[Docs] [txt|pdf] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits] [IPR]
Versions: (RFC 2326) 00 01 02 03 04 05 06 07
08 09 10 11 12 13 14 15 16 17 18 19
20 21 22 23 24 25 26 27 28 29 30 34
35 36 37 38 39 40 RFC 7826
MMUSIC Working Group H. Schulzrinne
Internet-Draft Columbia University
Intended status: Standards Track A. Rao
Expires: August 28, 2008 Cisco
R. Lanphier
Real Networks
M. Westerlund
Ericsson AB
M. Stiemerling (Ed.)
NEC
February 25, 2008
Real Time Streaming Protocol 2.0 (RTSP)
draft-ietf-mmusic-rfc2326bis-17.txt
Status of this Memo
By submitting this Internet-Draft, each author represents that any
applicable patent or other IPR claims of which he or she is aware
have been or will be disclosed, and any of which he or she becomes
aware will be disclosed, in accordance with Section 6 of BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt.
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
This Internet-Draft will expire on August 28, 2008.
Copyright Notice
Copyright (C) The IETF Trust (2008).
Schulzrinne, et al. Expires August 28, 2008 [Page 1]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Abstract
This memorandum defines RTSP version 2.0 which is a revision of the
Proposed Standard RTSP version 1.0 which is defined in RFC 2326.
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and provide a means for choosing delivery
mechanisms based upon RTP (RFC 3550).
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 8
1.1. Scope and Background . . . . . . . . . . . . . . . . . . 8
1.2. RTSP Specificication Update . . . . . . . . . . . . . . 9
1.3. Notational Conventions . . . . . . . . . . . . . . . . . 9
1.4. Terminology . . . . . . . . . . . . . . . . . . . . . . 10
2. RTSP Introduction . . . . . . . . . . . . . . . . . . . . . . 14
2.1. Protocol Properties . . . . . . . . . . . . . . . . . . 14
2.2. RTSP's Relationship to HTTP . . . . . . . . . . . . . . 15
2.3. Extending RTSP . . . . . . . . . . . . . . . . . . . . . 16
2.4. Overall Operation . . . . . . . . . . . . . . . . . . . 17
2.5. RTSP States . . . . . . . . . . . . . . . . . . . . . . 18
2.6. Relationship with Other Protocols . . . . . . . . . . . 19
3. RTSP Use Cases . . . . . . . . . . . . . . . . . . . . . . . 20
3.1. On-demand Playback of Stored Content . . . . . . . . . . 20
3.2. Unicast distribution of Live Content . . . . . . . . . . 21
3.3. On-demand Playback using Multicast . . . . . . . . . . . 22
3.4. Inviting an RTSP server into a conference . . . . . . . 22
3.5. Live Content using Multicast . . . . . . . . . . . . . . 23
4. Protocol Parameters . . . . . . . . . . . . . . . . . . . . . 25
4.1. RTSP Version . . . . . . . . . . . . . . . . . . . . . . 25
4.2. RTSP IRI and URI . . . . . . . . . . . . . . . . . . . . 25
4.3. Session Identifiers . . . . . . . . . . . . . . . . . . 27
4.4. SMPTE Relative Timestamps . . . . . . . . . . . . . . . 27
4.5. Normal Play Time . . . . . . . . . . . . . . . . . . . . 27
4.6. Absolute Time . . . . . . . . . . . . . . . . . . . . . 28
4.7. Feature-tags . . . . . . . . . . . . . . . . . . . . . . 28
4.8. Entity Tags . . . . . . . . . . . . . . . . . . . . . . 29
5. RTSP Message . . . . . . . . . . . . . . . . . . . . . . . . 30
5.1. Message Types . . . . . . . . . . . . . . . . . . . . . 30
5.2. Message Headers . . . . . . . . . . . . . . . . . . . . 30
Schulzrinne, et al. Expires August 28, 2008 [Page 2]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
5.3. Message Body . . . . . . . . . . . . . . . . . . . . . . 30
5.4. Message Length . . . . . . . . . . . . . . . . . . . . . 30
6. General Header Fields . . . . . . . . . . . . . . . . . . . . 32
7. Request . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
7.1. Request Line . . . . . . . . . . . . . . . . . . . . . . 33
7.2. Request Header Fields . . . . . . . . . . . . . . . . . 35
8. Response . . . . . . . . . . . . . . . . . . . . . . . . . . 37
8.1. Status-Line . . . . . . . . . . . . . . . . . . . . . . 37
8.1.1. Status Code and Reason Phrase . . . . . . . . . . . 37
8.2. Response Header Fields . . . . . . . . . . . . . . . . . 40
9. Entity . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
9.1. Entity Header Fields . . . . . . . . . . . . . . . . . . 43
9.2. Entity Body . . . . . . . . . . . . . . . . . . . . . . 44
10. Connections . . . . . . . . . . . . . . . . . . . . . . . . . 45
10.1. Reliability and Acknowledgements . . . . . . . . . . . . 45
10.2. Using Connections . . . . . . . . . . . . . . . . . . . 46
10.3. Closing Connections . . . . . . . . . . . . . . . . . . 47
10.4. Timing Out Connections and RTSP Messages . . . . . . . . 48
10.5. Showing Liveness . . . . . . . . . . . . . . . . . . . . 48
10.6. Use of IPv6 . . . . . . . . . . . . . . . . . . . . . . 49
11. Capability Handling . . . . . . . . . . . . . . . . . . . . . 50
12. Pipelining Support . . . . . . . . . . . . . . . . . . . . . 52
13. Method Definitions . . . . . . . . . . . . . . . . . . . . . 53
13.1. OPTIONS . . . . . . . . . . . . . . . . . . . . . . . . 54
13.2. DESCRIBE . . . . . . . . . . . . . . . . . . . . . . . . 55
13.3. SETUP . . . . . . . . . . . . . . . . . . . . . . . . . 57
13.3.1. Changing Transport Parameters . . . . . . . . . . . 59
13.4. PLAY . . . . . . . . . . . . . . . . . . . . . . . . . . 60
13.5. PAUSE . . . . . . . . . . . . . . . . . . . . . . . . . 65
13.6. TEARDOWN . . . . . . . . . . . . . . . . . . . . . . . . 68
13.7. GET_PARAMETER . . . . . . . . . . . . . . . . . . . . . 69
13.8. SET_PARAMETER . . . . . . . . . . . . . . . . . . . . . 70
13.9. REDIRECT . . . . . . . . . . . . . . . . . . . . . . . . 71
14. Embedded (Interleaved) Binary Data . . . . . . . . . . . . . 74
15. Status Code Definitions . . . . . . . . . . . . . . . . . . . 76
15.1. Success 1xx . . . . . . . . . . . . . . . . . . . . . . 76
15.1.1. 100 Continue . . . . . . . . . . . . . . . . . . . . 76
15.2. Success 2xx . . . . . . . . . . . . . . . . . . . . . . 76
15.2.1. 200 OK . . . . . . . . . . . . . . . . . . . . . . . 76
15.3. Redirection 3xx . . . . . . . . . . . . . . . . . . . . 76
15.3.1. 300 Multiple Choices . . . . . . . . . . . . . . . . 77
15.3.2. 301 Moved Permanently . . . . . . . . . . . . . . . 77
15.3.3. 302 Found . . . . . . . . . . . . . . . . . . . . . 77
15.3.4. 303 See Other . . . . . . . . . . . . . . . . . . . 77
15.3.5. 304 Not Modified . . . . . . . . . . . . . . . . . . 77
15.3.6. 305 Use Proxy . . . . . . . . . . . . . . . . . . . 78
15.4. Client Error 4xx . . . . . . . . . . . . . . . . . . . . 78
15.4.1. 400 Bad Request . . . . . . . . . . . . . . . . . . 78
Schulzrinne, et al. Expires August 28, 2008 [Page 3]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
15.4.2. 405 Method Not Allowed . . . . . . . . . . . . . . . 78
15.4.3. 451 Parameter Not Understood . . . . . . . . . . . . 78
15.4.4. 452 reserved . . . . . . . . . . . . . . . . . . . . 78
15.4.5. 453 Not Enough Bandwidth . . . . . . . . . . . . . . 79
15.4.6. 454 Session Not Found . . . . . . . . . . . . . . . 79
15.4.7. 455 Method Not Valid in This State . . . . . . . . . 79
15.4.8. 456 Header Field Not Valid for Resource . . . . . . 79
15.4.9. 457 Invalid Range . . . . . . . . . . . . . . . . . 79
15.4.10. 458 Parameter Is Read-Only . . . . . . . . . . . . . 79
15.4.11. 459 Aggregate Operation Not Allowed . . . . . . . . 79
15.4.12. 460 Only Aggregate Operation Allowed . . . . . . . . 79
15.4.13. 461 Unsupported Transport . . . . . . . . . . . . . 80
15.4.14. 462 Destination Unreachable . . . . . . . . . . . . 80
15.4.15. 463 Destination Prohibited . . . . . . . . . . . . . 80
15.4.16. 464 Data Transport Not Ready Yet . . . . . . . . . . 80
15.4.17. 470 Connection Authorization Required . . . . . . . 80
15.4.18. 471 Connection Credentials not accepted . . . . . . 80
15.4.19. 472 Failure to establish secure connection . . . . . 81
15.5. Server Error 5xx . . . . . . . . . . . . . . . . . . . . 81
15.5.1. 551 Option not supported . . . . . . . . . . . . . . 81
16. Header Field Definitions . . . . . . . . . . . . . . . . . . 82
16.1. Accept . . . . . . . . . . . . . . . . . . . . . . . . . 91
16.2. Accept-Credentials . . . . . . . . . . . . . . . . . . . 91
16.3. Accept-Encoding . . . . . . . . . . . . . . . . . . . . 92
16.4. Accept-Language . . . . . . . . . . . . . . . . . . . . 92
16.5. Accept-Ranges . . . . . . . . . . . . . . . . . . . . . 92
16.6. Allow . . . . . . . . . . . . . . . . . . . . . . . . . 92
16.7. Authorization . . . . . . . . . . . . . . . . . . . . . 93
16.8. Bandwidth . . . . . . . . . . . . . . . . . . . . . . . 93
16.9. Blocksize . . . . . . . . . . . . . . . . . . . . . . . 93
16.10. Cache-Control . . . . . . . . . . . . . . . . . . . . . 93
16.11. Connection . . . . . . . . . . . . . . . . . . . . . . . 96
16.12. Connection-Credentials . . . . . . . . . . . . . . . . . 96
16.13. Content-Base . . . . . . . . . . . . . . . . . . . . . . 97
16.14. Content-Encoding . . . . . . . . . . . . . . . . . . . . 97
16.15. Content-Language . . . . . . . . . . . . . . . . . . . . 97
16.16. Content-Length . . . . . . . . . . . . . . . . . . . . . 97
16.17. Content-Location . . . . . . . . . . . . . . . . . . . . 98
16.18. Content-Type . . . . . . . . . . . . . . . . . . . . . . 98
16.19. CSeq . . . . . . . . . . . . . . . . . . . . . . . . . . 98
16.20. Date . . . . . . . . . . . . . . . . . . . . . . . . . . 98
16.21. ETag . . . . . . . . . . . . . . . . . . . . . . . . . . 98
16.22. Expires . . . . . . . . . . . . . . . . . . . . . . . . 99
16.23. From . . . . . . . . . . . . . . . . . . . . . . . . . . 100
16.24. If-Match . . . . . . . . . . . . . . . . . . . . . . . . 100
16.25. If-Modified-Since . . . . . . . . . . . . . . . . . . . 100
16.26. If-None-Match . . . . . . . . . . . . . . . . . . . . . 101
16.27. Last-Modified . . . . . . . . . . . . . . . . . . . . . 101
Schulzrinne, et al. Expires August 28, 2008 [Page 4]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
16.28. Location . . . . . . . . . . . . . . . . . . . . . . . . 101
16.29. Pipelined-Requests . . . . . . . . . . . . . . . . . . . 101
16.30. Proxy-Authenticate . . . . . . . . . . . . . . . . . . . 102
16.31. Proxy-Authorization . . . . . . . . . . . . . . . . . . 102
16.32. Proxy-Require . . . . . . . . . . . . . . . . . . . . . 102
16.33. Proxy-Supported . . . . . . . . . . . . . . . . . . . . 103
16.34. Public . . . . . . . . . . . . . . . . . . . . . . . . . 103
16.35. Range . . . . . . . . . . . . . . . . . . . . . . . . . 104
16.36. Referer . . . . . . . . . . . . . . . . . . . . . . . . 106
16.37. Retry-After . . . . . . . . . . . . . . . . . . . . . . 106
16.38. Require . . . . . . . . . . . . . . . . . . . . . . . . 106
16.39. RTP-Info . . . . . . . . . . . . . . . . . . . . . . . . 107
16.40. Scale . . . . . . . . . . . . . . . . . . . . . . . . . 109
16.41. Speed . . . . . . . . . . . . . . . . . . . . . . . . . 109
16.42. Server . . . . . . . . . . . . . . . . . . . . . . . . . 110
16.43. Session . . . . . . . . . . . . . . . . . . . . . . . . 110
16.44. Supported . . . . . . . . . . . . . . . . . . . . . . . 111
16.45. Timestamp . . . . . . . . . . . . . . . . . . . . . . . 111
16.46. Transport . . . . . . . . . . . . . . . . . . . . . . . 112
16.47. Unsupported . . . . . . . . . . . . . . . . . . . . . . 117
16.48. User-Agent . . . . . . . . . . . . . . . . . . . . . . . 118
16.49. Vary . . . . . . . . . . . . . . . . . . . . . . . . . . 118
16.50. Via . . . . . . . . . . . . . . . . . . . . . . . . . . 118
16.51. WWW-Authenticate . . . . . . . . . . . . . . . . . . . . 118
17. Proxies . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
18. Caching . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
19. Security Framework . . . . . . . . . . . . . . . . . . . . . 122
19.1. RTSP and HTTP Authentication . . . . . . . . . . . . . . 122
19.2. RTSP over TLS . . . . . . . . . . . . . . . . . . . . . 122
19.3. Security and Proxies . . . . . . . . . . . . . . . . . . 123
19.3.1. Accept-Credentials . . . . . . . . . . . . . . . . . 124
19.3.2. User approved TLS procedure . . . . . . . . . . . . 125
20. Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
20.1. Base Syntax . . . . . . . . . . . . . . . . . . . . . . 127
20.2. RTSP Protocol Definition . . . . . . . . . . . . . . . . 129
20.2.1. Generic Protocol elements . . . . . . . . . . . . . 129
20.2.2. Message Syntax . . . . . . . . . . . . . . . . . . . 132
20.2.3. Header Syntax . . . . . . . . . . . . . . . . . . . 136
20.3. SDP extension Syntax . . . . . . . . . . . . . . . . . . 143
21. Security Considerations . . . . . . . . . . . . . . . . . . . 144
21.1. Remote denial of Service Attack . . . . . . . . . . . . 146
22. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 148
22.1. Feature-tags . . . . . . . . . . . . . . . . . . . . . . 148
22.1.1. Description . . . . . . . . . . . . . . . . . . . . 148
22.1.2. Registering New Feature-tags with IANA . . . . . . . 149
22.1.3. Registered entries . . . . . . . . . . . . . . . . . 149
22.2. RTSP Methods . . . . . . . . . . . . . . . . . . . . . . 149
22.2.1. Description . . . . . . . . . . . . . . . . . . . . 149
Schulzrinne, et al. Expires August 28, 2008 [Page 5]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
22.2.2. Registering New Methods with IANA . . . . . . . . . 149
22.2.3. Registered Entries . . . . . . . . . . . . . . . . . 150
22.3. RTSP Status Codes . . . . . . . . . . . . . . . . . . . 150
22.3.1. Description . . . . . . . . . . . . . . . . . . . . 150
22.3.2. Registering New Status Codes with IANA . . . . . . . 150
22.3.3. Registered Entries . . . . . . . . . . . . . . . . . 150
22.4. RTSP Headers . . . . . . . . . . . . . . . . . . . . . . 150
22.4.1. Description . . . . . . . . . . . . . . . . . . . . 150
22.4.2. Registering New Headers with IANA . . . . . . . . . 151
22.4.3. Registered entries . . . . . . . . . . . . . . . . . 151
22.5. Transport Header Registries . . . . . . . . . . . . . . 152
22.5.1. Transport Protocol Specification . . . . . . . . . . 152
22.5.2. Transport modes . . . . . . . . . . . . . . . . . . 153
22.5.3. Transport Parameters . . . . . . . . . . . . . . . . 154
22.6. Cache Directive Extensions . . . . . . . . . . . . . . . 154
22.7. Accept-Credentials . . . . . . . . . . . . . . . . . . . 155
22.7.1. Accept-Credentials policies . . . . . . . . . . . . 155
22.7.2. Accept-Credentials hash algorithms . . . . . . . . . 155
22.8. Range header formats . . . . . . . . . . . . . . . . . . 156
22.9. URI Schemes . . . . . . . . . . . . . . . . . . . . . . 156
22.9.1. The rtsp URI Scheme . . . . . . . . . . . . . . . . 156
22.9.2. The rtsps URI Scheme . . . . . . . . . . . . . . . . 157
22.9.3. The rtspu URI Scheme . . . . . . . . . . . . . . . . 158
22.10. SDP attributes . . . . . . . . . . . . . . . . . . . . . 159
23. References . . . . . . . . . . . . . . . . . . . . . . . . . 160
23.1. Normative References . . . . . . . . . . . . . . . . . . 160
23.2. Informative References . . . . . . . . . . . . . . . . . 162
Appendix A. Examples . . . . . . . . . . . . . . . . . . . . . . 165
A.1. Media on Demand (Unicast) . . . . . . . . . . . . . . . 165
A.2. Media on Demand using Pipelining . . . . . . . . . . . . 169
A.3. Media on Demand (Unicast) . . . . . . . . . . . . . . . 171
A.4. Single Stream Container Files . . . . . . . . . . . . . 175
A.5. Live Media Presentation Using Multicast . . . . . . . . 177
A.6. Capability Negotiation . . . . . . . . . . . . . . . . . 178
Appendix B. RTSP Protocol State Machine . . . . . . . . . . . . 180
B.1. States . . . . . . . . . . . . . . . . . . . . . . . . . 180
B.2. State variables . . . . . . . . . . . . . . . . . . . . 180
B.3. Abbreviations . . . . . . . . . . . . . . . . . . . . . 180
B.4. State Tables . . . . . . . . . . . . . . . . . . . . . . 181
Appendix C. Media Transport Alternatives . . . . . . . . . . . . 186
C.1. RTP . . . . . . . . . . . . . . . . . . . . . . . . . . 186
C.1.1. AVP . . . . . . . . . . . . . . . . . . . . . . . . 186
C.1.2. AVP/UDP . . . . . . . . . . . . . . . . . . . . . . 186
C.1.3. AVPF/UDP . . . . . . . . . . . . . . . . . . . . . . 187
C.1.4. SAVP/UDP . . . . . . . . . . . . . . . . . . . . . . 188
C.1.5. SAVPF/UDP . . . . . . . . . . . . . . . . . . . . . 188
C.1.6. RTCP usage with RTSP . . . . . . . . . . . . . . . . 188
C.2. RTP over TCP . . . . . . . . . . . . . . . . . . . . . . 189
Schulzrinne, et al. Expires August 28, 2008 [Page 6]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C.2.1. Interleaved RTP over TCP . . . . . . . . . . . . . . 189
C.2.2. RTP over independent TCP . . . . . . . . . . . . . . 189
C.2.3. Handling NPT Jumps in the RTP Media Layer . . . . . 193
C.2.4. Handling RTP Timestamps after PAUSE . . . . . . . . 196
C.2.5. RTSP / RTP Integration . . . . . . . . . . . . . . . 198
C.2.6. Scaling with RTP . . . . . . . . . . . . . . . . . . 198
C.2.7. Maintaining NPT synchronization with RTP
timestamps . . . . . . . . . . . . . . . . . . . . . 198
C.2.8. Continuous Audio . . . . . . . . . . . . . . . . . . 198
C.2.9. Multiple Sources in an RTP Session . . . . . . . . . 198
C.2.10. Usage of SSRCs and the RTCP BYE Message During an
RTSP Session . . . . . . . . . . . . . . . . . . . . 198
C.3. Future Additions . . . . . . . . . . . . . . . . . . . . 199
Appendix D. Use of SDP for RTSP Session Descriptions . . . . . . 200
D.1. Definitions . . . . . . . . . . . . . . . . . . . . . . 200
D.1.1. Control URI . . . . . . . . . . . . . . . . . . . . 200
D.1.2. Media Streams . . . . . . . . . . . . . . . . . . . 201
D.1.3. Payload Type(s) . . . . . . . . . . . . . . . . . . 202
D.1.4. Format-Specific Parameters . . . . . . . . . . . . . 202
D.1.5. Directionality of media stream . . . . . . . . . . . 202
D.1.6. Range of Presentation . . . . . . . . . . . . . . . 203
D.1.7. Time of Availability . . . . . . . . . . . . . . . . 204
D.1.8. Connection Information . . . . . . . . . . . . . . . 204
D.1.9. Entity Tag . . . . . . . . . . . . . . . . . . . . . 204
D.2. Aggregate Control Not Available . . . . . . . . . . . . 205
D.3. Aggregate Control Available . . . . . . . . . . . . . . 205
D.4. RTSP external SDP delivery . . . . . . . . . . . . . . . 206
Appendix E. Minimal RTSP Implementation . . . . . . . . . . . . 208
E.1. Minimal Core Implementation . . . . . . . . . . . . . . 208
E.2. Recommended Core Implementation . . . . . . . . . . . . 208
E.3. The Basic Playback Feature Support . . . . . . . . . . . 209
E.3.1. Client . . . . . . . . . . . . . . . . . . . . . . . 209
E.3.2. Server . . . . . . . . . . . . . . . . . . . . . . . 209
E.3.3. Proxy . . . . . . . . . . . . . . . . . . . . . . . 210
E.4. Secure Transport . . . . . . . . . . . . . . . . . . . . 210
Appendix F. Requirements for Unreliable Transport of RTSP . . . 211
Appendix G. Backwards Compatibility Considerations . . . . . . . 213
G.1. Play Request in Play mode . . . . . . . . . . . . . . . 213
G.2. Using Persistent Connections . . . . . . . . . . . . . . 213
Appendix H. Open Issues . . . . . . . . . . . . . . . . . . . . 214
Appendix I. Changes . . . . . . . . . . . . . . . . . . . . . . 216
I.1. Changes needing to be updated . . . . . . . . . . . . . 221
Appendix J. Acknowledgements . . . . . . . . . . . . . . . . . . 223
J.1. Contributors . . . . . . . . . . . . . . . . . . . . . . 223
Appendix K. RFC Editor Consideration . . . . . . . . . . . . . . 225
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 226
Intellectual Property and Copyright Statements . . . . . . . . . 227
Schulzrinne, et al. Expires August 28, 2008 [Page 7]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
1. Introduction
1.1. Scope and Background
This memo defines version 2.0 of the Real Time Streaming Protocol
(RTSP 2.0) which is an application-level protocol for control over
the delivery of data with real-time properties, typically streaming
media. Streaming media is, for instance, video on demand or audio
life streaming. Put simply, RTSP acts as a "network remote control"
for multimedia servers, as you know it from your TV set.
The protocol operates between RTSP 2.0 clients and servers, but also
supports the usage of RTSP 2.0 proxies between clients and servers.
Basically, clients can request information about streaming media from
servers, by asking for a description of the media or use media
description provided externally. Based on the media description
clients can request to play out the media, pause it, or stop it
completely, as known from a regular TV remote control. The requested
media can consist of multiple audio and video streams that are
delivered as a time- synchronized stream from servers to clients.
This memorandum describes the use of RTSP over a reliable connection
based transport level protocol, such as TCP. For security, TLS over
a connection oriented transport is supported.
There is no notion of an RTSP connection in the protocol. Instead,
an RTSP server maintains a session labeled by an identifier to
associate groups of media streams and their states. An RTSP session
is not tied to a transport-level connection such as a TCP connection.
During a session, a client may open and close multiple reliable
transport connections to the server to issue RTSP requests for that
session.
The set of streams to be controlled in an RTSP session is defined by
a presentation description. This memorandum does not define a format
for the presentation description. However Appendix D describes how
SDP [RFC4566] is used for this purpose. The streams controlled by
RTSP may use RTP [RFC3550] for their data transport, but the
operation of RTSP does not depend on the transport mechanism used to
carry continuous media. RTSP is intentionally similar in syntax and
operation to HTTP/1.1 [RFC2616] so that extension mechanisms to HTTP
can in most cases also be applied to RTSP.
The RTSP 2.0 protocol supports the following operations:
Schulzrinne, et al. Expires August 28, 2008 [Page 8]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Retrieval of media from media server: The client can either request
a presentation description via RTSP DESCRIBE, HTTP or some
other method. If the presentation is being multicast, the
presentation description contains the multicast addresses and
ports to be used for the continuous media. If the presentation
is to be sent only to the client via unicast, the client
provides the destination.
Invitation of a media server to a conference: A media server can be
"invited" to join an existing conference to play back media
into the presentation. This mode is useful, for example, in
distributed teaching applications. Several parties in the
conference may take turns "pushing the remote control buttons".
Note: This functionality will require RTSP external application
level functionality.
RTSP requests may be handled by proxies, tunnels and caches as in
HTTP/1.1 [RFC2616].
1.2. RTSP Specificication Update
This memorandum specifies RTSP 2.0 which is an update of RTSP 1.0, a
proposed standard defined in [RFC2326]. The goal of this version is
to correct the many flaws that have been identified in RTSP 1.0 since
its publication. The corrections are such that backwards
compatibility was impossible. Thus a new version was deemed the most
appropriate solution to get a more functional protocol. There are no
plans to revise RTSP 1.0. Appendix I catalogs the changes of this
version in relation to RTSP 1.0.
RTSP 2.0 has reduced functionality compared to RTSP 1.0 and aims at
specifying the RTSP core, functionality and rules for extensions, and
basic interaction with the media delivery protocol RTP [RFC3550].
Any other functionality would need to be published as extension
documents. This specification provides rules for such extensions and
defines registries to avoid naming collisions.
1.3. Notational Conventions
Since many of the definitions and syntax are identical to HTTP/1.1,
this specification only points to the section where they are defined
rather than copying it. For brevity, [HX.Y] is to be taken to refer
to Section X.Y of the current HTTP/1.1 specification ([RFC2616]).
All the mechanisms specified in this document are described in both
prose and the Augmented Backus-Naur form (ABNF) described in detail
in [RFC4234].
Schulzrinne, et al. Expires August 28, 2008 [Page 9]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Indented and smaller-type paragraphs are used to provide informative
background and motivation. This is intended to give readers who were
not involved with the formulation of the specification an
understanding of why things are the way they are in RTSP.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
The word, "unspecified" is used to indicate functionality or features
that are not defined in this specification. Such functionality
cannot be used in a standardized manner without further definition in
an extension specification to RTSP.
1.4. Terminology
Some of the terminology has been adopted from HTTP/1.1 [RFC2616].
Terms not listed here are defined as in HTTP/1.1.
Aggregate control: The concept of controlling multiple streams using
a single timeline, generally maintained by the server. A client,
for example, uses aggregate control when it issues a single play
or pause message to simultaneously control both the audio and
video in a movie. A session which is under aggregate control is
referred to as an aggregated session.
Aggregate control URI: The URI used in an RTSP request to refer to
and control an aggregated session. It normally, but not always,
corresponds to the presentation URI specified in the session
description. See Section 13.3 for more information.
Conference: A multiparty, multimedia presentation, where "multi"
implies greater than or equal to one.
Client: The client requests media service from the media server.
Connection: A transport layer virtual circuit established between
two programs for the purpose of communication.
Container file: A file which may contain multiple media streams
which often constitutes a presentation when played together. The
concept of a container file is not embedded in the protocol.
However, RTSP servers may offer aggregate control on the media
streams within these files.
Schulzrinne, et al. Expires August 28, 2008 [Page 10]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Continuous media: Data where there is a timing relationship between
source and sink; that is, the sink needs to reproduce the timing
relationship that existed at the source. The most common examples
of continuous media are audio and motion video. Continuous media
can be real-time (interactive or conversational), where there is a
"tight" timing relationship between source and sink, or streaming
(playback), where the relationship is less strict.
Entity: The information transferred as the payload of a request or
response. An entity consists of meta-information in the form of
entity-header fields and content in the form of an entity-body, as
described in Section 9.
Feature-tag: A tag representing a certain set of functionality, i.e.
a feature.
IRI: Internationalized Resource Identifier, is the same as an URI,
with the exception that it allows characters from the whole
Universal Character Set (Unicode/ISO 10646), rather than the US-
ASCII only. See [RFC3987] for more information.
Live: Normally used to describe a presentation or session with media
coming from an ongoing event. This generally results in the
session having an unbound or only loosely defined duration, and
sometimes no seek operations are possible.
Media initialization: Datatype/codec specific initialization. This
includes such things as clock rates, color tables, etc. Any
transport-independent information which is required by a client
for playback of a media stream occurs in the media initialization
phase of stream setup.
Media parameter: Parameter specific to a media type that may be
changed before or during stream playback.
Media server: The server providing playback services for one or more
media streams. Different media streams within a presentation may
originate from different media servers. A media server may reside
on the same host or on a different host from which the
presentation is invoked.
Media server indirection: Redirection of a media client to a
different media server.
(Media) stream: A single media instance, e.g., an audio stream or a
video stream as well as a single whiteboard or shared application
group. When using RTP, a stream consists of all RTP and RTCP
packets created by a source within an RTP session.
Schulzrinne, et al. Expires August 28, 2008 [Page 11]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Message: The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined in
Section 20 and transmitted over a connection or a connectionless
transport.
Non-Aggregated Control: Control of a single media stream. This is
only possible in RTSP sessions with a single media.
Participant: Member of a conference. A participant may be a
machine, e.g., a playback server.
Presentation: A set of one or more streams presented to the client
as a complete media feed and described by a presentation
description as defined below. Presentations with more than one
media stream are often handled in RTSP under aggregate control.
Presentation description: A presentation description contains
information about one or more media streams within a presentation,
such as the set of encodings, network addresses and information
about the content. Other IETF protocols such as SDP ([RFC4566])
use the term "session" for a presentation. The presentation
description may take several different formats, including but not
limited to the session description protocol format, SDP.
Response: An RTSP response. If an HTTP response is meant, that is
indicated explicitly.
Request: An RTSP request. If an HTTP request is meant, that is
indicated explicitly.
Request-URI: The URI used in a request to indicate the resource on
which the request is to be performed.
RTSP agent: Refers to either an RTSP client, an RTSP server, or an
RTSP Proxy. In this specification, there are many capabilities
that are common to these three entities such as the capability to
send requests or receive responses. This term will be used when
describing functionality that is applicable to all three of these
entities.
RTSP session: A stateful abstraction upon which the main control
methods of RTSP operate. An RTSP session is a server entity; it
is created, maintained and destroyed by the server. It is
established by an RTSP server upon the completion of a successful
SETUP request (when a 200 OK response is sent) and is labelled
with a session identifier at that time. The session exists until
timed out by the server or explicitly removed by a TEARDOWN
request. An RTSP session is a stateful entity; an RTSP server
Schulzrinne, et al. Expires August 28, 2008 [Page 12]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 13]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
2. RTSP Introduction
2.1. Protocol Properties
RTSP has the following properties:
Extendable: New methods and parameters can be easily added to RTSP.
Easy to parse: RTSP can be parsed by standard HTTP or MIME parsers.
Secure: RTSP re-uses web security mechanisms, either at the
transport level (TLS, [RFC4346]) or within the protocol itself.
All HTTP authentication mechanisms such as basic ([RFC2616]) and
digest authentication ([RFC2617]) are directly applicable.
Transport-independent: RTSP does not preclude the use of unreliable
datagram protocol (UDP) ([RFC0768]) as it would be possible to
implement application-level reliability. The use of a
connectionless datagram protocol such as UDP requires additional
definition that may be provided as extensions to the core RTSP
specification. The reliable stream protocol TCP ([RFC0793]) and
the secured reliable stream protocol TLS over TCP [RFC4346] are
the currently defined transport protocols for RTSP messages.
Media-delivery protocol independent: The operation of RTSP does not
depend on the transport mechanism used to carry continuous media.
While most real-time media will use RTP as a transport protocol,
RTSP does not preclude the use of other protocols such as MPEG-2
[ISO.13818-1.2000]. The use of other protocols requires
additional definition that may be provided as extensions to the
core RTSP specification.
Multi-server capable: Each media stream within a presentation can
reside on a different server. The client automatically
establishes several concurrent control sessions with the different
media servers. Media synchronization in those cases is performed
at the transport level.
Separation of stream control and conference initiation: Stream
control is divorced from inviting a media server to a conference.
In particular, SIP [RFC3261] or H.323 [ITU.H323.1996] may be used
to invite a server to a conference; however, the exact procedures
are unspecified.
Suitable for professional applications: RTSP supports frame- level
accuracy through SMPTE time stamps to allow remote digital
editing.
Schulzrinne, et al. Expires August 28, 2008 [Page 14]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Presentation description neutral: The protocol does not impose a
particular presentation description or metafile format and can
convey the type of format to be used. However, the presentation
description is required to contain at least one RTSP URI.
Proxy and firewall friendly: The protocol should be readily handled
by both application and transport-layer (SOCKS [RFC1961])
firewalls. A firewall may need to understand the SETUP method to
open a "hole" for the media stream.
HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so that
the existing infrastructure can be reused. This infrastructure
includes PICS (Platform for Internet Content Selection
[W3C.REC-PICS-services] [W3C.REC-PICS-labels]) for associating
labels with content. However, RTSP does not just add methods to
HTTP since controlling continuous media requires server state in
most cases.
Appropriate server control: If a client can start a stream, it needs
to be able to stop a stream. Servers should not start streaming
to clients in such a way that clients cannot stop the stream.
Transport negotiation: The client can negotiate the transport method
prior to actually needing to process a continuous media stream.
2.2. RTSP's Relationship to HTTP
RTSP is intentionally similar in syntax and operation to HTTP/1.1
[RFC2616] so that extension mechanisms to HTTP can in most cases also
be applied to RTSP. However, RTSP differs in a number of important
aspects from HTTP:
* RTSP introduces a number of new methods and has a different
protocol identifier.
* RTSP has the notion of a session built into the protocol.
* An RTSP server needs to maintain state in almost all cases, as
opposed to the stateless nature of HTTP.
* Both an RTSP server and client can issue requests.
* Data is usually carried out-of-band by a different protocol.
Session descriptions returned in a DESCRIBE response (see
Section 13.2) and interleaving of RTP with RTSP over TCP are
exceptions to this rule (see Section 14).
Schulzrinne, et al. Expires August 28, 2008 [Page 15]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
8859-1, consistent with HTML internationalization efforts
[RFC2070].
* The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1
[RFC2616] carries only the absolute path in the request and
puts the host name in a separate header field.
This makes "virtual hosting" easier, where a single host with
one IP address hosts several document trees.
2.3. Extending RTSP
Since not all media servers have the same functionality, media
servers by necessity will support different sets of requests. For
example:
o A server may not be capable of seeking (absolute positioning) if
it is to support live events only.
o Some servers may not support setting stream parameters and thus
not support GET_PARAMETER and SET_PARAMETER.
o Some server may support an RTSP extension.
It is up to the creators of presentation descriptions not to ask the
impossible of a server. This situation is similar in HTTP/1.1
[RFC2616], where the methods described in [H19.5] are not likely to
be supported across all servers.
RTSP can be extended in three ways, listed here in order of the
magnitude of changes supported:
o Existing methods can be extended with new parameters, e.g.
headers, as long as these parameters can be safely ignored by the
recipient. If the client needs negative acknowledgement when a
method extension is not supported, a tag corresponding to the
extension may be added in the field of the Require or Proxy-
Require headers (see Section 16.32).
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 16]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
o A new version of the protocol can be defined, allowing almost all
aspects (except the position of the protocol version number) to
change. A new version of the protocol MUST be registered through
an IETF standard track document.
The basic capability discovery mechanism can be used to both discover
support for a certain feature and to ensure that a feature is
available when performing a request. For detailed explanation of
this see Section 11.
2.4. Overall Operation
Each presentation and media stream is identified by an RTSP URI. The
overall presentation and the properties of the media the presentation
is composed of are defined by a presentation description file, the
format of which is outside the scope of this specification. The
presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored
on the media server.
For the purposes of this specification, a presentation description is
assumed to describe one or more presentations, each of which
maintains a common time axis. For simplicity of exposition and
without loss of generality, it is assumed that the presentation
description contains exactly one such presentation. A presentation
may contain several media streams.
The presentation description file contains a description of the media
streams making up the presentation, including their encodings,
language, and other parameters that enable the client to choose the
most appropriate combination of media. In this presentation
description, each media stream that is individually controllable by
RTSP is identified by an RTSP URI, which points to the media server
handling that particular media stream and names the stream stored on
that server. Several media streams can be located on different
servers; for example, audio and video streams can be split across
servers for load sharing. The description also enumerates which
transport methods the server is capable of.
Besides the media parameters, the network destination address and
port need to be determined. Several modes of operation can be
distinguished:
Unicast: The media is transmitted to the source of the RTSP request
or the requested destination, with the port number chosen by the
client. Alternatively, the media is transmitted on the same
reliable stream as RTSP.
Schulzrinne, et al. Expires August 28, 2008 [Page 17]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Multicast, server chooses address: The media server picks the
multicast address and port. This is the typical case for a live
or near-media-on-demand transmission.
Multicast, client chooses address: If the server is to participate
in an existing multicast conference, the multicast address, port
and encryption key are given by the conference description,
established by means outside the scope of this specification, for
example by a SIP created conference.
2.5. RTSP States
RTSP controls a stream which may be sent via a separate protocol,
independent of the control channel. For example, RTSP control may be
transported on a TCP connection while the media data is conveyed via
UDP. Thus, data delivery continues even if no RTSP requests are
received by the media server. Also, during its lifetime a single
media stream may be controlled by RTSP requests issued sequentially
on different TCP connections. Therefore, the server needs to
maintain "session state" to be able to correlate RTSP requests with a
stream. The state transitions are described in Appendix A.
Many methods in RTSP do not contribute to state. However, the
following play a central role in defining the allocation and usage of
stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, and
TEARDOWN.
SETUP: Causes the server to allocate resources for a stream and
create an RTSP session.
PLAY: Starts data transmission on a stream allocated via SETUP.
PAUSE: Temporarily halts a stream without freeing server resources.
REDIRECT: Indicates that the session should be moved to a new server
or location
TEARDOWN: Frees resources associated with the stream. The RTSP
session ceases to exist on the server.
RTSP methods that contribute to state use the Session header field
(Section 16.44) to identify the RTSP session whose state is being
manipulated. The server generates session identifiers in response to
SETUP requests (Section 13.3).
Schulzrinne, et al. Expires August 28, 2008 [Page 18]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
2.6. Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may
interact with HTTP in that the initial contact with streaming content
will often be made through a web page. The current protocol
specification aims to allow different hand-off points between a web
server and the media server implementing RTSP. For example, the
presentation description can be retrieved using HTTP or RTSP, which
reduces round trips in web-browser-based scenarios, yet also allows
for stand alone RTSP servers and clients which do not rely on HTTP at
all. However, RTSP differs fundamentally from HTTP in that most data
delivery takes place out-of-band in a different protocol. HTTP is an
asymmetric protocol where the client issues requests and the server
responds. In RTSP, both the media client and media server can issue
requests. RTSP requests are also stateful; they may set parameters
and continue to control a media stream long after the request has
been acknowledged.
Re-using HTTP functionality has advantages in at least two areas,
namely security and proxies. The requirements are very similar, so
having the ability to adopt HTTP work on caches, proxies and
authentication is valuable.
RTSP assumes the existence of a presentation description format that
can express both static and temporal properties of a presentation
containing several media streams. Session Description Protocol (SDP)
[RFC4566] is generally the format of choice; however, RTSP is not
bound to it. For data delivery, most real-time media will use RTP as
a transport protocol. While RTSP works well with RTP, it is not tied
to RTP.
Schulzrinne, et al. Expires August 28, 2008 [Page 19]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
3. RTSP Use Cases
This section describes the most important and considered use cases
for RTSP. They are listed in descending order of importance in
regards to ensuring that all necessary functionality is present.
This specification only fully supports usage of the two first. Also
in these first two cases, there are special cases or exceptions that
are not supported without extensions, e.g. the redirection of media
to another address than the controlling entity.
3.1. On-demand Playback of Stored Content
An RTSP capable server stores content suitable for being streamed to
a client. A client desiring playback of any of the stored content
uses RTSP to set up the media transport required to deliver the
desired content. RTSP is then used to initiate, halt and manipulate
the actual transmission (playout) of the content. RTSP is also
required to provide necessary description and synchronization
information for the content.
The above high level description can be broken down into a number of
functions that RTSP needs to be capable of.
Presentation Description: Provide initialization information about
the presentation (content); for example, which media codecs are
needed for the content. Other information that is important
includes the number of media stream the presentation contains,
the transport protocols used for the media streams, and
identifiers for these media streams. This information is
required before setup of the content is possible and to
determine if the client is even capable of using the content.
This information need not be sent using RTSP; other external
protocols can be used to transmit the transport presentation
descriptions. Two good examples are the use of HTTP [RFC2616]
or email to fetch or receive presentation descriptions like SDP
[RFC4566]
Setup: Set up some or all of the media streams in a presentation.
The setup itself consist of selecting the protocol for media
transport and the necessary parameters for the protocol, like
addresses and ports.
Control of Transmission: After the necessary media streams have been
established the client can request the server to start
transmitting the content. The client must be allowed to start
or stop the transmission of the content at arbitrary times.
The client must also be able to start the transmission at any
Schulzrinne, et al. Expires August 28, 2008 [Page 20]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
point in the timeline of the presentation.
Synchronization: For media transport protocols like RTP [RFC3550] it
might be beneficial to carry synchronization information within
RTSP. This may be due to either the lack of inter-media
synchronization within the protocol itself, or the potential
delay before the synchronization is established (which is the
case for RTP when using RTCP).
Termination: Terminate the established contexts.
For this use case there are a number of assumptions about how it
works. These are:
On-Demand content: The content is stored at the server and can be
accessed at any time during a time period when it is intended
to be available.
Independent sessions: A server is capable of serving a number of
clients simultaneously, including from the same piece of
content at different points in that presentations time-line.
Unicast Transport: Content for each individual client is transmitted
to them using unicast traffic.
It is also possible to redirect the media traffic to a different
destination than that of the entity controlling the traffic.
However, allowing this without appropriate mechanisms for checking
that the destination approves of this allows for distributed denial
of service attacks (DDoS).
3.2. Unicast distribution of Live Content
This use cases is similar to the above on-demand content case (see
Section 3.1) the difference is the nature of the content itself.
Live content is continuously distributed as it becomes available from
a source; i.e., the main difference from on-demand is that one starts
distributing content before the end of it has become available to the
server.
In many cases the consumer of live content is only interested in
consuming what is actually happens "now"; i.e., very similar to
broadcast TV. However in this case it is assumed that there exist no
broadcast or multicast channel to the users, and instead the server
functions as a distribution node, sending the same content to
multiple receivers, using unicast traffic between server and client.
This unicast traffic and the transport parameters are individually
negotiated for each receiving client.
Schulzrinne, et al. Expires August 28, 2008 [Page 21]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Another aspect of live content is that it often has a very limited
time of availability, as it is only is available for the duration of
the event the content covers. An example of such a live content
could be a music concert which lasts 2 hour and starts at a
predetermined time. Thus there is need to announce when and for how
long the live content is available.
In some cases, the server providing live content may be saving some
or all of the content to allow clients to pause the stream and resume
it from the paused point, or to "rewind" and play continuously from a
point earlier than the live point. Hence, this use case does not
necessarily exclude playing from other than the live point of the
stream, playing with scales other than 1.0, etc.
3.3. On-demand Playback using Multicast
It is possible to use RTSP to request that media be delivered to a
multicast group. The entity setting up the session (the controller)
will then control when and what media is delivered to the group.
This use case has some potential for denial of service attacks by
flooding a multicast group. Therefore, a mechanism is needed to
indicate that the group actually accepts the traffic from the RTSP
server.
An open issue in this use case is how one ensures that all receivers
listening to the multicast or broadcast receives the session
presentation configuring the receivers.
3.4. Inviting an RTSP server into a conference
If one has an established conference or group session, it is possible
to have an RTSP server distribute media to the whole group.
Transmission to the group is simplest when controlled by a single
participant or leader of the conference. Shared control might be
possible, but would require further investigation and possibly
extensions.
This use case assumes that there exists either multicast or a
conference focus that redistribute media to all participants.
This use case is intended to be able to handle the following
scenario: A conference leader or participant (hereafter called the
controller) has some pre-stored content on an RTSP server that he
wants to share with the group. The controller sets up an RTSP
session at the streaming server for this content and retrieves the
session description for the content. The destination for the media
content is set to the shared multicast group or conference focus.
When desired by the controller, he/she can start and stop the
Schulzrinne, et al. Expires August 28, 2008 [Page 22]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
transmission of the media to the conference group.
There are several issues with this use case that are not solved by
this core specification for RTSP:
Denial of service: To avoid an RTSP server from being an unknowing
participant in a denial of service attack the server needs to
be able to verify the destination's acceptance of the media.
Such a mechanism to verify the approval of received media does
not yet exist; instead, only policies can be used, which can be
made to work in controlled environments.
Distributing the presentation description to all participants in the
group: To enable a media receiver to correctly decode the content
the media configuration information needs to be distributed
reliably to all participants. This will most likely require
support from an external protocol.
Passing control of the session: If it is desired to pass control of
the RTSP session between the participants, some support will be
required by an external protocol to exchange state information
and possibly floor control of who is controlling the RTSP
session.
If there interest in this use case, further work is required on the
necessary extensions.
3.5. Live Content using Multicast
This use case in its simplest form does not require any use of RTSP
at all; this is what multicast conferences being announced with SAP
and SDP are intended to handle. However in use cases where more
advanced features like access control to the multicast session are
desired, RTSP could be used for session establishment.
A client desiring to join a live multicasted media session with
cryptographic (encryption) access control could use RTSP in the
following way. The source of the session announces the session and
gives all interested an RTSP URI. The client connects to the server
and requests the presentation description, allowing configuration for
reception of the media. In this step it is possible for the client
to use secured transport and any desired level of authentication; for
example, for billing or access control. An RTSP link also allows for
load balancing between multiple servers.
If these were the only goals, they could be achieved by simply using
HTTP. However, for cases where the sender likes to keep track of
each individual receiver of a session, and possibly use the session
Schulzrinne, et al. Expires August 28, 2008 [Page 23]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 a
method like ANNOUNCE (see [RFC2326]) might be suitable; however, it
will require a RTSP extension to revive the method.
Schulzrinne, et al. Expires August 28, 2008 [Page 24]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
4. Protocol Parameters
4.1. RTSP Version
HTTP specification section [H3.1] applies, with "HTTP" replaced by
"RTSP". This specification defines version 2.0 of RTSP.
4.2. RTSP IRI and URI
RTSP 2.0 defines and registers three URI schemas "rtsp", "rtsps" and
"rtspu". The usage of the last, "rtspu", is unspecified in RTSP 2.0,
and is defined here to register and reserve the URI scheme that is
defined in RTSP 1.0. The "rtspu" scheme indicates undefined
transport of the RTSP messages over unreliable transport (UDP). The
syntax of "rtsp" and "rtsps" URIs has been changed from RTSP 1.0.
This specification also defines the format of the RTSP IRI [RFC3987]
that can be used as RTSP resource identifiers and locators, in web
pages, user interfaces, on paper, etc. However, the RTSP request
message format only allows usage of the absolute URI format. The
RTSP IRI format SHALL use the rules and transformation for IRIs
defined in [RFC3987]. This way RTSP 2.0 URIs for request can be
produced from an RTSP IRI.
The RTSP IRI and URI are both syntax restricted compared to the
generic syntax defined in [RFC3986] and RFC [RFC3987]:
o An absolute URI requires the authority part; i.e., a host identity
must be provided.
o Parameters in the path element are prefixed with the reserved
separator ";".
The RTSP URI and IRI is case sensitive, with the exception of those
parts that [RFC3986] and [RFC3987] defines as case-insensitive; for
example, the scheme and host part.
The fragment identifier is used as defined in sections 3.5 and 4.3 of
[RFC3986], i.e. the fragment is to be stripped from the URI by the
requestor and not included in the request. The user agent also needs
to interpret the value of the fragment based on the media type the
request relates to; i.e., the media type indicated in Content-Type
header in the response to DESCRIBE.
The syntax of any URI query string is unspecified and responder
(usually the server) specific. The query is, from the requestor's
perspective, an opaque string and needs to be handled as such.
Schulzrinne, et al. Expires August 28, 2008 [Page 25]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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
[RFC4346], see (Section 19).
For the scheme "rtsp", if no port number is provided in the authority
part of the URI port number 554 SHALL be used. For the scheme
"rtsps", the TCP port 322 is registered and SHALL be assumed.
A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions of URIs
[RFC3986]. URIs may refer to a stream or an aggregate of streams;
i.e., a presentation. Accordingly, requests described in
(Section 13) can apply to either the whole presentation or an
individual stream within the presentation. Note that some request
methods can only be applied to streams, not presentations, and vice
versa.
For example, the RTSP URI:
rtsp://media.example.com:554/twister/audiotrack
may identify the audio stream within the presentation "twister",
which can be controlled via RTSP requests issued over a TCP
connection to port 554 of host media.example.com.
Also, the RTSP URI:
rtsp://media.example.com:554/twister
identifies the presentation "twister", which may be composed of audio
and video streams, but could also be something else like a random
media redirector.
This does not imply a standard way to reference streams in URIs.
The presentation description defines the hierarchical
relationships in the presentation and the URIs for the individual
streams. A presentation description may name a stream "a.mov" and
the whole presentation "b.mov".
The path components of the RTSP URI are opaque to the client and do
not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be used
with non-RTSP media control protocols simply by replacing the
scheme in the URI.
Schulzrinne, et al. Expires August 28, 2008 [Page 26]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
4.3. Session Identifiers
Session identifiers are strings of any arbitrary length but with a
minimum length of 8 characters. A session identifier MUST be chosen
cryptographically random (see [RFC4086]) and MUST be at least 8
characters long (can contain a maximum of 48 bits of entropy) to make
guessing it more difficult. It is RECOMMENDED that it contains 128
bits of entropy, i.e. approxamitely 22 characters from a high quality
generator. (see Section 21.) However, it needs to be noted that the
session identifier does not provide any security against session
hijacking unless it is kept confidential between client, server and
trusted proxies.
4.4. SMPTE Relative Timestamps
A SMPTE relative timestamp expresses time relative to the start of
the clip. Relative timestamps are expressed as SMPTE time codes for
frame-level access accuracy. The time code has the format
hours:minutes:seconds:frames.subframes,
with the origin at the start of the clip. The default smpte format
is "SMPTE 30 drop" format, with frame rate is 29.97 frames per
second. Other SMPTE codes MAY be supported (such as "SMPTE 25")
through the use of alternative use of "smpte-type". For SMPTE 30,
the "frames" field in the time value can assume the values 0 through
29. The difference between 30 and 29.97 frames per second is handled
by dropping the first two frame indices (values 00 and 01) of every
minute, except every tenth minute. If the frame and the subframe
values are zero, they may be omitted. Subframes are measured in one-
hundredth of a frame.
Examples:
smpte=10:12:33:20-
smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01
smpte-25=10:07:00-10:07:33:05.01
4.5. Normal Play Time
Normal play time (NPT) indicates the stream absolute position
relative to the beginning of the presentation, not to be confused
with the Network Time Protocol (NTP) [RFC1305]. The timestamp
consists of a decimal fraction. The part left of the decimal may be
expressed in either seconds or hours, minutes, and seconds. The part
right of the decimal point measures fractions of a second.
Schulzrinne, et al. Expires August 28, 2008 [Page 27]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
The beginning of a presentation corresponds to 0.0 seconds. Negative
values are not defined. The special constant "now" is defined as the
current instant of a live event. It MAY only be used for live
events, and SHALL NOT be used for on-demand (i.e., non-live) content.
NPT is defined as in DSM-CC [ISO.13818-6.1995]: "Intuitively, NPT is
the clock the viewer associates with a program. It is often
digitally displayed on a VCR. NPT advances normally when in normal
play mode (scale = 1), advances at a faster rate when in fast scan
forward (high positive scale ratio), decrements when in scan reverse
(high negative scale ratio) and is fixed in pause mode. NPT is
(logically) equivalent to SMPTE time codes."
Examples:
npt=123.45-125
npt=12:05:35.3-
npt=now-
The syntax conforms to ISO 8601 [ISO.8601.2000]. The npt-sec
notation is optimized for automatic generation, the npt-hhmmss
notation for consumption by human readers. The "now" constant
allows clients to request to receive the live feed rather than the
stored or time-delayed version. This is needed since neither
absolute time nor zero time are appropriate for this case.
4.6. Absolute Time
Absolute time is expressed as ISO 8601 [ISO.8601.2000] timestamps,
using UTC (GMT). Fractions of a second may be indicated.
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC:
19961108T143720.25Z
4.7. Feature-tags
Feature-tags are unique identifiers used to designate features in
RTSP. These tags are used in Require (Section 16.38), Proxy-Require
(Section 16.32), Proxy-Supported (Section 16.33), Unsupported
(Section 16.47), and 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.,
Schulzrinne, et al. Expires August 28, 2008 [Page 28]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
"com.example.mynewfeature" is an apt name for a feature whose
inventor can be reached at "example.com"), or register the new
feature-tag with the Internet Assigned Numbers Authority (IANA) (see
IANA Section 22).
The usage of feature-tags is further described in Section 11 that
deals with capability handling.
4.8. Entity Tags
Entity tags are opaque strings that are used to compare two entities
from the same resource, for example in caches or to optimize setup
after a redirect. Further explanation is present in [H3.11]. For an
explanation of how to compare entity tags see [H13.3]. Entity tags
can be carried in the ETag header (see Section 16.21) or in SDP (see
Appendix D.1.9).
Entity tags are used in RTSP to make some methods conditional. The
methods are made conditional through the inclusion of headers, see
Section 16.24 and Section 16.26. Note that RTSP entity tags apply to
the complete presentation; i.e., both the session description and the
individual media streams. Thus entity tags can be used to verify at
setup time after a redirect that the same session description applies
to the media at the new location using the If-Match header.
Schulzrinne, et al. Expires August 28, 2008 [Page 29]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
5. RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 3629 [RFC3629]). Lines SHALL be terminated by
CRLF.
Text-based protocols make it easier to add optional parameters in
a self-describing manner. Since the number of parameters and the
frequency of commands is low, processing efficiency is not a
concern. Text-based protocols, if done carefully, also allow easy
implementation of research prototypes in scripting languages such
as Tcl, Visual Basic and Perl.
The ISO 10646 character set avoids tricky character set switching,
but is invisible to the application as long as US-ASCII is being
used. This is also the encoding used for RTCP [RFC3550]. ISO 8859-1
translates directly into Unicode with a high-order octet of zero.
ISO 8859-1 characters with the most-significant bit set are
represented as 1100001x 10xxxxxx. (See RFC 3629 [RFC3629])
Requests contain methods, the object the method is operating upon and
parameters to further describe the method. Methods are idempotent
unless otherwise noted. Methods are also designed to require little
or no state maintenance at the media server.
5.1. Message Types
See [H4.1].
5.2. Message Headers
See [H4.2].
5.3. Message Body
See [H4.3].
Unlike HTTP, the presence of a message-body in either a request or a
response MUST be signaled by the inclusion of a Content-Length header
field (see Section 16.16).
5.4. Message Length
When a message body is included with a message, the length of that
body is determined by one of the following (in order of precedence):
1. Any response message which MUST NOT include a message body (such
as the 1xx, 204, and 304 responses) is always terminated by the
Schulzrinne, et al. Expires August 28, 2008 [Page 30]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
first empty line after the header fields, regardless of the
entity-header fields present in the message. (Note: An empty
line is a line with nothing preceding the CRLF.)
2. If a Content-Length header field (Section 16.16) is present, its
value in bytes represents the length of the message-body. If
this header field is not present, a value of zero is assumed.
Unlike an HTTP message, an RTSP message MUST contain a Content-Length
header field whenever it contains a message body. Note that RTSP
does not support the HTTP/1.1 "chunked" transfer coding (see
[H3.6.1]).
Given the moderate length of presentation descriptions returned,
the server should always be able to determine its length, even if
it is generated dynamically, making the chunked transfer encoding
unnecessary.
Schulzrinne, et al. Expires August 28, 2008 [Page 31]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
6. General Header Fields
See [H4.5], except that the Pragma, Trailer, Transfer-Encoding,
Upgrade, and Warning headers are not defined. RTSP further defines
the CSeq, Pipelined-Requests, Proxy-Supported and Timestamp headers.
The general headers are listed in Table 1:
+--------------------+--------------------+
| Header Name | Defined in Section |
+--------------------+--------------------+
| Cache-Control | Section 16.10 |
| | |
| Connection | Section 16.11 |
| | |
| CSeq | Section 16.19 |
| | |
| Date | Section 16.20 |
| | |
| Pipelined-Requests | Section 16.29 |
| | |
| Proxy-Supported | Section 16.33 |
| | |
| Supported | Section 16.44 |
| | |
| Timestamp | Section 16.45 |
| | |
| Via | Section 16.50 |
+--------------------+--------------------+
Table 1: The general headers used in RTSP
Schulzrinne, et al. Expires August 28, 2008 [Page 32]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
7. Request
A request message uses the format outlined below regardless of the
direction of a request, client to server or server to client:
o Request line, containing the method to be applied to the resource,
the identifier of the resource, and the protocol version in use;
o Zero or more Header lines, that can be of the following types:
general (Section 6), request (Section 7.2), or entity
(Section 9.1);
o One empty line (CRLF) to indicate the end of the header section;
o Optionally a message body (entity), consisting of one or more
lines. The length of the message body in bytes is indicated by
the Content-Length entity header.
7.1. Request Line
The request line provides the key information about the request: what
method, on what resources and using which RTSP version. The methods
that are defined by this specification are listed in Table 2.
Schulzrinne, et al. Expires August 28, 2008 [Page 33]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
+---------------+--------------------+
| Method | Defined in Section |
+---------------+--------------------+
| DESCRIBE | Section 13.2 |
| | |
| GET_PARAMETER | Section 13.7 |
| | |
| OPTIONS | Section 13.1 |
| | |
| PAUSE | Section 13.5 |
| | |
| PLAY | Section 13.4 |
| | |
| REDIRECT | Section 13.9 |
| | |
| SETUP | Section 13.3 |
| | |
| SET_PARAMETER | Section 13.8 |
| | |
| TEARDOWN | Section 13.6 |
+---------------+--------------------+
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.
For example:
Schulzrinne, et al. Expires August 28, 2008 [Page 34]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
OPTIONS * RTSP/2.0
An OPTIONS in this form will determine the capabilities of the server
or the proxy that first receives the request. If the capability of
the specific server needs to be determined, without regard to the
capability of an intervening proxy, the server should be addressed
explicitly with an absolute URI that contains the server's address.
For example:
OPTIONS rtsp://example.com RTSP/2.0
7.2. Request Header Fields
The RTSP headers in Table 3 can be included in a request, as request
headers, to modify the specifics of the request. Some of these
headers may also be used in the response to a request, as response
headers, to modify the specifics of a response (Section 8.2).
+--------------------+--------------------+
| Header | Defined in Section |
+--------------------+--------------------+
| Accept | Section 16.1 |
| | |
| Accept-Credentials | Section 16.2 |
| | |
| Accept-Encoding | Section 16.3 |
| | |
| Accept-Language | Section 16.4 |
| | |
| Authorization | Section 16.7 |
| | |
| Bandwidth | Section 16.8 |
| | |
| Blocksize | Section 16.9 |
| | |
| From | Section 16.23 |
| | |
| If-Match | Section 16.24 |
| | |
| If-Modified-Since | Section 16.25 |
| | |
| If-None-Match | Section 16.26 |
| | |
| Proxy-Require | Section 16.32 |
| | |
| Range | Section 16.35 |
| | |
Schulzrinne, et al. Expires August 28, 2008 [Page 35]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| Referer | Section 16.36 |
| | |
| Require | Section 16.38 |
| | |
| Scale | Section 16.40 |
| | |
| Session | Section 16.43 |
| | |
| Speed | Section 16.41 |
| | |
| Supported | Section 16.44 |
| | |
| Transport | Section 16.46 |
| | |
| User-Agent | Section 16.48 |
+--------------------+--------------------+
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 correct processing of the header.
Schulzrinne, et al. Expires August 28, 2008 [Page 36]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
8. Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version.
Also, RTSP defines additional status codes and does not define some
of the HTTP codes. The valid response codes and the methods they can
be used with are listed in Table 4.
After receiving and interpreting a request message, the recipient
responds with an RTSP response message.
8.1. Status-Line
The first line of a Response message is the Status-Line, consisting
of the protocol version followed by a numeric status code and the
textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence.
<RTSP-Version> SP <Status-Code> SP <Reason-Phrase> CRLF
8.1.1. Status Code and Reason Phrase
The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully
defined in Section 15. The Reason-Phrase is intended to give a short
textual description of the Status-Code. The Status-Code is intended
for use by automata and the Reason-Phrase is intended for the human
user. The client is not required to examine or display the Reason-
Phrase.
The first digit of the Status-Code defines the class of response.
The last two digits do not have any categorization role. There are 5
values for the first digit:
1xx: Informational - Request received, continuing process
2xx: Success - The action was successfully received, understood, and
accepted
3rr: Redirection - Further action needs to be taken in order to
complete the request
4xx: Client Error - The request contains bad syntax or cannot be
fulfilled
Schulzrinne, et al. Expires August 28, 2008 [Page 37]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
5xx: Server Error - The server failed to fulfill an apparently valid
request
The individual values of the numeric status codes defined for
RTSP/2.0, and an example set of corresponding Reason-Phrases, are
presented in Table 4. The reason phrases listed here are only
recommended; they may be replaced by local equivalents without
affecting the protocol. Note that RTSP adopts most HTTP/1.1
[RFC2616] status codes and adds RTSP-specific status codes starting
at x50 to avoid conflicts with newly defined HTTP status codes.
RTSP status codes are extensible. RTSP applications are not required
to understand the meaning of all registered status codes, though such
understanding is obviously desirable. However, applications MUST
understand the class of any status code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if an
unrecognized status code of 431 is received by the client, it can
safely assume that there was something wrong with its request and
treat the response as if it had received a 400 status code. In such
cases, user agents SHOULD present to the user the entity returned
with the response, since that entity is likely to include human-
readable information which will explain the unusual status.
+------+----------------------------------------+-----------------+
| Code | Reason | Method |
+------+----------------------------------------+-----------------+
| 100 | Continue | all |
| | | |
| | | |
| 200 | OK | all |
| | | |
| | | |
| 300 | Multiple Choices | all |
| | | |
| 301 | Moved Permanently | all |
| | | |
| 302 | Found | all |
| | | |
| 303 | See Other | all |
| | | |
| 305 | Use Proxy | all |
| | | |
| | | |
| 400 | Bad Request | all |
| | | |
| 401 | Unauthorized | all |
Schulzrinne, et al. Expires August 28, 2008 [Page 38]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| 402 | Payment Required | all |
| | | |
| 403 | Forbidden | all |
| | | |
| 404 | Not Found | all |
| | | |
| 405 | Method Not Allowed | all |
| | | |
| 406 | Not Acceptable | all |
| | | |
| 407 | Proxy Authentication Required | all |
| | | |
| 408 | Request Timeout | all |
| | | |
| 410 | Gone | all |
| | | |
| 411 | Length Required | all |
| | | |
| 412 | Precondition Failed | DESCRIBE, SETUP |
| | | |
| 413 | Request Entity Too Large | all |
| | | |
| 414 | Request-URI Too Long | all |
| | | |
| 415 | Unsupported Media Type | all |
| | | |
| 451 | Parameter Not Understood | SET_PARAMETER |
| | | |
| 452 | reserved | n/a |
| | | |
| 453 | Not Enough Bandwidth | SETUP |
| | | |
| 454 | Session Not Found | all |
| | | |
| 455 | Method Not Valid In This State | all |
| | | |
| 456 | Header Field Not Valid | all |
| | | |
| 457 | Invalid Range | PLAY, PAUSE |
| | | |
| 458 | Parameter Is Read-Only | SET_PARAMETER |
| | | |
| 459 | Aggregate Operation Not Allowed | all |
| | | |
| 460 | Only Aggregate Operation Allowed | all |
| | | |
| 461 | Unsupported Transport | all |
| | | |
Schulzrinne, et al. Expires August 28, 2008 [Page 39]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| 462 | Destination Unreachable | all |
| | | |
| 463 | Destination Prohibited | SETUP |
| | | |
| 464 | Data Transport Not Ready Yet | PLAY |
| | | |
| 470 | Connection Authorization Required | all |
| | | |
| 471 | Connection Credentials not accepted | all |
| | | |
| 472 | Failure to establish secure connection | all |
| | | |
| | | |
| 500 | Internal Server Error | all |
| | | |
| 501 | Not Implemented | all |
| | | |
| 502 | Bad Gateway | all |
| | | |
| 503 | Service Unavailable | all |
| | | |
| 504 | Gateway Timeout | all |
| | | |
| 505 | RTSP Version Not Supported | all |
| | | |
| 551 | Option not support | all |
+------+----------------------------------------+-----------------+
Table 4: Status codes and their usage with RTSP methods
8.2. Response Header Fields
The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in
the Status-Line. These header fields give information about the
server and about further access to the resource identified by the
Request-URI. All headers currently classified as response headers
are listed in Table 5.
Schulzrinne, et al. Expires August 28, 2008 [Page 40]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
+------------------------+--------------------+
| Header | Defined in Section |
+------------------------+--------------------+
| Accept-Credentials | Section 16.2 |
| | |
| Accept-Ranges | Section 16.5 |
| | |
| Connection-Credentials | Section 16.12 |
| | |
| ETag | Section 16.21 |
| | |
| Location | Section 16.28 |
| | |
| Proxy-Authenticate | Section 16.30 |
| | |
| Public | Section 16.34 |
| | |
| Range | Section 16.35 |
| | |
| Retry-After | Section 16.37 |
| | |
| RTP-Info | Section 16.39 |
| | |
| Scale | Section 16.40 |
| | |
| Session | Section 16.43 |
| | |
| Server | Section 16.42 |
| | |
| Speed | Section 16.41 |
| | |
| Transport | Section 16.46 |
| | |
| Unsupported | Section 16.47 |
| | |
| Vary | Section 16.49 |
| | |
| WWW-Authenticate | Section 16.51 |
+------------------------+--------------------+
Table 5: The RTSP response headers
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However the usage
of feature-tags in the request allows the responding party to learn
the capability of the receiver of the response. New or experimental
header fields MAY be given the semantics of response-header fields if
all parties in the communication recognize them to be response-header
Schulzrinne, et al. Expires August 28, 2008 [Page 41]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
fields. Unrecognized header fields in responses are treated as
entity-header fields.
Schulzrinne, et al. Expires August 28, 2008 [Page 42]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
9. Entity
Request and Response messages MAY transfer an entity if not otherwise
restricted by the request method or response status code. An entity
consists of entity-header fields and an entity-body, although some
responses will only include the entity-headers.
The SETPARAMETER and GET_PARAMETER request and response, and DESCRIBE
response MAY have an entity. All 4xx and 5xx responses MAY also have
an entity.
In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the entity.
9.1. Entity Header Fields
Entity-header fields define meta-information about the entity-body
or, if no body is present, about the resource identified by the
request. The entity header fields are listed in Table 6.
+------------------+--------------------+
| Header | Defined in Section |
+------------------+--------------------+
| Allow | Section 16.6 |
| | |
| Content-Base | Section 16.13 |
| | |
| Content-Encoding | Section 16.14 |
| | |
| Content-Language | Section 16.15 |
| | |
| Content-Length | Section 16.16 |
| | |
| Content-Location | Section 16.17 |
| | |
| Content-Type | Section 16.18 |
| | |
| Expires | Section 16.22 |
| | |
| Last-Modified | Section 16.27 |
+------------------+--------------------+
Table 6: The RTSP entity headers
The extension-header mechanism allows additional entity-header fields
to be defined without changing the protocol, but these fields cannot
be assumed to be recognizable by the recipient. Unrecognized header
fields SHOULD be ignored by the recipient and forwarded by proxies.
Schulzrinne, et al. Expires August 28, 2008 [Page 43]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
9.2. Entity Body
See [H7.2] with the addition that an RTSP message with an entity body
MUST include the Content-Type and Content-Length headers.
Schulzrinne, et al. Expires August 28, 2008 [Page 44]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
10. Connections
RTSP requests can be transmitted using the two different connection
scenarios listed below:
o persistent - a transport connection is used for several request/
response transactions;
o transient - a transport connection is used for a single request/
response transaction.
RFC 2326 attempted to specify an optional mechanism for transmitting
RTSP messages in connectionless mode over a transport protocol such
as UDP. However, it was not specified in sufficient detail to allow
for interoperable implementations. In an attempt to reduce
complexity and scope, and due to lack of interest, RTSP 2.0 does not
attempt to define a mechanism for supporting RTSP over UDP or other
connectionless transport protocols. A side-effect of this is that
RTSP requests SHALL NOT be sent to multicast groups since no
connection can be established with a specific receiver in multicast
environments.
Certain RTSP headers, such as the CSeq header (Section 16.19), which
may appear to be relevant only to connectionless transport scenarios
are still retained and must be implemented according to the
specification. In the case of CSeq, it is quite useful for matching
responses to requests if the requests are pipelined (see Section 12).
It is also useful in proxies for keeping track of the different
requests when aggregating several client requests on a single TCP
connection.
10.1. Reliability and Acknowledgements
When RTSP messages are transmitted using reliable transport
protocols, they MUST NOT be retransmitted at the RTSP protocol level.
Instead, the implementation must rely on the underlying transport to
provide reliability. The RTSP implementation may use any indication
of reception acknowledgement of the message from the underlying
transport protocols to optimize the RTSP behavior.
If both the underlying reliable transport such as TCP and the RTSP
application retransmit requests, each packet loss or message loss
may result in two retransmissions. The receiver typically cannot
take advantage of the application-layer retransmission since the
transport stack will not deliver the application-layer
retransmission before the first attempt has reached the receiver.
If the packet loss is caused by congestion, multiple
retransmissions at different layers will exacerbate the
Schulzrinne, et al. Expires August 28, 2008 [Page 45]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
congestion.
Lack of acknowledgement of an RTSP request should be handled within
the constraints of the connection timeout considerations described
below (Section 10.4).
10.2. Using Connections
A TCP transport can be used for both persistent connections (for
several message exchanges) and transient connections (for a single
message exchange). Implementations of this specification MUST
support RTSP over TCP. The scheme of the RTSP URI (Section 4.2)
indicates the default port that the server will listen on.
A server MUST handle both persistent and transient connections.
Transient connections facilitate mechanisms for fault tolerance.
They also allow for application layer mobility. A server and
client pair that support transient connections can survive the
loss of a TCP connection; e.g., due to a NAT timeout. When the
client has discovered that the TCP connection has been lost, it
can set up a new one when there is need to communicate again.
A persistent connection MAY be used for all transactions between the
server and client, including messages for multiple RTSP sessions.
However a persistent connection MAY also be closed after a few
message exchanges. For example, a client may use a persistent
connection for the initial SETUP and PLAY message exchanges in a
session and then close the connection. Later, when the client wishes
to send a new request, such as a PAUSE for the session, a new
connection would be opened. This connection may either be transient
or persistent.
An RTSP agent SHOULD NOT have more than one connection to the server
at any given point. If a client or proxy handles multiple RTSP
sessions on the same server, it SHOULD use only one connection for
managing those sessions.
This saves connection resources on the server. It also reduces
complexity by and enabling the server to maintain less state about
its sessions and connections.
Unlike HTTP, RTSP allows a server to send requests to a client.
However, this can be supported only if a client establishes a
persistent connection with the server. In cases where a persistent
connection does not exist between a server and its client, due to the
lack of a signalling channel the server may be forced to drop an RTSP
session without notifying the client. An example of such a case is
Schulzrinne, et al. Expires August 28, 2008 [Page 46]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
when the server desires to send a REDIRECT request for an RTSP
session to the client but is not able to do so because it cannot
reach the client.
Without a persistent connection between the client and the server,
the media server has no reliable way of reaching the client.
Also, this is the only way that requests from a server to its
client are likely to traverse firewalls.
In light of the above, it is RECOMMENDED that clients use persistent
connections whenever possible. A client that supports persistent
connections MAY "pipeline" its requests (see Section 12).
10.3. Closing Connections
The client MAY close a connection at any point when no outstanding
request/response transactions exist for any RTSP session being
managed through the connection. The server, however, SHOULD NOT
close a connection until all RTSP sessions being managed through the
connection have been timed out (Section 16.43). A server SHOULD NOT
close a connection immediately after responding to a session-level
TEARDOWN request for the last RTSP session being controlled through
the connection. Instead, it should wait for a reasonable amount of
time for the client to receive the TEARDOWN response, take
appropriate action, and initiate the connection closing. The server
SHOULD wait at least 10 seconds after sending the TEARDOWN response
before closing the connection.
This is to ensure that the client has time to issue a SETUP for a
new session on the existing connection after having torn the last
one down. 10 seconds should give the client ample opportunity get
its message to the server.
A server SHOULD NOT close the connection directly as a result of
responding to a request with an error code.
Certain error responses such as "460 Only Aggregate Operation
Allowed" (Section 15.4.12) are used for negotiating capabilities
of a server with respect to content or other factors. In such
cases, it is inefficient for the server to close a connection on
an error response. Also, such behavior would prevent
implementation of advanced/special types of requests or result in
extra overhead for the client when testing for new features. On
the flip side, keeping connections open after sending an error
response poses a Denial of Service security risk (Section 21).
If a server closes a connection while the client is attempting to
send a new request, the client will have to close its current
Schulzrinne, et al. Expires August 28, 2008 [Page 47]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
connection, establish a new connection and send its request over the
new connection.
An RTSP message should not be terminated by closing the connection.
Such a message MAY be considered to be incomplete by the receiver and
discarded. An RTSP message is properly terminated as defined in
Section 5.
10.4. Timing Out Connections and RTSP Messages
Receivers of a request (responder) SHOULD respond to requests in a
timely manner even when a reliable transport such as TCP is used.
Similarly, the sender of a request (requestor) SHOULD wait for a
sufficient time for a response before concluding that the responder
will not be acting upon its request.
A responder SHOULD respond to all requests within 5 seconds. If the
responder recognizes that processing of a request will take longer
than 5 seconds, it SHOULD send a 100 (Continue) response as soon as
possible. It SHOULD continue sending a 100 response every 5 seconds
thereafter until it is ready to send the final response to the
requestor. After sending a 100 response, the receiver MUST send a
final response indicating the success or failure of the request.
A requestor SHOULD wait at least 10 seconds for a response before
concluding that the responder will not be responding to its request.
After receiving a 100 response, the requestor SHOULD continue waiting
for further responses. If more than 10 seconds elapses without
receiving any response, the requestor MAY assume that the responder
is unresponsive and abort the connection.
A requestor SHOULD wait longer than 10 seconds for a response if it
is experiencing significant transport delays on its connection to the
responder. The requestor is capable of determining the RTT of the
request/response cycle using the Timestamp header (Section 16.45) in
any RTSP request.
10.5. Showing Liveness
The mechanisms for showing liveness of the client is, any RTSP
request with a Session header, if RTP & RTCP is used an RTCP message,
or through any other used media protocol capable of indicating
liveness of the RTSP client. It is RECOMMENDED that a client does
not wait to the last second of the timeout before trying to send a
liveness message. The RTSP message may be lost or when using
reliable protocols, such as TCP, the message may take some time to
arrive safely at the receiver. To show liveness between RTSP request
issued to accomplish other things, the following mechanisms can be
Schulzrinne, et al. Expires August 28, 2008 [Page 48]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
used, in descending order of preference:
RTCP: If RTP is used for media transport RTCP SHOULD be used. If
RTCP is used to report transport statistics, it SHALL also work
as keep alive. The server can determine the client by used
network address and port together with the fact that the client
is reporting on the servers SSRC(s). A downside of using RTCP
is that it only gives statistical guarantees to reach the
server. However that probability is so low that it can be
ignored in most cases. For example, a session with 60 seconds
timeout and enough bitrate assigned to RTCP messages to send a
message from client to server on average every 5 seconds. That
client have for a network with 5 \% packet loss, the
probability to fail showing liveness sign in that session
within the timeout interval of 2.4*E-16. In sessions with
shorter timeout times, or much higher packet loss, or small
RTCP bandwidths SHOULD also use any of the mechanisms below.
SETPARAMETER: When using SETPARAMETER for keep alive, no body SHOULD
be included. This method is the RECOMMENDED RTSP method to use
in request only intended to perform keep-alive.
OPTIONS: This method does also work. However it causes the server
to perform more unnecessary processing and result in bigger
responses than necessary for the task. The reason for this is
that the server needs to determine what capabilities that are
associated with the media resource to correctly populate the
Public and Allow headers.
The timeout parameter MAY be included in a SETUP response, and SHALL
NOT be included in requests. The server uses it to indicate to the
client how long the server is prepared to wait between RTSP commands
or other signs of life before closing the session due to lack of
activity (see below and Appendix B). The timeout is measured in
seconds, with a default of 60 seconds. The length of the session
timeout SHALL NOT be changed in a established session.
10.6. Use of IPv6
Explicit IPv6 support was not present in RTSP 1.0 (RFC 2326). RTSP
2.0 has been updated for explicit IPv6 support. Implementations of
RTSP 2.0 MUST understand literal IPv6 addresses in URIs and headers.
Schulzrinne, et al. Expires August 28, 2008 [Page 49]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
11. Capability Handling
This section describes the available capability handling mechanism
which allows RTSP to be extended. Extensions to this version of the
protocol are basically done in two ways. First, new headers can be
added. Secondly, new methods can be added. The capability handling
mechanism is designed to handle both cases.
When a method is added, the involved parties can use the OPTIONS
method to discover wether it is supported. This is done by issuing a
OPTIONS request to the other party. Depending on the URI it will
either apply in regards to a certain media resource, the whole server
in general, or simply the next hop. The OPTIONS response MUST
contain a Public header which declares all methods supported for the
indicated resource.
It is not necessary to use OPTIONS to discover support of a method,
the client could simply try the method. If the receiver of the
request does not support the method it will respond with an error
code indicating the the method is either not implemented (501) or
does not apply for the resource (405). The choice between the two
discovery methods depends on the requirements of the service.
Feature-Tags are defined to handle functionality additions that are
not new methods. Each feature-tag represents a certain block of
functionality. The amount of functionality that a feature-tag
represents can vary significantly. A feature-tag can for example
represent the functionality a single RTSP header provides. Another
feature-tag can represent much more functionality, such as the
"play.basic" feature-tag which represents the minimal playback
implementation.
Feature-tags are used to determine wether the client, server or proxy
supports the functionality that is necessary to achieve the desired
service. To determine support of a feature-tag, several different
headers can be used, each explained below:
Supported: The supported header is used to determine the complete
set of functionality that both client and server have. The
intended usage is to determine before one needs to use a
functionality that it is supported. It can be used in any
method, however OPTIONS is the most suitable one as it at the
same time determines all methods that are implemented. When
sending a request the requestor declares all its capabilities
by including all supported feature-tags. This results in that
the receiver learns the requestors feature support. The
receiver then includes its set of features in the response.
Schulzrinne, et al. Expires August 28, 2008 [Page 50]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Proxy-Supported: The Proxy-Supported header is used similar to the
Supported header, but instead of giving the supported
functionality of the client or server it provides both the
requestor and the responder a view of what functionality the
proxy chain between the two supports. Proxies are required to
add this header whenever the Supported header is present, but
proxies may independently of the requestor add it.
Require: The Require header can be included in any request where the
end-point, i.e. the client or server, is required to understand
the feature to correctly perform the request. This can, for
example, be a SETUP request where the server is required to
understand a certain parameter to be able to set up the media
delivery correctly. Ignoring this parameter would not have the
desired effect and is not acceptable. Therefore the end-point
receiving a request containing a Require MUST negatively
acknowledge any feature that it does not understand and not
perform the request. The response in cases where features are
not supported are 551 (Option Not Supported). Also the
features that are not supported are given in the Unsupported
header in the response.
Proxy-Require: This method has the same purpose and workings as
Require except that it only applies to proxies and not the end-
point. Features that needs to be supported by both proxies and
end-point needs to be included in both the Require and Proxy-
Require header.
Unsupported: This header is used in a 551 error response, to
indicate which feature(s) that was not supported. Such a
response is only the result of the usage of the Require and/or
Proxy-Require header where one or more feature where not
supported. This information allows the requestor to make the
best of situations as it knows which features are not
supported.
Schulzrinne, et al. Expires August 28, 2008 [Page 51]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
12. Pipelining Support
Pipelining is a general method to improve performance of request
response protocols by allowing the requesting entity to have more
than one request outstanding and send them over the same persistent
connection. For RTSP where the relative order of requests will
matter it is important to maintain the order of the requests.
Because of this the the responding entity SHALL process the incoming
requests in their sending order. The sending order can be determined
by the CSeq header and its sequence number. For TCP the delivery
order will be the same as the sending order. The processing of the
request SHALL also have been finished before processing the next
request from the same entity. The responses MUST be sent in the
order the requests was processed.
RTSP 2.0 has extended support for pipelining compared to RTSP 1.0.
The major improvement is to allow all requests to setup and initiate
media playback to be pipelined after each other. This is
accomplished by the utilization of the Pipelined-Requests header (see
Section 16.29). This header allows a client to request that two or
more requests is to be processed in the same RTSP session context
which the first request creates. In other words a client can request
that two or more media streams are set-up and then played without
needing to wait for a single response. This speeds up the initial
startup time for an RTSP session with at least one RTT.
If a pipelined request builds on the succesful completion of one or
more prior requests the requestor must verify that all requests were
executed as expected. A common example will be two SETUP requests
and a PLAY request. In case one of the SETUP fails unexpectedly, the
PLAY request can still be succesfully executed. However, not as
expected by the requesting client as only a single media instead of
two will be played. In this case the client can send a PAUSE
request, correct the failing SETUP request and then request it to be
played.
Schulzrinne, et al. Expires August 28, 2008 [Page 52]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
13. Method Definitions
The method indicates what is to be performed on the resource
identified by the Request-URI. The method name is case-sensitive.
New methods may be defined in the future. Method names SHALL NOT
start with a $ character (decimal 24) and MUST be a token as defined
by the ABNF [RFC4234] 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 |
| | | | | |
| REDIRECT | S -> C | P,S | optional | required |
| | | | | |
| SETUP | C -> S | S | required | required |
| | | | | |
| SET_PARAMETER | C -> S | P,S | required | optional |
| | | | | |
| | S -> C | | | |
| | | | | |
| TEARDOWN | C -> S | P,S | required | required |
+---------------+-----------+--------+--------------+---------------+
Table 7: Overview of RTSP methods, their direction, and what objects
(P: presentation, S: stream) they operate on. Legend: R=Respond,
Sd=Send, Opt: Optional, Req: Required
Note on Table 7: GET_PARAMETER is recommended, but not required.
For example, a fully functional server can be built to deliver
media without any parameters. SET_PARAMETER is required however
due to its usage for keep-alive. PAUSE is now required due to
that it is the only way of getting out of the state machines play
state without terminating the whole session.
Schulzrinne, et al. Expires August 28, 2008 [Page 53]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
13.1. OPTIONS
The semantics of the RTSP OPTIONS method is equivalent to that of the
HTTP OPTIONS method described in [H9.2]. In RTSP however, OPTIONS is
bi-directional, in that a client can request it to a server and vice
versa. A client MUST implement the capability to send an OPTIONS
request and a server or a proxy MUST implement the capability to
respond to an OPTIONS request. The client, server or proxy MAY also
implement the converse of their required capability.
An OPTIONS request may be issued at any time. Such a request does
not modify the session state. However, it may prolong the session
lifespan (see below). The URI in an OPTIONS request determines the
scope of the request and the corresponding response. If the Request-
URI refers to a specific media resource on a given host, the scope is
limited to the set of methods supported for that media resource by
the indicated RTSP agent. A Request-URI with only the host address
limits the scope to the specified RTSP agent's general capabilities
without regard to any specific media. If the Request-URI is an
asterisk ("*"), the scope is limited to the general capabilities of
the next hop (i.e. the RTSP agent in direct communication with the
request sender).
Regardless of scope of the request, the Public header MUST always be
included in the OPTIONS response listing the methods that are
supported by the responding RTSP agent. In addition, if the scope of
the request is limited to a media resource, the Allow header MUST be
included in the response to enumerate the set of methods that are
allowed for that resource unless the set of methods completely
matches the set in the Public header. If the given resource is not
available, the RTSP agent SHOULD return an appropriate response code
such as 3rr or 4xx. The Supported header MAY be included in the
request to query the set of features that are supported by the
responding RTSP agent.
The OPTIONS method can be used to keep an RTSP session alive.
However, it is not the preferred means of session keep-alive
signalling, see Section 16.43. 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:
Schulzrinne, et al. Expires August 28, 2008 [Page 54]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C->S: OPTIONS * RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Require:
Proxy-Require: gzipped-messages
Supported: play.basic
S->C: RTSP/2.0 200 OK
CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Supported: play.basic, implicit-play, gzipped-messages
Server: PhonyServer/1.1
Note that some of the feature-tags in Require and Proxy-Require are
fictional features.
13.2. DESCRIBE
The DESCRIBE method is used to retrieve the description of a
presentation or media object from a server. The Request-URI of the
DESCRIBE request identifies the media resource of interest. The
client MAY include the Accept header in the request to list the
description formats that it understands. The server SHALL respond
with a description of the requested resource and return the
description in the entity of the response. The DESCRIBE reply-
response pair constitutes the media initialization phase of RTSP.
Example:
Schulzrinne, et al. Expires August 28, 2008 [Page 55]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.1
Content-Type: application/sdp
Content-Length: 367
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 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
m=application 32416 UDP WB
a=orient:portrait
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.)
o via some form of a user interface
Schulzrinne, et al. Expires August 28, 2008 [Page 56]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 three
different cases; Create an RTSP session, add a media to a session,
and change the transport parameters of already set up media stream.
Using SETUP to add media to an existing session, when the session is
in PLAY state, is unspecified. Otherwise SETUP can be used in all
three states; INIT, and READY, for both purposes and in PLAY to
change the transport parameters.
The Transport header, see Section 16.46, specifies the transport
parameters acceptable to the client for data transmission; the
response will contain the transport parameters selected by the
server. This allows the client to enumerate in descending order of
preference the transport mechanisms and parameters acceptable to it,
while the server can select the most appropriate. It is expected
that the session description format used will enable the client to
select a limited number possible configurations that are offered to
the server to choose from. All transport related parameters shall be
included in the Transport header, the use of other headers for this
purpose is discouraged due to middleboxes, such as firewalls or NATs.
For the benefit of any intervening firewalls, a client SHALL indicate
the known transport parameters, even if it has no influence over
these parameters, for example, where the server advertises a fixed
multicast address as destination.
Since SETUP includes all transport initialization information,
firewalls and other intermediate network devices (which need this
information) are spared the more arduous task of parsing the
DESCRIBE response, which has been reserved for media
initialization.
The client SHALL include the Accept-Ranges header in the request
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
Schulzrinne, et al. Expires August 28, 2008 [Page 57]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
request. If the client does not support a time format necessary for
the presentation the server SHALL respond using 456 (Header Field Not
Valid for Resource) and include the Accept-Ranges header with the
range unit formats supported for the resource.
In a SETUP response the server SHALL include the Accept-Ranges header
(see Section 16.5) to indicate which time formats that are acceptable
to use for this media resource.
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: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.1
Session: 47112344;timeout=60
Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589";
src_addr="192.0.2.241:6256"/"192.0.2.241:6257";
ssrc=2A3F93ED
Accept-Ranges: NPT
In the above example the client wants to create an RTSP session
containing the media resource "rtsp://example.com/foo/bar/baz.rm".
The transport parameters acceptable to the client is either RTP/AVP/
UDP (UDP per default) to be received on client port 4588 and 4589 or
RTP/AVP interleaved on the RTSP control channel. The server selects
the RTP/AVP/UDP transport and adds the ports it will send and
received RTP and RTCP from, and the RTP SSRC that will be used by the
server.
The server MUST generate a session identifier in response to a
successful SETUP request, unless a SETUP request to a server includes
a session identifier, in which case the server MUST bundle this setup
request into the existing session (aggregated session) or return
error 459 (Aggregate Operation Not Allowed) (see Section 15.4.11).
An Aggregate control URI MUST be used to control an aggregated
session. This URI MUST be different from the stream control URIs of
the individual media streams included in the aggregate. The
Aggregate control URI is to be specified by the session description
if the server supports aggregated control and aggregated control is
desired for the session. However even if aggregated control is
offered the client MAY chose to not set up the session in aggregated
control. If an Aggregate control URI is not specified in the session
Schulzrinne, et al. Expires August 28, 2008 [Page 58]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
description, it is normally an indication that non-aggregated control
should be used. The SETUP of media streams in an aggregate which has
not been given an aggregated control URI is unspecified.
While the session ID sometimes has enough information for
aggregate control of a session, the Aggregate control URI is still
important for some methods such as SETPARAMETER 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 16.43. 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.
o If RTP is used as a transport for the underlying media streams, an
RTCP sender or receiver report from the client(s) for any of the
media streams in that RTSP session. RTCP Sender Reports may for
example be received in sessions where the server is invited into a
conference session and is as valid for keep-alive.
If a SETUP request on a session fails for any reason, the session
state, as well as transport and other parameters for associated
streams SHALL remain unchanged from their values as if the SETUP
request had never been received by the server.
13.3.1. Changing Transport Parameters
A client MAY issue a SETUP request for a stream that is already set
up or playing in the session to change transport parameters, which a
server MAY allow. If it does not allow changing of parameters, it
MUST respond with error 455 (Method Not Valid In This State).
Reasons to support changing transport parameters, is to allow for
application layer mobility and flexibility to utilize the best
available transport as it becomes available. If a client receives a
455 when trying to change transport parameters while the server is in
play state, it MAY try to put the server in ready state using PAUSE,
before issuing the SETUP request again. If also that fails the
Schulzrinne, et al. Expires August 28, 2008 [Page 59]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
changing of transport parameters will require that the client
performs a TEARDOWN of the affected media and then setting it up
again. In aggregated session avoiding tearing down all the media at
the same time will avoid the creation of a new session.
All transport parameters MAY be changed. However the primary usage
expected is to either change transport protocol completely, like
switching from Interleaved TCP mode to UDP or vise versa or change
delivery address.
In a SETUP response for a request to change the transport parameters
while in Play state, the server SHALL include the Range to indicate
from what point the new transport parameters are used. Further, if
RTP is used for delivery, the server SHALL also include the RTP-Info
header to indicate from what timestamp and RTP sequence number the
change has taken place. If both RTP-Info and Range is included in
the response the "rtp_time" parameter and range MUST be for the
corresponding time, i.e. be used in the same way as for PLAY to
ensure the correct synchronization information is available.
If the transport parameters change while in PLAY state results in a
change of synchronization related information, for example changing
RTP SSRC, the server MUST provide in the SETUP response the necessary
synchronization information. However the server is RECOMMENDED to
avoid changing the synchronization information if possible.
13.4. PLAY
The PLAY method tells the server to start sending data via the
mechanism specified in SETUP. 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.
For aggregated sessions where the initial SETUP request (creating a
session) is followed by one or more additional SETUP request, a PLAY
request MAY be pipelined after those additional SETUP requests
without awaiting their responses. This procedure can reduce the
delay from start of session establishment until media play-out has
started with one round trip time. However an client needs to be
aware that using this procedure will result in the playout of the
server state established at the time of processing the PLAY, i.e.,
after the processing of all the requests prior to the PLAY request in
the pipeline. This may not be the intended one due to failure of any
of the prior requests. However a client easily determine this based
on the responses from those requests. In case of failure the client
can halt the media playout using PAUSE and try to establish the
intended state again before issuing another PLAY request.
Schulzrinne, et al. Expires August 28, 2008 [Page 60]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
In an aggregated session the PLAY request MUST contain an aggregated
control URI. A server SHALL responde with error 460 (Only Aggregate
Operation Allowed) if the client PLAY Request-URI is for one of the
media. The media in an aggregate SHALL be played in sync. If a
client want individual control of the media it needs to use separate
RTSP sessions for each media.
Upon receipt of the PLAY request, the server SHALL position the
normal play time to the beginning of the range specified in the
received Range header and delivers stream data until the end of the
range if given, or until a new PLAY request is received, else to the
end of the media is reached. To allow for precise composition
multiple ranges MAY be specified in one PLAY Request. The range
values are valid if all given ranges are part of any media within the
aggregate. If a given range value points outside of the media, the
response SHALL be the 457 (Invalid Range) error code.
The below example will first play seconds 10 through 15, then,
immediately following, seconds 20 to 25, and finally seconds 30
through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0
CSeq: 835
Session: 12345678
Range: npt=10-15, npt=20-25, npt=30-
User-Agent: PhonyClient/1.2
See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It SHALL start
playing a stream from the beginning (npt=0-) unless the stream has
been paused or is currently playing. If a stream has been paused via
PAUSE, stream delivery resumes at the pause point. If a stream is
currently playing, the new PLAY begins at the current stream
position. The stream SHALL play until the end of the media.
The Range header MUST NOT contain a time parameter. The usage of
time in PLAY method has been deprecated. If a request with time
parameter is received the server SHOULD respond with a 457 (Invalid
Range) to indicate that the time parameter is not supported.
Server MUST include a "Range" header in any PLAY response. The
response MUST use the same format as the request's range header
contained. If no Range header was in the request, the NPT time
format SHOULD be used unless the client showed support for an other
format more appropriate. Also for a session with live media streams
the Range header MUST indicate a valid time. It is RECOMMENDED that
normal play time is used, either the "now" indicator, for example
Schulzrinne, et al. Expires August 28, 2008 [Page 61]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
"npt=now-", or the time since session start as an open interval, e.g.
"npt=96.23-". An absolute time value (clock) for the corresponding
time MAY be given, i.e. "clock=20030213T143205Z-". The UTC clock
format SHOULD only be used if client has shown support for it.
For an on-demand stream, the server MUST reply with the actual range
that will be played back, i.e. for which duration any media (having
content at this time) is delivered. This may differ from the
requested range if alignment of the requested range to valid frame
boundaries is required for the media source. Note that some media
streams in an aggregate may need to be delivered from even earlier
points. Also, some media format have a very long duration per
individual data unit, therefore it might be necessary for the client
to parse the data unit, and select where to start.
Example: Single audio stream (MIDI)
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: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: npt=3.52-
RTP-Info:url="rtsp://example.com/audio"
ssrc=0D12F123:seq=14783;rtptime=2345962545
S->C: RTP Packet TS=2345962545 => NPT=3.52
Duration=4.15 seconds
In this example the client receives the first media packet that
stretches all the way up and past the requested playtime. Thus, it
is the client's decision if to render to the user the time between
3.52 and 7.05, or to skip it. In most cases it is probably most
suitable to not render that time period.
For live media sources it might be impossible to specify from which
point in time all media streams carrying active content can actually
be delivered. Therefore a server MAY specify a start time (or now-)
in the range header, for which not all media will be available from.
If no range is specified in the request, the start position SHALL
still be returned in the reply. If the medias that are part of an
aggregate has different lengths, the PLAY request SHALL be performed
as long as the given range is valid for any media, for example the
longest media. Media will be sent whenever it is available for the
Schulzrinne, et al. Expires August 28, 2008 [Page 62]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
given play-out point.
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 16.39.
After playing the desired range, the presentation does NOT transition
to the READY state, media delivery simply stops. A PAUSE request
MUST be issued before the stream enters the READY state. A PLAY
request while the stream is still in the PLAYING state is legal, and
can be issued without an intervening PAUSE request. Such a request
SHALL replace the current PLAY action with the new one requested,
i.e. being handle the same as the request was received in ready
state. In the case the first time range in Range header has a open
start time (-endtime), the server SHALL continue to play from where
it currently was until the specified end point. This is useful to
change ongoing playback to play another sequence, or end at another
point than in the previous request.
A client desiring to play the media from the beginning MUST send a
PLAY request with a Range header pointing at the beginning, e.g.
npt=0-. If a PLAY request is received without a Range header when
media delivery has stopped at the end, the server SHOULD respond with
a 457 "Invalid Range" error response. In that response the current
pause point in a Range header SHALL be included.
The following example plays the whole presentation starting at SMPTE
time code 0:10:20 until the end of the clip. Note: The RTP-Info
headers has been broken into several lines to fit the page.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/2.0
CSeq: 833
Session: 12345678
Range: smpte=0:10:20-
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 833
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: smpte=0:10:22-0:15:45
RTP-Info:url="rtsp://example.com/twister.en"
ssrc=0D12F123:seq=14783;rtptime=2345962545
For playing back a recording of a live presentation, it may be
desirable to use clock units:
Schulzrinne, et al. Expires August 28, 2008 [Page 63]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 23 Jan 1997 15:35:06 GMT
Server:PhonyServer 1.0
Range: clock=19961108T142300Z-19961108T143520Z
RTP-Info:url="rtsp://example.com/meeting.en"
ssrc=0D12F123:seq=53745;rtptime=484589019
All range specifiers in this specification allow for ranges with
unspecified begin times (e.g. "npt=-30"). When used in a PLAY
request, the server treats this as a request to start/resume playback
from the current pause point, ending at the end time specified in the
Range header. If the pause point is located later than the given end
value, a 457 (Invalid Range) response SHALL be given.
The possibility to replace a current PLAY request with a new one
replaces two RTSP 1.0 functions:
o The queued play functionality described in RFC 2326 [RFC2326] is
removed and multiple ranges can be used to achieve a similar
functionality.
o The use of PLAY for keep-alive signaling, i.e. PLAY request
without a range header in PLAY state, has also been deprecated.
Instead a client can use, SETPARAMETER (recommended) or OPTIONS
(allowed) for keep alive.
An example of using PLAY request to change the behavior, if a server
has received requests to play ranges 10 to 15 and then 13 to 20 (that
is, overlapping ranges), a PLAY request 4 seconds after the first
would take effect while the server plays the first range. Thus
changing the behavior to continue to play to 25 seconds, i.e. the
played range equal play with range: npt=10-25. If the second PLAY
request would arrive as the second range in the first request was
playing, then the equivalent request would be play with range:npt=10-
15,npt=13-25.
Schulzrinne, et al. Expires August 28, 2008 [Page 64]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: 12345678
Range: npt=10-15, npt=13-20
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: npt=10-15, npt=13-20
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792482193
Session: 12345678
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 835
Session: 12345678
Range: npt=-25
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:09 GMT
Server: PhonyServer 1.0
Range: npt=14-25
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934239921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792842193
Session: 12345678
13.5. PAUSE
The PAUSE request causes the stream delivery to immediately be
interrupted (halted). A PAUSE request MUST be done either with the
aggregated control URI for aggregated sessions, resulting in all
media being halted, or the media URI for non-aggregated sessions.
Any attempt to do muting of a single media with an PAUSE request in
an aggregated session SHALL be responded with error 460 (Only
Aggregate Operation Allowed). After resuming playback,
synchronization of the tracks MUST be maintained. Any server
resources are kept, though servers MAY close the session and free
resources after being paused for the duration specified with the
timeout parameter of the Session header in the SETUP message.
Schulzrinne, et al. Expires August 28, 2008 [Page 65]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Date: 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 SHALL resume from the pause point and play until media
end.
The pause point after any PAUSE request SHALL be returned to the
client by adding a Range header with what remains unplayed of the
PLAY request's ranges, i.e. including all the remaining ranges part
of multiple range specification. If one desires to resume playing a
ranged request, one simply includes the Range header from the PAUSE
response.
Schulzrinne, et al. Expires August 28, 2008 [Page 66]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: npt=10-30
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=4FAD8726:seq=57654;rtptime=2792482193
Session: 12345678
After 11 seconds, i.e. at 21 seconds into the presentation:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 835
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:09 GMT
Server: PhonyServer 1.0
Range: npt=21-30
Session: 12345678
If a client issues a PAUSE request and the server acknowledges and
enters the READY state, the proper server response, if the player
issues another PAUSE, is still 200 OK. The 200 OK response MUST
include the Range header with the current pause point. See examples
below:
Schulzrinne, et al. Expires August 28, 2008 [Page 67]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 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
13.6. TEARDOWN
The TEARDOWN client to server request stops the stream delivery for
the given URI, freeing the resources associated with it. A TEARDOWN
request MAY be performed on either an aggregated or a media control
URI. However some restrictions apply depending on the current state.
The TEARDOWN request SHALL contain a Session header indicating what
session the request applies to.
A TEARDOWN using the aggregated control URI or the media URI in a
session under non-aggregated control (single media session) MAY be
done in any state (Ready, and Play). A successful request SHALL
result in that media delivery is immediately halted and the session
state is destroyed. This SHALL be indicated through the lack of a
Session header in the response.
A TEARDOWN using a media URI in an aggregated session MAY only be
done in Ready state. Such a request only removes the indicated media
stream and associated resources from the session. This may result in
that a session returns to non-aggregated control, due to that it only
contains a single media after the requests completion. A session
that will exist after the processing of the TEARDOWN request SHALL in
the response to that TEARDOWN request contain a Session header. Thus
the presence of the Session header indicates to the receiver of the
response if the session is still existing or has been removed.
Schulzrinne, et al. Expires August 28, 2008 [Page 68]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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. GET_PARAMETER
The GET_PARAMETER request retrieves the value of a 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. The content of the
reply and response is left to the implementation.
The method MAY also be used without a body (entity). If the request
is successful, i.e., a 200 OK response is received, then the keep-
alive timer has been updated. Any non-required header present in
such a request may or may not been processed. To allow a client to
determine if any such header has been processed, it is necessary to
use a feature-tag and the Require header. Due to this reason it is
RECOMMENDED that any parameters to be retrieved are sent in the body,
rather than using any header.
Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 431
Content-Type: text/parameters
Session: 12345678
Content-Length: 26
User-Agent: PhonyClient/1.2
packets_received
jitter
C->S: RTSP/2.0 200 OK
CSeq: 431
Content-Length: 38
Content-Type: text/parameters
packets_received: 10
jitter: 0.3838
Schulzrinne, et al. Expires August 28, 2008 [Page 69]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
The "text/parameters" section is only an example type for a body
carrying parameters.
13.8. SET_PARAMETER
This method requests to set the value of a parameter or a set of
parameters for a presentation or stream specified by the URI. The
method MAY also be used without a body (entity). It is the
RECOMMENDED method to use in request sent for the sole purpose of
updating the keep-alive timer. If this request is successful, i.e. a
200 OK response is received, then the keep-alive timer has been
updated. Any non-required header present in such a request may or
may not been processed. To allow a client to determine if any such
header has been processed, it is necessary to use a feature tag and
the Require header. Due to this reason it is RECOMMENDED that any
parameters are sent in the body, rather than using any header.
A request is RECOMMENDED to only contain a single parameter to allow
the client to determine why a particular request failed. If the
request contains several parameters, the server MUST only act on the
request if all of the parameters can be set successfully. A server
MUST allow a parameter to be set repeatedly to the same value, but it
MAY disallow changing parameter values. If the receiver of the
request does not understand or cannot locate a parameter, error 451
(Parameter Not Understood) SHALL be used. In the case a parameter is
not allowed to change, the error code is 458 (Parameter Is Read-
Only). The response body SHOULD contain only the parameters that
have errors. Otherwise no body SHALL be returned.
Note: transport parameters for the media stream MUST only be set with
the SETUP command.
Restricting setting transport parameters to SETUP is for the
benefit of firewalls.
The parameters are split in a fine-grained fashion so that there
can be more meaningful error indications. However, it may make
sense to allow the setting of several parameters if an atomic
setting is desirable. Imagine device control where the client
does not want the camera to pan unless it can also tilt to the
right angle at the same time.
Example:
Schulzrinne, et al. Expires August 28, 2008 [Page 70]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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
The "text/parameters" section is only an example type for
parameters. This method is intentionally loosely defined with the
intention that the reply content and response content will be
defined by the one desiring to use this mechanism.
13.9. REDIRECT
The REDIRECT method is issued by a server to inform a client that it
required to connect to another server location to access the resource
indicated by the Request-URI. The presence of the Session header in
a REDIRECT request indicates the scope of the request, and determines
the specific semantics of the request.
A REDIRECT request with a Session header has end-to-end (i.e. server
to client) scope and applies only to the given session. Any
intervening proxies SHOULD NOT disconnect the control channel while
there are other remaining end-to-end sessions. The OPTIONAL Location
header, if included in such a request, SHALL contain a complete
absolute URI pointing to the resource to which the client SHOULD
reconnect. Specifically, the Location SHALL NOT contain just the
host and port. A client may receive a REDIRECT request with a
Session header, if and only if, an end-to-end session has been
established.
A client may receive a REDIRECT request without a Session header at
any time when it has communication or a connection established with a
server. The scope of such a request is limited to the next-hop (i.e.
the RTSP agent in direct communication with the server) and applies,
as well, to the control connection between the next-hop RTSP agent
and the server. A REDIRECT request without a Session header
indicates that all sessions and pending requests being managed via
the control connection MUST be redirected. The OPTIONAL Location
header, if included in such a request, SHOULD contain an absolute URI
Schulzrinne, et al. Expires August 28, 2008 [Page 71]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
with only the host address and the OPTIONAL port number of the server
to which the RTSP agent SHOULD reconnect. Any intervening proxies
SHOULD do all of the following in the order listed:
1. respond to the REDIRECT request
2. disconnect the control channel from the requesting server
3. connect to the server at the given host address
4. pass the REDIRECT request to each applicable client (typically
those clients with an active session or an unanswered request)
Note: The proxy is responsible for accepting REDIRECT responses
from its clients; these responses MUST NOT be passed on to either
the original server or the redirected server.
The lack of a Location header in any REDIRECT request is indicative
of the server no longer being able to fulfill the current request and
having no alternatives for the client to continue with its normal
operation. It is akin to a server initiated TEARDOWN that applies
both to sessions as well as the general connection associated with
that client.
When the Range header is not included in a REDIRECT request, the
client SHOULD perform the redirection immediately and return a
response to the server. The server can consider the session as
terminated and can free any associated state after it receives the
successful (2xx) response. The server MAY close the signalling
connection upon receiving the response and the client SHOULD close
the signalling connection after sending the 2xx response. The
exception to this is when the client has several sessions on the
server being managed by the given signalling connection. In this
case, the client SHOULD close the connection when it has received and
responded to REDIRECT requests for all the sessions managed by the
signalling connection.
If the OPTIONAL Range header is included in a REDIRECT request, it
indicates when the redirection takes effect. The range value MUST be
an open ended single value, e.g. npt=59-, indicating the play out
time when redirection SHALL occur. Alternatively, a range with a
time= parameter indicates the wall clock time by when the redirection
MUST take place. When the time= parameter is present in the range,
any range value MUST be ignored even though it MUST be syntactically
correct. To allow a client to determine that redirect time without
being time synchronized with the server, the server SHALL include a
Date header in the request. When the indicated redirect point is
reached, a client MUST issue a TEARDOWN request and SHOULD close the
Schulzrinne, et al. Expires August 28, 2008 [Page 72]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
signalling connection after receiving a 2xx response. The normal
connection considerations apply for the server.
The differentiation of REDIRECT requests with and without range
headers is to allow for clear and explicit state handling. As the
state in the server needs to be kept until the point of
redirection, the handling becomes more clear if the client is
required to TEARDOWN the session at the redirect point.
If the REDIRECT request times out following the rules in Section 10.4
the server MAY terminate the session or transport connection that
would be redirected by the request. This is a safeguard against
misbehaving clients that refuses to respond to a REDIRECT request.
That should not provide any benefit.
After a REDIRECT request has been processed, a client that wants to
continue to send or receive media for the resource identified by the
Request-URI will have to establish a new session with the designated
host. If the URI given in the Location header is a valid resource
URI, a client SHOULD issue a DESCRIBE request for the URI.
Note: The media resource indicated by the Location header can be
identical, slightly different or totally different. This is the
reason why a new DESCRIBE request SHOULD be issued.
If the Location header contains only a host address, the client MAY
assume that the media on the new server is identical to the media on
the old server, i.e. all media configuration information from the old
session is still valid except for the host address. However the
usage of conditional SETUP using ETag identifiers are RECOMMENDED to
verify the assumption.
This example request redirects traffic for this session to the new
server at the given absolute time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 732
Location: rtsp://s2.example.com:8001
Range: npt=0- ;time=19960213T143205Z
Session: uZ3ci0K+Ld-M
C->S: RTSP/2.0 200 OK
CSeq: 732
User-Agent: PhonyClient/1.2
Schulzrinne, et al. Expires August 28, 2008 [Page 73]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
14. Embedded (Interleaved) Binary Data
In order to fulfill certain requirements on the network side, e.g. in
conjunction with network address translators that block RTP traffic
over UDP, it may be necessary to interleave RTSP messages and media
stream data. This interleaving should generally be avoided unless
necessary since it complicates client and server operation and
imposes additional overhead. Also head of line blocking may cause
problems. Interleaved binary data SHOULD only be used if RTSP is
carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte channel identifier,
followed by the length of the encapsulated binary data as a binary,
two-byte integer in network byte order. The stream data follows
immediately afterwards, without a CRLF, but including the upper-layer
protocol headers. Each $ block SHALL contain exactly one upper-layer
protocol data unit, e.g., one RTP packet.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| "$" = 24 | Channel ID | Length in bytes |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Length number of bytes of binary data :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The channel identifier is defined in the Transport header with the
interleaved parameter (Section 16.46).
When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. The usage of RTCP messages is
indicated by including a range containing a second channel in the
interleaved parameter of the Transport header, see Section 16.46. If
RTCP is used, packets SHALL be sent on the first available channel
higher than the RTP channel. The channels are bi-directional and
therefore RTCP traffic are sent on the second channel in both
directions.
RTCP is sometime needed for synchronization when two or more
streams are interleaved in such a fashion. Also, this provides a
convenient way to tunnel RTP/RTCP packets through the TCP control
connection when required by the network configuration and transfer
them onto UDP when possible.
Schulzrinne, et al. Expires August 28, 2008 [Page 74]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;unicast;interleaved=5-6
Session: 12345678
Accept-Ranges: NPT
C->S: PLAY rtsp://example.com/bar.file RTSP/2.0
CSeq: 3
Session: 12345678
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 3
Session: 12345678
Date: 05 Jun 1997 18:59:15 GMT
RTP-Info: url="rtsp://example.com/bar.file"
ssrc=0D12F123:seq=232433;rtptime=972948234
Range: npt=0-56.8
S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
S->C: $006{2 byte length}{"length" bytes RTCP packet}
Schulzrinne, et al. Expires August 28, 2008 [Page 75]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
15. Status Code Definitions
Where applicable, HTTP status [H10] codes are reused. Status codes
that have the same meaning are not repeated here. See Table 4 for a
listing of which status codes may be returned by which requests. All
error messages, 4xx and 5xx MAY return a body containing further
information about the error.
15.1. Success 1xx
15.1.1. 100 Continue
See, [H10.1.1].
15.2. Success 2xx
15.2.1. 200 OK
See, [H10.2.1].
15.3. Redirection 3xx
The notation "3rr" indicates response codes from 300 to 399 inclusive
which are meant for redirection. The response code 304 is excluded
from this set, as it is not used for redirection.
See [H10.3] for definition of status code 300 to 305. However
comments are given for some to how they apply to RTSP.
Within RTSP, redirection may be used for load balancing or
redirecting stream requests to a server topologically closer to the
client. Mechanisms to determine topological proximity are beyond the
scope of this specification.
A 3rr code MAY be used to respond to any request. It is RECOMMENDED
that they are used if necessary before a session is established, i.e.
in response to DESCRIBE or SETUP. However in cases where a server is
not able to send a REDIRECT request to the client, the server MAY
need to resort to using 3rr responses to inform a client with a
established session about the need for redirecting the session. If
an 3rr response is received for an request in relation to a
established session, the client SHOULD send a TEARDOWN request for
the session, and MAY reestablish the session using the resource
indicated by the Location.
If the the Location header is used in a response it SHALL contain an
absolute URI pointing out the media resource the client is redirected
to, the URI SHALL NOT only contain the host name.
Schulzrinne, et al. Expires August 28, 2008 [Page 76]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
15.3.1. 300 Multiple Choices
See [H10.3.1].
15.3.2. 301 Moved Permanently
The request resource are moved permanently and resides now at the URI
given by the location header. The user client SHOULD redirect
automatically to the given URI. This response MUST NOT contain a
message-body. The Location header MUST be included in the response.
15.3.3. 302 Found
The requested resource resides temporarily at the URI given by the
Location header. The Location header MUST be included in the
response. This response is intended to be used for many types of
temporary redirects; e.g., load balancing. It is RECOMMENDED that
the server set the reason phrase to something more meaningful than
"Found" in these cases. The user client SHOULD redirect
automatically to the given URI. This response MUST NOT contain a
message-body.
This example shows a client being redirected to a different server:
C->S: SETUP rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 2
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: NPT, SMPTE, UTC
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 302 Try Other Server
CSeq: 2
Location: rtsp://s2.example.com:8001/fizzle/foo
15.3.4. 303 See Other
This status code SHALL NOT be used in RTSP. However, it was allowed
to use in RTSP 1.0 (RFC 2326).
15.3.5. 304 Not Modified
If the client has performed a conditional DESCRIBE or SETUP (see
Section 16.25) and the requested resource has not been modified, the
server SHOULD send a 304 response. This response MUST NOT contain a
message-body.
The response MUST include the following header fields:
Schulzrinne, et al. Expires August 28, 2008 [Page 77]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
o Date
o ETag and/or Content-Location, if the header(s) would have been
sent in a 200 response to the same request.
o Expires, Cache-Control, and/or Vary, if the field-value might
differ from that sent in any previous response for the same
variant.
This response is independent for the DESCRIBE and SETUP requests.
That is, a 304 response to DESCRIBE does NOT imply that the resource
content is unchanged (only the session description) and a 304
response to SETUP does NOT imply that the resource description is
unchanged. The ETag and If-Match headers may be used to link the
DESCRIBE and SETUP in this manner.
15.3.6. 305 Use Proxy
See [H10.3.6].
15.4. Client Error 4xx
15.4.1. 400 Bad Request
The request could not be understood by the server due to malformed
syntax. The client SHOULD NOT repeat the request without
modifications [H10.4.1]. If the request does not have a CSeq header,
the server MUST NOT include a CSeq in the response.
15.4.2. 405 Method Not Allowed
The method specified in the request is not allowed for the resource
identified by the Request-URI. The response MUST include an Allow
header containing a list of valid methods for the requested resource.
This status code is also to be used if a request attempts to use a
method not indicated during SETUP.
15.4.3. 451 Parameter Not Understood
The recipient of the request does not support one or more parameters
contained in the request. When returning this error message the
sender SHOULD return a entity body containing the offending
parameter(s).
15.4.4. 452 reserved
This error code was removed from RFC 2326 [RFC2326] and is obsolete.
Schulzrinne, et al. Expires August 28, 2008 [Page 78]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
15.4.5. 453 Not Enough Bandwidth
The request was refused because there was insufficient bandwidth.
This may, for example, be the result of a resource reservation
failure.
15.4.6. 454 Session Not Found
The RTSP session identifier in the Session header is missing,
invalid, or has timed out.
15.4.7. 455 Method Not Valid in This State
The client or server cannot process this request in its current
state. The response SHALL contain an Allow header to make error
recovery possible.
15.4.8. 456 Header Field Not Valid for Resource
The server could not act on a required request header. For example,
if PLAY contains the Range header field but the stream does not allow
seeking. This error message may also be used for specifying when the
time format in Range is impossible for the resource. In that case
the Accept-Ranges header SHALL be returned to inform the client of
which format(s) that are allowed.
15.4.9. 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the
presentation.
15.4.10. 458 Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can be read but not
modified. When returning this error message the sender SHOULD return
a entity body containing the offending parameter(s).
15.4.11. 459 Aggregate Operation Not Allowed
The requested method may not be applied on the URI in question since
it is an aggregate (presentation) URI. The method may be applied on
a media URI.
15.4.12. 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URI in question since
it is not an aggregate control (presentation) URI. The method may be
applied on the aggregate control URI.
Schulzrinne, et al. Expires August 28, 2008 [Page 79]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
15.4.13. 461 Unsupported Transport
The Transport field did not contain a supported transport
specification.
15.4.14. 462 Destination Unreachable
The data transmission channel could not be established because the
client address could not be reached. This error will most likely be
the result of a client attempt to place an invalid dest_addr
parameter in the Transport field.
15.4.15. 463 Destination Prohibited
The data transmission channel was not established because the server
prohibited access to the client address. This error is most likely
the result of a client attempt to redirect media traffic to another
destination with a dest_addr parameter in the Transport header.
15.4.16. 464 Data Transport Not Ready Yet
The data transmission channel to the media destination is not yet
ready for carrying data. However the responding entity still expects
that the data transmission channel will be established at this point
in time. Note however that this may result in a permanent failure
like 462 "Destination Unreachable".
An example when this error may occur is in the case a client sends a
PLAY request to a server prior to ensuring that the TCP connections
negotiated for carrying media data was successful established (In
violation of this specification). The server would use this error
code to indicate that the requested action could not be performed due
to the failure of completing the connection establishment.
15.4.17. 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.
15.4.18. 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.
Schulzrinne, et al. Expires August 28, 2008 [Page 80]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
15.4.19. 472 Failure to establish secure connection
A proxy fails to establish a secure connection to the next hop RTSP
agent. This is primarily caused by a fatal failure at the TLS
handshake, for example due to server not accepting any cipher suits.
15.5. Server Error 5xx
15.5.1. 551 Option not supported
A feature-tag given in the Require or the Proxy-Require fields was
not supported. The Unsupported header SHALL be returned stating the
feature for which there is no support.
Schulzrinne, et al. Expires August 28, 2008 [Page 81]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 | |
| | | | | |
| REDIRECT | S -> C | P,S | RDR | |
| | | | | |
| SETUP | C -> S | S | STP | |
| | | | | |
| SETPARAMETER | C -> S, S -> C | P,S | SPR | R,r |
| | | | | |
| TEARDOWN | C -> S | P,S | TRD | |
+---------------+----------------+--------+---------+------+
Table 8: Overview of RTSP methods, their direction, and what objects
(P: presentation, S: stream) they operate on. Body notes if a method
is allowed to carry body and in which direction, R = Request,
r=response. Note: It is allowed for all error messages 4xx and 5xx to
have a body
The general syntax for header fields is covered in Section 5.2. This
section lists the full set of header fields along with notes on
meaning, and usage. The syntax definition for header fields are
present in Section 20.2.3. Throughout this section, we use [HX.Y] to
refer to Section X.Y of the current HTTP/1.1 specification RFC 2616
[RFC2616]. Examples of each header field are given.
Information about header fields in relation to methods and proxy
processing is summarized in Table 9, Table 10, Table 11, and
Table 12.
The "where" column describes the request and response types in which
the header field can be used. Values in this column are:
Schulzrinne, et al. Expires August 28, 2008 [Page 82]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
R: header field may only appear in requests;
r: header field may only appear in responses;
2xx, 4xx, etc.: A numerical value or range indicates response codes
with which the header field can be used;
c: header field is copied from the request to the response.
An empty entry in the "where" column indicates that the header field
may be present in both requests and responses.
The "proxy" column describes the operations a proxy may perform on a
header field. An empty proxy column indicates that the proxy SHALL
NOT do any changes to that header, all allowed operations are
explicitly stated:
a: A proxy can add or concatenate the header field if not present.
m: A proxy can modify an existing header field value.
d: A proxy can delete a header field value.
r: A proxy needs to be able to read the header field, and thus
this header field cannot be encrypted.
The rest of the columns relate to the presence of a header field in a
method. The method names when abbreviated, are according to Table 8:
c: Conditional; requirements on the header field depend on the
context of the message.
m: The header field is mandatory.
m*: The header field SHOULD be sent, but clients/servers need to be
prepared to receive messages without that header field.
o: The header field is optional.
*: The header field is SHALL be present if the message body is not
empty. See Section 16.16, Section 16.18 and Section 5.3 for
details.
-: The header field is not applicable.
"Optional" means that a Client/Server MAY include the header field in
a request or response. The Client/Server behavior when receiving
such headers varies, for some it may ignore the header field, in
Schulzrinne, et al. Expires August 28, 2008 [Page 83]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
other case it is request to process the header. This is regulated by
the method and header descriptions. Example of such headers that
require processing are the Require and Proxy-Require header fields
discussed in Section 16.38 and Section 16.32. A "mandatory" header
field MUST be present in a request, and MUST be understood by the
Client/Server receiving the request. A mandatory response header
field MUST be present in the response, and the header field MUST be
understood by the Client/Server processing the response. "Not
applicable" means that the header field MUST NOT be present in a
request. If one is placed in a request by mistake, it MUST be
ignored by the Client/Server receiving the request. Similarly, a
header field labeled "not applicable" for a response means that the
Client/Server MUST NOT place the header field in the response, and
the Client/Server MUST ignore the header field in the response.
An RTSP agent SHALL ignore extension headers that are not understood.
The From and Location header fields contain an URI. If the URI
contains a comma, or semicolon, the URI MUST be enclosed in double
quotas ("). Any URI parameters are contained within these quotas.
If the URI is not enclosed in double quotas, any semicolon- delimited
parameters are header-parameters, not URI parameters.
+----------------+------+-----+-----+-----+------+-----+------+-----+
| Header | Wher | Pro | DES | OPT | SETU | PLA | PAUS | TRD |
| | e | xy | | | P | Y | E | |
+----------------+------+-----+-----+-----+------+-----+------+-----+
| Accept | R | | o | - | - | - | - | - |
| | | | | | | | | |
| Accept-Credent | R | r | o | o | o | o | o | o |
| ials | | | | | | | | |
| | | | | | | | | |
| Accept-Encodin | R | r | o | - | - | - | - | - |
| g | | | | | | | | |
| | | | | | | | | |
| Accept-Languag | R | r | o | - | - | - | - | - |
| e | | | | | | | | |
| | | | | | | | | |
| Accept-Ranges | R | r | - | - | m | - | - | - |
| | | | | | | | | |
| Accept-Ranges | r | r | - | - | o | - | - | - |
| | | | | | | | | |
| Accept-Ranges | 456 | r | - | - | - | o | - | - |
| | | | | | | | | |
| Allow | r | am | c | c | c | - | - | - |
| | | | | | | | | |
| Allow | 405 | am | m | m | m | m | m | m |
| | | | | | | | | |
Schulzrinne, et al. Expires August 28, 2008 [Page 84]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| 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 | | * | - | - | - | - | - |
Schulzrinne, et al. Expires August 28, 2008 [Page 85]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| Content-Type | 4xx, | | * | * | * | * | * | * |
| | 5xx | | | | | | | |
| | | | | | | | | |
| CSeq | Rc | rm | m | m | m | m | m | m |
| | | | | | | | | |
| Date | | am | o | o | o | o | o | o |
| | | | | | | | | |
| ETag | r | r | o | - | o | - | - | - |
| | | | | | | | | |
| Expires | r | r | o | - | - | - | - | - |
| | | | | | | | | |
| From | R | r | o | o | o | o | o | o |
| | | | | | | | | |
| If-Match | R | r | - | - | o | - | - | - |
| | | | | | | | | |
| If-Modified-Si | R | r | o | - | o | - | - | - |
| nce | | | | | | | | |
| | | | | | | | | |
| If-None-Match | R | r | o | - | - | - | - | - |
| | | | | | | | | |
| Last-Modified | r | r | o | - | - | - | - | - |
| | | | | | | | | |
| Location | 3rr | | o | o | o | o | o | o |
+----------------+------+-----+-----+-----+------+-----+------+-----+
Table 9: Overview of RTSP header fields (A-L) related to methods
DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.
+------------+-------+------+----+-----+-------+------+-------+-----+
| Header | Where | Prox | DE | OPT | SETUP | PLAY | PAUSE | TRD |
| | | y | S | | | | | |
+------------+-------+------+----+-----+-------+------+-------+-----+
| Pipelined- | | amdr | - | o | o | o | o | o |
| Requests | | | | | | | | |
| | | | | | | | | |
| Proxy- | 407 | amr | m | m | m | m | m | m |
| Authentica | | | | | | | | |
| te | | | | | | | | |
| | | | | | | | | |
| Proxy- | R | rd | o | o | o | o | o | o |
| Authorizat | | | | | | | | |
| ion | | | | | | | | |
| | | | | | | | | |
| Proxy- | R | ar | o | o | o | o | o | o |
| Require | | | | | | | | |
| | | | | | | | | |
| Proxy- | r | r | c | c | c | c | c | c |
| Require | | | | | | | | |
Schulzrinne, et al. Expires August 28, 2008 [Page 86]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| Proxy- | R | amr | c | c | c | c | c | c |
| Supported | | | | | | | | |
| | | | | | | | | |
| Proxy- | r | | c | c | c | c | c | c |
| Supported | | | | | | | | |
| | | | | | | | | |
| Public | r | admr | - | m | - | - | - | - |
| | | | | | | | | |
| Public | 501 | admr | m | m | m | m | m | m |
| | | | | | | | | |
| Range | R | | - | - | - | o | - | - |
| | | | | | | | | |
| Range | r | | - | - | c | m | m | - |
| | | | | | | | | |
| Referer | R | | o | o | o | o | o | o |
| | | | | | | | | |
| Require | R | | o | o | o | o | o | o |
| | | | | | | | | |
| Retry-Afte | 3rr,5 | | o | o | o | - | - | - |
| r | 03 | | | | | | | |
| | | | | | | | | |
| RTP-Info | r | | - | - | c | c | - | - |
| | | | | | | | | |
| Scale | | | - | - | - | o | - | - |
| | | | | | | | | |
| Session | R | r | - | o | o | m | m | m |
| | | | | | | | | |
| Session | r | r | - | c | m | m | m | o |
| | | | | | | | | |
| Server | R | r | - | o | - | - | - | - |
| | | | | | | | | |
| Server | r | r | o | o | o | o | o | o |
| | | | | | | | | |
| Speed | | | - | - | - | o | - | - |
| | | | | | | | | |
| Supported | R | amr | o | o | o | o | o | o |
| | | | | | | | | |
| Supported | r | amr | c | c | c | c | c | c |
| | | | | | | | | |
| Timestamp | R | admr | o | o | o | o | o | o |
| | | | | | | | | |
| Timestamp | c | admr | m | m | m | m | m | m |
| | | | | | | | | |
| Transport | | amr | - | - | m | - | - | - |
| | | | | | | | | |
| Unsupporte | r | | c | c | c | c | c | c |
| d | | | | | | | | |
| | | | | | | | | |
Schulzrinne, et al. Expires August 28, 2008 [Page 87]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| User-Agent | R | | m* | m* | m* | m* | m* | m* |
| | | | | | | | | |
| Vary | r | | c | c | c | c | c | c |
| | | | | | | | | |
| Via | R | amr | o | o | o | o | o | o |
| | | | | | | | | |
| Via | c | dr | m | m | m | m | m | m |
| | | | | | | | | |
| WWW- | 401 | | m | m | m | m | m | m |
| Authentica | | | | | | | | |
| te | | | | | | | | |
+------------+-------+------+----+-----+-------+------+-------+-----+
Table 10: Overview of RTSP header fields (P-W) related to methods
DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.
+------------------------+---------+-------+-----+-----+-----+
| Header | Where | Proxy | GPR | SPR | RDR |
+------------------------+---------+-------+-----+-----+-----+
| Accept-Credentials | R | r | o | o | o |
| | | | | | |
| Allow | 405 | amr | m | m | m |
| | | | | | |
| Authorization | R | | o | o | o |
| | | | | | |
| Bandwidth | R | | - | o | - |
| | | | | | |
| Blocksize | R | | - | o | - |
| | | | | | |
| Connection | | | o | o | o |
| | | | | | |
| Connection-Credentials | 470,407 | ar | o | o | o |
| | | | | | |
| Content-Base | R | | o | o | - |
| | | | | | |
| 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 | - |
Schulzrinne, et al. Expires August 28, 2008 [Page 88]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| 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 |
| | | | | | |
| Date | R | a | o | o | m |
| | | | | | |
| Date | r | am | o | o | o |
| | | | | | |
| From | R | r | o | o | o |
| | | | | | |
| Last-Modified | R | r | - | - | - |
| | | | | | |
| Last-Modified | r | r | o | - | - |
| | | | | | |
| Location | 3rr | | o | o | o |
| | | | | | |
| Location | 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 |
| | | | | | |
Schulzrinne, et al. Expires August 28, 2008 [Page 89]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| Proxy-Supported | r | | c | c | c |
| | | | | | |
| Public | 501 | admr | m | m | m |
+------------------------+---------+-------+-----+-----+-----+
Table 11: Overview of RTSP header fields (A-P) related to methods
GETPARAMETER, SETPARAMETER, and REDIRECT.
+------------------+---------+-------+-----+-----+-----+
| Header | Where | Proxy | GPR | SPR | RDR |
+------------------+---------+-------+-----+-----+-----+
| Range | R | | - | - | o |
| | | | | | |
| Referer | R | | o | o | o |
| | | | | | |
| Require | R | r | o | o | o |
| | | | | | |
| Retry-After | 3rr,503 | | o | o | - |
| | | | | | |
| Scale | | | - | - | - |
| | | | | | |
| Session | R | r | o | o | o |
| | | | | | |
| Session | r | r | c | c | o |
| | | | | | |
| 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 |
Schulzrinne, et al. Expires August 28, 2008 [Page 90]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
| WWW-Authenticate | 401 | | m | m | m |
+------------------+---------+-------+-----+-----+-----+
Table 12: Overview of RTSP header fields (R-W) related to methods
GETPARAMETER, SETPARAMETER, and REDIRECT.
16.1. Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
See [H14.1] for syntax.
Example of use:
Accept: application/example q=1.0, application/sdp
16.2. Accept-Credentials
The Accept-Credentials header is a request header used to indicate to
any trusted intermediary how to handle further secured connections to
proxies or servers. See Section 19 for the usage of this header. It
SHALL NOT be included in server to client requests.
In a request the header SHALL contain the method (User, Proxy, or
Any) for approving credentials selected by the requestor. The method
SHALL NOT be changed by any proxy, unless it is "proxy" when a proxy
MAY change it to "user" to take the role of user approving each
further hop. If the method is "User" the header contains zero or
more of credentials that the client accepts. The header may contain
zero credentials in the first RTSP request to a RTSP server when
using the "User" method. This as the client has not yet received any
credentials to accept. Each credential SHALL consist of one URI
identifying the proxy or server, the hash algorithm identifier, and
the hash over that entity's DER encoded certificate [RFC3280] in
Base64 [RFC4648]. All RTSP clients and proxies SHALL implement the
SHA-256[FIPS-pub-180-2] algorithm for computation of the hash of the
DER encoded certificate. The SHA-256 algorithm is identified by the
token "sha-256".
The intention with allowing for other hash algorithms is to enable
the future retirement of algorithms that are not implemented
somewhere else than here. Thus the definition of future algorithms
for this purpose is intended to be extremely limited. A feature tag
can be used to ensure that support for the replacement algorithm
exist.
Example:
Schulzrinne, et al. Expires August 28, 2008 [Page 91]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Accept-Credentials:User
"rtsps://proxy2.example.com/";sha-256;exaIl9VMbQMOFGClx5rXnPJKVNI=,
"rtsps://server.example.com/";sha-256;lurbjj5khhB0NhIuOXtt4bBRH1M=
16.3. Accept-Encoding
See [H14.3].
16.4. Accept-Language
See [H14.4]. Note that the language specified applies to the
presentation description and any reason phrases, not the media
content.
16.5. Accept-Ranges
The Accept-Ranges request and response-header field allows indication
of the format supported in the Range header. The client SHALL
include the header in SETUP requests to indicate which formats it
support to receive in PLAY and PAUSE responses, and REDIRECT
requests. The server SHALL include the header in SETUP and 456 error
responses to indicate the formats supported for the resource
indicated by the request URI.
Accept-Ranges: NPT, SMPTE
This header has the same syntax as [H14.5] and the syntax is defined
in Section 20.2.3. However, new range-units are defined.
16.6. Allow
The Allow entity-header field lists the methods supported by the
resource identified by the Request-URI. The purpose of this field is
to strictly inform the recipient of valid methods associated with the
resource. An Allow header field MUST be present in a 405 (Method Not
Allowed) response. See [H14.7] for syntax definition. The Allow
header MUST also be present in all OPTIONS responses where the
content of the header will not include exactly the same methods as
listed in the Public header.
The Allow SHALL also be included in SETUP and DESCRIBE responses, if
the methods allowed for the resource is different than the minimal
implementation set.
Example of use:
Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE
Schulzrinne, et al. Expires August 28, 2008 [Page 92]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
16.7. Authorization
See [H14.8].
16.8. Bandwidth
The Bandwidth request-header field describes the estimated bandwidth
available to the client, expressed as a positive integer and measured
in bits per second. The bandwidth available to the client may change
during an RTSP session, e.g., due to mobility, congestion, etc.
Example:
Bandwidth: 62360
16.9. Blocksize
The Blocksize request-header field is sent from the client to the
media server asking the server for a particular media packet size.
This packet size does not include lower-layer headers such as IP,
UDP, or RTP. The server is free to use a blocksize which is lower
than the one requested. The server MAY truncate this packet size to
the closest multiple of the minimum, media-specific block size, or
override it with the media-specific size if necessary. The block
size MUST be a positive decimal number, measured in octets. The
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 93]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 94]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 95]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
16.11. Connection
See [H14.10]. The use of the connection option "close" in RTSP
messages SHOULD be limited to error messages when the server is
unable to recover and therefore see it necessary to close the
connection. The reason is that the client has the choice of
continuing using a connection indefinitely, as long as it sends valid
messages.
16.12. Connection-Credentials
The Connection-Credentials response header is used to carry the chain
of credentials of any next hop that need to be approved by the
requestor. It SHALL only be used in server to client responses.
The Connection-Credentials header in an RTSP response SHALL, if
included, contain the credential information (in form of a list of
certificates providing the chain of certification) of the next hop
that an intermediary needs to securely connect to. The header MUST
include the URI of the next hop (proxy or server) and a base64
[RFC4648] encoded binary structure containg a sequence of DER encoded
X.509v3 certificates[RFC3280] .
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
from the chain, under the assumption that the remote end must already
possess it in order to validate it in any case.
Example:
Schulzrinne, et al. Expires August 28, 2008 [Page 96]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC...
Where MIIDNTCC... is a BASE64 encoding of the following structure:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number of certifcates | Size of certificate #1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Size of certificate #2 | Size of certificate #3 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #1 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #2 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #3 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
16.13. Content-Base
The Content-Base entity-header field may be used to specify the base
URI for resolving relative URIs within the entity.
Content-Base: rtsp://media.example.com/movie/twister
If no Content-Base field is present, the base URI of an entity is
defined either by its Content-Location (if that Content-Location URI
is an absolute URI) or the URI used to initiate the request, in that
order of precedence. Note, however, that the base URI of the
contents within the entity-body may be redefined within that entity-
body.
16.14. Content-Encoding
See [H14.11].
16.15. Content-Language
See [H14.12].
16.16. Content-Length
The Content-Length general-header field contains the length of the
body (entity) of the message (i.e. after the double CRLF following
the last header). Unlike HTTP, it MUST be included in all messages
that carry body beyond the header portion of the message. If it is
missing, a default value of zero is assumed. It is interpreted
according to [H14.13].
Schulzrinne, et al. Expires August 28, 2008 [Page 97]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
16.17. Content-Location
See [H14.14].
16.18. Content-Type
See [H14.17]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and
parameter-value types.
16.19. CSeq
The CSeq general-header field specifies the sequence number for an
RTSP request-response pair. This field MUST be present in all
requests and responses. For every RTSP request containing the given
sequence number, the corresponding response will have the same
number. Any retransmitted request MUST contain the same sequence
number as the original (i.e. the sequence number is not incremented
for retransmissions of the same request). For each new RTSP request
the CSeq value SHALL be incremented by one. The initial sequence
number MAY be any number, however it is RECOMMENDED to start at 0.
Each sequence number series is unique between each requester and
responder, i.e. the client has one series for its request to a server
and the server has another when sending request to the client. Each
requester and responder is identified with its network address.
Proxies that aggregate several sessions on the same transport will
regularly need to renumber the CSeq header field in requests and
responses to fulfill the rules for the header.
Example:
CSeq: 239
16.20. Date
See [H14.18]. An RTSP message containing a body MUST include a Date
header if the sending host has a clock. Servers SHOULD include a
Date header in all other RTSP messages.
16.21. ETag
The ETag response header MAY be included in DESCRIBE or SETUP
responses. The entity tags (Section 4.8) returned in a DESCRIBE
response, and the one in SETUP refers to the presentation, i.e. both
the returned session description and the media stream. This allows
for verification that one has the right session description to a
media resource at the time of the SETUP request. However it has the
disadvantage that a change in any of the parts results in
Schulzrinne, et al. Expires August 28, 2008 [Page 98]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
invalidation of all the parts.
If the ETag is provided both inside the entity, e.g. within the
"a=etag" attribute in SDP, and in the response message, then both
tags SHALL be identical. It is RECOMMENDED that the ETag is
primarily given in the RTSP response message, to ensure that caches
can use the ETag without requiring content inspection. However for
session descriptions that are distributed outside of RTSP, for
example using HTTP, etc. it will be necessary to include the entity
tag in the session description as specified in Appendix D.1.9.
SETUP and DESCRIBE requests can be made conditional upon the ETag
using the headers If-Match (Section 16.24) and If-None-Match (
Section 16.26).
16.22. Expires
The Expires entity-header field gives a date and time after which the
description or media-stream should be considered stale. The
interpretation depends on the method:
DESCRIBE response: The Expires header indicates a date and time
after which the presentation description (body) SHOULD be
considered stale.
SETUP response: The Expires header indicate a date and time after
which the media stream SHOULD be considered stale.
A stale cache entry may not normally be returned by a cache (either a
proxy cache or an user agent cache) unless it is first validated with
the origin server (or with an intermediate cache that has a fresh
copy of the entity). See Section 18 for further discussion of the
expiration model.
The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.
The format is an absolute date and time as defined by HTTP-date in
[H3.3]; it MUST be in RFC1123-date format:
An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/2.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as having occurred in the past
(i.e., already expired).
Schulzrinne, et al. Expires August 28, 2008 [Page 99]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
The presence of an Expires header field with a date value of some
time in the future on a media stream that otherwise would by default
be non-cacheable indicates that the media stream is cacheable, unless
indicated otherwise by a Cache-Control header field (Section 16.10).
16.23. From
See [H14.22].
16.24. If-Match
See [H14.24].
The If-Match request-header field is especially useful for ensuring
the integrity of the presentation description, in both the case where
it is fetched via means external to RTSP (such as HTTP), or in the
case where the server implementation is guaranteeing the integrity of
the description between the time of the DESCRIBE message and the
SETUP message. By including the ETag given in or with the session
description in a SETUP request, the client ensures that resources set
up are matching the description. A SETUP request for which the ETag
validation check fails, SHALL responde using 412 (Precondition
Failed).
This validation check is also very useful if a session has been
redirected from one server to another.
16.25. If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE
and SETUP methods to make them conditional. If the requested variant
has not been modified since the time specified in this field, a
description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (Not Modified)
response SHALL be returned without any message-body.
An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
Schulzrinne, et al. Expires August 28, 2008 [Page 100]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
16.26. If-None-Match
See [H14.26].
This request header can be used with one or several entity tags to
make DESCRIBE requests conditional. A new session description is
retrieved only if another entity than the ones already available
would be included. If the entity available for delivery is matching
the one the client already has, then a 304 (Not Modified) response is
given.
16.27. Last-Modified
The Last-Modified entity-header field indicates the date and time at
which the origin server believes the presentation description or
media stream was last modified. See [H14.29]. For the methods
DESCRIBE, the header field indicates the last modification date and
time of the description, for SETUP that of the media stream.
16.28. Location
See [H14.30].
16.29. Pipelined-Requests
The Pipelined-Requests general header is used to indicate that a
request is to be executed in the context created by previous
requests. The primary usage of this header is to allow pipelining of
SETUP requests so that any additional SETUP request after the first
one doesn't need to wait for the session ID to be sent back to the
requesting entity. The header contains a unique identifier that is
scoped by the persistent connection used to send the requests.
Upon receiving a request with the Pipelined-Requests the responding
entity SHALL look up if there exist a binding between this Pipelined-
Requests identifier for the current persistent connection and an RTSP
session ID. If that exist then the received request is processed the
same way as if it did contain the Session header with the looked up
session ID. If there doesn't exist a mapping and no Session header
is included in the request, the responding entity SHALL create a
binding upon the succesful completion of a session creating request,
i.e. SETUP. If the request failed to create an RTSP session no
binding SHALL be created. In case the request contains both a
Session header and the Pipelined-Requests header the Pipelined-
Requests SHALL be ignored.
Note: Based on the above definition at least the first request
containing a new unique Pipelined-Requests will be required to be a
Schulzrinne, et al. Expires August 28, 2008 [Page 101]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
SETUP request (unless the protocol is extended with new methods of
creating a session). After that first one, additional SETUP requests
or request of any type using the RTSP session context may include the
Pipelined-Requests header.
For all responses to request that contained the Pipelined-Requests,
the Session header and the Pipelined-Requests SHALL both be included,
assuming that it is allowed for that response and that the binding
between the header values exist. Pipelined-Requests SHOULD NOT be
used in requests after that the client has received the RTSP Session
ID. This as using the real session ID allows the request to be used
also in cases the persistent connection has been terminated and a new
connection is needed.
It is the sender of the request that is responsible for using a
previously unused identifier within this transport connection scope
when a new RTSP session is to be cretated with this method. A server
side binding SHALL be deleted upon the termination of the related
RTSP session. Note: Although this definition would allow for reusing
previously used pipelining identifiers, this is NOT RECOMMENDED to
allow for better error handling and logging.
RTSP Proxies may need to translate Pipelined-Requests identifier
values from incoming request to outgoing to allow for aggregation of
requests onto a persistent connection.
16.30. Proxy-Authenticate
See [H14.33].
16.31. Proxy-Authorization
See [H14.34].
16.32. Proxy-Require
The Proxy-Require request-header field is used to indicate proxy-
sensitive features that MUST be supported by the proxy. Any Proxy-
Require header features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client using the
Unsupported header. The proxy SHALL use the 551 (Option Not
Supported) status code in the response. Any feature-tag included in
the Proxy-Require does not apply to the end-point (server or client).
To ensure that a feature is supported by both proxies and servers the
tag needs to be included in also a Require header.
See Section 16.38 for more details on the mechanics of this message
and a usage example.
Schulzrinne, et al. Expires August 28, 2008 [Page 102]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Example of use:
Proxy-Require: play.basic
16.33. Proxy-Supported
The Proxy-Supported header field enumerates all the extensions
supported by the proxy using feature-tags. The header carries the
intersection of extensions supported by the forwarding proxies. The
Proxy-Supported header MAY be included in any request by a proxy. It
SHALL be added by any proxy if the Supported header is present in a
request. When present in a request, the receiver MUST in the
response copy the received Proxy-Supported header.
The Proxy-Supported header field contains a list of feature-tags
applicable to proxies, as described in Section 4.7. The list are the
intersection of all feature-tags understood by the proxies. To
achieve an intersection, the proxy adding the Proxy-Supported header
includes all proxy feature-tags it understands. Any proxy receiving
a request with the header, checks the list and removes any feature-
tag it do not support. A Proxy-Supported header present in the
response SHALL NOT be touched by the proxies.
Example:
C->P1: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
User-Agent: PhonyClient/1.2
P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
Via: 2.0 prox1.example.com
P2->S: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-blech
Via: 2.0 prox1.example.com, 2.0 prox2.example.com
S->C: RTSP/2.0 200 OK
Supported: foo, bar, baz
Proxy-Supported: proxy-foo, proxy-blech
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
Via: 2.0 prox1.example.com, 2.0 prox2.example.com
16.34. 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
Schulzrinne, et al. Expires August 28, 2008 [Page 103]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
sender's capabilities to the recipient. The methods listed may or
may not be applicable to the Request-URI; the Allow header field
(section 14.7) MAY be used to indicate methods allowed for a
particular URI.
Example of use:
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
In the event that there are proxies between the sender and the
recipient of a response, each intervening proxy MUST modify the
Public header field to remove any methods that are not supported via
that proxy. The resulting Public header field will contain an
intersection of the sender's methods and the methods allowed through
by the intervening proxies.
In general proxies should allow all methods to transparently pass
through from the sending RTSP agent to the receiving RTSP agent,
but there may be cases where this is not desirable for a given
proxy. Modification of the Public response header field by the
intervening proxies ensures that the request sender gets an
accurate response indicating the methods that can be used on the
target agent via the proxy chain.
16.35. Range
The Range header specifies a time range in PLAY (Section 13.4), PAUSE
(Section 13.5), SETUP (Section 13.3), and REDIRECT (Section 13.9)
requests and responses.
The range can be specified in a number of units. This specification
defines smpte (Section 4.4), npt (Section 4.5), and clock
(Section 4.6) range units. While byte ranges [H14.35.1] and other
extended units MAY be used, their behavior is unspecified since they
are not normally meaningful in RTSP. Servers supporting the Range
header MUST understand the NPT range format and SHOULD understand the
SMPTE range format. If the Range header is sent in a time format
that is not understood, the recipient SHOULD return 456 (Header Field
Not Valid for Resource) and include an Accept-Ranges header
indicating the supported time formats for the given resource.
The Range header MAY contain a time parameter in UTC, specifying the
time at which the operation is to be made effective. This
functionality SHALL be used only with the REDIRECT method.
Example:
Range: clock=19960213T143205Z-;time=19970123T143720Z
Schulzrinne, et al. Expires August 28, 2008 [Page 104]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
The notation is similar to that used for the HTTP/1.1 [RFC2616]
byte-range header. It allows clients to select an excerpt from
the media object, and to play from a given point to the end as
well as from the current location to a given point.
Ranges are half-open intervals, including the first point, but
excluding the second point. In other words, a range of A-B starts
exactly at time A, but stops just before B. Only the start time of a
media unit such as a video or audio frame is relevant. For example,
assume that video frames are generated every 40 ms. A range of 10.0-
10.1 would include a video frame starting at 10.0 or later time and
would include a video frame starting at 10.08, even though it lasted
beyond the interval. A range of 10.0-10.08, on the other hand, would
exclude the frame at 10.08.
By default, range intervals increase, where the second point is
larger than the first point.
Example:
Range: npt=10-15
However, range intervals can also decrease if the Scale header (see
Section 16.40) 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.
Schulzrinne, et al. Expires August 28, 2008 [Page 105]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
16.36. Referer
See [H14.36]. The URI refers to that of the presentation
description, typically retrieved via HTTP.
16.37. Retry-After
See [H14.37].
16.38. Require
The Require request-header field is used by clients or servers to
ensure that the other end-point supports features that are required
in respect to this request. It can also be used to query if the
other end-point supports certain features, however the use of the
Supported (Section 16.44) is much more effective in this purpose.
The server MUST respond to this header by using the Unsupported
header to negatively acknowledge those feature-tags which are NOT
supported. The response SHALL use the error code 551 (Option Not
Supported). This header does not apply to proxies, for the same
functionality in respect to proxies see Proxy-Require header
(Section 16.32).
This is to make sure that the client-server interaction will
proceed without delay when all features are understood by both
sides, and only slow down if features are not understood (as in
the example below). For a well-matched client-server pair, the
interaction proceeds quickly, saving a round-trip often required
by negotiation mechanisms. In addition, it also removes state
ambiguity when the client requires features that the server does
not understand.
Example (Not complete):
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/2.0 551 Option not supported
CSeq: 302
Unsupported: funky-feature
In this example, "funky-feature" is the feature-tag which indicates
to the client that the fictional Funky-Parameter field is required.
The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.
Schulzrinne, et al. Expires August 28, 2008 [Page 106]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Proxies and other intermediary devices SHALL ignore this header. If
a particular extension requires that intermediate devices support it,
the extension should be tagged in the Proxy-Require field instead
(see Section 16.32).
16.39. RTP-Info
The RTP-Info response-header field is used to set RTP-specific
parameters in the PLAY response. For streams using RTP as transport
protocol the RTP-Info header SHOULD be part of a 200 response to
PLAY.
The exclusion of the RTP-Info in a PLAY response for RTP
transported media will result in that a client needs to
synchronize the media streams using RTCP. This may have negative
impact as the RTCP can be lost, and does not need to be
particulary timely in their arrival. Also functionality as
informing the client from which packet a seek has occurred is
affected.
The RTP-Info MAY also be included in SETUP responses to provide
synchronization information when changing transport parameters, see
Section 13.3.
The header can carry the following parameters:
url: Indicates the stream URI which for which the following RTP
parameters correspond, this URI MUST be the same used in the
SETUP request for this media stream. Any relative URI SHALL
use the Request-URI as base URI. This parameter SHALL be
present.
ssrc: The Synchronization source (SSRC) that the RTP timestamp and
sequence number provide applies to. This parameter SHALL be
present.
seq: Indicates the sequence number of the first packet of the stream
that is direct result of the request. This allows clients to
gracefully deal with packets when seeking. The client uses
this value to differentiate packets that originated before the
seek from packets that originated after the seek. Note that a
client may not receive the packet with the expressed sequence
number, and instead packets with a higher sequence number, due
to packet loss or reordering. This parameter is RECOMMENDED to
be present.
Schulzrinne, et al. Expires August 28, 2008 [Page 107]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
rtptime: SHALL indicate the RTP timestamp value corresponding to the
start time value in the Range response header, or if not
explicitly given the implied start point. The client uses this
value to calculate the mapping of RTP time to NPT or other
media timescale. This parameter SHOULD be present to ensure
inter-media synchronization is achieved. There exist no
requirement that any received RTP packet will have the same RTP
timestamp value as the one in the parameter used to establish
synchronization.
A mapping from RTP timestamps to NTP timestamps (wall clock) is
available via RTCP. However, this information is not sufficient
to generate a mapping from RTP timestamps to media clock time
(NPT, etc.). Furthermore, in order to ensure that this
information is available at the necessary time (immediately at
startup or after a seek), and that it is delivered reliably, this
mapping is placed in the RTSP control channel.
In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to NTP,
using initial RTCP sender reports to do the mapping, and later
reports to check drift against the mapping.
Example:
Range:npt=3.25-15
RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;
rtptime=12345678,url="rtsp://example.com/foo/video"
ssrc=9A9DE123:seq=30211;rtptime=29567112
Lets assume that Audio uses a 16kHz RTP timestamp clock and Video
a 90kHz RTP timestamp clock. Then the media synchronization is
depicted in the following way.
NPT 3.0---3.1---3.2-X-3.3---3.4---3.5---3.6
Audio PA A
Video V PV
X: NPT time value = 3.25, from Range header.
A: RTP timestamp value for Audio from RTP-Info header (12345678).
V: RTP timestamp value for Video from RTP-Info header (29567112).
PA: RTP audio packet carrying an RTP timestamp of 12344878. Which
corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2
PV: RTP video packet carrying an RTP timestamp of 29573412. Which
corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32
Schulzrinne, et al. Expires August 28, 2008 [Page 108]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
16.40. Scale
A scale value of 1 indicates normal play at the normal forward
viewing rate. If not 1, the value corresponds to the rate with
respect to normal viewing rate. For example, a ratio of 2 indicates
twice the normal viewing rate ("fast forward") and a ratio of 0.5
indicates half the normal viewing rate. In other words, a ratio of 2
has normal play time increase at twice the wallclock rate. For every
second of elapsed (wallclock) time, 2 seconds of content will be
delivered. A negative value indicates reverse direction. For
certain media transports this may require certain considerations to
work consistent, see Appendix C.1 for description on how RTP handles
this.
Unless requested otherwise by the Speed parameter, the data rate
SHOULD not be changed. Implementation of scale changes depends on
the server and media type. For video, a server may, for example,
deliver only key frames or selected key frames. For audio, for
example, it may time-scale the audio while preserving pitch or, less
desirably, deliver fragments of audio.
The server should try to approximate the viewing rate, but may
restrict the range of scale values that it supports. The response
MUST contain the actual scale value chosen by the server.
If the server does not implement the possibility to scale, it will
not return a Scale header. A server supporting Scale operations for
PLAY SHALL indicate this with the use of the "play.scale" feature-
tags.
When indicating a negative scale for a reverse playback, the Range
header MUST indicate a decreasing range as described in
Section 16.35.
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
Range: npt=15-10
16.41. Speed
The Speed request-header field requests the server to deliver data to
the client at a particular speed, contingent on the server's ability
and desire to serve the media stream at the given speed.
Implementation by the server is OPTIONAL. The default is the bit
rate of the stream.
The parameter value is expressed as a decimal ratio, e.g., a value of
2.0 indicates that data is to be delivered twice as fast as normal.
Schulzrinne, et al. Expires August 28, 2008 [Page 109]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
A speed of zero is invalid. All speeds may not be possible to
support. Therefore the actual used speed MUST be included in the
response. The lack of a response header is indication of lack of
support from the server of this functionality. Support of the speed
functionality are indicated by the "play.speed" feature-tag.
Example:
Speed: 2.5
Use of this field changes the bandwidth used for data delivery. It
is meant for use in specific circumstances where preview of the
presentation at a higher or lower rate is necessary. Implementors
should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss
rates. If the data transport is performed over non-dedicated best-
effort networks the sender is required to perform congestion control
of the stream(s). This can result in that the communicated speed is
impossible to maintain.
16.42. Server
See [H14.38], however the header syntax is corrected in
Section 20.2.3.
16.43. Session
The Session request-header and response-header field identifies an
RTSP session. An RTSP session is created by the server as a result
of a successful SETUP request and in the response the session
identifier is given to the client. The RTSP session exist until
destroyed by a TEARDOWN or timed out by the server.
The session identifier is chosen by the server (see Section 4.3) and
MUST be returned in the SETUP response. Once a client receives a
session identifier, it SHALL be included in any request related to
that session. This means that the Session header MUST be included in
a request using the following methods: PLAY, PAUSE, and TEARDOWN, and
MAY be included in SETUP, OPTIONS, SETPARAMETER, GET_PARAMETER, and
REDIRECT, and SHALL NOT be included in DESCRIBE. In an RTSP response
the session header SHALL be included in methods, SETUP, PLAY, and
PAUSE, and MAY be included in methods, TEARDOWN, and REDIRECT, and if
included in the request of the following methods it SHALL also be
included in the response, OPTIONS, GET_PARAMETER, and SETPARAMETER,
and SHALL NOT be included in DESCRIBE.
Note that a session identifier identifies an RTSP session across
Schulzrinne, et al. Expires August 28, 2008 [Page 110]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
transport sessions or connections. RTSP requests for a given session
can use different URIs (Presentation and media URIs). Note, that
there are restrictions depending on the session which URIs that are
acceptable for a given method. However, multiple "user" sessions for
the same URI from the same client will require use of different
session identifiers.
The session identifier is needed to distinguish several delivery
requests for the same URI coming from the same client.
The response 454 (Session Not Found) SHALL be returned if the session
identifier is invalid.
16.44. Supported
The Supported header field enumerates all the extensions supported by
the client or server using feature tags. The header carries the
extensions supported by the message sending entity. The Supported
header MAY be included in any request. When present in a request,
the receiver MUST respond with its corresponding Supported header.
Note, also in 4xx and 5xx responses is the supported header included.
The Supported header field contains a list of feature-tags, described
in Section 4.7, that are understood by the client or server.
Example:
C->S: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
Supported: bar, blech, baz
16.45. Timestamp
The Timestamp general-header field describes when the agent sent the
request. The value of the timestamp is of significance only to the
agent and may use any timescale. The responding agent MUST echo the
exact same value and MAY, if it has accurate information about this,
add a floating point number indicating the number of seconds that has
elapsed since it has received the request. The timestamp is used by
the agent to compute the round-trip time to the responding agent so
that it can adjust the timeout value for retransmissions. It also
resolves retransmission ambiguities for unreliable transport of RTSP.
Schulzrinne, et al. Expires August 28, 2008 [Page 111]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
16.46. Transport
The Transport request and response header field indicates which
transport protocol is to be used and configures its parameters such
as destination address, compression, multicast time-to-live and
destination port for a single stream. It sets those values not
already determined by a presentation description.
Transports are comma separated, listed in order of preference.
Parameters may be added to each transport, separated by a semicolon.
The server SHOULD return a Transport response-header field in the
response to indicate the values actually chosen. The Transport
header field MAY also be used to change certain transport parameters.
A server MAY refuse to change parameters of an existing stream.
A Transport request header field MAY contain a list of transport
options acceptable to the client, in the form of multiple transport-
spec entries. In that case, the server MUST return the single
(transport-spec) which was actually chosen. The number of transport-
spec entries is expected to be limited as the client will get
guidance on what configurations that are possible from the
presentation description.
A transport-spec transport option may only contain one of any given
parameter within it. Parameters MAY be given in any order.
Additionally, it may only contain the unicast or the multicast
transport type parameter. Unknown parameters SHALL be ignored. The
requester need to ensure that the responder understands the
parameters through the use of feature tags and the Require header.
Any parameters part of future extensions requires clarification if
they are safe to ignore in accordance to this specification, or are
required to be understood. If a parameter is required to be
understood, then a feature-tag MUST be defined for the functionality
and used in the Require or Proxy-Require headers.
The Transport header field is restricted to describing a single
media stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a
multitude of session description formats greatly simplifies
designs of firewalls.
The general syntax for the transport specifier is a list of slash
separated tokens:
Value1/Value2/Value3...
Which for RTP transports take the form:
RTP/profile/lower-transport.
Schulzrinne, et al. Expires August 28, 2008 [Page 112]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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:
dest_addr: The presence of this parameter and its values indicates
the destination address or addresses (host address and port
pairs for IP flows) necessary for the media transport.
No dest_addr: The lack of the dest_addr parameter indicates that the
server SHALL send media to same address for which the RTSP
messages originates. Does not work for transports requiring
explicitly given destination ports.
The choice of method for indicating where the media is to be
delivered depends on the use case. In many 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.
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.
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
Schulzrinne, et al. Expires August 28, 2008 [Page 113]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
multiple entities is unspecified.
The client originating the RTSP request MAY specify the
destination address of the stream recipient with the host
address part of the tuple. When the destination address is
specified, the recipient may be a different party than the
originator of the request. To avoid becoming the unwitting
perpetrator of a remote-controlled denial-of-service attack, a
server MUST perform security checks (see Section 21.1) and
SHOULD log such attempts before allowing the client to direct a
media stream to a recipient address not chosen by the server.
Implementations cannot rely on TCP as reliable means of client
identification. If the server does not allow the host address
part of the tuple to be set, it SHALL return 463 (Destination
Prohibited).
The host address part of the tuple MAY be empty, for example
":58044", in cases when only destination port is desired to be
specified.
src_addr: A general source address parameter that can contain one or
more address specifications. Each combination of Protocol/
Profile/Lower Transport needs to have the format and
interpretation of its address specification defined. For RTP/
AVP/UDP and RTP/AVP/TCP, the address specification is a tuple
containing a host address and port.
This parameter MUST be specified by the server if it transmits
media packets from another address than the one RTSP messages
are sent to. This will allow the client to verify source
address and give it a destination address for its RTCP feedback
packets if RTP is used. The address or addresses indicated in
the src_addr parameter SHOULD be used both for sending and
receiving of the media streams data packets. The main reasons
are threefold: First, indicating the port and source address(s)
lets the receiver know where from the packets is expected to
originate. Secondly, traversal of NATs are greatly simplified
when traffic is flowing symmetrically over a NAT binding.
Thirdly, certain NAT traversal mechanisms, needs to know to
which address and port to send so called "binding packets" from
the receiver to the sender, thus creating a address binding in
the NAT that the sender to receiver packet flow can use.
This information may also be available through SDP.
However, since this is more a feature of transport than
media initialization, the authoritative source for this
information should be in the SETUP response.
Schulzrinne, et al. Expires August 28, 2008 [Page 114]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
mode: The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY. The RECORD value was defined in
RFC 2326 and is in this specification unspecified but reserved.
interleaved: The interleaved parameter implies mixing the media
stream with the control stream in whatever protocol is being
used by the control stream, using the mechanism defined in
Section 14. The argument provides the channel number to be
used in the $ statement and MUST be present. This parameter
MAY be specified as a range, e.g., interleaved=4-5 in cases
where the transport choice for the media stream requires it,
e.g. for RTP with RTCP. The channel number given in the
request are only a guidance from the client to the server on
what channel number(s) to use. The server MAY set any valid
channel number in the response. The declared channel(s) are
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. When included in requests the value
indicate the TTL value that the client desires to use. In
response the value actually being used is returned. A server
will need to consider what values that are reasonable and also
the authority of the user to set this value.
RTP-specific:
These parameters are MAY only be used if the media transport protocol
is RTP.
ssrc: The ssrc parameter, if included in a SETUP response, indicates
the RTP SSRC [RFC3550] value(s) that will be used by the media
server for RTP packets within the stream. It is expressed as
an eight digit hexadecimal value.
The ssrc parameter SHALL NOT be specified in requests. The
functionality of specifying the ssrc parameter in a SETUP
request is deprecated as it is incompatible with the
specification of RTP in RFC 3550[RFC3550]. If the parameter is
Schulzrinne, et al. Expires August 28, 2008 [Page 115]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
included in the Transport header of a SETUP request, the server
MAY ignore it, and choose appropriate SSRCs for the stream.
The server MAY set the ssrc parameter in the Transport header
of the response.
The parameters defined below MAY only be used if the media transport
protocol if the lower-level transport is connection-oriented (such as
TCP). However, these parameters MUST NOT be used when interleaving
data over the RTSP control connection.
setup: Clients use the setup parameter on the Transport line in a
SETUP request, to indicate the roles it wishes to play in a TCP
connection. This parameter is adapted from [RFC4145]. We
discuss the use of this parameter in RTP/AVP/TCP non-
interleaved transport in Appendix C.2.2; the discussion below
is limited to syntactic issues. Clients may specify the
following values for the setup parameter: ["active":] The
client will initiate an outgoing connection. ["passive":] The
client will accept an incoming connection. ["actpass":] The
client is willing to accept an incoming connection or to
initiate an outgoing connection.
If a client does not specify a setup value, the "active" value
is assumed.
In response to a client SETUP request where the setup parameter
is set to "active", a server's 2xx reply MUST assign the setup
parameter to "passive" on the Transport header line.
In response to a client SETUP request where the setup parameter
is set to "passive", a server's 2xx reply MUST assign the setup
parameter to "active" on the Transport header line.
In response to a client SETUP request where the setup parameter
is set to "actpass", a server's 2xx reply MUST assign the setup
parameter to "active" or "passive" on the Transport header
line.
Note that the "holdconn" value for setup is not defined for
RTSP use, and MUST NOT appear on a Transport line.
connection: Clients use the setup parameter on the Transport line in
a SETUP request, to indicate the SETUP request prefers the
reuse of an existing connection between client and server (in
which case the client sets the "connection" parameter to
"existing"), or that the client requires the creation of a new
connection between client and server (in which cast the client
sets the "connection" parameter to "new"). Typically, clients
Schulzrinne, et al. Expires August 28, 2008 [Page 116]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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).
The combination of transport protocol, profile and lower transport
needs to be defined. A number of combinations are defined in the
Appendix C.
Below is a usage example, showing a client advertising the capability
to handle multicast or unicast, preferring multicast. Since this is
a unicast-only stream, the server responds with the proper transport
parameters for unicast.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Transport: RTP/AVP;multicast;mode="PLAY",
RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";mode="PLAY"
Accept-Ranges: NPT, SMPTE, UTC
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";src_addr="192.0.2.224:6256"/
"192.0.2.224:6257";mode="PLAY"
Accept-Ranges: NPT
16.47. Unsupported
The Unsupported response-header field lists the features not
supported by the server. In the case where the feature was specified
via the Proxy-Require field (Section 16.32), 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).
Schulzrinne, et al. Expires August 28, 2008 [Page 117]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
The request SHALL NOT be forwarded.
See Section 16.38 for a usage example.
16.48. User-Agent
See [H14.43] for explanation, however the syntax is clarified due to
an error in RFC 2616. A Client SHOULD include this header in all
RTSP messages it sends.
16.49. Vary
See [H14.44].
16.50. Via
See [H14.45].
16.51. WWW-Authenticate
See [H14.47].
Schulzrinne, et al. Expires August 28, 2008 [Page 118]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
Caching Proxy: This type of proxy is used to reduce the workload on
servers and connections. By caching a presentation, both
description and media streams the proxy can serve a client
content without requesting it from the server once it has been
cached and hasn't become stale. See the caching Section 18.
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, and
often also media stream translation or redirection.
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. It can also provide network owners with a
logging and audit point for RTSP sessions, e.g. for
corporations that tracks or limits their employees access to
certain type of content.
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.
The existence of proxies must always be considered when developing
new RTSP extensions. There must be definition of how proxies may
Schulzrinne, et al. Expires August 28, 2008 [Page 119]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
handle the extension, if it is required to understand it, thus
requiring a feature-tag to be used in the Proxy-Require header.
Schulzrinne, et al. Expires August 28, 2008 [Page 120]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
18. Caching
In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the
exception of the presentation description returned by DESCRIBE.
(Since the responses for anything but DESCRIBE and GET_PARAMETER do
not return any data, caching is not really an issue for these
requests.) However, it is desirable for the continuous media data,
typically delivered out-of-band with respect to RTSP, to be cached,
as well as the session description.
On receiving a SETUP or PLAY request, a proxy ascertains whether it
has an up-to-date copy of the continuous media content and its
description. It can determine whether the copy is up-to-date by
issuing a SETUP or DESCRIBE request, respectively, and comparing the
Last-Modified header with that of the cached copy. If the copy is
not up-to-date, it modifies the SETUP transport parameters as
appropriate and forwards the request to the origin server.
Subsequent control commands such as PLAY or PAUSE then pass the proxy
unmodified. The proxy delivers the continuous media data to the
client, while possibly making a local copy for later reuse. The
exact behavior allowed to the cache is given by the cache-response
directives described in Section 16.10. A cache MUST answer any
DESCRIBE requests if it is currently serving the stream to the
requestor, as it is possible that low-level details of the stream
description may have changed on the origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
through" variety. Rather than retrieving the whole resource from the
origin server, the cache simply copies the streaming data as it
passes by on its way to the client. Thus, it does not introduce
additional latency.
To the client, an RTSP proxy cache appears like a regular media
server, to the media origin server like a client. Just as an HTTP
cache has to store the content type, content language, and so on for
the objects it caches, a media cache has to store the presentation
description. Typically, a cache eliminates all transport-references
(that is, e.g. multicast information) from the presentation
description, since these are independent of the data delivery from
the cache to the client. Information on the encodings remains the
same. If the cache is able to translate the cached media data, it
would create a new presentation description with all the encoding
possibilities it can offer.
Schulzrinne, et al. Expires August 28, 2008 [Page 121]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
It should be stressed that using the HTTP authentication alone does
not provide full control message security. Therefore, in
environments requiring tighter security for the control messages, TLS
SHOULD be used, see Section 19.2.
19.2. RTSP over TLS
RTSP SHALL follow the same guidelines with regards to TLS [RFC4346]
usage as specified for HTTP, see [RFC2818]. RTSP over TLS is
separated from unsecured RTSP both on URI level and port level.
Instead of using the "rtsp" scheme identifier in the URI, the "rtsps"
scheme identifier MUST be used to signal RTSP over TLS. If no port
is given in a URI with the "rtsps" scheme, port 322 SHALL be used for
TLS over TCP/IP.
When a client tries to setup an insecure channel to the server (using
the "rtsp" URI), and the policy for the resource requires a secure
channel, the server SHALL redirect the client to the secure service
by sending a 301 redirect response code together with the correct
Location URI (using the "rtsps" scheme). A user or client MAY
upgrade a non secured URI to a secured by changing the scheme from
"rtsp" to "rtsps". A server implementing support for "rtsps" SHALL
allow this.
It should be noted that TLS allows for mutual authentication (when
using both server and client certificates). Still, one of the more
common way TLS is used is to only provide server side authentication
(often to avoid client certificates). TLS is then used in addition
to HTTP authentication, providing transport security and server
authentication, while HTTP Authentication is used to authenticate the
client.
RTSP includes the possibility to keep a TCP session up between the
Schulzrinne, et al. Expires August 28, 2008 [Page 122]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
client and server, throughout the RTSP session lifetime. It may be
convenient to keep the TCP session, not only to save the extra setup
time for TCP, but also the extra setup time for TLS (even if TLS uses
the resume function, there will be almost two extra roundtrips).
Still, when TLS is used, such behavior introduces extra active state
in the server, not only for TCP and RTSP, but also for TLS. This may
increase the vulnerability to DoS attacks.
In addition to these recommendations, Section 19.3 gives further
recommendations of TLS usage with proxies.
19.3. Security and Proxies
The nature of a proxy is often to act as a "man-in-the-middle", while
security is often about preventing the existence of a "man-in-the-
middle". This section provides clients with the possibility to use
proxies even when applying secure transports (TLS) between the RTSP
agents. The TLS proxy mechanism allows for server and proxy
identification using certificates. However, the client can not be
identified based on certificates. The client needs to select between
using the procedure specified below or using a TLS connection
directly (by-passing any proxies) to the server. The choice may be
dependent on policies.
There are basically two categories of proxies, the transparent
proxies (of which the client is not aware) and the non-transparent
proxies (of which the client is aware). An infrastructure based on
proxies requires that the trust model is such that both client and
servers can trust the proxies to handle the RTSP messages correctly.
To be able to trust a proxy, the client and server also needs to be
aware of the proxy. Hence, transparent proxies cannot generally be
seen as trusted and will not work well with security (unless they
work only at transport layer). In the rest of this section any
reference to proxy will be to a non-transparent proxy, which inspects
or manipulate the RTSP messages.
HTTP Authentication is built on the assumption of proxies and can
provide user-proxy authentication and proxy-proxy/server
authentication in addition to the client-server authentication.
When TLS is applied and a proxy is used, the client will connect to
the proxy's address when connecting to any RTSP server. This implies
that for TLS, the client will authenticate the proxy server and not
the end server. Note that when the client checks the server
certificate in TLS, it MUST check the proxy's identity (URI or
possibly other known identity) against the proxy's identity as
presented in the proxy's Certificate message.
Schulzrinne, et al. Expires August 28, 2008 [Page 123]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
The problem is that for a proxy accepted by the client, the proxy
needs to be provided information on which grounds it should accept
the next-hop certificate. Both the proxy and the user may have rules
for this, and the user have the possibility to select the desired
behavior. To handle this case, the Accept-Credentials header (See
Section 16.2) is used, where the client can force the proxy/proxies
to relay back the chain of certificates used to authenticate any
intermediate proxies as well as the server. Given the assumption
that the proxies are viewed as trusted, it gives the user a
possibility to enforce policies to each trusted proxy of whether it
should accept the next entity in the chain.
A proxy MUST use TLS for the next hop if the RTSP request includes a
"rtsps" URI. TLS MAY be applied on intermediate links (e.g. between
client and proxy, or between proxy and proxy), even if the resource
and the end server does not require to use it. The proxy SHALL when
initiating the next hop TLS connection use the incomming TLS
connections CipherSuite list, only modified by removing any cipher
suits that the proxy does not support. In case a proxy fails to
establish a TLS connection due to cipher suite mismatch between proxy
and next hop proxy or server, this is indicated using error code 472
(Failure to establish secure connection).
19.3.1. Accept-Credentials
The Accept-Credentials header can be used by the client to distribute
simple authorization policies to intermediate proxies. The client
includes the Accept-Credentials header to dictate how the proxy
treats the server/next proxy certificate. There are currently three
methods defined:
Any, which means that the proxy (or proxies) SHALL accept whatever
certificate presented. This is of course not a recommended
option to use, but may be useful in certain circumstances (such
as testing).
Proxy, which means that the proxy (or proxies) MUST use its own
policies to validate the certificate and decide whether to
accept it or not. This is convenient in cases where the user
has a strong trust relation with the proxy. Reason why a
strong trust relation may exist are; personal/company proxy,
proxy has a out-of-band policy configuration mechanism.
User, which means that the proxy (or proxies) MUST send credential
information about the next hop to the client for authorization.
The client can then decide whether the proxy should accept the
certificate or not. See Section 19.3.2 for further details.
Schulzrinne, et al. Expires August 28, 2008 [Page 124]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
If the Accept-Credentials header is not included in the RTSP request
from the client, then the "Proxy" method SHALL be used as default.
If an other method than the "Proxy" is to be used, then the Accept-
Credentials header SHALL be included in all of the RTSP request from
the client. This is because it cannot be assumed that the proxy
always keeps the TLS state or the users previously preference between
different RTSP messages (in particular if the time interval between
the messages is long).
With the "Any" and "Proxy" methods the proxy will apply the policy as
defined for respectively method. If the policy do not accept the
credentials of the next hop, the entity SHALL respond with a message
using status code 471 (Connection Credentials not accepted).
An RTSP request in the direction server to client MUST NOT include
the Accept-Credential header. As for the non-secured communication,
the possibility for these request depends on the presence of a client
established connection. However if the server to client request is
in relation to a session established over a TLS secured channel, if
MUST be sent in a TLS secured connection. That secured connection
MUST also be the one used by the last client to server request. If
no such transport connection exist at the time when the server desire
to send the request, it silently fails.
Further policies MAY be defined and registered, but should be done so
with caution.
19.3.2. User approved TLS procedure
For the "User" method each proxy MUST perform the the following
procedure for each RTSP request:
o Setup the TLS session to the next hop if not already present (i.e.
run the TLS handshake, but do not send the RTSP request).
o Extract the peer certificate chain for the TLS session.
o Check if a matching identity and hash of the peer certificate is
present in the Accept-Credentials header. If present, send the
message to the next hop, and conclude these procedures. If not,
go to the next step.
o The proxy responds to the RTSP request with a 470 or 407 response
code. The 407 response code MAY be used when the proxy requires
both user and connection authorization from user or client. In
this message the proxy SHALL include a Connection-Credentials
header, see Section 16.12 with the next hop's identity and
certificate.
Schulzrinne, et al. Expires August 28, 2008 [Page 125]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
The client MUST upon receiving a 470 or 407 response with Connection-
Credentials header take the decision on whether to accept the
certificate or not (if it cannot do so, the user SHOULD be
consulted). If the certificate is accepted, the client has to again
send the RTSP request. In that request the client has to include the
Accept-Credentials header including the hash over the DER encoded
certificate for all trusted proxies in the chain.
Example:
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Ranges: NPT, SMPTE, UTC
Accept-Credentials: User
P->C: RTSP/2.0 470 Connection Authorization Required
CSeq: 2
Connection-Credentials: "rtsps://test.example.org";
MIIDNTCCAp...
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Credentials: User "rtsps://test.example.org";sha-256;
dPYD7txpoGTbAqZZQJ+vaeOkyH4=
Accept-Ranges: NPT, SMPTE, UTC
P->S: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Via: RTSP/2.0 proxy.example.org
Accept-Credentials: User "rtsps://test.example.org";sha-256;
dPYD7txpoGTbAqZZQJ+vaeOkyH4=
Accept-Ranges: NPT, SMPTE, UTC
One implication of this process is that the connection for secured
RTSP messages may take significantly more round-trip times for the
first message. An complete extra message exchange between the proxy
connecting to the next hop and the client results because of the
process for approval for each hop. However after the first message
exchange the remaining message should not be delayed, if each message
contains the chain of proxies that the requestor accepts. The
procedure of including the credentials in each request rather than
building state in each proxy, avoids the need for revocation
procedures.
Schulzrinne, et al. Expires August 28, 2008 [Page 126]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
20. Syntax
The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)
as defined in RFC 4234 [RFC4234]. It uses the basic definitions
present in RFC 4234.
Please note that ABNF strings, e.g. "Accept", are case insensitive
as specified in section 2.3 of RFC 4234.
20.1. Base Syntax
RTSP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
This is intended to behave exactly as HTTP/1.1 as described in RFC
2616 [RFC2616]. The SWS construct is used when linear white space is
optional, generally between tokens and separators.
To separate the header name from the rest of value, a colon is used,
which, by the above rule, allows whitespace before, but no line
break, and whitespace after, including a linebreak. The HCOLON
defines this construct.
OCTET = %x00-FF ; any 8-bit sequence of data
CHAR = %x01-7F ; any US-ASCII character (octets 1 - 127)
UPALPHA = %x41-5A ; any US-ASCII uppercase letter "A".."Z"
LOALPHA = %x61-7A ;any US-ASCII lowercase letter "a".."z"
ALPHA = UPALPHA / LOALPHA
DIGIT = %x30-39 ; any US-ASCII digit "0".."9"
CTL = %x00-1F / %x7F ; any US-ASCII control charac7ter
; (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
Schulzrinne, et al. Expires August 28, 2008 [Page 127]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
LWS = [CRLF] 1*( SP / HT )
SWS = [LWS] ; sep whitespace
HCOLON = *( SP / HT ) ":" SWS
TEXT = %x20-7D / %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 / %x23-7D / %x80-FF ; any TEXT except <">
quoted-pair = BACKSLASH CHAR
ctext = %x20-27 / %x2A-7D
/ %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 / "+" / "/"
Schulzrinne, et al. Expires August 28, 2008 [Page 128]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
SLASH = SWS "/" SWS ; slash
EQUAL = SWS "=" SWS ; equal
LPAREN = SWS "(" SWS ; left parenthesis
RPAREN = SWS ")" SWS ; right parenthesis
COMMA = SWS "," SWS ; comma
SEMI = SWS ";" SWS ; semicolon
COLON = SWS ":" SWS ; colon
LDQUOT = SWS DQ ; open double quotation mark
RDQUOT = DQ SWS ; close double quotation mark
RAQUOT = ">" SWS ; right angle quote
LAQUOT = SWS "<" ; left angle quote
TEXT-UTF8char = %x21-7E / UTF8-NONASCII
UTF8-NONASCII = %xC0-DF 1UTF8-CONT
/ %xE0-EF 2UTF8-CONT
/ %xF0-F7 3UTF8-CONT
/ %xF8-FB 4UTF8-CONT
/ %xFC-FD 5UTF8-CONT
UTF8-CONT = %x80-BF
20.2. RTSP Protocol Definition
20.2.1. Generic Protocol elements
Schulzrinne, et al. Expires August 28, 2008 [Page 129]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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>
Schulzrinne, et al. Expires August 28, 2008 [Page 130]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
RTSP-URI = schemes ":" URI-rest
RTSP-URI-Ref = RTSP-URI / RTSP-Relative
schemes = "rtsp" / "rtsps" / scheme
scheme = < As defined in RFC 3986>
URI-rest = hier-part [ "?" query ]
hier-part = "//" authority path-abempty
RTSP-Relative = relative-part [ "?" query ]
relative-part = "//" authority path-abempty
/ path-absolute
/ path-noscheme
/ path-empty
authority = < As defined in RFC 3986>
query = < As defined in RFC 3986>
path = path-abempty ; begins with "/" or is empty
/ path-absolute ; begins with "/" but not "//"
/ path-noscheme ; begins with a non-colon segment
/ path-rootless ; begins with a segment
/ path-empty ; zero characters
path-abempty = *( "/" segment )
path-absolute = "/" [ segment-nz *( "/" segment ) ]
path-noscheme = segment-nz-nc *( "/" segment )
path-rootless = segment-nz *( "/" segment )
path-empty = 0<pchar>
segment = *pchar [";" *pchar]
segment-nz = ( 1*pchar [";" *pchar]) / (";" *pchar)
segment-nz-nc = ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc)
; non-zero-length segment without any colon ":"
pchar = unreserved / pct-encoded / sub-delims / ":" / "@"
pchar-nc = unreserved / pct-encoded / sub-delims / "@"
sub-delims = "!" / "$" / "&" / "'" / "(" / ")"
/ "*" / "+" / "," / "="
Schulzrinne, et al. Expires August 28, 2008 [Page 131]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
smpte-range = smpte-type "=" smpte-range-spec
; See section 3.4
smpte-range-spec = ( smpte-time "-" [ smpte-time ] )
/ ( "-" smpte-time )
smpte-type = "smpte" / "smpte-30-drop"
/ "smpte-25" / smpte-type-extension
; other timecodes may be added
smpte-type-extension = token
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
[ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]
npt-range = "npt=" npt-range-spec
npt-range-spec = ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
npt-time = "now" / npt-sec / npt-hhmmss
npt-sec = 1*DIGIT [ "." *DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
npt-hh = 1*DIGIT ; any positive number
npt-mm = 1*2DIGIT ; 0-59
npt-ss = 1*2DIGIT ; 0-59
utc-range = "clock=" utc-range-spec
utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
utc-time = utc-date "T" utc-clock "Z"
utc-date = 8DIGIT
utc-clock = 6DIGIT [ "." fraction ]
fraction = 1*DIGIT
feature-tag = token
session-id = 1*256( ALPHA / DIGIT / safe )
extension-header = header-name HCOLON header-value
header-name = token
header-value = *(TEXT-UTF8char / UTF8-CONT / LWS)
20.2.2. Message Syntax
Schulzrinne, et al. Expires August 28, 2008 [Page 132]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
RTSP-message = Request / Response ;RTSP/2.0 messages
Request = Request-Line
*(general-header
/ request-header
/ entity-header )
CRLF
[ message-body ]
Response = Status-Line
*( general-header
/ response-header
/ entity-header )
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"
/ "REDIRECT"
/ "SETUP"
/ "SET_PARAMETER"
/ "TEARDOWN"
/ extension-method
extension-method = token
Request-URI = "*" / RTSP-URI
RTSP-Version = "RTSP/" 1*DIGIT "." 1*DIGIT
message-body = 1*OCTET
Status-Code = "100" ; Continue
/ "200" ; OK
/ "300" ; Multiple Choices
/ "301" ; Moved Permanently
/ "302" ; Found
/ "303" ; See Other
/ "304" ; Not Modified
/ "305" ; Use Proxy
/ "400" ; Bad Request
/ "401" ; Unauthorized
/ "402" ; Payment Required
Schulzrinne, et al. Expires August 28, 2008 [Page 133]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
/ "403" ; Forbidden
/ "404" ; Not Found
/ "405" ; Method Not Allowed
/ "406" ; Not Acceptable
/ "407" ; Proxy Authentication Required
/ "408" ; Request Time-out
/ "410" ; Gone
/ "411" ; Length Required
/ "412" ; Precondition Failed
/ "413" ; Request Entity Too Large
/ "414" ; Request-URI Too Large
/ "415" ; Unsupported Media Type
/ "451" ; Parameter Not Understood
/ "452" ; reserved
/ "453" ; Not Enough Bandwidth
/ "454" ; Session Not Found
/ "455" ; Method Not Valid in This State
/ "456" ; Header Field Not Valid for Resource
/ "457" ; Invalid Range
/ "458" ; Parameter Is Read-Only
/ "459" ; Aggregate operation not allowed
/ "460" ; Only aggregate operation allowed
/ "461" ; Unsupported Transport
/ "462" ; Destination Unreachable
/ "463" ; Destination Prohibited
/ "464" ; Data Transport Not Ready Yet
/ "470" ; Connection Authorization Required
/ "471" ; Connection Credentials not accepted
/ "472" ; Failure to establish secure connection
/ "500" ; Internal Server Error
/ "501" ; Not Implemented
/ "502" ; Bad Gateway
/ "503" ; Service Unavailable
/ "504" ; Gateway Time-out
/ "505" ; RTSP Version not supported
/ "551" ; Option not supported
/ extension-code
extension-code = 3DIGIT
Reason-Phrase = *TEXT
general-header = Cache-Control
/ Connection
/ CSeq
/ Date
Schulzrinne, et al. Expires August 28, 2008 [Page 134]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
/ Pipelined-Requests
/ Proxy-Supported
/ 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
/ Proxy-Require
/ Range
/ Referer
/ Require
/ Scale
/ Session
/ Speed
/ Supported
/ Transport
/ User-Agent
/ extension-header
response-header = Accept-Credentials
/ Accept-Ranges
/ Connection-Credentials
/ ETag
/ Location
/ Proxy-Authenticate
/ Public
/ Range
/ Retry-After
/ RTP-Info
/ Scale
/ Session
/ Server
/ Speed
/ Transport
/ Unsupported
/ Vary
/ WWW-Authenticate
Schulzrinne, et al. Expires August 28, 2008 [Page 135]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
/ extension-header
entity-header = Allow
/ Content-Base
/ Content-Encoding
/ Content-Language
/ Content-Length
/ Content-Location
/ Content-Type
/ Expires
/ Last-Modified
/ extension-header
20.2.3. Header Syntax
All header syntaxes not defined in this section are defined in
section 14 of the HTTP 1.1 specification [RFC2616].
Accept = "Accept" HCOLON
[ accept-range *(COMMA accept-range) ]
accept-range = media-range *(SEMI accept-param)
media-range = ( "*/*"
/ ( m-type SLASH "*" )
/ ( m-type SLASH m-subtype )
) *( SEMI m-parameter )
accept-param = ("q" EQUAL qvalue) / generic-param
qvalue = ( "0" [ "." *3DIGIT ] )
/ ( "1" [ "." *3("0") ] )
Accept-Credentials = "Accept-Credentials" HCOLON cred-decision CRLF
cred-decision = ("User" [LWS cred-info])
/ "Proxy"
/ "Any"
/ token [LWS 1*TEXT] ; For future extensions
cred-info = cred-info-data *(COMMA cred-info-data)
cred-info-data = DQ RTSP-URI DQ SEMI hash-alg SEMI base64
hash-alg = "sha-256" / extension-alg
extension-alg = token
Accept-Encoding = "Accept-Encoding" HCOLON
[ encoding *(COMMA encoding) ]
encoding = codings *(SEMI accept-param)
codings = content-coding / "*"
content-coding = token
Accept-Language = "Accept-Language" HCOLON
[ language *(COMMA language) ]
language = language-range *(SEMI accept-param)
language-range = ( ( 1*8ALPHA *( "-" 1*8ALPHA ) ) / "*" )
Accept-Ranges = "Accept-Ranges" HCOLON acceptable-ranges CRLF
Schulzrinne, et al. Expires August 28, 2008 [Page 136]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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)
auth-scheme = token
Bandwidth = "Bandwidth" HCOLON 1*DIGIT CRLF
Blocksize = "Blocksize" HCOLON 1*DIGIT CRLF
Schulzrinne, et al. Expires August 28, 2008 [Page 137]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Cache-Control = "Cache-Control" HCOLON cache-directive CRLF
*(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 CRLF
cred-chain = DQ RTSP-URI DQ SEMI base64
Connection = "Connection" HCOLON (connection-token)
*(COMMA connection-token) CRLF
connection-token = token
Content-Base = "Content-Base" HCOLON RTSP-URI-Ref CRLF
Content-Encoding = "Content-Encoding" HCOLON
content-coding *(COMMA content-coding)
Content-Language = "Content-Language" HCOLON
language-tag *(COMMA language-tag)
language-tag = primary-tag *( "-" subtag )
primary-tag = 1*8ALPHA
subtag = 1*8ALPHA
Content-Length = "Content-Length" HCOLON 1*DIGIT
Content-Location = "Content-Location" HCOLON RTSP-URI-Ref
Content-Type = ( "Content-Type" / "c" ) HCOLON media-type
media-type = m-type SLASH m-subtype *(SEMI m-parameter)
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
Schulzrinne, et al. Expires August 28, 2008 [Page 138]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 1*DIGIT CRLF
Date = "Date" HCOLON RTSP-date
RTSP-date = rfc1123-date ; HTTP-date
rfc1123-date = wkday "," SP date1 SP time SP "GMT"
date1 = 2DIGIT SP month SP 4DIGIT
; day month year (e.g., 02 Jun 1982)
time = 2DIGIT ":" 2DIGIT ":" 2DIGIT
; 00:00:00 - 23:59:59
wkday = "Mon" / "Tue" / "Wed"
/ "Thu" / "Fri" / "Sat" / "Sun"
month = "Jan" / "Feb" / "Mar" / "Apr"
/ "May" / "Jun" / "Jul" / "Aug"
/ "Sep" / "Oct" / "Nov" / "Dec"
ETag = "ETag" HCOLON entity-tag
Expires = "Expires" HCOLON delta-seconds
From = "From" HCOLON from-spec
from-spec = ( name-addr / addr-spec ) *( SEMI from-param )
name-addr = [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec = RTSP-URI / absolute-URI
absolute-URI = < As defined in RFC 3986>
display-name = *(token LWS)/ quoted-string
from-param = tag-param / generic-param
tag-param = "tag" EQUAL token
If-Match = "If-Match" HCOLON ( "*" / entity-tag-list)
entity-tag-list = entity-tag *(COMMA entity-tag)
entity-tag = [ weak ] opaque-tag
weak = "W/"
opaque-tag = quoted-string
If-Modified-Since = "If-Modified-Since" HCOLON RTSP-date
If-None-Match = "If-None-Match" HCOLON ("*" / entity-tag-list)
Last-Modified = "Last-Modified" HCOLON RTSP-date
Location = "Location" HCOLON RTSP-URI
Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id
startup-id = 1*8DIGIT
Schulzrinne, et al. Expires August 28, 2008 [Page 139]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Proxy-Authenticate = "Proxy-Authenticate" HCOLON challenge
challenge = ("Digest" LWS digest-cln *(COMMA digest-cln))
/ other-challenge
other-challenge = auth-scheme LWS auth-param
*(COMMA auth-param)
digest-cln = realm / domain / nonce
/ opaque / stale / algorithm
/ qop-options / auth-param
realm = "realm" EQUAL realm-value
realm-value = quoted-string
domain = "domain" EQUAL LDQUOT URI
*( 1*SP URI ) RDQUOT
URI = RTSP-URI / RTSP-URI-Ref
nonce = "nonce" EQUAL nonce-value
nonce-value = quoted-string
opaque = "opaque" EQUAL quoted-string
stale = "stale" EQUAL ( "true" / "false" )
algorithm = "algorithm" EQUAL ("MD5" / "MD5-sess" / token)
qop-options = "qop" EQUAL LDQUOT qop-value
*("," qop-value) RDQUOT
qop-value = "auth" / "auth-int" / token
Proxy-Require = "Proxy-Require" HCOLON feature-tag CRLF
*(COMMA feature-tag)
Proxy-Supported = "Proxy-Supported" HCOLON feature-tag
*(COMMA feature-tag) CRLF
Public = "Public" HCOLON Method *(COMMA Method) CRLF
Range = "Range" HCOLON ranges-list [exec-time] CRLF
ranges-list = ranges-spec *(COMMA ranges-spec)
exec-time = SEMI "time" EQUAL utc-time
ranges-spec = npt-range / utc-range / smpte-range
/ range-ext
range-ext = extension-format "=" range-value
range-value = 1*(rtsp-unreserved / quoted-string / ":" )
Referer = "Referer" HCOLON RTSP-URI-Ref
Require = "Require" HCOLON feature-tag-list CRLF
feature-tag-list = feature-tag *(COMMA feature-tag)
Schulzrinne, et al. Expires August 28, 2008 [Page 140]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
RTP-Info = "RTP-Info" HCOLON rtsp-info-spec
*(COMMA rtsp-info-spec) CRLF
rtsp-info-spec = stream-url 1*ssrc-parameter
stream-url = "url" EQUAL DQ RTSP-URI-Ref DQ
ssrc-parameter = LWS "ssrc" EQUAL ssrc HCOLON
ri-parameter *(SEMI ri-parameter)
ri-parameter = "seq" EQUAL 1*DIGIT
/ "rtptime" EQUAL 1*DIGIT
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 ] CRLF
Speed = "Speed" HCOLON 1*DIGIT [ "." *DIGIT ] CRLF
Server = "Server" HCOLON ( product / comment )
*(LWS (product / comment)) CRLF
product = token [SLASH product-version]
product-version = token
comment = LPAREN *( ctext / quoted-pair) RPAREN
Session = "Session" HCOLON session-id
[ SEMI "timeout" EQUAL delta-seconds ] CRLF
Supported = "Supported" HCOLON [feature-tag-list] CRLF
Timestamp = "Timestamp" HCOLON timestamp-value LWS [delay]
timestamp-value = *DIGIT [ "." *DIGIT ]
delay = *DIGIT [ "." *DIGIT ]
Transport = "Transport" HCOLON transport-spec
*(COMMA transport-spec) CRLF
transport-spec = transport-id *tr-parameter
transport-id = trans-id-rtp / other-trans
trans-id-rtp = "RTP/" profile ["/" lower-transport]
; no LWS is allowed inside transport-id
other-trans = token *("/" token)
Schulzrinne, et al. Expires August 28, 2008 [Page 141]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
profile = "AVP" / "SAVP" / "AVPF" / token
lower-transport = "TCP" / "UDP" / token
tr-parameter = SEMI ( "unicast" / "multicast" )
/ SEMI "interleaved" EQUAL channel [ "-" channel ]
/ SEMI "append"
/ SEMI "ttl" EQUAL ttl
/ SEMI "layers" EQUAL 1*DIGIT
/ SEMI "ssrc" EQUAL ssrc *(SLASH ssrc)
/ SEMI "client_ssrc" EQUAL ssrc
/ SEMI "mode" EQUAL mode-spec
/ SEMI "dest_addr" EQUAL addr-list
/ SEMI "src_addr" EQUAL addr-list
/ SEMI trn-param-ext
/ SEMI "setup" EQUAL contrans-setup
/ SEMI "connection" EQUAL contrans-con
contrans-setup = "active" / "passive" / "actpass"
contrans-con = "new" / "existing"
trn-param-ext = par-name [EQUAL trn-par-value]
par-name = token
trn-par-value = *(rtsp-unreserved / DQ *TEXT DQ)
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>
Schulzrinne, et al. Expires August 28, 2008 [Page 142]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Unsupported = "Unsupported" HCOLON feature-tag-list CRLF
User-Agent = "User-Agent" HCOLON ( product / comment )
0*(LWS (product / comment)) CRLF
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
20.3. SDP extension Syntax
This section defines in ABNF the SDP extensions defined for RTSP.
See Appendix D for the definition of the extensions in text.
control-attribute = "a=control:" *SP RTSP-URI
a-range-def = "a=range:" ranges-spec CRLF
a-etag-def = "a=etag:" entity-tag CRLF
Schulzrinne, et al. Expires August 28, 2008 [Page 143]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
21. Security Considerations
Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security considerations outlined in [H15]
apply. Specifically, please note the following:
Abuse of Server Log Information: RTSP and HTTP servers will
presumably have similar logging mechanisms, and thus should be
equally guarded in protecting the contents of those logs, thus
protecting the privacy of the users of the servers. See
[H15.1.1] for HTTP server recommendations regarding server
logs.
Transfer of Sensitive Information: There is no reason to believe
that information transferred or controlled via RTSP may be any
less sensitive than that normally transmitted via HTTP.
Therefore, all of the precautions regarding the protection of
data privacy and user privacy apply to implementors of RTSP
clients, servers, and proxies. See [H15.1.2] for further
details.
Attacks Based On File and Path Names: Though RTSP URIs are opaque
handles that do not necessarily have file system semantics, it
is anticipated that many implementations will translate
portions of the Request-URIs directly to file system calls. In
such cases, file systems SHOULD follow the precautions outlined
in [H15.5], such as checking for ".." in path components.
Personal Information: RTSP clients are often privy to the same
information that HTTP clients are (user name, location, etc.)
and thus should be equally sensitive. See [H15.1] for further
recommendations.
Privacy Issues Connected to Accept Headers: Since may of the same
"Accept" headers exist in RTSP as in HTTP, the same caveats
outlined in [H15.1.4] with regards to their use should be
followed.
DNS Spoofing: Presumably, given the longer connection times
typically associated to RTSP sessions relative to HTTP
sessions, RTSP client DNS optimizations should be less
prevalent. Nonetheless, the recommendations provided in
[H15.3] are still relevant to any implementation which attempts
to rely on a DNS-to-IP mapping to hold beyond a single use of
the mapping.
Schulzrinne, et al. Expires August 28, 2008 [Page 144]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Location Headers and Spoofing: If a single server supports multiple
organizations that do not trust each another, then it needs to
check the values of Location and Content-Location header fields
in responses that are generated under control of said
organizations to make sure that they do not attempt to
invalidate resources over which they have no authority.
([H15.4])
In addition to the recommendations in the current HTTP specification
(RFC 2616 [RFC2616], as of this writing) and also of the previous
RFC2068 [RFC2068], future HTTP specifications may provide additional
guidance on security issues.
The following are added considerations for RTSP implementations.
Concentrated denial-of-service attack: The protocol offers the
opportunity for a remote-controlled denial-of-service attack.
See Section 21.1.
Session hijacking: Since there is no or little relation between a
transport layer connection and an RTSP session, it is possible
for a malicious client to issue requests with random session
identifiers which would affect unsuspecting clients. The
server SHOULD use a large, random and non-sequential session
identifier to minimize the possibility of this kind of attack.
However, unless the RTSP signalling always are confedentiality
protected, e.g. using TLS, an on-path attacker will be able to
hijack a session. For real session security, client
authentication needs to be performed.
Authentication: Servers SHOULD implement both basic and digest
[RFC2617] authentication. In environments requiring tighter
security for the control messages, the transport layer
mechanism TLS (RFC 4346 [RFC4346]) 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
Schulzrinne, et al. Expires August 28, 2008 [Page 145]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
local security policy.
Scope of Multicast: If RTSP is used to control the transmission of
media onto a multicast network it is need to consider the scope
that delivery has. RTSP supports the TTL Transport header
parameter to indicate this scope. However such scope control
is risk as it may be set to large and distribute media beyond
the intended scope.
TLS through proxies: If one uses the possibility to connect TLS in
multiple legs (Section 19.3 one really needs to be aware of the
trust model. That procedure requires full faith and trust in
all proxies that one allows to connect through. They are man
in the middle and has access to all that goes on over the TLS
connection. Thus it is important to consider if that trust
model is acceptable in the actual application.
21.1. Remote denial of Service Attack
The attacker may initiate traffic flows to one or more IP addresses
by specifying them as the destination in SETUP requests. While the
attacker's IP address may be known in this case, this is not always
useful in prevention of more attacks or ascertaining the attackers
identity. Thus, an RTSP server MUST only allow client-specified
destinations for RTSP-initiated traffic flows if the server has
ensured that the specified destination address accepts receiving
media through different security mechanisms. Security mechanism that
are acceptable in an increased generality are; verification of the
client's identity, either against a database of known users using
RTSP authentication mechanisms (preferably digest authentication or
stronger); a list of addresses that accept to be media destinations,
especially considering user identity; and media path based
verification.
The server SHOULD NOT allow the destination field to be set unless a
mechanism exists in the system to authorize the request originator to
direct streams to the recipient. It is preferred that this
authorization be performed by the media recipient (destination)
itself and the credentials passed along to the server. However, in
certain cases, such as when recipient address is a multicast group,
or when the recipient is unable to communicate with the server in an
out-of-band manner, this may not be possible. In these cases server
may chose another method such as a server-resident authorization list
to ensure that the request originator has the proper credentials to
request stream delivery to the recipient.
One solution that performs the necessary verification of acceptance
of media suitable for unicast based delivery is the ICE based NAT
Schulzrinne, et al. Expires August 28, 2008 [Page 146]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 147]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
22. IANA Considerations
This section sets up a number of registries for RTSP 2.0 that should
be maintained by IANA. For each registry there is a description on
what it is required to contain, what specification is needed when
adding a entry with IANA, and finally the entries that this document
needs to register. See also the Section 2.3 "Extending RTSP". There
is also an IANA registration of two SDP attributes.
The sections describing how to register an item uses some of the
requirements level described in RFC 2434 [RFC2434], namely "First
Come, First Served", "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 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.
Schulzrinne, et al. Expires August 28, 2008 [Page 148]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
22.1.2. Registering New Feature-tags with IANA
The registering of feature-tags is done on a first come, first served
basis.
The name of the feature MUST follow these rules: The name may be of
any length, but SHOULD be no more than twenty characters long. The
name MUST NOT contain any spaces, or control characters. The
registration SHALL indicate if the feature-tag applies to clients,
servers, or proxies only or any combinations of these. Any
proprietary feature SHALL have as the first part of the name a vendor
tag, which identifies the organization.
22.1.3. Registered entries
The following feature-tags are in this specification defined and
hereby registered. The change control belongs to the IETF.
play.basic: The minimal implementation for playback operations
according to Appendix E. 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 standard track
document. 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,
Schulzrinne, et al. Expires August 28, 2008 [Page 149]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 9 methods: DESCRIBE,
GET_PARAMETER, OPTIONS, PAUSE, PLAY, REDIRECT, SETUP, SETPARAMETER,
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 track
document. A specification for a new status code MUST specify the
following:
o The requested number.
o A description what the status code means and the expected behavior
of the sender and receiver of the code.
22.3.3. Registered Entries
RFCXXX, registers the numbered status code defined in the ABNF entry
"Status-Code" except "extension-code" in Section 20.2.2.
22.4. RTSP Headers
22.4.1. Description
By specifying new headers a method(s) can be enhanced in many
different ways. An unknown header will be ignored by the receiving
entity. If the new header is vital for a certain functionality, a
feature-tag for the functionality can be created and demanded to be
used by the counter-part with the inclusion of a Require header
carrying the feature-tag.
Schulzrinne, et al. Expires August 28, 2008 [Page 150]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
22.4.2. Registering New Headers with IANA
A public available specification is required to register a header.
The specification SHOULD be a standards document, preferable an IETF
RFC.
The specification MUST 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].
o 3GPP-QoE-Metrics defined in [3gpp-26234].
Schulzrinne, et al. Expires August 28, 2008 [Page 151]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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. Transport Header Registries
The transport header contains a number of parameters which have
possibilities for future extensions. Therefore registries for these
needs to be defined.
22.5.1. Transport Protocol Specification
A registry for the parameter transport-protocol specification SHALL
be defined with the following rules:
o Registering require an public available standards specification.
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 152]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
RTP/SAVP/UDP: the same as RTP/SAVP.
RTP/SAVPF: Use of the RTP[RFC3550] protocol for media transport in
combination with the "[I-D.ietf-avt-profile-savpf] over UDP.
The usage is explained in RFC XXXX, appendix Appendix C.1.
RTP/SAVPF/UDP: the same as RTP/SAVPF.
RTP/AVP/TCP: Use of the RTP[RFC3550] protocol for media transport in
combination with the "RTP profile for audio and video
conferences with minimal control"[RFC3551] over TCP. The usage
is explained in RFC XXXX, appendix Appendix C.2.2.
RTP/AVPF/TCP: Use of the RTP[RFC3550] protocol for media transport
in combination with the "Extended RTP Profile for RTCP-based
Feedback (RTP/AVPF)"[RFC4585] over TCP. The usage is explained
in RFC XXXX, appendix Appendix C.2.2.
RTP/SAVP/TCP: Use of the RTP[RFC3550] protocol for media transport
in combination with the "The Secure Real-time Transport
Protocol (SRTP)" [RFC3711] over TCP. The usage is explained in
RFC XXXX, appendix Appendix C.2.2.
RTP/SAVPF/TCP: Use of the RTP[RFC3550] protocol for media transport
in combination with the "[I-D.ietf-avt-profile-savpf] over TCP.
The usage is explained in RFC XXXX, appendix Appendix C.2.2.
22.5.2. Transport modes
A registry for the transport parameter mode SHALL be defined with the
following rules:
o Registering requires an IETF standard tracks document.
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 153]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
22.5.3. Transport Parameters
A registry for parameters that may be included in the Transport
header SHALL be defined with the following rules:
o Registering required a Open Standards document.
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 16.46.
22.6. Cache Directive Extensions
There exist a number of cache directives which can be sent in the
Cache-Control header. A registry for this cache directives SHALL be
defined with the following rules:
o Registering requires an IETF standard tracks document.
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:
Schulzrinne, et al. Expires August 28, 2008 [Page 154]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
min-fresh:
must-revalidate:
proxy-revalidate:
max-age:
22.7. Accept-Credentials
The security framework's TLS connection mechanism has two
registerable entities.
22.7.1. Accept-Credentials policies
In Section 19.3.1 three policies for how to handle certificates.
Further policies may be defined and SHALL be registered with IANA
using the following rules:
o Registering requires an IETF standard tracks document.
o A registration is required 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
22.7.2. Accept-Credentials hash algorithms
The Accept-Credentials header (See Section 16.2) allows for the usage
of other algorithms for hashing the DER records of accepted entities.
The registration of any future algorithm is expected to be extremely
rare and could also be an interoperability problem. Therefore the
XXX bare for registering new algorithms is placed intentional high.
Any registration of a new hash algorithm SHALL meet the following
requirement:
Schulzrinne, et al. Expires August 28, 2008 [Page 155]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
o Registration requires an IETF standard track document.
o A definition of the algorithm and its identifier meeting the
"token" ABNF requirement.
22.8. Range header formats
The Range header allows for different range formats. New ones may be
registered, but moderation should be applied as it makes
interoperability more difficult. A registration SHALL fulfill the
following requirements:
o A publicly available standards document.
o A ABNF definition of the range format that fulfils the "range-ext"
definition.
o A Contact person for the registration.
o Rules for how one handles the range when using a negative Scale.
22.9. URI Schemes
This specification defines two URI schemes ("rtsp" and "rtsps") and
reserves a third one ("rtspu"). Registrations are following RFC
4395[RFC4395].
22.9.1. 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 (SETPARAMETER and
GET_PARAMETER, and termination of the session context created
(TEARDOWN).
Schulzrinne, et al. Expires August 28, 2008 [Page 156]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 (XXX).
Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC
2326), RTSP 2.0 (RFC XXXX)
Interoperability considerations: The change in URI syntax performed
between RTSP 1.0 and 2.0 can create interoperability issues.
Security considerations: All the security threats identified in
Section 7 of RFC 3986 applies also to this scheme. They needs
to be reviewed and considered in any implementation utilizing
this scheme.
Contact: Magnus Westerlund, magnus.westerlund@ericsson.com
Author/Change controller: IETF MMUSIC WG
References: RFC 2326, RFC 3986, RFC 3987, RFC XXXX
22.9.2. 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 (SETPARAMETER 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.
Schulzrinne, et al. Expires August 28, 2008 [Page 157]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC
2326), RTSP 2.0 (RFC XXXX)
Interoperability considerations: The change in URI syntax performed
between RTSP 1.0 and 2.0 can create interoperability issues.
Security considerations: All the security threats identified in
Section 7 of RFC 3986 applies also to this scheme. They needs
to be reviewed and considered in any implementation utilizing
this scheme.
Contact: Magnus Westerlund, magnus.westerlund@ericsson.com
Author/Change controller: IETF MMUSIC WG
References: RFC 2326, RFC 3986, RFC 3987, RFC XXXX
22.9.3. The rtspu URI Scheme
URI scheme name: rtspu
Status: Permanent
URI scheme syntax: See Section 3.2 of RFC 2326.
URI scheme semantics: The rtspu scheme is used to indicate resources
accessible through the usage of the Real-time Streaming
Protocol (RTSP) over unrelaible datagram transport. RTSP
allows different operations on the resource identified by the
URI, but the primary purpose is the streaming delivery of the
resource to a client. However the operations that are
currently defined are: Describing the resource for the purpose
of configuring the receiving entity (DESCRIBE), configuring the
delivery method and its addressing (SETUP), controlling the
delivery (PLAY and PAUSE), reading or setting of resource
related parameters (SETPARAMETER 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)
Schulzrinne, et al. Expires August 28, 2008 [Page 158]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 MMUSIC WG
References: RFC 2326, RFC 3986, RFC 3987
22.10. SDP attributes
This specification defines two SDP [RFC4566] attributes that it is
requested that IANA register.
SDP Attribute ("att-field"):
Attribute name: range
Long form: Media Range Attribute
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: See ABNF definition.
Attribute name: control
Long form: RTSP control URI
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: Absolute or Relative URIs.
Attribute name: etag
Long form: Entity Tag
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: See ABNF definition
Schulzrinne, et al. Expires August 28, 2008 [Page 159]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
23. References
23.1. Normative References
[3gpp-26234]
Third Generation Partnership Project (3GPP), "Transparent
end-to-end Packet-switched Streaming Service (PSS);
Protocols and codecs; Technical Specification 26.234",
December 2002.
[FIPS-pub-180-2]
National Institute of Standards and Technology (NIST),
"Federal Information Processing Standards Publications
(FIPS PUBS) 180-2: Secure Hash Standard", Augusti 2002.
[I-D.ietf-avt-profile-savpf]
Ott, J. and E. Carrara, "Extended Secure RTP Profile for
RTCP-based Feedback (RTP/SAVPF)",
draft-ietf-avt-profile-savpf-12 (work in progress),
November 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.
[RFC2068] Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1",
RFC 2068, January 1997.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2434] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", BCP 26, RFC 2434,
October 1998.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
[RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A., and L. Stewart, "HTTP
Authentication: Basic and Digest Access Authentication",
RFC 2617, June 1999.
[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.
Schulzrinne, et al. Expires August 28, 2008 [Page 160]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
[RFC3280] Housley, R., Polk, W., Ford, W., and D. Solo, "Internet
X.509 Public Key Infrastructure Certificate and
Certificate Revocation List (CRL) Profile", RFC 3280,
April 2002.
[RFC3513] Hinden, R. and S. Deering, "Internet Protocol Version 6
(IPv6) Addressing Architecture", RFC 3513, April 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, November 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, January 2005.
[RFC3987] Duerst, M. and M. Suignard, "Internationalized Resource
Identifiers (IRIs)", RFC 3987, January 2005.
[RFC4086] Eastlake, D., Schiller, J., and S. Crocker, "Randomness
Requirements for Security", BCP 106, RFC 4086, June 2005.
[RFC4234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 4234, October 2005.
[RFC4346] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.1", RFC 4346, April 2006.
[RFC4395] Hansen, T., Hardie, T., and L. Masinter, "Guidelines and
Registration Procedures for New URI Schemes", BCP 115,
RFC 4395, February 2006.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Schulzrinne, et al. Expires August 28, 2008 [Page 161]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Description Protocol", RFC 4566, July 2006.
[RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
Carrara, "Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming
Protocol (RTSP)", RFC 4567, July 2006.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
[RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data
Encodings", RFC 4648, October 2006.
23.2. Informative References
[I-D.ietf-mmusic-rtsp-nat]
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-05 (work in progress),
July 2007.
[ISO.13818-1.2000]
International Organization for Standardization,
"Information technology - Generic coding of moving
pictures and associated audio information: Systems", ISO/
IEC 13818-1:2000, December 2000.
[ISO.13818-6.1995]
International Organization for Standardization,
"Information technology - Generic coding of moving
pictures and associated audio information - part 6:
Extension for digital storage media and control",
ISO Draft Standard 13818-6, November 1995.
[ISO.8601.2000]
International Organization for Standardization, "Data
elements and interchange formats - Information interchange
- Representation of dates and times", ISO/IEC Standard
8601, December 2000.
[ITU.H323.1996]
Schulzrinne, et al. Expires August 28, 2008 [Page 162]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
International Telecommunications Union, "Visual telephone
systems and equipment for local area networks which
provide a non-guaranteed quality of service", ITU-
T Recommendation H.323, May 1996.
[NOSSDAV-1997-1]
Schulzrinne, H., "A comprehensive multimedia control
architecture for the Internet", May 1997.
[RFC1123] Braden, R., "Requirements for Internet Hosts - Application
and Support", STD 3, RFC 1123, October 1989.
[RFC1305] Mills, D., "Network Time Protocol (Version 3)
Specification, Implementation", RFC 1305, March 1992.
[RFC1644] Braden, B., "T/TCP -- TCP Extensions for Transactions
Functional Specification", RFC 1644, July 1994.
[RFC1961] McMahon, P., "GSS-API Authentication Method for SOCKS
Version 5", RFC 1961, June 1996.
[RFC2070] Yergeau, F., Nicol, G., Adams, G., and M. Duerst,
"Internationalization of the Hypertext Markup Language",
RFC 2070, January 1997.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326, April 1998.
[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.
[W3C.REC-PICS-labels]
Miller, J., Krauskopf, T., Resnick, P., and W. Treese,
"PICS label distribution label syntax and communication
protocols", W3C REC-PICS-labels-961031.
[W3C.REC-PICS-services]
Miller, J., Resnick, P., and D. Singer, "Rating services
Schulzrinne, et al. Expires August 28, 2008 [Page 163]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
and rating systems (and their machine readable
descriptions)", W3C REC-PICS-services-961031,
October 1996.
Schulzrinne, et al. Expires August 28, 2008 [Page 164]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix A. Examples
This section contains several different examples trying to illustrate
possible ways of using RTSP. The examples can also help with the
understanding of how functions of RTSP work. However remember that
this is examples and the normative and syntax description in the
other sections takes precedence. Please also note that many of the
example contain syntax illegal line breaks to accommodate the
formatting restriction that the RFC series impose.
A.1. Media on Demand (Unicast)
The is an example of media on demand streaming of a media stored in a
container file. For purposes of this example, a container file is a
storage entity in which multiple continuous media types pertaining to
the same end-user presentation are present. In effect, the container
file represents an RTSP presentation, with each of its components
being RTSP controlled media streams. Container files are a widely
used means to store such presentations. While the components are
transported as independent streams, it is desirable to maintain a
common context for those streams at the server end.
This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case
of any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent
selective retrieval of the streams by the client in order to preserve
the artistic effect of the combined media presentation. Similarly,
in such a tightly bound presentation, it is desirable to be able to
control all the streams via a single control message using an
aggregate URI.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URIs. In
a container file it is also desirable to not write any URI parts
which is not kept, when the container is distributed, like the host
and most of the path element. Therefore this example also uses the
"*" and relative URI in the delivered SDP.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to
the container file.
Schulzrinne, et al. Expires August 28, 2008 [Page 165]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 257
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
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
Schulzrinne, et al. Expires August 28, 2008 [Page 166]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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;dest_addr=":8000"/":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
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;dest_addr=":8002"/":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
Schulzrinne, et al. Expires August 28, 2008 [Page 167]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=0-10, npt=30-
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=0-10, npt=30-623.10
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1";
ssrc=4F312DD8:seq=54321;rtptime=2876889
C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 5
User-Agent: PhonyClient/1.2
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 5
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:36:01 GMT
Session: 12345678
Range: npt=34.57-623.10
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 6
User-Agent: PhonyClient/1.2
Range: npt=34.57-623.10
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 6
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:36:01 GMT
Session: 12345678
Range: npt=34.57-623.10
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12555;rtptime=6330012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=55021;rtptime=3132889
Schulzrinne, et al. Expires August 28, 2008 [Page 168]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
A.2. Media on Demand using Pipelining
This example is basically the example above (Appendix A.1), but now
utilizing pipelining to speed up the setup. It requires only two
round trip times until the media starts flowing. First of all, the
session description is retrieved to determine what media resources
need to be setup. In the second step, one sends the necessary SETUP
requests and the PLAY request to initiate media delivery.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to
the container file.
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 257
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
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
Schulzrinne, et al. Expires August 28, 2008 [Page 169]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
Accept-Ranges: NPT, SMPTE, UTC
Pipelined-Requests: 7654
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=0-10, npt=30-
Session: 12345678
Pipelined-Requests: 7654
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001;
src_addr="192.0.2.5:9000"/"192.0.2.5:9001"
ssrc=93CB001E
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 23 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT
Pipelined-Requests: 7654
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003;
src_addr="192.0.2.5:9002"/"192.0.2.5:9003";
ssrc=A813FC13
Session: 12345678
Expires: 24 Jan 1997 15:35:13 GMT
Date: 23 Jan 1997 15:35:13 GMT
Accept-Range: NPT
Pipelined-Requests: 7654
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=0-10, npt=30-623.10
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1";
ssrc=4F312DD8:seq=54321;rtptime=2876889
Schulzrinne, et al. Expires August 28, 2008 [Page 170]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 171]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.0 200 OK
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 264
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
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=":3056"/":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
Schulzrinne, et al. Expires August 28, 2008 [Page 172]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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=":3058"/":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=":3058"/":3059";
src_addr="192.0.2.5:5002"/"192.0.2.5:5003"
Date: 23 Jan 1997 15:35:12 GMT
Server: PhonyServer/1.0
Cache-Control: public
Expires: 24 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT, SMPTE
C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Session: 23456789
Range: smpte=0:10:00-
V->C: RTSP/2.0 200 OK
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-1:49:23
RTP-Info: url="rtsp://video.example.com/twister/video"
ssrc=A17E189D:seq=12312232;rtptime=78712811
Server: PhonyServer/2.0
Date: 23 Jan 1997 15:35:13 GMT
Schulzrinne, et al. Expires August 28, 2008 [Page 173]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Session: 12345678
Range: smpte=0:10:00-
A->C: RTSP/2.0 200 OK
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-1:49:23
RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
ssrc=3D124F01:seq=876655;rtptime=1032181
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:13 GMT
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 12345678
A->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:36:52 GMT
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 23456789
V->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/2.0
Date: 23 Jan 1997 15:36:52 GMT
Even though the audio and video track are on two different servers,
may start at slightly different times, and may drift with respect to
each other, the client can perform initial synchronize of the two
media using RTP-Info and Range received in the PLAY responses. If
the two servers are time synchronized the RTCP packets can also be
used to maintain synchronization.
Schulzrinne, et al. Expires August 28, 2008 [Page 174]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 148
Server: PhonyServer/1.0
Date: 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
t=0 0
a=control: *
m=audio 0 RTP/AVP 0
a=control:streamid=0
Schulzrinne, et al. Expires August 28, 2008 [Page 175]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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=":6970"/":6971";
src_addr="192.0.2.5:6970"/"192.0.2.5:6971";
mode="PLAY";ssrc=EAB98712
CSeq: 2
Session: 2034820394
Expires: 23 Jan 1997 16:00:00 GMT
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:07 GMT
Accept-Ranges: NPT
C->S: PLAY rtsp://foo.com/test.wav/ RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 2034820394
S->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:08 GMT
Session: 2034820394
Range: npt=0-600
RTP-Info: url="rtsp://foo.com/test.wav/streamid=0"
ssrc=0D12F123:seq=981888;rtptime=3781123
Note the different URI in the SETUP command, and then the switch back
to the aggregate URI in the PLAY command. This makes complete sense
when there are multiple streams with aggregate control, but is less
than intuitive in the special case where the number of streams is
one. However the server has declared that the aggregated control URI
in the SDP and therefore this is legal.
In this case, it is also required that servers accept implementations
that use the non-aggregated interpretation and use the individual
media URI, like this:
C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Schulzrinne, et al. Expires August 28, 2008 [Page 176]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 182
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:06 GMT
Supported: play.basic
v=0
o=- 2890844526 2890842807 IN IP4 192.0.2.5
s=RTSP Session
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-
Schulzrinne, et al. Expires August 28, 2008 [Page 177]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 23 Jan 1997 15:35:06 GMT
Transport: RTP/AVP;multicast;dest_addr="224.2.0.1:3456"/"
224.2.0.1:3457";ttl=16
Session: 0456804596
Accept-Ranges: NPT, UTC
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 3
Session: 0456804596
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:07 GMT
Session: 0456804596
Range:npt=1256-
RTP-Info: url="rtsp://live.example.com/concert/audio"
ssrc=0D12F123:seq=1473; rtptime=80000
A.6. Capability Negotiation
This examples illustrate how the client and server determines their
capability to support a special feature, in this case "play.scale".
The server, through the clients request and the included Supported
header, learns the client supports RTSP 2.0, and also supports the
playback time scaling feature of RTSP. The server's response
contains the following feature related information to the client; it
supports the basic playback (play.basic), the extended functionality
of time scaling of content (play.scale), and one "example.com"
proprietary feature (com.example.flight). The client also learns the
methods supported (Public header) by the server for the indicated
resource.
Schulzrinne, et al. Expires August 28, 2008 [Page 178]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0
CSeq: 1
Supported: play.basic, play.scale
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 1
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
Server: PhonyServer/2.0
Supported: play.basic, play.scale, com.example.flight
When the client sends its SETUP request it tells the server that it
is requires support of the play.scale feature for this session by
including the Require header.
C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":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=":3056"/":3057";
src_addr="192.0.2.5:5000"/"192.0.2.5:5001"
Server: PhonyServer/2.0
Accept-Ranges: NPT, SMPTE
Schulzrinne, et al. Expires August 28, 2008 [Page 179]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix B. RTSP Protocol State Machine
The RTSP session state machine describes the behavior of the protocol
from RTSP session initialization through RTSP session termination.
The State machine is defined on a per session basis which is uniquely
identified by the RTSP session identifier. The session may contain
one or more media streams depending on state. If a single media
stream is part of the session it is in non-aggregated control. If
two or more is part of the session it is in aggregated control.
The below state machine is a normative description of the protocols
behavior. However, in case of ambiguity with the earlier parts of
this specification, the description in the earlier parts SHALL take
precedence.
B.1. States
The state machine contains three states, described below. For each
state there exist a table which shows which requests and events that
is allowed and if they will result in a state change.
Init: Initial state no session exist.
Ready: Session is ready to start playing.
Play: Session is playing, i.e. sending media stream data in the
direction S->C.
B.2. State variables
This representation of the state machine needs more than its state to
work. A small number of variables are also needed and is explained
below.
NRM: The number of media streams part of this session.
RP: Resume point, the point in the presentation time line at which
a request to continue will resume from. A time format for the
variable is not mandated.
B.3. Abbreviations
To make the state tables more compact a number of abbreviations are
used, which are explained below.
Schulzrinne, et al. Expires August 28, 2008 [Page 180]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
IFI: IF Implemented.
md: Media
PP: Pause Point, the point in the presentation time line at which
the presentation was paused.
Prs: Presentation, the complete multimedia presentation.
RedP: Redirect Point, the point in the presentation time line at
which a REDIRECT was specified to occur.
SES: Session.
B.4. State Tables
This section contains a table for each state. The table contains all
the requests and events that this state is allowed to act on. The
events which is method names are, unless noted, requests with the
given method in the direction client to server (C->S). In some cases
there exist one or more requisite. The response column tells what
type of response actions should be performed. Possible actions that
is requested for an event includes: response codes, e.g. 200, headers
that MUST be included in the response, setting of state variables, or
setting of other session related parameters. The new state column
tells which state the state machine changes to.
The response to valid request meeting the requisites is normally a
2xx (SUCCESS) unless other noted in the response column. The
exceptions needs to be given a response according to the response
column. If the request does not meet the requisite, is erroneous or
some other type of error occur the appropriate response code MUST be
sent. If the response code is a 4xx the session state is unchanged.
A response code of 3rr will result in that the session is ended and
its state is changed to Init. A response code of 304 results in no
state change. However there exist restrictions to when a 3rr
response may be used. A 5xx response SHALL not result in any change
of the session state, except if the error is not possible to recover
from. A unrecoverable error SHALL result the ending of the session.
As it in the general case can't be determined if it was a
unrecoverable error or not the client will be required to test. In
the case that the next request after a 5xx is responded with 454
(Session Not Found) the client knows that the session has ended.
The server will timeout the session after the period of time
specified in the SETUP response, if no activity from the client is
detected. Therefore there exist a timeout event for all states
except Init.
Schulzrinne, et al. Expires August 28, 2008 [Page 181]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 |
| | | |
| SETPARAMETER | 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.
Schulzrinne, et al. Expires August 28, 2008 [Page 182]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
+--------------+-----------------+-----------+----------------------+
| Action | Requisite | New State | Response |
+--------------+-----------------+-----------+----------------------+
| SETUP | New URI | Ready | NRM +=1 |
| | | | |
| SETUP | URI Setup prior | Ready | Change transport |
| | | | param |
| | | | |
| TEARDOWN | Prs URI, | Init | No session hdr, NRM |
| | | | = 0 |
| | | | |
| TEARDOWN | md URI,NRM=1 | Init | No Session hdr, NRM |
| | | | = 0 |
| | | | |
| TEARDOWN | md URI,NRM>1 | Ready | Session hdr, NRM -= |
| | | | 1 |
| | | | |
| PLAY | Prs URI, No | Play | Play from RP |
| | range | | |
| | | | |
| PLAY | Prs URI, Range | Play | According to range |
| | | | |
| PAUSE | Prs URI | Ready | Return PP |
| | | | |
| SC:REDIRECT | Range hdr | Ready | Set RedP |
| | | | |
| SC:REDIRECT | no range hdr | Init | Session is removed |
| | | | |
| Timeout | | Init | |
| | | | |
| RedP reached | | Init | TEARDOWN of session |
+--------------+-----------------+-----------+----------------------+
Table 15: State: Ready
In the Ready state, see Table 15, some of the actions are depending
on the number of media streams (NRM) in the session, i.e. aggregated
or non-aggregated control. A setup request in the ready state can
either add one more media stream to the session or if the media
stream (same URI) already is part of the session change the transport
parameters. TEARDOWN is depending on both the Request-URI and the
number of media stream within the session. If the Request-URI is the
presentations URI the whole session is torn down. If a media URI is
used in the TEARDOWN request and more than one media exist in the
session, the session will remain and a session header MUST be
returned in the response. If only a single media stream remains in
the session when performing a TEARDOWN with a media URI the session
is removed. The number of media streams remaining after tearing down
Schulzrinne, et al. Expires August 28, 2008 [Page 183]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 |
| | | | |
| 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
The Play state table, see Table 16, is the largest. The table
contains an number of requests that has presentation URI as a
Schulzrinne, et al. Expires August 28, 2008 [Page 184]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
Schulzrinne, et al. Expires August 28, 2008 [Page 185]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix C. Media Transport Alternatives
This section defines how certain combinations of protocols, profiles
and lower transports are used. This includes the usage of the
Transport header's source and destination address parameters
"src_addr" and "dest_addr".
C.1. RTP
This section defines the interaction of RTSP with respect to the RTP
protocol [RFC3550]. It also defines any necessary media transport
signalling with regards to RTP.
The available RTP profiles and lower layer transports are described
below along with rules on signalling the available combinations.
C.1.1. AVP
The usage of the "RTP Profile for Audio and Video Conferences with
Minimal Control" [RFC3551] when using RTP for media transport over
different lower layer transport protocols is defined below in regards
to RTSP.
One such case is defined within this document, the use of embedded
(interleaved) binary data as defined in Section 14. The usage of
this method is indicated by include the "interleaved" parameter.
When using embedded binary data the "src_addr" and "dest_addr" SHALL
NOT be used. This addressing and multiplexing is used as defined
with use of channel numbers and the interleaved parameter.
C.1.2. AVP/UDP
This part describes sending of RTP [RFC3550] over lower transport
layer UDP [RFC0768] according to the profile "RTP Profile for Audio
and Video Conferences with Minimal Control" defined in RFC 3551
[RFC3551]. This profiles requires one or two uni- or bi-directional
UDP flows per media stream. The first UDP flow is for RTP and the
second is for RTCP. Embedding of RTP data with the RTSP messages, in
accordance with Section 14, SHOULD NOT be performed when RTSP
messages are transported over unreliable transport protocols, like
UDP [RFC0768].
The RTP/UDP and RTCP/UDP flows can be established using the Transport
header's "src_addr", and "dest_addr" parameters.
In RTSP PLAY mode, the transmission of RTP packets from client to
server is unspecified. The behavior in regards to such RTP packets
Schulzrinne, et al. Expires August 28, 2008 [Page 186]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
MAY be defined in future.
The "src_addr" and "dest_addr" parameters are used in the following
way for media playback, i.e. Mode=PLAY:
o The "src_addr" and "dest_addr" parameters MUST contain either 1 or
2 address specifications.
o Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST
contain either:
* both an address and a port number, or
* a port number without an address.
o The first address and port pair given in either of the parameters
applies to the RTP stream. The second address and port pair if
present applies to the RTCP stream.
o The RTP/UDP packets from the server to the client SHALL be sent to
the address and port given by first address and port pair of the
"dest_addr" parameter.
o The RTCP/UDP packets from the server to the client SHALL be sent
to the address and port given by the second address and port pair
of the "dest_addr" parameter. If no second pair is specified RTCP
SHALL NOT be sent.
o The RTCP/UDP packets from the client to the server SHALL be sent
to the address and port given by the second address and port pair
of the "src_addr" parameter. If no second pair is given RTCP
SHALL NOT be sent.
o The RTP/UDP packets from the client to the server SHALL be sent to
the address and port given by the first address and port pair of
the "src_addr" parameter.
o RTP and RTCP Packets SHOULD be sent from the corresponding
receiver port, i.e. RTCP packets from server should be sent from
the "src_addr" parameters second address port pair.
C.1.3. AVPF/UDP
The RTP profile "Extended RTP Profile for RTCP-based Feedback (RTP/
AVPF)"[RFC4585] MAY be used as RTP profiles in session using RTP.
All that is defined for AVP SHALL also apply for AVPF.
The usage of AVPF is indicated by the media initialization protocol
Schulzrinne, et al. Expires August 28, 2008 [Page 187]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
used. In the case of SDP it is indicated by media lines (m=)
containing the profile RTP/AVPF. That SDP MAY also contain further
AVPF related SDP attributes configuring the AVPF session regarding
reporting interval and feedback messages that shall be used that
SHALL be followed.
C.1.4. SAVP/UDP
The RTP profile "The Secure Real-time Transport Protocol (SRTP)"
[RFC3711] is an RTP profile (SAVP) that MAY be used in RTSP sessions
using RTP. All that is defined for AVP SHALL also apply for SAVP.
The usage of SRTP requires that a security association is
established. The RECOMMENDED mechanism for establishing that
security association is to use MIKEY with RTSP as defined in RFC 4567
[RFC4567].
C.1.5. SAVPF/UDP
The RTP profile "Extended Secure RTP Profile for RTCP-based Feedback
(RTP/SAVPF)" [I-D.ietf-avt-profile-savpf] is an RTP profile (SAVPF)
that MAY be used in RTSP sessions using RTP. All that is defined for
AVP SHALL also apply for SAVPF.
The usage of SRTP requires that a security association is
established. The RECOMMENDED mechanism for establishing that
security association is to use MIKEY[RFC3830] with RTSP as defined in
RFC 4567 [RFC4567].
C.1.6. RTCP usage with RTSP
RTCP has several usages when RTP is used for media transport as
explained below. Due to that RTCP SHALL be supported if an RTSP
agent handles RTP.
C.1.6.1. Media synchronization
RTCP provides media synchronization and clock drift compensation.
The first is available from RTP-Info header to accomplish the initial
synchronization. But to be able to handle any clockdrift between the
media streams, RTCP is needed.
C.1.6.2. RTSP Session keep-alive
RTCP traffic from the RTSP client to the RTSP server SHALL function
as keep-alive. Which requires an RTSP server supporting RTP to use
the received RTCP packets as indications that the client desires the
related RTSP session to be kept alive.
Schulzrinne, et al. Expires August 28, 2008 [Page 188]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C.1.6.3. Bit-rate adapation
RTCP Receiver Rerports and any additional feedback from the client
SHALL be used adapt the bit-rate used over the transport for all
cases when RTP is sent over UDP. A RTP sender without reserved
resources SHALL NOT use more than its fair share of the available
resources. This can be determined by comparing on short to medium
term (some seconds) the used bit-rate and adapt it so that the RTP
sender sends at a bit-rate comparable to what a TCP sender would
achieve on average over the same path.
C.2. RTP over TCP
Transport of RTP over TCP can be done in two ways, over independent
TCP connections using RFC 4571 [RFC4571] or interleaved in the RTSP
control connection. In both cases the protocol SHALL be "rtp" and
the lower layer SHALL be TCP. The profile may be any of the above
specified ones; AVP, AVPF, SAVP or SAVPF.
C.2.1. Interleaved RTP over TCP
The use of embedded (interleaved) binary data transported on the RTSP
connection is possible as specified in Section 14. When using this
declared combination of interleaved binary data the RTSP messages
MUST be transported over TCP. TLS may or may not be used.
One should however consider that this will result that all media
streams go through any proxy. Using independent TCP connections can
avoid that issue.
C.2.2. RTP over independent TCP
In this Appendix, we describe the sending of RTP [RFC3550] over lower
transport layer TCP [RFC0793] according to "Framing Real-time
Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over
Connection-Oriented Transport" [RFC4571]. This Appendix adapts the
guidelines for using RTP over TCP within SIP/SDP [RFC4145] to work
with RTSP.
A client codes the support of RTP over independent TCP by specifying
an RTP/AVP/TCP transport option without an interleaved parameter in
the Transport line of a SETUP request. This transport option MUST
include the "unicast" parameter.
If the client wishes to use RTP with RTCP, two ports (or two address/
port pairs) are specified by the dest_addr parameter. If the client
wishes to use RTP without RTCP, one port (or one address/port pair)
is specified by the dest_addr parameter. Ordering rules of dest_addr
Schulzrinne, et al. Expires August 28, 2008 [Page 189]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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 16.46) on
the Transport line to be "active", or it MAY omit the setup
parameter, as active is the default. If the client signals the
active role, the ports for all dest_addr values MUST be set to 9 (the
discard port).
If the client wishes to play the passive role in TCP connection
initiation, it MUST set the "setup" parameter on the Transport line
to be "passive". If the client is able to assume the active or the
passive role, it MUST set the "setup" parameter on the Transport line
to be "actpass". In either case, the dest_addr port value for RTP
MUST be set to the TCP port number on which the client is expecting
to receive the RTP stream connection, and the dest_addr port value
for RTCP MUST be set to the TCP port number on which the client is
expecting to receive the RTCP stream connection.
If upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, a
server decides to accept this requested option, the 2xx reply MUST
contain a Transport option that specifies RTP/AVP/TCP (without using
the interleaved parameter, and with using the unicast parameter).
The dest_addr parameter value MUST be echoed from the parameter value
in the client request unless the destination address (only port) was
not provided in which can the server MAY include the source address
of the RTSP TCP connection with the port number unchanged.
In addition, the server reply MUST set the setup parameter on the
Transport line, to indicate the role the server will play in the
connection setup. Permissible values are "active" (if a client set
"setup" to "passive" or "actpass") and "passive" (if a client set
"setup" to "active" or "actpass").
If a server sets "setup" to "passive", the "src_addr" in the reply
MUST indicate the ports the server is willing to receive an RTP
connection and (if the client requested an RTCP connection by
specifying two dest_addr ports or address/port pairs) and RTCP
connection. If a server sets "setup" to "active", the ports
specified in "src_addr" MUST be set to 9. The server MAY use the
"ssrc" parameter, following the guidance in Section 16.46. 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-
Schulzrinne, et al. Expires August 28, 2008 [Page 190]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
interleaved RTP/AVP/TCP media stream, the active party SHOULD
initiate the TCP connection as soon as possible. The client SHALL
NOT send a PLAY request prior to the establishment of all the TCP
connections negotiated using SETUP for the session. In case the
server receives a PLAY request in a session that has not yet
established all the TCP connections, it SHALL respond using the 464
"Data Transport Not Ready Yet" (Section 15.4.16) error code.
Once the PLAY request for a media resource transported over non-
interleaved RTP/AVP/TCP occurs, media begins to flow from server to
client over the RTP TCP connection, and RTCP packets flow
bidirectionally over the RTCP TCP connection. As in the RTP/UDP
case, client to server traffic on the TCP port is unspecified by this
memo. The packets that travel on these connections SHALL be framed
using the protocol defined in [RFC4571], not by the framing defined
for interleaving RTP over the RTSP control connection defined in
Section 14.
A successful PAUSE request for a media being transported over RTP/
AVP/TCP pauses the flow of packets over the connections, without
closing the connections. A successful TEARDOWN request signals that
the TCP connections for RTP and RTCP are to be closed as soon as
possible.
Subsequent SETUP requests on an already-SETUP RTP/AVP/TCP URI may be
ambiguous in the following way: does the client wish to open up new
TCP RTP and RTCP connections for the URI, or does the client wish to
continue using the existing TCP RTP and RTCP connections? The client
SHOULD use the "connection" parameter (defined in Section 16.46) 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:
Schulzrinne, et al. Expires August 28, 2008 [Page 191]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 257
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
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
Schulzrinne, et al. Expires August 28, 2008 [Page 192]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Accept-Ranges: NPT, SMPTE, UTC
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9";
src_addr="192.0.2.5:9000"/"192.0.2.5:9001"
setup=passive;connection=new;ssrc=93CB001E
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 23 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT
C->M: TCP Connection Establishment
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=0-10, npt=30-
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=0-10, npt=30-623.10
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=1";
ssrc=4F312DD8:seq=54321;rtptime=2876889
C.2.3. Handling NPT Jumps in the RTP Media Layer
RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP
media layer[RFC3550]. Such control allows jumps to be created in NPT
timeline of the RTSP session. For example, jumps in NPT can be
caused by multiple ranges in the range specifier of a PLAY request or
through a "seek" opertaion on an RTSP session which involves a PLAY,
PAUSE, PLAY scenario where a new NPT is set for the session. The
media layer rendering the RTP stream should not be affected by jumps
in NPT. Thus, both RTP sequence numbers and RTP timestamps MUST be
continuous and monotonic across jumps of NPT.
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
Schulzrinne, et al. Expires August 28, 2008 [Page 193]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
agent may believe later packets to be duplicates of packets just
played out.
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 = 49, rtptime = 39200, NPT time = 14.9s
Immediately after the end of the play range, the client follows up
with a request to PLAY from a new NPT.
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 5
Session: abcdefg
Range: npt=18-20
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 5
Session: abcdefg
Range: npt=18-20
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=50;rtptime=40100
The ensuing RTP data stream is depicted below:
Schulzrinne, et al. Expires August 28, 2008 [Page 194]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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). To avoid this gap in playback due to the time it takes
to perform RTSP requests, a PLAY request with multiple ranges needs
to be specified. That would result in the following example:
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 4
Session: abcdefg
Range: npt=10-15;npt=18-20
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 4
Session: abcdefg
Range: npt=10-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
. . .
S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
S -> C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
S -> C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
. . .
S -> C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s
Schulzrinne, et al. Expires August 28, 2008 [Page 195]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C.2.4. Handling RTP Timestamps after PAUSE
During a PAUSE / PLAY interaction in an RTSP session, the duration of
time for which the RTP transmission was halted MUST be reflected in
the RTP timestamp of each RTP stream. The duration can be calculated
for each RTP stream as the time elapsed from when the last RTP packet
was sent before the PAUSE request was received and when the first RTP
packet was sent after the subsequent PLAY request was received. The
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:
Schulzrinne, et al. Expires August 28, 2008 [Page 196]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0
CSeq: 5
Session: abdcdefg
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 5
Session: abcdefg
Range: npt=10.4-15
20 seconds elapse and then the client sends a PLAY request. In
addition the server requires 15 ms to process the request:
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 6
Session: abcdefg
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 6
Session: abcdefg
Range: npt=10.4-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=4;rtptime=164400
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s
First, NPT 10 through 10.3 is played, then a PAUSE is received by the
server. After 20 seconds a PLAY is received by the server which take
15ms to process. The duration of time for which the session was
paused is reflected in the RTP timestamp of the RTP packets sent
after this PLAY request.
A client can use the RTSP range header and RTP-Info header to map NPT
time of a presentation with the RTP timestamp.
Note: In RFC 2326 [RFC2326], this matter was not clearly defined and
was misunderstood commonly. However for RTSP 2.0 it is expected that
this will be handled correctly and no exception handling will be
required.
Schulzrinne, et al. Expires August 28, 2008 [Page 197]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
C.2.5. RTSP / RTP Integration
For certain datatypes, tight integration between the RTSP layer and
the RTP layer will be necessary. This by no means precludes the
above restrictions. Combined RTSP/RTP media clients should use the
RTP-Info field to determine whether incoming RTP packets were sent
before or after a seek or before or after a PAUSE.
C.2.6. Scaling with RTP
For scaling (see Section 16.40), 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 16.41) of one, the
server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame.
Note: The above scaling puts requirements on the media codec or a
media stream to support it. For example motion JPEG or other non-
predictive video coding can easier handle the above example.
C.2.7. Maintaining NPT synchronization with RTP timestamps
The client can maintain a correct display of NPT (Normal Play Time)
by noting the RTP timestamp value of the first packet arriving after
repositioning. The sequence parameter of the RTP-Info
(Section 16.39) header provides the first sequence number of the next
segment.
C.2.8. Continuous Audio
For continuous audio, the server SHOULD set the RTP marker bit at the
beginning of serving a new PLAY request or at jumps in timeline.
This allows the client to perform playout delay adaptation.
C.2.9. Multiple Sources in an RTP Session
Note that more than one SSRC MAY be sent in the media stream. If it
happens all sources are expected to be rendered simultaneously.
C.2.10. Usage of SSRCs and the RTCP BYE Message During an RTSP Session
The RTCP BYE message indicates the end of use of a given SSRC. If
all sources leave an RTP session, it can, in most cases, be assumed
to have ended. Therefore, a client or server SHALL NOT send a RTCP
BYE message until it has finished using a SSRC. A server SHOULD keep
using a SSRC until the RTP session is terminated. Prolonging the use
of a SSRC allows the established synchronization context associated
Schulzrinne, et al. Expires August 28, 2008 [Page 198]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
with that SSRC to be used to synchronize subsequent PLAY requests
even if the PLAY response is late.
An SSRC collision with the SSRC that transmits media does also have
consequences, as it will force the media sender to change its SSRC in
accordance with the RTP specification[RFC3550]. This will result in
a loss of synchronization context, and require any receiver to wait
for RTCP sender reports for all media requiring synchronization
before being able to play out synchronized. Due to these reasons a
client joining a session should take care to not select the same SSRC
as the server. Any SSRC signalled in the Transport header SHOULD be
avoided. A client detecting a collision prior to sending any RTP or
RTCP messages can also select a new SSRC.
C.3. Future Additions
It is the intention that any future protocol or profile regarding
both for media delivery and lower transport should be easy to add to
RTSP. This section provides the necessary steps that needs to be
meet.
The following things needs to be considered when adding a new
protocol of profile for use with RTSP:
o The protocol or profile needs to define a name tag representing
it. This tag is required to be a ABNF "token" to be possible to
use in the Transport header specification.
o The useful combinations of protocol/profile/lower-layer needs to
be defined and 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.
See the IANA section (Section 22) for information how to register new
attributes.
Schulzrinne, et al. Expires August 28, 2008 [Page 199]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix D. Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, [RFC4566]) may be used to
describe streams or presentations in RTSP. This description is
typically returned in reply to a DESCRIBE request on an URI from a
server to a client, or received via HTTP from a server to a client.
This appendix describes how an SDP file determines the operation of
an RTSP session. SDP as is provides no mechanism by which a client
can distinguish, without human guidance, between several media
streams to be rendered simultaneously and a set of alternatives
(e.g., two audio streams spoken in different languages). However the
SDP extension "Grouping of Media Lines in the Session Description
Protocol (SDP)" [RFC3388] may provide such functionality depending on
need. Also future grouping semantics may in the future be developed.
D.1. Definitions
The terms "session-level", "media-level" and other key/attribute
names and values used in this appendix are to be used as defined in
SDP (RFC 4566 [RFC4566]):
D.1.1. Control URI
The "a=control:" attribute is used to convey the control URI. This
attribute is used both for the session and media descriptions. If
used for individual media, it indicates the URI to be used for
controlling that particular media stream. If found at the session
level, the attribute indicates the URI for aggregate control
(presentation URI). The session level URI SHALL be different from
any media level URI. The presence of a session level control
attribute SHALL be interpreted as support for aggregated control.
The control attribute SHALL be present on media level unless the
presentation only contains a single media stream, in which case the
attribute MAY only be present on the session level.
ABNF for the attribute is defined in Section 20.3.
Example:
a=control:rtsp://example.com/foo
This attribute MAY contain either relative or absolute URIs,
following the rules and conventions set out in RFC 3986 [RFC3986].
Implementations SHALL look for a base URI in the following order:
1. the RTSP Content-Base field;
Schulzrinne, et al. Expires August 28, 2008 [Page 200]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
2. the RTSP Content-Location field;
3. the RTSP Request-URI.
If this attribute contains only an asterisk (*), then the URI SHALL
be treated as if it were an empty embedded URI, and thus inherit the
entire base URI.
The URI handling for SDPs from container files need special
consideration. For example lets assume that a container file has the
URI: "rtsp://example.com/container.mp4". Lets further assume this
URI is the base URI, and that there is a absolute media level URI:
"rtsp://example.com/container.mp4/trackID=2". A relative media level
URI that resolves in accordance with RFC 3986 [RFC3986] to the above
given media URI is: "container.mp4/trackID=2". It is usually not
desirable to need to include in or modify the SDP stored within the
container file with the server local name of the container file. To
avoid this, one can modify the base URI used to include a trailing
slash, e.g. "rtsp://example.com/container.mp4/". In this case the
relative URI for the media will only need to be: "trackID=2".
However this will also mean that using "*" in the SDP will result in
control URI including the trailing slash, i.e.
"rtsp://example.com/container.mp4/".
Note: The usage of TrackID in the above is not an standardized
form, but one example out of several similar strings such as
TrackID, Track_ID, StreamID that is used by different server
vendors to indicate a particular piece of media inside a container
file.
D.1.2. Media Streams
The "m=" field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate
synchronization. If the session is over multicast, the port number
indicated SHOULD be used for reception. The client MAY try to
override the destination port, through the Transport header. The
servers MAY allow this, the response will indicate if allowed or not.
If the session is unicast, the port number is the ones RECOMMENDED by
the server to the client, about which receiver ports to use; the
client MUST still include its receiver ports in its SETUP request.
The client MAY ignore this recommendation. If the server has no
preference, it SHOULD set the port number value to zero.
The "m=" lines contain information about what transport protocol,
profile, and possibly lower-layer is to be used for the media stream.
The combination of transport, profile and lower layer, like RTP/AVP/
UDP needs to be defined for how to be used with RTSP. The currently
Schulzrinne, et al. Expires August 28, 2008 [Page 201]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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.
The direction attributes are not commonly used in SDPs for RTSP, but
may occur. "a=recvonly" in a SDP provided to the RTSP client SHALL
indicate that media delivery will only occur in the direction from
the RTSP server to the client. In SDP provided to the RTSP client
that lacks any of the directionality attributes (a=recvonly,
a=sendonly, a=sendrecv) SHALL behave as if the "a=recvonly" attribute
Schulzrinne, et al. Expires August 28, 2008 [Page 202]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
was received. Note that this overrules the normal default rule
defined in SDP[RFC4566]. The usage of "a=sendonly" or "a=sendrecv"
is not defined, nor is the interpretation of SDP by other entities
than the RTSP client.
D.1.6. Range of Presentation
The "a=range" attribute defines the total time range of the stored
session or an individual media. Non-seekable live sessions can be
indicated, while the length of live sessions can be deduced from the
"t" and "r" SDP parameters.
The attribute is both a session and a media level attribute. For
presentations that contains media streams of the same durations, the
range attribute SHOULD only be used at session-level. In case of
different length the range attribute MUST be given at media level for
all media, and SHOULD NOT be given at session level. If the
attribute is present at both media level and session level the media
level values SHALL be used.
Note: Usually one will specify the same length for all media, even if
there isn't media available for the full duration on all media.
However that requires that the server accepts PLAY requests within
that range.
Servers SHALL take care to provide RTSP Range (see Section 16.35)
values that are consistent with what is presented in the SDP for the
content. There are no reason for non dynamic content, like media
clips provided on demand to have inconsistent values. Inconsistent
values between the SDP and the actual values for the content handled
by the server is likely to generate some failure, like 457 "Invalid
Range", in case the client uses PLAY requests with a Range header.
In case the content is dynamic in length and it is infeasible to
provide a correct value in the SDP the server is recommended to
describe this as non-seekable content (see below). The server MAY
override that property in the response to a PLAY request using the
correct values in the Range header.
The unit is specified first, followed by the value range. The units
and their values are as defined in Section 4.4, Section 4.5 and
Section 4.6 and MAY be extended with further formats. Any open ended
range (start-), i.e. without stop range, is of unspecified duration
and SHALL be considered as non-seekable content unless this property
is overridden. Multiple instances carrying different clock formats
MAY be included at either session or media level.
ABNF for the attribute is defined in Section 20.3.
Schulzrinne, et al. Expires August 28, 2008 [Page 203]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Examples:
a=range:npt=0-34.4368
a=range:clock=19971113T2115-19971113T2203
Non seekable stream of unknown duration:
a=range:npt=0-
D.1.7. Time of Availability
The "t=" field MUST contain suitable values for the start and stop
times for both aggregate and non-aggregate stream control. The
server SHOULD indicate a stop time value for which it guarantees the
description to be valid, and a start time that is equal to or before
the time at which the DESCRIBE request was received. It MAY also
indicate start and stop times of 0, meaning that the session is
always available.
For sessions that are of live type, i.e. specific start time, unknown
stop time, likely unseekable, the "t=" and "r=" field SHOULD be used
to indicate the start time of the event. The stop time SHOULD be
given so that the live event will have ended at that time, while
still not be unnecessary long into the future.
D.1.8. Connection Information
In SDP, the "c=" field contains the destination address for the media
stream. For on-demand unicast streams and some multicast streams,
the destination address MAY be specified by the client via the SETUP
request, thus overriding any specified address. To identify streams
without a fixed destination address, where the client is required to
specify a destination address, the "c=" field SHOULD be set to a null
value. For addresses of type "IP4", this value SHALL be "0.0.0.0",
and for type "IP6", this value SHALL be "0:0:0:0:0:0:0:0", i.e. the
unspecified address according to RFC 3513 [RFC3513].
D.1.9. Entity Tag
The optional "a=etag" attribute identifies a version of the session
description. It is opaque to the client. SETUP requests may include
this identifier in the If-Match field (see Section 16.24) to only
allow session establishment if this attribute value still corresponds
to that of the current description. The attribute value is opaque
and may contain any character allowed within SDP attribute values.
ABNF for the attribute is defined in Section 20.3.
Example:
a=etag:158bb3e7c7fd62ce67f12b533f06b83a
Schulzrinne, et al. Expires August 28, 2008 [Page 204]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
One could argue that the "o=" field provides identical
functionality. However, it does so in a manner that would put
constraints on servers that need to support multiple session
description types other than SDP for the same piece of media
content.
D.2. Aggregate Control Not Available
If a presentation does not support aggregate control no session level
"a=control:" attribute is specified. For a SDP with multiple media
sections specified, each section will have its own control URI
specified via the "a=control:" attribute.
Example:
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.56
s=I came from a web page
e=adm@example.com
c=IN IP4 0.0.0.0
t=0 0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid
Note that the position of the control URI in the description implies
that the client establishes separate RTSP control sessions to the
servers audio.com and video.com.
It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media
client through non-RTSP means. This is necessary as there is no
mechanism to indicate that the client should request more detailed
media stream information via DESCRIBE.
D.3. Aggregate Control Available
In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level
"a=control:" attributes, which are used to specify the stream URIs,
and a session-level "a=control:" attribute which is used as the
Request-URI for aggregate control. If the media-level URI is
relative, it is resolved to absolute URIs according to Appendix D.1.1
above.
Example:
Schulzrinne, et al. Expires August 28, 2008 [Page 205]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Base: rtsp://example.com/movie/
Content-Length: 228
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
t=0 0
a=control:*
m=video 8002 RTP/AVP 31
a=control:trackID=1
m=audio 8004 RTP/AVP 3
a=control:trackID=2
In this example, the client is required to establish a single RTSP
session to the server, and uses the URIs
rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to set up the video and audio
streams, respectively. The URI rtsp://example.com/movie/, which is
resolved from the "*", controls the whole presentation (movie).
A client is not required to issues SETUP requests for all streams
within an aggregate object. Servers should allow the client to ask
for only a subset of the streams.
D.4. RTSP external SDP delivery
There are some considerations that needs to be made when the session
description is delivered to client outside of RTSP, for example in
HTTP or email.
First of all the SDP needs to contain absolute URIs, relative will in
most cases not work as the delivery will not correctly forward the
base URI. And as SDP might be temporarily stored on file system
before being loaded into an RTSP capable client, thus if possible to
transport the base URI it still would need to be merged into the
file.
Schulzrinne, et al. Expires August 28, 2008 [Page 206]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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].
Schulzrinne, et al. Expires August 28, 2008 [Page 207]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix E. Minimal RTSP Implementation
This section defines the minimal implementation requirements for RTSP
agents.
E.1. Minimal Core Implementation
The minimal core implementation is what is required to negotiate the
usage of any other features. A minimal core implementation is not
supporting any other feature set will be useless as the minimal
implementation doesn't deliver any service. All feature sets SHALL
include the minimal core.
A minimal core implementation SHALL support the following
functionalities:
o Establishing a connection between RTSP agents using TCP.
o Implement the reception and response to the OPTIONS method.
o Implement the handling of all headers mandatory or conditional in
regards to the usage of the OPTIONS method. See Table 9 and
Table 10. This include at least the capability to ignore unknown
headers.
o Implement the headers related to capability negotiation and
exchange:
* Require
* Supported
* Proxy-Require
* Proxy-Supported
* Unsupported
E.2. Recommended Core Implementation
A RTSP Agent is also RECOMMENDED to support the following:
o RTSP basic and digest authentication: The 401 response, the WWW-
Authenticate and Authorization headers, and both Basic and Digest
authentication methods as defined by [RFC2617].
o Secure RTSP message transport as specified by Appendix E.4.
Schulzrinne, et al. Expires August 28, 2008 [Page 208]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
E.3. The Basic Playback Feature Support
This section defines what is required to be supported for clients,
proxies and servers to be supporting the "play.basic" feature-tag.
E.3.1. Client
A play.basic supporting client SHALL implement the following:
o The RTSP methods as required by Table 7.
o All the RTSP headers that are required required or conditional in
requests or responses to method required to be supported according
to Table 9, Table 10, Table 11, and Table 12 and in addition the
following headers:
* Content-Base
* Content-Encoding and at least the Identity method.
* Content-Location
* Location
* Range and the npt time format
* RTP-Info
o Handling of all Status code categories.
o Media delivery using RTP/AVP over UDP.
A play.basic supporting client is also RECOMMENDED to support the
following:
o Expires header
o From header
E.3.2. Server
A play.basic supporting server SHALL implement the following:
o The RTSP methods as required by Table 7.
o Reception and responding to all headers specified in Section 16.
The implementation of functionality provided by all these header
with the following exceptions:
Schulzrinne, et al. Expires August 28, 2008 [Page 209]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
* Scale
* Speed
* Blocksize
o Media delivery using RTP/AVP over UDP.
A play.basic supporting Server is also RECOMMENDED to support the
following:
o XXX Editor's note: empty element in minimal.text!
E.3.3. Proxy
A play.basic supporting proxy SHALL implement the following:
o At least passing through all the methods listed in Table 7.
o The handling of all RTSP headers that are required to be handled
by the server and clients supporting "play.basic" and in addition
the following headers:
* Cache-Control
* Expires
* Via
E.4. Secure Transport
Any Client, Proxy or Server supporting secure transport of RTSP
messages and usage of the "rtsps" URI scheme SHALL implement:
o The Accept-Credentials and Connection-Credentials headers;
o TLS over TCP.
Schulzrinne, et al. Expires August 28, 2008 [Page 210]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix F. Requirements for Unreliable Transport of RTSP
This section provides anyone intending to define how to transport of
RTSP messages over a unreliable transport protocol with some
information learned by the attempt in RFC 2326 [RFC2326]. RFC 2326
define both an URI scheme and some basic functionality for transport
of RTSP messages over UDP, however it was not sufficient for reliable
usage and successful interoperability.
The RTSP scheme defined for unreliable transport of RTSP messages was
"rtspu". It has been reserved by this specification as at least one
commercial implementation exist, thus avoiding any collisions in the
name space.
The following considerations should exist for operation of RTSP over
an unreliable transport protocol:
o Request shall be acknowledged by the receiver. If there is no
acknowledgement, the sender may resend the same message after a
timeout of one round-trip time (RTT). Any retransmissions due to
lack of acknowledgement must carry the same sequence number as the
original request.
o The round-trip time can be estimated as in TCP (RFC 1123)
[RFC1123], with an initial round-trip value of 500 ms. An
implementation may cache the last RTT measurement as the initial
value for future connections.
o If RTSP is used over a small-RTT LAN, standard procedures for
optimizing initial TCP round trip estimates, such as those used in
T/TCP (RFC 1644) [RFC1644], can be beneficial.
o The Timestamp header (Section 16.45) is used to avoid the
retransmission ambiguity problem XXY Need ref for Stev94:TCP and
obviates the need for Karn's algorithm.
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
Schulzrinne, et al. Expires August 28, 2008 [Page 211]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
using an unreliable protocol like UDP than for TCP.
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 16.45
Schulzrinne, et al. Expires August 28, 2008 [Page 212]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix G. Backwards Compatibility Considerations
This section contains notes on issues about backwards compatibility
with clients or servers being implemented according to RFC 2326
[RFC2326]. Note that there exist no requirement to implement RTSP
1.0, in fact we recommend against it as it is difficult to do in an
interoperable way.
A server implementing RTSP/2.0 MUST include a RTSP-Version of
RTSP/2.0 in all responses to requests containing RTSP-Version
RTSP/2.0. If a server receives a RTSP/1.0 request, it MAY respond
with a RTSP/1.0 response if it chooses to support RFC 2326. If the
server chooses not to support RFC 2326, it SHOULD respond with a 505
(RTSP Version not supported) status code. A server MUST NOT respond
to a RTSP-Version RTSP/1.0 request with a RTSP-Version RTSP/2.0
response.
Clients implementing RTSP/2.0 MAY use an OPTIONS request with a RTSP-
Version of 2.0 to determine whether a server supports RTSP/2.0. If
the server responds with either a RTSP-Version of 1.0 or a status
code of 505 (RTSP Version not supported), the client will have to use
RTSP/1.0 requests if it chooses to support RFC 2326.
G.1. Play Request in Play mode
The behavior in the server when a Play is received in Play mode has
changed (Section 13.4). In RFC 2326, the new PLAY request would be
queued until the current Play completed. Any new PLAY request now
take effect immediately replacing the previous request.
G.2. Using Persistent Connections
Some server implementations of RFC 2326 maintain a one-to-one
relationship between a connection and an RTSP session. Such
implementations require clients to use a persistent connection to
communicate with the server and when a client closes its connection,
the server may remove the RTSP session. This is worth noting if a
RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.
Schulzrinne, et al. Expires August 28, 2008 [Page 213]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix H. Open Issues
This section contains a list of open issues that still needs to be
resolved. However also any open issues in the bug tracker at
http://rtspspec.sourceforge.net should also be considered.
1. Should the SMPTE range format be updated to support the 50 and
60 frames per second modes?
2. Should we define a recommended format for error message bodies?
3. Today there is no recommended or required format for 300
response entities containing URI lists. Should one be defined?
4. Should the dest_addr parameter in the Transport header in
responses include the destination used by the server?
5. Should a IPv6 multicast scope parameter for the Transport header
be defined? This would be similar to TTL.
6. The Expires header (Section 16.22 contains the below paragraph:
Expires header field with a date value of some time in the
future on a media stream that otherwise would by default be non-
cacheable indicates that the media stream is cacheable, unless
indicated otherwise by a Cache-Control header field
(Section 16.10).
Is there any purpose for this in RTSP, or could we remove this
statement and instead rely on the Cache-Control header?
7. Should proxies strip out the credentials for themselves when
forwarding messages with Accept-Credentials?
8. Is Session ID combined with TLS a sufficient mechanism to
prevent hijacking?
9. Move to start TLS mechanism like the one defined in RFC 2817?
10. Look into the GRID communities proxy-certs and see how this
relates to the current TLS proxy solution.
11. Resolve Eric Rescorlas security comments on the Proxy TLS
solution:
1. There doesn't seem to be any way to communicate your cipher
suite preferences.
Schulzrinne, et al. Expires August 28, 2008 [Page 214]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
2. I don't see how certificate-based client authentication
works. Is it supposed to?
3. You need to provide the entire cert chain in Connection-
Credentials, not just the certificate.
12. Consider to switch to SHA256 instead of SHA1 for the digest over
the DER encoded certs.
13. Resolve the following Stephen Farrel issue: "C. I don't
understand how the client-side proxies can be expected to know
enough about proxies existing toward the server. If they don't
then I'm not sure how they can be expected to make any decision
that's better than would be the case were policy to be dealt
with solely on a hop-by-hop basis. Maybe I'm missing something
that can provide that information?"
14. Resolve the following Stephen Farrel issue: "D. The "User"
policy model is that a client presents acceptable name/URIs and
digests to the proxy. TLS doesn't really provide a way for that
proxy, as a client, to ask the server for the "right"
certificate, so I suspect there's a gap here that'll be hard to
fill. (If the client imposed a constraint as to the root-CA
that had to be used then that'd map to the next TLS connection,
but maybe it'd be too coarse-grained?)"
Schulzrinne, et al. Expires August 28, 2008 [Page 215]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix I. Changes
Compared to RTSP 1.0 (RFC 2326), the below changes has been made when
defining RTSP 2.0. Note that this list does not reflect minor
changes in wording or correction of typographical errors.
o The 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
supporting this version.
Schulzrinne, et al. Expires August 28, 2008 [Page 216]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
* Clarified the text on the transport headers "dest_addr"
parameters regarding what security precautions the server is
required to perform.
o The Range formats has been changed in the following way:
* The NPT format has been given a initial NPT identifier that
must now be used.
* All formats now support initial open ended formats of type
"npt=-10".
o RTSP message handling has been changed in the following way:
* RTSP messages now uses URIs rather then URLs.
* It has been clarified that a 4xx message due to missing CSeq
header shall be returned without a CSeq header.
* Rules for how to handle timing out RTSP messages has been
added.
* Extended Pipelining rules allowing for quick session startup.
o The HTTP references has been updated to RFC 2616 and RFC 2617.
This has resulted in that the Public, and the Content-Base header
needed to be defined in the RTSP specification. Known effects on
RTSP due to HTTP clarifications:
* Content-Encoding header can include encoding of type
"identity".
o The state machine section has completely been rewritten. It
includes now more details and are also more clear about the model
used.
o A IANA section has been included with contains a number of
registries and their rules. This will allow us to use IANA to
keep track of RTSP extensions.
o Than transport of RTSP messages has seen the following changes:
* The use of UDP for RTSP message transport has been deprecated
due to missing interest and to broken specification.
* 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
Schulzrinne, et al. Expires August 28, 2008 [Page 217]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
time.
* Strong recommendations why server and clients should use
persistent connections has also been added.
* There is now a requirement on the servers to handle non-
persistent connections as this provides fault tolerance.
* Added wording on the usage of Connection:Close for RTSP.
* specified usage of TLS for RTSP messages, including a scheme to
approve a proxies TLS connection to the next hop.
o The following header related changes have been made:
* Accept-Ranges response header is added. This header clarifies
which range formats that can be used for a resource.
* Changed the Range header to allow multiple ranges for creating
editing list.
* Fixed the missing definitions for the Cache-Control header.
Also added to the syntax definition the missing delta-seconds
for max-stale and min-fresh parameters.
* Put requirement on CSeq header that the value is increased by
one for each new RTSP request. A Recommendation to start at 1
has also been added.
* Added requirement that the Date header must be used for all
messages with entity and the Server should always include it.
* Removed possibility of using Range header with Scale header to
indicate when it is to be activated, since it can't work as
defined. Also added rule that lack of Scale header in response
indicates lack of support for the header. Feature-tags for
scaled playback has been defined.
* The Speed header must now be responded to indicate support and
the actual speed going to be used. A feature-tag is defined.
Notes on congestion control was also added.
* The Supported header was borrowed from SIP to help with the
feature negotiation in RTSP.
* Clarified that the Timestamp header can be used to resolve
retransmission ambiguities.
Schulzrinne, et al. Expires August 28, 2008 [Page 218]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
* The Session header text has been expanded with a explanation on
keep alive and which methods to use. SET_PARAMETER is now
recommended to use if only keep-alive within RTSP is desired.
* It has been clarified how the Range header formats is used to
indicate pause points in the PAUSE response.
* Clarified that RTP-Info URIs that are relative, uses the
Request-URI as base URI. Also clarified that used URI must be
that one that was used in the SETUP request. They are now also
required to be quoted. The header also expresses the SSRC for
the provided RTP timestamp and sequence number values.
* Added text that requires the Range to always be present in PLAY
responses. Clarified what should be sent in case of live
streams.
* The headers table has been updated using a structured borrowed
from SIP. Those tables carries much more information and
should provide a good overview of the available headers.
* It has been is clarified that any message with a message body
is required to have a Content-Length header. This was the case
in RFC 2326 but could be misinterpreted.
* To resolve functionality around ETag. The ETag and If-None-
Match header has been added from HTTP with necessary
clarification in regards to RTSP operation.
* Imported the Public header from HTTP RFC 2068 [RFC2068] since
it has been removed from HTTP due to lack of use. Public is
used quite frequently in RTSP.
* Clarified rules for populating the Public header so that it is
an intersection of the capabilities of all the RTSP agents in a
chain.
o The Protocol Syntax has been changed in the following way:
* All BNF definitions are updated according to the rules defined
in RFC 4234 [RFC4234] and has been gathered in a separate
Section 20.
* The BNF for the User-Agent and Server headers has been
corrected so now only the description is in the HTTP
specification.
Schulzrinne, et al. Expires August 28, 2008 [Page 219]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
* Some definitions in the introduction regarding the RTSP session
has been changed.
* The protocol has been made fully IPv6 capable. Certain of the
functionality, like using explicit IPv6 addresses in fields
requires that the protocol support this updated specification.
* Added a fragment part to the RTSP URI. This seem to be
indicated by the note below the definition however it was not
part of the BNF.
* The CHAR rule has been changed to exclude NULL.
o The Status codes has been changed in the following way:
* The use of status code 303 "See Other" has been deprecated as
it does not make sense to use in RTSP.
* When sending response 451 and 458 the response body should
contain the offending parameters.
* Clarification on when a 3rr redirect status code can be
received has been added. This includes receiving 3rr as a
result of request within a established session. This provides
clarification to a previous unspecified behavior.
* Removed the 201 (Created) and 250 (Low On Storage Space) status
codes as they are only relevant to recording, which is
deprecated.
o The following functionality has been deprecated from the protocol:
* The use of Queued Play.
* The use of PLAY method for keep-alive in play state.
* The RECORD and ANNOUNCE methods and all related functionality.
Some of the syntax has been removed.
* The possibility to use timed execution of methods with the time
parameter in the Range header.
* The description on how rtspu works is not part of the core
specification and will require external description. Only that
it exist is defined here and some requirements for the the
transport is provided.
Schulzrinne, et al. Expires August 28, 2008 [Page 220]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
o The following changes has been made in relation to methods:
* The OPTIONS method has been clarified with regards to the use
of the Public and Allow headers.
* The RECORD and ANNOUNCE methods are removed as they are lacking
implementation and not considered necessary in the core
specification. Any work on these methods should be done as a
extension document to RTSP.
* Added text clarifying the usage of SET_PARAMETER for keep-alive
and usage without any body.
* PLAY method is now allowed to be pipelined with the pipelining
of one or more SETUP requests following the initial that
generates the session for aggregated control.
o Wrote a new section about how to setup different media transport
alternatives and their profiles, and lower layer protocols. This
resulted that the appendix on RTP interaction was moved there
instead in the part describing RTP. The section also includes
guidelines what to think of when writing usage guidelines for new
protocols and profiles.
o Setup and usage of independent TCP connections for transport of
RTP has been specified.
o Added a new section describing the available mechanisms to
determine if functionality is supported, called "Capability
Handling". Renamed option-tags to feature-tags.
o Added a contributors section with people who have contributed
actual text to the specification.
o Added a section Use Cases that describes the major use cases for
RTSP.
o Clarified the usage of a=range and how to indicate live content
that are not seekable with this header.
o Text specifying the special behavior of PLAY for live content.
I.1. Changes needing to be updated
The minimal implementation specification has been changed:
o Required Timestamp, Via, and Unsupported headers for a minimal
server implementation.
Schulzrinne, et al. Expires August 28, 2008 [Page 221]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
o Recommended that Cache-Control, Expires and Date headers be
supported by server implementations.
Schulzrinne, et al. Expires August 28, 2008 [Page 222]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix J. Acknowledgements
This memorandum defines RTSP version 2.0 which is a revision of the
Proposed Standard RTSP version 1.0 which is defined in [RFC2326].
The authors of this RFC are Henning Schulzrinne, Anup Rao, and Robert
Lanphier.
Both RTSP version 1.0 and RTSP versio 2.0 borrow format and
descriptions from HTTP/1.1.
This document has benefited greatly from the comments of all those
participating in the MMUSIC-WG. In addition to those already
mentioned, the following individuals have contributed to this
specification:
Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari,
Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V.
Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets,
Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal
Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov,
Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith,
Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen
Chesire, David Walker, Geetha Srikantan, Stephan Wenger, Pekka Pessi,
Jae-Hwan Kim, Holger Schmidt, Stephen Farrell, Xavier Marjou, Joe
Pallas and Mela Martti.
J.1. Contributors
The following people have made written contributions that were
included in the specification:
o Tom Marshall contributed text on the usage of 3rr status codes.
o Thomas Zheng contributed text on the usage of the Range in PLAY
responses.
o Sean Sheedy contributed text on the timeout behavior of RTSP
messages and connections, and the 463 status code.
o Fredrik Lindholm contributed text about the RTSP security
framework.
o John Lazzaro contributed the text for RTP over Independent TCP.
Schulzrinne, et al. Expires August 28, 2008 [Page 223]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
o Aravind Narasimhan contributed by rewriting Media Transport
Alternatives (Appendix C) and editorial improvements on a number
of places in the specification.
Schulzrinne, et al. Expires August 28, 2008 [Page 224]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Appendix K. RFC Editor Consideration
Please replace RFC XXXX with the RFC number this specification
recieves.
Please replace RFC YYYY with the RFC number that SAVPF
[I-D.ietf-avt-profile-savpf] receives.
Schulzrinne, et al. Expires August 28, 2008 [Page 225]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
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
Real Networks
Seattle, WA
USA
Email: robla@robla.net
Magnus Westerlund
Ericsson AB
Torshamsgatan 23
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
Schulzrinne, et al. Expires August 28, 2008 [Page 226]
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) February 2008
Full Copyright Statement
Copyright (C) The IETF Trust (2008).
This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors
retain all their rights.
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND
THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS
OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Intellectual Property
The IETF takes no position regarding the validity or scope of any
Intellectual Property Rights or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; nor does it represent that it has
made any independent effort to identify any such rights. Information
on the procedures with respect to rights in RFC documents can be
found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any
assurances of licenses to be made available, or the result of an
attempt made to obtain a general license or permission for the use of
such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository at
http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at
ietf-ipr@ietf.org.
Acknowledgment
Funding for the RFC Editor function is provided by the IETF
Administrative Support Activity (IASA).
Schulzrinne, et al. Expires August 28, 2008 [Page 227]
Html markup produced by rfcmarkup 1.129b, available from
https://tools.ietf.org/tools/rfcmarkup/