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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
Obsoletes: RFC 2326                                               A. Rao
(if approved)                                                      Cisco
Intended status: Standards Track                             R. Lanphier
Expires: December 21, 2009
                                                           M. Westerlund
                                                             Ericsson AB
                                                    M. Stiemerling (Ed.)
                                                                     NEC
                                                           June 19, 2009


                Real Time Streaming Protocol 2.0 (RTSP)
                    draft-ietf-mmusic-rfc2326bis-21

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
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   This Internet-Draft will expire on December 21, 2009.



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Copyright Notice

   Copyright (c) 2009 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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Abstract

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

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


Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .  11
     1.1.   Notes on Copyright . . . . . . . . . . . . . . . . . . .  11
   2.  Protocol Overview . . . . . . . . . . . . . . . . . . . . . .  13
     2.1.   Content Description  . . . . . . . . . . . . . . . . . .  13
     2.2.   Session Establishment  . . . . . . . . . . . . . . . . .  14
     2.3.   Media Delivery Control . . . . . . . . . . . . . . . . .  15
     2.4.   Session Parameter Manipulations  . . . . . . . . . . . .  17
     2.5.   Media Delivery . . . . . . . . . . . . . . . . . . . . .  17
       2.5.1.   Media Delivery Manipulations . . . . . . . . . . . .  18
     2.6.   Session Maintenance and Termination  . . . . . . . . . .  20
     2.7.   Extending RTSP . . . . . . . . . . . . . . . . . . . . .  21
   3.  Document Conventions  . . . . . . . . . . . . . . . . . . . .  23
     3.1.   Notational Conventions . . . . . . . . . . . . . . . . .  23
     3.2.   Terminology  . . . . . . . . . . . . . . . . . . . . . .  23
   4.  Protocol Parameters . . . . . . . . . . . . . . . . . . . . .  27
     4.1.   RTSP Version . . . . . . . . . . . . . . . . . . . . . .  27
     4.2.   RTSP IRI and URI . . . . . . . . . . . . . . . . . . . .  27
     4.3.   Session Identifiers  . . . . . . . . . . . . . . . . . .  29
     4.4.   SMPTE Relative Timestamps  . . . . . . . . . . . . . . .  29
     4.5.   Normal Play Time . . . . . . . . . . . . . . . . . . . .  30
     4.6.   Absolute Time  . . . . . . . . . . . . . . . . . . . . .  31
     4.7.   Feature-Tags . . . . . . . . . . . . . . . . . . . . . .  31
     4.8.   Message Body Tags  . . . . . . . . . . . . . . . . . . .  31
     4.9.   Media Properties . . . . . . . . . . . . . . . . . . . .  32
       4.9.1.   Random Access  . . . . . . . . . . . . . . . . . . .  33
       4.9.2.   Retention  . . . . . . . . . . . . . . . . . . . . .  33
       4.9.3.   Content Modifications  . . . . . . . . . . . . . . .  33
       4.9.4.   Supported Scale Factors  . . . . . . . . . . . . . .  34
       4.9.5.   Mapping to the Attributes  . . . . . . . . . . . . .  34
   5.  RTSP Message  . . . . . . . . . . . . . . . . . . . . . . . .  35



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     5.1.   Message Types  . . . . . . . . . . . . . . . . . . . . .  35
     5.2.   Message Headers  . . . . . . . . . . . . . . . . . . . .  36
     5.3.   Message Body . . . . . . . . . . . . . . . . . . . . . .  36
     5.4.   Message Length . . . . . . . . . . . . . . . . . . . . .  37
   6.  General Header Fields . . . . . . . . . . . . . . . . . . . .  38
   7.  Request . . . . . . . . . . . . . . . . . . . . . . . . . . .  39
     7.1.   Request Line . . . . . . . . . . . . . . . . . . . . . .  39
     7.2.   Request Header Fields  . . . . . . . . . . . . . . . . .  41
   8.  Response  . . . . . . . . . . . . . . . . . . . . . . . . . .  43
     8.1.   Status-Line  . . . . . . . . . . . . . . . . . . . . . .  43
       8.1.1.   Status Code and Reason Phrase  . . . . . . . . . . .  43
     8.2.   Response Headers . . . . . . . . . . . . . . . . . . . .  46
   9.  Message Body  . . . . . . . . . . . . . . . . . . . . . . . .  49
     9.1.   Message Body Header Fields . . . . . . . . . . . . . . .  49
     9.2.   Message Body . . . . . . . . . . . . . . . . . . . . . .  50
   10. Connections . . . . . . . . . . . . . . . . . . . . . . . . .  51
     10.1.  Reliability and Acknowledgements . . . . . . . . . . . .  51
     10.2.  Using Connections  . . . . . . . . . . . . . . . . . . .  52
     10.3.  Closing Connections  . . . . . . . . . . . . . . . . . .  54
     10.4.  Timing Out Connections and RTSP Messages . . . . . . . .  55
     10.5.  Showing Liveness . . . . . . . . . . . . . . . . . . . .  55
     10.6.  Use of IPv6  . . . . . . . . . . . . . . . . . . . . . .  56
   11. Capability Handling . . . . . . . . . . . . . . . . . . . . .  57
   12. Pipelining Support  . . . . . . . . . . . . . . . . . . . . .  59
   13. Method Definitions  . . . . . . . . . . . . . . . . . . . . .  60
     13.1.  OPTIONS  . . . . . . . . . . . . . . . . . . . . . . . .  61
     13.2.  DESCRIBE . . . . . . . . . . . . . . . . . . . . . . . .  62
     13.3.  SETUP  . . . . . . . . . . . . . . . . . . . . . . . . .  64
       13.3.1.  Changing Transport Parameters  . . . . . . . . . . .  67
     13.4.  PLAY . . . . . . . . . . . . . . . . . . . . . . . . . .  68
       13.4.1.  General Usage  . . . . . . . . . . . . . . . . . . .  68
       13.4.2.  Aggregated Sessions  . . . . . . . . . . . . . . . .  72
       13.4.3.  Updating current PLAY Requests . . . . . . . . . . .  73
       13.4.4.  Playing On-Demand Media  . . . . . . . . . . . . . .  74
       13.4.5.  Playing Dynamic On-Demand Media  . . . . . . . . . .  75
       13.4.6.  Playing Live Media . . . . . . . . . . . . . . . . .  75
       13.4.7.  Playing Live with Recording  . . . . . . . . . . . .  76
       13.4.8.  Playing Live with Time-Shift . . . . . . . . . . . .  76
     13.5.  PLAY_NOTIFY  . . . . . . . . . . . . . . . . . . . . . .  77
       13.5.1.  End-of-Stream  . . . . . . . . . . . . . . . . . . .  78
       13.5.2.  Media-Properties-Update  . . . . . . . . . . . . . .  79
       13.5.3.  Scale-Change . . . . . . . . . . . . . . . . . . . .  80
     13.6.  PAUSE  . . . . . . . . . . . . . . . . . . . . . . . . .  81
     13.7.  TEARDOWN . . . . . . . . . . . . . . . . . . . . . . . .  84
       13.7.1.  Client to Server . . . . . . . . . . . . . . . . . .  84
       13.7.2.  Server to Client . . . . . . . . . . . . . . . . . .  85
     13.8.  GET_PARAMETER  . . . . . . . . . . . . . . . . . . . . .  86
     13.9.  SET_PARAMETER  . . . . . . . . . . . . . . . . . . . . .  87



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     13.10. REDIRECT . . . . . . . . . . . . . . . . . . . . . . . .  89
   14. Embedded (Interleaved) Binary Data  . . . . . . . . . . . . .  92
   15. Status Code Definitions . . . . . . . . . . . . . . . . . . .  94
     15.1.  Success 1xx  . . . . . . . . . . . . . . . . . . . . . .  94
       15.1.1.  100 Continue . . . . . . . . . . . . . . . . . . . .  94
     15.2.  Success 2xx  . . . . . . . . . . . . . . . . . . . . . .  94
       15.2.1.  200 OK . . . . . . . . . . . . . . . . . . . . . . .  94
     15.3.  Redirection 3xx  . . . . . . . . . . . . . . . . . . . .  94
       15.3.1.  301 Moved Permanently  . . . . . . . . . . . . . . .  95
       15.3.2.  302 Found  . . . . . . . . . . . . . . . . . . . . .  95
       15.3.3.  303 See Other  . . . . . . . . . . . . . . . . . . .  95
       15.3.4.  304 Not Modified . . . . . . . . . . . . . . . . . .  95
       15.3.5.  305 Use Proxy  . . . . . . . . . . . . . . . . . . .  96
     15.4.  Client Error 4xx . . . . . . . . . . . . . . . . . . . .  96
       15.4.1.  400 Bad Request  . . . . . . . . . . . . . . . . . .  96
       15.4.2.  401 Unauthorized . . . . . . . . . . . . . . . . . .  96
       15.4.3.  402 Payment Required . . . . . . . . . . . . . . . .  97
       15.4.4.  403 Forbidden  . . . . . . . . . . . . . . . . . . .  97
       15.4.5.  404 Not Found  . . . . . . . . . . . . . . . . . . .  97
       15.4.6.  405 Method Not Allowed . . . . . . . . . . . . . . .  97
       15.4.7.  406 Not Acceptable . . . . . . . . . . . . . . . . .  97
       15.4.8.  407 Proxy Authentication Required  . . . . . . . . .  98
       15.4.9.  408 Request Timeout  . . . . . . . . . . . . . . . .  98
       15.4.10. 410 Gone . . . . . . . . . . . . . . . . . . . . . .  98
       15.4.11. 411 Length Required  . . . . . . . . . . . . . . . .  98
       15.4.12. 412 Precondition Failed  . . . . . . . . . . . . . .  99
       15.4.13. 413 Request Message Body Too Large . . . . . . . . .  99
       15.4.14. 414 Request-URI Too Long . . . . . . . . . . . . . .  99
       15.4.15. 415 Unsupported Media Type . . . . . . . . . . . . .  99
       15.4.16. 451 Parameter Not Understood . . . . . . . . . . . .  99
       15.4.17. 452 reserved . . . . . . . . . . . . . . . . . . . .  99
       15.4.18. 453 Not Enough Bandwidth . . . . . . . . . . . . . . 100
       15.4.19. 454 Session Not Found  . . . . . . . . . . . . . . . 100
       15.4.20. 455 Method Not Valid in This State . . . . . . . . . 100
       15.4.21. 456 Header Field Not Valid for Resource  . . . . . . 100
       15.4.22. 457 Invalid Range  . . . . . . . . . . . . . . . . . 100
       15.4.23. 458 Parameter Is Read-Only . . . . . . . . . . . . . 100
       15.4.24. 459 Aggregate Operation Not Allowed  . . . . . . . . 100
       15.4.25. 460 Only Aggregate Operation Allowed . . . . . . . . 100
       15.4.26. 461 Unsupported Transport  . . . . . . . . . . . . . 101
       15.4.27. 462 Destination Unreachable  . . . . . . . . . . . . 101
       15.4.28. 463 Destination Prohibited . . . . . . . . . . . . . 101
       15.4.29. 464 Data Transport Not Ready Yet . . . . . . . . . . 101
       15.4.30. 465 Notification Reason Unknown  . . . . . . . . . . 101
       15.4.31. 470 Connection Authorization Required  . . . . . . . 101
       15.4.32. 471 Connection Credentials not accepted  . . . . . . 102
       15.4.33. 472 Failure to establish secure connection . . . . . 102
     15.5.  Server Error 5xx . . . . . . . . . . . . . . . . . . . . 102



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       15.5.1.  500 Internal Server Error  . . . . . . . . . . . . . 102
       15.5.2.  501 Not Implemented  . . . . . . . . . . . . . . . . 102
       15.5.3.  502 Bad Gateway  . . . . . . . . . . . . . . . . . . 102
       15.5.4.  503 Service Unavailable  . . . . . . . . . . . . . . 102
       15.5.5.  504 Gateway Timeout  . . . . . . . . . . . . . . . . 103
       15.5.6.  505 RTSP Version Not Supported . . . . . . . . . . . 103
       15.5.7.  551 Option not supported . . . . . . . . . . . . . . 103
   16. Header Field Definitions  . . . . . . . . . . . . . . . . . . 104
     16.1.  Accept . . . . . . . . . . . . . . . . . . . . . . . . . 113
     16.2.  Accept-Credentials . . . . . . . . . . . . . . . . . . . 114
     16.3.  Accept-Encoding  . . . . . . . . . . . . . . . . . . . . 114
     16.4.  Accept-Language  . . . . . . . . . . . . . . . . . . . . 115
     16.5.  Accept-Ranges  . . . . . . . . . . . . . . . . . . . . . 116
     16.6.  Allow  . . . . . . . . . . . . . . . . . . . . . . . . . 116
     16.7.  Authorization  . . . . . . . . . . . . . . . . . . . . . 117
     16.8.  Bandwidth  . . . . . . . . . . . . . . . . . . . . . . . 118
     16.9.  Blocksize  . . . . . . . . . . . . . . . . . . . . . . . 118
     16.10. Cache-Control  . . . . . . . . . . . . . . . . . . . . . 118
     16.11. Connection . . . . . . . . . . . . . . . . . . . . . . . 121
     16.12. Connection-Credentials . . . . . . . . . . . . . . . . . 121
     16.13. Content-Base . . . . . . . . . . . . . . . . . . . . . . 122
     16.14. Content-Encoding . . . . . . . . . . . . . . . . . . . . 122
     16.15. Content-Language . . . . . . . . . . . . . . . . . . . . 123
     16.16. Content-Length . . . . . . . . . . . . . . . . . . . . . 124
     16.17. Content-Location . . . . . . . . . . . . . . . . . . . . 124
     16.18. Content-Type . . . . . . . . . . . . . . . . . . . . . . 124
     16.19. CSeq . . . . . . . . . . . . . . . . . . . . . . . . . . 125
     16.20. Date . . . . . . . . . . . . . . . . . . . . . . . . . . 125
     16.21. Expires  . . . . . . . . . . . . . . . . . . . . . . . . 126
     16.22. From . . . . . . . . . . . . . . . . . . . . . . . . . . 127
     16.23. If-Match . . . . . . . . . . . . . . . . . . . . . . . . 127
     16.24. If-Modified-Since  . . . . . . . . . . . . . . . . . . . 128
     16.25. If-None-Match  . . . . . . . . . . . . . . . . . . . . . 128
     16.26. Last-Modified  . . . . . . . . . . . . . . . . . . . . . 129
     16.27. Location . . . . . . . . . . . . . . . . . . . . . . . . 129
     16.28. Media-Properties . . . . . . . . . . . . . . . . . . . . 130
     16.29. Media-Range  . . . . . . . . . . . . . . . . . . . . . . 132
     16.30. MTag . . . . . . . . . . . . . . . . . . . . . . . . . . 132
     16.31. Notify-Reason  . . . . . . . . . . . . . . . . . . . . . 133
     16.32. Pipelined-Requests . . . . . . . . . . . . . . . . . . . 133
     16.33. Proxy-Authenticate . . . . . . . . . . . . . . . . . . . 134
     16.34. Proxy-Authorization  . . . . . . . . . . . . . . . . . . 134
     16.35. Proxy-Require  . . . . . . . . . . . . . . . . . . . . . 135
     16.36. Proxy-Supported  . . . . . . . . . . . . . . . . . . . . 135
     16.37. Public . . . . . . . . . . . . . . . . . . . . . . . . . 136
     16.38. Range  . . . . . . . . . . . . . . . . . . . . . . . . . 137
     16.39. Referer  . . . . . . . . . . . . . . . . . . . . . . . . 138
     16.40. Retry-After  . . . . . . . . . . . . . . . . . . . . . . 139



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     16.41. Request-Status . . . . . . . . . . . . . . . . . . . . . 139
     16.42. Require  . . . . . . . . . . . . . . . . . . . . . . . . 139
     16.43. RTP-Info . . . . . . . . . . . . . . . . . . . . . . . . 140
     16.44. Scale  . . . . . . . . . . . . . . . . . . . . . . . . . 142
     16.45. Seek-Style . . . . . . . . . . . . . . . . . . . . . . . 143
     16.46. Speed  . . . . . . . . . . . . . . . . . . . . . . . . . 145
     16.47. Server . . . . . . . . . . . . . . . . . . . . . . . . . 145
     16.48. Session  . . . . . . . . . . . . . . . . . . . . . . . . 146
     16.49. Supported  . . . . . . . . . . . . . . . . . . . . . . . 147
     16.50. Terminate-Reason . . . . . . . . . . . . . . . . . . . . 147
     16.51. Timestamp  . . . . . . . . . . . . . . . . . . . . . . . 148
     16.52. Transport  . . . . . . . . . . . . . . . . . . . . . . . 148
     16.53. Unsupported  . . . . . . . . . . . . . . . . . . . . . . 155
     16.54. User-Agent . . . . . . . . . . . . . . . . . . . . . . . 155
     16.55. Vary . . . . . . . . . . . . . . . . . . . . . . . . . . 156
     16.56. Via  . . . . . . . . . . . . . . . . . . . . . . . . . . 156
     16.57. WWW-Authenticate . . . . . . . . . . . . . . . . . . . . 157
   17. Proxies . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
     17.1.  Proxies and Protocol Extensions  . . . . . . . . . . . . 159
   18. Caching . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
     18.1.  Validation Model (HTTP)  . . . . . . . . . . . . . . . . 162
       18.1.1.  Last-Modified Dates  . . . . . . . . . . . . . . . . 163
       18.1.2.  Message Body Tag Cache Validators  . . . . . . . . . 163
       18.1.3.  Weak and Strong Validators . . . . . . . . . . . . . 163
       18.1.4.  Rules for When to Use Entity Tags and
                Last-Modified Dates  . . . . . . . . . . . . . . . . 166
       18.1.5.  Non-validating Conditionals  . . . . . . . . . . . . 167
     18.2.  Invalidation After Updates or Deletions (HTTP) . . . . . 167
   19. Security Framework  . . . . . . . . . . . . . . . . . . . . . 169
     19.1.  RTSP and HTTP Authentication . . . . . . . . . . . . . . 169
     19.2.  RTSP over TLS  . . . . . . . . . . . . . . . . . . . . . 169
     19.3.  Security and Proxies . . . . . . . . . . . . . . . . . . 170
       19.3.1.  Accept-Credentials . . . . . . . . . . . . . . . . . 171
       19.3.2.  User approved TLS procedure  . . . . . . . . . . . . 172
   20. Syntax  . . . . . . . . . . . . . . . . . . . . . . . . . . . 174
     20.1.  Base Syntax  . . . . . . . . . . . . . . . . . . . . . . 174
     20.2.  RTSP Protocol Definition . . . . . . . . . . . . . . . . 176
       20.2.1.  Generic Protocol elements  . . . . . . . . . . . . . 176
       20.2.2.  Message Syntax . . . . . . . . . . . . . . . . . . . 179
       20.2.3.  Header Syntax  . . . . . . . . . . . . . . . . . . . 183
     20.3.  SDP extension Syntax . . . . . . . . . . . . . . . . . . 192
   21. Security Considerations . . . . . . . . . . . . . . . . . . . 193
     21.1.  Remote denial of Service Attack  . . . . . . . . . . . . 195
   22. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 197
     22.1.  Feature-tags . . . . . . . . . . . . . . . . . . . . . . 197
       22.1.1.  Description  . . . . . . . . . . . . . . . . . . . . 197
       22.1.2.  Registering New Feature-tags with IANA . . . . . . . 198
       22.1.3.  Registered entries . . . . . . . . . . . . . . . . . 198



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     22.2.  RTSP Methods . . . . . . . . . . . . . . . . . . . . . . 198
       22.2.1.  Description  . . . . . . . . . . . . . . . . . . . . 198
       22.2.2.  Registering New Methods with IANA  . . . . . . . . . 198
       22.2.3.  Registered Entries . . . . . . . . . . . . . . . . . 199
     22.3.  RTSP Status Codes  . . . . . . . . . . . . . . . . . . . 199
       22.3.1.  Description  . . . . . . . . . . . . . . . . . . . . 199
       22.3.2.  Registering New Status Codes with IANA . . . . . . . 199
       22.3.3.  Registered Entries . . . . . . . . . . . . . . . . . 199
     22.4.  RTSP Headers . . . . . . . . . . . . . . . . . . . . . . 199
       22.4.1.  Description  . . . . . . . . . . . . . . . . . . . . 199
       22.4.2.  Registering New Headers with IANA  . . . . . . . . . 200
       22.4.3.  Registered entries . . . . . . . . . . . . . . . . . 200
     22.5.  Accept-Credentials . . . . . . . . . . . . . . . . . . . 201
       22.5.1.  Accept-Credentials policies  . . . . . . . . . . . . 201
       22.5.2.  Accept-Credentials hash algorithms . . . . . . . . . 201
     22.6.  Cache-Control  Cache Directive Extensions  . . . . . . . 202
     22.7.  Media Properties . . . . . . . . . . . . . . . . . . . . 202
       22.7.1.  Description  . . . . . . . . . . . . . . . . . . . . 203
       22.7.2.  Registration Rules . . . . . . . . . . . . . . . . . 203
       22.7.3.  Registered Values  . . . . . . . . . . . . . . . . . 203
     22.8.  Notify-Reason header . . . . . . . . . . . . . . . . . . 203
       22.8.1.  Description  . . . . . . . . . . . . . . . . . . . . 203
       22.8.2.  Registration Rules . . . . . . . . . . . . . . . . . 203
       22.8.3.  Registered Values  . . . . . . . . . . . . . . . . . 204
     22.9.  Range header formats . . . . . . . . . . . . . . . . . . 204
     22.10. Terminate-Reason Header  . . . . . . . . . . . . . . . . 204
       22.10.1. Redirect Reasons . . . . . . . . . . . . . . . . . . 204
       22.10.2. Terminate-Reason Header Parameters . . . . . . . . . 205
     22.11. RTP-Info header parameters . . . . . . . . . . . . . . . 205
       22.11.1. Description  . . . . . . . . . . . . . . . . . . . . 205
       22.11.2. Registration Rules . . . . . . . . . . . . . . . . . 205
       22.11.3. Registered Values  . . . . . . . . . . . . . . . . . 205
     22.12. Seek-Style Policies  . . . . . . . . . . . . . . . . . . 206
       22.12.1. Description  . . . . . . . . . . . . . . . . . . . . 206
       22.12.2. Registration Rules . . . . . . . . . . . . . . . . . 206
       22.12.3. Registered Values  . . . . . . . . . . . . . . . . . 206
     22.13. Transport Header Registries  . . . . . . . . . . . . . . 206
       22.13.1. Transport Protocol Specification . . . . . . . . . . 207
       22.13.2. Transport modes  . . . . . . . . . . . . . . . . . . 208
       22.13.3. Transport Parameters . . . . . . . . . . . . . . . . 208
     22.14. URI Schemes  . . . . . . . . . . . . . . . . . . . . . . 209
       22.14.1. The rtsp URI Scheme  . . . . . . . . . . . . . . . . 209
       22.14.2. The rtsps URI Scheme . . . . . . . . . . . . . . . . 210
       22.14.3. The rtspu URI Scheme . . . . . . . . . . . . . . . . 211
     22.15. SDP attributes . . . . . . . . . . . . . . . . . . . . . 211
     22.16. Media Type Registration for text/parameters  . . . . . . 212
   23. References  . . . . . . . . . . . . . . . . . . . . . . . . . 214
     23.1.  Normative References . . . . . . . . . . . . . . . . . . 214



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     23.2.  Informative References . . . . . . . . . . . . . . . . . 216
   Appendix A.  Examples . . . . . . . . . . . . . . . . . . . . . . 218
     A.1.   Media on Demand (Unicast)  . . . . . . . . . . . . . . . 218
     A.2.   Media on Demand using Pipelining . . . . . . . . . . . . 221
     A.3.   Media on Demand (Unicast)  . . . . . . . . . . . . . . . 224
     A.4.   Single Stream Container Files  . . . . . . . . . . . . . 228
     A.5.   Live Media Presentation Using Multicast  . . . . . . . . 230
     A.6.   Capability Negotiation . . . . . . . . . . . . . . . . . 231
   Appendix B.  RTSP Protocol State Machine  . . . . . . . . . . . . 233
     B.1.   States . . . . . . . . . . . . . . . . . . . . . . . . . 233
     B.2.   State variables  . . . . . . . . . . . . . . . . . . . . 233
     B.3.   Abbreviations  . . . . . . . . . . . . . . . . . . . . . 233
     B.4.   State Tables . . . . . . . . . . . . . . . . . . . . . . 234
   Appendix C.  Media Transport Alternatives . . . . . . . . . . . . 239
     C.1.   RTP  . . . . . . . . . . . . . . . . . . . . . . . . . . 239
       C.1.1.   AVP  . . . . . . . . . . . . . . . . . . . . . . . . 239
       C.1.2.   AVP/UDP  . . . . . . . . . . . . . . . . . . . . . . 239
       C.1.3.   AVPF/UDP . . . . . . . . . . . . . . . . . . . . . . 240
       C.1.4.   SAVP/UDP . . . . . . . . . . . . . . . . . . . . . . 241
       C.1.5.   SAVPF/UDP  . . . . . . . . . . . . . . . . . . . . . 241
       C.1.6.   RTCP usage with RTSP . . . . . . . . . . . . . . . . 241
     C.2.   RTP over TCP . . . . . . . . . . . . . . . . . . . . . . 242
       C.2.1.   Interleaved RTP over TCP . . . . . . . . . . . . . . 243
       C.2.2.   RTP over independent TCP . . . . . . . . . . . . . . 243
     C.3.   Handling Media Clock Time Jumps in the RTP Media Layer . 247
     C.4.   Handling RTP Timestamps after PAUSE  . . . . . . . . . . 251
     C.5.   RTSP / RTP Integration . . . . . . . . . . . . . . . . . 253
     C.6.   Scaling with RTP . . . . . . . . . . . . . . . . . . . . 253
     C.7.   Maintaining NPT synchronization with RTP timestamps  . . 253
     C.8.   Continuous Audio . . . . . . . . . . . . . . . . . . . . 253
     C.9.   Multiple Sources in an RTP Session . . . . . . . . . . . 253
     C.10.  Usage of SSRCs and the RTCP BYE Message During an
            RTSP Session . . . . . . . . . . . . . . . . . . . . . . 253
     C.11.  Future Additions . . . . . . . . . . . . . . . . . . . . 254
   Appendix D.  Use of SDP for RTSP Session Descriptions . . . . . . 255
     D.1.   Definitions  . . . . . . . . . . . . . . . . . . . . . . 255
       D.1.1.   Control URI  . . . . . . . . . . . . . . . . . . . . 255
       D.1.2.   Media Streams  . . . . . . . . . . . . . . . . . . . 256
       D.1.3.   Payload Type(s)  . . . . . . . . . . . . . . . . . . 257
       D.1.4.   Format-Specific Parameters . . . . . . . . . . . . . 257
       D.1.5.   Directionality of media stream . . . . . . . . . . . 257
       D.1.6.   Range of Presentation  . . . . . . . . . . . . . . . 258
       D.1.7.   Time of Availability . . . . . . . . . . . . . . . . 259
       D.1.8.   Connection Information . . . . . . . . . . . . . . . 259
       D.1.9.   Message Body Tag . . . . . . . . . . . . . . . . . . 259
     D.2.   Aggregate Control Not Available  . . . . . . . . . . . . 260
     D.3.   Aggregate Control Available  . . . . . . . . . . . . . . 260
     D.4.   RTSP external SDP delivery . . . . . . . . . . . . . . . 261



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   Appendix E.  RTSP Use Cases . . . . . . . . . . . . . . . . . . . 263
     E.1.   On-demand Playback of Stored Content . . . . . . . . . . 263
     E.2.   Unicast Distribution of Live Content . . . . . . . . . . 264
     E.3.   On-demand Playback using Multicast . . . . . . . . . . . 265
     E.4.   Inviting an RTSP server into a conference  . . . . . . . 265
     E.5.   Live Content using Multicast . . . . . . . . . . . . . . 266
   Appendix F.  Text format for Parameters . . . . . . . . . . . . . 268
   Appendix G.  Requirements for Unreliable Transport of RTSP  . . . 269
   Appendix H.  Backwards Compatibility Considerations . . . . . . . 271
     H.1.   Play Request in Play mode  . . . . . . . . . . . . . . . 271
     H.2.   Using Persistent Connections . . . . . . . . . . . . . . 271
   Appendix I.  Open Issues  . . . . . . . . . . . . . . . . . . . . 272
   Appendix J.  Changes  . . . . . . . . . . . . . . . . . . . . . . 273
   Appendix K.  Acknowledgements . . . . . . . . . . . . . . . . . . 280
     K.1.   Contributors . . . . . . . . . . . . . . . . . . . . . . 280
   Appendix L.  RFC Editor Consideration . . . . . . . . . . . . . . 282
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . . 283


































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1.  Introduction

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

   The protocol operates between RTSP 2.0 clients and servers, but also
   supports the usage of proxies placed between clients and servers.
   Clients can request information about streaming media from servers,
   by asking for a description of the media or use media description
   provided externally.  Then the media delivery protocol is used to
   establish the media streams described by the media description.
   Clients can then request to play out the media, pause it, or stop it
   completely, as known from a regular DVD player 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.

   RTSP 2.0 is an replacement of RTSP 1.0 [RFC2326] that obsoletes that
   specification.  This protocol is based on RTSP 1.0 but not backwards
   compatible other than in the basic version negotiation mechanism.
   The changes are documented in Appendix J.  There are many reasons why
   RTSP 2.0 can't be backwards compatible with RTSP 1.0 but some of the
   main ones are; that most header that needed to be extensible did not
   define the allowed syntax preventing safe deployment of extensions;
   the changed behavior of the PLAY method when received in playing
   state; changed behavior of the extensibility model and its mechanism;
   the change of syntax for some headers.  The summary is that there are
   so many small details that changing version become necessary to
   enable clarification and consistent behavior.

   This document is structured in the way that it begins with an
   overview of the protocol operations and its functions in an informal
   way.  Then a set of definitions of used terms and document
   conventions is introduced.  Then comes the actual protocol
   specification.  In the appendix some functionality that isn't core
   RTSP defined, but still important to enable some usage, like RTP and
   SDP usage with RTSP.  This is followed by a number of informational
   parts discussing the changes, use cases, different considerations or
   motivations.

1.1.  Notes on Copyright

   The IETF has adopted new IPR contributor rules in [RFC5378], which
   results in a changed model of copyright.  The baseline is that "The



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   IETF Trust and the IETF must obtain the right to publish an IETF
   Contribution as an RFC or an Internet-Draft from the Contributors."
   (taken from Section 3.1 of [RFC5378]).

   This memo has plenty of text taken from [RFC2326] and thus the
   associated copyright.  Magnus Westerlund has solicited the authors of
   [RFC2326] and this memo to transfer the copyright to the new model,
   i.e., to the IETF trust and the IETF.  Most of the authors have
   responded and transferred their copyright.  However, not all of them
   have.  This is the first reason for the currently used boiler plate
   (and thus the current status), i.e., with pre5378Trust200902.  See
   also this document [IETF-Trust-License-Policy] for more information.

   Furthermore, this memo does contain text that has been copied and
   modified from [RFC2616].  Older versions of this memo solely linked
   to the particular places.  Linking to the HTTP/1.1 specification was
   not appropriate anymore, as the text was not fitting to RTSP 2.0
   needs and had to be adapted.  Thus text copied from HTTP/1.1 is still
   under copyright prior to [RFC5378].
































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2.  Protocol Overview

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

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

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

2.1.  Content Description

   RTSP exists to provide access to multi-media content, but tries to be
   agnostic to the media type or the actual media delivery protocol that
   is used.  To enable a client to implement a complete system, an RTSP-
   external mechanism for describing the content and the delivery
   protocol(s) is used.  RTSP assumes that this description is either
   delivered completely out of bands or as a data object in the response
   to a client's request using the DESCRIBE method (Section 13.2).

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

   o  Number of media streams





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

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

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

   o  Media encoding information enabling client to correctly decode it
      upon reception

   o  An aggregate control resource identifier

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

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

2.2.  Session Establishment

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

   The client creates an RTSP session by sending an request using the
   SETUP method (Section 13.3) to the server.  In the SETUP request the
   client also includes all the transport parameter necessary to enable
   the media delivery protocol to function in the "Transport" header
   (Section 16.52).  This includes parameters that are pre-established
   by the content description but necessary for any middlebox to
   correctly handle the media delivery protocols.  The Transport header
   in a request may contain multiple alternatives for media delivery in
   a prioritized list, which the server can select from.  These
   alternatives are typically based on information in the content
   description.

   The server determines if the media resource is available upon
   receiving a SETUP request and if any of the transport parameter
   specifications are acceptable.  If that is successful, an RTSP
   session context is created and the relevant parameters and state is
   stored.  An identifier is created for the RTSP session and included
   in the response in the Session header (Section 16.48).  The SETUP
   response includes a Transport header that specifies which of the
   alternatives that have been selected and relevant parameters.



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

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

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

2.3.  Media Delivery Control

   After having established an RTSP session, the client can start
   controlling the media delivery.  The basic operations are Start by
   using the PLAY method (Section 13.4) and Halt by using the PAUSE
   method (Section 13.6).  PLAY also allows for choosing the starting
   media position from which the server should deliver the media.  The
   positioning is done using the Range header (Section 16.38) that
   supports several different time formats: Normal Play Time
   (Section 4.5), SMPTE Timestamps (Section 4.4) and absolute time
   (Section 4.6).  The Range header does further allow the client to
   specify a position where delivery should end, thus allowing a
   specific interval to be delivered.

   The support for positioning/searching within a content depends on the
   content's media properties.  Content exists in a number of different
   types, such as: on-demand, live, and live with simultaneous
   recording.  Even within these categories there are differences in how
   the content is generated and distributed, which affect how it can be
   accessed for playback.  The properties applicable for the RTSP
   session are provided by the server in the SETUP response using the



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   Media-Properties header (Section 16.28).  These are expressed using
   one or several independent attributes.  A first attribute is Random
   Access, which expresses if positioning can be done, and with what
   granularity.  Another aspect is whether the content will change
   during the lifetime of the session.  While on-demand content will
   provided in its completeness from the beginning, a live stream being
   recorded results in that the length of the accessible content grows
   as the session goes on.  There also exist content that is dynamically
   built by another protocol than RTSP and thus also changes in steps
   during the session, but maybe not continuously.  Furthermore, when
   content is recorded, there are cases where not the complete content
   is maintained, but, for example, only the last hour.  All these
   properties result in the need for mechanisms that will be discussed
   below.

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

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

   For live sessions that are being recorded the client will need to
   keep track of how the recording progresses.  Upon session
   establishment the client will learn the current duration of the
   recording from the Media-Range header.  As the recording is ongoing
   the content grows in direct relation to the passed time.  Therefore,
   each server's response to a PLAY request will contain the current
   Media-Range header.  The server should also send regularly every 5
   minutes the current media range in a PLAY_NOTIFY request.  If the
   live transmission ends, the server must send a PLAY_NOTIFY request
   with the updated Media-Properties indicating that the content stopped
   being a recorded live session and instead become a on-demand content.
   The request also contains the final media range.  While the live
   delivery continues the client can request to play what is delivered
   just now by using the NPT timescale symbol "now", or it can request a



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   specific point in the available content by an explicit range request
   for that point.  If the requested point is outside of the available
   interval the server will adjust the position to the closest available
   point, i.e., either at the beginning or the end.

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

2.4.  Session Parameter Manipulations

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

   RTSP 2.0 does not have a strong mechanism for providing negotiation
   of which headers, or parameters and their formats, that can be used.
   However, responses will indicate request headers or parameters that
   are not supported.  A priori determination of what features are
   available needs to be done through out-of-band mechanisms, like the
   session description, or through the usage of feature tags
   (Section 4.7).

2.5.  Media Delivery

   The delivery of media to the RTSP client is done with a protocol
   outside of RTSP and this protocol is determined during the session
   establishment.  This document specifies how media is delivered with
   RTP over UDP, TCP or the RTSP control connection.  Additional
   protocols may be specified in the future based on demand.

   The usage of RTP as media delivery protocol requires some additional



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   information to function well.  The PLAY responses contains
   synchronization information to enable reliable and timely deliver of
   how a client should synchronize different sources in the different
   RTP sessions.  It also provides a mapping between RTP timestamps and
   the content time scale.  When the server is notifying the client
   about the end of the media delivery requested using PLAY, it sends a
   PLAY_NOTIFY request to the client.  The PLAY_NOTIFY request includes
   information about the last RTP sequence numbers for each stream, and
   thus enables correct handling of the buffer drainage at the end.

2.5.1.  Media Delivery Manipulations

   The basic playback functionality of RTSP is to request content for a
   particular range to be delivered to the client in a pace that enables
   playback as intended by the creator.  However, RTSP can also
   manipulate how this delivery is done to the client in two ways.

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

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

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

   Scale is used for fast forward or slow motion control as it changes
   the amount of content timescale that should be played back per time
   unit.  Scale > 1.0, means fast forward, e.g.  Scale=2.0 results in
   that 2 seconds of content is played back every second of playback.
   Scale = 1.0 is the default value that is used if no Scale is
   specified, i.e. playback at the contents original rate.  Scale values
   between 0 and 1.0 is providing for slow motion.  Scale can be
   negative to allow for reverse playback in either regular pace (Scale
   = -1.0) or fast backwards (Scale < -1.0) or slow motion backwards
   (-1.0 < Scale < 0).  Scale = 0 is equal to pause and is not allowed.

   In most cases the realization of scale means server side manipulation
   of the media to ensure that the client can actually play it back.
   These media manipulation and when they are needed are highly media
   type dependent.  Lets exemplify with two common media types audio and
   video.

   It is very difficult to modify the playback rate of audio.  A maximum
   of 10-30% is possible by changing the pitch-rate of speech.  Music
   goes out of tune if one tries to manipulate the playback rate by
   resampling it.  This is a well known problem and audio is commonly



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   muted or played back in short segments with skips to keep up with the
   current playback point.

   For video is possible to manipulate the number of frames that is
   displayed per second, but the rendering capabilities are often
   limited to certain frame rates.  The decoding, handling capabilities
   and bitrate of received encoded content also limits the number of
   frames that can be delivered.  Therefore, when providing fast
   forward, one generally picks a subset of the frames from the original
   content to be displayed.  However, the video encoding methods used
   will commonly limit the possibilities on which frames that can be
   chosen and still be decoded by the receiver.

   Due to the media restrictions, a particular content will commonly be
   restricted to a limited set of possible scale ratios.  To handle this
   correctly, RTSP has a mechanism to indicate the supported Scale
   ratios for the content.  To support aggregated or dynamic content,
   where this may change during the ongoing session and dependent on the
   location within the content, a mechanism for updating the media
   properties and the current used scale factor exist.

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

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




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

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

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

2.6.  Session Maintenance and Termination

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

   o  Media transport protocol keep-alive.  RTCP is possible to use when
      using RTP.

   o  Any RTSP request referencing the session context.

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



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   server through the inclusion of the timeout parameter in the session
   header.

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

   A client may keep the session alive indefinitely if allowed by the
   server, however it is recommend to release the session context when
   an extended period of time without media delivery activity has
   passed.  It can re-establish the session context if required later.
   One issue is that what is extended periods of time is dependent on
   the server and its usage.  It is recommended that the client
   terminates the session before 10*times the session timeout value has
   passed.  A server may terminate the session after one session timeout
   period without any client activity beyond keep-alive.  When a server
   terminates the session context, it does that by sending a TEARDOWN
   request indicating the reason.

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

2.7.  Extending RTSP

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

   The client can learn the servers capability through the usage of the
   OPTIONS method (Section 13.1) and the Supported header
   (Section 16.49).  It can also try and possibly fail by using new



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   methods or require that particular features are supported using the
   Require or Proxy-Require header.

   The RTSP protocol in itself 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, for example,
      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.

   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 a detailed explanation of
   this see Section 11.

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

















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3.  Document Conventions

3.1.  Notational Conventions

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

3.2.  Terminology

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

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

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

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






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

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

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

   IRI:  Internationalized Resource Identifier, is 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) 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.  A message is either a Request or a Response.

   Message Body:  The information transferred as the payload of a
      message.  A message body consists of meta-information in the form
      of message-header and content in the form of an message-body, as
      described in Section 9.

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

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

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

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

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

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

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

   RTSP session:  A stateful abstraction upon which the main control
      methods of RTSP operate.  An RTSP session is a 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



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      request.  An RTSP session is a stateful entity; an RTSP server
      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.





























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4.  Protocol Parameters

4.1.  RTSP Version

   This specification defines version 2.0 of RTSP.

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

   The <minor> number is incremented when the changes made to the
   protocol add features which do not change the general message parsing
   algorithm, but which may add to the message semantics and imply
   additional capabilities of the sender.  The <major> number is
   incremented when the format of a message within the protocol is
   changed.  The version of an RTSP message is indicated by an RTSP-
   Version field in the first line of the message.  Note that the major
   and minor numbers MUST be treated as separate integers and that each
   MAY be incremented higher than a single digit.  Thus, RTSP/2.4 is a
   lower version than RTSP/2.13, which in turn is lower than RTSP/12.3.
   Leading zeros MUST be ignored by recipients and MUST NOT be sent.

4.2.  RTSP IRI and URI

   RTSP 2.0 defines and registers three URI schemes "rtsp", "rtsps" and
   "rtspu".  The usage of the last, "rtspu", is unspecified in RTSP 2.0,
   and is defined here to register 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 MUST 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.



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   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 IRI by the
   requester and not included in the request URI.  The user agent needs
   to interpret the value of the fragment based on the media type the
   request relates to; i.e., the media type indicated in Content-Type
   header in the response to DESCRIBE.

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

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

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




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   Also, the RTSP URI:

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

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

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

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

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

4.3.  Session Identifiers

   Session identifiers are strings of 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. approximately 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
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   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.

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

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

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



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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 16.53) 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.,
   "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.  Message Body Tags

   Message body tags are opaque strings that are used to compare two
   message bodies from the same resource, for example in caches or to
   optimize setup after a redirect.  Message body tags can be carried in
   the MTag header (see Section 16.30) or in SDP (see Appendix D.1.9).

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

   Message body tags are used in RTSP to make some methods conditional.
   The methods are made conditional through the inclusion of headers,
   see Section 16.23 and Section 16.25.  Note that RTSP message body
   tags apply to the complete presentation; i.e., both the session



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   description and the individual media streams.  Thus message body tags
   can be used to verify at setup time after a redirect that the same
   session description applies to the media at the new location using
   the If-Match header.

4.9.  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 differences between a
   number of supported usages.

   On-demand:  Media that has a fixed (given) duration that doesn't
      change during the life time of the RTSP session and 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 seekable, only the content presently being delivered can be
      accessed.

   Live with Recording:  A Live stream that is combined with a server
      side capability to store and retain the content of the live
      session 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.





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4.9.1.  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 provided to express the worst case distance between
      random access points.

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

   No seeking:  Seeking is not possible at all.

4.9.2.  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
      is in existence.

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

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

4.9.3.  Content Modifications

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

   Immutable:  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.





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   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.

4.9.4.  Supported Scale Factors

   The content is often limiting the possible rates of scale that can be
   supported when delivering the media.  To enable the client to know
   what values or ranges of scale operations that the whole content or
   the current position supports a media properties attribute for this
   is defined.  It contains a list with the values and/or ranges that
   are supported.  The attribute is named "Scales".  It may be updated
   at any point in the content due to content consisting of spliced
   pieces or content being dynamically updated by out of bands
   mechanisms.

4.9.5.  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:
      Immutable, Retention: unlimited or time limited.

   Dynamic On-demand:  Random Access: Random Access=3s, Content
      Modifications: Dynamic, Retention: unlimited 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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5.  RTSP Message

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

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

   The ISO 10646 character set avoids 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 [RFC0822] 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, 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

   RTSP header fields (see Section 16) include general-header, request-
   header, response-header, and entity-header fields.

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

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

   Unknown message headers MUST be ignored by a RTSP server or client.
   An RTSP Proxy MUST forward unknown message headers.  Message headers
   defined outside of this specification that are required to be
   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

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

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

   The presence of a message-body in a request is signaled by the
   inclusion of a Content-Length header field in the RTSP message.  A
   message-body MUST NOT be included in a request or response if the
   specification of the particular method (see Method Definitions
   (Section 13)) does not allow sending an message body.  A server
   SHOULD read and forward a message-body on any request; if the request
   method does not include defined semantics for a message body, then
   the message-body SHOULD be ignored when handling the request.




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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
       message-header fields present in the message.  (Note: An empty
       line is a line with nothing preceding the CRLF.)

   2.  If a Content-Length header(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 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

   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 16.28      |
                |                    |                    |
                | Media-Range        | Section 16.29      |
                |                    |                    |
                | Pipelined-Requests | Section 16.32      |
                |                    |                    |
                | Proxy-Supported    | Section 16.36      |
                |                    |                    |
                | Seek-Style         | Section 16.45      |
                |                    |                    |
                | Supported          | Section 16.49      |
                |                    |                    |
                | Timestamp          | Section 16.51      |
                |                    |                    |
                | Via                | Section 16.56      |
                +--------------------+--------------------+

                 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 message
      body(Section 9.1);

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

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

7.1.  Request Line

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




























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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 16.22      |
                |                    |                    |
                | If-Match           | Section 16.23      |
                |                    |                    |
                | If-Modified-Since  | Section 16.24      |
                |                    |                    |
                | If-None-Match      | Section 16.25      |
                |                    |                    |
                | Notify-Reason      | Section 16.31      |
                |                    |                    |



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                | Proxy-Require      | Section 16.35      |
                |                    |                    |
                | Range              | Section 16.38      |
                |                    |                    |
                | Terminate-Reason   | Section 16.50      |
                |                    |                    |
                | 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 16.52      |
                |                    |                    |
                | User-Agent         | Section 16.54      |
                +--------------------+--------------------+

                     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.


















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8.  Response

   After receiving and interpreting a request message, the recipient
   responds with an RTSP response message.  The final response is
   exactly one message, and final responses are any using the response
   code classes from the list; 2xx, 3xx, 4xx and 5xx classes.  Only for
   responses using the response code class 1xx, is it allowed to send
   one or more 1xx response messages prior to the final response
   message.

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

8.1.  Status-Line

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

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

8.1.1.  Status Code and Reason Phrase

   The Status-Code element is a 3-digit integer result code of the
   attempt to understand and satisfy the request.  These codes are fully
   defined in Section 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







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

   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 message body
   returned with the response, since that message body is likely to
   include human-readable information which will explain the unusual
   status.

    +------+----------------------------------------+-----------------+
    | Code | Reason                                 | Method          |
    +------+----------------------------------------+-----------------+
    | 100  | Continue                               | all             |
    |      |                                        |                 |
    |      |                                        |                 |
    | 200  | OK                                     | all             |
    |      |                                        |                 |
    |      |                                        |                 |
    | 301  | Moved Permanently                      | all             |
    |      |                                        |                 |
    | 302  | Found                                  | all             |
    |      |                                        |                 |
    | 304  | Not Modified                           | all             |
    |      |                                        |                 |
    | 305  | Use Proxy                              | all             |
    |      |                                        |                 |
    |      |                                        |                 |
    | 400  | Bad Request                            | all             |



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    | 401  | Unauthorized                           | all             |
    |      |                                        |                 |
    | 402  | Payment Required                       | all             |
    |      |                                        |                 |
    | 403  | Forbidden                              | all             |
    |      |                                        |                 |
    | 404  | Not Found                              | all             |
    |      |                                        |                 |
    | 405  | Method Not Allowed                     | all             |
    |      |                                        |                 |
    | 406  | Not Acceptable                         | all             |
    |      |                                        |                 |
    | 407  | Proxy Authentication Required          | all             |
    |      |                                        |                 |
    | 408  | Request Timeout                        | all             |
    |      |                                        |                 |
    | 410  | Gone                                   | all             |
    |      |                                        |                 |
    | 411  | Length Required                        | all             |
    |      |                                        |                 |
    | 412  | Precondition Failed                    | DESCRIBE, SETUP |
    |      |                                        |                 |
    | 413  | Request Message Body Too Large         | all             |
    |      |                                        |                 |
    | 414  | Request-URI Too Long                   | all             |
    |      |                                        |                 |
    | 415  | Unsupported Media Type                 | all             |
    |      |                                        |                 |
    | 451  | Parameter Not Understood               | SET_PARAMETER   |
    |      |                                        |                 |
    | 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             |
    |      |                                        |                 |



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    | 461  | Unsupported Transport                  | all             |
    |      |                                        |                 |
    | 462  | Destination Unreachable                | all             |
    |      |                                        |                 |
    | 463  | Destination Prohibited                 | SETUP           |
    |      |                                        |                 |
    | 464  | Data Transport Not Ready Yet           | PLAY            |
    |      |                                        |                 |
    | 465  | Notification Reason Unknown            | PLAY_NOTIFY     |
    |      |                                        |                 |
    | 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 Headers

   The response-header allow the request recipient to pass additional
   information about the response which cannot be placed in the Status-
   Line.  This header 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      |
              |                        |                    |
              | MTag                   | Section 16.30      |
              |                        |                    |
              | Location               | Section 16.27      |
              |                        |                    |
              | 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 16.52      |
              |                        |                    |
              | Unsupported            | Section 16.53      |
              |                        |                    |
              | Vary                   | Section 16.55      |
              |                        |                    |
              | WWW-Authenticate       | Section 16.57      |
              +------------------------+--------------------+

                    Table 5: The RTSP response headers

   Response-headers 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 MAY be given the semantics of response-header if all parties
   in the communication recognize them to be response-header.



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   Unrecognized headers in responses are treated as message-headers.


















































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9.  Message Body

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

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

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

9.1.  Message Body Header Fields

   message-header fields define meta-information about the message-body
   or, if no body is present, about the resource identified by the
   request.  The message body 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 16.21      |
                 |                  |                    |
                 | Last-Modified    | Section 16.26      |
                 +------------------+--------------------+

                  Table 6: The RTSP message body headers

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



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

9.2.  Message Body

   RTSP message with an message body MUST include the Content-Type and
   Content-Length headers if a message body is included.

   When an message body is included with a message, the data type of
   that body is determined via the header fields Content-Type and
   Content-Encoding.

   Content-Type specifies the media type of the underlying data.
   Content-Encoding may be used to indicate any additional content
   codings applied to the data, usually for the purpose of data
   compression, that are a property of the requested resource.  There is
   no default encoding.

   The Content-Length of a message is the length of the message-body.
































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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 MUST NOT be sent to multicast groups since no
   connection can be established with a specific receiver in multicast
   environments.

   Certain RTSP headers, such as the CSeq header (Section 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 is RECOMMENDED be used for all transactions
   between the server and client, including messages for multiple RTSP
   sessions.  However a persistent connection MAY be closed after a few
   message exchanges.  For example, a client may use a persistent
   connection for the initial SETUP and PLAY message 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.

   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 silently discard RTSP
   messages, and may even drop an RTSP session without notifying the



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   client.  An example of such a case is when the server desires to send
   a REDIRECT request for an RTSP session to the client but is not able
   to do so because it cannot reach the client.  A server that attempt
   to send a request to a client that has no connection currently to the
   server SHOULD discard the request directly, it MAY queue it for later
   delivery.  However, if the server queue the request it should when
   adding additional requests to the queue ensure to remove older
   requests that are now redundant.

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

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

   To provide some guidance on what is reasonable the following
   guidelines are given.  An RTSP agent should not have more than 10
   outstanding requests per RTSP session.  An RTSP agent should not have
   more than 10 outstanding requests that aren't related to an RTSP
   session or that are requesting to create an RTSP session.



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   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 15.4.25) are used for negotiating capabilities
      of a server with respect to content or other factors.  In such
      cases, it is inefficient for the server to close a connection on
      an error response.  Also, such behavior would prevent
      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
   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.





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10.4.  Timing Out Connections and RTSP Messages

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

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

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

   A requester SHOULD wait longer than 10 seconds for a response if it
   is experiencing significant transport delays on its connection to the
   responder.  The requester is capable of determining the RTT of the
   request/response cycle using the Timestamp header (Section 16.51) in
   any RTSP request.

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

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



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   issued to accomplish other things, the following mechanisms can be
   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 MUST 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 is also usable, but it causes the server to
         perform more unnecessary processing and result in bigger
         responses than necessary for the task.  The reason is that the
         server needs to determine the capabilities associated with the
         media resource to correctly populate the Public and Allow
         headers.

   The timeout parameter MAY be included in a SETUP response, and MUST
   NOT be included in requests.  The server uses it to indicate to the
   client how long the server is prepared to wait between RTSP commands
   or other signs of life before closing the session due to lack of
   activity (see below and Appendix B).  The timeout is measured in
   seconds, with a default of 60 seconds.  The length of the session
   timeout MUST NOT be changed in 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 whether 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 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 whether the client, server or
   proxy supports the functionality that is necessary to achieve the
   desired service.  To determine support of a feature-tag, several
   different headers can be used, each explained below:

   Supported:  This header is used to determine the complete set of
         functionality that both client and server have.  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 requester declares all its capabilities by
         including all supported feature-tags.  This results in that the
         receiver learns the requesters feature support.  The receiver
         then includes its set of features in the response.




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   Proxy-Supported:  This header is used similar to the Supported
         header, but instead of giving the supported functionality of
         the client or server it provides both the requester and the
         responder a view of 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 requester add it.

   Require:  The 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 header 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 features were not supported.  Such a response is
         only the result of the usage of the Require and/or Proxy-
         Require header where one or more feature where not supported.
         This information allows the requester 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 responding entity MUST process the incoming
   requests in their sending order.  The sending order can be determined
   by the CSeq header and its sequence number.  For TCP the delivery
   order will be the same as the sending order.  The processing of the
   request MUST 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 are processed in the same RTSP session context which
   the first request creates.  In other words a client can request that
   two or more media streams are set-up and then played without needing
   to wait for a single response.  This speeds up the initial startup
   time for an RTSP session with at least one RTT.

   If a pipelined request builds on the successful completion of one or
   more prior requests the requester must verify that all requests were
   executed as expected.  A common example will be two SETUP requests
   and a PLAY request.  In case one of the SETUP fails unexpectedly, the
   PLAY request can still be successfully executed.  However, 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 MUST 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      |
   |               |           |        |              |               |
   |               | S -> C    |        | 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 optional, but SET_PARAMETER is
      required 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 who's 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 still needs to perform the minimal
   required processing, like adding the Via header.

13.1.  OPTIONS

   The semantics of the RTSP OPTIONS method is similar to that of the
   HTTP OPTIONS method described in [H9.2].  In RTSP however, OPTIONS is
   bi-directional, in that a client can 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
   responding RTSP agent.




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   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 MUST respond
   with a description of the requested resource and return the
   description in the message body 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 possible
   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 16.52, specifies the media
   transport parameters acceptable to the client for data transmission;
   the response will contain the transport parameters selected by the
   server.  This allows the client to enumerate in descending order of
   preference the transport mechanisms and parameters acceptable to it,
   while the server can select the most appropriate.  It is expected
   that the session description format used will enable the client to
   select a limited number 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 MUST indicate
   the known transport parameters, even if it has no influence over
   these parameters, for example, where the server advertises a fixed
   multicast address as destination.

      Since SETUP includes all transport initialization information,
      firewalls and other intermediate network devices (which need this
      information) are spared the more arduous task of parsing the
      DESCRIBE response, which has been reserved for media
      initialization.

   The client MUST 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 MUST respond using 456 (Header Field Not
   Valid for Resource) and include the Accept-Ranges header with the
   range unit formats supported for the resource.

   In a SETUP response the server MUST include the Accept-Ranges header
   (see Section 16.5) to indicate which time formats that are acceptable
   to use for this media resource.

   The SETUP response 200 OK MUST include the Media-Properties header
   (see Section 16.28 ).  The combination of the parameters of the
   Media-Properties header indicate the nature of the content present in
   the session (see also Section 4.9).  For example, a live stream with
   time shifting is indicated by

   o  Random Access set to Random-Access,

   o  Content Modifications set to Time Progressing,

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

   The SETUP response 200 OK MUST include the Media-Range header (see
   Section 16.29) 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 15.4.24).
   An Aggregate control URI MUST be used to control an aggregated
   session.  This URI MUST be different from the stream control URIs of
   the individual media streams included in the aggregate.  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 carries enough information for
      aggregate control of a session, the Aggregate control URI is still
      important for some methods such as SET_PARAMETER where the control
      URI enables the resource in question to be easily identified.  The
      Aggregate control URI is also useful for proxies, enabling them to
      route the request to the appropriate server, and for logging,
      where it is useful to note the actual resource that a request was
      operating on.

   A session will exist until it is either removed by a TEARDOWN request
   or is timed-out by the server.  The server MAY remove a session that
   has not demonstrated liveness signs from the client(s) within a
   certain timeout period.  The default timeout value is 60 seconds; the
   server MAY set this to a different value and indicate so in the
   timeout field of the Session header in the SETUP response.  For
   further discussion see Section 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 MUST remain unchanged from their values as if the SETUP
   request had never been received by the server.

13.3.1.  Changing Transport Parameters

   A client MAY issue a SETUP request for a stream that is already set
   up or playing in the session to change transport parameters, which a
   server MAY allow.  If it does not allow changing of parameters, it
   MUST respond with error 455 (Method Not Valid In This State).
   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 MUST include the Range to indicate
   from what point the new transport parameters are used.  Further, if
   RTP is used for delivery, the server MUST 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 start point in the Range
   header MUST be for the corresponding time, i.e. be used in the same
   way as for PLAY to ensure the correct synchronization information is
   available.

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



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   avoid changing the synchronization information if possible.

13.4.  PLAY

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

13.4.1.  General Usage

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

   Upon receipt of the PLAY request, the server MUST position the normal
   play time to the beginning of the range specified in the received
   Range header 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.  If no Range header is present in the PLAY
   request the server shall play from current pause point until the end
   of media.  The pause point defaults at start to the beginning of the
   media.  For media that is time-progressing and has no retention, the
   pause point will always be set equal to NPT "now", i.e. current
   playback point.  The pause point may also be set to a particular
   point in the media by the PAUSE method, see Section 13.6.  The pause
   point for media that is currently playing is equal to the current
   media position.  For time-progressing media with time-limited
   retention, if the pause point represents a position that is older
   than what is retained by the server, the pause point will be moved to
   the oldest retained.

   What range values is valid depends on the type of content.  For
   content that isn't time progressing the range value is valid if the
   given range is part of any media within the aggregate.  In other
   words the valid media range for the aggregate is the super-set of all
   of the media components in the aggregate.  If a given range value
   points outside of the media, the response MUST be the 457 (Invalid
   Range) error code and include the Media-Range header (Section 16.29)
   with the valid range for the media.  For time progressing content
   where the client request a start point prior to what is retained, the
   start point is adjusted to the oldest retained content.  For a start
   point that is beyond the media front edge, i.e. beyond the current
   value for "now", the server shall adjust the start value to the
   current front edge.  The Range headers end point value may point
   beyond the current media edge.  In that case the server shall deliver
   media from the requested (and possibly adjusted) start point until
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   the specified stop point.  Please note that if one simply want to
   play from a particular start point until the end of media using an
   Range header with an implicit stop point is recommended.

   For media with random access properties a client may express its
   preference on which policy for start point selection the server shall
   use.  This is done by including the Seek-Style header (Section 16.45)
   in the PLAY request.

   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 MUST be included.

   All range specifiers in this specification allow for ranges with
   implicit start point (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 MUST be given.

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

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

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

   For any error response to a PLAY request, the server's response
   depends on the current session state.  If the session is in ready
   state, the current pause-point is returned using Range header with
   the pause point as the explicit start-point and an implicit end-



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   point.  For time-progressing content where the pause-point moves with
   real-time due to limited retention, the current pause point is
   returned.  For sessions in playing state, the current playout point
   and the remaining parts of the range request is returned.  For any
   media with retention longer than 0 seconds the currently valid Media-
   Range header shall also be included in the response.

   A PLAY response MAY include a header(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 which is 10
   m prior (3.51) and the RTP-Info header that contains the necessary
   parameters for the RTP stack.
   C->S: PLAY rtsp://example.com/audio RTSP/2.0
         CSeq: 836
         Session: 12345678
         Range: npt=3.52-
         User-Agent: PhonyClient/1.2

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

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

   The server reply with the actual start point that will be delivered.
   This may differ from the requested range if alignment of the
   requested range to valid frame boundaries is required for the media
   source.  Note that some media streams in an aggregate may need to be
   delivered from even earlier points.  Also, some media 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 server shall also indicate which policy it uses for
   selecting the actual start point by including a Seek-Style header.

   In the following example the client receives the first media packet
   that stretches all the way up and past the requested playtime.  Thus,



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

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

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

   After playing the desired range, the presentation does NOT 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
   MUST 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 range in Range header has a implicit start time
   (-endtime), the server MUST continue to play from where it currently
   was until the specified end point.  This is useful to change 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.
















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   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
         Seek-Style: Next
         RTP-Info:url="rtsp://example.com/twister.en"
            ssrc=0D12F123:seq=14783;rtptime=2345962545

   For playing back a recording of a live presentation, it may be
   desirable to use clock units:
   C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/2.0
         CSeq: 835
         Session: 12345678
         Range: clock=19961108T142300Z-19961108T143520Z
         User-Agent: PhonyClient/1.2

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

13.4.2.  Aggregated Sessions

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

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

   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



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   delay from start of session establishment until media play-out has
   started with one round trip time.  However, a client needs to be
   aware that using this procedure will result in the playout of the
   server state established at the time of processing the PLAY, i.e.,
   after the processing of all the requests prior to the PLAY request in
   the pipeline.  This may not be the intended one due to failure of any
   of the prior requests.  However, a client can easily determine this
   based on the responses from those requests.  In case of failure the
   client can halt the media playout using PAUSE and try to establish
   the intended state again before issuing another PLAY request.

13.4.3.  Updating current PLAY Requests

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

   The important difference compared to a PLAY request in ready state is
   the handling of the current play point and how the range header in
   request is constructed.  The session is actively playing media and
   the play point 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 implicit
   start point and a explicit stop value (Z), e.g. npt=-60, which
   indicate that it MUST convert the range specifier being played prior
   to this PLAY request (X to Y) into (X to Z) and continue as this was
   the request originally played.

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











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

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

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

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Date: Thu, 23 Jan 1997 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=14-25
           Seek-Style: Next
           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934239921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=789DAF12:seq=57654;rtptime=2792842193
           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 16.28):

   o  Random-Access property is set to Random Access;

   o  Content Modifications set to Immutable;

   o  Retention set Unlimited or Time-Limited.

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



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   Section 13.4.1.

13.4.5.  Playing Dynamic On-Demand Media

   Dynamic on-demand media is indicated by the content of the Media-
   Properties header in the SETUP response by (see also Section 16.28):

   o  Random-Access set to Random Access;

   o  Content Modifications set to dynamic;

   o  Retention 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 MUST include the
   current Media-Range header (see Section 16.29).

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 16.28):

   o  Random-Access set to no-seeking;

   o  Content Modifications set to Time-Progressing;

   o  Retention with Time-Duration set to 0.0.

   For live media, the SETUP response 200 OK MUST include the Media-
   Range header (see Section 16.29).

   A client MAY send PLAY requests without the Range header, if the
   request include the Range header it MUST use a symbolic value
   representing "now".  For NPT that range specification is "npt=now-".
   The server MUST include the Range header in the response and it MUST
   indicate an explicit time value and not a symbolic value.  In other
   words npt=now- is not a valid to use in the response.  Instead the
   time since session start is recommended expressed as an open
   interval, e.g. "npt=96.23-".  An absolute time value (clock) for the



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   corresponding time MAY be given, i.e. "clock=20030213T143205Z-".  The
   UTC clock format can only be used if client has shown support for it
   using the Accept-Ranges header.

13.4.7.  Playing Live with Recording

   Certain media server may offer recording services of live sessions to
   their clients.  This recording would normally be from the beginning
   of the media session.  Clients can randomly access the media between
   now and the beginning of the media session.  This live media with
   recording is indicated by the content of the Media-Properties header
   in the SETUP response by (see also Section 16.28):

   o  Random-Access set to random-access;

   o  Content Modifications set to Time-Progressing;

   o  Retention set to Time-limited or Unlimited

   The SETUP response 200 OK MUST include the Media-Range header (see
   Section 16.29) 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 header indicates the currently available media window
   around the current time.  This window can cover recorded content in
   the past (seen from current time in the media) or recorded content in
   the future (seen from current time in the media).  The server adjusts
   the 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 16.28):

   o  Random-Access set to random-access;

   o  Content Modifications set to Time-Progressing;





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

   The SETUP response 200 OK MUST include the Media-Range header (see
   Section 16.29) for this type of media.  For live media with recording
   the Range header indicates the current time in the media and the
   Media Range indicates a window around the current time.  This window
   can cover recorded content in the past (seen from current time in the
   media) or recorded content in the future (seen from current time in
   the media).  The server adjusts the play point to the requested
   border of the window, if the client requests a play point that is
   located outside the recording windows, e.g., if requested too far in
   the past, the server selects the oldest range in the recording.  The
   considerations in Section 13.5.3 apply, if a client requests 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 asynchronously 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 in this case a need to obey some limitations
   when adding new Notify-Reasons that intend to use a message body: The
   server can send any type of message body, but it is not ensured that
   the client can understand the received message body.  This is related
   to DESCRIBE (see Section 13.2 ), but in this particular case the
   client can state its acceptable message bodies by using the Accept
   header.  In the case of PLAY_NOTIFY, the server does not know which
   message bodies are understood by the client.

   The Notify-Reason header (see Section 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 MUST respond with an



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   465 (Notification Reason Unknown) (Section 15.4.30) error code.
   Servers can send PLAY_NOTIFY with these types:

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

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

   o  scale-change (see Section 13.5.3).

13.5.1.  End-of-Stream

   A PLAY_NOTIFY request with Notify-Reason header set to end-of-stream
   indicates the completion or near completion of the PLAY request and
   the ending delivery of the media stream(s).  The request MUST NOT be
   issued unless the server is in the playing state.  The end of the
   media stream delivery notification may be used to indicate either a
   successful completion of the PLAY request currently being served, or
   to indicate some error resulting in failure to complete the request.
   The Request-Status header (Section 16.41) MUST be included to
   indicate which request the notification is for and its completion
   status.  The message response status codes (Section 8.1.1) are used
   to indicate how the PLAY request concluded.  The sender of a
   PALY_NOTIFY can issue an updated PALY_NOTIFY, in the case of a
   PLAY_NOTIFY sent with wrong information.  For instance, a PLAY_NOTIFY
   was issued before reaching the end-of-stream, but some error occurred
   resulting in that the previously sent PLAY_NOTIFY contained a wrong
   time when the stream will end.  In this case a new PLAY_NOTIFY MUST
   be sent including the correct status for the completion and all
   additional information.

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

   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.




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   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:

     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 16.28) and/or the available media range
   that can be played as indicated by Media-Range (Section 16.29).
   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 MUST be sent for media that are time-progressing
   every time an 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 notifications 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 MUST 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
   requested range.

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

   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

   The server may be forced to change the rate, when a client request
   playback at Scale (Section 16.44) values other than 1.0 (normal
   playback rate).  For time progressing media with some retention, i.e.
   the server stores already sent content, a client requesting to play
   with Scale values larger than 1 may catch up with the front end of
   the media.  The server will then be unable to continue to provide
   with content at Scale larger than 1 as content is only made available
   by the server at Scale=1.  Another case is when Scale < 1 and the
   media retention is time-duration limited.  In this case the playback
   point can reach the oldest media unit available, and further playback
   at this scale becomes impossible as there will be no media available.
   To avoid having the client 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.

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

   To minimize impact on playback in any of the above cases the server
   MUST modify the playback properties and set Scale to a supportable
   value and continue delivery the media.  When doing this modification
   it MUST send a PLAY_NOTIFY message with the Notify-Reason header set
   to "scale-change".  The request MUST 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 wallclock time of the change.
   For time progressing content also the Media-Range and the Media-



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   Properties at this point in time MUST be included.  The Media-
   Properties header MUST be included if the scale change was due to the
   content changing what scale values that is supported.

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

   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 MUST 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:







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

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


























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

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

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

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=21-30
           Session: 12345678

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















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

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

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

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

13.7.  TEARDOWN

13.7.1.  Client to Server

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

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

   A TEARDOWN using a media URI in an aggregated session 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
   contains a single media after the requests completion.  A session
   that will exist after the processing of the TEARDOWN request MUST in
   the response to that TEARDOWN request contain a Session header.  Thus
   the presence of the Session header indicates to the receiver of the



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   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

13.7.2.  Server to Client

   The server can send TEARDOWN requests in the server to client
   direction to indicate that the server has been forced to terminate
   the ongoing session.  This may happen for several reasons such as,
   server maintenance without available backup, or session have been
   inactive for extended periods of time.  The reason is provided in the
   Terminate-Reason header (Section 16.50).

   When a RTSP client has maintained a RTSP session that otherwise is
   inactive for an extended period of time the server may reclaim the
   resources.  That is done by issuing a REDIRECT request with the
   Terminate-Reason set to "Session-Timeout".  This MAY be done when the
   client has been inactive in the RTSP session for more than one
   Session Timeout period (Section 16.48).  However, the server is
   RECOMMENDED to not perform this operation until an extended period of
   inactivity has passed.  The time period is considered extended when
   it is 10 times the Session Timeout period.  Consideration of the
   application of the server and its content should be performed when
   configuring what is considered as extended periods of time.

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

   The TEARDOWN request is normally done on the session aggregate
   control URI and MUST include the following headers; Session and
   Terminate-Reason headers.  The request only applies to the session
   identified in the Session header.  The server may include a message
   to the client's user with the "user-msg" parameter.

   The TEARDOWN request may alternatively be done on the wild card URI *
   and without any session header.  The scope of such a request is



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   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.  This request
   indicates that all sessions and pending requests being managed via
   the control connection are terminated.  Any intervening proxies
   SHOULD do all of the following in the order listed:

   1.  respond to the TEARDOWN request

   2.  disconnect the control channel from the requesting server

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

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

13.8.  GET_PARAMETER

   The GET_PARAMETER request retrieves the value of any specified
   parameter or parameters for a presentation or stream specified in the
   URI.  If the Session header is present in a request, the value of a
   parameter MUST be retrieved in the specified session context.  There
   are two ways of specifying the parameters to be retrieved.  The first
   is by including headers which have been defined such that you can use
   them for this purpose.  Header for this purpose should allow empty,
   or stripped value parts to avoid having to specify bogus data when
   indicating the desire to retrieve a value.  The successful completion
   of the request should also be evident from any filled out values in
   the response.  The Media-Range header (Section 16.29) is one such
   header.  The other is to specify a message body that lists the
   parameter(s) that are desirable to retrieve.  The Content-Type header
   (Section 16.18) is used to specify which format the message body has.

   The headers that MAY be used for retrieving their current value using
   GET_PARAMETER are:

   o  Accept-Ranges

   o  Media-Range

   o  Media-Properties

   o  Range

   o  RTP-Info




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   The method MAY also be used without a message body 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.  Normally the presence of
   filled out values in the header will be indication that the header
   has been processed.  However, for cases when this is difficult to
   determine, it is recommended to use a feature-tag and the Require
   header.  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) MUST be sent
   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
           Session: 12345678
           Content-Length: 38
           Content-Type: text/parameters

           packets_received: 10
           jitter: 0.3838

13.9.  SET_PARAMETER

   This method requests to set the value of a parameter or a set of
   parameters for a presentation or stream specified by the URI.  The
   method MAY also be used without a message body.  It is the
   RECOMMENDED method to 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



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   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) MUST be used.  In the case a parameter is
   not allowed to change, the error code is 458 (Parameter Is Read-
   Only).  The response body MUST contain only the parameters that have
   errors.  Otherwise no body MUST 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





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13.10.  REDIRECT

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

   The REDIRECT request SHALL contain a Terminate-Reason header
   (Section 16.50) to inform the client of the reason for the request.
   Additional parameters related to the reason may also be included.
   The intention here is to allow an server administrator to do a
   controlled shutdown of the RTSP server.  That requires sufficient
   time to inform all entities having associated state with the server
   and for them to perform a controlled migration from this server to a
   fall back server.

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

   A client may receive a REDIRECT request without a Session header at
   any time when it has communication or a connection established with a
   server.  The scope of such a request is limited to the next-hop (i.e.
   the RTSP agent in direct communication with the server) and applies
   to all sessions controlled, as well as the 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 REQUIRED
   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




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

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

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

   When no Terminate-Reason "time" parameter are included in a REDIRECT
   request, the client SHALL perform the redirection immediately and
   return a response to the server.  The server shall consider the
   session as terminated and can free any associated state after it
   receives the successful (2xx) response.  The server MAY close the
   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.

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

      The differentiation of REDIRECT requests with and without range
      header 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



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   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 MTag identifiers are RECOMMENDED to
   verify the assumption.

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

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

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
























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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 MUST 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 16.52).

   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 interval containing a second channel in the
   interleaved parameter of the Transport header, see Section 16.52.  If
   RTCP is used, packets MUST 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
           Media-Properties: Random-Access=0.2, Unmutable, Unlimited

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

     S->C: RTSP/2.0 200 OK
           CSeq: 3
           Session: 12345678
           Date: Thu, 05 Jun 1997 18:59:15 GMT
           RTP-Info: url="rtsp://example.com/bar.file"
             ssrc=0D12F123:seq=232433;rtptime=972948234
           Range: npt=0-56.8
           Seek-Style: RAP

     S->C: $005{2 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

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

15.2.  Success 2xx

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

15.2.1.  200 OK

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

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.

   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.

   An 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 an established session about the need for redirecting the
   session.  If an 3rr response is received for a request in relation to



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   an established session, the client SHOULD send a TEARDOWN request for
   the session, and MAY reestablish the session using the resource
   indicated by the Location.

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

15.3.1.  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.

15.3.2.  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

15.3.3.  303 See Other

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

15.3.4.  304 Not Modified

   If the client has performed a conditional DESCRIBE or SETUP (see
   Section 16.24) and the requested resource has not been modified, the
   server SHOULD send a 304 response.  This response MUST NOT contain a



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   message-body.

   The response MUST include the following header fields:

   o  Date

   o  MTag 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 MTag and If-Match headers may be used to link the
   DESCRIBE and SETUP in this manner.

15.3.5.  305 Use Proxy

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

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.  If the request does not have a CSeq header, the
   server MUST NOT include a CSeq in the response.

15.4.2.  401 Unauthorized

   The request requires user authentication.  The response MUST include
   a WWW-Authenticate header (Section 16.57) field containing a
   challenge applicable to the requested resource.  The client MAY
   repeat the request with a suitable Authorization header field.  If
   the request already included Authorization credentials, then the 401
   response indicates that authorization has been refused for those
   credentials.  If the 401 response contains the same challenge as the
   prior response, and the user agent has already attempted
   authentication at least once, then the user SHOULD be presented the
   entity that was given in the response, since that entity might



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   include relevant diagnostic information.  HTTP access authentication
   is explained in [RFC2617].

15.4.3.  402 Payment Required

   This code is reserved for future use.

15.4.4.  403 Forbidden

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

15.4.5.  404 Not Found

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

15.4.6.  405 Method Not Allowed

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

15.4.7.  406 Not Acceptable

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

   The response SHOULD include an message body containing a list of
   available entity characteristics and location(s) from which the user
   or user agent can choose the one most appropriate.  The entity format
   is specified by the media type given in the Content-Type header
   field.  Depending upon the format and the capabilities of the user
   agent, selection of the most appropriate choice MAY be performed



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   automatically.  However, this specification does not define any
   standard for such automatic selection.

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

15.4.8.  407 Proxy Authentication Required

   This code is similar to 401 (Unauthorized) (Section 15.4.2), but
   indicates that the client must first authenticate itself with the
   proxy.  The proxy MUST return a Proxy-Authenticate header field
   (Section 16.33) containing a challenge applicable to the proxy for
   the requested resource.

15.4.9.  408 Request Timeout

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

15.4.10.  410 Gone

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

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

15.4.11.  411 Length Required

   The server refuses to accept the request without a defined Content-
   Length.  The client MAY repeat the request if it adds a valid
   Content-Length header field containing the length of the message-body
   in the request message.





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15.4.12.  412 Precondition Failed

   The precondition given in one or more of the request-header fields
   evaluated to false when it was tested on the server.  This response
   code allows the client to place preconditions on the current resource
   meta information (header field data) and thus prevent the requested
   method from being applied to a resource other than the one intended.

15.4.13.  413 Request Message Body Too Large

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

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

15.4.14.  414 Request-URI Too Long

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

15.4.15.  415 Unsupported Media Type

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

15.4.16.  451 Parameter Not Understood

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

15.4.17.  452 reserved

   This error code was removed from RFC 2326 [RFC2326] as it is
   obsolete.  This error code MUST NOT be used anymore.



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15.4.18.  453 Not Enough Bandwidth

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

15.4.19.  454 Session Not Found

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

15.4.20.  455 Method Not Valid in This State

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

15.4.21.  456 Header Field Not Valid for Resource

   The server could not act on a required request header.  For example,
   if PLAY contains the Range header field but the stream does not allow
   seeking.  This error message may also be used for specifying when the
   time format in Range is impossible for the resource.  In that case
   the Accept-Ranges header MUST be returned to inform the client of
   which format(s) that are allowed.

15.4.22.  457 Invalid Range

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

15.4.23.  458 Parameter Is Read-Only

   The parameter to be set by SET_PARAMETER can be read but not
   modified.  When returning this error message the sender SHOULD return
   a message body containing the offending parameter(s).

15.4.24.  459 Aggregate Operation Not Allowed

   The requested method may not be applied on the URI in question since
   it is an aggregate (presentation) URI.  The method may be applied on
   a media URI.

15.4.25.  460 Only Aggregate Operation Allowed

   The requested method may not be applied on the URI in question since
   it is not an aggregate control (presentation) URI.  The method may be
   applied on the aggregate control URI.



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15.4.26.  461 Unsupported Transport

   The Transport field did not contain a supported transport
   specification.

15.4.27.  462 Destination Unreachable

   The data transmission channel could not be established because the
   client address could not be reached.  This error will most likely be
   the result of a client attempt to place an invalid dest_addr
   parameter in the Transport field.

15.4.28.  463 Destination Prohibited

   The data transmission channel was not established because the server
   prohibited access to the client address.  This error is most likely
   the result of a client attempt to redirect media traffic to another
   destination with a dest_addr parameter in the Transport header.

15.4.29.  464 Data Transport Not Ready Yet

   The data transmission channel to the media destination is not yet
   ready for carrying data.  However the responding 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.

15.4.30.  465 Notification Reason Unknown

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

15.4.31.  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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15.4.32.  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.

15.4.33.  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

   Response status codes beginning with the digit "5" indicate cases in
   which the server is aware that it has erred or is incapable of
   performing the request The server SHOULD include an entity containing
   an explanation of the error situation, and whether it is a temporary
   or permanent condition.  User agents SHOULD display any included
   entity to the user.  These response codes are applicable to any
   request method.

15.5.1.  500 Internal Server Error

   The server encountered an unexpected condition which prevented it
   from fulfilling the request.

15.5.2.  501 Not Implemented

   The server does not support the functionality required to fulfill the
   request.  This is the appropriate response when the server does not
   recognize the request method and is not capable of supporting it for
   any resource.

15.5.3.  502 Bad Gateway

   The server, while acting as a gateway or proxy, received an invalid
   response from the upstream server it accessed in attempting to
   fulfill the request.

15.5.4.  503 Service Unavailable

   The server is currently unable to handle the request due to a
   temporary overloading or maintenance of the server.  The implication
   is that this is a temporary condition which will be alleviated after
   some delay.  If known, the length of the delay MAY be indicated in a
   Retry-After header.  If no Retry-After is given, the client SHOULD
   handle the response as it would for a 500 response.



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         Note: The existence of the 503 status code does not imply that
         a server must use it when becoming overloaded.  Some servers
         may wish to simply refuse the connection.

15.5.5.  504 Gateway Timeout

   The server, while acting as a proxy, did not receive a timely
   response from the upstream server specified by the URI or some other
   auxiliary server (e.g.  DNS) it needed to access in attempting to
   complete the request.

15.5.6.  505 RTSP Version Not Supported

   The server does not support, or refuses to support, the RTSP protocol
   version that was used in the request message.  The server is
   indicating that it is unable or unwilling to complete the request
   using the same major version as the client other than with this error
   message.  The response SHOULD contain an message body describing why
   that version is not supported and what other protocols are supported
   by that server.

15.5.7.  551 Option not supported

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

























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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
   informational 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 MUST
   NOT do any changes to that header, all allowed operations are
   explicitly stated:

   a:    A proxy can add or concatenate the header field if not present.

   m:    A proxy can modify an existing header field value.

   d:    A proxy can delete a header field value.

   r:    A proxy needs to be able to read the header field, and thus
         this header field cannot be encrypted.

   The rest of the columns relate to the presence of a header field in a
   method.  The method names when abbreviated, are according to Table 8:

   c:    Conditional; requirements on the header field depend on the
         context of the message.

   m:    The header field is mandatory.

   m*:   The header field SHOULD be sent, but clients/servers need to be
         prepared to receive messages without that header field.

   o:    The header field is optional.

   *:    The header field MUST be present if the message body is not
         empty.  See Section 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 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 MUST 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
   quotes (").  Any URI parameters are contained within these quotes.
   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   |
   |                |      |     |     |     |      |     |      |     |
   | MTag           | 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 | OP | SETU | PLAY | PAUSE | TRD |
   |              |       | y    | S  | T  | P    |      |       |     |
   +--------------+-------+------+----+----+------+------+-------+-----+
   | Media-       |       |      | -  | -  | r    | r    | r     | -   |
   | Properties   |       |      |    |    |      |      |       |     |
   |              |       |      |    |    |      |      |       |     |
   | Media- Range |       |      | -  | -  | r    | r    | r     | -   |
   |              |       |      |    |    |      |      |       |     |
   | Pipelined-   |       | amdr | -  | o  | o    | o    | o     | o   |
   | Requests     |       |      |    |    |      |      |       |     |
   |              |       |      |    |    |      |      |       |     |
   | Proxy-       | 407   | amr  | m  | m  | m    | m    | m     | m   |
   | Authenticate |       |      |    |    |      |      |       |     |
   |              |       |      |    |    |      |      |       |     |
   | Proxy-       | R     | rd   | o  | o  | o    | o    | o     | o   |
   | Authorizatio |       |      |    |    |      |      |       |     |
   | n            |       |      |    |    |      |      |       |     |
   |              |       |      |    |    |      |      |       |     |




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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     | -   |
   |              |       |      |    |    |      |      |       |     |
   | Terminate-Re | R     | r    | -  | -  | -    | -    | -     | -   |
   | ason         |       |      |    |    |      |      |       |     |
   |              |       |      |    |    |      |      |       |     |
   | Referer      | R     |      | o  | o  | o    | o    | o     | o   |
   |              |       |      |    |    |      |      |       |     |
   | Request-     | R     |      | -  | -  | -    | -    | -     | -   |
   | Status       |       |      |    |    |      |      |       |     |
   |              |       |      |    |    |      |      |       |     |
   | Require      | R     |      | o  | o  | o    | o    | o     | o   |
   |              |       |      |    |    |      |      |       |     |
   | Retry-After  | 3rr,5 |      | o  | o  | o    | -    | -     | -   |
   |              | 03    |      |    |    |      |      |       |     |
   |              |       |      |    |    |      |      |       |     |
   | Retry-After  | 413   |      | o  | o  | o    | o    | o     | o   |
   |              |       |      |    |    |      |      |       |     |
   | 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  | -    | -    | -     | -   |



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   | Server       | r     | r    | o  | o  | o    | o    | o     | o   |
   |              |       |      |    |    |      |      |       |     |
   | Speed        |       |      | -  | -  | -    | o    | -     | -   |
   |              |       |      |    |    |      |      |       |     |
   | 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    | -    | -     | -   |
   |              |       |      |    |    |      |      |       |     |
   | Unsupported  | r     |      | c  | c  | c    | c    | c     | c   |
   |              |       |      |    |    |      |      |       |     |
   | 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   |
   | Authenticate |       |      |    |    |      |      |       |     |
   +--------------+-------+------+----+----+------+------+-------+-----+

     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   | -   |
   |                        |         |       |     |     |     |     |



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   | Content-Base           | R       |       | o   | o   | -   | -   |
   |                        |         |       |     |     |     |     |
   | 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   | -   | -   | -   |
   |                        |         |       |     |     |     |     |



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   | Location               | 3rr     |       | o   | o   | o   | -   |
   |                        |         |       |     |     |     |     |
   | 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   |
    |                  |             |       |     |     |     |     |
    | Terminate-Reason | R           | r     | -   | -   | m   | -   |
    |                  |             |       |     |     |     |     |
    | Referer          | R           |       | o   | o   | o   | -   |
    |                  |             |       |     |     |     |     |
    | Request-Status   | R           |       | -   | -   | -   | m   |
    |                  |             |       |     |     |     |     |
    | Require          | R           | r     | o   | o   | o   | -   |
    |                  |             |       |     |     |     |     |
    | Retry-After      | 3rr,413,503 |       | o   | o   | -   | -   |
    |                  |             |       |     |     |     |     |
    | Retry-After      | 413         |       | o   | o   | o   | o   |



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    | Scale            |             |       | -   | -   | -   | c   |
    |                  |             |       |     |     |     |     |
    | Seek-Style       |             |       | -   | -   | -   | -   |
    |                  |             |       |     |     |     |     |
    | Session          | R           | r     | o   | o   | o   | m   |
    |                  |             |       |     |     |     |     |
    | 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 Section 20.2.3 for the syntax.

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



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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
   MUST NOT be included in server to client requests.

   In a request the header MUST contain the method (User, Proxy, or Any)
   for approving credentials selected by the requester.  The method MUST
   NOT be changed by any proxy, unless it is "proxy" when a proxy MAY
   change it to "user" to take the role of user approving each further
   hop.  If the method is "User" the header contains zero or more of
   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 MUST 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 MUST implement the
   SHA-256[FIPS-pub-180-2] algorithm for computation of the hash of the
   DER encoded certificate.  The SHA-256 algorithm is identified by the
   token "sha-256".

   The intention with allowing for other hash algorithms is to enable
   the future retirement of algorithms that are not implemented
   somewhere else than here.  Thus the definition of future algorithms
   for this purpose is intended to be extremely limited.  A feature tag
   can be used to ensure that support for the replacement algorithm
   exist.

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

16.3.  Accept-Encoding

   The Accept-Encoding request-header field is similar to Accept, but
   restricts the content-codings that are acceptable in the response.

   A server tests whether a content-coding is acceptable, according to
   an Accept-Encoding field, using these rules:

   1.  If the content-coding is one of the content-codings listed in the
       Accept-Encoding field, then it is acceptable, unless it is
       accompanied by a qvalue of 0.  (As defined in section 3.9, a
       qvalue of 0 means "not acceptable.")




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   2.  The special "*" symbol in an Accept-Encoding field matches any
       available content-coding not explicitly listed in the header
       field.

   3.  If multiple content-codings are acceptable, then the acceptable
       content-coding with the highest non-zero qvalue is preferred.

   4.  The "identity" content-coding is always acceptable, unless
       specifically refused because the Accept-Encoding field includes
       "identity;q=0", or because the field includes "*;q=0" and does
       not explicitly include the "identity" content-coding.  If the
       Accept-Encoding field-value is empty, then only the "identity"
       encoding is acceptable.

   If an Accept-Encoding field is present in a request, and if the
   server cannot send a response which is acceptable according to the
   Accept-Encoding header, then the server SHOULD send an error response
   with the 406 (Not Acceptable) status code.

   If no Accept-Encoding field is present in a request, the server MAY
   assume that the client will accept any content coding.  In this case,
   if "identity" is one of the available content-codings, then the
   server SHOULD use the "identity" content-coding, unless it has
   additional information that a different content-coding is meaningful
   to the client.

16.4.  Accept-Language

   The Accept-Language request-header field is similar to Accept, but
   restricts the set of natural languages that are preferred as a
   response to the request.  Note that the language specified applies to
   the presentation description and any reason phrases, but not the
   media content.

   A language tag identifies a natural language spoken, written, or
   otherwise conveyed by human beings for communication of information
   to other human beings.  Computer languages are explicitly excluded.
   The syntax and registry of RTSP 2.0 language tags is the same as that
   defined by [RFC4646].

   Each language-range MAY be given an associated quality value which
   represents an estimate of the user's preference for the languages
   specified by that range.  The quality value defaults to "q=1".  For
   example:







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      Accept-Language: da, en-gb;q=0.8, en;q=0.7

   would mean: "I prefer Danish, but will accept British English and
   other types of English."  A language-range matches a language-tag if
   it exactly equals the tag, or if it exactly equals a prefix of the
   tag such that the first tag character following the prefix is "-".
   The special range "*", if present in the Accept-Language field,
   matches every tag not matched by any other range present in the
   Accept-Language field.

      Note: This use of a prefix matching rule does not imply that
      language tags are assigned to languages in such a way that it is
      always true that if a user understands a language with a certain
      tag, then this user will also understand all languages with tags
      for which this tag is a prefix.  The prefix rule simply allows the
      use of prefix tags if this is the case.

   The language quality factor assigned to a language-tag by the Accept-
   Language field is the quality value of the longest language- range in
   the field that matches the language-tag.  If no language- range in
   the field matches the tag, the language quality factor assigned is 0.
   If no Accept-Language header is present in the request, the server
   SHOULD assume that all languages are equally acceptable.  If an
   Accept-Language header is present, then all languages which are
   assigned a quality factor greater than 0 are acceptable.

16.5.  Accept-Ranges

   The Accept-Ranges request and response-header field allows indication
   of the format supported in the Range header.  The client MUST include
   the header in SETUP requests to indicate which formats it support to
   receive in PLAY and PAUSE responses, and REDIRECT requests.  The
   server MUST include the header in SETUP and 456 error responses to
   indicate the formats supported for the resource indicated by the
   request URI.  The header MAY be included in GET_PARAMETER request and
   response pairs.  The GET_PARAMETER request MUST contain a Session
   header to identify the session context the request are related to.
   The requester and responder will indicate their capabilities
   regarding Range formats respectively.
      Accept-Ranges: NPT, SMPTE

   The syntax is defined in Section 20.2.3.

16.6.  Allow

   The Allow message-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



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   resource.  An Allow header field MUST be present in a 405 (Method Not
   Allowed) response.  The Allow header MUST also be present in all
   OPTIONS responses where the content of the header will not include
   exactly the same methods as listed in the Public header.

   The Allow MUST 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

   An RTSP client that wishes to authenticate itself with a server,
   usually, but not necessarily, after receiving a 401 response, does so
   by including an Authorization request-header field with the request.
   The Authorization field value consists of credentials containing the
   authentication information of the user agent for the realm of the
   resource being requested.

   If a request is authenticated and a realm specified, the same
   credentials SHOULD be valid for all other requests within this realm
   (assuming that the authentication scheme itself does not require
   otherwise, such as credentials that vary according to a challenge
   value or using synchronized clocks).

   When a shared cache (see Section 18) receives a request containing an
   Authorization field, it MUST NOT return the corresponding response as
   a reply to any other request, unless one of the following specific
   exceptions holds:

   1.  If the response includes the "maxage" cache-control directive,
       the cache MAY use that response in replying to a subsequent
       request.  But (if the specified maximum age has passed) a proxy
       cache MUST first revalidate it with the origin server, using the
       request-headers from the new request to allow the origin server
       to authenticate the new request.  (This is the defined behavior
       for maxage.)  If the response includes "maxage=0", the proxy MUST
       always revalidate it before re-using it.

   2.  If the response includes the "must-revalidate" cache-control
       directive, the cache MAY use that response in replying to a
       subsequent request.  But if the response is stale, all caches
       MUST first revalidate it with the origin server, using the
       request-headers from the new request to allow the origin server
       to authenticate the new request.




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   3.  If the response includes the "public" cache-control directive, it
       MAY be returned in reply to any subsequent request.

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
   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.






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   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
         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.




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   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.

   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.






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16.11.  Connection

   The Connection general-header field allows the sender to specify
   options that are desired for that particular connection and MUST NOT
   be communicated by proxies over further connections.

   RTSP 2.0 proxies MUST parse the Connection header field before a
   message is forwarded and, for each connection-token in this field,
   remove any header field(s) from the message with the same name as the
   connection-token.  Connection options are signaled by the presence of
   a connection-token in the Connection header field, not by any
   corresponding additional header field(s), since the additional header
   field may not be sent if there are no parameters associated with that
   connection option.

   Message headers listed in the Connection header MUST NOT include end-
   to-end headers, such as Cache-Control.

   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
   requester.  It MUST only be used in server to client responses.

   The Connection-Credentials header in an RTSP response MUST, if
   included, contain the credential information (in form of a list of
   certificates providing the chain of certification) of the next hop
   that an intermediary needs to securely connect to.  The header MUST
   include the URI of the next hop (proxy or server) and a base64
   [RFC4648] encoded binary structure containing a sequence of DER
   encoded X.509v3 certificates[RFC5280] .

   The binary structure starts with the number of certificates
   (NR_CERTS) included as a 16 bit unsigned integer.  This is followed
   by NR_CERTS number of 16 bit unsigned integers providing the size in
   octets of each DER encoded certificate.  This is followed by NR_CERTS
   number of DER encoded X.509v3 certificates in a sequence (chain).
   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



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   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 certificates       | Size of certificate #1        |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       | Size of certificate #2        | Size of certificate #3        |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       : DER Encoding of Certificate #1                                :
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       : DER Encoding of Certificate #2                                :
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       : DER Encoding of Certificate #3                                :
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

16.13.  Content-Base

   The Content-Base message-header field may be used to specify the base
   URI for resolving relative URIs within the message body.
   Content-Base: rtsp://media.example.com/movie/twister
   If no Content-Base field is present, the base URI of an message body
   is defined either by its Content-Location (if that Content-Location
   URI is an absolute URI) or the URI used to initiate the request, in
   that order of precedence.  Note, however, that the base URI of the
   contents within the message-body may be redefined within that
   message-body.

16.14.  Content-Encoding

   The Content-Encoding header field is used as a modifier to the media-
   type.  When present, its value indicates what additional content
   codings have been applied to the message body, and thus what decoding
   mechanisms must be applied in order to obtain the media-type
   referenced by the Content-Type header field.  Content-Encoding is
   primarily used to allow a document to be compressed without losing
   the identity of its underlying media type.

   The content-coding is a characteristic of the entity identified by
   the Request-URI.  Typically, the message body is stored with this
   encoding and is only decoded before rendering or analogous usage.



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   However, a non-transparent proxy MAY modify the content-coding if the
   new coding is known to be acceptable to the recipient, unless the
   "no-transform" cache-control directive is present in the message.

   If the content-coding of an message body is not "identity", then the
   response MUST include a Content-Encoding entity-header that lists the
   non-identity content-coding(s) used.

   If the content-coding of an message body in a request message is not
   acceptable to the origin server, the server SHOULD respond with a
   status code of 415 (Unsupported Media Type).

   If multiple encodings have been applied to a message body, the
   content codings MUST be listed in the order in which they were
   applied.  Additional information about the encoding parameters MAY be
   provided by other header fields not defined by this specification.

16.15.  Content-Language

   The Content-Language header field describes the natural language(s)
   of the intended audience for the enclosed message body.  Note that
   this might not be equivalent to all the languages used within the
   message body.

   Language tags are mentioned in Section 16.4.  The primary purpose of
   Content-Language is to allow a user to identify and differentiate
   entities according to the user's own preferred language.  Thus, if
   the body content is intended only for a Danish-literate audience, the
   appropriate field is

      Content-Language: da

   If no Content-Language is specified, the default is that the content
   is intended for all language audiences.  This might mean that the
   sender does not consider it to be specific to any natural language,
   or that the sender does not know for which language it is intended.

   Multiple languages MAY be listed for content that is intended for
   multiple audiences.  For example, a rendition of the "Treaty of
   Waitangi," presented simultaneously in the original Maori and English
   versions, would call for

      Content-Language: mi, en

   However, just because multiple languages are present within an entity
   does not mean that it is intended for multiple linguistic audiences.
   An example would be a beginner's language primer, such as "A First
   Lesson in Latin," which is clearly intended to be used by an English-



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   literate audience.  In this case, the Content-Language would properly
   only include "en".

   Content-Language MAY be applied to any media type -- it is not
   limited to textual documents.

16.16.  Content-Length

   The Content-Length general-header field contains the length of the
   message body of the RTSP message (i.e. after the double CRLF
   following the last header).  Unlike HTTP, it MUST be included in all
   messages that carry a message body beyond the header portion of the
   RTSP message.  If it is missing, a default value of zero is assumed.
   Any Content-Length greater than or equal to zero is a valid value.

16.17.  Content-Location

   The Content-Location header field MAY be used to supply the resource
   location for the entity enclosed in the message when that entity is
   accessible from a location separate from the requested resource's
   URI.  A server SHOULD provide a Content-Location for the variant
   corresponding to the response entity; especially in the case where a
   resource has multiple entities associated with it, and those entities
   actually have separate locations by which they might be individually
   accessed, the server SHOULD provide a Content-Location for the
   particular variant which is returned.

   The Content-Location value is not a replacement for the original
   requested URI; it is only a statement of the location of the resource
   corresponding to this particular entity at the time of the request.
   Future requests MAY specify the Content-Location URI as the request-
   URI if the desire is to identify the source of that particular
   entity.

   A cache cannot assume that an entity with a Content-Location
   different from the URI used to retrieve it can be used to respond to
   later requests on that Content-Location URI.  However, the Content-
   Location can be used to differentiate between multiple entities
   retrieved from a single requested resource.

   If the Content-Location is a relative URI, the relative URI is
   interpreted relative to the Request-URI.

16.18.  Content-Type

   The Content-Type header indicates the media type of the message body
   sent to the recipient.  Note that the content types suitable for RTSP
   are likely to be restricted in practice to presentation descriptions



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   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 MUST 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

   The Date header field represents the date and time at which the
   message was originated.  The inclusion of the Date header in RTSP
   message follows these rules:

   o  An RTSP message, sent either by the client or the server,
      containing a body MUST include a Date header, if the sending host
      has a clock;

   o  Clients and servers are RECOMMENDED to include a Date header in
      all other RTSP messages, if the sending host has a clock;

   o  If the server does not have a clock that can provide a reasonable
      approximation of the current time, its responses MUST NOT include
      a Date header field.  In this case, this rule MUST be followed:
      Some origin server implementations might not have a clock
      available.  An origin server without a clock MUST NOT assign
      Expires or Last- Modified values to a response, unless these
      values were associated with the resource by a system or user with
      a reliable clock.  It MAY assign an Expires value that is known,
      at or before server configuration time, to be in the past (this
      allows "pre-expiration" of responses without storing separate



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      Expires values for each resource).

   A received message that does not have a Date header field MUST be
   assigned one by the recipient if the message will be cached by that
   recipient .  An RTSP implementation without a clock MUST NOT cache
   responses without revalidating them on every use.  An RTSP cache,
   especially a shared cache, SHOULD use a mechanism, such as NTP, to
   synchronize its clock with a reliable external standard.

   The RTSP-date sent in a Date header SHOULD NOT represent a date and
   time subsequent to the generation of the message.  It SHOULD
   represent the best available approximation of the date and time of
   message generation, unless the implementation has no means of
   generating a reasonably accurate date and time.  In theory, the date
   ought to represent the moment just before the entity is generated.
   In practice, the date can be generated at any time during the message
   origination without affecting its semantic value.

16.21.  Expires

   The Expires message-header field gives a date and time after which
   the description or media-stream should be considered stale.  The
   interpretation depends on the method:

   DESCRIBE response:  The Expires header indicates a date and time
         after which the presentation description (body) SHOULD be
         considered stale.

   SETUP response:  The Expires header indicate a date and time after
         which the media stream SHOULD be considered stale.

   A stale cache entry may not normally be returned by a cache (either a
   proxy cache or an user agent cache) unless it is first validated with
   the origin server (or with an intermediate cache that has a fresh
   copy of the message body).  See Section 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 RTSP-date:

   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



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   (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.

16.22.  From

   The From request-header field, if given, SHOULD contain an Internet
   e-mail address for the human user who controls the requesting user
   agent.  The address SHOULD be machine-usable, as defined by "mailbox"
   in [RFC1123].

   This header field MAY be used for logging purposes and as a means for
   identifying the source of invalid or unwanted requests.  It SHOULD
   NOT be used as an insecure form of access protection.  The
   interpretation of this field is that the request is being performed
   on behalf of the person given, who accepts responsibility for the
   method performed.  In particular, robot agents SHOULD include this
   header so that the person responsible for running the robot can be
   contacted if problems occur on the receiving end.

   The Internet e-mail address in this field MAY be separate from the
   Internet host which issued the request.  For example, when a request
   is passed through a proxy the original issuer's address SHOULD be
   used.

   The client SHOULD NOT send the From header field without the user's
   approval, as it might conflict with the user's privacy interests or
   their site's security policy.  It is strongly recommended that the
   user be able to disable, enable, and modify the value of this field
   at any time prior to a request.

16.23.  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 MTag 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 MTag



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   validation check fails, MUST response using 412 (Precondition
   Failed).

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

16.24.  If-Modified-Since

   The If-Modified-Since request-header field is used with the DESCRIBE
   and SETUP methods to make them conditional.  If the requested variant
   has not been modified since the time specified in this field, a
   description will not be returned from the server (DESCRIBE) or a
   stream will not be set up (SETUP).  Instead, a 304 (Not Modified)
   response MUST be returned without any message-body.

   An example of the field is:
     If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT

16.25.  If-None-Match

   This request header can be used with one or several message body tags
   to make DESCRIBE requests conditional.  A client that has one or more
   message bodies previously obtained from the resource, can verify that
   none of those entities is current by including a list of their
   associated message body tags in the If-None-Match header field.  The
   purpose of this feature is to allow efficient updates of cached
   information with a minimum amount of transaction overhead.  As a
   special case, the value "*" matches any current entity of the
   resource.

   If any of the message body tags match the message body tag of the
   message body that would have been returned in the response to a
   similar DESCRIBE request (without the If-None-Match header) on that
   resource, or if "*" is given and any current entity exists for that
   resource, then the server MUST NOT perform the requested method,
   unless required to do so because the resource's modification date
   fails to match that supplied in an If-Modified-Since header field in
   the request.  Instead, if the request method was DESCRIBE, the server
   SHOULD respond with a 304 (Not Modified) response, including the
   cache- related header fields (particularly MTag) of one of the
   message bodies that matched.  For all other request methods, the
   server MUST respond with a status of 412 (Precondition Failed).

   See Section 18.1.3 for rules on how to determine if two message body
   tags match.

   If none of the message body tags match, then the server MAY perform
   the requested method as if the If-None-Match header field did not



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   exist, but MUST also ignore any If-Modified-Since header field(s) in
   the request.  That is, if no message body tags match, then the server
   MUST NOT return a 304 (Not Modified) response.

   If the request would, without the If-None-Match header field, result
   in anything other than a 2xx or 304 status, then the If-None-Match
   header MUST be ignored.  (See Section 18.1.4 for a discussion of
   server behavior when both If-Modified-Since and If-None-Match appear
   in the same request.)

   The meaning of "If-None-Match: *" is that the method MUST NOT be
   performed if the representation selected by the origin server (or by
   a cache, possibly using the Vary mechanism, see Section 16.55)
   exists, and SHOULD be performed if the representation does not exist.
   This feature is intended to be useful in preventing races between PUT
   operations.

   The result of a request having both an If-None-Match header field and
   either an If-Match or an If-Unmodified-Since header fields is
   undefined by this specification.

16.26.  Last-Modified

   The Last-Modified message-header field indicates the date and time at
   which the origin server believes the presentation description or
   media stream was last modified.  For the method DESCRIBE, the header
   field indicates the last modification date and time of the
   description, for SETUP that of the media stream.

   An origin server MUST NOT send a Last-Modified date which is later
   than the server's time of message origination.  In such cases, where
   the resource's last modification would indicate some time in the
   future, the server MUST replace that date with the message
   origination date.

   An origin server SHOULD obtain the Last-Modified value of the entity
   as close as possible to the time that it generates the Date value of
   its response.  This allows a recipient to make an accurate assessment
   of the entity's modification time, especially if the entity changes
   near the time that the response is generated.

   RTSP servers SHOULD send Last-Modified whenever feasible.

16.27.  Location

   The Location response-header field is used to redirect the recipient
   to a location other than the Request-URI for completion of the
   request or identification of a new resource.  For 3xx responses, the



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   location SHOULD indicate the server's preferred URI for automatic
   redirection to the resource.  The field value consists of a single
   absolute URI.

   Note: The Content-Location header field (Section 16.17) differs from
   Location in that the Content-Location identifies the original
   location of the entity enclosed in the request.  It is therefore
   possible for a response to contain header fields for both Location
   and Content-Location.  Also see Section 18.2 for cache requirements
   of some methods.

16.28.  Media-Properties

   This general header is used in SETUP response or PLAY_NOTIFY requests
   to indicate the media's properties that currently are applicable to
   the RTSP session.  PLAY_NOTIFY MAY be used to modify these properties
   at any point.  However, the client SHOULD have received the update
   prior to any action related to the new media properties take affect.
   For aggregated sessions the Media-Properties header will be returned
   in each SETUP response.  The header received in the latest response
   is the one that applies on the whole session from this point until
   any future update.  The header MAY be included without value in
   GET_PARAMETER requests to the server with a Session header included
   to query the current Media-Properties for the session.  The responder
   MUST include the current session's media properties.

   The media properties expressed by this header is the one applicable
   to all media in the RTSP session.  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
   multiple properties SHOULD NOT be used.  The list of properties that
   can be expressed MAY be extended at any time.  Unknown property
   values MUST be ignored.

   This specification defines the following 4 groups and their property
   values:







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   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 beginning only.

      No-Seeking:  No seeking is possible.

   Content Modifications

      Immutable:  The content will not be changed during the life-time
         of the RTSP session.

      Dynamic:  The content may be changed based on external methods or
         triggers

      Time-Progressing  The media accessible progress as wallclock time
         progresses.

   Retention

      Unlimited:  Content will be retained for the duration of the life-
         time of the RTSP session.

      Time-Limited:  Content will be retained at least until the
         specified wallclock time.  The time must be provided in the
         absolute time format specified in Section 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.

   Supported Scale:

      Scales:  A quoted comma separated list of one or more decimal
         values or ranges of scale values supported by the content.  A
         range has a start and stop value separated by a colon.  A range
         indicates that the content supports fine grained selection of
         scale values.  Fine grained allows for steps at least as small
         as one tenth of a scale value.  Negative values are supported.
         The value 0 have no meaning and must not be used.




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

   On-demand:
   Media-Properties: Random-Access=2.5s, Unlimited, Immutable,
        Scales="-20, -10, -4, 0.5:1.5, 4, 8, 10, 15, 20"

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

16.29.  Media-Range

   The Media-Range general header is used to give the range of the media
   at the time of sending the RTSP message.  This header MUST be
   included in SETUP response, and PLAY and PAUSE response for media
   that are Time-Progressing, and PLAY and PAUSE response after any
   change for media that are Dynamic, and in PLAY_NOTIFY request that
   are sent due to Media-Property-Update.  Media-Range header without
   any range specifications MAY be included in GET_PARAMETER requests to
   the server to request the current range.  The server MUST in this
   case include the current range at the time of sending the response.

   The header MUST include range specifications for all time formats
   supported for the media, as indicated in Accept-Ranges header
   (Section 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-continuous 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 wallclock progresses so will also the media range.
   However it shall be assumed that media time progress in direct
   relationship to wallclock time (with the exception of clock skew) so
   that a reasonably accurate estimation of the media range can be
   calculated.

16.30.  MTag

   The MTag response header MAY be included in DESCRIBE or SETUP
   responses.  The message body 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 MTag is provided both inside the message body, e.g. within the



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   "a=mtag" attribute in SDP, and in the response message, then both
   tags MUST be identical.  It is RECOMMENDED that the MTag is primarily
   given in the RTSP response message, to ensure that caches can use the
   MTag without requiring content inspection.  However for session
   descriptions that are distributed outside of RTSP, for example using
   HTTP, etc. it will be necessary to include the message body tag in
   the session description as specified in Appendix D.1.9.

   SETUP and DESCRIBE requests can be made conditional upon the MTag
   using the headers If-Match (Section 16.23) and If-None-Match (
   Section 16.25).

16.31.  Notify-Reason

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

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 does not 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.

   Upon receiving a request with the Pipelined-Requests the responding
   entity MUST look up if there exist a binding between this Pipelined-
   Requests identifier for the current persistent connection and an RTSP
   session ID.  If that exists then the received request is processed
   the same way as if it 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 MUST create
   a binding upon the successful completion of a session creating
   request, i.e.  SETUP.  If the request failed to create an RTSP
   session no binding MUST be created.  In case the request contains
   both a Session header and the Pipelined-Requests header the
   Pipelined-Requests MUST be ignored.

   Note: Based on the above definition at least the first request
   containing a new unique Pipelined-Requests will be required to be a
   SETUP request (unless the protocol is extended with new methods of
   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.




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   For all responses to request that contained the Pipelined-Requests,
   the Session header and the Pipelined-Requests MUST 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 created with this method.  A server
   side binding MUST 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

   The Proxy-Authenticate response-header field MUST be included as part
   of a 407 (Proxy Authentication Required) response.  The field value
   consists of a challenge that indicates the authentication scheme and
   parameters applicable to the proxy for this Request-URI.

   The HTTP access authentication process is described in [RFC2617].
   Unlike WWW-Authenticate, the Proxy-Authenticate header field applies
   only to the current connection and SHOULD NOT be passed on to
   downstream clients.  However, an intermediate proxy might need to
   obtain its own credentials by requesting them from the downstream
   client, which in some circumstances will appear as if the proxy is
   forwarding the Proxy-Authenticate header field.

16.34.  Proxy-Authorization

   The Proxy-Authorization request-header field allows the client to
   identify itself (or its user) to a proxy which requires
   authentication.  The Proxy-Authorization field value consists of
   credentials containing the authentication information of the user
   agent for the proxy and/or realm of the resource being requested.

   The HTTP access authentication process is described in [RFC2617].
   Unlike Authorization, the Proxy-Authorization header field applies
   only to the next outbound proxy that demanded authentication using
   the Proxy- Authenticate field.  When multiple proxies are used in a



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   chain, the Proxy-Authorization header field is consumed by the first
   outbound proxy that was expecting to receive credentials.  A proxy
   MAY relay the credentials from the client request to the next proxy
   if that is the mechanism by which the proxies cooperatively
   authenticate a given request.

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 MUST use the 551 (Option Not
   Supported) status code in the response.  Any feature-tag included in
   the Proxy-Require does not apply to the end-point (server or client).
   To ensure that a feature is supported by both proxies and servers the
   tag needs to be included in also a Require header.

   See Section 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
   MUST be added by any proxy if the Supported header is present in a
   request.  When present in a request, the receiver MUST in the
   response copy the received Proxy-Supported header.

   The Proxy-Supported header field contains a list of feature-tags
   applicable to proxies, as described in Section 4.7.  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 MUST 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 pro.example.com

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

     S->C:  RTSP/2.0 200 OK
            Supported: foo, bar, baz
            Proxy-Supported: proxy-foo, proxy-blech
            Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
            Via: 2.0 pro.example.com, 2.0 prox2.example.com

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 16.6) MAY be used to indicate methods allowed for a
   particular URI.

   Example of use:
      Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN

   In the event that there are proxies between the sender and the
   recipient of a response, each intervening proxy MUST modify the
   Public header field to remove any methods that are not supported via
   that proxy.  The resulting Public header field will contain an
   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.  It MAY be
   included in GET_PARAMETER request from he client to the server with
   only a Range format and no value to request the current media
   position independent if the session is in playing or ready state in
   the included format.  The server SHALL if supporting that range
   format respond with the current playing point or pause point as the
   start of the range.  If an explicit stop point was used in the
   previous PLAY request, then that value shall be included as stop
   point.  Note that if the server is currently under any type of media
   playback manipulation affecting the interpretation of Range, like
   Scale, that is also required to be included in any GET_PARAMETER
   response to provide complete information.

   The range can be specified in a number of units.  This specification
   defines smpte (Section 4.4), 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.

   Example:
     Range: clock=19960213T143205Z-

   The Range header contains a range of one single range format.  A
   range is a half-open interval with a start and an end point,
   including the start point, but excluding the end point.  A range may
   either be fully specified with explicit values for start point and
   end point, or have either start or end point be implicit.  An
   implicit start point indicates the session's pause point, and if no
   pause point is set the start of the content.  An implicit end point
   indicates the end of the content.  The usage of both implicit start
   and end point is not allowed in the same range header, however, the
   exclusion of the range header has that meaning, i.e. from pause point
   (or start) until end of content.

      Regarding the half-open intervals; a range of A-B starts exactly
      at time A, but ends just before B. Only the start time of a media
      unit such as a video or audio frame is relevant.  For example,
      assume that video frames are generated every 40 ms.  A range of
      10.0-10.1 would include a video frame starting at 10.0 or later



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      time and would include a video frame starting at 10.08, even
      though it lasted beyond the interval.  A range of 10.0-10.08, on
      the other hand, would exclude the frame at 10.08.

      Please note the difference between NPT time scales' "now" and an
      implicit start value.  Implicit value reference the current pause-
      point.  While "now" is the currently ongoing time.  In a time-
      progressing session with recording (retention for some or full
      time) the pause point may be 2 min into the session while now
      could be 1 hour into the session.

   By default, range intervals increase, where the second point is
   larger than the first point.

   Example:
       Range: npt=10-15

   However, range intervals can also decrease if the Scale header (see
   Section 16.44) indicates a negative scale value.  For example, this
   would be the case when a playback in reverse is desired.

   Example:
       Scale: -1
       Range: npt=15-10

   Decreasing ranges are still half open intervals as described above.
   Thus, for range A-B, A is closed and B is open.  In the above
   example, 15 is closed and 10 is open.  An exception to this rule is
   the case when B=0 in a decreasing range.  In this case, the range is
   closed on both ends, as otherwise there would be no way to reach 0 on
   a reverse playback for formats that have such a notion, like NPT and
   SMPTE.

   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

   The Referer request-header field allows the client to specify, for
   the server's benefit, the address (URI) of the resource from which
   the Request-URI was obtained (the "referrer", although the header
   field is misspelled.)  The URI refers to that of the presentation



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   description, typically retrieved via HTTP.  The Referer request-
   header allows a server to generate lists of back-links to resources
   for interest, logging, optimized caching, etc.  It also allows
   obsolete or mistyped links to be traced for maintenance.  The Referer
   field MUST NOT be sent if the Request-URI was obtained from a source
   that does not have its own URI, such as input from the user keyboard.

   If the field value is a relative URI, it SHOULD be interpreted
   relative to the Request-URI.  The URI MUST NOT include a fragment.

   See [H15.1.3] for security considerations on Referer.

16.40.  Retry-After

   The Retry-After response-header field can be used with a 503 (Service
   Unavailable) response to indicate how long the service is expected to
   be unavailable to the requesting client.  This field MAY also be used
   with any 3xx (Redirection) response to indicate the minimum time the
   user-agent is asked wait before issuing the redirected request.  The
   value of this field can be either an RTSP-date or an integer number
   of seconds (in decimal) after the time of the response.

   Example:
   Retry-After: Fri, 31 Dec 1999 23:59:59 GMT
   Retry-After: 120

   In the latter example, the delay is 2 minutes.

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 provides 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"

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



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   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 MUST 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 MUST ignore this header.  If a
   particular extension requires that intermediate devices support it,
   the extension should be tagged in the Proxy-Require field instead
   (see Section 16.35).  See discussion in the proxies section
   (Section 17.1) about when to consider that a feature requires proxy
   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



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   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
      particularly timely in their arrival.  Also functionality as
      informing the client from which packet a seek has occurred is
      affected.

   The RTP-Info MAY be included in SETUP responses to provide
   synchronization information when changing transport parameters, see
   Section 13.3.  The RTP-Info header MAY also be included in
   GET_PARAMETER requests from client to server without any value to
   indicate a request for this information.  In such a case the Range
   header MUST also be included in the request.  The server SHALL
   respond if the session is in playing state with the Range header
   filled in with the current playback point and with the corresponding
   RTP-Info values.

   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 MUST use
         the Request-URI as base URI.  This parameter MUST be present.

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

   seq:  Indicates the sequence number of the first packet of the stream
         that is direct result of the request.  This allows clients to
         gracefully deal with packets when seeking.  The client uses
         this value to differentiate packets that originated before the
         seek from packets that originated after the seek.  Note that a
         client may not receive the packet with the expressed sequence
         number, and instead packets with a higher sequence number, due
         to packet loss or reordering.  This parameter is RECOMMENDED to
         be present.

   rtptime:  MUST indicate the RTP timestamp value corresponding to the
         start time value in the Range response header, or if not
         explicitly given the implied start point.  The client uses this
         value to calculate the mapping of RTP time to NPT or other
         media timescale.  This parameter SHOULD be present to ensure
         inter-media synchronization is achieved.  There exist no



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         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 (wallclock) is
      available via RTCP.  However, this information is not sufficient
      to generate a mapping from RTP timestamps to media clock time
      (NPT, etc.).  Furthermore, in order to ensure that this
      information is available at the necessary time (immediately at
      startup or after a seek), and that it is delivered reliably, this
      mapping is placed in the RTSP control channel.

      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 content time increase at twice the playback time.  For every
   second of elapsed (wallclock) time, 2 seconds of content will be



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   delivered.  A negative value indicates reverse direction.  For
   certain media transports this may require certain considerations to
   work consistent, see Appendix C.1 for description on how RTP handles
   this.

   The transmitted data rate SHOULD NOT be changed by selection of a
   different scale value.  The resulting bit-rate should be in
   reasonably close to the nominal bit-rate of the content for Scale =
   1.  The server has to actively manipulate the data when needed to
   meet the bitrate constraints.  Implementation of scale changes
   depends on the 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, or completely mute the
   audio.

   The server and content may restrict the range of scale values that it
   supports.  The supported values are indicated by the Media-Properties
   header (Section 16.28).  The client SHOULD only indicate values
   indicated to be supported.  However, as the values may change as the
   content progresses a requested value may no longer be valid when the
   request arrives.  Thus an non-supported value in a request does not
   generate an error, only forces the server to choose the closest
   value.  The response MUST always contain the actual scale value
   chosen by the server.

   If the server does not implement the possibility to scale, it will
   not return a Scale header.  A server supporting Scale operations for
   PLAY MUST indicate this with the use of the "play.scale" feature-tag.

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




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   Seek-Style is a general header that MAY be included in any PLAY
   request to indicate the client's preference for any media stream that
   has random access properties.  The server MUST always include the
   header in any PLAY response for media with random access properties
   to indicate what policy was applied.  A Server that receives a
   unknown Seek-Style policy MUST ignore it and select the server
   default policy.

   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 continuous 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 continuous framed media that would mean the first next
      frame after the provided time.  For discrete media the first unit
      that is to be rendered after the provided time.  The main usage 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 MUST 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 severely degraded unless additional
   data is available as, for example, already buffered, or through other
   side channels.





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16.46.  Speed

   The Speed request-header field requests the server to deliver
   specific amounts of nominal media time per unit of delivery time,
   contingent on the server's ability and desire to serve the media
   stream at the given speed.  The client requests the delivery speed to
   be within a given range with an upper and lower bound.  The server
   SHALL delivery at the highest possible speed within the range, but
   not faster than the upper-bound, for which the underlying network
   path can support the resulting transport data rates.  As long as any
   speed value within the given range can be provided the server SHALL
   NOT modify the media quality.  Only if the server is unable to
   delivery media at the speed value provided by the lower bound shall
   it reduce the media quality.

   Implementation of the Speed functionality by the server is OPTIONAL.
   The server can indicate its support through a feature-tag,
   play.scale.  The lack of a Speed header in the response is an
   indication of lack of support of this functionality.

   The speed parameter values are expressed as a positive decimal value,
   e.g., a value of 2.0 indicates that data is to be delivered twice as
   fast as normal.  A speed value of zero is invalid.  The range is
   specified in the form "lower bound - upper bound".  The lower bound
   value may be smaller or equal to the upper bound.  All speeds may not
   be possible to support.  Therefore the server MAY modify the
   requested values to the closest supported.  The actual supported
   speed MUST be included in the response.  Note however that the use
   cases may vary and that Speed value ranges such as 0.7 - 0.8,
   0.3-2.0, 1.0-2.5, 2.5-2.5 all has their usage.

   Example:
     Speed: 1.0 - 2.5
   Use of this header changes the bandwidth used for data delivery.  It
   is meant for use in specific circumstances where delivery of the
   presentation at a higher or lower rate is desired.  The main use
   cases are buffer operations or local scale operations.  Implementors
   should keep in mind that bandwidth for the session may be negotiated
   beforehand (by means other than RTSP), and therefore re-negotiation
   may be necessary.  To perform Speed operations the server needs to
   ensure that the network path can support the resulting bit-rate.
   Thus the media transport needs to support feedback so that the server
   can react and adapt to the available bitrate.

16.47.  Server

   The Server response-header field contains information about the
   software used by the origin server to handle the request.  The field



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   can contain multiple product tokens and comments identifying the
   server and any significant subproducts.  The product tokens are
   listed in order of their significance for identifying the
   application.

   Example:
   Server: PhonyServer/1.0

   If the response is being forwarded through a proxy, the proxy
   application MUST NOT modify the Server response-header.  Instead, it
   SHOULD include a Via field (Section 16.56).

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, REDIRECT or timed out by the server.

   The session identifier is chosen by the server (see Section 4.3) and
   MUST be returned in the SETUP response.  Once a client receives a
   session identifier, it MUST be included in any request related to
   that session.  This means that the Session header MUST be included in
   a request using the following methods: PLAY, PAUSE, and TEARDOWN, and
   MAY be included in SETUP, OPTIONS, SET_PARAMETER, GET_PARAMETER, and
   REDIRECT, and MUST NOT be included in DESCRIBE.  In an RTSP response
   the session header MUST be included in methods, SETUP, PLAY, and
   PAUSE, and MAY be included in methods, TEARDOWN, and REDIRECT, and if
   included in the request of the following methods it MUST also be
   included in the response, OPTIONS, GET_PARAMETER, and SET_PARAMETER,
   and MUST NOT be included in DESCRIBE.

   Note that a session identifier identifies an RTSP session across
   transport sessions or connections.  RTSP requests for a given session
   can use different URIs (Presentation and media URIs).  Note, that
   there are restrictions depending on the session which URIs that are
   acceptable for a given method.  However, multiple "user" sessions for
   the same URI from the same client will require use of different
   session identifiers.

      The session identifier is needed to distinguish several delivery
      requests for the same URI coming from the same client.

   The response 454 (Session Not Found) MUST be returned if the session
   identifier is invalid.

   The header MAY include the session timeout period.  If not explicitly



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   provided this value is set to 60 seconds.  As this affects how often
   session keep-alives are needed values smaller than 30 seconds are not
   recommended.  However larger that default values can be useful in
   applications of RTSP that have inactive but established sessions for
   longer time periods.

      60 seconds was chosen as session timeout value due to: Resulting
      in not to frequent keep-alive messages and having low sensitivity
      to variations in request response timing.  If one reduces the
      timeout value to below 30 seconds the corresponding request
      response timeout becomes a significant part of the session
      timeout. 60 seconds also allows for reasonably rapid recovery of
      committed server resources in case of client failure.

16.49.  Supported

   The Supported header 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 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.  Terminate-Reason

   The Terminate-Reason request header allows the server when sending a
   REDIRECT or TERMINATE request to provide a reason for the session
   termination and any additional information.  This specification
   identifies three reasons for Redirections and may be extended in the
   future:

   Server-Admin:  The server needs to be shutdown for some
      administrative reason.






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   Session-Timeout:  A client's session is kept alive for extended
      periods of time and the server has determined that it needs to
      reclaim the resources associated with this session.

   Internal-Error  An internal error that is impossible to recover from
      has occurred forcing the server to terminate the session.

   The Server may provide additional parameters containing information
   around the redirect.  This specification defines the following ones.

   time:  Provides a wallclock time when the server will stop provide
      any service.

   user-msg:  An UTF-8 text string with a message from the server to the
      user.  This message SHOULD be displayed to the user.

16.51.  Timestamp

   The Timestamp general-header describes when the agent sent the
   request.  The value of the timestamp is of significance only to the
   agent and may use any timescale.  The responding agent MUST echo the
   exact same value and MAY, if it has accurate information about this,
   add a floating point number indicating the number of seconds that has
   elapsed since it has received the request.  The timestamp 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.

16.52.  Transport

   The Transport request and response header indicates which transport
   protocol is to be used and configures its parameters such as
   destination address, compression, multicast time-to-live and
   destination port for a single stream.  It sets those values not
   already determined by a presentation description.

   A Transport request header MAY contain a list of transport options
   acceptable to the client, in the form of multiple transport
   specification entries.  Transport specifications are comma separated,
   listed in decreasing order of preference.  Parameters may be added to
   each transport specification, separated by a semicolon.  The server
   MUST return a Transport response-header 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 MUST be
   responded using the status code 461 (Unsupported Transport)
   (Section 15.4.26).  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



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   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 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 need to be understood in a transport
   specification, if not, the transport specification MUST be ignored.
   RTSP proxies of any type that uses or modifies the transport
   specification, e.g. access proxy or security proxy, MUST remove
   specifications with unknown parameters before forwarding the RTSP
   message.  If that result in no remaining transport specification the
   proxy shall send a 461 (Unsupported Transport) (Section 15.4.26)
   response without any Transport header.

      The Transport header is restricted to describing a single media
      stream.  (RTSP can also control multiple streams as a single
      entity.)  Making it part of RTSP rather than relying on a
      multitude of session description formats greatly simplifies
      designs of firewalls.

   The general syntax for the transport specifier is a list of slash
   separated tokens:
   Value1/Value2/Value3...
   Which for RTP transports take the form:
   RTP/profile/lower-transport.

   The default value for the "lower-transport" parameters is specific to
   the profile.  For RTP/AVP, the default is UDP.

   There are two different methods for how to specify where the media
   should be delivered for unicast transport:

   dest_addr:  The presence of this parameter and its values indicates
         the destination address or addresses (host address and port
         pairs for IP flows) necessary for the media transport.

   No dest_addr:  The lack of the dest_addr parameter indicates that the
         server MUST send media to 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



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   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 Description Information:  The information
         included in the transport header can all be coming from the
         session description, e.g. the SDP c= and m= line.  This
         mitigates some of the security issues of the previous methods
         as it is the session provider that picks the multicast group
         and scope.  The client MUST include the information if it is
         available in the session description.

   No dest_addr:  The behavior when no explicit multicast group is
         present in a request is not defined.

   An RTSP proxy will need to take care.  If the media is not desired to
   be routed through the proxy, the proxy will need to introduce the
   destination indication.

   Below are the configuration parameters associated with transport:

   General parameters:

   unicast / multicast:  This parameter is a mutually exclusive
         indication of whether unicast or multicast delivery will be
         attempted.  One of the two values MUST be specified.  Clients
         that are capable of handling both unicast and multicast
         transmission 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.







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   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 MUST 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
         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



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         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 interval, e.g., interleaved=4-5 in cases
         where the transport choice for the media stream requires it,
         e.g. for RTP with RTCP.  The channel number given in the
         request 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
         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.



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   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 MUST NOT be specified in requests.  The
         functionality of specifying the ssrc parameter in a SETUP
         request is deprecated as it is incompatible with the
         specification of RTP in RFC 3550[RFC3550].  If the parameter is
         included in the Transport header of a SETUP request, the server
         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 of 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-
         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



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         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).

   RTCP-mux:  Use to negotiate the usage of RTP and RTCP multiplexing
         [I-D.ietf-avt-rtp-and-rtcp-mux] on a single underlying
         transport stream.  The presence of this parameter in a SETUP
         request indicates the clients support and desire to use RTP and
         RTCP multiplexing.  The client MAY still include two transport
         streams in the Transport header specification to handle cases
         if RTP and RTCP multiplexing is not supported by the server.
         If the server supports the usage of RTP and RTCP multiplexing
         it SHALL include this parameter in the response and strip down
         the transport address negotiation to a single src_addr and
         dest_addr.  If the server does not support RTP and RTCP
         multiplexing is removes this parameter from the transport
         specification in response and treat the specification as if the
         parameter was not included.

   The combination of transport protocol, profile and lower transport
   needs to be defined.  A number of combinations are defined in the
   Appendix C.




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   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
           Media-Properties: Random-Access=0.6, Dynamic,
                             Time-Limited=20081128T165900

16.53.  Unsupported

   The Unsupported response-header 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 MUST NOT be forwarded.

   See Section 16.42 for a usage example.

16.54.  User-Agent

   The User-Agent request-header field contains information about the
   user agent originating the request.  This is for statistical
   purposes, the tracing of protocol violations, and automated
   recognition of user agents for the sake of tailoring responses to
   avoid particular user agent limitations.  User agents SHOULD include
   this field with requests.  The field can contain multiple product
   tokens and comments identifying the agent and any subproducts which
   form a significant part of the user agent.  By convention, the
   product tokens are listed in order of their significance for
   identifying the application.




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   Example:
   User-Agent: PhonyClient/1.2

16.55.  Vary

   The Vary field value indicates the set of request-header fields that
   fully determines, while the response is fresh, whether a cache is
   permitted to use the response to reply to a subsequent request
   without revalidation.  For uncacheable or stale responses, the Vary
   field value advises the user agent about the criteria that were used
   to select the representation.  A Vary field value of "*" implies that
   a cache cannot determine from the request headers of a subsequent
   request whether this response is the appropriate representation.

   An RTSP server SHOULD include a Vary header field with any cacheable
   response that is subject to server-driven negotiation.  Doing so
   allows a cache to properly interpret future requests on that resource
   and informs the user agent about the presence of negotiation on that
   resource.  A server MAY include a Vary header field with a non-
   cacheable response that is subject to server-driven negotiation,
   since this might provide the user agent with useful information about
   the dimensions over which the response varies at the time of the
   response.

   A Vary field value consisting of a list of field-names signals that
   the representation selected for the response is based on a selection
   algorithm which considers ONLY the listed request-header field values
   in selecting the most appropriate representation.  A cache MAY assume
   that the same selection will be made for future requests with the
   same values for the listed field names, for the duration of time for
   which the response is fresh.

   The field-names given are not limited to the set of standard request-
   header fields defined by this specification.  Field names are case-
   insensitive.

   A Vary field value of "*" signals that unspecified parameters not
   limited to the request-headers (e.g., the network address of the
   client), play a role in the selection of the response representation.
   The "*" value MUST NOT be generated by a proxy server; it may only be
   generated by an origin server.

16.56.  Via

   The Via general-header field MUST be used by proxies to indicate the
   intermediate protocols and recipients between the user agent and the
   server on requests, and between the origin server and the client on
   responses.  The field is intended to be used for tracking message



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   forwards, avoiding request loops, and identifying the protocol
   capabilities of all senders along the request/response chain.

   Multiple Via field values represents each proxy that has forwarded
   the message.  Each recipient MUST append its information such that
   the end result is ordered according to the sequence of forwarding
   applications.

   Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by
   default, forward the names and ports of hosts within the private/
   protected region.  This information SHOULD only be propagated if
   explicitly enabled.  If not enabled, the via-received of any host
   behind the firewall/NAT SHOULD be replaced by an appropriate
   pseudonym for that host.

   For organizations that have strong privacy requirements for hiding
   internal structures, a proxy MAY combine an ordered subsequence of
   Via header field entries with identical sent-protocol values into a
   single such entry.  Applications MUST NOT combine entries which have
   different received-protocol values.

16.57.  WWW-Authenticate

   The WWW-Authenticate response-header field MUST be included in 401
   (Unauthorized) response messages.  The field value consists of at
   least one challenge that indicates the authentication scheme(s) and
   parameters applicable to the Request-URI.

   The HTTP access authentication process is described in [RFC2617].
   User agents are advised to take special care in parsing the WWW-
   Authenticate field value as it might contain more than one challenge,
   or if more than one WWW-Authenticate header field is provided, the
   contents of a challenge itself can contain a comma-separated list of
   authentication parameters.

















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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 the description and media
         streams, i.e., the presentation, the proxy can serve a client
         with content, but without requesting it from the server once it
         has been cached and has not become stale.  See the caching
         Section 18.  This type of proxy is also expected to 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 an 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 deployments with private address spaces this proxy
         is likely to be combined with the access proxy.  Anyway, the
         functionality of this proxy is usually closely tied into
         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
   requester, as it is possible that low-level details of the stream
   description may have changed on the origin-server.

   Note that an RTSP cache, 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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18.1.   Validation Model (HTTP)

   When a cache has a stale entry that it would like to use as a
   response to a client's request, it first has to check with the origin
   server (or possibly an intermediate cache with a fresh response) to
   see if its cached entry is still usable.  We call this "validating"
   the cache entry.  Since we do not want to have to pay the overhead of
   retransmitting the full response if the cached entry is good, and we
   do not want to pay the overhead of an extra round trip if the cached
   entry is invalid, the RTSP protocol supports the use of conditional
   methods.

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

   The server then checks that validator against the current validator
   for the entity, and, if they match (see Section 18.1.3), it responds
   with a special status code (usually, 304 (Not Modified)) and no
   message body.  Otherwise, it returns a full response (including
   message body).  Thus, we avoid transmitting the full response if the
   validator matches, and we avoid an extra round trip if it does not
   match.

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

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

      Note: a response that lacks a validator may still be cached, and
      served from cache until it expires, unless this is explicitly
      prohibited by a cache-control directive (see Section 16.10).
      However, a cache cannot do a conditional retrieval if it does not
      have a validator for the entity, which means it will not be
      refreshable after it expires.







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18.1.1.  Last-Modified Dates

   The Last-Modified header (Section 16.26) value is often used as a
   cache validator.  In simple terms, a cache entry is considered to be
   valid if the entity has not been modified since the Last-Modified
   value.

18.1.2.  Message Body Tag Cache Validators

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

   Message body tags are described in Section 5.3

18.1.3.  Weak and Strong Validators

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

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

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








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

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

      Support for weak validators is optional.  However, weak validators
      allow for more efficient caching of equivalent objects; for
      example, a hit counter on a site is probably good enough if it is
      updated every few days or weeks, and any value during that period
      is likely "good enough" to be equivalent.

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

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

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

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

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

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

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

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





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

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

   OR

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

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

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

   OR

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

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

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

   This method relies on the fact that if two different responses were
   sent by the origin server during the same second, but both had the
   same Last-Modified time, then at least one of those responses would
   have a Date value equal to its Last-Modified time.  The arbitrary 60-
   second limit guards against the possibility that the Date and Last-
   Modified values are generated from different clocks, or at somewhat
   different times during the preparation of the response.  An
   implementation MAY use a value larger than 60 seconds, if it is
   believed that 60 seconds is too short.

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







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18.1.4.  Rules for When to Use Entity Tags and Last-Modified Dates

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

   RTSP origin servers:

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

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

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

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

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

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

   RTSP clients:

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

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




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

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

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

18.1.5.  Non-validating Conditionals

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

18.2.  Invalidation After Updates or Deletions (HTTP)

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

   There is no way for the RTSP protocol to guarantee that all such
   cache entries are marked invalid.  For example, the request that
   caused the change at the origin server might not have gone through
   the proxy where a cache entry is stored.  However, several rules help
   reduce the likelihood of erroneous behavior.

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

   Some HTTP methods MUST cause a cache to invalidate an entity.  This
   is either the entity referred to by the Request-URI, or by the



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   Location or Content-Location headers (if present).  These methods
   are:

   o  DESCRIBE

   o  SETUP

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

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





































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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 MUST 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 MUST follow the same guidelines with regards to TLS [RFC5246]
   usage as specified for HTTP, see [RFC2818].  RTSP over TLS is
   separated from unsecured RTSP both on 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 MUST be used for
   TLS over TCP/IP.

   When a client tries to setup an insecure channel to the server (using
   the "rtsp" URI), and the policy for the resource requires a secure
   channel, the server MUST redirect the client to the secure service by
   sending a 301 redirect response code together with the correct
   Location URI (using the "rtsps" scheme).  A user or client MAY
   upgrade a non secured URI to a secured by changing the scheme from
   "rtsp" to "rtsps".  A server implementing support for "rtsps" MUST
   allow this.

   It should be noted that TLS allows for mutual authentication (when
   using both server and client certificates).  Still, one of the more
   common 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 round trips).
   Still, when TLS is used, such behavior introduces extra active state
   in the server, not only for TCP and RTSP, but also for TLS.  This may
   increase the vulnerability to DoS attacks.

   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 MUST when
   initiating the next hop TLS connection use the incoming TLS
   connections cipher suite 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) MUST 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 MUST be used as default.  If
   another method than the "Proxy" is to be used, then the Accept-
   Credentials header MUST be included in all of the RTSP request from
   the client.  This is because it cannot be assumed that the proxy
   always keeps the TLS state or the users previous preference between
   different RTSP messages (in particular if the time interval between
   the messages is long).

   With the "Any" and "Proxy" methods the proxy will apply the policy as
   defined for respectively method.  If the policy does not accept the
   credentials of the next hop, the entity MUST respond with a message
   using status code 471 (Connection Credentials not accepted).

   An RTSP request in the direction server to client MUST NOT include
   the Accept-Credential header.  As for the non-secured communication,
   the possibility for these requests depends on the presence of a
   client established connection.  However if the server to client
   request is in relation to a session established over a TLS secured
   channel, it MUST be sent in a TLS secured connection.  That secured
   connection MUST also be the one used by the last client to server
   request.  If no such transport connection exist at the time when the
   server desires 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 following procedure
   for each RTSP request:

   o  Setup the TLS session to the next hop if not already present (i.e.
      run the TLS handshake, but do not send the RTSP request).

   o  Extract the peer certificate chain for the TLS session.

   o  Check if a matching identity and hash of the peer certificate is
      present in the Accept-Credentials header.  If present, send the
      message to the next hop, and conclude these procedures.  If not,
      go to the next step.

   o  The proxy responds to the RTSP request with a 470 or 407 response
      code.  The 407 response code MAY be used when the proxy requires
      both user and connection authorization from user or client.  In
      this message the proxy MUST include a Connection-Credentials
      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 requester 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 values can be folded onto multiple lines if the
   continuation line begins with a space or horizontal tab.  All linear
   white space, including folding, has the same semantics as SP.  A
   recipient MAY replace any linear white space with a single SP before
   interpreting the field value or forwarding the message downstream.
   This is intended to behave exactly as HTTP/1.1 as described in RFC
   2616 [RFC2616].  The SWS construct is used when linear white space is
   optional, generally between tokens and separators.

   To separate the header name from the rest of value, a colon is used,
   which, by the above rule, allows whitespace before, but no line
   break, and whitespace after, including a line break.  The HCOLON
   defines this construct.

   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
   MINUS    =  SWS "-" SWS ; minus/dash
   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-REQ-URI   =  schemes ":" URI-req-rest
   RTSP-URI-Ref   =  RTSP-URI / RTSP-Relative
   RTSP-REQ-Ref   =  RTSP-REQ-URI / RTSP-REQ-Rel
   schemes        =  "rtsp" / "rtsps" / scheme
   scheme         =  < As defined in RFC 3986>
   URI-rest       =  hier-part [ "?" query ] [ "#" fragment ]
   URI-req-rest   =  hier-part [ "?" query ]
                     ; Note fragment part not allowed in requests
   hier-part      =  "//" authority path-abempty

   RTSP-Relative  =  relative-part [ "?" query ] [ "#" fragment ]
   RTSP-REQ-Rel   =  relative-part [ "?" query ]
   relative-part  =  "//" authority path-abempty
                     / path-absolute
                     / path-noscheme
                     / path-empty

   authority      =  < As defined in RFC 3986>
   query          =  < As defined in RFC 3986>
   fragment       =  < 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  =  "smpte" 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
                   /  message-header) CRLF)
                   CRLF
                   [ message-body ]

   Response     = Status-Line
                  *((general-header
                  /  response-header
                  /  message-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-REQ-URI
   RTSP-Version =  "RTSP/" 1*DIGIT "." 1*DIGIT

   message-body = 1*OCTET

   Status-Code  =  "100"  ; Continue
                /  "200"  ; OK
                /  "301"  ; Moved Permanently
                /  "302"  ; Found
                /  "303"  ; See Other
                /  "304"  ; Not Modified
                /  "305"  ; Use Proxy
                /  "400"  ; Bad Request
                /  "401"  ; Unauthorized
                /  "402"  ; Payment Required



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                /  "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 Message Body Too Large
                /  "414"  ; Request-URI Too Large
                /  "415"  ; Unsupported Media Type
                /  "451"  ; Parameter Not Understood
                /  "452"  ; reserved
                /  "453"  ; Not Enough Bandwidth
                /  "454"  ; Session Not Found
                /  "455"  ; Method Not Valid in This State
                /  "456"  ; Header Field Not Valid for Resource
                /  "457"  ; Invalid Range
                /  "458"  ; Parameter Is Read-Only
                /  "459"  ; Aggregate operation not allowed
                /  "460"  ; Only aggregate operation allowed
                /  "461"  ; Unsupported Transport
                /  "462"  ; Destination Unreachable
                /  "463"  ; Destination Prohibited
                /  "464"  ; Data Transport Not Ready Yet
                /  "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
                   /  Terminate-Reason
                   /  Transport
                   /  User-Agent
                   /  extension-header












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

   message-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-type-range *(SEMI accept-param)
   media-type-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-REQ-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-REQ-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-REQ-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"

   Expires        =  "Expires" HCOLON RTSP-date
   From           =  "From" HCOLON from-spec
   from-spec      =  ( name-addr / addr-spec ) *( SEMI from-param )
   name-addr      =  [ display-name ] LAQUOT addr-spec RAQUOT
   addr-spec      =  RTSP-REQ-URI / absolute-URI
   absolute-URI   =  < As defined in RFC 3986>
   display-name   =  *(token LWS) / quoted-string
   from-param     =  tag-param / generic-param
   tag-param      =  "tag" EQUAL token
   If-Match       =  "If-Match" HCOLON ("*" / message-tag-list)
   message-tag-list =  message-tag *(COMMA message-tag)
   message-tag      =  [ weak ] opaque-tag
   weak             =  "W/"
   opaque-tag       =  quoted-string
   If-Modified-Since  =  "If-Modified-Since" HCOLON RTSP-date
   If-None-Match    =  "If-None-Match" HCOLON ("*" / message-tag-list)
   Last-Modified    =  "Last-Modified" HCOLON RTSP-date
   Location         =  "Location" HCOLON RTSP-REQ-URI
   Media-Properties = "Media-Properties" HCOLON [media-prop-list]
   media-prop-list  = media-prop-value *(COMMA media-prop-value)



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   media-prop-value = ("Random-Access" [EQUAL POS-FLOAT])
                    / "Begining-Only"
                    / "No-Seeking"
                    / "Immutable"
                    / "Dynamic"
                    / "Time-Progressing"
                    / "Unlimited"
                    / ("Time-Limited" EQUAL utc-range-spec)
                    / ("Time-Duration" EQUAL POS-FLOAT)
                    / ("Scales" EQUAL scale-value-list)
                    / media-prop-ext
   media-prop-ext   = token [EQUAL (1*rtsp-unreserved / quoted-string)]
   scale-value-list = DQ scale-entry *(COMMA scale-entry) DQ
   scale-entry      = scale-value / (scale-value COLON scale-value)
   scale-value      =  ["-"] 1*8DIGIT ["." 1*8DIGIT]
   Media-Range      = "Media-Range" HCOLON [ranges-list]
   ranges-list      =  ranges-spec *(COMMA ranges-spec)
   MTag             =  "MTag" HCOLON message-tag
   Notify-Reason    = "Notify-Reason" HCOLON Notify-Reas-val
   Notify-Reas-val  = "end-of-stream"
                    / "media-properties-update"
                    / "scale-change"
                    / Notify-Reason-extension
   Notify-Reason-extension  = token
   Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id
   startup-id  = 1*8DIGIT

























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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 RTSP-REQ-Ref
                          *(1*SP RTSP-REQ-Ref ) RDQUOT
  nonce                =  "nonce" EQUAL nonce-value
  nonce-value          =  quoted-string
  opaque               =  "opaque" EQUAL quoted-string
  stale                =  "stale" EQUAL ( "true" / "false" )
  algorithm            =  "algorithm" EQUAL ("MD5" / "MD5-sess" / token)
  qop-options          =  "qop" EQUAL LDQUOT qop-value
                          *("," qop-value) RDQUOT
  qop-value            =  "auth" / "auth-int" / token
  Proxy-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-spec

  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-REQ-Ref
  Request-Status       =  "Request-Status" HCOLON req-status-info
  req-status-info      =  cseq-info LWS status-info LWS reason-info
  cseq-info            =  "cseq" EQUAL cseq-nr
  status-info          =  "status" EQUAL Status-Code
  reason-info          =  "reason" EQUAL 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-REQ-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 POS-FLOAT MINUS POS-FLOAT
   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]


















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   Terminate-Reason      =  "Terminate-Reason" HCOLON TR-Info
   TR-Info              =  TR-Reason *(SEMI TR-Parameter)
   TR-Reason            =  "Session-Timeout"
                        /  "Server-Admin"
                        /  "Internal-Error"
                        /  token
   TR-Parameter         =  TR-time / TR-user-msg / generic-param
   TR-time              =  "time" EQUAL utc-time
   TR-user-msg          =  "user-msg" EQUAL quoted-string

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

   Transport        =  "Transport" HCOLON transport-spec
                       *(COMMA transport-spec)
   transport-spec   =  transport-id *trns-parameter
   transport-id     =  trans-id-rtp / other-trans
   trans-id-rtp     =  "RTP/" profile ["/" lower-transport]
                       ; no LWS is allowed inside transport-id
   other-trans      =  token *("/" token)






























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profile             = "AVP" / "SAVP" / "AVPF" / token
lower-transport     = "TCP" / "UDP" / token
trns-parameter      = (SEMI ( "unicast" / "multicast" ))
                    / (SEMI "interleaved" EQUAL channel [ "-" channel ])
                    / (SEMI "ttl" EQUAL ttl)
                    / (SEMI "layers" EQUAL 1*DIGIT)
                    / (SEMI "ssrc" EQUAL ssrc *(SLASH ssrc))
                    / (SEMI "mode" EQUAL mode-spec)
                    / (SEMI "dest_addr" EQUAL addr-list)
                    / (SEMI "src_addr" EQUAL addr-list)
                    / (SEMI trn-param-ext)
                    / (SEMI "setup" EQUAL contrans-setup)
                    / (SEMI "connection" EQUAL contrans-con)
                    / (SEMI "RTCP-mux")
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-REQ-REF

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

   a-mtag-def          =  "a=mtag:" message-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
   also.

   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 confidentiality
         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.

   Resource Exhaustion  As RTSP is a stateful protocol and establish
         resource usages on the server there is a clear possibility to
         attack the server by trying to overbook these resources to
         perform an denial of service attack.  This attack can be both
         against ongoing sessions and to prevent others from
         establishing sessions.  RTSP agents will need to have mechanism
         to prevent single peers from consuming extensive amounts of
         resources.

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 mechanisms
   that are acceptable in an increased generality are:

   o  Verification of the client's identity, either against a database
      of known users using RTSP authentication mechanisms (preferably
      digest authentication or stronger)

   o  A list of addresses that accept to be media destinations,
      especially considering user identity

   o  Media path based verification

   The server SHOULD NOT allow the destination field to be set unless a



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   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 the
   server may chose another method such as a server-resident
   authorization list to ensure that the request originator has the
   proper credentials to request stream delivery to the recipient.

   One solution that performs the necessary verification of acceptance
   of media suitable for unicast based delivery is the ICE based NAT
   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 2.7 "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 MUST indicate if the feature-tag applies to clients,
   servers, or proxies only or any combinations of these.  Any
   proprietary feature MUST 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

   RFCXXXX, 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 registrable
   entities.

22.5.1.  Accept-Credentials policies

   In Section 19.3.1 three policies for how to handle certificates are
   specified.  Further policies may be defined and MUST 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 cause interoperability problems.  Therefore the
   bar for registering new algorithms is intentionally placed high.

   Any registration of a new hash algorithm MUST 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 these cache directives MUST 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 innovation will result in
   new use cases or media streams with properties not covered by the
   ones specified here.  Thus new media properties can be specified.  As
   new media properties may need a substantial amount of new definitions
   to correctly specify behavior for this property the bar is intended
   to be high.

22.7.2.  Registration Rules

   Registering new media property MUST fulfill the following
   requirements

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

   o  Have an 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 the RTSP protocol as
      result of these changes.

22.7.3.  Registered Values

   This specification registers the 9 values listed in Section 16.28.

22.8.  Notify-Reason header

22.8.1.  Description

   Notify-Reason values used to indicate why a notification was sent.
   They may also imply that certain headers are required for the client
   to act properly upon the information the notification carries.  New
   notification behaviors need to be described to result in
   interoperable usage, thus a specification of each new value is
   required.

22.8.2.  Registration Rules

   Registrations for new Notify-Reason value MUST 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 effect 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 MUST fulfill the
   following requirements:

   o  Follow the Specification Required policy.

   o  An ABNF definition of the range format that fulfills 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.  Terminate-Reason Header

   The Terminate-Reason header (Section 16.50) has two registries for
   extensions.

22.10.1.  Redirect Reasons

   Registrations are done under the policy of Expert Review.  The
   registered value needs to follow syntax, i.e. be a token.  The



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   specification needs to provide definition of what the procedures that
   is to be followed when a client receives this redirect reason.  This
   specification registers two values:

   o  Session-Timeout

   o  Server-Admin

22.10.2.  Terminate-Reason Header Parameters

   Registrations are done under the policy of Specification Required.
   The registrations must define a syntax for the parameter that also
   follows the allowed by the RTSP 2.0 specification.  A contact person
   is also required.  This specification registers:

   o  time

   o  user-msg

22.11.  RTP-Info header parameters

22.11.1.  Description

   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
   requires specification required.

22.11.2.  Registration Rules

   Registrations for new Notify-Reason value MUST 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

22.11.3.  Registered Values

   This specification registers 2 parameter value pairs:






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   o  seq

   o  rtptime

22.12.  Seek-Style Policies

22.12.1.  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.

22.12.2.  Registration Rules

   Registrations of new Seek-Style polices MUST 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.

22.12.3.  Registered Values

   This specification registers 3 values:

   o  RAP

   o  First-Prior

   o  Next

22.13.  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.







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22.13.1.  Transport Protocol Specification

   A registry for the parameter transport-protocol specification MUST 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.

   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.




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   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.

22.13.2.  Transport modes

   A registry for the transport parameter mode MUST 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:

   PLAY: See RFC XXXX.

22.13.3.  Transport Parameters

   A registry for parameters that may be included in the Transport
   header MUST 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 16.52.





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22.14.  URI Schemes

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

22.14.1.  The rtsp URI Scheme

   URI scheme name:  rtsp

   Status:  Permanent

   URI scheme syntax:  See Section 20.2.1 of RFC XXXX.

   URI scheme semantics:  The rtsp scheme is used to indicate resources
         accessible through the usage of the Real-time Streaming
         Protocol (RTSP).  RTSP allows different operations on the
         resource identified by the URI, but the primary purpose is the
         streaming delivery of the resource to a client.  However the
         operations that are currently defined are: 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)

   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 need
         to be reviewed and considered in any implementation utilizing
         this scheme.

   Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com

   Author/Change controller:  IETF






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   References:  RFC 2326, RFC 3986, RFC 3987, RFC XXXX

22.14.2.  The rtsps URI Scheme

   URI scheme name:  rtsps

   Status:  Permanent

   URI scheme syntax:  See Section 20.2.1 of RFC XXXX.

   URI scheme semantics:  The rtsps scheme is used to indicate resources
         accessible through the usage of the Real-time Streaming
         Protocol (RTSP) over TLS.  RTSP allows different operations on
         the resource identified by the URI, but the primary purpose is
         the streaming delivery of the resource to a client.  However
         the operations that are currently defined are: 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)

   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 need
         to be reviewed and considered in any implementation utilizing
         this scheme.

   Contact:  Magnus Westerlund, magnus.westerlund@ericsson.com

   Author/Change controller:  IETF

   References:  RFC 2326, RFC 3986, RFC 3987, RFC XXXX








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22.14.3.  The rtspu URI Scheme

   URI scheme name:  rtspu

   Status:  Permanent

   URI scheme syntax:  See Section 3.2 of RFC 2326.

   URI scheme semantics:  The rtspu scheme is used to indicate resources
         accessible through the usage of the Real-time Streaming
         Protocol (RTSP) over unreliable datagram transport.  RTSP
         allows different operations on the resource identified by the
         URI, but the primary purpose is the streaming delivery of the
         resource to a client.  However the operations that are
         currently defined are: 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.

   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

22.15.  SDP attributes

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




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   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:     mtag
        Long form:          Message Tag
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             See ABNF definition


22.16.  Media Type Registration for text/parameters

   Type name:  text

   Subtype name:  parameters

   Required parameters:

   Optional parameters:

   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



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      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 F.

   Applications that use this media type:  Applications that use RTSP
      and have additional parameters they like to read and set using the
      RTSP GET_PARAMETER and SET_PARAMETER methods.

   Additional information:

   Magic number(s):

   File extension(s):

   Macintosh file type code(s):

   Person & email address to contact for further information:  Magnus
      Westerlund (magnus.westerlund@ericsson.com)

   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", August 2002.

   [I-D.ietf-avt-rtp-and-rtcp-mux]
              Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port",
              draft-ietf-avt-rtp-and-rtcp-mux-07 (work in progress),
              August 2007.

   [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.



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   [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)",
              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 35,
              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.



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   [RFC4646]  Phillips, A. and M. Davis, "Tags for Identifying
              Languages", BCP 47, RFC 4646, September 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
              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.

   [RFC5378]  Bradner, S. and J. Contreras, "Rights Contributors Provide
              to the IETF Trust", BCP 78, RFC 5378, November 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.

   [IETF-Trust-License-Policy]
              IETF, "IETF TRUST Legal Provisions Relating to IETF
              Documents", 2009.

   [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.




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   [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.

   [RFC0822]  Crocker, D., "Standard for the format of ARPA Internet
              text messages", STD 11, RFC 822, August 1982.

   [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.

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

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

   [RFC2663]  Srisuresh, P. and M. Holdrege, "IP Network Address
              Translator (NAT) Terminology and Considerations",
              RFC 2663, August 1999.

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, October 2000.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

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



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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 priorization 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
         Media-Properties: Random-Access=0.02, Unmutable, Unlimited

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
         Session: 12345678
         Accept-Ranges: NPT, SMPTE, 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
         Media-Properties: Random-Access=0.8, Unmutable, Unlimited

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




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   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=30-623.10
         Seek-Style: RAP
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12345;rtptime=3450012,
           url="rtsp://example.com/twister.3gp/trackID=1"
            ssrc=4F312DD8:seq=54321;rtptime=2876889

   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
         Seek-Style: Next
         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
         Seek-Style: Next
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12555;rtptime=6330012,
           url="rtsp://example.com/twister.3gp/trackID=1"
            ssrc=4F312DD8:seq=55021;rtptime=3132889

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



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   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

   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"



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         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-
         Seek-Style: RAP
         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
         Media-Properties: Random-Access=0.2, Unmutable, Unlimited

   M->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;
                    dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
                    src_addr="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
         Media-Properties: Random-Access=0.8, Unmutable, Unlimited

   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=0-623.10
         Seek-Style: RAP
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12345;rtptime=3450012,



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           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
         Media-Properties: Random-Access=0.02, Unmutable, Unlimited










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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
         Media-Properties: Random-Access=1.2, Unmutable, Unlimited

   C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 23456789
         Range: smpte=0:10:00-

   V->C: RTSP/2.0 200 OK
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-1:49:23
         Seek-Style: First-Prior
         RTP-Info: url="rtsp://video.example.com/twister/video"
                   ssrc=A17E189D:seq=12312232;rtptime=78712811
         Server: PhonyServer/2.0
         Date: 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
         Seek-Style: First-Prior
         RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
                   ssrc=3D124F01:seq=876655;rtptime=1032181
         Server: PhonyServer/1.0
         Date: 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
   that may start at slightly different times and may drift with respect
   to each other over time, the client can perform initial
   synchronization of the two media using RTP-Info and Range received in
   the PLAY responses.  If the two servers are time synchronized the
   RTCP packets can also be used to maintain synchronization.






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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
         Media-Properties: Begining-Only, Unmutable, Unlimited


   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
         Seek-Style: RAP
         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:





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   C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 2034820394

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
         Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0


   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0
         CSeq: 3
         Session: 0456804596
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:07 GMT
         Session: 0456804596
         Seek-Style: Next
         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
   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
         Media-Properties: Random-Access=0.8, Unmutable, Unlimited

















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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 MUST 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
   are 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 need to be given a response according to the response
   column.  If the request does not meet the requisite, is erroneous or
   some other type of error 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 MUST NOT result in any change
   of the session state, except if the error is not possible to recover
   from.  A unrecoverable error MUST 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



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   tearing down 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" MUST
   NOT be used.  This addressing and multiplexing is used as defined
   with use of channel numbers and the interleaved parameter.

C.1.2.  AVP/UDP

   This part describes sending of RTP [RFC3550] over lower transport
   layer UDP [RFC0768] according to the profile "RTP Profile for Audio
   and Video Conferences with Minimal Control" defined in RFC 3551
   [RFC3551].  This profile requires one or two uni- or bi-directional
   UDP flows per media stream.  The first UDP flow is for RTP and the
   second is for RTCP.  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 MUST 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 MUST 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
      MUST NOT be sent.

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

   o  The RTP/UDP packets from the client to the server MUST be sent to
      the address and port given by the first address 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 MUST 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 MUST
   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 MUST 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 MUST 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 MUST be supported if an RTSP agent
   handles RTP.

C.1.6.1.  Media synchronization

   RTCP provides media synchronization and clock drift compensation.
   The initial media synchronization is available from RTP-Info header.
   However to be able to handle any clock drift between the media
   streams, RTCP is needed.

C.1.6.2.  RTSP Session keep-alive

   RTCP traffic from the RTSP client to the RTSP server MUST function as
   keep-alive.  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 adaption

   RTCP Receiver reports and any additional feedback from the client
   MUST be used adapt the bit-rate used over the transport for all cases
   when RTP is sent over UDP.  An RTP sender without reserved resources
   MUST NOT use more than its fair share of the available resources.
   This can be determined by comparing on short to medium term (some
   seconds) the used bit-rate and adapt it so that the RTP sender sends
   at a bit-rate comparable to what a TCP sender would achieve on
   average over the same path.

C.1.6.4.  RTP and RTCP Multiplexing

   RTSP can be used to negotiate the usage of RTP and RTCP multiplexing
   as described in [I-D.ietf-avt-rtp-and-rtcp-mux].  This allows servers
   and client to reduce the amount of resources required for the session
   by only requiring one underlying transport stream per media stream
   instead of two when using RTP and RTCP.  This lessens the server port
   consumption and also the necessary state and keep-alive work when
   operating across Network and Address Translators [RFC2663].

   Content must be prepared with some consideration for RTP and RTCP
   multiplexing, mainly ensuring that the RTP payload types used does
   not collide with the ones used for RTCP packet types this option
   likely needs explicit support from the content unless the RTP payload
   types can be remapped by the server and that is correctly reflected
   in the session description.  Beyond that support of this feature
   should come at little cost and much gain.

   It is recommended that if the content and server supports RTP and
   RTCP multiplexing that this is indicated in the session description,
   for example using the SDP attribute "a=rtcp-mux".  If the SDP message
   contains the a=rtcp-mux attribute for a media stream, the server MUST
   support RTP and RTCP multiplexing.  If indicated or otherwise desired
   by the client it can include the Transport parameter "RTCP-mux" in
   any transport specification where it desires to use RTCP-mux.  The
   server will indicate if it supports RTCP-mux.  Server and Client
   SHOULD support RTP and RTCP multiplexing.

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 MUST be "rtp" and the
   lower layer MUST be TCP.  The profile may be any of the above
   specified ones; AVP, AVPF, SAVP or SAVPF.





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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
   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 16.52) 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.




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   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 16.52.  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-
   interleaved RTP/AVP/TCP media stream, the active party SHOULD
   initiate the TCP connection as soon as possible.  The client MUST NOT
   send a PLAY request prior to the establishment of all the TCP
   connections negotiated using SETUP for the session.  In case the
   server receives a PLAY request in a session that has not yet
   established all the TCP connections, it MUST respond using the 464
   "Data Transport Not Ready Yet" (Section 15.4.29) error code.

   Once the PLAY request for a media resource transported over non-
   interleaved RTP/AVP/TCP occurs, media begins to flow from server to
   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 MUST 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/



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   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 16.52) 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
            Accept-Ranges: NPT, SMPTE, UTC



















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      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
            Media-Properties: Random-Access=0.8, Unmutable, Unlimited

      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=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=30-623.10
            Seek-Style: First-Prior
            RTP-Info:  url="rtsp://example.com/twister.3gp/trackID=1"
               ssrc=4F312DD8:seq=54321;rtptime=2876889

C.3.  Handling Media Clock Time Jumps in the RTP Media Layer

   RTSP allows media clients to control selected, non-contiguous
   sections of media presentations, rendering those streams with an RTP
   media layer [RFC3550].  Two cases occur, the first is when a new PLAY
   request replaces an old ongoing request and the new request results
   in a jump in the media.  This should produce in the RTP layer a
   continuous media stream.  A client may also directly following a
   completed PLAY request perform a new PLAY request.  This will result
   in some gap in the media layer.  The below text will look into both
   cases.

   A PLAY request that replaces a ongoing request allows the media layer
   rendering the RTP stream without being affected by jumps in media
   clock time.  The RTP timestamps for the new media range is set so
   that they become continuous with the previous media range in the
   previous request.  The RTP sequence number for the first packet in



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   the new range will be the next following the last packet in the
   previous range, i.e. monotonically increasing.  The goal is to allow
   the media rendering layer to work without interruption or
   reconfiguration across the jumps in media clock.  This should be
   possible in all cases of replaced PLAY requests for media that has
   random-access properties.  In this case care is needed to align
   frames or similar media dependent structures.

   In cases where jumps in media clock time are a result of RTSP
   signalling operations arriving after a completed PLAY operation, the
   request timing will result in that media becomes non-continuous.  The
   server becomes unable to send the media so that it 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.  Note also that in this case the RTP sequence number should
   be the next packet number.  If not, the RTCP packet loss reporting
   will indicate as loss all packets not received between the point of
   pausing and later resuming.  This may trigger congestion avoidance
   mechanisms.  An allowed exception from the above recommendation on
   monotonically increasing RTP sequence number is live media streams,
   likely being relayed.  In this case, when the client resumes
   delivery, it will get the media that is currently being delivered to
   the server itself.  For this type of basic delivery of live streams
   to multiple users over unicast, individual rewriting of RTP sequence
   numbers becomes quite a burden.  For solutions that anyway caches
   media, timeshifts, etc, the rewriting should be a minor issue.

   The goal when handling jumps in media clock time is that the provided
   stream is continuous without gaps in RTP timestamp or sequence
   number.  However, when delivery has been halted for some reason the
   RTP timestamp when resuming MUST represent the duration the delivery
   was halted.  RTP sequence number MUST generally be the next number,
   i.e. monotonically increasing modulo 65536.  For media resources with
   the properties Time-Progressing and Time-Duration=0.0 the server MAY
   create RTP media streams with RTP sequence number jumps in them due
   to client first halting delivery and later resuming it (PAUSE and



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   then later PLAY).  However, servers utilizing this exception must
   take into consideration the resulting RTCP receiver reports that
   likely contains loss report for all the packets part of the
   discontinuity.  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 on how the server acts in each case.

      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.

   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: abcdefgh
        Range: npt=10-15
        User-Agent: PhonyClient/1.2

      S->C: RTSP/2.0 200 OK
        CSeq: 4
        Session: abcdefgh
        Range: npt=10-15
        RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                  ssrc=0D12F123:seq=0;rtptime=0

   The ensuing RTP data stream is depicted below:


      S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s
      S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s
       . . .
      S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s


   Upon the completion of the requested playback the server sends a
   PLAY_NOFIFY







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        S->C: PLAY_NOTIFY rtsp://example.com/fizzle RTSP/2.0
              CSeq: 45
              Notify-Reason: end-of-stream
              Request-Status: cseq=4 status=200 reason="OK"
              Range: npt=-15
              RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                 ssrc=0D12F123:seq=49;rtptime=39200
              Session: abcdefgh

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

   Upon the completion of the play range, the client follows up with a
   request to PLAY from a new NPT.

   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
         CSeq: 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

   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).




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C.4.  Handling RTP Timestamps after PAUSE

   During a PAUSE / PLAY interaction in an RTSP session, the duration of
   time for which the RTP transmission was halted MUST be reflected in
   the RTP timestamp of each RTP stream.  The duration can be calculated
   for each RTP stream as the time elapsed from when the last RTP packet
   was sent before the PAUSE request was received and when the first RTP
   packet was sent after the subsequent PLAY request was received.  The
   duration includes all latency incurred and processing time required
   to complete the request.

      The RTP RFC [RFC3550] states that: The RTP timestamp for each unit
      [packet] would be related to the wallclock time at which the unit
      becomes current on the virtual presentation timeline.

      In order to satisfy the requirements of [RFC3550], the RTP
      timestamp space needs to increase continuously with real time.
      While this is not optimal for stored media, it is required for RTP
      and RTCP to function as intended.  Using a continuous RTP
      timestamp space allows the same timestamp model for both stored
      and live media and allows better opportunity to integrate both
      types of media under a single control.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero.

   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15
         RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=0;rtptime=0

   The ensuing RTP data stream is depicted below:

      S -> C: RTP packet - seq = 0, rtptime = 0,    NPT time = 10s
      S -> C: RTP packet - seq = 1, rtptime = 800,  NPT time = 10.1s
      S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
      S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s

   The client then sends a PAUSE request:



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   C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0
         CSeq: 5
         Session: abcdefg
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 5
         Session: abcdefg
         Range: npt=10.4-15

   20 seconds elapse and then the client sends a PLAY request.  In
   addition the server requires 15 ms to process the request:

   C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
         CSeq: 6
         Session: abcdefg
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 6
         Session: abcdefg
         Range: npt=10.4-15
         RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=4;rtptime=164400

   The ensuing RTP data stream is depicted below:

      S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
      S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
      S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s

   First, NPT 10 through 10.3 is played, then a PAUSE is received by the
   server.  After 20 seconds a PLAY is received by the server which 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.

   Note Further: To ensure correct media decoding and usually jitter-
   buffer handling reseting some of the state when issuing a PLAY
   request is needed.



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C.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.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.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.8.  Continuous Audio

   For continuous audio, the server SHOULD set the RTP marker bit at the
   beginning of serving a new PLAY request or at jumps in timeline.
   This allows the client to perform playout delay adaptation.

C.9.  Multiple Sources in an RTP Session

   Note that more than one SSRC MAY be sent in the media stream.  If it
   happens all sources are expected to be rendered simultaneously.

C.10.  Usage of SSRCs and the RTCP BYE Message During an RTSP Session

   The RTCP BYE message indicates the end of use of a given SSRC.  If
   all sources leave an RTP session, it can, in most cases, be assumed
   to have ended.  Therefore, a client or server MUST NOT send 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.11.  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 or profile for use with RTSP:

   o  The protocol or profile needs to define a name tag representing
      it.  This tag is required to be 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 MUST be different from any
   media level URI.  The presence of a session level control attribute
   MUST be interpreted as support for aggregated control.  The control
   attribute MUST be present on media level unless the presentation only
   contains a single media stream, in which case the attribute MAY 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 MUST 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 MUST be
   treated as if it were an empty embedded URI, and thus inherit the
   entire base URI.

      Note, RFC 2326 was very unclear on the processing of relative URI
      and several RTSP 1.0 implementations at the point of publishing
      this document did not perform RFC 3986 processing to determine the
      resulting URI, instead simple concatenation is common.  To avoid
      this issue completely it is recommended to use absolute URI in the
      SDP.

   The URI handling for SDPs from container files need special
   consideration.  For example 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 numbers are the ones RECOMMENDED
   by the server to the client, about which receiver ports to use; the
   client MUST still include its receiver ports in its SETUP request.



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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 which transport protocol,
   profile, and possibly lower-layer is to be used for the media stream.
   The combination of transport, profile and lower layer, like RTP/AVP/
   UDP needs to be defined for how to be used with RTSP.  The currently
   defined combinations are defined in Appendix C, further combinations
   MAY be specified.

   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 MUST
   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) MUST 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 MUST be used.

   Note: Usually one will specify the same length for all media, even if
   there isn't media available for the full duration on all media.
   However that requires that the server accepts PLAY requests within
   that range.

   Servers MUST take care to provide RTSP Range (see Section 16.38)
   values that are consistent with what is presented in the SDP for the
   content.  There is no reason for non dynamic content, like media
   clips provided on demand to have inconsistent values.  Inconsistent
   values between the SDP and the actual values for the content handled
   by the server is likely to generate some failure, like 457 "Invalid
   Range", in case the client uses PLAY requests with a Range header.
   In case the content is dynamic in length and it is infeasible to
   provide a correct value in the SDP the server is recommended to
   describe this as non-seekable content (see below).  The server MAY
   override that property in the response to a PLAY request using the
   correct values in the Range header.

   The unit is specified first, followed by the value range.  The units
   and their values are as defined in Section 4.4, 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 MUST be considered as non-seekable content unless this property
   is overridden.  Multiple instances carrying different clock formats
   MAY be included at either session or media level.

   ABNF for the attribute is defined in Section 20.3.

   Examples:
     a=range:npt=0-34.4368
     a=range:clock=19971113T211503Z-19971113T220300Z
     Non seekable stream of unknown duration:
     a=range:npt=0-

D.1.7.  Time of Availability

   The "t=" field 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 MUST be "0.0.0.0",
   and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" (can also be
   written as "::"), i.e. the unspecified address according to RFC 4291
   [RFC4291].

D.1.9.  Message Body Tag

   The optional "a=mtag" attribute identifies a version of the session
   description.  It is opaque to the client.  SETUP requests may include
   this identifier in the If-Match field (see Section 16.23) to only
   allow session establishment if this attribute value still corresponds
   to that of the current description.  The attribute value is opaque



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   and may contain any character allowed within SDP attribute values.

   ABNF for the attribute is defined in Section 20.3.

   Example:
     a=mtag:"158bb3e7c7fd62ce67f12b533f06b83a"

      One could argue that the "o=" field provides identical
      functionality.  However, it does so in a manner that would put
      constraints on servers that need to support multiple session
      description types other than SDP for the same piece of media
      content.

D.2.  Aggregate Control Not Available

   If a presentation does not support aggregate control no session level
   "a=control:" attribute is specified.  For a SDP with multiple media
   sections specified, each section will have its own control URI
   specified via the "a=control:" attribute.

   Example:
   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



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   Request-URI for aggregate control.  If the media-level URI is
   relative, it is resolved to absolute URIs according to Appendix D.1.1
   above.

   Example:
   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



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   base URI.  And as SDP might be temporarily stored on file system
   before being loaded into an RTSP capable client, thus if possible to
   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.  RTSP Use Cases

   This Appendix describes the most important and considered use cases
   for RTSP.  They are listed in descending order of importance in
   regards to ensuring that all necessary functionality is present.
   This specification only fully supports usage of the two first.  Also
   in these first two cases, there are special cases or exceptions that
   are not supported without extensions, e.g. the redirection of media
   to another address than the controlling entity.

E.1.  On-demand Playback of Stored Content

   An RTSP capable server stores content suitable for being streamed to
   a client.  A client desiring playback of any of the stored content
   uses RTSP to set up the media transport required to deliver the
   desired content.  RTSP is then used to initiate, halt and manipulate
   the actual transmission (playout) of the content.  RTSP is also
   required to provide necessary description and synchronization
   information for the content.

   The above high level description can be broken down into a number of
   functions that RTSP needs to be capable of.

   Presentation Description:  Provide initialization information about
         the presentation (content); for example, which media codecs are
         needed for the content.  Other information that is important
         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).

E.2.  Unicast Distribution of Live Content

   This use case is similar to the above on-demand content case (see
   Appendix E.1) the difference is the nature of the content itself.
   Live content is continuously distributed as it becomes available from
   a source; i.e., the main difference from on-demand is that one starts
   distributing content before the end of it has become available to the
   server.

   In many cases the consumer of live content is only interested in
   consuming what 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.

E.3.  On-demand Playback using Multicast

   It is possible to use RTSP to request that media be delivered to a
   multicast group.  The entity setting up the session (the controller)
   will then control when and what media is delivered to the group.
   This use case has some potential for denial of service attacks by
   flooding a multicast group.  Therefore, a mechanism is needed to
   indicate that the group actually accepts the traffic from the RTSP
   server.

   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.

E.4.  Inviting an RTSP server into a conference

   If one has an established conference or group session, it is possible
   to have an RTSP server distribute media to the whole group.
   Transmission to the group is simplest when controlled by a single
   participant or leader of the conference.  Shared control might be
   possible, but would require further investigation and possibly
   extensions.

   This use case assumes that there exists either multicast or a
   conference focus that redistribute media to all participants.

   This use case is intended to be able to handle the following
   scenario: A conference leader or participant (hereafter called the
   controller) has some pre-stored content on an RTSP server that he
   wants to share with the group.  The controller sets up an RTSP
   session at the streaming server for this content and retrieves the
   session description for the content.  The destination for the media
   content is set to the shared multicast group or conference focus.



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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.

E.5.  Live Content using Multicast

   This use case in its simplest form does not require any use of RTSP
   at all; this is what multicast conferences being announced with SAP
   [RFC2974] and SDP are intended to handle.  However in use cases where
   more advanced features like access control to the multicast session
   are desired, RTSP could be used for session establishment.

   A client desiring to join a live multicasted media session with
   cryptographic (encryption) access control could use RTSP in the
   following way.  The source of the session announces the session and
   gives all interested an RTSP URI.  The client connects to the server
   and requests the presentation description, allowing configuration for
   reception of the media.  In this step it is possible for the client
   to use secured transport and any desired level of authentication; for
   example, for billing or access control.  An RTSP link also allows for
   load balancing between multiple servers.

   If these were the only goals, they could be achieved by simply using
   HTTP.  However, for cases where the sender likes to keep track of



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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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Appendix F.  Text format for Parameters

   A resource of type "text/parameters" consists of either 1) a list of
   parameters (for a query) or 2) a list of parameters and associated
   values (for an response or setting of the parameter).  Each entry of
   the list is a single line of text.  Parameters are separated from
   values by a colon.  The parameter name MUST only use US-ASCII visible
   characters while the values are UTF-8 text strings.  The media type
   registration template is in Section 22.16.

   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 value part, thus usage of UTF-8 strings may not
   be supported.  However, as individual parameters are not defined, the
   using application anyway needs to have out-of-band agreement or using
   feature-tag to determine if the end-point supports the parameters.

   The ABNF [RFC5234] grammar for "text/parameters" content is:

   file             = *((parameter / parameter-value) CRLF)
   parameter        = 1*visible-except-colon
   parameter-value  = parameter *WSP ":" value
   visible-except-colon = %x21-39 / %x3B-7E    ; VCHAR - ":"
   value            = *(TEXT-UTF8char / WSP)
   TEXT-UTF8char    =  %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 G.  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 16.51) 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 16.51












































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Appendix H.  Backwards Compatibility Considerations

   This section contains notes on issues about backwards compatibility
   with clients or servers being implemented according to RFC 2326
   [RFC2326].  Note that there 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 MUST 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.

H.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.

H.2.  Using Persistent Connections

   Some server implementations of RFC 2326 maintain a one-to-one
   relationship between a connection and an RTSP session.  Such
   implementations require clients to use a persistent connection to
   communicate with the server and when a client closes its connection,
   the server may remove the RTSP session.  This is worth noting if a
   RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.













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Appendix I.  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 J.  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.

      *  The 300 (Multiple Choices) response code has been removed.

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

      *  Extended Pipelining rules allowing for quick session startup.

   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.

      *  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 message body and the Server should always include
         it.

      *  Removed possibility of using Range header with Scale header to
         indicate when it is to be activated, since it can't work as
         defined.  Also added rule that lack of Scale header in response
         indicates lack of support for the header.  Feature-tags for
         scaled playback has been defined.

      *  The Speed header must now be responded to indicate support and
         the actual speed going to be used.  A feature-tag is defined.
         Notes on congestion control was also added.

      *  The Supported header was borrowed from SIP [RFC3261] to help
         with the feature negotiation in RTSP.

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



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      *  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 MTag.  The MTag 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.

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

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



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      *  The ABNF 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 ABNF.

      *  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.

      *  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



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         it exist is defined here and some requirements for 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.

      *  REDIRECT has been expanded and diversified for different
         situations.

   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.

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




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   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 K.  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 version 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, Martti Mela, and Patrick Hoffman.

K.1.  Contributors

   The following people have made written contributions that were
   included in the specification:

   o  Tom Marshall contributed text on the usage of 3rr status codes.

   o  Thomas Zheng contributed text on the usage of the Range in PLAY
      responses and proposed an earlier version of the PLAY_NOTIFY
      method.

   o  Sean Sheedy contributed text on the timeout behavior of RTSP
      messages and connections, the 463 status code, and proposed an
      earlier version of the PLAY_NOTIFY method.

   o  Greg Sherwood proposed an earlier version of the PLAY_NOTIFY
      method.





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   o  Fredrik Lindholm contributed text about the RTSP security
      framework.

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

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











































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Appendix L.  RFC Editor Consideration

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















































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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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