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
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Copyright Notice
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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
29. The difference between 30 and 29.97 frames per second is handled
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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
the provided end-point, or the end of the media is reached prior to
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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.
Schulzrinne, et al. Expires December 21, 2009 [Page 214]
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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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