . .  10
     1.3.   Notational Conventions . . . . . . . . . . . . . . . . .  10
     1.4.   Terminology  . . . . . . . . . . . . . . . . . . . . . .  11
     1.5.   Media Properties . . . . . . . . . . . . . . . . . . . .  14
       1.5.1.   Random Access  . . . . . . . . . . . . . . . . . . .  15
       1.5.2.   Retention  . . . . . . . . . . . . . . . . . . . . .  15
       1.5.3.   Content Modifications  . . . . . . . . . . . . . . .  16
       1.5.4.   Mapping to the Attributes  . . . . . . . . . . . . .  16
   2.  RTSP Introduction . . . . . . . . . . . . . . . . . . . . . .  17
     2.1.   Protocol Properties  . . . . . . . . . . . . . . . . . .  17
     2.2.   RTSP's Relationship to HTTP  . . . . . . . . . . . . . .  18
     2.3.   Extending RTSP . . . . . . . . . . . . . . . . . . . . .  19
     2.4.   Overall Operation  . . . . . . . . . . . . . . . . . . .  20
     2.5.   RTSP States  . . . . . . . . . . . . . . . . . . . . . .  21
     2.6.   Relationship with Other Protocols  . . . . . . . . . . .  22
   3.  RTSP Use Cases  . . . . . . . . . . . . . . . . . . . . . . .  23
     3.1.   On-demand Playback of Stored Content . . . . . . . . . .  23
     3.2.   Unicast distribution of Live Content . . . . . . . . . .  24
     3.3.   On-demand Playback using Multicast . . . . . . . . . . .  25
     3.4.   Inviting an RTSP server into a conference  . . . . . . .  25
     3.5.   Live Content using Multicast . . . . . . . . . . . . . .  26
   4.  Protocol Parameters . . . . . . . . . . . . . . . . . . . . .  28
     4.1.   RTSP Version . . . . . . . . . . . . . . . . . . . . . .  28
     4.2.   RTSP IRI and URI . . . . . . . . . . . . . . . . . . . .  28
     4.3.   Session Identifiers  . . . . . . . . . . . . . . . . . .  30
     4.4.   SMPTE Relative Timestamps  . . . . . . . . . . . . . . .  30
     4.5.   Normal Play Time . . . . . . . . . . . . . . . . . . . .  30
     4.6.   Absolute Time  . . . . . . . . . . . . . . . . . . . . .  31



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     4.7.   Feature-tags . . . . . . . . . . . . . . . . . . . . . .  31
     4.8.   Entity Tags  . . . . . . . . . . . . . . . . . . . . . .  32
   5.  RTSP Message  . . . . . . . . . . . . . . . . . . . . . . . .  33
     5.1.   Message Types  . . . . . . . . . . . . . . . . . . . . .  33
     5.2.   Message Headers  . . . . . . . . . . . . . . . . . . . .  34
     5.3.   Message Body . . . . . . . . . . . . . . . . . . . . . .  34
     5.4.   Message Length . . . . . . . . . . . . . . . . . . . . .  34
   6.  General Header Fields . . . . . . . . . . . . . . . . . . . .  35
   7.  Request . . . . . . . . . . . . . . . . . . . . . . . . . . .  36
     7.1.   Request Line . . . . . . . . . . . . . . . . . . . . . .  36
     7.2.   Request Header Fields  . . . . . . . . . . . . . . . . .  38
   8.  Response  . . . . . . . . . . . . . . . . . . . . . . . . . .  40
     8.1.   Status-Line  . . . . . . . . . . . . . . . . . . . . . .  40
       8.1.1.   Status Code and Reason Phrase  . . . . . . . . . . .  40
     8.2.   Response Header Fields . . . . . . . . . . . . . . . . .  43
   9.  Entity  . . . . . . . . . . . . . . . . . . . . . . . . . . .  46
     9.1.   Entity Header Fields . . . . . . . . . . . . . . . . . .  46
     9.2.   Entity Body  . . . . . . . . . . . . . . . . . . . . . .  47
   10. Connections . . . . . . . . . . . . . . . . . . . . . . . . .  48
     10.1.  Reliability and Acknowledgements . . . . . . . . . . . .  48
     10.2.  Using Connections  . . . . . . . . . . . . . . . . . . .  49
     10.3.  Closing Connections  . . . . . . . . . . . . . . . . . .  50
     10.4.  Timing Out Connections and RTSP Messages . . . . . . . .  51
     10.5.  Showing Liveness . . . . . . . . . . . . . . . . . . . .  51
     10.6.  Use of IPv6  . . . . . . . . . . . . . . . . . . . . . .  52
   11. Capability Handling . . . . . . . . . . . . . . . . . . . . .  53
   12. Pipelining Support  . . . . . . . . . . . . . . . . . . . . .  55
   13. Method Definitions  . . . . . . . . . . . . . . . . . . . . .  56
     13.1.  OPTIONS  . . . . . . . . . . . . . . . . . . . . . . . .  57
     13.2.  DESCRIBE . . . . . . . . . . . . . . . . . . . . . . . .  58
     13.3.  SETUP  . . . . . . . . . . . . . . . . . . . . . . . . .  60
       13.3.1.  Changing Transport Parameters  . . . . . . . . . . .  63
     13.4.  PLAY . . . . . . . . . . . . . . . . . . . . . . . . . .  64
       13.4.1.  General Usage  . . . . . . . . . . . . . . . . . . .  64
       13.4.2.  Aggregated Sessions  . . . . . . . . . . . . . . . .  68
       13.4.3.  Updating current PLAY Requests . . . . . . . . . . .  68
       13.4.4.  Playing On-Demand Media  . . . . . . . . . . . . . .  70
       13.4.5.  Playing Dynamic On-Demand Media  . . . . . . . . . .  71
       13.4.6.  Playing Live Media . . . . . . . . . . . . . . . . .  71
       13.4.7.  Playing Live with Recording  . . . . . . . . . . . .  72
       13.4.8.  Playing Live with Time-Shift . . . . . . . . . . . .  72
     13.5.  PLAY_NOTIFY  . . . . . . . . . . . . . . . . . . . . . .  73
       13.5.1.  End-of-Stream  . . . . . . . . . . . . . . . . . . .  74
       13.5.2.  Media-Properties-Update  . . . . . . . . . . . . . .  75
       13.5.3.  Scale-Change . . . . . . . . . . . . . . . . . . . .  76
     13.6.  PAUSE  . . . . . . . . . . . . . . . . . . . . . . . . .  77
     13.7.  TEARDOWN . . . . . . . . . . . . . . . . . . . . . . . .  79
     13.8.  GET_PARAMETER  . . . . . . . . . . . . . . . . . . . . .  80



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     13.9.  SET_PARAMETER  . . . . . . . . . . . . . . . . . . . . .  81
     13.10. REDIRECT . . . . . . . . . . . . . . . . . . . . . . . .  82
   14. Embedded (Interleaved) Binary Data  . . . . . . . . . . . . .  86
   15. Status Code Definitions . . . . . . . . . . . . . . . . . . .  88
     15.1.  Success 1xx  . . . . . . . . . . . . . . . . . . . . . .  88
       15.1.1.  100 Continue . . . . . . . . . . . . . . . . . . . .  88
     15.2.  Success 2xx  . . . . . . . . . . . . . . . . . . . . . .  88
       15.2.1.  200 OK . . . . . . . . . . . . . . . . . . . . . . .  88
     15.3.  Redirection 3xx  . . . . . . . . . . . . . . . . . . . .  88
       15.3.1.  300 Multiple Choices . . . . . . . . . . . . . . . .  89
       15.3.2.  301 Moved Permanently  . . . . . . . . . . . . . . .  89
       15.3.3.  302 Found  . . . . . . . . . . . . . . . . . . . . .  89
       15.3.4.  303 See Other  . . . . . . . . . . . . . . . . . . .  89
       15.3.5.  304 Not Modified . . . . . . . . . . . . . . . . . .  89
       15.3.6.  305 Use Proxy  . . . . . . . . . . . . . . . . . . .  90
     15.4.  Client Error 4xx . . . . . . . . . . . . . . . . . . . .  90
       15.4.1.  400 Bad Request  . . . . . . . . . . . . . . . . . .  90
       15.4.2.  405 Method Not Allowed . . . . . . . . . . . . . . .  90
       15.4.3.  451 Parameter Not Understood . . . . . . . . . . . .  90
       15.4.4.  452 reserved . . . . . . . . . . . . . . . . . . . .  90
       15.4.5.  453 Not Enough Bandwidth . . . . . . . . . . . . . .  91
       15.4.6.  454 Session Not Found  . . . . . . . . . . . . . . .  91
       15.4.7.  455 Method Not Valid in This State . . . . . . . . .  91
       15.4.8.  456 Header Field Not Valid for Resource  . . . . . .  91
       15.4.9.  457 Invalid Range  . . . . . . . . . . . . . . . . .  91
       15.4.10. 458 Parameter Is Read-Only . . . . . . . . . . . . .  91
       15.4.11. 459 Aggregate Operation Not Allowed  . . . . . . . .  91
       15.4.12. 460 Only Aggregate Operation Allowed . . . . . . . .  91
       15.4.13. 461 Unsupported Transport  . . . . . . . . . . . . .  92
       15.4.14. 462 Destination Unreachable  . . . . . . . . . . . .  92
       15.4.15. 463 Destination Prohibited . . . . . . . . . . . . .  92
       15.4.16. 464 Data Transport Not Ready Yet . . . . . . . . . .  92
       15.4.17. 465 Notification Reason Unknown  . . . . . . . . . .  92
       15.4.18. 470 Connection Authorization Required  . . . . . . .  92
       15.4.19. 471 Connection Credentials not accepted  . . . . . .  93
       15.4.20. 472 Failure to establish secure connection . . . . .  93
     15.5.  Server Error 5xx . . . . . . . . . . . . . . . . . . . .  93
       15.5.1.  551 Option not supported . . . . . . . . . . . . . .  93
   16. Header Field Definitions  . . . . . . . . . . . . . . . . . .  94
     16.1.  Accept . . . . . . . . . . . . . . . . . . . . . . . . . 103
     16.2.  Accept-Credentials . . . . . . . . . . . . . . . . . . . 103
     16.3.  Accept-Encoding  . . . . . . . . . . . . . . . . . . . . 104
     16.4.  Accept-Language  . . . . . . . . . . . . . . . . . . . . 104
     16.5.  Accept-Ranges  . . . . . . . . . . . . . . . . . . . . . 104
     16.6.  Allow  . . . . . . . . . . . . . . . . . . . . . . . . . 105
     16.7.  Authorization  . . . . . . . . . . . . . . . . . . . . . 105
     16.8.  Bandwidth  . . . . . . . . . . . . . . . . . . . . . . . 105
     16.9.  Blocksize  . . . . . . . . . . . . . . . . . . . . . . . 105



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     16.10. Cache-Control  . . . . . . . . . . . . . . . . . . . . . 106
     16.11. Connection . . . . . . . . . . . . . . . . . . . . . . . 108
     16.12. Connection-Credentials . . . . . . . . . . . . . . . . . 108
     16.13. Content-Base . . . . . . . . . . . . . . . . . . . . . . 109
     16.14. Content-Encoding . . . . . . . . . . . . . . . . . . . . 109
     16.15. Content-Language . . . . . . . . . . . . . . . . . . . . 109
     16.16. Content-Length . . . . . . . . . . . . . . . . . . . . . 110
     16.17. Content-Location . . . . . . . . . . . . . . . . . . . . 110
     16.18. Content-Type . . . . . . . . . . . . . . . . . . . . . . 110
     16.19. CSeq . . . . . . . . . . . . . . . . . . . . . . . . . . 110
     16.20. Date . . . . . . . . . . . . . . . . . . . . . . . . . . 110
     16.21. ETag . . . . . . . . . . . . . . . . . . . . . . . . . . 111
     16.22. Expires  . . . . . . . . . . . . . . . . . . . . . . . . 111
     16.23. From . . . . . . . . . . . . . . . . . . . . . . . . . . 112
     16.24. If-Match . . . . . . . . . . . . . . . . . . . . . . . . 112
     16.25. If-Modified-Since  . . . . . . . . . . . . . . . . . . . 112
     16.26. If-None-Match  . . . . . . . . . . . . . . . . . . . . . 113
     16.27. Last-Modified  . . . . . . . . . . . . . . . . . . . . . 113
     16.28. Location . . . . . . . . . . . . . . . . . . . . . . . . 113
     16.29. Media-Properties . . . . . . . . . . . . . . . . . . . . 113
     16.30. Media-Range  . . . . . . . . . . . . . . . . . . . . . . 115
     16.31. Notify-Reason  . . . . . . . . . . . . . . . . . . . . . 115
     16.32. Pipelined-Requests . . . . . . . . . . . . . . . . . . . 115
     16.33. Proxy-Authenticate . . . . . . . . . . . . . . . . . . . 116
     16.34. Proxy-Authorization  . . . . . . . . . . . . . . . . . . 117
     16.35. Proxy-Require  . . . . . . . . . . . . . . . . . . . . . 117
     16.36. Proxy-Supported  . . . . . . . . . . . . . . . . . . . . 117
     16.37. Public . . . . . . . . . . . . . . . . . . . . . . . . . 118
     16.38. Range  . . . . . . . . . . . . . . . . . . . . . . . . . 119
     16.39. Referer  . . . . . . . . . . . . . . . . . . . . . . . . 120
     16.40. Retry-After  . . . . . . . . . . . . . . . . . . . . . . 120
     16.41. Request-Status . . . . . . . . . . . . . . . . . . . . . 120
     16.42. Require  . . . . . . . . . . . . . . . . . . . . . . . . 121
     16.43. RTP-Info . . . . . . . . . . . . . . . . . . . . . . . . 122
     16.44. Scale  . . . . . . . . . . . . . . . . . . . . . . . . . 123
     16.45. Seek-Style . . . . . . . . . . . . . . . . . . . . . . . 124
     16.46. Speed  . . . . . . . . . . . . . . . . . . . . . . . . . 125
     16.47. Server . . . . . . . . . . . . . . . . . . . . . . . . . 126
     16.48. Session  . . . . . . . . . . . . . . . . . . . . . . . . 126
     16.49. Supported  . . . . . . . . . . . . . . . . . . . . . . . 127
     16.50. Timestamp  . . . . . . . . . . . . . . . . . . . . . . . 127
     16.51. Transport  . . . . . . . . . . . . . . . . . . . . . . . 128
     16.52. Unsupported  . . . . . . . . . . . . . . . . . . . . . . 134
     16.53. User-Agent . . . . . . . . . . . . . . . . . . . . . . . 134
     16.54. Vary . . . . . . . . . . . . . . . . . . . . . . . . . . 134
     16.55. Via  . . . . . . . . . . . . . . . . . . . . . . . . . . 135
     16.56. WWW-Authenticate . . . . . . . . . . . . . . . . . . . . 135
   17. Proxies . . . . . . . . . . . . . . . . . . . . . . . . . . . 136



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     17.1.  Proxies and Protocol Extensions  . . . . . . . . . . . . 137
   18. Caching . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
   19. Security Framework  . . . . . . . . . . . . . . . . . . . . . 140
     19.1.  RTSP and HTTP Authentication . . . . . . . . . . . . . . 140
     19.2.  RTSP over TLS  . . . . . . . . . . . . . . . . . . . . . 140
     19.3.  Security and Proxies . . . . . . . . . . . . . . . . . . 141
       19.3.1.  Accept-Credentials . . . . . . . . . . . . . . . . . 142
       19.3.2.  User approved TLS procedure  . . . . . . . . . . . . 143
   20. Syntax  . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
     20.1.  Base Syntax  . . . . . . . . . . . . . . . . . . . . . . 145
     20.2.  RTSP Protocol Definition . . . . . . . . . . . . . . . . 147
       20.2.1.  Generic Protocol elements  . . . . . . . . . . . . . 147
       20.2.2.  Message Syntax . . . . . . . . . . . . . . . . . . . 150
       20.2.3.  Header Syntax  . . . . . . . . . . . . . . . . . . . 154
     20.3.  SDP extension Syntax . . . . . . . . . . . . . . . . . . 162
   21. Security Considerations . . . . . . . . . . . . . . . . . . . 163
     21.1.  Remote denial of Service Attack  . . . . . . . . . . . . 165
   22. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 167
     22.1.  Feature-tags . . . . . . . . . . . . . . . . . . . . . . 167
       22.1.1.  Description  . . . . . . . . . . . . . . . . . . . . 167
       22.1.2.  Registering New Feature-tags with IANA . . . . . . . 168
       22.1.3.  Registered entries . . . . . . . . . . . . . . . . . 168
     22.2.  RTSP Methods . . . . . . . . . . . . . . . . . . . . . . 168
       22.2.1.  Description  . . . . . . . . . . . . . . . . . . . . 168
       22.2.2.  Registering New Methods with IANA  . . . . . . . . . 168
       22.2.3.  Registered Entries . . . . . . . . . . . . . . . . . 169
     22.3.  RTSP Status Codes  . . . . . . . . . . . . . . . . . . . 169
       22.3.1.  Description  . . . . . . . . . . . . . . . . . . . . 169
       22.3.2.  Registering New Status Codes with IANA . . . . . . . 169
       22.3.3.  Registered Entries . . . . . . . . . . . . . . . . . 169
     22.4.  RTSP Headers . . . . . . . . . . . . . . . . . . . . . . 169
       22.4.1.  Description  . . . . . . . . . . . . . . . . . . . . 169
       22.4.2.  Registering New Headers with IANA  . . . . . . . . . 170
       22.4.3.  Registered entries . . . . . . . . . . . . . . . . . 170
     22.5.  Accept-Credentials . . . . . . . . . . . . . . . . . . . 171
       22.5.1.  Accept-Credentials policies  . . . . . . . . . . . . 171
       22.5.2.  Accept-Credentials hash algorithms . . . . . . . . . 171
     22.6.  Cache-Control  Cache Directive Extensions  . . . . . . . 172
     22.7.  Media Properties . . . . . . . . . . . . . . . . . . . . 172
       22.7.1.  Description  . . . . . . . . . . . . . . . . . . . . 173
       22.7.2.  Registration Rules . . . . . . . . . . . . . . . . . 173
       22.7.3.  Registered Values  . . . . . . . . . . . . . . . . . 173
     22.8.  Notify-Reason header . . . . . . . . . . . . . . . . . . 173
       22.8.1.  Description  . . . . . . . . . . . . . . . . . . . . 173
       22.8.2.  Registration Rules . . . . . . . . . . . . . . . . . 173
       22.8.3.  Registered Values  . . . . . . . . . . . . . . . . . 174
     22.9.  Range header formats . . . . . . . . . . . . . . . . . . 174
     22.10. RTP-Info header parameters . . . . . . . . . . . . . . . 174



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       22.10.1. Desctiption  . . . . . . . . . . . . . . . . . . . . 174
       22.10.2. Registration Rules . . . . . . . . . . . . . . . . . 175
       22.10.3. Registered Values  . . . . . . . . . . . . . . . . . 175
     22.11. Seek-Style Policies  . . . . . . . . . . . . . . . . . . 175
       22.11.1. Description  . . . . . . . . . . . . . . . . . . . . 175
       22.11.2. Registration Rules . . . . . . . . . . . . . . . . . 175
       22.11.3. Registered Values  . . . . . . . . . . . . . . . . . 176
     22.12. Transport Header Registries  . . . . . . . . . . . . . . 176
       22.12.1. Transport Protocol Specification . . . . . . . . . . 176
       22.12.2. Transport modes  . . . . . . . . . . . . . . . . . . 177
       22.12.3. Transport Parameters . . . . . . . . . . . . . . . . 178
     22.13. URI Schemes  . . . . . . . . . . . . . . . . . . . . . . 178
       22.13.1. The rtsp URI Scheme  . . . . . . . . . . . . . . . . 178
       22.13.2. The rtsps URI Scheme . . . . . . . . . . . . . . . . 179
       22.13.3. The rtspu URI Scheme . . . . . . . . . . . . . . . . 180
     22.14. SDP attributes . . . . . . . . . . . . . . . . . . . . . 181
     22.15. Media Type Registration for text/parameters  . . . . . . 182
   23. References  . . . . . . . . . . . . . . . . . . . . . . . . . 184
     23.1.  Normative References . . . . . . . . . . . . . . . . . . 184
     23.2.  Informative References . . . . . . . . . . . . . . . . . 186
   Appendix A.  Examples . . . . . . . . . . . . . . . . . . . . . . 189
     A.1.   Media on Demand (Unicast)  . . . . . . . . . . . . . . . 189
     A.2.   Media on Demand using Pipelining . . . . . . . . . . . . 193
     A.3.   Media on Demand (Unicast)  . . . . . . . . . . . . . . . 195
     A.4.   Single Stream Container Files  . . . . . . . . . . . . . 199
     A.5.   Live Media Presentation Using Multicast  . . . . . . . . 201
     A.6.   Capability Negotiation . . . . . . . . . . . . . . . . . 202
   Appendix B.  RTSP Protocol State Machine  . . . . . . . . . . . . 204
     B.1.   States . . . . . . . . . . . . . . . . . . . . . . . . . 204
     B.2.   State variables  . . . . . . . . . . . . . . . . . . . . 204
     B.3.   Abbreviations  . . . . . . . . . . . . . . . . . . . . . 204
     B.4.   State Tables . . . . . . . . . . . . . . . . . . . . . . 205
   Appendix C.  Media Transport Alternatives . . . . . . . . . . . . 210
     C.1.   RTP  . . . . . . . . . . . . . . . . . . . . . . . . . . 210
       C.1.1.   AVP  . . . . . . . . . . . . . . . . . . . . . . . . 210
       C.1.2.   AVP/UDP  . . . . . . . . . . . . . . . . . . . . . . 210
       C.1.3.   AVPF/UDP . . . . . . . . . . . . . . . . . . . . . . 211
       C.1.4.   SAVP/UDP . . . . . . . . . . . . . . . . . . . . . . 212
       C.1.5.   SAVPF/UDP  . . . . . . . . . . . . . . . . . . . . . 212
       C.1.6.   RTCP usage with RTSP . . . . . . . . . . . . . . . . 212
     C.2.   RTP over TCP . . . . . . . . . . . . . . . . . . . . . . 213
       C.2.1.   Interleaved RTP over TCP . . . . . . . . . . . . . . 213
       C.2.2.   RTP over independent TCP . . . . . . . . . . . . . . 213
       C.2.3.   Handling Media Clock Time Jumps in the RTP Media
                Layer  . . . . . . . . . . . . . . . . . . . . . . . 217
       C.2.4.   Handling RTP Timestamps after PAUSE  . . . . . . . . 220
       C.2.5.   RTSP / RTP Integration . . . . . . . . . . . . . . . 223
       C.2.6.   Scaling with RTP . . . . . . . . . . . . . . . . . . 223



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       C.2.7.   Maintaining NPT synchronization with RTP
                timestamps . . . . . . . . . . . . . . . . . . . . . 223
       C.2.8.   Continuous Audio . . . . . . . . . . . . . . . . . . 223
       C.2.9.   Multiple Sources in an RTP Session . . . . . . . . . 223
       C.2.10.  Usage of SSRCs and the RTCP BYE Message During an
                RTSP Session . . . . . . . . . . . . . . . . . . . . 223
     C.3.   Future Additions . . . . . . . . . . . . . . . . . . . . 224
   Appendix D.  Use of SDP for RTSP Session Descriptions . . . . . . 225
     D.1.   Definitions  . . . . . . . . . . . . . . . . . . . . . . 225
       D.1.1.   Control URI  . . . . . . . . . . . . . . . . . . . . 225
       D.1.2.   Media Streams  . . . . . . . . . . . . . . . . . . . 226
       D.1.3.   Payload Type(s)  . . . . . . . . . . . . . . . . . . 227
       D.1.4.   Format-Specific Parameters . . . . . . . . . . . . . 227
       D.1.5.   Directionality of media stream . . . . . . . . . . . 227
       D.1.6.   Range of Presentation  . . . . . . . . . . . . . . . 228
       D.1.7.   Time of Availability . . . . . . . . . . . . . . . . 229
       D.1.8.   Connection Information . . . . . . . . . . . . . . . 229
       D.1.9.   Entity Tag . . . . . . . . . . . . . . . . . . . . . 229
     D.2.   Aggregate Control Not Available  . . . . . . . . . . . . 230
     D.3.   Aggregate Control Available  . . . . . . . . . . . . . . 230
     D.4.   RTSP external SDP delivery . . . . . . . . . . . . . . . 231
   Appendix E.  Text format for Parameters . . . . . . . . . . . . . 233
   Appendix F.  Requirements for Unreliable Transport of RTSP  . . . 234
   Appendix G.  Backwards Compatibility Considerations . . . . . . . 236
     G.1.   Play Request in Play mode  . . . . . . . . . . . . . . . 236
     G.2.   Using Persistent Connections . . . . . . . . . . . . . . 236
   Appendix H.  Open Issues  . . . . . . . . . . . . . . . . . . . . 237
   Appendix I.  Changes  . . . . . . . . . . . . . . . . . . . . . . 238
   Appendix J.  Acknowledgements . . . . . . . . . . . . . . . . . . 245
     J.1.   Contributors . . . . . . . . . . . . . . . . . . . . . . 245
   Appendix K.  RFC Editor Consideration . . . . . . . . . . . . . . 247
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . . 248
   Intellectual Property and Copyright Statements  . . . . . . . . . 249


















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1.  Introduction

1.1.  Scope and Background

   This memo defines version 2.0 of the Real Time Streaming Protocol
   (RTSP 2.0) which is an application-level protocol for 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, as you know it from your TV set.

   The protocol operates between RTSP 2.0 clients and servers, but also
   supports the usage of RTSP 2.0 proxies between clients and servers.
   Basically, clients can request information about streaming media from
   setranrvers, by asking for a description of the media or use media
   description provided externally.  Based on the media description
   clients can request to play out the media, pause it, or stop it
   completely, as known from a regular TV remote control.  The requested
   media can consist of multiple audio and video streams that are
   delivered as a time-synchronized streams from servers to clients.

   This memorandum describes the use of RTSP over a reliable connection
   based transport level protocol, such as TCP.  For security, TLS over
   a connection oriented transport is supported.

   There is no dependency on an special RTSP connection in the protocol.
   Instead, an RTSP server maintains a session labeled by an identifier
   to associate groups of media streams and their states.  An RTSP
   session is not tied to a transport-level connection such as a TCP
   connection.  During a session, a client may open and close multiple
   reliable transport connections to the server to issue RTSP requests
   for that session.

   The set of streams to be controlled in an RTSP session is defined by
   a presentation description.  This memorandum does not define a format
   for the presentation description.  However Appendix D describes how
   SDP [RFC4566] is used for this purpose.  The streams controlled by
   RTSP may use RTP [RFC3550] for their data transport, but the
   operation of RTSP does not depend on the transport mechanism used to
   carry continuous media.  RTSP is intentionally similar in syntax and
   operation to HTTP/1.1 [RFC2616] so that extension mechanisms to HTTP
   may also be applied to RTSP.

   The RTSP 2.0 protocol supports the following operations:







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   Retrieval of media from media server:  The client can either request
         a presentation description via RTSP DESCRIBE, HTTP or some
         other method.  If the presentation is being multicast, the
         presentation description contains the multicast addresses and
         ports to be used for the continuous media.  If the presentation
         is to be sent only to the client via unicast, the client
         provides the destination.

   Invitation of a media server to a conference:  A media server can be
         "invited" to join an existing conference to play back media
         into the presentation.  This mode is useful, for example, in
         distributed teaching applications.  Several parties in the
         conference may take turns "pushing the remote control buttons".
         Note: This functionality will require RTSP external application
         level functionality.

   RTSP requests may be handled by proxies, tunnels and caches as in
   HTTP/1.1 [RFC2616].

1.2.  RTSP Specificication Update

   This memorandum specifies RTSP 2.0 which is an update of RTSP 1.0, a
   proposed standard defined in [RFC2326].  The goal of this version is
   to correct the many flaws that have been identified in RTSP 1.0 since
   its publication.  The corrections are such that backwards
   compatibility was impossible.  Thus a new version was deemed the most
   appropriate solution to get a more functional protocol.  There are no
   plans to revise RTSP 1.0.  Appendix I catalogs the changes of this
   version in relation to RTSP 1.0.

   RTSP 2.0 as specified in this memo has reduced functionality compared
   to RTSP 1.0 and aims at specifying the RTSP core, functionality and
   rules for extensions, and basic interaction with the media delivery
   protocol RTP [RFC3550].

   Any other functionality would need to be published as extension
   documents.  This specification provides rules for such extensions and
   defines registries to avoid naming collisions.

1.3.  Notational Conventions

   Since some of the definitions and syntax 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 current 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



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   in [RFC5234].

   Indented and smaller-type 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", "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.

1.4.  Terminology

   Some of the terminology has been adopted from HTTP/1.1 [RFC2616].
   Terms not listed here are defined as in HTTP/1.1.

   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.

   Conference:  A multiparty, multimedia presentation, where "multi"
      implies greater than or equal to one.

   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.

   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.





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   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
      (playback), where the relationship is less strict.

   Entity:  The information transferred as the payload of a request or
      response.  An entity consists of meta-information in the form of
      entity-header fields and content in the form of an entity-body, as
      described in Section 9.

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

   IRI:  Internationalized Resource Identifier, is the same as an URI,
      with the exception that it 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 playback.

   Media server:  The server providing playback 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 server indirection:  Redirection of a media client to a
      different media server.

   (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.



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   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 or a connectionless
      transport.

   Non-Aggregated Control:  Control of a single media stream.  This is
      only possible in RTSP sessions with a single media.

   Participant:  Member of a conference.  A participant may be a
      machine, e.g., a playback server.

   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.  If an HTTP response is meant, that is
      indicated explicitly.

   Request:  An RTSP request.  If an HTTP request is meant, that 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 server entity; it
      is created, maintained and destroyed by the 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 labelled
      with a session identifier at that time.  The session exists until
      timed out by the server or explicitly removed by a TEARDOWN
      request.  An RTSP session is a stateful entity; an RTSP server



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      maintains an explicit session state machine (see Appendix A) 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.

   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 an 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.

1.5.  Media Properties

   When RTSP handles media it is important to consider the different
   properties a media instance for playback can have.  This
   specification considers the below listed media properties in its
   protocol operations.  They are derived from the differencies 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 are 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
      within the media i.e. randomly access any range of the media
      stream to playback.

   Dynamic On-demand:  This is a variation of the on-demand case where
      external methods are used to manipulate the actual content of the
      media setup for the RTSP session.  The main example is where a
      playlist determines the content of the session.

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




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   Live with Recording:  A Live stream that is combined with a server
      side capability to store and retain the content of the live
      session for random access playback within the part of the already
      recorded content.  The actual behavior of the media stream is very
      much depending 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 be a sliding window that goes forwards while
      data is made available and content that is older than the
      limitation will be discarded.

   Considering the above usages one get the following media properties
   and their different instance values.

1.5.1.  Random Access

   Random Access, i.e. if one can request that the playback point is
   moved from one point in the media duration to another.  The following
   different values are considered:

   Random Access:  Yes the media are 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 provied to express the worst case distance between
      random access points.

   Return To Start:  Seeking is only possible to begining of the
      content.

   No seeking:  Seeking is not possible at all.

1.5.2.  Retention

   Media may have different retention policy in place that affect the
   operation on the media.  The following different media retention
   policies are envisioned and taken into consideration where
   applicable.

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

   Time Limited:  The media will at least not be removed before given
      wall clock time.  After that time it may or may not be available
      any more.




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   Duration limited:  Each indiviudal unit of the media will be retained
      for the specified duration.

1.5.3.  Content Modifications

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

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

   Dynamic:  Between explicit updates the media content will not change,
      but the content may change due to external methods or triggers,
      such as playlists.

   Time Progressing:  As times progress new content will become
      available.  If the content also is retained it will become longer
      and longer as everything between the start point and the point in
      currently being made available can be accessed.

1.5.4.  Mapping to the Attributes

   This section exemplifies how one would map the above listed usages to
   the properties and their values.

   On-demand:  Random Access: Random Access=5s, Content Modifications:
      Unmutable, Retention: unlimted or time limited.

   Dynamic On-demand:  Random Access: Random Access=3s, Content
      Modifications: Dynamic, Retention: unlimted or time limited.

   Live:  Random Access: No seeking, Content Modifications: Time
      Progressing, Retention: Duration limited=0.0s

   Live with Recording:  Random Access: Random Access=3s, Content
      Modifications: Time Progressing, Retention: Duration limited=2H















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2.  RTSP Introduction

2.1.  Protocol Properties

   RTSP has the following properties:

   Extendable:  New methods and parameters can be easily added to RTSP.

   Secure:  RTSP re-uses web security mechanisms, either at the
      transport level (TLS, [RFC5246]) or within the protocol itself.
      All HTTP authentication mechanisms such as basic ([RFC2616]) and
      digest authentication ([RFC2617]) are directly applicable.

   Transport-independent:  RTSP does not preclude the use of unreliable
      datagram protocol (UDP) ([RFC0768]) as it would be possible to
      implement application-level reliability.  The use of a
      connectionless datagram protocol such as UDP requires additional
      definition that may be provided as extensions to the core RTSP
      specification.  The reliable stream protocol TCP ([RFC0793]) and
      the secured reliable stream protocol TLS over TCP [RFC5246] are
      the currently defined transport protocols for RTSP messages.

   Media-delivery protocol independent:  The operation of RTSP does not
      depend on the transport mechanism used to carry continuous media.
      While most real-time media will use RTP as a transport protocol,
      RTSP does not preclude the use of other protocols such as MPEG-2
      [ISO.13818-1.2000].  The use of other protocols requires
      additional definition that may be provided as extensions to the
      core RTSP specification.

   Multi-server capable:  Each media stream within a presentation can
      reside on a different server.  The client automatically
      establishes several concurrent control sessions with the different
      media servers.  Media synchronization in those cases is performed
      at the transport level.

   Separation of stream control and conference initiation:  Stream
      control is divorced from inviting a media server to a conference.
      In particular, SIP [RFC3261] or H.323 [ITU.H323.1996] may be used
      to invite a server to a conference; however, the exact procedures
      are unspecified.

   Suitable for professional applications:  RTSP supports frame- level
      accuracy through SMPTE time stamps to allow remote digital
      editing.






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   Presentation description neutral:  The protocol does not impose a
      particular presentation description or metafile format and can
      convey the type of format to be used.  However, the presentation
      description is required to contain at least one RTSP URI.

   Proxy and firewall friendly:  The protocol should be readily handled
      by both application and transport-layer (SOCKS [RFC1961])
      firewalls.  A firewall may need to understand the SETUP method to
      open a "hole" for the media stream.

   HTTP-friendly:  Where sensible, RTSP reuses HTTP concepts, so that
      the existing infrastructure can be reused.  This infrastructure
      includes PICS (Platform for Internet Content Selection
      [W3C.REC-PICS-services] [W3C.REC-PICS-labels]) for associating
      labels with content.  However, RTSP does not just add methods to
      HTTP since controlling continuous media requires server state in
      most cases.

   Appropriate server control:  If a client can start a stream, it needs
      to be able to stop a stream.  Servers should not start streaming
      to clients in such a way that clients cannot stop the stream.

   Transport negotiation:  The client can negotiate the transport method
      prior to actually needing to process a continuous media stream.

2.2.  RTSP's Relationship to HTTP

   RTSP is intentionally similar in syntax and operation to HTTP/1.1
   [RFC2616] so that extension mechanisms to HTTP can in some cases also
   be applied to RTSP.  However, RTSP differs in a number of important
   aspects from HTTP:



      *  RTSP introduces a number of new methods and has a different
         protocol identifier.

      *  RTSP has the notion of a session built into the protocol.

      *  An RTSP server needs to maintain state in almost all cases, as
         opposed to the stateless nature of HTTP.

      *  Both an RTSP server and client can issue requests.

      *  Data is usually carried out-of-band by a different protocol.
         Session descriptions returned in a DESCRIBE response (see
         Section 13.2) and interleaving of RTP with RTSP over TCP are
         exceptions to this rule (see Section 14).



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      *  RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
         8859-1, consistent with HTML internationalization efforts
         [RFC2070].

      *  The Request-URI always contains the absolute URI.  Because of
         backward compatibility with a historical blunder, HTTP/1.1
         [RFC2616] carries only the absolute path in the request and
         puts the host name in a separate header field.
         This makes "virtual hosting" easier, where a single host with
         one IP address hosts several document trees.

2.3.  Extending RTSP

   Since not all media servers have the same functionality, media
   servers by necessity will support different sets of requests.  For
   example:

   o  A server may not be capable of seeking (absolute positioning) if
      it is to support live events only.

   o  Some servers may not support setting stream parameters and thus
      not support GET_PARAMETER and SET_PARAMETER.

   o  Some server may support an RTSP extension.

   It is up to the creators of presentation descriptions not to ask the
   impossible of a server.  This situation is similar in HTTP/1.1
   [RFC2616], where the methods described in [H19.5] are not likely to
   be supported across all servers.

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

   o  Existing methods can be extended with new parameters, e.g.
      headers, as long as these parameters can be safely ignored by the
      recipient.  If the client needs negative acknowledgement 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 (see Section 16.35).

   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.





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   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 standard 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 detailed explanation of
   this see Section 11.

2.4.  Overall Operation

   Each presentation and media stream is identified by an RTSP URI.  The
   overall presentation and the properties of the media the presentation
   is composed of are defined by a presentation description file, the
   format of which is outside the scope of this specification.  The
   presentation description file may be obtained by the client using
   HTTP or other means such as email and may not necessarily be stored
   on the media server.

   For the purposes of this specification, a presentation description is
   assumed to describe one or more presentations, each of which
   maintains a common time axis.  For simplicity of exposition and
   without loss of generality, it is assumed that the presentation
   description contains exactly one such presentation.  A presentation
   may contain several media streams.

   The presentation description file contains a description of the media
   streams making up the presentation, including their encodings,
   language, and other parameters that enable the client to choose the
   most appropriate combination of media.  In this presentation
   description, each media stream that is individually controllable by
   RTSP is identified by an RTSP URI, which points to the media server
   handling that particular media stream and names the stream stored on
   that server.  Several media streams can be located on different
   servers; for example, audio and video streams can be split across
   servers for load sharing.  The description also enumerates which
   transport methods the server is capable of.

   Besides the media parameters, the network destination address and
   port need to be determined.  Several modes of operation can be
   distinguished:

   Unicast:  The media is transmitted to the source of the RTSP request
      or the requested destination, with the port number chosen by the
      client.  Alternatively, the media is transmitted on the same
      reliable stream as RTSP.




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   Multicast, server chooses address:  The media server picks the
      multicast address and port.  This is the typical case for a live
      or near-media-on-demand transmission.  The address is provided by
      the session description.

   Multicast, client chooses address:  If the server is to participate
      in an existing multicast conference, the multicast address, port
      and encryption key are given by the conference description,
      established by means outside the scope of this specification, for
      example by a SIP created conference.

2.5.  RTSP States

   RTSP controls a stream which may be sent via a separate protocol,
   independent of the control channel.  For example, RTSP control may be
   transported on a TCP connection while the media data is conveyed via
   UDP.  Thus, data delivery continues even if no RTSP requests are
   received by the media server.  Also, during its lifetime a single
   media stream may be controlled by RTSP requests issued sequentially
   on different TCP connections.  Therefore, the server needs to
   maintain "session state" to be able to correlate RTSP requests with a
   stream.  The state transitions are described in Appendix A.

   Many methods in RTSP do not contribute to state.  However, the
   following play a central role in defining the allocation and usage of
   stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, and
   TEARDOWN.

   SETUP:  Causes the server to allocate resources for a stream and
      create an RTSP session.

   PLAY:  Starts data transmission on a stream allocated via SETUP.

   PAUSE:  Temporarily halts a stream without freeing server resources.

   REDIRECT:  Indicates that the session should be moved to a new server
      or location

   TEARDOWN:  Frees resources associated with the stream.  The RTSP
      session ceases to exist on the server.

   RTSP methods that contribute to state use the Session header field
   (Section 16.49) to identify the RTSP session whose state is being
   manipulated.  The server generates session identifiers in response to
   SETUP requests (Section 13.3).






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2.6.  Relationship with Other Protocols

   RTSP has some overlap in functionality with HTTP.  It also may
   interact with HTTP in that the initial contact with streaming content
   will often be made through a web page.  The current protocol
   specification aims to allow different hand-off points between a web
   server and the media server implementing RTSP.  For example, the
   presentation description can be retrieved using HTTP or RTSP, which
   reduces round trips in web-browser-based scenarios, yet also allows
   for stand alone RTSP servers and clients which do not rely on HTTP at
   all.  However, RTSP differs fundamentally from HTTP in that most data
   delivery takes place out-of-band in a different protocol.  HTTP is an
   asymmetric protocol where the client issues requests and the server
   responds.  In RTSP, both the media client and media server can issue
   requests.  RTSP requests are also stateful; they may set parameters
   and continue to control a media stream long after the request has
   been acknowledged.

   Re-using HTTP functionality has advantages in at least two areas,
   namely security and proxies.  The requirements are very similar, so
   having the ability to adopt HTTP work on caches, proxies and
   authentication is valuable.

   RTSP assumes the existence of a presentation description format that
   can express both static and temporal properties of a presentation
   containing several media streams.  Session Description Protocol (SDP)
   [RFC4566] is generally the format of choice; however, RTSP is not
   bound to it.  For data delivery, most real-time media will use RTP as
   a transport protocol.  While RTSP works well with RTP, it is not tied
   to RTP.





















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3.  RTSP Use Cases

   This section 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
   to another address than the controlling entity.

3.1.  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
         includes the number of media stream 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 consist 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



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         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.

   It is also possible to redirect the media traffic to a different
   destination than that of the entity 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).

3.2.  Unicast distribution of Live Content

   This use cases is similar to the above on-demand content case (see
   Section 3.1) 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 is actually happens "now"; i.e., very similar to
   broadcast TV.  However in this case it is assumed that there exist 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.



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   Another aspect of live content is that it often has a very limited
   time of availability, as it is only is 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 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.

3.3.  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.

   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 memo has to rely on a
   external solution to solve this issue.

3.4.  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.



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   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 information
         and possibly floor control of who is controlling the RTSP
         session.

   If there interest in this use case, further work is required on the
   necessary extensions.

3.5.  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



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   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 know
   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.




































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4.  Protocol Parameters

4.1.  RTSP Version

   HTTP specification section [H3.1] applies, with "HTTP" replaced by
   "RTSP".  This specification defines version 2.0 of RTSP.

4.2.  RTSP IRI and URI

   RTSP 2.0 defines and registers three URI schemas "rtsp", "rtsps" and
   "rtspu".  The usage of the last, "rtspu", is unspecified in RTSP 2.0,
   and is defined here to register and reserve the URI scheme that is
   defined in RTSP 1.0.  The "rtspu" scheme indicates undefined
   transport of the RTSP messages over unreliable transport (UDP).  The
   syntax of "rtsp" and "rtsps" URIs has been changed from RTSP 1.0.

   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 SHALL use the rules and transformation for IRIs
   defined in [RFC3987].  This way RTSP 2.0 URIs for request can be
   produced from an RTSP IRI.

   The RTSP IRI and URI are both syntax restricted compared to the
   generic syntax defined in [RFC3986] and RFC [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 RTSP URI and IRI is case sensitive, with the exception of those
   parts that [RFC3986] and [RFC3987] defines as case-insensitive; for
   example, the scheme and host part.

   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 URI by the
   requestor and not included in the request.  The user agent also 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 requestor's
   perspective, an opaque string and needs to be handled as such.




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   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 port number 554 SHALL be used.  For the scheme
   "rtsps", the TCP port 322 is registered and SHALL be assumed.

   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:

      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.





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4.3.  Session Identifiers

   Session identifiers are strings of any arbitrary length but with a
   minimum length of 8 characters.  A session identifier MUST be chosen
   cryptographically random (see [RFC4086]) and MUST be at least 8
   characters long (can contain a maximum of 48 bits of entropy) to make
   guessing it more difficult.  It is RECOMMENDED that it contains 128
   bits of entropy, i.e. approxamitely 22 characters from a high quality
   generator. (see Section 21.)  However, it needs to be noted that the
   session identifier does not provide any security against session
   hijacking unless it is kept confidential between client, server and
   trusted proxies.

4.4.  SMPTE Relative Timestamps

   A SMPTE relative timestamp expresses time relative to the start of
   the clip.  Relative timestamps are expressed as SMPTE time codes 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 alternative 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

4.5.  Normal Play Time

   Normal play time (NPT) indicates the stream absolute position
   relative to the beginning of the presentation, not to be confused
   with the Network Time Protocol (NTP) [RFC1305].  The timestamp
   consists of a decimal fraction.  The part left of the decimal may be
   expressed in either seconds or hours, minutes, and seconds.  The part
   right of the decimal point measures fractions of a second.




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   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 SHALL 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
   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
   (high 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 conforms to ISO 8601 [ISO.8601.2000].  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.

4.6.  Absolute Time

   Absolute time is expressed as ISO 8601 [ISO.8601.2000] timestamps,
   using UTC (GMT).  Fractions of a second may be indicated.

   Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
   UTC:

     19961108T143720.25Z

4.7.  Feature-tags

   Feature-tags are unique identifiers used to designate features in
   RTSP.  These tags are used in Require (Section 16.42), Proxy-Require
   (Section 16.35), Proxy-Supported (Section 16.36), and Unsupported
   (Section 16.52) header fields.

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

   The creator of a new RTSP feature-tag should either prefix the
   feature-tag with a reverse domain name (e.g.,



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   "com.example.mynewfeature" is an apt name for a feature whose
   inventor can be reached at "example.com"), or register the new
   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.

4.8.  Entity Tags

   Entity tags are opaque strings that are used to compare two entities
   from the same resource, for example in caches or to optimize setup
   after a redirect.  Further explanation is present in [H3.11].  For an
   explanation of how to compare entity tags see [H13.3].  Entity tags
   can be carried in the ETag header (see Section 16.21) or in SDP (see
   Appendix D.1.9).

   Entity tags are used in RTSP to make some methods conditional.  The
   methods are made conditional through the inclusion of headers, see
   Section 16.24 and Section 16.26.  Note that RTSP entity tags apply to
   the complete presentation; i.e., both the session description and the
   individual media streams.  Thus entity 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.



























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5.  RTSP Message

   RTSP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 3629 [RFC3629]).  Lines SHALL 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 tricky 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].  ISO 8859-1
   translates directly into Unicode with a high-order octet of zero.
   ISO 8859-1 characters with the most-significant bit set are
   represented as 1100001x 10xxxxxx.  (See RFC 3629 [RFC3629])

   Requests contain methods, 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 consist of requests from client to server or server to
   client and responses in the reverse direction.  Request Section 7 and
   Response Section 8 messages use the generic message format of RFC 822
   [9] for transferring entities (the payload of the message).  Both
   types of message 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 header fields, and
   possibly a message-body.

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

   In the interest of robustness, servers SHOULD ignore any empty
   line(s) received where a Request-Line is expected.  In other words,
   if the server is reading the protocol stream at the beginning of a
   message and receives a CRLF first, it should ignore the CRLF.





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5.2.  Message Headers

   See [H4.2].

   Unknown message headers SHALL be ignored by a RTSP server or client.
   An RTSP Proxy SHALL forward unknown message headers.  Message headers
   defined outside of this specification that are required to be
   interpret by the RTSP agent will need to use feature tags
   (Section 4.7) and include it in the appropriate Require
   (Section 16.42) or Proxy-Require (Section 16.35) header.

5.3.  Message Body

   See [H4.3].

   Unlike HTTP, the presence of a message-body in either a request or a
   response MUST be signaled by the inclusion of a Content-Length header
   field (see Section 16.16).

5.4.  Message Length

   When a message body is included with a message, the length of that
   body is determined by one of the following (in order of precedence):

   1.  Any response message which MUST NOT include a message body (such
       as the 1xx, 204, and 304 responses) is always terminated by the
       first empty line after the header fields, regardless of the
       entity-header fields present in the message.  (Note: An empty
       line is a line with nothing preceding the CRLF.)

   2.  If a Content-Length header field (Section 16.16) is present, its
       value in bytes represents the length of the message-body.  If
       this header field is not present, a value of zero is assumed.

   Unlike an HTTP message, an RTSP message MUST contain a Content-Length
   header field 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.








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6.  General Header Fields

   See [H4.5], except that the Pragma, Trailer, Transfer-Encoding,
   Upgrade, and Warning headers are not defined.  RTSP further defines
   the CSeq, Pipelined-Requests, Proxy-Supported and Timestamp headers.
   The general headers are listed in Table 1:

                +--------------------+--------------------+
                | Header Name        | Defined in Section |
                +--------------------+--------------------+
                | Cache-Control      | Section 16.10      |
                |                    |                    |
                | Connection         | Section 16.11      |
                |                    |                    |
                | CSeq               | Section 16.19      |
                |                    |                    |
                | Date               | Section 16.20      |
                |                    |                    |
                | Media-Properties   | Section 16.29      |
                |                    |                    |
                | Media-Range        | Section 16.30      |
                |                    |                    |
                | Pipelined-Requests | Section 16.32      |
                |                    |                    |
                | Proxy-Supported    | Section 16.36      |
                |                    |                    |
                | Seek-Style         | Section 16.45      |
                |                    |                    |
                | Supported          | Section 16.49      |
                |                    |                    |
                | Timestamp          | Section 16.50      |
                |                    |                    |
                | Via                | Section 16.55      |
                +--------------------+--------------------+

                 Table 1: The general headers used in RTSP















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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 (Section 6), request (Section 7.2), or entity
      (Section 9.1);

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

   o  Optionally a message body (entity), consisting of one or more
      lines.  The length of the message body in bytes is indicated by
      the Content-Length entity 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.




























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                  +---------------+--------------------+
                  | 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> <Request-URI> <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 (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.



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   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.  Some of these
   headers may also be used in the response to a request, as response
   headers, to modify the specifics of a response (Section 8.2).

                +--------------------+--------------------+
                | Header             | Defined in Section |
                +--------------------+--------------------+
                | Accept             | Section 16.1       |
                |                    |                    |
                | Accept-Credentials | Section 16.2       |
                |                    |                    |
                | Accept-Encoding    | Section 16.3       |
                |                    |                    |
                | Accept-Language    | Section 16.4       |
                |                    |                    |
                | Authorization      | Section 16.7       |
                |                    |                    |
                | Bandwidth          | Section 16.8       |
                |                    |                    |
                | Blocksize          | Section 16.9       |
                |                    |                    |
                | From               | Section 16.23      |
                |                    |                    |
                | If-Match           | Section 16.24      |
                |                    |                    |
                | If-Modified-Since  | Section 16.25      |
                |                    |                    |
                | If-None-Match      | Section 16.26      |
                |                    |                    |
                | Notify-Reason      | Section 16.31      |
                |                    |                    |



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                | Proxy-Require      | Section 16.35      |
                |                    |                    |
                | Range              | Section 16.38      |
                |                    |                    |
                | Referer            | Section 16.39      |
                |                    |                    |
                | Request-Status     | Section 16.41      |
                |                    |                    |
                | Require            | Section 16.42      |
                |                    |                    |
                | Scale              | Section 16.44      |
                |                    |                    |
                | Session            | Section 16.48      |
                |                    |                    |
                | Speed              | Section 16.46      |
                |                    |                    |
                | Supported          | Section 16.49      |
                |                    |                    |
                | Transport          | Section 16.51      |
                |                    |                    |
                | User-Agent         | Section 16.53      |
                +--------------------+--------------------+

                     Table 3: The RTSP request headers

   Detailed headers definition are provided in Section 16.

   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. actually happens.




















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8.  Response

   [H6] applies except that HTTP-Version is replaced by RTSP-Version.
   Also, RTSP defines additional status codes and does not define some
   of the HTTP codes.  The valid response codes and the methods they can
   be used with are listed in Table 4.

   After receiving and interpreting a request message, the recipient
   responds with an RTSP response message.

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 15.  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

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







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   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 newly defined HTTP 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 the exception that an
   unrecognized response 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 entity returned
   with the response, since that entity is likely to include human-
   readable information which will explain the unusual status.

    +------+----------------------------------------+-----------------+
    | Code | Reason                                 | Method          |
    +------+----------------------------------------+-----------------+
    | 100  | Continue                               | all             |
    |      |                                        |                 |
    |      |                                        |                 |
    | 200  | OK                                     | all             |
    |      |                                        |                 |
    |      |                                        |                 |
    | 300  | Multiple Choices                       | all             |
    |      |                                        |                 |
    | 301  | Moved Permanently                      | all             |
    |      |                                        |                 |
    | 302  | Found                                  | all             |
    |      |                                        |                 |
    | 303  | See Other                              | all             |
    |      |                                        |                 |
    | 305  | Use Proxy                              | all             |
    |      |                                        |                 |
    |      |                                        |                 |
    | 400  | Bad Request                            | all             |
    |      |                                        |                 |
    | 401  | Unauthorized                           | all             |



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    | 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             |
    |      |                                        |                 |
    | 411  | Length Required                        | all             |
    |      |                                        |                 |
    | 412  | Precondition Failed                    | DESCRIBE, SETUP |
    |      |                                        |                 |
    | 413  | Request Entity Too Large               | all             |
    |      |                                        |                 |
    | 414  | Request-URI Too Long                   | all             |
    |      |                                        |                 |
    | 415  | Unsupported Media Type                 | all             |
    |      |                                        |                 |
    | 451  | Parameter Not Understood               | SET_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     |
    |      |                                        |                 |
    | 458  | Parameter Is Read-Only                 | SET_PARAMETER   |
    |      |                                        |                 |
    | 459  | Aggregate Operation Not Allowed        | all             |
    |      |                                        |                 |
    | 460  | Only Aggregate Operation Allowed       | all             |
    |      |                                        |                 |
    | 461  | Unsupported Transport                  | all             |
    |      |                                        |                 |



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    | 462  | Destination Unreachable                | all             |
    |      |                                        |                 |
    | 463  | Destination Prohibited                 | SETUP           |
    |      |                                        |                 |
    | 464  | Data Transport Not Ready Yet           | PLAY            |
    |      |                                        |                 |
    | 465  | Notification Reason Unknown            | PLAY_NOTIFY     |
    |      |                                        |                 |
    | 470  | Connection Authorization Required      | all             |
    |      |                                        |                 |
    | 471  | Connection Credentials not accepted    | all             |
    |      |                                        |                 |
    | 472  | Failure to establish secure connection | all             |
    |      |                                        |                 |
    |      |                                        |                 |
    | 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 support                     | all             |
    +------+----------------------------------------+-----------------+

          Table 4: Status codes and their usage with RTSP methods

8.2.  Response Header Fields

   The response-header fields allow the request recipient to pass
   additional information about the response which cannot be placed in
   the Status-Line.  These header fields give 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.











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              +------------------------+--------------------+
              | Header                 | Defined in Section |
              +------------------------+--------------------+
              | Accept-Credentials     | Section 16.2       |
              |                        |                    |
              | Accept-Ranges          | Section 16.5       |
              |                        |                    |
              | Connection-Credentials | Section 16.12      |
              |                        |                    |
              | ETag                   | Section 16.21      |
              |                        |                    |
              | Location               | Section 16.28      |
              |                        |                    |
              | Proxy-Authenticate     | Section 16.33      |
              |                        |                    |
              | Public                 | Section 16.37      |
              |                        |                    |
              | Range                  | Section 16.38      |
              |                        |                    |
              | Retry-After            | Section 16.40      |
              |                        |                    |
              | RTP-Info               | Section 16.43      |
              |                        |                    |
              | Scale                  | Section 16.44      |
              |                        |                    |
              | Session                | Section 16.48      |
              |                        |                    |
              | Server                 | Section 16.47      |
              |                        |                    |
              | Speed                  | Section 16.46      |
              |                        |                    |
              | Transport              | Section 16.51      |
              |                        |                    |
              | Unsupported            | Section 16.52      |
              |                        |                    |
              | Vary                   | Section 16.54      |
              |                        |                    |
              | WWW-Authenticate       | Section 16.56      |
              +------------------------+--------------------+

                    Table 5: The RTSP response headers

   Response-header field 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.  New or experimental
   header fields MAY be given the semantics of response-header fields if
   all parties in the communication recognize them to be response-header



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   fields.  Unrecognized header fields in responses are treated as
   entity-header fields.

















































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9.  Entity

   Request and Response messages MAY transfer an entity if not otherwise
   restricted by the request method or response status code.  An entity
   consists of entity-header fields and an entity-body, although some
   responses will only include the entity-headers.

   The SET_PARAMETER and GET_PARAMETER request and response, and
   DESCRIBE response MAY have an entity.  All 4xx and 5xx responses MAY
   also have an entity.

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

9.1.  Entity Header Fields

   Entity-header fields define meta-information about the entity-body
   or, if no body is present, about the resource identified by the
   request.  The entity header fields are listed in Table 6.

                 +------------------+--------------------+
                 | Header           | Defined in Section |
                 +------------------+--------------------+
                 | Allow            | Section 16.6       |
                 |                  |                    |
                 | Content-Base     | Section 16.13      |
                 |                  |                    |
                 | Content-Encoding | Section 16.14      |
                 |                  |                    |
                 | Content-Language | Section 16.15      |
                 |                  |                    |
                 | Content-Length   | Section 16.16      |
                 |                  |                    |
                 | Content-Location | Section 16.17      |
                 |                  |                    |
                 | Content-Type     | Section 16.18      |
                 |                  |                    |
                 | Expires          | Section 16.22      |
                 |                  |                    |
                 | Last-Modified    | Section 16.27      |
                 +------------------+--------------------+

                     Table 6: The RTSP entity headers

   The extension-header mechanism allows additional entity-header fields
   to be defined without changing the protocol, but these fields cannot
   be assumed to be recognizable by the recipient.  Unrecognized header
   fields SHOULD be ignored by the recipient and forwarded by proxies.



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9.2.  Entity Body

   See [H7.2] with the addition that an RTSP message with an entity body
   MUST include the Content-Type and Content-Length headers.















































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10.  Connections

   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 a 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 SHALL 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 16.19), 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

   When 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 acknowledgement 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
      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



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      congestion.

   Lack of acknowledgement 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)
   indicates the default port that the server will listen on.

   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 MAY be used for all transactions between the
   server and client, including messages for multiple RTSP sessions.
   However a persistent connection MAY also be closed after a few
   message exchanges.  For example, a client may use a persistent
   connection for the initial SETUP and PLAY message 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 SHOULD NOT have more than one connection to the server
   at any given point.  If a client or proxy handles multiple RTSP
   sessions on the same server, it SHOULD use only one connection for
   managing those sessions.

      This saves connection resources on the server.  It also reduces
      complexity by and enabling the server to maintain less state about
      its sessions and connections.

   Unlike HTTP, 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 signalling channel the server may be forced to drop an RTSP
   session without notifying the client.  An example of such a case is



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   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.

      Without a persistent connection between the client and the server,
      the media server has no reliable way of reaching the client.
      Also, this is the only way that requests from a server to its
      client are likely to traverse firewalls.

   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).

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 16.48).  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, it should wait for a reasonable amount of
   time for the client to receive the TEARDOWN response, take
   appropriate action, 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 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 15.4.12) 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
      implementation of advanced/special types of requests or result in
      extra overhead for the client when testing for new features.  On
      the flip side, keeping connections open after sending an error
      response poses a Denial of Service security risk (Section 21).

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



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   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 (requestor) 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
   requestor.  After sending a 100 response, the receiver MUST send a
   final response indicating the success or failure of the request.

   A requestor 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 requestor SHOULD continue waiting
   for further responses.  If more than 10 seconds elapses without
   receiving any response, the requestor MAY assume that the responder
   is unresponsive and abort the connection.

   A requestor SHOULD wait longer than 10 seconds for a response if it
   is experiencing significant transport delays on its connection to the
   responder.  The requestor is capable of determining the RTT of the
   request/response cycle using the Timestamp header (Section 16.50) in
   any RTSP request.

10.5.  Showing Liveness

   The mechanisms for showing liveness of the client is, any RTSP
   request with a Session header, if RTP & RTCP is used an RTCP message,
   or through any other used media 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 be lost or when using
   reliable protocols, such as TCP, the message may take some time to
   arrive safely at the receiver.  To show liveness between RTSP request
   issued to accomplish other things, the following mechanisms can be



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   used, in descending order of preference:

   RTCP: If RTP is used for media transport RTCP SHOULD be used.  If
         RTCP is used to report transport statistics, it SHALL also work
         as keep alive.  The server can determine the client by used
         network address and port together with the fact that the client
         is reporting on the servers SSRC(s).  A downside of using RTCP
         is that it only gives statistical guarantees to reach the
         server.  However that probability is so low that it can be
         ignored in most cases.  For example, 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 have for a network with 5 % packet loss, the probability
         to fail showing liveness sign in that session within the
         timeout interval of 2.4*E-16.  In sessions with shorter timeout
         times, or much higher packet loss, or small RTCP bandwidths
         SHOULD also use any of the mechanisms below.

   SET_PARAMETER:  When using SET_PARAMETER for keep alive, no body
         SHOULD be included.  This method is the RECOMMENDED RTSP method
         to use in request only intended to perform keep-alive.

   OPTIONS:  This method does also work.  However it causes the server
         to perform more unnecessary processing and result in bigger
         responses than necessary for the task.  The reason for this is
         that the server needs to determine what capabilities that are
         associated with the media resource to correctly populate the
         Public and Allow headers.

   The timeout parameter MAY be included in a SETUP response, and SHALL
   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 below and Appendix B).  The timeout is measured in
   seconds, with a default of 60 seconds.  The length of the session
   timeout SHALL NOT be changed in a established session.

10.6.  Use of IPv6

   Explicit IPv6 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 headers.









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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 wether it is supported.  This is done by issuing a
   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
   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,
   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 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 which represents the minimal playback
   implementation.

   Feature-tags are used to determine wether 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:  The supported header is used to determine the complete
         set of functionality that both client and server have.  The
         intended usage is to determine before one needs to use a
         functionality that it is supported.  It can be used in any
         method, however OPTIONS is the most suitable one as it at the
         same time determines all methods that are implemented.  When
         sending a request the requestor declares all its capabilities
         by including all supported feature-tags.  This results in that
         the receiver learns the requestors feature support.  The
         receiver then includes its set of features in the response.




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   Proxy-Supported:  The Proxy-Supported header is used similar to the
         Supported header, but instead of giving the supported
         functionality of the client or server it provides both the
         requestor and the responder a view of what functionality the
         proxy chain between the two supports.  Proxies are required to
         add this header whenever the Supported header is present, but
         proxies may independently of the requestor add it.

   Require:  The Require header can be included in any request where the
         end-point, i.e. the client or server, is required to understand
         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 method has the same purpose and workings as
         Require except that it only applies to proxies and not the end-
         point.  Features that needs to be supported by both proxies and
         end-point needs to be included in both the Require and Proxy-
         Require header.

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
















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12.  Pipelining Support

   Pipelining is a general method to improve performance of request
   response protocols by allowing the requesting entity 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 the responding entity SHALL 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 as the sending order.  The processing of the
   request SHALL also have been finished before processing the next
   request from the same entity.  The responses MUST be sent in the
   order the requests was processed.

   RTSP 2.0 has extended support for pipelining compared to RTSP 1.0.
   The major improvement is to allow all requests to setup and initiate
   media playback to be pipelined after each other.  This is
   accomplished by the utilization of the Pipelined-Requests header (see
   Section 16.32).  This header allows a client to request that two or
   more requests is to be 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 with at least one RTT.

   If a pipelined request builds on the succesful completion of one or
   more prior requests the requestor 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 succesfully executed.  However, not 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.
















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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 SHALL NOT
   start with a $ character (decimal 24) and MUST be a token as defined
   by the ABNF [RFC5234] in the syntax chapter Section 20.  The methods
   are summarized in Table 7.

   +---------------+-----------+--------+--------------+---------------+
   | method        | direction | object | Server req.  | Client req.   |
   +---------------+-----------+--------+--------------+---------------+
   | DESCRIBE      | C -> S    | P,S    | recommended  | recommended   |
   |               |           |        |              |               |
   | GET_PARAMETER | C -> S    | P,S    | optional     | optional      |
   |               |           |        |              |               |
   |               | S -> C    |        |              |               |
   |               |           |        |              |               |
   | OPTIONS       | C -> S    | P,S    | R=Req,       | Sd=Req, R=Opt |
   |               |           |        | Sd=Opt       |               |
   |               |           |        |              |               |
   |               | S -> C    |        |              |               |
   |               |           |        |              |               |
   | 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    |        |              |               |
   |               |           |        |              |               |
   | TEARDOWN      | C -> S    | P,S    | required     | required      |
   +---------------+-----------+--------+--------------+---------------+

   Table 7: Overview of RTSP methods, their direction, and what objects
     (P: presentation, S: stream) they operate on. Legend: R=Respond,
                   Sd=Send, Opt: Optional, Req: Required








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      Note on Table 7: GET_PARAMETER is recommended, but not required.
      For example, a fully functional server can be built to deliver
      media without any parameters.  SET_PARAMETER is required however
      due to its usage for keep-alive.  PAUSE is now required due to
      that it is the only way of getting out of the state machines 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 whos main function is to log or audit and not modify
   transport or media handling in any way MAY forward RTSP messages with
   unknown methods.  Note, the proxy stil needs to perform the minimal
   required processing, like adding the Via header.

13.1.  OPTIONS

   The semantics of the RTSP OPTIONS method is equivalent to that of the
   HTTP OPTIONS method described in [H9.2].  In RTSP however, OPTIONS is
   bi-directional, in that a client can request it 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.  The client, server or proxy MAY also
   implement the converse of their required capability.

   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 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



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   responding RTSP agent.

   The OPTIONS method can be used to keep an RTSP session alive.
   However, it is not the preferred means of session keep-alive
   signalling, see Section 16.48.  An OPTIONS request intended for
   keeping alive an RTSP session MUST include the Session header with
   the associated session ID.  Such a request SHOULD also use the media
   or the aggregated control URI as the Request-URI.

   Example:

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

     S->C:  RTSP/2.0 200 OK
            CSeq: 1
            Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
            Supported: play.basic, implicit-play, gzipped-messages
            Server: PhonyServer/1.1


   Note that some of the feature-tags in Require and Proxy-Require are
   fictional 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 SHALL respond
   with a description of the requested resource and return the
   description in the entity of the response.  The DESCRIBE reply-
   response pair constitutes the media initialization phase of RTSP.

   Example:











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     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=mhandley 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=recvonly
           a=control:*
           t=2873397496 2873404696
           m=audio 3456 RTP/AVP 0
           a=control:audio
           m=video 2232 RTP/AVP 31
           a=control:video

   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 usage of 3rr
   responses are recommended.

      By forcing a DESCRIBE response to contain all media initialization
      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.

   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.)




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   o  via some form of a 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.

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

13.3.  SETUP

   The SETUP request for an 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.  SETUP can be used in all
   three states; INIT, and READY, for both purposes and in PLAY to
   change the transport parameters.  There is also a third possibile
   usage for the SETUP method which is not specified in this memo:
   adding a media to a session.  Using SETUP to add media to an existing
   session, when the session is in PLAY state, is unspecified.

   The Transport header, see Section 16.51, specifies the 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 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 discouraged due to middleboxes, such as firewalls or NATs.

   For the benefit of any intervening firewalls, a client SHALL 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
      DESCRIBE response, which has been reserved for media
      initialization.

   The client SHALL include the Accept-Ranges header in the request



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   indicating all supported unit formats in the Range header.  This
   allows the server to know which format it may use in future session
   related responses, such as PLAY response without any range in the
   request.  If the client does not support a time format necessary for
   the presentation the server SHALL 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 SHALL include the Accept-Ranges header
   (see Section 16.5) to indicate which time formats that are acceptable
   to use for this media resource.

   The SETUP response 200 OK SHALL include the Media-Properties header
   (see Section 16.29 ).  The combination of the parameters of the
   Media-Properties header indicate the nature of the content present in
   the session (see also Section 1.5).  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 SHALL include the Media-Range header (see
   Section 16.30) if the media is Time-Progressing.

   A basic example for SETUP:

     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, UTC
           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="192.0.2.241:6256"/
                      "192.0.2.241:6257"; ssrc=2A3F93ED
           Accept-Ranges: NPT
           Media-Properties: Random-Access=3.2, Time-Progressing,
                             Time-Duration=3600.0



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           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 is either RTP/AVP/
   UDP (UDP per default) to be received on client port 4588 and 4589 or
   RTP/AVP interleaved on the RTSP control channel.  The server selects
   the RTP/AVP/UDP transport and adds the ports it will send and
   received 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, 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 15.4.11).
   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.  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.  The SETUP of media streams in an aggregate which has
   not been given an aggregated control URI is unspecified.

      While the session ID sometimes has 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 16.48.  Signs of liveness for an RTSP
   session are:

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



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   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 as 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 SHALL 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).
   Reasons to support changing transport parameters, is to allow for
   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 also that fails the
   changing of transport parameters will require that the client
   performs a TEARDOWN of the affected media and then setting it up
   again.  In 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 transport protocol completely, like
   switching from Interleaved TCP mode to UDP or vise versa or change
   delivery address.

   In a SETUP response for a request to change the transport parameters
   while in Play state, the server SHALL include the Range to indicate
   from what point the new transport parameters are used.  Further, if
   RTP is used for delivery, the server SHALL also include the RTP-Info
   header to indicate from what timestamp and RTP sequence number the
   change has taken place.  If both RTP-Info and Range is included in
   the response the "rtp_time" parameter and range 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.



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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 SHALL position the
   normal play time to the beginning of the range specified in the
   received Range header and delivers stream data until the end of the
   range if given, or until a new PLAY request is received, else to the
   end of the media is reached.  To allow for precise composition
   multiple ranges MAY be specified in one PLAY Request.  The range
   values are valid if all given ranges are part of any media within the
   aggregate.  If a given range value points outside of the media, the
   response SHALL be the 457 (Invalid Range) error code.

   The below example will first play seconds 10 through 15, then,
   immediately following, seconds 20 to 25, and finally seconds 30
   through the end.

     C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0
           CSeq: 835
           Session: 12345678
           Range: npt=10-15, npt=20-25, npt=30-
           Seek-Style: RAP
           User-Agent: PhonyClient/1.2

   See the description of the PAUSE request for further examples.

   A PLAY request without a Range header is legal.  It SHALL start
   playing a stream from the beginning (npt=0-) unless the stream has
   been paused or is currently playing.  If a stream has been paused via
   PAUSE, stream delivery resumes at the pause point.  If a stream is
   currently playing, the new PLAY begins at the current stream
   position.  The stream SHALL play until the end of the media.  The
   Range header MUST NOT contain a time parameter.  The usage of time in
   PLAY method has been deprecated.  If a request with time parameter is
   received the server SHOULD respond with a 457 (Invalid Range) to
   indicate that the time parameter is not supported.  If no range is
   specified in the request, the start position SHALL still be returned
   in the reply.  If the medias that are part of an aggregate has



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   different lengths, the PLAY request SHALL be performed as long as the
   given range is valid for any media, for example the longest media.
   Media will be sent whenever it is available for the given play-out
   point.

   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 when
   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 in a Range header SHALL be included.

   All range specifiers in this specification allow for ranges with
   unspecified begin times (e.g. "npt=-30").  When used in a PLAY
   request, the server treats this as a request to start/resume playback
   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 SHALL be given.

   Server MUST include a "Range" header in any PLAY response.  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 respone.  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.  For time-progressing
   content where the pause-point moves with real-time due to limited
   retention, the current resume point is returned.  For sessions in
   playing state, the current playout point and the remaining parts of
   the range request is returned.

   A PLAY response MAY include a header(s) carrying synchronization
   information.  As the information necessary is dependent on the media
   transport format, further rules specifying the header and its usage
   is needed.  For RTP the RTP-Info header is specified, see
   Section 16.43, 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 (equal to
   the requested in this case) and the RTP-Info header that contains the
   necessary parameters for the RTP stack.




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   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=3.52-324.39
         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

   For media with random-acces properties, the server MUST reply with
   the actual range that will be played back, i.e. for which duration
   any media (having content at this time) is 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 format 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 client can express its desired handling by the server by
   including the Seek-Style header (Section 16.45) in the PLAY request,
   if desired.

   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 if 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 to not render that time period.
















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   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-
         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 transition
   to the READY state, media delivery simply stops.  A PAUSE request
   MUST be issued before the stream enters the READY state.  A PLAY
   request while the stream is still in the PLAYING state is legal, and
   can be issued without an intervening PAUSE request.  Such a request
   SHALL replace the current PLAY action with the new one requested,
   i.e. being handle the same as the request was received in ready
   state.  In the case the first time range in Range header has a open
   start time (-endtime), the server SHALL continue to play from where
   it currently was until the specified end point.  This is useful to
   change ongoing playback to play another sequence, or end 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 to fit the page.

   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
         Server: PhonyServer 1.0
         Range: smpte=0:10:22-0:15:45
         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



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   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
         Server:PhonyServer 1.0
         Range: clock=19961108T142300Z-19961108T143520Z
         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 SHALL responde with error 460 (Only Aggregate
   Operation Allowed) if the client PLAY Request-URI is for one of the
   media.  The media in an aggregate SHALL be played in sync.  If a
   client wants individual control of the media it needs to use separate
   RTSP sessions for each media.

   For aggregated sessions where the initial SETUP request (creating a
   session) is followed by one or more additional SETUP request, 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 an 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 may not be the intended one due to failure of any
   of the prior requests.  However a client 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 request while the stream is In PLAYING state
   and thus updating their request.  The possibility to replace a
   current PLAY request with a new one replaces the following RTSP 1.0
   functions based on PLAY in play state:



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   o  The queued play functionality described in RFC 2326 [RFC2326] is
      removed and multiple ranges can be used to achieve a similar
      functionality in combination with the possibility to replace
      previous play messages.

   o  The use of PLAY for keep-alive signaling, i.e.  PLAY request
      without a range header in PLAY state, has also been deprecated.
      Instead a client can use, SET_PARAMETER (recommended) or OPTIONS
      (allowed) for keep alive.

   The important difference compared to a PLAY request in ready state is
   the handling of the current playpoint and how the range header in
   request is constructed.  The session is actively playing media and
   the playpoint will be moving making the exact time a request will
   take action is hard to predict.  Depending on how the PLAY header
   appears two different cases exist: total replacement or continuation.
   A total replacement is signalled 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 open start
   point and a explict stop value (Z), e.g. npt=-60, which indicate that
   it SHALL 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.  For both total replacement and continuation the
   PLAY request SHALL remove any additional range specifiers present in
   the previous request and add any that is present in the new PLAY
   request.

   An example of this behavior.  The server has received requests to
   play ranges 10 to 15 and then 13 to 20 (that is, overlapping ranges).
   If the new PLAY request arrives at the server 4 seconds after the
   previous one, it will take effect while the server plays the first
   range (10-15).  Thus changing the behavior of this range to continue
   to play to 25 seconds, i.e. the equivalent single request would be
   PLAY with range: npt=10-25.  Note that the second range (13-20) is
   deleted and never comes into effect.  If the new PLAY request would
   arrive as the second range in the first request was playing (13-20
   and shown below), then the equivalent single request would be play
   with range:npt=10-15,npt=13-25.









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     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-15, npt=13-20
           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=10-15, npt=13-20
           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
           Server: PhonyServer 1.0
           Range: npt=14-25
           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
           Session: 12345678

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 16.29):

   o  Random-Access property is set to Random Acces;

   o  Content Modifications set to unmutable;

   o  Retetion set Unlimited or Time-Limited.

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




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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 16.29):

   o  Random-Access set to Random Access;

   o  Content Modifications set to dynamic;

   o  Retetion set 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 ways for the client to get informed about changed of media
   resources in play state, if the resource was changed.  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 SHALL include
   the current Media-Range header (see Section 16.30).

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 16.29):

   o  Random-Access set to no-seeking;

   o  Content Modifications set to Time-Progressing;

   o  Retetion with Time-Duration set to 0.0.

   For live media, the SETUP response 200 OK SHALL include the Media-
   Range header (see Section 16.30).

   A client MAY send PLAY requests without the Range header, if the
   request include the Range header it SHALL use a symbolic value
   representing "now".  For NPT that range specification is "npt=now-".
   The server SHALL include the Range header in the response and it MUST
   indicate a explict time value and not a symbolic value.  In other
   words npt=now- is not a valid to use in the respone.  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
   UTC clock format SHOULD only be used if client has shown support for



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   it.

13.4.7.  Playing Live with Recording

   Certain media server may offer recording services of live sessions to
   their clients.  This recording would normally be from the begining of
   the media session.  Clients can randomly access the media between now
   and the begining 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 16.29):

   o  Random-Access set to random-access;

   o  Content Modifications set to Time-Progressing;

   o  Retetion set to Time-limited or Unlimited

   The SETUP response 200 OK SHALL include the Media-Range header (see
   Section 16.30) for this type of media.  For live media with recording
   the Range header indicates the current playback time in the media and
   the Media Range 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
   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 to 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 playback at Scale (Section 16.44) values other
   than 1.0 (Normal playback rate) while playing live media with
   recording.

13.4.8.  Playing Live with Time-Shift

   Certain media server 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 16.29):

   o  Random-Access set to random-access;

   o  Content Modifications set to Time-Progressing;

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




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   The SETUP response 200 OK SHALL include the Media-Range header (see
   Section 16.30) 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 to 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 playback
   at Scale (Section 16.44) values other than 1.0 (Normal playback rate)
   while playing live media with time-shift.

13.5.  PLAY_NOTIFY

   The PLAY_NOTIFY method is issued by a server to inform a client about
   an ansynchronous 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 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.  There is the 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 (seeSection 13.2 ), but there the client can state its
   acceptable message bodies by using the Accept header.  In case of
   PLAY_NOTIFY, the server does not know which message bodies are
   understood by the client.

   The Notify-Reason header (see Section 16.31) specifies the reason why
   the server sends the PLAY_NOTIFY request.  This is extensible and new
   reasons MAY be added in the future.  In case the client does not
   understand the reason for the notification it SHALL respond with an
   465 (Notification Reason Unknown) (Section 15.4.17) error code.
   Servers can send PLAY_NOTIFY with these types:




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

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

   o  and 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 end of the media streams has been reached or will be in
   the near future for the given session aggregate.  The request SHALL
   NOT be issued unless the server is in the playing state.  The end of
   the media stream delivery notification may be for either succesful
   completion of the PLAY request currently being served or indicate
   some error resulting in failure to complete the request.  The
   Request-Status header (Section 16.41) SHALL 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.  In case a PLAY_NOTIFY was issues
   prior to the actual completion and some error occured resulting in
   that the previosuly sent was in error a new Notification MUST 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.  The Range header indicates the point in
   the stream or streams where delivery was/are ending with the
   timescale that has been used by the client in the PLAY request being
   fulfilled.  For normal play time it is not alllowed to use "now" as
   server do know the real ending time of the media stream and now
   carries no information to determine what has/will be delivered.  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.

   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.

   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:






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     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/audio"
              ssrc=0D12F123:seq=14783;rtptime=2345962545
           Session: uZ3ci0K+Ld-M

     C->S: RTSP/2.0 200 OK
           CSeq: 854
           User-Agent: PhonyClient/1.2

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 16.29) and/or the available media range
   that can be played as indicated by Media-Range (Section 16.30).
   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.

   This notification SHALL be sent for media that are time-progressing
   every time a event happens that changes the basis for making
   estimations on how the media range progress.  In addition it is
   RECOMMENDED that the server sends these notification 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.  Requests for the just mentioned reasons SHALL include Media-
   Range header to provide current Media duration and the Range header
   to indicate the current playing point and any remaining parts of the
   requsted range.

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















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     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=0-1:37:21.394
           Range: npt=1:15:49.873-

     C->S: RTSP/2.0 200 OK
           CSeq: 854
           User-Agent: PhonyClient/1.2

13.5.3.  Scale-Change

   When a client request playback at Scale (Section 16.44) values other
   than 1.0 (Normal playback rate) then the server may be forced to
   changed the 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 front end of
   the media.  The server will be unable to continue provide content at
   Scale larger than 1 as content 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 playback point can reach the
   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 loose any media, the scale will need to be adjusted
   to the same rate which the media is removed from the storage buffer,
   commonly scale=1.0.

   To minimize impact on playback in any of the above cases the server
   SHALL modify the playback properties and set Scale to a supportable
   value (commonly 1.0) and continue delivery the media.  When doing
   this modification it MUST send a PLAY_NOTIFY message with the Notify-
   Reason header set to "Scale-Change".  The request SHALL contain a
   Range header with the media time where the change took effect, a
   Scale header with the new value in use, Session header with the ID
   for the session it applies to and a Date header with the server wall
   clock time of the change.  For time progressing content also the
   Media-Range and the Media-Properties at this point in time SHALL be
   included.

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




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   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=0-1:37:21.394
           Range: npt=1:37:21.394-
           Scale: 1
           RTP-Info: url="rtsp://example.com/fizzle/foo/audio"
               ssrc=0D12F123:rtptime=2345962545

     C->S: RTSP/2.0 200 OK
           CSeq: 854
           User-Agent: PhonyClient/1.2

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.
   Any attempt to do muting of a single media with an PAUSE request in
   an aggregated session SHALL be responded 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-

   The PAUSE request causes stream delivery to be interrupted
   immediately on receipt of the message and the pause point is set to



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   the current point in the presentation.  That pause point in the media
   stream needs to be maintained.  A subsequent PLAY request without
   Range header SHALL resume from the pause point and play until media
   end.

   The pause point after any PAUSE request SHALL be returned to the
   client by adding a Range header with what remains unplayed of the
   PLAY request's ranges, i.e. including all the remaining ranges part
   of multiple range specification.  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.  Any play-
   request including symbolic values, such as the NPT timescale's "now"
   MUST be resolved into the actual stream position where the pause
   point is.  For example a Play request with a range specification of
   "npt=now-" will need to be responded with an explicit value such as
   "npt=157.321-".  For media that is time-progressing and has retention
   duration=0 the follow-up PLAY request to start media delivery again,
   will need to use "npt=now-" and not the answer in the pause-respone.
     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
           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 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=21-30
           Session: 12345678




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   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:

     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

   The TEARDOWN client to server request stops the stream delivery for
   the given URI, freeing the resources associated with it.  A TEARDOWN
   request MAY be performed on either an aggregated or a media control
   URI.  However some restrictions apply depending on the current state.
   The TEARDOWN request SHALL contain a Session header indicating what
   session the request applies to.

   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 SHALL
   result in that media delivery is immediately halted and the session
   state is destroyed.  This SHALL be indicated through the lack of a
   Session header in the response.

   A TEARDOWN using a media URI in an aggregated session MAY 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
   that a session returns to non-aggregated control, due to that it only



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   contains a single media after the requests completion.  A session
   that will exist after the processing of the TEARDOWN request SHALL in
   the response to that TEARDOWN request contain a Session header.  Thus
   the presence of the Session header indicates to the receiver of the
   response if the session is still existing 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

13.8.  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
   exist two ways of specifying the parameters to retrive.  The first is
   by including headers that has been defined such that you can use them
   for this purpose.  Header for this purpose should allow empty, or
   stripped value parts to avoid having to specify bogus data when
   indicating the desire to retrive a value.  The succesful completion
   of the request should also be evident from any filled out values in
   the response.  The Media-Range header (Section 16.30) is one such
   header.  The other is to specify a body (entity) that lists the
   parameter(s) that are desirable to retrieve.  The Content-Type header
   (Section 16.18) is used to specify which format the entity has.

   The method MAY also be used without a body (entity) or any header
   that request parameters for keep-alive purpose.  Any request that 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 been processed.  Normaly 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.  Due to this reason it is usually easier if any parameters to
   be retrieved are sent in the body, rather than using any header.

   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) SHALL be sent



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   including a body listing these parameters that wasn't understood.  If
   all parameters are understood their value is filled in and returned
   in the response message body.

   Example:

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

           packets_received
           jitter

     C->S: RTSP/2.0 200 OK
           CSeq: 431
           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 body (entity).  It is the
   RECOMMENDED method to use in 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 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.

   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) SHALL be used.  In the case a parameter is
   not allowed to change, the error code is 458 (Parameter Is Read-
   Only).  The response body SHALL contain only the parameters that have



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   errors.  Otherwise no body SHALL be returned.

   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
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff

     S->C: RTSP/2.0 451 Parameter Not Understood
           CSeq: 421
           Content-length: 10
           Content-type: text/parameters

           barparam: barstuff

13.10.  REDIRECT

   The REDIRECT method is issued by a server to inform a client that it
   required to connect to another server location to access the resource
   indicated by the Request-URI.  The presence of the Session header in
   a REDIRECT request indicates the scope of the request, and determines
   the specific semantics of the request.

   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 OPTIONAL Location
   header, if included in such a request, SHALL contain a complete
   absolute URI pointing to the resource to which the client SHOULD
   reconnect.  Specifically, the Location SHALL NOT contain just the
   host and port.  A client may receive a REDIRECT request with a



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   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,
   as well, to the control 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 control connection MUST be redirected.  The OPTIONAL 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

   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.

   The lack of a Location header in any REDIRECT request is indicative
   of the server no longer being able to fulfill the current request and
   having no alternatives for the client to continue with its normal
   operation.  It is akin to a server initiated TEARDOWN that applies
   both to sessions as well as the general connection associated with
   that client.

   When the Range header is not included in a REDIRECT request, the
   client SHOULD perform the redirection immediately and return a
   response to the server.  The server can consider the session as
   terminated and can free any associated state after it receives the
   successful (2xx) response.  The server MAY close the signalling
   connection upon receiving the response and the client SHOULD close
   the signalling 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 signalling 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
   signalling connection.



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   If the OPTIONAL Range header is included in a REDIRECT request, it
   indicates when the redirection takes effect.  The range value MUST be
   an open ended single value, e.g. npt=59-, indicating the play out
   time when redirection SHALL occur.  Alternatively, a range with a
   time= parameter indicates the wall clock time by when the redirection
   MUST take place.  When the time= parameter is present in the range,
   any range value MUST be ignored even though it MUST be syntactically
   correct.  To allow a client to determine that redirect time without
   being time synchronized with the server, the server SHALL include a
   Date header in the request.  When the indicated redirect point is
   reached, a client MUST issue a TEARDOWN request and SHOULD close the
   signalling connection after receiving a 2xx response.  The normal
   connection considerations apply for the server.

      The differentiation of REDIRECT requests with and without range
      headers is to allow for clear and explicit state handling.  As the
      state in the server needs to be kept until the point of
      redirection, the handling becomes more clear if the client is
      required to TEARDOWN the session at the redirect point.

   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 refuses to respond to a REDIRECT request.
   That should not provide any benefit.

   After a REDIRECT request has been processed, a client that wants to
   continue to send or 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.

      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 ETag identifiers are RECOMMENDED to
   verify the assumption.

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






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     S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 732
           Location: rtsp://s2.example.com:8001
           Range: npt=0- ;time=19960213T143205Z
           Session: uZ3ci0K+Ld-M

     C->S: RTSP/2.0 200 OK
           CSeq: 732
           User-Agent: PhonyClient/1.2










































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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.

   Stream data such as RTP packets is encapsulated by an ASCII dollar
   sign (24 decimal), followed by a one-byte channel identifier,
   followed by the length of the encapsulated binary data as a binary,
   two-byte integer in network byte order.  The stream data follows
   immediately afterwards, without a CRLF, but including the upper-layer
   protocol headers.  Each $ block SHALL contain exactly one upper-layer
   protocol data unit, e.g., one RTP packet.
       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
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      | "$" = 24      | Channel ID    | Length in bytes               |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      : Length number of bytes of binary data                         :
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The channel identifier is defined in the Transport header with the
   interleaved parameter (Section 16.51).

   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 a range containing a second channel in the
   interleaved parameter of the Transport header, see Section 16.51.  If
   RTCP is used, packets SHALL be sent on the first available channel
   higher than the RTP channel.  The channels are bi-directional and
   therefore RTCP traffic are sent on the second channel in both
   directions.

      RTCP is sometime 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 TCP control
      connection when required by the network configuration and transfer
      them onto UDP when possible.








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     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, UTC
           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

     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 18:59:15 GMT
           RTP-Info: url="rtsp://example.com/bar.file"
             ssrc=0D12F123:seq=232433;rtptime=972948234
           Range: npt=0-56.8

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






















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15.  Status Code Definitions

   Where applicable, HTTP status [H10] codes are reused.  Status codes
   that have the same meaning are not repeated here.  See Table 4 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.

15.1.  Success 1xx

15.1.1.  100 Continue

   See, [H10.1.1].

15.2.  Success 2xx

15.2.1.  200 OK

   See, [H10.2.1].

15.3.  Redirection 3xx

   The notation "3rr" indicates response codes from 300 to 399 inclusive
   which are meant for redirection.  The response code 304 is excluded
   from this set, as it is not used for redirection.

   See [H10.3] for definition of status code 300 to 305.  However
   comments are given for some to how they apply to RTSP.

   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.  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 a
   established session about the need for redirecting the session.  If
   an 3rr response is received for an request in relation to a
   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 the Location header is used in a response it SHALL contain an
   absolute URI pointing out the media resource the client is redirected
   to, the URI SHALL NOT only contain the host name.



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15.3.1.  300 Multiple Choices

   See [H10.3.1].

15.3.2.  301 Moved Permanently

   The request resource are 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.

15.3.3.  302 Found

   The requested resource resides temporarily at the URI given by the
   Location header.  The Location header MUST be included in the
   response.  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, UTC
           User-Agent: PhonyClient/1.2

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

15.3.4.  303 See Other

   This status code SHALL NOT be used in RTSP.  However, it was allowed
   to use in RTSP 1.0 (RFC 2326).

15.3.5.  304 Not Modified

   If the client has performed a conditional DESCRIBE or SETUP (see
   Section 16.25) 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:




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   o  Date

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

   o  Expires, Cache-Control, and/or Vary, 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 ETag and If-Match headers may be used to link the
   DESCRIBE and SETUP in this manner.

15.3.6.  305 Use Proxy

   See [H10.3.6].

15.4.  Client Error 4xx

15.4.1.  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 [H10.4.1].  If the request does not have a CSeq header,
   the server MUST NOT include a CSeq in the response.

15.4.2.  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.

15.4.3.  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 entity body containing the offending
   parameter(s).

15.4.4.  452 reserved

   This error code was removed from RFC 2326 [RFC2326] and is obsolete.




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15.4.5.  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.

15.4.6.  454 Session Not Found

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

15.4.7.  455 Method Not Valid in This State

   The client or server cannot process this request in its current
   state.  The response SHALL contain an Allow header to make error
   recovery possible.

15.4.8.  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 SHALL be returned to inform the client of
   which format(s) that are allowed.

15.4.9.  457 Invalid Range

   The Range value given is out of bounds, e.g., beyond the end of the
   presentation.

15.4.10.  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 entity body containing the offending parameter(s).

15.4.11.  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.

15.4.12.  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.



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15.4.13.  461 Unsupported Transport

   The Transport field did not contain a supported transport
   specification.

15.4.14.  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.

15.4.15.  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.

15.4.16.  464 Data Transport Not Ready Yet

   The data transmission channel to the media destination is not yet
   ready for carrying data.  However the responding entity still expects
   that the data transmission channel will be established at this 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 successful 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.

15.4.17.  465 Notification Reason Unknown

   The client has received a PLAY_NOTIFY (Section 13.5) with a Notify-
   Reason header (Section 16.31) indicates a reson that are unknown to
   the client.

15.4.18.  470 Connection Authorization Required

   The secured connection attempt need user or client authorization
   before proceeding.  The next hops certificate is included in this
   response in the Accept-Credentials header.






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15.4.19.  471 Connection Credentials not accepted

   When performing a secure connection over multiple connections, a
   intermediary has refused to connect to the next hop and carry out the
   request due to unacceptable credentials for the used policy.

15.4.20.  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 suits.

15.5.  Server Error 5xx

15.5.1.  551 Option not supported

   A feature-tag given in the Require or the Proxy-Require fields was
   not supported.  The Unsupported header SHALL be returned stating the
   feature for which there is no support.
































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16.  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         | P,S    | OPT     |      |
       |               |                |        |         |      |
       |               | S -> C         |        |         |      |
       |               |                |        |         |      |
       | 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     |      |
       +---------------+----------------+--------+---------+------+

   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: It is allowed for all error messages 4xx and 5xx to
                                have a body

   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, we use [HX.Y] to
   refer to Section X.Y of the current 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:




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   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.

   An empty entry in the "where" column indicates that the header field
   may be present in both requests and responses.

   The "proxy" column describes the operations a proxy may perform on a
   header field.  An empty proxy column indicates that the proxy SHALL
   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:

   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 is SHALL be present if the message body is not
         empty.  See Section 16.16, Section 16.18 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



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   other case it is request to process the header.  This is regulated by
   the method and header descriptions.  Example of such headers that
   require processing are the Require and Proxy-Require header fields
   discussed in Section 16.42 and Section 16.35.  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 SHALL ignore extension headers that are not understood.

   The From and Location header fields contain an URI.  If the URI
   contains a comma, or semicolon, the URI MUST be enclosed in double
   quotas (").  Any URI parameters are contained within these quotas.
   If the URI is not enclosed in double quotas, any semicolon- delimited
   parameters are header-parameters, not URI parameters.

   +----------------+------+-----+-----+-----+------+-----+------+-----+
   | Header         | Wher | Pro | DES | OPT | SETU | PLA | PAUS | TRD |
   |                | e    | xy  |     |     | P    | Y   | E    |     |
   +----------------+------+-----+-----+-----+------+-----+------+-----+
   | Accept         | R    |     | o   | -   | -    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Accept-Credent | R    | r   | o   | o   | o    | o   | o    | o   |
   | ials           |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Accept-Encodin | R    | r   | o   | -   | -    | -   | -    | -   |
   | g              |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Accept-Languag | R    | r   | o   | -   | -    | -   | -    | -   |
   | e              |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Accept-Ranges  | R    | r   | -   | -   | m    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Accept-Ranges  | r    | r   | -   | -   | o    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Accept-Ranges  | 456  | r   | -   | -   | -    | o   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Allow          | r    | am  | c   | c   | c    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Allow          | 405  | am  | m   | m   | m    | m   | m    | m   |
   |                |      |     |     |     |      |     |      |     |



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   | Authorization  | R    |     | o   | o   | o    | o   | o    | o   |
   |                |      |     |     |     |      |     |      |     |
   | Bandwidth      | R    |     | o   | o   | o    | o   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Blocksize      | R    |     | o   | -   | o    | o   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Cache-Control  |      | r   | o   | -   | o    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Connection     |      |     | o   | o   | o    | o   | o    | o   |
   |                |      |     |     |     |      |     |      |     |
   | Connection-Cre | 470, | ar  | o   | o   | o    | o   | o    | o   |
   | dentials       | 407  |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Base   | r    |     | o   | -   | -    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Content-Base   | 4xx, |     | o   | o   | o    | o   | o    | o   |
   |                | 5xx  |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Encodi | R    | r   | -   | -   | -    | -   | -    | -   |
   | ng             |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Encodi | r    | r   | o   | -   | -    | -   | -    | -   |
   | ng             |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Encodi | 4xx, | r   | o   | o   | o    | o   | o    | o   |
   | ng             | 5xx  |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Langua | R    | r   | -   | -   | -    | -   | -    | -   |
   | ge             |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Langua | r    | r   | o   | -   | -    | -   | -    | -   |
   | ge             |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Langua | 4xx, | r   | o   | o   | o    | o   | o    | o   |
   | ge             | 5xx  |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Length | r    | r   | *   | -   | -    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Content-Length | 4xx, | r   | *   | *   | *    | *   | *    | *   |
   |                | 5xx  |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Locati | r    |     | o   | -   | -    | -   | -    | -   |
   | on             |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Locati | 4xx, |     | o   | o   | o    | o   | o    | o   |
   | on             | 5xx  |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | Content-Type   | r    |     | *   | -   | -    | -   | -    | -   |



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   | Content-Type   | 4xx, |     | *   | *   | *    | *   | *    | *   |
   |                | 5xx  |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | CSeq           | Rc   | rm  | m   | m   | m    | m   | m    | m   |
   |                |      |     |     |     |      |     |      |     |
   | Date           |      | am  | o   | o   | o    | o   | o    | o   |
   |                |      |     |     |     |      |     |      |     |
   | ETag           | r    | r   | o   | -   | o    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Expires        | r    | r   | o   | -   | -    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | From           | R    | r   | o   | o   | o    | o   | o    | o   |
   |                |      |     |     |     |      |     |      |     |
   | If-Match       | R    | r   | -   | -   | o    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | If-Modified-Si | R    | r   | o   | -   | o    | -   | -    | -   |
   | nce            |      |     |     |     |      |     |      |     |
   |                |      |     |     |     |      |     |      |     |
   | If-None-Match  | R    | r   | o   | -   | -    | -   | -    | -   |
   |                |      |     |     |     |      |     |      |     |
   | Last-Modified  | r    | r   | 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 | Prox | DE | OPT | SETUP | PLAY | PAUSE | TRD |
   |            |       | y    | S  |     |       |      |       |     |
   +------------+-------+------+----+-----+-------+------+-------+-----+
   | Media-     |       |      | -  | -   | r     | r    | r     | -   |
   | Properties |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Media-     |       |      | -  | -   | r     | r    | r     | -   |
   | Range      |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Pipelined- |       | amdr | -  | o   | o     | o    | o     | o   |
   | Requests   |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Proxy-     | 407   | amr  | m  | m   | m     | m    | m     | m   |
   | Authentica |       |      |    |     |       |      |       |     |
   | te         |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Proxy-     | R     | rd   | o  | o   | o     | o    | o     | o   |
   | Authorizat |       |      |    |     |       |      |       |     |
   | ion        |       |      |    |     |       |      |       |     |



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   | Proxy-     | R     | ar   | o  | o   | o     | o    | o     | o   |
   | Require    |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Proxy-     | r     | r    | c  | c   | c     | c    | c     | c   |
   | Require    |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Proxy-     | R     | amr  | c  | c   | c     | c    | c     | c   |
   | Supported  |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Proxy-     | r     |      | c  | c   | c     | c    | c     | c   |
   | Supported  |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Public     | r     | admr | -  | m   | -     | -    | -     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Public     | 501   | admr | m  | m   | m     | m    | m     | m   |
   |            |       |      |    |     |       |      |       |     |
   | Range      | R     |      | -  | -   | -     | o    | -     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Range      | r     |      | -  | -   | c     | m    | m     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Referer    | R     |      | o  | o   | o     | o    | o     | o   |
   |            |       |      |    |     |       |      |       |     |
   | Request-   | R     |      | -  | -   | -     | -    | -     | -   |
   | Status     |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | Require    | R     |      | o  | o   | o     | o    | o     | o   |
   |            |       |      |    |     |       |      |       |     |
   | Retry-Afte | 3rr,5 |      | o  | o   | o     | -    | -     | -   |
   | r          | 03    |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | RTP-Info   | r     |      | -  | -   | c     | c    | -     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Scale      |       |      | -  | -   | -     | o    | -     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Seek-Style | R     |      | -  | -   | -     | o    | -     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Seek-Style | r     |      | -  | -   | -     | m    | -     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Session    | R     | r    | -  | o   | o     | m    | m     | m   |
   |            |       |      |    |     |       |      |       |     |
   | Session    | r     | r    | -  | c   | m     | m    | m     | o   |
   |            |       |      |    |     |       |      |       |     |
   | Server     | R     | r    | -  | o   | -     | -    | -     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Server     | r     | r    | o  | o   | o     | o    | o     | o   |
   |            |       |      |    |     |       |      |       |     |
   | Speed      |       |      | -  | -   | -     | o    | -     | -   |
   |            |       |      |    |     |       |      |       |     |



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   | Supported  | R     | amr  | o  | o   | o     | o    | o     | o   |
   |            |       |      |    |     |       |      |       |     |
   | Supported  | r     | amr  | c  | c   | c     | c    | c     | c   |
   |            |       |      |    |     |       |      |       |     |
   | Timestamp  | R     | admr | o  | o   | o     | o    | o     | o   |
   |            |       |      |    |     |       |      |       |     |
   | Timestamp  | c     | admr | m  | m   | m     | m    | m     | m   |
   |            |       |      |    |     |       |      |       |     |
   | Transport  |       | amr  | -  | -   | m     | -    | -     | -   |
   |            |       |      |    |     |       |      |       |     |
   | Unsupporte | r     |      | c  | c   | c     | c    | c     | c   |
   | d          |       |      |    |     |       |      |       |     |
   |            |       |      |    |     |       |      |       |     |
   | User-Agent | R     |      | m* | m*  | m*    | m*   | m*    | m*  |
   |            |       |      |    |     |       |      |       |     |
   | Vary       | r     |      | c  | c   | c     | c    | c     | c   |
   |            |       |      |    |     |       |      |       |     |
   | 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   |
   | Authentica |       |      |    |     |       |      |       |     |
   | te         |       |      |    |     |       |      |       |     |
   +------------+-------+------+----+-----+-------+------+-------+-----+

     Table 10: Overview of RTSP header fields (P-W) related to methods
           DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.

   +------------------------+---------+-------+-----+-----+-----+-----+
   | Header                 | Where   | Proxy | GPR | SPR | RDR | PNY |
   +------------------------+---------+-------+-----+-----+-----+-----+
   | Accept-Credentials     | R       | r     | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Allow                  | 405     | amr   | m   | m   | m   | -   |
   |                        |         |       |     |     |     |     |
   | Authorization          | R       |       | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Bandwidth              | R       |       | -   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Blocksize              | R       |       | -   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Connection             |         |       | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Connection-Credentials | 470,407 | ar    | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Base           | R       |       | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |



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   | Content-Base           | r       |       | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Base           | 4xx,5xx |       | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Encoding       | R       | r     | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Encoding       | r       | r     | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Encoding       | 4xx,5xx | r     | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Language       | R       | r     | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Language       | r       | r     | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Language       | 4xx,5xx | r     | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Length         | R       | r     | *   | *   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Length         | r       | r     | *   | *   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Length         | 4xx,5xx | r     | *   | *   | *   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Location       | R       |       | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Location       | r       |       | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Location       | 4xx,5xx |       | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Type           | R       |       | *   | *   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Type           | r       |       | *   | *   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Content-Type           | 4xx     |       | *   | *   | *   | -   |
   |                        |         |       |     |     |     |     |
   | CSeq                   | R,c     | mr    | m   | m   | m   | m   |
   |                        |         |       |     |     |     |     |
   | Date                   | R       | a     | o   | o   | m   | -   |
   |                        |         |       |     |     |     |     |
   | Date                   | r       | am    | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | From                   | R       | r     | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Last-Modified          | R       | r     | -   | -   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Last-Modified          | r       | r     | o   | -   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Location               | 3rr     |       | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |



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   | Location               | R       |       | -   | -   | m   | -   |
   |                        |         |       |     |     |     |     |
   | Media-Properties       |         |       | -   | -   | -   |     |
   |                        |         |       |     |     |     |     |
   | Media-Range            | R       |       | o   | -   | -   | c   |
   |                        |         |       |     |     |     |     |
   | Media-Range            | r       |       | c   | -   | -   | -   |
   |                        |         |       |     |     |     |     |
   | Notify-Reason          | R       |       | -   | -   | -   | m   |
   |                        |         |       |     |     |     |     |
   | Pipelined-Requests     |         | amdr  | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Proxy-Authenticate     | 407     | amr   | m   | m   | m   | -   |
   |                        |         |       |     |     |     |     |
   | Proxy-Authorization    | R       | rd    | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Proxy-Require          | R       | ar    | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | Proxy-Require          | r       | r     | c   | c   | c   | -   |
   |                        |         |       |     |     |     |     |
   | 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, PLAY_NOTIFY, and REDIRECT.

      +------------------+---------+-------+-----+-----+-----+-----+
      | Header           | Where   | Proxy | GPR | SPR | RDR | PNY |
      +------------------+---------+-------+-----+-----+-----+-----+
      | Range            | R       |       | -   | -   | o   | m   |
      |                  |         |       |     |     |     |     |
      | Referer          | R       |       | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Request-Status   | R       |       | -   | -   | -   | m   |
      |                  |         |       |     |     |     |     |
      | Require          | R       | r     | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Retry-After      | 3rr,503 |       | o   | o   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Scale            |         |       | -   | -   | -   | c   |
      |                  |         |       |     |     |     |     |
      | Seek-Style       |         |       | -   | -   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Session          | R       | r     | o   | o   | o   | m   |



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      | Session          | r       | r     | c   | c   | o   | m   |
      |                  |         |       |     |     |     |     |
      | Server           | R       | r     | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Server           | r       | r     | o   | o   | -   | -   |
      |                  |         |       |     |     |     |     |
      | Supported        | R       | adrm  | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Supported        | r       | adrm  | c   | c   | c   | -   |
      |                  |         |       |     |     |     |     |
      | Timestamp        | R       | adrm  | o   | o   | o   | -   |
      |                  |         |       |     |     |     |     |
      | Timestamp        | c       | adrm  | m   | m   | m   | -   |
      |                  |         |       |     |     |     |     |
      | Unsupported      | r       | arm   | c   | c   | c   | -   |
      |                  |         |       |     |     |     |     |
      | User-Agent       | R       | r     | m*  | m*  | -   | -   |
      |                  |         |       |     |     |     |     |
      | User-Agent       | r       | r     | -   | -   | m*  | -   |
      |                  |         |       |     |     |     |     |
      | Vary             | r       |       | c   | c   | -   | -   |
      |                  |         |       |     |     |     |     |
      | 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,  PLAY_NOTIFY, and REDIRECT.

16.1.  Accept

   The Accept request-header field can be used to specify certain
   presentation description content types which are acceptable for the
   response.

   See [H14.1] for syntax.

   Example of use:
     Accept: application/example ;q=1.0, application/sdp

16.2.  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



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   SHALL NOT be included in server to client requests.

   In a request the header SHALL contain the method (User, Proxy, or
   Any) for approving credentials selected by the requestor.  The method
   SHALL 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 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 as the client has not yet received any
   credentials to accept.  Each credential SHALL consist of one URI
   identifying the proxy or server, the hash algorithm identifier, and
   the hash over that entity's DER encoded certificate [RFC5280] in
   Base64 [RFC4648].  All RTSP clients and proxies SHALL 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
   exist.

   Example:
   Accept-Credentials:User
     "rtsps://proxy2.example.com/";sha-256;exaIl9VMbQMOFGClx5rXnPJKVNI=,
     "rtsps://server.example.com/";sha-256;lurbjj5khhB0NhIuOXtt4bBRH1M=

16.3.  Accept-Encoding

   See [H14.3].

16.4.  Accept-Language

   See [H14.4].  Note that the language specified applies to the
   presentation description and any reason phrases, not the media
   content.

16.5.  Accept-Ranges

   The Accept-Ranges request and response-header field allows indication
   of the format supported in the Range header.  The client SHALL
   include the header in SETUP requests to indicate which formats it
   support to receive in PLAY and PAUSE responses, and REDIRECT
   requests.  The server SHALL include the header in SETUP and 456 error
   responses to indicate the formats supported for the resource



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   indicated by the request URI.
      Accept-Ranges: NPT, SMPTE

   This header has the same syntax as [H14.5] and the syntax is defined
   in Section 20.2.3.  However, new range-units are defined.

16.6.  Allow

   The Allow entity-header field lists the methods supported by the
   resource identified by the Request-URI.  The purpose of this field is
   to strictly inform the recipient of valid methods associated with the
   resource.  An Allow header field MUST be present in a 405 (Method Not
   Allowed) response.  See [H14.7] for syntax definition.  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.

   The Allow SHALL also be included in SETUP and DESCRIBE responses, if
   the methods allowed for the resource is different than the minimal
   implementation set.

   Example of use:
      Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE

16.7.  Authorization

   See [H14.8].

16.8.  Bandwidth

   The Bandwidth request-header field describes the estimated bandwidth
   available to the client, expressed as a positive integer and measured
   in bits per second.  The bandwidth available to the client may change
   during an RTSP session, e.g., due to mobility, congestion, etc.

   Example:
     Bandwidth: 62360

16.9.  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,
   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



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   server only returns an error (4xx) if the value is syntactically
   invalid.

16.10.  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 SETUP request and its
   response.  Note: Cache-Control does not govern the caching of
   responses as for HTTP, instead it applies to the media stream
   identified by the SETUP request.  The RTSP requests are generally not
   cacheable, for further information see Section 18.  Below is the
   description of the cache directives that can be included in the
   Cache-Control header.

   no-cache:  Indicates that the media stream 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
         enforcing that the content can't be cached.

   public:  Indicates that the media stream is cacheable by any cache.

   private:  Indicates that the media stream 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
         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.  Therefore,
         if a response includes the no-transform directive, an
         intermediate cache or proxy MUST NOT change the encoding of the
         stream.  Unlike HTTP, RTSP does not provide for partial



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         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 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 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 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 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
         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
         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.



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   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 entity 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 entity with a 200 (OK)
   response.

16.11.  Connection

   See [H14.10].  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 see 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.

16.12.  Connection-Credentials

   The Connection-Credentials response header is used to carry the chain
   of credentials of any next hop that need to be approved by the
   requestor.  It SHALL only be used in server to client responses.

   The Connection-Credentials header in an RTSP response SHALL, 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
   [RFC4648] encoded binary structure containg 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).



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   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:

        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 certifcates        | 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                                :
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

16.13.  Content-Base

   The Content-Base entity-header field may be used to specify the base
   URI for resolving relative URIs within the entity.
   Content-Base: rtsp://media.example.com/movie/twister
   If no Content-Base field is present, the base URI of an entity 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 entity-body may be redefined within that entity-
   body.

16.14.  Content-Encoding

   See [H14.11].

16.15.  Content-Language

   See [H14.12].




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16.16.  Content-Length

   The Content-Length general-header field contains the length of the
   body (entity) of the message (i.e. after the double CRLF following
   the last header).  Unlike HTTP, it MUST be included in all messages
   that carry body beyond the header portion of the message.  If it is
   missing, a default value of zero is assumed.  It is interpreted
   according to [H14.13].

16.17.  Content-Location

   See [H14.14].

16.18.  Content-Type

   See [H14.17].  Note that the content types suitable for RTSP are
   likely to be restricted in practice to presentation descriptions and
   parameter-value types.

16.19.  CSeq

   The CSeq general-header field specifies the sequence number for an
   RTSP request-response pair.  This field MUST be present in all
   requests and responses.  For every RTSP request containing the given
   sequence number, the corresponding response will have the same
   number.  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).  For each new RTSP request
   the CSeq value SHALL be incremented by one.  The initial sequence
   number MAY 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 request to a server
   and the server has another when sending request to the client.  Each
   requester and responder is identified with its network address.

   Proxies that aggregate several sessions on the same transport will
   regularly need to renumber the CSeq header field in requests and
   responses to fulfill the rules for the header.

   Example:
   CSeq: 239

16.20.  Date

   See [H14.18].  An RTSP message containing a body MUST include a Date
   header if the sending host has a clock.  Servers SHOULD include a
   Date header in all other RTSP messages.




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16.21.  ETag

   The ETag response header MAY be included in DESCRIBE or SETUP
   responses.  The entity tags (Section 4.8) 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.  However it has the
   disadvantage that a change in any of the parts results in
   invalidation of all the parts.

   If the ETag is provided both inside the entity, e.g. within the
   "a=etag" attribute in SDP, and in the response message, then both
   tags SHALL be identical.  It is RECOMMENDED that the ETag is
   primarily given in the RTSP response message, to ensure that caches
   can use the ETag 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 entity
   tag in the session description as specified in Appendix D.1.9.

   SETUP and DESCRIBE requests can be made conditional upon the ETag
   using the headers If-Match (Section 16.24) and If-None-Match (
   Section 16.26).

16.22.  Expires

   The Expires entity-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 entity).  See Section 18 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 HTTP-date in



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   [H3.3]; it MUST be in RFC1123-date format:

   An example of its use is
     Expires: Thu, 01 Dec 1994 16:00:00 GMT

   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.

16.23.  From

   See [H14.22].

16.24.  If-Match

   See [H14.24].

   The If-Match request-header field is especially useful for ensuring
   the integrity of the presentation description, in both the case where
   it is fetched via means external to RTSP (such as HTTP), or in the
   case where the server implementation is guaranteeing the integrity of
   the description between the time of the DESCRIBE message and the
   SETUP message.  By including the ETag given in or with the session
   description in a SETUP request, the client ensures that resources set
   up are matching the description.  A SETUP request for which the ETag
   validation check fails, SHALL responde using 412 (Precondition
   Failed).

   This validation check is also very useful if a session has been
   redirected from one server to another.

16.25.  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 SHALL be returned without any message-body.

   An example of the field is:



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     If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT

16.26.  If-None-Match

   See [H14.26].

   This request header can be used with one or several entity tags to
   make DESCRIBE requests conditional.  A new session description is
   retrieved only if another entity than the ones already available
   would be included.  If the entity available for delivery is matching
   the one the client already has, then a 304 (Not Modified) response is
   given.

16.27.  Last-Modified

   The Last-Modified entity-header field indicates the date and time at
   which the origin server believes the presentation description or
   media stream was last modified.  See [H14.29].  For the methods
   DESCRIBE, the header field indicates the last modification date and
   time of the description, for SETUP that of the media stream.

16.28.  Location

   See [H14.30].

16.29.  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 MUST have received the update
   prior to any action related to the new media properties take affect.
   For aggregated sessions the Media-Properties header will be returned
   in each SETUP response and the one that applies is the response to
   the last request.

   The media properties expressed by this header is the one applicable
   to all media in the RTSP session.  So 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 are enforced within the header.
   Property values should be grouped into a single group that handles a
   particular orthogonal property.  Values or groups that express



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   multiple properties SHOULD NOT be used.  The list of properties that
   can be expressed MAY be extended at any time.  Unknown property
   values SHALL be ignored.

   This specification defines the following 3 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
         the longest duration between any two random access points in
         the media.

      Begining-Only:  Seeking is limited to the begining only.

      No-Seeking:  No seeking is possible.

   Content Modifications

      Unmutable:  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 accesible progress as wall clock 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 wall clock time.  The time must be provided in the
         absolute time format specified in Section Section 4.6.

      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.

   An Example of this header for first an on-demand content and then a
   live stream without recording.




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   On-demand:
   Media-Properties: Random-Acess=2.5s, Unlimited, Unmutable

   Live stream without recording/timeshifting:
   Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0.0

16.30.  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 SHALL 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 value.  The server SHALL in this
   case include the curent value at the time of sending the response.

   The header SHALL include range specification for all time formats
   supported for the media, as indicated in Accept-Ranges header
   (Section 16.5) 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-continous range.

   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 wall clock progresses so will also the media range.
   However it shall be assumed that media time progress in direct
   relationship to wall clock time (with the exception of clock skew) so
   that a reasoanbly accurate estiamation of the media range can be
   calculated.

16.31.  Notify-Reason

   The Notify Reason header is solely used in the PLAY_NOTIFY method.
   It indiciates the reason why the server has sent the asynchronous
   PLAY_NOTIFY request (see Section 13.5).

16.32.  Pipelined-Requests

   The Pipelined-Requests general header is used to indicate that a
   request is to be executed in the context created by previous
   requests.  The primary usage of this header is to allow pipelining of
   SETUP requests so that any additional SETUP request after the first
   one doesn't need to wait for the session ID to be sent back to the
   requesting entity.  The header contains a unique identifier that is
   scoped by the persistent connection used to send the requests.




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   Upon receiving a request with the Pipelined-Requests the responding
   entity SHALL look up if there exist a binding between this Pipelined-
   Requests identifier for the current persistent connection and an RTSP
   session ID.  If that exist then the received request is processed the
   same way as if it did contain the Session header with the looked up
   session ID.  If there doesn't exist a mapping and no Session header
   is included in the request, the responding entity SHALL create a
   binding upon the succesful completion of a session creating request,
   i.e.  SETUP.  If the request failed to create an RTSP session no
   binding SHALL be created.  In case the request contains both a
   Session header and the Pipelined-Requests header the Pipelined-
   Requests SHALL 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
   creating a session).  After that first one, additional SETUP requests
   or request of any type using the RTSP session context may include the
   Pipelined-Requests header.

   For all responses to request that contained the Pipelined-Requests,
   the Session header and the Pipelined-Requests SHALL both be included,
   assuming that it is allowed for that response and that the binding
   between the header values exist.  Pipelined-Requests SHOULD NOT be
   used in requests after that the client has received the RTSP Session
   ID.  This as using the real session ID allows the request to be used
   also in cases the persistent connection has been terminated and a new
   connection is needed.

   It is the sender of the request that is responsible for using a
   previously unused identifier within this transport connection scope
   when a new RTSP session is to be cretated with this method.  A server
   side binding SHALL be deleted upon the termination of the related
   RTSP session.  Note: Although this definition would allow for reusing
   previously used pipelining identifiers, this is NOT RECOMMENDED to
   allow for better error handling and logging.

   RTSP Proxies may need to translate Pipelined-Requests identifier
   values from incoming request to outgoing to allow for aggregation of
   requests onto a persistent connection.

16.33.  Proxy-Authenticate

   See [H14.33].







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16.34.  Proxy-Authorization

   See [H14.34].

16.35.  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 SHALL 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 16.42 for more details on the mechanics of this message
   and a usage example.  See discussion in the proxies section
   (Section 17.1) about when to consider that a feature requires proxy
   support.

   Example of use:
      Proxy-Require: play.basic

16.36.  Proxy-Supported

   The Proxy-Supported 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
   SHALL 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 4.7.  The list are the
   intersection of all feature-tags understood by the proxies.  To
   achieve an intersection, the proxy adding the Proxy-Supported header
   includes all proxy feature-tags it understands.  Any proxy receiving
   a request with the header, checks the list and removes any feature-
   tag it do not support.  A Proxy-Supported header present in the
   response SHALL NOT be touched by the proxies.

   Example:







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     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 prox1.example.com

    P2->S:  OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech
            Proxy-Supported: proxy-foo, proxy-blech
            Via: 2.0 prox1.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 prox1.example.com, 2.0 prox2.example.com

16.37.  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 14.7) 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
   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.




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16.38.  Range

   The Range 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.

   The range can be specified in a number of units.  This specification
   defines smpte (Section 4.4), npt (Section 4.5), and clock
   (Section 4.6) range units.  While 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.

   The Range header MAY contain a time parameter in UTC, specifying the
   time at which the operation is to be made effective.  This
   functionality SHALL be used only with the REDIRECT method.

   Example:
     Range: clock=19960213T143205Z-;time=19970123T143720Z

      The notation is similar to that used for the HTTP/1.1 [RFC2616]
      byte-range header.  It allows clients to select an excerpt from
      the media object, and to play from a given point to the end as
      well as from the current location to a given point.

   Ranges are half-open intervals, including the first point, but
   excluding the second point.  In other words, a range of A-B starts
   exactly at time A, but stops 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.

   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 16.44) indicates a negative scale value.  For example, this
   would be the case when a playback in reverse is desired.



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   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.

   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.

16.39.  Referer

   See [H14.36].  The URI refers to that of the presentation
   description, typically retrieved via HTTP.

16.40.  Retry-After

   See [H14.37].

16.41.  Request-Status

   This request header is used to indicate the end result for requests
   that takes time to complete, such a 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 is reports on using the
   CSeq number for the session indicated by the Session header in the
   request.  It provies both a numerical status code (according to
   Section 8.1.1) and a human readable reason phrase.

   Example:
   Request-Status: cseq=63 status=500 reason="Media data unavailable"








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16.42.  Require

   The Require request-header field is used by clients or servers 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 (Section 16.49) is much more effective in this purpose.
   The server MUST respond to this header by using the Unsupported
   header to negatively acknowledge those feature-tags which are NOT
   supported.  The response SHALL use the error code 551 (Option Not
   Supported).  This header does not apply to proxies, for the same
   functionality in respect to proxies see Proxy-Require header
   (Section 16.35) 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 what a server, do need to understand also the
   server 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.

   Proxies and other intermediary devices SHALL 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 16.35).  See discussion in the proxies section
   (Section 17.1) about when to consider that a feature requires proxy



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   support.

16.43.  RTP-Info

   The RTP-Info response-header field is used to set RTP-specific
   parameters in the PLAY response.  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 that a client needs to
      synchronize the media streams using RTCP.  This may have negative
      impact as the RTCP can be lost, and does not need to be
      particulary timely in their arrival.  Also functionality as
      informing the client from which packet a seek has occurred is
      affected.

   The RTP-Info MAY also be included in SETUP responses to provide
   synchronization information when changing transport parameters, see
   Section 13.3.

   The header can carry the following parameters:

   url:  Indicates the stream URI which for which the following RTP
         parameters correspond, this URI MUST be the same used in the
         SETUP request for this media stream.  Any relative URI SHALL
         use the Request-URI as base URI.  This parameter SHALL be
         present.

   ssrc: The Synchronization source (SSRC) that the RTP timestamp and
         sequence number provide applies to.  This parameter SHALL 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:  SHALL 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



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         inter-media synchronization is achieved.  There exist 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 NTP timestamps (wall clock) 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.

      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

16.44.  Scale

   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 normal play time increase at twice the wallclock rate.  For every



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   second of elapsed (wallclock) time, 2 seconds of content 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.

   Unless requested otherwise by the Speed parameter, the data rate
   SHOULD NOT be changed.  Implementation of scale changes depends on
   the server and media type.  For video, a server may, for example,
   deliver only key frames or selected key frames.  For audio, for
   example, it may time-scale the audio while preserving pitch or, less
   desirably, deliver fragments of audio.

   The server should try to approximate the viewing rate, but may
   restrict the range of scale values that it supports.  The response
   MUST 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 SHALL indicate this with the use of the "play.scale" feature-
   tags.

   When indicating a negative scale for a reverse playback, the Range
   header MUST indicate a decreasing range as described in
   Section 16.38.

   Example of playing in reverse at 3.5 times normal rate:
     Scale: -3.5
     Range: npt=15-10

16.45.  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 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 SHALL always include the
   header in any PLAY response for media with random access properties
   to indicate what policy was applied.  A Server that receives a
   unknown Seek-Style policy SHALL ignore it and select the server
   default policy.




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   This specification defines the following seek policies that may be
   requested:

   RAP:  Random Access Point (RAP) is the behavior of requesting the
      server to locate the closest previous random access point that
      exist 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.

   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 continous media
      that is media that will be render during the requested start time
      of the range.

   Next:  The next media units after the provided start time of the
      range.  For continous 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 is
      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 SHALL only be
      used on the client's explicit request.

   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 severly degraded unless additional
   data is available as, for example, already buffered, or through other
   side channels.

16.46.  Speed

   The Speed request-header field requests the server to deliver data to
   the client at a particular speed, contingent on the server's ability
   and desire to serve the media stream at the given speed.
   Implementation by the server is OPTIONAL.  The default is the bit
   rate of the stream.

   The parameter value is expressed as a decimal ratio, e.g., a value of



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   2.0 indicates that data is to be delivered twice as fast as normal.
   A speed of zero is invalid.  All speeds may not be possible to
   support.  Therefore the actual used speed MUST be included in the
   response.  The lack of a response header is indication of lack of
   support from the server of this functionality.  Support of the speed
   functionality are indicated by the "play.speed" feature-tag.

   Example:
     Speed: 2.5

   Use of this field changes the bandwidth used for data delivery.  It
   is meant for use in specific circumstances where preview of the
   presentation at a higher or lower rate is necessary.  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.  When data is delivered over UDP, it is highly
   recommended that means such as RTCP be used to track packet loss
   rates.  If the data transport is performed over non-dedicated best-
   effort networks the sender is required to perform congestion control
   of the stream(s).  This can result in that the communicated speed is
   impossible to maintain.

16.47.  Server

   See [H14.38], however the header syntax is corrected in
   Section 20.2.3.

16.48.  Session

   The Session request-header and response-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 exist until
   destroyed by a TEARDOWN 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 SHALL 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 SHALL NOT be included in DESCRIBE.  In an RTSP response
   the session header SHALL 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 SHALL also be
   included in the response, OPTIONS, GET_PARAMETER, and SET_PARAMETER,
   and SHALL NOT be included in DESCRIBE.




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   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) SHALL be returned if the session
   identifier is invalid.

16.49.  Supported

   The Supported header field enumerates all the extensions supported by
   the client or server using feature tags.  The header carries the
   extensions supported by the message sending entity.  The Supported
   header MAY be included in any request.  When present in a request,
   the receiver MUST respond with its corresponding Supported header.
   Note, also in 4xx and 5xx responses is the supported header included.

   The Supported header field contains a list of feature-tags, described
   in Section 4.7, that are understood by the client or server.

   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

16.50.  Timestamp

   The Timestamp general-header field 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 is used by
   the agent to compute the round-trip time to the responding agent so
   that it can adjust the timeout value for retransmissions.  It also
   resolves retransmission ambiguities for unreliable transport of RTSP.





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16.51.  Transport

   The Transport request and response header field 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 field 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.  Parameters may be added to
   each transport specififcation, separated by a semicolon.  The server
   MUST return a Transport response-header field in the response to
   indicate the values actually chosen if any.  If not transport
   specification is supported no transport header is returned and the
   request SHALL be responded using the status code 461 (Unsupported
   Transport) (Section 15.4.13).  In case more than one transport
   specification was present in the request, the server MUST return the
   single (transport-spec) which was actually chosen if any.  The number
   of transport-spec entries is expected to be limited as the client
   will get guidance on what configurations that are possible from the
   presentation description.

   The Transport header field MAY also be used in subsequent SETUP
   requests to change transport parameters.  A server MAY refuse to
   change parameters of an existing stream.

   A transport specification may only contain one of any given parameter
   within it.  Parameters MAY be given in any order.  Additionally, it
   may only contain either of the unicast or the multicast transport
   type parameter.  All parameters needs to be understood in a transport
   specification, if not the transport specification SHALL be ignored.
   RTSP proxies of any type that uses or modifies the transport
   specification, e.g. acces proxy or security proxy, SHALL remove
   specifications with unknown parameters before forwarding the RTSP
   message.  If that result in no remaing transport specification the
   proxy shall send a 461 (Unsupported Transport) (Section 15.4.13)
   response without any Transport header.

      The Transport header field 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 specifier is a list of slash
   separated tokens:



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   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 SHALL send media to same address for which the RTSP
         messages originates.  Does not work for transports requiring
         explicitly given destination ports.

   The choice of method for indicating where the media is to be
   delivered depends on the use case.  In some case 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:

   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 needs to be addressed if
         using this method because a RTSP server can be used as a DoS
         attacker on a existing multicast group.

   dest_addr using Session Decription Information:  The information
         included in the transport header can all be comming 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 SHALL include the information if it is
         available in the session description.

   No dest_addr:  The lack of an explicit multicast group request the
         server to decide the group address and its scope.  For this to
         work the server needs to have a context about what scope that
         works.  This method is currently under specified.

   An RTSP proxy will need to take care.  If the media is not desired to



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   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 needs 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 entity per transport spec is intended.  The usage
         of multiple destination to distribute a single media to
         multiple entities is unspecified.

         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 21.1) 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 SHALL return 463 (Destination
         Prohibited).

         The host address part of the tuple MAY be empty, for example
         ":58044", in cases when only destination port is desired to be
         specified.  Responses to request 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



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         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 streams 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 are 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 a address binding in
         the NAT that the sender to receiver packet flow can use.


               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.  Valid values are PLAY and RECORD.  If not
         provided, the default is PLAY.  The RECORD value was defined in
         RFC 2326 and is in this specification unspecified but reserved.

   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 $ statement and MUST be present.  This parameter
         MAY be specified as a range, 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 are 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



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         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 sends 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.

   Multicast-specific:

   ttl:  multicast time-to-live for IPv4.  When included in requests the
         value indicate the TTL value that the client request 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 function are not needed for IPv6 as
         the scoping is part of the address.

   RTP-specific:

   These parameters are 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 SHALL 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
         MAY ignore it, and choose appropriate SSRCs for the stream.
         The server MAY set the ssrc parameter in the Transport header
         of the response.

   The parameters defined below MAY only be used if the media transport
   protocol if the lower-level transport is connection-oriented (such as
   TCP).  However, these parameters MUST NOT be used when interleaving
   data over the RTSP control 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].  We
         discuss the use of this parameter in RTP/AVP/TCP non-



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         interleaved transport in Appendix C.2.2; the discussion below
         is limited to 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 setup parameter on the Transport line in
         a SETUP request, to indicate the SETUP request prefers the
         reuse of an existing connection between client and server (in
         which case the client sets the "connection" parameter to
         "existing"), or that the client requires 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.

         The default value of "connection" is "existing", for all SETUP
         requests (initial and subsequent).



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   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, UTC
           User-Agent: PhonyClient/1.2

     S->C: RTSP/2.0 200 OK
           CSeq: 302
           Date: Thu, 23 Jan 1997 15:35:06 GMT
           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

16.52.  Unsupported

   The Unsupported response-header field lists the features not
   supported by the server.  In the case where the feature was specified
   via the Proxy-Require field (Section 16.35), 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 SHALL NOT be forwarded.

   See Section 16.42 for a usage example.

16.53.  User-Agent

   See [H14.43] for explanation, however the syntax is clarified due to
   an error in RFC 2616.  A Client SHOULD include this header in all
   RTSP messages it sends.

16.54.  Vary

   See [H14.44].





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16.55.  Via

   See [H14.45].

16.56.  WWW-Authenticate

   See [H14.47].












































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17.  Proxies

   RTSP Proxies are RTSP agents that sit 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.  Proxies are also
   introduced for several different reasons and the below are often
   combined.

   Caching Proxy:  This type of proxy is used to reduce the workload on
         servers and connections.  By caching a presentation, both
         description and media streams the proxy can serve a client
         content without requesting it from the server once it has been
         cached and hasn't become stale.  See the caching Section 18.
         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.

   Translator Proxy:  This type of proxy is used to ensure that a RTSP
         client get 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 a RTSP
         client get 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 proxies are required to redirect the media to
         themselves or a controlled gateway that perform 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 request that the
         necessary pinholes in the firewall is opened when a client in
         the protected network want to access media streams on the
         external side.This proxy can also limit the clients access to
         certain type of content.  This proxy can perform its function
         without redirecting the media between the server and client.
         However, in deployements 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
         understand all aspects of how the media transport.




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   Auditing Proxy:  RTSP proxies can also provide network owners with a
         logging and audit point for RTSP sessions, e.g. for
         corporations that tracks 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 type of proxies can be used also 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 type of
   deployment, and instead secured sessions do by-pass of the proxies.

   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.  Thus resulting in blocking further development of
   RTSP and its usage.

17.1.  Proxies and Protocol Extensions

   The existence of proxies must always be considered when developing
   new RTSP extensions.  Most type of proxies will need to implement any
   new method to operate correct in the presence of that extension.  New
   headers will be possible to introduce without being blocked by
   proxies not yet updated.  However, it is important to consider if
   this header and its function is required to be understood by the
   proxy or can be forwarded.  If the header needs to be understood a
   feature-tag representing the functionality needs to be included in
   the Proxy-Require header.  Below are guidelines for analysis if the
   header needs to be understood.  The transport header and its
   parameters also shows that headers that are extensible and requires
   correct interpretation in the proxy also requires handling rules.

   When defining a new RTSP header it needs to be considered if RTSP
   proxies are required to understand them to achieve correct
   functionality.  Determining this is not easy as the functionality for
   proxies are widely varied as can be understood from the above list of
   functionality.  When evaluating this one can dived the functionality
   into three main categories:






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

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

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

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
































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18.  Caching

   In HTTP, response-request 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 behavior allowed to the cache is given by the cache-response
   directives described in Section 16.10.  A cache MUST answer any
   DESCRIBE requests if it is currently serving the stream to the
   requestor, as it is possible that low-level details of the stream
   description may have changed on the origin-server.

   Note that an RTSP cache, unlike the HTTP 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 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
   (that is, 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.







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19.  Security Framework

   The RTSP security framework consists of two high level components:
   the pure authentication mechanisms based on HTTP authentication, and
   the 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] and also in [H15].
   Servers SHOULD implement both basic and digest [RFC2617]
   authentication.  Client SHALL implement both basic and digest
   authentication [RFC2617] so that Server who 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.

19.2.  RTSP over TLS

   RTSP SHALL 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 URI level and 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 SHALL 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 SHALL 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" SHALL
   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 way 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.



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   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 roundtrips).
   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.

   In addition to these recommendations, Section 19.3 gives further
   recommendations of TLS usage with proxies.

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 can not 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 basically 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).  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 needs 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 transport layer).  In the rest of this section any
   reference to proxy will be to a non-transparent proxy, which inspects
   or manipulate 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.



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   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 have the possibility to select the desired
   behavior.  To handle this case, the Accept-Credentials header (See
   Section 16.2) is used, where the client can force the proxy/proxies
   to relay back the chain of certificates used to authenticate any
   intermediate proxies as well as the server.  Given the assumption
   that the proxies are viewed as trusted, it gives the user a
   possibility to enforce policies to each trusted proxy of whether it
   should accept the next entity in the chain.

   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 does not require to use it.  The proxy SHALL when
   initiating the next hop TLS connection use the incomming TLS
   connections CipherSuite list, only modified by removing any cipher
   suits 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) SHALL accept whatever
         certificate 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.  Reason 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.




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   If the Accept-Credentials header is not included in the RTSP request
   from the client, then the "Proxy" method SHALL be used as default.
   If an other method than the "Proxy" is to be used, then the Accept-
   Credentials header SHALL be included in all of the RTSP request from
   the client.  This is because it cannot be assumed that the proxy
   always keeps the TLS state or the users previously 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 respectively method.  If the policy do not accept the
   credentials of the next hop, the entity SHALL 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-Credential header.  As for the non-secured communication,
   the possibility for these request 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, if
   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 exist at the time when the server desire
   to send the request, it silently fails.

   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 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 SHALL include a Connection-Credentials
      header, see Section 16.12 with the next hop's identity and
      certificate.



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   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).  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, UTC
         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: 2
         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, UTC
   P->S: 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"
         Via: RTSP/2.0 proxy.example.org
         Accept-Credentials: User "rtsps://test.example.org";sha-256;
         dPYD7txpoGTbAqZZQJ+vaeOkyH4=
         Accept-Ranges: NPT, SMPTE, UTC

   One implication of this process is that the connection for secured
   RTSP messages may take significantly more round-trip times for the
   first message.  An 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 after the first message
   exchange the remaining message should not be delayed, if each message
   contains the chain of proxies that the requestor accepts.  The
   procedure of including the credentials in each request rather than
   building state in each proxy, avoids the need for revocation
   procedures.




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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.

20.1.  Base Syntax

   RTSP header field 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 linebreak.  The HCOLON
   defines this construct.

   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)
   DQ              =  %x22  ; US-ASCII double-quote mark (34)
   BACKSLASH       =  %x5C  ; US-ASCII backslash (92)
   CRLF            =  CR LF











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   LWS             =  [CRLF] 1*( SP / HT )
   SWS             =  [LWS] ; sep whitespace
   HCOLON          =  *( SP / HT ) ":" SWS
   TEXT            =  %x20-7E / %x80-FF  ; any OCTET except CTLs
   tspecials       =  "(" / ")" / "<" / ">" / "@"
                   /  "," / ";" / ":" / BACKSLASH / DQ
                   /  "/" / "[" / "]" / "?" / "="
                   /  "{" / "}" / 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   =  ( DQ *qdtext DQ )
   qdtext          =  %x20-21 / %x23-7E / %x80-FF  ; any TEXT except <">
   quoted-pair     =  BACKSLASH CHAR
   ctext           =  %x20-27 / %x2A-7E
                   /  %x80-FF  ; any OCTET except CTLs, "(" and ")"
   generic-param   =  token [ EQUAL gen-value ]
   gen-value       =  token / host / quoted-string

   safe            =  "$" / "-" / "_" / "." / "+"
   extra           =  "!" / "*" / "'" / "(" / ")" / ","
   rtsp-extra      =  "!" / "*" / "'" / "(" / ")"

   HEX             =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F"
                   /  "a" / "b" / "c" / "d" / "e" / "f"
   LHEX            =  DIGIT / %x61-66 ;lowercase a-f
   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 / "+" / "/"
















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   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
   LDQUOT   =  SWS DQ ; open double quotation mark
   RDQUOT   =  DQ SWS ; close double quotation mark
   RAQUOT   =  ">" SWS ; right angle quote
   LAQUOT   =  SWS "<" ; left angle quote

   TEXT-UTF8char    =  %x21-7E / UTF8-NONASCII
   UTF8-NONASCII    =  %xC0-DF 1UTF8-CONT
                    /  %xE0-EF 2UTF8-CONT
                    /  %xF0-F7 3UTF8-CONT
                    /  %xF8-FB 4UTF8-CONT
                    /  %xFC-FD 5UTF8-CONT
   UTF8-CONT        =  %x80-BF

   FLOAT            = ["-"] 1*39DIGIT ["." 1*46DIGIT]
   POS-FLOAT        = 1*39DIGIT ["." 1*46DIGIT]

20.2.  RTSP Protocol Definition

20.2.1.  Generic Protocol elements

























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   RTSP-IRI       =  schemes ":" IRI-rest
   IRI-rest       =  ihier-part [ "?" iquery ] [ "#" ifragment ]
   ihier-part     =  "//" iauthority ipath-abempty
   RTSP-IRI-ref   =  RTSP-IRI / irelative-ref
   irelative-ref  =  irelative-part [ "?" iquery ] [ "#" ifragment ]
   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 ":"

   ipchar         =  iunreserved / pct-encoded / sub-delims / ":" / "@"
   ipchar-nc      =  iunreserved / pct-encoded / sub-delims / "@"

   iquery         =  < As defined in RFC 3987>
   ifragment      =  < As defined in RFC 3987>
   iunreserved    =  < As defined in RFC 3987>
   pct-encoded    =  < As defined in RFC 3987>














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   RTSP-URI       =  schemes ":" URI-rest
   RTSP-URI-Ref   =  RTSP-URI / RTSP-Relative
   schemes        =  "rtsp" / "rtsps" / scheme
   scheme         =  < As defined in RFC 3986>
   URI-rest       =  hier-part [ "?" query ]
   hier-part      =  "//" authority path-abempty

   RTSP-Relative  =  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 ":"

   pchar          =  unreserved / pct-encoded / sub-delims / ":" / "@"
   pchar-nc       =  unreserved / pct-encoded / sub-delims / "@"

   sub-delims     =  "!" / "$" / "&" / "'" / "(" / ")"
                     / "*" / "+" / "," / "="













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   smpte-range        =  smpte-type "=" smpte-range-spec
                         ; See section 3.4
   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  =  token
   smpte-time         =  1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
                         [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]


   npt-range        =  "npt=" npt-range-spec
   npt-range-spec   =  ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
   npt-time         =  "now" / npt-sec / npt-hhmmss
   npt-sec          =  1*DIGIT [ "." *DIGIT ]
   npt-hhmmss       =  npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
   npt-hh           =  1*DIGIT   ; any positive number
   npt-mm           =  1*2DIGIT  ; 0-59
   npt-ss           =  1*2DIGIT  ; 0-59


   utc-range        =  "clock=" 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 [ "." fraction ]
   fraction         =  1*DIGIT


   feature-tag       =  token

   session-id        =  1*256( ALPHA / DIGIT / safe )

   extension-header  =   header-name HCOLON header-value
   header-name       =  token
   header-value      =  *(TEXT-UTF8char / UTF8-CONT / LWS)

20.2.2.  Message Syntax












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   RTSP-message  = Request / Response  ; RTSP/2.0 messages

   Request       = Request-Line
                   *((general-header
                   /  request-header
                   /  entity-header) CRLF)
                   CRLF
                   [ message-body ]

   Response     = Status-Line
                  *((general-header
                  /  response-header
                  /  entity-header) CRLF)
                  CRLF
                  [ message-body ]

   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-URI
   RTSP-Version =  "RTSP/" 1*DIGIT "." 1*DIGIT

   message-body = 1*OCTET

   Status-Code  =  "100"  ; Continue
                /  "200"  ; OK
                /  "300"  ; Multiple Choices
                /  "301"  ; Moved Permanently
                /  "302"  ; Found
                /  "303"  ; See Other
                /  "304"  ; Not Modified
                /  "305"  ; Use Proxy
                /  "400"  ; Bad Request
                /  "401"  ; Unauthorized



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                /  "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
                /  "411"  ; Length Required
                /  "412"  ; Precondition Failed
                /  "413"  ; Request Entity 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
                /  "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   =  *TEXT









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   general-header  =  Cache-Control
                   /  Connection
                   /  CSeq
                   /  Date
                   /  Media-Properties
                   /  Media-Range
                   /  Pipelined-Requests
                   /  Proxy-Supported
                   /  Seek-Style
                   /  Supported
                   /  Timestamp
                   /  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-Require
                   /  Range
                   /  Referer
                   /  Request-Status
                   /  Require
                   /  Scale
                   /  Session
                   /  Speed
                   /  Supported
                   /  Transport
                   /  User-Agent
                   /  extension-header













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   response-header  =  Accept-Credentials
                    /  Accept-Ranges
                    /  Connection-Credentials
                    /  ETag
                    /  Location
                    /  Proxy-Authenticate
                    /  Public
                    /  Range
                    /  Retry-After
                    /  RTP-Info
                    /  Scale
                    /  Session
                    /  Server
                    /  Speed
                    /  Transport
                    /  Unsupported
                    /  Vary
                    /  WWW-Authenticate
                    /  extension-header

   entity-header    =  Allow
                    /  Content-Base
                    /  Content-Encoding
                    /  Content-Language
                    /  Content-Length
                    /  Content-Location
                    /  Content-Type
                    /  Expires
                    /  Last-Modified
                    /  extension-header

20.2.3.  Header Syntax

   All header syntaxes not defined in this section are defined in
   section 14 of the HTTP 1.1 specification [RFC2616].

   Accept          =  "Accept" HCOLON
                      [ accept-range *(COMMA accept-range) ]
   accept-range    =  media-range *(SEMI accept-param)
   media-range     =  ( "*/*"
                      / ( m-type SLASH "*" )
                      / ( m-type SLASH m-subtype )
                       ) *( SEMI m-parameter )
   accept-param    =  ("q" EQUAL qvalue) / generic-param
   qvalue          =  ( "0" [ "." *3DIGIT ] )
                   /  ( "1" [ "." *3("0") ] )
   Accept-Credentials   =  "Accept-Credentials" HCOLON cred-decision
   cred-decision     =  ("User" [LWS cred-info])



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                     /  "Proxy"
                     /  "Any"
                     /  (token [LWS 1*TEXT]) ; For future extensions
   cred-info         =  cred-info-data *(COMMA cred-info-data)

   cred-info-data    =  DQ RTSP-URI DQ SEMI hash-alg SEMI base64
   hash-alg          =  "sha-256" / extension-alg
   extension-alg     =  token
   Accept-Encoding   =  "Accept-Encoding" HCOLON
                        [ encoding *(COMMA encoding) ]
   encoding          =  codings *(SEMI accept-param)
   codings           =  content-coding / "*"
   content-coding    =  token
   Accept-Language   =  "Accept-Language" HCOLON
                        [ language *(COMMA language) ]
   language          =  language-range *(SEMI accept-param)
   language-range    =  (1*8ALPHA *( "-" 1*8ALPHA)) / "*"
   Accept-Ranges     =  "Accept-Ranges" HCOLON acceptable-ranges
   acceptable-ranges =  (range-unit *(COMMA range-unit))
                     /  "none"
   range-unit        =  "NPT" / "SMPTE" / "UTC" / extension-format
   extension-format  =  token
   Allow             =  "Allow" HCOLON [Method *(COMMA Method)]
   Authorization     =  "Authorization" HCOLON credentials
   credentials       =  ("Digest" LWS digest-response)
                     /  other-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  =  Request-URI
                        ; by HTTP/1.1
   message-qop       =  "qop" EQUAL qop-value
   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)



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   auth-scheme       =  token

   Bandwidth         =  "Bandwidth" HCOLON 1*DIGIT

   Blocksize         =  "Blocksize" HCOLON 1*DIGIT

   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*DIGIT

   Connection-Credentials = "Connection-Credentials" HCOLON cred-chain
   cred-chain          =  DQ RTSP-URI DQ SEMI base64

   Connection         =  "Connection" HCOLON (connection-token)
                         *(COMMA connection-token)
   connection-token   =  token

   Content-Base       =  "Content-Base" HCOLON RTSP-URI-Ref
   Content-Encoding   =  "Content-Encoding" HCOLON
                         content-coding *(COMMA content-coding)
   Content-Language   =  "Content-Language" HCOLON
                         language-tag *(COMMA language-tag)
   language-tag       =  primary-tag *( "-" subtag )
   primary-tag        =  1*8ALPHA
   subtag             =  1*8ALPHA
   Content-Length     =  "Content-Length" HCOLON 1*DIGIT
   Content-Location   =  "Content-Location" HCOLON RTSP-URI-Ref
   Content-Type       =  ( "Content-Type" / "c" ) HCOLON media-type
   media-type         =  m-type SLASH m-subtype *(SEMI m-parameter)



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   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      =  rfc1123-date ; HTTP-date
   rfc1123-date   =  wkday "," SP date1 SP time SP "GMT"
   date1          =  2DIGIT SP month SP 4DIGIT
                     ; day month year (e.g., 02 Jun 1982)
   time           =  2DIGIT ":" 2DIGIT ":" 2DIGIT
                     ; 00:00:00 - 23:59:59
   wkday          =  "Mon" / "Tue" / "Wed"
                  /  "Thu" / "Fri" / "Sat" / "Sun"
   month          =  "Jan" / "Feb" / "Mar" / "Apr"
                  /  "May" / "Jun" / "Jul" / "Aug"
                  /  "Sep" / "Oct" / "Nov" / "Dec"

   ETag           =  "ETag" HCOLON entity-tag
   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-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 ("*" / entity-tag-list)
   entity-tag-list  =  entity-tag *(COMMA entity-tag)
   entity-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 ("*" / entity-tag-list)
   Last-Modified    =  "Last-Modified" HCOLON RTSP-date
   Location         =  "Location" HCOLON RTSP-URI
   Media-Properties = "Media-Properties" HCOLON media-prop-list



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   media-prop-list  = media-prop-value *(COMMA media-prop-value)
   media-prop-value = ("Random-Access" [EQUAL POS-FLOAT])
                    / "Begining-Only"
                    / "No-Seeking"
                    / "Unmutable"
                    / "Dynamic"
                    / "Time-Progressing"
                    / "Unlimited"
                    / ("Time-Limited" EQUAL utc-range-spec)
                    / ("Time-Duration" EQUAL POS-FLOAT)
                    / media-prop-ext
   media-prop-ext   = token [EQUAL (1*rtsp-unreserved / quoted-string)]
   Media-Range      = "Media-Range" HCOLON [ranges-list]
   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






























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  Proxy-Authenticate   =  "Proxy-Authenticate" HCOLON challenge-list
  challenge-list       =  challenge *(COMMA challenge)
  challenge            =  ("Digest" LWS digest-cln *(COMMA digest-cln))
                       /  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 URI
                          *(1*SP URI) RDQUOT
  URI                  =  RTSP-URI / RTSP-URI-Ref
  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-Require        =  "Proxy-Require" HCOLON feature-tag
                          *(COMMA feature-tag)

  Proxy-Supported      =  "Proxy-Supported" HCOLON feature-tag
                          *(COMMA feature-tag)

  Public               =  "Public" HCOLON Method *(COMMA Method)

  Range                =  "Range" HCOLON ranges-list [exec-time]
  ranges-list          =  ranges-spec *(COMMA ranges-spec)
  exec-time            =  SEMI "time" EQUAL utc-time
  ranges-spec          =  npt-range / utc-range / smpte-range
                       /  range-ext
  range-ext            =  extension-format "=" range-value
  range-value          =  1*(rtsp-unreserved / quoted-string / ":" )

  Referer              =  "Referer" HCOLON RTSP-URI-Ref
  Request-Status       =  "Request-Status" HCOLON status-info
  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 DQ Reason-Phrase DQ
  Require              =  "Require" HCOLON feature-tag-list
  feature-tag-list     =  feature-tag *(COMMA feature-tag)





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   RTP-Info         =  "RTP-Info" HCOLON rtsp-info-spec
                       *(COMMA rtsp-info-spec)
   rtsp-info-spec   =  stream-url 1*ssrc-parameter
   stream-url       =  "url" EQUAL DQ RTSP-URI-Ref DQ
   ssrc-parameter   =  LWS "ssrc" EQUAL ssrc HCOLON
                       ri-parameter *(SEMI ri-parameter)
   ri-parameter     =  ("seq" EQUAL 1*DIGIT)
                    /  ("rtptime" EQUAL 1*DIGIT)
                    /  generic-param

   Retry-After      =  "Retry-After" HCOLON delta-seconds
                       [ comment ] *( SEMI retry-param )
   retry-param      =  ("duration" EQUAL delta-seconds)
                    /  generic-param

   Scale            =  "Scale" HCOLON ["-"] 1*DIGIT [ "." *DIGIT ]
   Seek-Style       =  "Seek-Style" HCOLON Seek-S-values
   Seek-S-values    =  "RAP"
                    /  "First-Prior"
                    /  "Next"
                    /  Seek-S-value-ext
   Seek-S-value-ext =  token
   Speed            =  "Speed" HCOLON 1*DIGIT [ "." *DIGIT ]

   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 ]

   Supported        =  "Supported" HCOLON [feature-tag-list]


   Timestamp        =  "Timestamp" HCOLON timestamp-value LWS [delay]
   timestamp-value  =  *DIGIT [ "." *DIGIT ]
   delay            =  *DIGIT [ "." *DIGIT ]

   Transport        =  "Transport" HCOLON transport-spec
                       *(COMMA transport-spec)
   transport-spec   =  transport-id *tr-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)




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profile             = "AVP" / "SAVP" / "AVPF" / token
lower-transport     = "TCP" / "UDP" / token
tr-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 trn-param-ext)
                    / (SEMI "setup" EQUAL contrans-setup)
                    / (SEMI "connection" EQUAL contrans-con)
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
mode-spec           = ( DQ mode *(COMMA mode) DQ )
mode                = "PLAY" / token
addr-list           = quoted-addr *(SLASH quoted-addr)
quoted-addr         = DQ (host-port / extension-addr) DQ
host-port           = host [":" port]
                    / ":" port
extension-addr      = 1*qdtext
host                = < As defined in RFC 3986>
port                = < As defined in RFC 3986>





















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   Unsupported     = "Unsupported" HCOLON feature-tag-list

   User-Agent      = "User-Agent" HCOLON ( product / comment )
                   0*(LWS (product / comment))

   Vary            = "Vary" HCOLON ( "*" / field-name-list)
   field-name-list = field-name *(COMMA field-name)
   field-name      = token
   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-branch
                   / 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-branch      = "branch" EQUAL token
   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-URI

   a-range-def         =  "a=range:" ranges-spec CRLF

   a-etag-def          =  "a=etag:" entity-tag CRLF











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21.  Security Considerations

   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security considerations outlined in [H15]
   apply.  Specifically, please note the following:

   Abuse of Server Log Information:  RTSP and HTTP servers will
         presumably have similar logging mechanisms, and thus should be
         equally guarded in protecting the contents of those logs, thus
         protecting the privacy of the users of the servers.  See
         [H15.1.1] for HTTP server recommendations regarding server
         logs.

   Transfer of Sensitive Information:  There is no reason to believe
         that information transferred or controlled via RTSP may be any
         less sensitive than that normally transmitted via 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.

   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 [H15.5], 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 may of the same
         "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.






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   Location Headers and Spoofing:  If a single server supports multiple
         organizations that do not trust each another, then it needs to
         check the values of Location and 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
   RFC2068 [RFC2068], future HTTP specifications may provide additional
   guidance on security issues.

   The following are added considerations for RTSP implementations.

   Concentrated denial-of-service attack:  The protocol offers the
         opportunity for a remote-controlled denial-of-service attack.
         See Section 21.1.

   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 would affect unsuspecting clients.  The
         server SHOULD use a large, random and non-sequential session
         identifier to minimize the possibility of this kind of attack.
         However, unless the RTSP signalling always are confedentiality
         protected, e.g. using TLS, an on-path attacker will be able to
         hijack a session.  For real session security, client
         authentication needs to be performed.

   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.

   Stream issues:  RTSP only provides for stream control.  Stream
         delivery issues are not covered in this section, nor in the
         rest of this draft.  RTSP implementations will most likely rely
         on other protocols such as RTP, IP multicast, RSVP and IGMP,
         and should address security considerations brought up in those
         and other applicable specifications.

   Persistently suspicious behavior:  RTSP servers SHOULD return error
         code 403 (Forbidden) upon receiving a single instance of
         behavior which is deemed a security risk.  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 clients which are deemed to be in violation of



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         local security policy.

   Scope of Multicast:  If RTSP is used to control the transmission of
         media onto a multicast network it is need to consider the scope
         that delivery has.  RTSP supports the TTL Transport header
         parameter to indicate this scope.  However such scope control
         is risk as it may be set to large and distribute media beyond
         the intended scope.

   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 that one allows to connect through.  They are man
         in the middle and has 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.

21.1.  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 attackers
   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 mechanism that
   are acceptable in an increased generality are; verification of the
   client's identity, either against a database of known users using
   RTSP authentication mechanisms (preferably digest authentication or
   stronger); a list of addresses that accept to be media destinations,
   especially considering user identity; and media path based
   verification.

   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 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 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 ICE based NAT



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   traversal method described in [I-D.ietf-mmusic-rtsp-nat].  By using
   random passwords and username the probability of unintended
   indication as a valid media destination is very low.  If the server
   include in its STUN requests a cookie (consisting of random material)
   that is the destination echo back the solution is also safe against
   having a off-path attacker being able to spoof the STUN checks.
   Leaving this solution vulnerable only to on-path attackers that can
   see the STUN requests go to the target of attack.

   For delivery to multicast addresses there is need for another
   solution which is not specified here.








































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22.  IANA Considerations

   This section sets up a number of registries for RTSP 2.0 that should
   be maintained by IANA.  For each registry there is a description on
   what it is required to contain, what specification is needed when
   adding a entry with IANA, and finally the entries that this document
   needs to register.  See also the Section 2.3 "Extending RTSP".  There
   is also an IANA registration of two SDP attributes.

   The sections describing how to register an item uses some of the
   requirements level described in RFC 5226 [RFC5226], namely "First
   Come, First Served", "Expert Review, "Specification Required", and
   "Standards Action".

   A registration request to IANA MUST contain the following
   information:

   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

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.







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22.1.2.  Registering New Feature-tags with IANA

   The registering of feature-tags is done on a first come, first served
   basis.

   The name of the feature MUST follow these rules: The name may be of
   any length, but SHOULD be no more than twenty characters long.  The
   name MUST NOT contain any spaces, or control characters.  The
   registration SHALL indicate if the feature-tag applies to clients,
   servers, or proxies only or any combinations of these.  Any
   proprietary feature SHALL have as the first part of the name a vendor
   tag, which identifies the organization.

22.1.3.  Registered entries

   The following feature-tags are in this specification defined and
   hereby registered.  The change control belongs to the IETF.

   play.basic:  The minimal implementation for playback operations
         according to this specification.  Applies for both clients,
         servers and proxies.

   play.scale:  Support of scale operations for media playback.  Applies
         only for servers.

   play.speed:  Support of the speed functionality for playback.
         Applies only for servers.

22.2.  RTSP Methods

22.2.1.  Description

   What a method is, is described in section Section 13.  Extending the
   protocol with new methods allow for totally new functionality.

22.2.2.  Registering New Methods with IANA

   A new method MUST be registered through an IETF Standards Action.
   The reason is that new methods may radically change the protocols
   behavior and purpose.

   A specification for a new RTSP method MUST consist of the following
   items:

   o  A method name which follows the ABNF rules for methods.

   o  A clear specification on what action and response a request with
      the method will result in.  Which directions the method is used,



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      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 registered headers that
      are allowed to use with the method in request or/and response.

   o  Describe how the method relates to network proxies.

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.

22.3.  RTSP Status Codes

22.3.1.  Description

   A status code is the three digit numbers used to convey information
   in RTSP response messages, seeSection 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 can only be registered by an IETF Standards Action.
   A specification for a new status code MUST specify the following:

   o  The requested number.

   o  A description what the status code means and the expected behavior
      of the sender and receiver of the code.

22.3.3.  Registered Entries

   RFCXXX, registers the numbered status code defined in the ABNF entry
   "Status-Code" except "extension-code" 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
   entity.  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.




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22.4.2.  Registering New Headers with IANA

   Registrations in the registry can be done following the Expert Review
   policy.  A specification SHOULD be provided, preferable 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 specification SHOULD 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 16 in RFCXXXX are to be registered.

   Furthermore the following RTSP headers defined in other
   specifications are registered:

   o  x-wap-profile defined in [3gpp-26234].

   o  x-wap-profile-diff defined in [3gpp-26234].

   o  x-wap-profile-warning defined in [3gpp-26234].

   o  x-predecbufsize defined in [3gpp-26234].

   o  x-initpredecbufperiod defined in [3gpp-26234].

   o  x-initpostdecbufperiod defined in [3gpp-26234].

   o  3gpp-videopostdecbufsize defined in [3gpp-26234].

   o  3GPP-Link-Char defined in [3gpp-26234].

   o  3GPP-Adaptation defined in [3gpp-26234].





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   o  3GPP-QoE-Metrics defined in [3gpp-26234].

   o  3GPP-QoE-Feedback defined in [3gpp-26234].

   The use of "x-" is NOT RECOMMENDED but the above headers in the
   register list was 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

   In Section 19.3.1 three policies for how to handle certificates.
   Further policies may be defined and SHALL be registered with IANA
   using the following rules:

   o  Registering requires an IETF Standards Action

   o  A registration is required to name a contact person.

   o  Name of the policy.

   o  A describing text that explains how the policy works for handling
      the certificates.

   This specification registers the following values:

   Any

   Proxy

   User

22.5.2.  Accept-Credentials hash algorithms

   The Accept-Credentials header (See Section 16.2) 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 be an interoperability problem.  Therefore the
   bar for registering new algorithms is placed intentional high.

   Any registration of a new hash algorithm SHALL fulfill the following
   requirement:

   o  Follow the IETF Standards Action policy.




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   o  A definition of the algorithm and its identifier meeting the
      "token" ABNF requirement.

22.6.  Cache-Control  Cache Directive Extensions

   There exist a number of cache directives which can be sent in the
   Cache-Control header.  A registry for this cache directives SHALL be
   defined with the following rules:

   o  Registering requires an IETF Standards Action.

   o  A registration is required to contain a contact person.

   o  Name of the directive and a definition of the value, if any.

   o  Specification if it is an request or response directive.

   o  A describing text that explains how the cache directive is used
      for RTSP controlled media streams.

   This specification registers the following values:

   no-cache:

   public:

   private:

   no-transform:

   only-if-cached:

   max-stale:

   min-fresh:

   must-revalidate:

   proxy-revalidate:

   max-age:

22.7.  Media Properties








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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 was in mind when writing the specification are defined.
   However, it can be expected that further inovation will result in new
   use cases or media streams with properties not covered by the one
   specified here.  Thus new ones can be specified.  As new media
   properties may need 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 new media property SHALL fulfill the following
   requirements

   o  Follow the Specification Required policy and get the approval of
      the designated Expert.

   o  Have a ABNF definition of the media property value name that meets
      "media-prop-ext" definition

   o  A Contact Person for the Registration

   o  Description of all changes to the behavior of RTSP protocol as
      result of these changes.

22.7.3.  Registered Values

   This specification registers the 9 values listed in Section 16.29.

22.8.  Notify-Reason header

22.8.1.  Description

   Notify-Reason values are the way to indicate why a notification was
   sent.  It may also imply that certain headers shall or should be
   present required for the client to act upon the information the
   notification carries.  New notification behaviors do need to be
   described to result in interoperable usage, thus specification are
   required.

22.8.2.  Registration Rules

   Registrations for new Notify-Reason value SHALL fulfill the following
   requirements




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   o  Follow the Specification Required policy and get the approval of
      the designated Expert.

   o  Have a ABNF definition of the Notify reason value name that meets
      "Notify-Reason-extension" definition

   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.

22.8.3.  Registered Values

   This specification registers 3 values defined in the Notify-Reas-val
   ABNFSection 20.2.3:

   o  end-of-stream

   o  media-properties-update

   o  scale-change

22.9.  Range header formats

   The Range header allows for different range formats.  New ones may be
   registered, but moderation should be applied as it makes
   interoperability more difficult.  A registration SHALL fulfill the
   following requirements:

   o  Follow the Specification Required policy.

   o  A ABNF definition of the range format that fulfils the "range-ext"
      definition.

   o  A Contact person for the registration.

   o  Rules for how one handles the range when using a negative Scale.

22.10.  RTP-Info header parameters

22.10.1.  Desctiption

   The RTP-Info header (Section 16.43) 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 registration



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   requires specification required.

22.10.2.  Registration Rules

   Registrations for new Notify-Reason value SHALL fulfill the following
   requirements

   o  Follow the Specification Required policy and get the approval of
      the designated Expert.

   o  Have a ABNF definition that meets the "generic-param" definition

   o  A Contact Person for the Registration

22.10.3.  Registered Values

   This specification registers 2 parameter value pairs:

   o  seq

   o  rtptime

22.11.  Seek-Style Policies

22.11.1.  Description

   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.

22.11.2.  Registration Rules

   Registrations of new Seek-Style polcies SHALL fulfill the following
   requirements

   o  Follow the Specification Required policy.

   o  Have a ABNF definition of the Seek-Style policy name that meets
      "Seek-S-value-ext" definition

   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.





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22.11.3.  Registered Values

   This specification registers 3 values:

   o  RAP

   o  First-Prior

   o  Next

22.12.  Transport Header Registries

   The transport header contains a number of parameters which have
   possibilities for future extensions.  Therefore registries for these
   needs to be defined.

22.12.1.  Transport Protocol Specification

   A registry for the parameter transport-protocol specification SHALL
   be defined with the following rules:

   o  Registering uses the policy of Specification Required.

   o  A contact person or organization with address and email.

   o  A value definition that are following the ABNF syntax definition.

   o  A describing text that explains how the registered value are used
      in RTSP.

   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 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 Appendix C.1.

   RTP/AVPF/UDP:  the same as RTP/AVPF.






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   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 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 "[RFC5124] over UDP.  The usage is
         explained in RFC XXXX, appendix 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 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 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 Appendix C.2.2.

   RTP/SAVPF/TCP:  Use of the RTP[RFC3550] protocol for media transport
         in combination with the "[RFC5124] over TCP.  The usage is
         explained in RFC XXXX, appendix Appendix C.2.2.

22.12.2.  Transport modes

   A registry for the transport parameter mode SHALL be defined with the
   following rules:

   o  Registering requires an IETF Standards Action.

   o  A contact person or organization with address and email.

   o  A value definition that are following the ABNF token definition.

   o  A describing text that explains how the registered value are used
      in RTSP.

   This specification registers 1 value:




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   PLAY: See RFC XXXX.

22.12.3.  Transport Parameters

   A registry for parameters that may be included in the Transport
   header SHALL be defined with the following rules:

   o  Registering uses the Specification Required policy.

   o  A value definition that are following the ABNF token definition.

   o  A describing text that explains how the registered value are used
      in RTSP.

   This specification registers all the transport parameters defined in
   Section 16.51.

22.13.  URI Schemes

   This specification defines two URI schemes ("rtsp" and "rtsps") and
   reserves a third one ("rtspu").  Registrations are following RFC
   4395[RFC4395].

22.13.1.  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: Describing the
         resource for the purpose of configuring the receiving entity
         (DESCRIBE), configuring the delivery method and its addressing
         (SETUP), controlling the delivery (PLAY and PAUSE), reading or
         setting of resource related parameters (SET_PARAMETER and
         GET_PARAMETER, and termination of the session context created
         (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.




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   Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC
         2326), RTSP 2.0 (RFC XXXX)

   Interoperability considerations:  The change in URI syntax performed
         between RTSP 1.0 and 2.0 can create interoperability issues.

   Security considerations:  All the security threats identified in
         Section 7 of RFC 3986 applies 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, RFC 3986, RFC 3987, RFC XXXX

22.13.2.  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: Describing the
         resource for the purpose of configuring the receiving entity
         (DESCRIBE), configuring the delivery method and its addressing
         (SETUP), controlling the delivery (PLAY and PAUSE), reading or
         setting of resource related parameters (SET_PARAMETER and
         GET_PARAMETER, and termination of the session context created
         (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.

   Applications/protocols that use this URI scheme name:  RTSP 1.0 (RFC
         2326), RTSP 2.0 (RFC XXXX)







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   Interoperability considerations:  The change in URI syntax performed
         between RTSP 1.0 and 2.0 can create interoperability issues.

   Security considerations:  All the security threats identified in
         Section 7 of RFC 3986 applies 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, RFC 3986, RFC 3987, RFC XXXX

22.13.3.  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 unrelaible 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: Describing the resource for the purpose
         of configuring the receiving entity (DESCRIBE), configuring the
         delivery method and its addressing (SETUP), controlling the
         delivery (PLAY and PAUSE), reading or setting of resource
         related parameters (SET_PARAMETER and GET_PARAMETER, and
         termination of the session context created (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.

   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.






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   Security considerations:  All the security threats identified in
         Section 7 of RFC 3986 applies 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, RFC 3986, RFC 3987

22.14.  SDP attributes

   This specification defines three SDP [RFC4566] attributes that it is
   requested that IANA register.

   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
        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
        Values:             Absolute or Relative URIs.

        Attribute name:     etag
        Long form:          Entity 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







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22.15.  Media Type Registration for text/parameters

   Type name:  text

   Subtype name:  parameters

   Required parameters:

   Optional parameters:

   Encoding considerations:

   Security considerations:  This format may carry any type of
      parameters.  Some can clear have security requirements, like
      privacy, confidentiality or integrity requirements.  The format
      has no built in security protection.  For the usage it was defined
      the transport can be protected between server and client using
      TLS.  However, care must be take to consider if also the proxies
      are trusted with the parameters in case hop-by-hop security is
      used.  If stored as file in file system the necessary precautions
      needs to be taken in relation to the parameters requirements
      including object security such as S/MIME [RFC3851].

   Interoperability considerations:  This media type was mentioned as a
      fictional example in RFC 2326 but was not formally specified.
      This have resulted in usage of this media type which may not match
      its formal definition.

   Published specification:  RFC XXXX, Appendix E.

   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
      Westerlund (magnus.westerlund@ericsson.com)







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   Intended usage:   Common

   Restrictions on usage:   None

   Author:  Magnus Westerlund (magnus.westerlund@ericsson.com)

   Change controller:  IETF

   Addition Notes:










































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23.  References

23.1.  Normative References

   [3gpp-26234]
              Third Generation Partnership Project (3GPP), "Transparent
              end-to-end Packet-switched Streaming Service (PSS);
              Protocols and codecs; Technical Specification 26.234",
              December 2002.

   [FIPS-pub-180-2]
              National Institute of Standards and Technology (NIST),
              "Federal Information Processing Standards Publications
              (FIPS PUBS) 180-2: Secure Hash Standard", Augusti 2002.

   [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.

   [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.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",



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              RFC 3711, March 2004.

   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              August 2004.

   [RFC3851]  Ramsdell, B., "Secure/Multipurpose Internet Mail
              Extensions (S/MIME) Version 3.1 Message Specification",
              RFC 3851, July 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 115,
              RFC 4395, February 2006.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [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.

   [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.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for



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              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

   [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.

23.2.  Informative References

   [I-D.ietf-mmusic-rtsp-nat]
              Goldberg, J., Westerlund, M., and T. Zeng, "An Network
              Address Translator (NAT) Traversal mechanism for media
              controlled  by Real-Time Streaming Protocol (RTSP)",
              draft-ietf-mmusic-rtsp-nat-07 (work in progress),
              July 2008.

   [ISO.13818-1.2000]
              International Organization for Standardization,
              "Information technology - Generic coding of moving
              pictures and associated audio information: Systems", ISO/
              IEC 13818-1:2000, December 2000.

   [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.

   [ITU.H323.1996]
              International Telecommunications Union, "Visual telephone



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              systems and equipment for local area networks which
              provide a non-guaranteed quality of service", ITU-
              T Recommendation H.323, May 1996.

   [NOSSDAV-1997-1]
              Schulzrinne, H., "A comprehensive multimedia control
              architecture for the Internet", May 1997.

   [RFC1123]  Braden, R., "Requirements for Internet Hosts - Application
              and Support", STD 3, RFC 1123, October 1989.

   [RFC1305]  Mills, D., "Network Time Protocol (Version 3)
              Specification, Implementation", RFC 1305, March 1992.

   [RFC1644]  Braden, B., "T/TCP -- TCP Extensions for Transactions
              Functional Specification", RFC 1644, July 1994.

   [RFC1961]  McMahon, P., "GSS-API Authentication Method for SOCKS
              Version 5", RFC 1961, June 1996.

   [RFC2068]  Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
              Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1",
              RFC 2068, January 1997.

   [RFC2070]  Yergeau, F., Nicol, G., Adams, G., and M. Duerst,
              "Internationalization of the Hypertext Markup Language",
              RFC 2070, January 1997.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [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.

   [RFC3388]  Camarillo, G., Eriksson, G., Holler, J., and H.
              Schulzrinne, "Grouping of Media Lines in the Session
              Description Protocol (SDP)", RFC 3388, December 2002.

   [RFC4145]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in
              the Session Description Protocol (SDP)", RFC 4145,
              September 2005.

   [Stevens98]



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              Stevens, W., "Unix Networking Programming - Volume 1,
              second edition", 1998.

   [W3C.REC-PICS-labels]
              Miller, J., Krauskopf, T., Resnick, P., and W. Treese,
              "PICS label distribution label syntax and communication
              protocols", W3C REC-PICS-labels-961031.

   [W3C.REC-PICS-services]
              Miller, J., Resnick, P., and D. Singer, "Rating services
              and rating systems (and their machine readable
              descriptions)", W3C REC-PICS-services-961031,
              October 1996.






































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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
   this is examples and the normative and syntax description in the
   other sections takes precedence.  Please also note that many of the
   example contain syntax illegal line breaks to accommodate the
   formatting restriction that the RFC series impose.

A.1.  Media on Demand (Unicast)

   The 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 is 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.

   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.








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   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: Thu, 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 271
         Content-Base: rtsp://example.com/twister.3gp/
         Expires: 24 Jan 1997 15:35:06 GMT

         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=0-0:10:34.10
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control: trackID=1
         m=video 0 RTP/AVP 26
         a=control: trackID=4

























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   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, UTC

   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="192.0.2.5:9000"/"192.0.2.5:9001"
         Session: 12345678
         Expires: 24 Jan 1997 15:35:12 GMT
         Date: 23 Jan 1997 15:35:12 GMT
         Accept-Ranges: NPT

   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, UTC

   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="192.0.2.5:9002"/"192.0.2.5:9003";

         Session: 12345678
         Expires: 24 Jan 1997 15:35:13 GMT
         Date: 23 Jan 1997 15:35:13 GMT
         Accept-Range: NPT














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   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 4
         User-Agent: PhonyClient/1.2
         Range: npt=0-10, npt=30-
         Session: 12345678

   M->C: RTSP/2.0 200 OK
         CSeq: 4
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:14 GMT
         Session: 12345678
         Range: npt=0-10, npt=30-623.10
         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

   M->C: RTSP/2.0 200 OK
         CSeq: 5
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:01 GMT
         Session: 12345678
         Range: npt=34.57-623.10

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 6
         User-Agent: PhonyClient/1.2
         Range: npt=34.57-623.10
         Session: 12345678

   M->C: RTSP/2.0 200 OK
         CSeq: 6
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:01 GMT
         Session: 12345678
         Range: npt=34.57-623.10
         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






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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: Thu, 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 271
         Content-Base: rtsp://example.com/twister.3gp/
         Expires: 24 Jan 1997 15:35:06 GMT

         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=0-0:10:34.10
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control: trackID=1
         m=video 0 RTP/AVP 26
         a=control: trackID=4

   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, UTC
         Pipelined-Requests: 7654




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   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, UTC
         Pipelined-Requests: 7654

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 4
         User-Agent: PhonyClient/1.2
         Range: npt=0-10, npt=30-
         Session: 12345678
         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="192.0.2.5:9000"/"192.0.2.5:9001";
                    ssrc=93CB001E
         Session: 12345678
         Expires: 24 Jan 1997 15:35:12 GMT
         Date: 23 Jan 1997 15:35:12 GMT
         Accept-Ranges: NPT
         Pipelined-Requests: 7654

   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="192.0.2.5:9002"/"192.0.2.5:9003";
                    ssrc=A813FC13
         Session: 12345678
         Expires: 24 Jan 1997 15:35:13 GMT
         Date: 23 Jan 1997 15:35:13 GMT
         Accept-Range: NPT
         Pipelined-Requests: 7654

   M->C: RTSP/2.0 200 OK
         CSeq: 4
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:14 GMT
         Session: 12345678
         Range: npt=0-10, npt=30-623.10
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"



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            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.  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.
































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   C->W: GET /twister.sdp HTTP/1.1
         Host: www.example.com
         Accept: application/sdp

   W->C: HTTP/1.0 200 OK
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 278
         Expires: 23 Jan 1998 15:35:06 GMT

         v=0
         o=- 2890844526 2890842807 IN IP4 192.0.2.5
         s=RTSP Session
         e=adm@example.com
         c=IN IP4 0.0.0.0
         a=range:npt=0-1:49:34
         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, UTC

   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="192.0.2.5:5000"/"192.0.2.5:5001"
         Date: 23 Jan 1997 15:35:12 GMT
         Server: PhonyServer/1.0
         Expires: 24 Jan 1997 15:35:12 GMT
         Cache-Control: public
         Accept-Ranges: NPT, SMPTE











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   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, UTC

   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="192.0.2.5:5002"/"192.0.2.5:5003"
         Date: 23 Jan 1997 15:35:12 GMT
         Server: PhonyServer/1.0
         Cache-Control: public
         Expires: 24 Jan 1997 15:35:12 GMT
         Accept-Ranges: NPT, SMPTE

   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
         RTP-Info: url="rtsp://video.example.com/twister/video"
                   ssrc=A17E189D:seq=12312232;rtptime=78712811
         Server: PhonyServer/2.0
         Date: 23 Jan 1997 15:35:13 GMT

















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   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
         RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
                   ssrc=3D124F01:seq=876655;rtptime=1032181
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:13 GMT



   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: 23 Jan 1997 15:36:52 GMT

   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: 23 Jan 1997 15:36:52 GMT


   Even though the audio and video track are on two different servers,
   may start at slightly different times, and may drift with respect to
   each other, the client can perform initial synchronize 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.







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A.4.  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 are an example of how a
   multi-stream server might expect a single-stream file to be served:

   C->S: DESCRIBE rtsp://foo.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.com/test.wav/
         Content-type: application/sdp
         Content-length: 163
         Server: PhonyServer/1.0
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Expires: 23 Jan 1997 17:00:00 GMT

         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



















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   C->S: SETUP rtsp://foo.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, UTC

   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="192.0.2.5:6970"/"192.0.2.5:6971";
             mode="PLAY";ssrc=EAB98712
         CSeq: 2
         Session: 2034820394
         Expires: 23 Jan 1997 16:00:00 GMT
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:07 GMT
         Accept-Ranges: NPT


   C->S: PLAY rtsp://foo.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: 23 Jan 1997 15:35:08 GMT
         Session: 2034820394
         Range: npt=0-600
         RTP-Info: url="rtsp://foo.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 that 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:

   C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2



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A.5.  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>
           ...
           <href "Streamed Live Music performance"
              src="rtsp://live.example.com/concert/audio">
           ...
         </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: Thu, 23 Jan 1997 15:35:06 GMT
         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 224.2.0.1/16
         a=control: rtsp://live.example.com/concert/audio
         a=range:npt=0-










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   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0
         CSeq: 2
         Transport: RTP/AVP;multicast
         Accept-Ranges: NPT, SMPTE, UTC
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Date: Thu, 23 Jan 1997 15:35:06 GMT
         Transport: RTP/AVP;multicast;
              dest_addr="224.2.0.1:3456"/"224.2.0.1:3457";ttl=16
         Session: 0456804596
         Accept-Ranges: NPT, UTC


   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: 23 Jan 1997 15:35:07 GMT
         Session: 0456804596
         Range:npt=1256-
         RTP-Info: url="rtsp://live.example.com/concert/audio"
                   ssrc=0D12F123:seq=1473; rtptime=80000

A.6.  Capability Negotiation

   This examples illustrate how the client and server determines 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 playback (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.








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   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
         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
   is 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, UTC
         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="192.0.2.5:5000"/"192.0.2.5:5001"
         Server: PhonyServer/2.0
         Accept-Ranges: NPT, SMPTE


















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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.

   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 a normative description of the protocols
   behavior.  However, in case of ambiguity with the earlier parts of
   this specification, the description in the earlier parts SHALL take
   precedence.

B.1.  States

   The state machine contains three states, described below.  For each
   state there exist a table which shows which requests and events that
   is allowed and if they will result in a state change.

   Init: Initial state no session exist.

   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 is 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 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.






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   IFI:  IF Implemented.

   md:   Media

   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 is 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
   is requested for an event includes: response codes, e.g. 200, headers
   that MUST 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 valid request meeting the requisites is normally a
   2xx (SUCCESS) unless other noted in the response column.  The
   exceptions needs 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 occur the appropriate response code MUST 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 exist restrictions to when a 3rr
   response may be used.  A 5xx response SHALL NOT result in any change
   of the session state, except if the error is not possible to recover
   from.  A unrecoverable error SHALL result 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.

   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 exist a timeout event for all states
   except Init.



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   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 MUST 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
   allows 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.



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   +--------------+-----------------+-----------+----------------------+
   | Action       | Requisite       | New State | Response             |
   +--------------+-----------------+-----------+----------------------+
   | 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     | Play      | Play from RP         |
   |              | range           |           |                      |
   |              |                 |           |                      |
   | PLAY         | Prs URI, Range  | Play      | According to range   |
   |              |                 |           |                      |
   | PAUSE        | Prs URI         | Ready     | Return PP            |
   |              |                 |           |                      |
   | SC:REDIRECT  | Range hdr       | Ready     | Set RedP             |
   |              |                 |           |                      |
   | SC:REDIRECT  | no range hdr    | Init      | Session is removed   |
   |              |                 |           |                      |
   | Timeout      |                 | Init      |                      |
   |              |                 |           |                      |
   | RedP reached |                 | Init      | TEARDOWN of 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 transport
   parameters.  TEARDOWN is depending on both the Request-URI and the
   number of media stream 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 exist in the
   session, the session will remain and a session header MUST be
   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



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   a media stream determines the new state.

   +--------------+-----------------+-----------+----------------------+
   | Action       | Requisite       | New State | Response             |
   +--------------+-----------------+-----------+----------------------+
   | PAUSE        | PrsURI          | Ready     | Set RP to present    |
   |              |                 |           | point                |
   |              |                 |           |                      |
   | PP reached   |                 | Ready     | RP = PP              |
   |              |                 |           |                      |
   | End of media | All media       | Play      | Set RP = End of      |
   |              |                 |           | media                |
   |              |                 |           |                      |
   | End of range |                 | Play      | Set RP = End of      |
   |              |                 |           | range                |
   |              |                 |           |                      |
   | PLAY         | Prs URI, No     | Play      | Play from present    |
   |              | range           |           | point                |
   |              |                 |           |                      |
   | PLAY         | Prs URI, Range  | Play      | According to range   |
   |              |                 |           |                      |
   | PLAY_NOTIFY  |                 | Play      | 200                  |
   |              |                 |           |                      |
   | SETUP        | New URI         | Play      | 455                  |
   |              |                 |           |                      |
   | SETUP        | Setuped URI     | Play      | 455                  |
   |              |                 |           |                      |
   | SETUP        | Setuped URI,    | Play      | Change transport     |
   |              | IFI             |           | 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  | Range hdr       | Play      | Set RedP             |
   |              |                 |           |                      |
   | SC:REDIRECT  | no range hdr    | Init      | Session is removed   |
   |              |                 |           |                      |
   | RedP reached |                 | Init      | TEARDOWN of session  |
   |              |                 |           |                      |
   | Timeout      |                 | Init      | Stop Media playout   |
   +--------------+-----------------+-----------+----------------------+

                           Table 16: State: Play




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   The Play state table, see Table 16, is the largest.  The table
   contains an number of requests that has presentation URI as a
   prerequisite on 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 remain in Play state.  An explicit PAUSE request MUST be sent
   to change the state to Ready.  It may appear that there exist an
   automatic transitions in "RedP reached" and "PP reached", however
   they are requested and acknowledge before they take place.  The time
   at which the transition will happen is known by looking at the range
   header.  If the client sends request close in time to these
   transitions it needs to be prepared for getting error message as the
   state may or may not have changed.


































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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
   signalling with regards to RTP.

   The available RTP profiles and lower layer transports are described
   below along with rules on signalling the available combinations.

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 include the "interleaved" parameter.

   When using embedded binary data the "src_addr" and "dest_addr" SHALL
   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].  This profiles 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.  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



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   MAY be defined in future.

   The "src_addr" and "dest_addr" parameters are used in the following
   way for media playback, i.e.  Mode=PLAY:

   o  The "src_addr" and "dest_addr" parameters MUST contain either 1 or
      2 address specifications.

   o  Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST
      contain either:

      *  both an address and a port number, or

      *  a port number without an address.

   o  The first address and port pair given in either of the parameters
      applies to the RTP stream.  The second address and port pair if
      present applies to the RTCP stream.

   o  The RTP/UDP packets from the server to the client SHALL be sent to
      the address and port given by first address and port pair of the
      "dest_addr" parameter.

   o  The RTCP/UDP packets from the server to the client SHALL be sent
      to the address and port given by the second address and port pair
      of the "dest_addr" parameter.  If no second pair is specified RTCP
      SHALL NOT be sent.

   o  The RTCP/UDP packets from the client to the server SHALL be sent
      to the address and port given by the second address and port pair
      of the "src_addr" parameter.  If no second pair is given RTCP
      SHALL NOT be sent.

   o  The RTP/UDP packets from the client to the server SHALL be sent to
      the address and port given by the first address and port pair of
      the "src_addr" parameter.

   o  RTP and RTCP Packets SHOULD be sent from the corresponding
      receiver port, i.e.  RTCP packets from server should be sent from
      the "src_addr" parameters second address port pair.

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 session using RTP.
   All that is defined for AVP SHALL also apply for AVPF.

   The usage of AVPF is indicated by the media initialization protocol



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   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 that shall be used that
   SHALL be followed.

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 SHALL also apply for SAVP.

   The usage of SRTP requires that a security association is
   established.  The RECOMMENDED mechanism for establishing that
   security association is to use MIKEY with RTSP as defined in RFC 4567
   [RFC4567].

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 AVP SHALL also
   apply for SAVPF.

   The usage of SRTP requires that a security association is
   established.  The RECOMMENDED mechanism for establishing that
   security association is to use MIKEY[RFC3830] with RTSP as defined in
   RFC 4567 [RFC4567].

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 SHALL be supported if an RTSP
   agent handles RTP.

C.1.6.1.  Media synchronization

   RTCP provides media synchronization and clock drift compensation.
   The first is available from RTP-Info header to accomplish the initial
   synchronization.  But to be able to handle any clockdrift 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 SHALL function
   as keep-alive.  Which 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.



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C.1.6.3.  Bit-rate adapation

   RTCP Receiver reports and any additional feedback from the client
   SHALL be used adapt the bit-rate used over the transport for all
   cases when RTP is sent over UDP.  A RTP sender without reserved
   resources SHALL 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.

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
   control connection.  In both cases the protocol SHALL be "rtp" and
   the lower layer SHALL 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.

   One should however consider that this will result that 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, we describe 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 ports (or two address/
   port pairs) are specified by the dest_addr parameter.  If the client
   wishes to use RTP without RTCP, one port (or one address/port pair)
   is specified by the dest_addr parameter.  Ordering rules of dest_addr



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   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 16.51) 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 for all dest_addr values 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 port value for RTP
   MUST be set to the TCP port number on which the client is expecting
   to receive the RTP stream connection, and the dest_addr port value
   for RTCP MUST be set to the TCP port number on which the client is
   expecting to receive the RTCP stream connection.

   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 can 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 RTP
   connection and (if the client requested an RTCP connection by
   specifying two dest_addr ports or address/port pairs) and RTCP
   connection.  If a server sets "setup" to "active", the ports
   specified in "src_addr" MUST be set to 9.  The server MAY use the
   "ssrc" parameter, following the guidance in Section 16.51.  Port
   ordering for src_addr follows the rules for RTP/AVP/UDP.

   For cases when servers have a public IP-address it is RECOMMENDED
   that the server take the passive role and the client the active role.
   This help in cases when the client is behind a NAT.

   After sending (receiving) a 2xx reply for a SETUP method for a non-



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   interleaved RTP/AVP/TCP media stream, the active party SHOULD
   initiate the TCP connection as soon as possible.  The client SHALL
   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 SHALL respond using the 464
   "Data Transport Not Ready Yet" (Section 15.4.16) 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
   client over the RTP TCP connection, and RTCP packets flow
   bidirectionally over the RTCP TCP connection.  As in the RTP/UDP
   case, client to server traffic on the TCP port is unspecified by this
   memo.  The packets that travel on these connections SHALL be framed
   using the protocol defined in [RFC4571], not by the framing defined
   for interleaving RTP over the RTSP control 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 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 RTP and RTCP connections for the URI, or does the client wish to
   continue using the existing TCP RTP and RTCP connections?  The client
   SHOULD use the "connection" parameter (defined in Section 16.51) on
   the Transport line to make its intention clear in the regard (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.

   Below, we rewrite part of the example media on demand example shown
   in Appendix A.1 to use RTP/AVP/TCP non-interleaved:












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      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: Thu, 23 Jan 1997 15:35:06 GMT
            Content-Type: application/sdp
            Content-Length: 227
            Content-Base: rtsp://example.com/twister.3gp/
            Expires: 24 Jan 1997 15:35:06 GMT

            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=0-0:10:34.10
            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




















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            Accept-Ranges: NPT, SMPTE, UTC

      M->C: RTSP/2.0 200 OK
            CSeq: 2
            Server: PhonyServer/1.0
            Transport: RTP/AVP/TCP;unicast;
                       dest_addr=":9"/":9";
                       src_addr="192.0.2.5:9000"/"192.0.2.5:9001"
                       setup=passive;connection=new;ssrc=93CB001E
            Session: 12345678
            Expires: 24 Jan 1997 15:35:12 GMT
            Date: 23 Jan 1997 15:35:12 GMT
            Accept-Ranges: NPT

      C->M: TCP Connection Establishment

      C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
            CSeq: 4
            User-Agent: PhonyClient/1.2
            Range: npt=0-10, npt=30-
            Session: 12345678

      M->C: RTSP/2.0 200 OK
            CSeq: 4
            Server: PhonyServer/1.0
            Date: 23 Jan 1997 15:35:14 GMT
            Session: 12345678
            Range: npt=0-10, npt=30-623.10
            RTP-Info:  url="rtsp://example.com/twister.3gp/trackID=1"
               ssrc=4F312DD8:seq=54321;rtptime=2876889

C.2.3.  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].  Such control allows jumps to be created in
   media clock (e.g.  NPT) timeline of the RTSP session.  The simple
   case of jumps in media clock time is caused by multiple ranges in the
   range specifier of a PLAY request.  This allows the media layer
   rendering the RTP stream without being affected by jumps in media
   clock time.  The RTP timestamps for the next media segment is set so
   that they become continuous with the previous media segment in the
   request.  The RTP sequence number for the first packet in the new
   segment will be the next following the last packet in the previous
   segment, 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 multiple ranges in a PLAY request for media that has random-



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   access properties.  It is also possible when one PLAY request
   replaces another ongoing for media with random-access properties.  In
   both cases care to align frames or similar media dependent structures
   are need but likely possible.

   In cases where jumps in media clock time are a result of RTSP
   signalling operations arriving during or after an PLAY operation, the
   processing delay or request timing can result in that media becomes
   non-continuous.  The server becomes unable to send the media so that
   it arrive 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,
   like live streams, any media clock jump is commonly result of
   correspondingly long pause of delivery.  The RTP timestamp will have
   increased in direct proportion to the duration of the paused
   delivery.  Not 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.

   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 SHALL represent the duration the delivery
   was halted.  RTP sequence number MUST be the next number.  A client
   can not rely on that a server will align when resuming playing even
   if it is RECOMMENDED.  The RTP-Info header will provide information
   if this have occurred or not.

      We 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.




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   As an example, assume a 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: 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 = 49, rtptime = 39200, NPT time = 14.9s


   Immediately after the end 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: 5
         Session: abcdefg
         Range: npt=18-20
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 5
         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



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   In this example, first, NPT 10 through 15 is played, then the client
   request 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).  To avoid this gap in playback due to the time it takes
   to perform RTSP requests, a PLAY request with multiple ranges needs
   to be specified.  That would result in the following example:

      C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15;npt=18-20
       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 = 49, rtptime = 39200, NPT time = 14.9s
      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


C.2.4.  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



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   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

   The client then sends a PAUSE request:











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   C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0
         CSeq: 5
         Session: abdcdefg
         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 take
   15ms 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.







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C.2.5.  RTSP / RTP Integration

   For certain datatypes, 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.2.6.  Scaling with RTP

   For scaling (see Section 16.44), RTP timestamps should correspond to
   the playback timing.  For example, when playing video recorded at 30
   frames/second at a scale of two and speed (Section 16.46) 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.2.7.  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 16.43) header provides the first sequence number of the next
   segment.

C.2.8.  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.2.9.  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.

C.2.10.  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 SHALL NOT send a 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



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   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 force the media sender to change its SSRC in
   accordance with the RTP specification[RFC3550].  This 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
   as the server.  Any SSRC signalled in the Transport header SHOULD be
   avoided.  A client detecting a collision prior to sending any RTP or
   RTCP messages can also select a new SSRC.

C.3.  Future Additions

   It is the intention that any future protocol or profile regarding
   both for 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 of profile for use with RTSP:

   o  The protocol or profile needs to define a name tag representing
      it.  This tag is required to be a 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.  May need new headers similar to RTP info to
      carry information.

   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.








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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 an 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.  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).  However the
   SDP extension "Grouping of Media Lines in the Session Description
   Protocol (SDP)" [RFC3388] may provide such functionality depending on
   need.  Also future grouping semantics may in the future be developed.

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 (RFC 4566 [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 SHALL be different from
   any media level URI.  The presence of a session level control
   attribute SHALL be interpreted as support for aggregated control.
   The control attribute SHALL be present on media level unless the
   presentation only contains a single media stream, in which case the
   attribute MAY only be present on the session level.

   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 SHALL look for a base URI in the following order:

   1.  the RTSP Content-Base field;





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   2.  the RTSP Content-Location field;

   3.  the RTSP Request-URI.

   If this attribute contains only an asterisk (*), then the URI SHALL
   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 lets assume that a container file has the
   URI: "rtsp://example.com/container.mp4".  Lets further assume this
   URI is the base URI, and that there is a 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 an 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.

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 number is 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.



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   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 what 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.

   Usage of grouping of media lines [RFC3388] to determine which media
   lines should or should not be included in a RTSP session is
   unspecified.

   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 RFC 3551 (Sections 5 and 6), or an MIME type
   registered with IANA, or an experimental encoding as specified in SDP
   (RFC 4566 [RFC4566]).  Codec-specific parameters are not specified in
   this field, but rather in the "fmtp" attribute described below.

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
   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"
   provides instructions on which 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 for media
   streams delivered from the RTSP server to the client would be given
   the "a=recvonly" attribute.



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   The direction attributes are not commonly used in SDPs for RTSP, but
   may occur. "a=recvonly" in a SDP provided to the RTSP client SHALL
   indicate that media delivery will only occur in the direction from
   the RTSP server to the client.  In SDP provided to the RTSP client
   that lacks any of the directionality attributes (a=recvonly,
   a=sendonly, a=sendrecv) SHALL behave as if the "a=recvonly" attribute
   was received.  Note that this overrules the normal default rule
   defined in SDP[RFC4566].  The usage of "a=sendonly" or "a=sendrecv"
   is not defined, nor is the interpretation of SDP by other entities
   than the RTSP client.

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, 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 contains media streams of the same durations, the
   range attribute SHOULD only be used at session-level.  In case of
   different length 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 SHALL 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 SHALL take care to provide RTSP Range (see Section 16.38)
   values that are consistent with what is presented in the SDP for the
   content.  There are 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, Section 4.5 and
   Section 4.6 and MAY be extended with further formats.  Any open ended
   range (start-), i.e. without stop range, is of unspecified duration



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   and SHALL 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 MUST contain suitable values for the start and stop
   times for both aggregate and non-aggregate stream control.  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, the "c=" field contains the destination address for the media
   stream.  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 SHALL be "0.0.0.0",
   and for type "IP6", this value SHALL be "0:0:0:0:0:0:0:0", i.e. the
   unspecified address according to RFC 4291 [RFC4291].

D.1.9.  Entity Tag

   The optional "a=etag" 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 16.24) 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.



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   ABNF for the attribute is defined in Section 20.3.

   Example:
     a=etag:"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:
   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.com/movie.aud
   m=audio 8004 RTP/AVP 3
   a=control:rtsp://video.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.com and video.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



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   above.

   Example:
   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: Thu, 23 Jan 1997 15:35:06 GMT
         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 required 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 issues 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.  RTSP external SDP delivery

   There are some considerations that needs to be made when the session
   description is delivered to client outside of RTSP, for example in
   HTTP or email.

   First of all the SDP needs to contain absolute URIs, relative will in
   most cases not work as the delivery will not correctly forward the
   base URI.  And as SDP might be temporarily stored on file system
   before being loaded into an RTSP capable client, thus if possible to



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   transport the base URI it still would need to be merged into the
   file.

   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 are 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].









































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Appendix E.  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 SHALL only use US-ASCII
   visable characters while the values are UTF-8 text strings.  The
   media type registration template is in Section 22.15.

   There exist 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 vaue 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    =  %x21-7E / UTF8-NONASCII
   UTF8-NONASCII    =  %xC0-DF 1UTF8-CONT
                    /  %xE0-EF 2UTF8-CONT
                    /  %xF0-F7 3UTF8-CONT
                    /  %xF8-FB 4UTF8-CONT
                    /  %xFC-FD 5UTF8-CONT
   UTF8-CONT        =  %x80-BF
   WSP              = <See RFC 5234> ; Space or HTAB
   VCHAR            = <See RFC 5234>
   CRLF             = <See RFC 5234>















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Appendix F.  Requirements for Unreliable Transport of RTSP

   This section provides anyone intending to define how to transport of
   RTSP messages over a unreliable transport protocol with some
   information learned by the attempt in RFC 2326 [RFC2326].  RFC 2326
   define both an URI scheme and some basic functionality for transport
   of RTSP messages over UDP, however it was not sufficient for reliable
   usage and successful interoperability.

   The RTSP scheme defined for unreliable transport of RTSP messages was
   "rtspu".  It has been reserved by this specification as at least one
   commercial implementation exist, thus avoiding any collisions in the
   name space.

   The following considerations should exist for operation of RTSP over
   an unreliable transport protocol:

   o  Request 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 1123)
      [RFC1123], 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  If RTSP is used over a small-RTT LAN, standard procedures for
      optimizing initial TCP round trip estimates, such as those used in
      T/TCP (RFC 1644) [RFC1644], can be beneficial.

   o  The Timestamp header (Section 16.50) is used to avoid the
      retransmission ambiguity problem [Stevens98].

   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.



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   There exist two RTSP headers thats primarily are intended for being
   used by the unreliable handling of RTSP messages and which will be
   maintained:

   o  [CSeq] See Section 16.19

   o  [Timestamp] See Section 16.50












































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Appendix G.  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 exist no requirement to implement RTSP
   1.0, in fact we recommend against it as it is difficult to do in an
   interoperable way.

   A server implementing RTSP/2.0 MUST include a RTSP-Version of
   RTSP/2.0 in all responses to requests containing RTSP-Version
   RTSP/2.0.  If a server receives a RTSP/1.0 request, it MAY respond
   with a RTSP/1.0 response if it chooses to support RFC 2326.  If the
   server chooses not to support RFC 2326, it SHALL respond with a 505
   (RTSP Version not supported) status code.  A server MUST NOT respond
   to a RTSP-Version RTSP/1.0 request with a 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 a 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.

G.1.  Play Request in Play mode

   The behavior in the server when a Play is received in Play mode 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
   take effect immediately replacing the previous request.

G.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.













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Appendix H.  Open Issues

   Open issues are filed and tracked in the bug and feature trackers at
   http://rtspspec.sourceforge.net.  Open issues are discussed on MMUSIC
   list.














































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Appendix I.  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 extension parameters are to be
         ignored.

      *  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 has been resolved.

      *  Syntax error with ";" for multicast and unicast has been
         resolved.

      *  Two new addressing parameters has been defined, src_addr and
         dest_addr.  These replaces 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 use 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 extend to allow multiple values for clients



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         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 a initial NPT identifier that
         must now be used.

      *  All formats now support initial open ended formats of type
         "npt=-10".

   o  RTSP message handling has been changed in the following way:

      *  RTSP messages now uses URIs rather then URLs.

      *  It has been clarified that a 4xx message due to missing CSeq
         header shall be returned without a CSeq header.

      *  Rules for how to handle timing out RTSP messages has been
         added.

      *  Extended Pipelining rules allowing for quick session startup.

   o  The HTTP references has been updated to RFC 2616 and RFC 2617.
      This has resulted in that the Public, and the Content-Base header
      needed to be 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 completely been rewritten.  It
      includes now more details and are also more clear about the model
      used.

   o  A IANA section has been included with contains a number of
      registries and their rules.  This will allow us to use IANA to
      keep track of RTSP extensions.

   o  Than 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.





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      *  The rules for how TCP connections is 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 has also been added.

      *  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 proxies 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.

      *  Changed the Range header to allow multiple ranges for creating
         editing list.

      *  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 1
         has also been added.

      *  Added requirement that the Date header must be used for all
         messages with entity 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 was also added.

      *  The Supported header was borrowed from SIP to help with the
         feature negotiation in RTSP.




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      *  Clarified that the Timestamp header can be used to resolve
         retransmission ambiguities.

      *  The Session header text has been expanded with a 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 is used to
         indicate pause points in the PAUSE response.

      *  Clarified that RTP-Info URIs that are relative, uses the
         Request-URI as base URI.  Also clarified that used URI must be
         that one that was used in the SETUP request.  They 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 structured borrowed
         from SIP.  Those tables carries much more information and
         should provide a good overview of the available headers.

      *  It has been is 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.

      *  To resolve functionality around ETag.  The ETag and If-None-
         Match header has 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.






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   o  The Protocol Syntax has been changed in the following way:

      *  All BNF definitions are updated according to the rules defined
         in RFC 5234 [RFC5234] and has been gathered in a separate
         Section 20.

      *  The BNF for the User-Agent and Server headers has been
         corrected so now only the description is in the HTTP
         specification.

      *  Some definitions in the introduction regarding the RTSP session
         has been changed.

      *  The protocol has been made fully IPv6 capable.  Certain of the
         functionality, like using explicit IPv6 addresses in fields
         requires that the protocol support this updated specification.

      *  Added a fragment part to the RTSP URI.  This seem to be
         indicated by the note below the definition however it was not
         part of the BNF.

      *  The CHAR rule has been changed to exclude NULL.

   o  The Status codes has 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 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.

   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.



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      *  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
         it exist is defined here and some requirements for the the
         transport is provided.

   o  The following changes has been made in relation to methods:

      *  The OPTIONS method has been clarified with regards to the use
         of the Public and Allow headers.

      *  The RECORD and ANNOUNCE methods are removed as they are lacking
         implementation and not considered necessary in the core
         specification.  Any work on these methods should be done as a
         extension document to RTSP.

      *  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.

   o  Wrote a new section about how to setup different media transport
      alternatives and their profiles, and lower layer protocols.  This
      resulted that the appendix on RTP interaction was moved there
      instead in the part describing RTP.  The section also includes
      guidelines what to think of 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.




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   o  Text specifying the special behavior of PLAY for live content.

   o  Added a new method PLAY_NOTIFY.  This method is used by the RTSP
      server to asynchronously notify clients about session changes.















































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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 this RFC are Henning Schulzrinne, Anup Rao, and Robert
   Lanphier.

   Both RTSP version 1.0 and RTSP versio 2.0 borrow format and
   descriptions from HTTP/1.1.

   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, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
   Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari,
   Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V.
   Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
   John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets,
   Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
   Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal
   Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov,
   Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith,
   Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen
   Chesire, David Walker, Geetha Srikantan, Stephan Wenger, Pekka Pessi,
   Jae-Hwan Kim, Holger Schmidt, Stephen Farrell, Xavier Marjou, Joe
   Pallas and Mela Martti.

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.

   o  Sean Sheedy contributed text on the timeout behavior of RTSP
      messages and connections, and the 463 status code.

   o  Fredrik Lindholm contributed text about the RTSP security
      framework.

   o  John Lazzaro contributed the text for RTP over Independent TCP.





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   o  Aravind Narasimhan contributed by rewriting Media Transport
      Alternatives (Appendix C) and editorial improvements on a number
      of places in the specification.
















































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Appendix K.  RFC Editor Consideration

   Please replace RFC XXXX with the RFC number this specification
   recieves.















































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Authors' Addresses

   Henning Schulzrinne
   Columbia University
   1214 Amsterdam Avenue
   New York, NY  10027
   USA

   Email: schulzrinne@cs.columbia.edu


   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: stiemerling@nw.neclab.eu









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Full Copyright Statement

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   contained in BCP 78, and except as set forth therein, the authors
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