MMUSIC Working Group H. Schulzrinne Internet-Draft Columbia University Obsoletes: 2326 (if approved) A. Rao Intended status: Standards Track Cisco Expires: March 15, 2014 R. Lanphier M. Westerlund Ericsson AB M. Stiemerling (Ed.) NEC September 11, 2013 Real Time Streaming Protocol 2.0 (RTSP) draft-ietf-mmusic-rfc2326bis-36 Abstract This memorandum defines RTSP version 2.0 which obsoletes RTSP version 1.0 defined in RFC 2326. The Real Time Streaming Protocol, or RTSP, is an application-level protocol for setup and control of the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP (RFC 3550). Status of this Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." This Internet-Draft will expire on March 15, 2014. Schulzrinne, et al. Expires March 15, 2014 [Page 1] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Copyright Notice Copyright (c) 2013 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. This document may contain material from IETF Documents or IETF Contributions published or made publicly available before November 10, 2008. The person(s) controlling the copyright in some of this material may not have granted the IETF Trust the right to allow modifications of such material outside the IETF Standards Process. Without obtaining an adequate license from the person(s) controlling the copyright in such materials, this document may not be modified outside the IETF Standards Process, and derivative works of it may not be created outside the IETF Standards Process, except to format it for publication as an RFC or to translate it into languages other than English. Schulzrinne, et al. Expires March 15, 2014 [Page 2] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 11 2. Protocol Overview . . . . . . . . . . . . . . . . . . . . . . 13 2.1. Presentation Description . . . . . . . . . . . . . . . . 13 2.2. Session Establishment . . . . . . . . . . . . . . . . . 14 2.3. Media Delivery Control . . . . . . . . . . . . . . . . . 15 2.4. Session Parameter Manipulations . . . . . . . . . . . . 17 2.5. Media Delivery . . . . . . . . . . . . . . . . . . . . . 18 2.5.1. Media Delivery Manipulations . . . . . . . . . . . . 18 2.6. Session Maintenance and Termination . . . . . . . . . . 20 2.7. Extending RTSP . . . . . . . . . . . . . . . . . . . . . 21 3. Document Conventions . . . . . . . . . . . . . . . . . . . . 23 3.1. Notational Conventions . . . . . . . . . . . . . . . . . 23 3.2. Terminology . . . . . . . . . . . . . . . . . . . . . . 23 4. Protocol Parameters . . . . . . . . . . . . . . . . . . . . . 27 4.1. RTSP Version . . . . . . . . . . . . . . . . . . . . . . 27 4.2. RTSP IRI and URI . . . . . . . . . . . . . . . . . . . . 27 4.3. Session Identifiers . . . . . . . . . . . . . . . . . . 30 4.4. Media Time Formats . . . . . . . . . . . . . . . . . . . 30 4.4.1. SMPTE Relative Timestamps . . . . . . . . . . . . . 30 4.4.2. Normal Play Time . . . . . . . . . . . . . . . . . . 31 4.4.3. Absolute Time . . . . . . . . . . . . . . . . . . . 31 4.5. Feature-Tags . . . . . . . . . . . . . . . . . . . . . . 32 4.6. Message Body Tags . . . . . . . . . . . . . . . . . . . 32 4.7. Media Properties . . . . . . . . . . . . . . . . . . . . 33 4.7.1. Random Access and Seeking . . . . . . . . . . . . . 34 4.7.2. Retention . . . . . . . . . . . . . . . . . . . . . 34 4.7.3. Content Modifications . . . . . . . . . . . . . . . 34 4.7.4. Supported Scale Factors . . . . . . . . . . . . . . 35 4.7.5. Mapping to the Attributes . . . . . . . . . . . . . 35 5. RTSP Message . . . . . . . . . . . . . . . . . . . . . . . . 36 5.1. Message Types . . . . . . . . . . . . . . . . . . . . . 36 5.2. Message Headers . . . . . . . . . . . . . . . . . . . . 37 5.3. Message Body . . . . . . . . . . . . . . . . . . . . . . 37 5.4. Message Length . . . . . . . . . . . . . . . . . . . . . 38 6. General Header Fields . . . . . . . . . . . . . . . . . . . . 39 7. Request . . . . . . . . . . . . . . . . . . . . . . . . . . . 41 7.1. Request Line . . . . . . . . . . . . . . . . . . . . . . 41 7.2. Request Header Fields . . . . . . . . . . . . . . . . . 43 8. Response . . . . . . . . . . . . . . . . . . . . . . . . . . 45 8.1. Status-Line . . . . . . . . . . . . . . . . . . . . . . 45 8.1.1. Status Code and Reason Phrase . . . . . . . . . . . 45 8.2. Response Headers . . . . . . . . . . . . . . . . . . . . 49 9. Message Body . . . . . . . . . . . . . . . . . . . . . . . . 50 9.1. Message-Body Header Fields . . . . . . . . . . . . . . . 50 9.2. Message Body . . . . . . . . . . . . . . . . . . . . . . 51 9.3. Message Body Format Negotiation . . . . . . . . . . . . 52 Schulzrinne, et al. Expires March 15, 2014 [Page 3] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 10. Connections . . . . . . . . . . . . . . . . . . . . . . . . . 53 10.1. Reliability and Acknowledgements . . . . . . . . . . . . 53 10.2. Using Connections . . . . . . . . . . . . . . . . . . . 54 10.3. Closing Connections . . . . . . . . . . . . . . . . . . 57 10.4. Timing Out Connections and RTSP Messages . . . . . . . . 58 10.5. Showing Liveness . . . . . . . . . . . . . . . . . . . . 59 10.6. Use of IPv6 . . . . . . . . . . . . . . . . . . . . . . 60 10.7. Overload Control . . . . . . . . . . . . . . . . . . . . 61 11. Capability Handling . . . . . . . . . . . . . . . . . . . . . 63 11.1. Feature Tag: play.basic . . . . . . . . . . . . . . . . 64 12. Pipelining Support . . . . . . . . . . . . . . . . . . . . . 66 13. Method Definitions . . . . . . . . . . . . . . . . . . . . . 67 13.1. OPTIONS . . . . . . . . . . . . . . . . . . . . . . . . 68 13.2. DESCRIBE . . . . . . . . . . . . . . . . . . . . . . . . 69 13.3. SETUP . . . . . . . . . . . . . . . . . . . . . . . . . 71 13.3.1. Changing Transport Parameters . . . . . . . . . . . 74 13.4. PLAY . . . . . . . . . . . . . . . . . . . . . . . . . . 75 13.4.1. General Usage . . . . . . . . . . . . . . . . . . . 75 13.4.2. Aggregated Sessions . . . . . . . . . . . . . . . . 80 13.4.3. Updating current PLAY Requests . . . . . . . . . . . 81 13.4.4. Playing On-Demand Media . . . . . . . . . . . . . . 83 13.4.5. Playing Dynamic On-Demand Media . . . . . . . . . . 84 13.4.6. Playing Live Media . . . . . . . . . . . . . . . . . 84 13.4.7. Playing Live with Recording . . . . . . . . . . . . 85 13.4.8. Playing Live with Time-Shift . . . . . . . . . . . . 85 13.5. PLAY_NOTIFY . . . . . . . . . . . . . . . . . . . . . . 86 13.5.1. End-of-Stream . . . . . . . . . . . . . . . . . . . 87 13.5.2. Media-Properties-Update . . . . . . . . . . . . . . 89 13.5.3. Scale-Change . . . . . . . . . . . . . . . . . . . . 90 13.6. PAUSE . . . . . . . . . . . . . . . . . . . . . . . . . 91 13.7. TEARDOWN . . . . . . . . . . . . . . . . . . . . . . . . 94 13.7.1. Client to Server . . . . . . . . . . . . . . . . . . 94 13.7.2. Server to Client . . . . . . . . . . . . . . . . . . 95 13.8. GET_PARAMETER . . . . . . . . . . . . . . . . . . . . . 96 13.9. SET_PARAMETER . . . . . . . . . . . . . . . . . . . . . 98 13.10. REDIRECT . . . . . . . . . . . . . . . . . . . . . . . . 100 14. Embedded (Interleaved) Binary Data . . . . . . . . . . . . . 103 15. Proxies . . . . . . . . . . . . . . . . . . . . . . . . . . . 105 15.1. Proxies and Protocol Extensions . . . . . . . . . . . . 106 15.2. Multiplexing and Demultiplexing of Messages . . . . . . 107 16. Caching . . . . . . . . . . . . . . . . . . . . . . . . . . . 108 16.1. Validation Model . . . . . . . . . . . . . . . . . . . . 108 16.1.1. Last-Modified Dates . . . . . . . . . . . . . . . . 110 16.1.2. Message Body Tag Cache Validators . . . . . . . . . 110 16.1.3. Weak and Strong Validators . . . . . . . . . . . . . 110 16.1.4. Rules for When to Use Message Body Tags and Last-Modified Dates . . . . . . . . . . . . . . . . 112 16.1.5. Non-validating Conditionals . . . . . . . . . . . . 114 Schulzrinne, et al. Expires March 15, 2014 [Page 4] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 16.2. Invalidation After Updates or Deletions . . . . . . . . 114 17. Status Code Definitions . . . . . . . . . . . . . . . . . . . 116 17.1. Success 1xx . . . . . . . . . . . . . . . . . . . . . . 116 17.1.1. 100 Continue . . . . . . . . . . . . . . . . . . . . 116 17.2. Success 2xx . . . . . . . . . . . . . . . . . . . . . . 116 17.2.1. 200 OK . . . . . . . . . . . . . . . . . . . . . . . 116 17.3. Redirection 3xx . . . . . . . . . . . . . . . . . . . . 116 17.3.1. 300 . . . . . . . . . . . . . . . . . . . . . . . . 117 17.3.2. 301 Moved Permanently . . . . . . . . . . . . . . . 117 17.3.3. 302 Found . . . . . . . . . . . . . . . . . . . . . 117 17.3.4. 303 See Other . . . . . . . . . . . . . . . . . . . 118 17.3.5. 304 Not Modified . . . . . . . . . . . . . . . . . . 118 17.3.6. 305 Use Proxy . . . . . . . . . . . . . . . . . . . 118 17.4. Client Error 4xx . . . . . . . . . . . . . . . . . . . . 118 17.4.1. 400 Bad Request . . . . . . . . . . . . . . . . . . 118 17.4.2. 401 Unauthorized . . . . . . . . . . . . . . . . . . 119 17.4.3. 402 Payment Required . . . . . . . . . . . . . . . . 119 17.4.4. 403 Forbidden . . . . . . . . . . . . . . . . . . . 119 17.4.5. 404 Not Found . . . . . . . . . . . . . . . . . . . 119 17.4.6. 405 Method Not Allowed . . . . . . . . . . . . . . . 119 17.4.7. 406 Not Acceptable . . . . . . . . . . . . . . . . . 120 17.4.8. 407 Proxy Authentication Required . . . . . . . . . 120 17.4.9. 408 Request Timeout . . . . . . . . . . . . . . . . 120 17.4.10. 410 Gone . . . . . . . . . . . . . . . . . . . . . . 120 17.4.11. 412 Precondition Failed . . . . . . . . . . . . . . 121 17.4.12. 413 Request Message Body Too Large . . . . . . . . . 121 17.4.13. 414 Request-URI Too Long . . . . . . . . . . . . . . 121 17.4.14. 415 Unsupported Media Type . . . . . . . . . . . . . 121 17.4.15. 451 Parameter Not Understood . . . . . . . . . . . . 121 17.4.16. 452 reserved . . . . . . . . . . . . . . . . . . . . 122 17.4.17. 453 Not Enough Bandwidth . . . . . . . . . . . . . . 122 17.4.18. 454 Session Not Found . . . . . . . . . . . . . . . 122 17.4.19. 455 Method Not Valid in This State . . . . . . . . . 122 17.4.20. 456 Header Field Not Valid for Resource . . . . . . 122 17.4.21. 457 Invalid Range . . . . . . . . . . . . . . . . . 122 17.4.22. 458 Parameter Is Read-Only . . . . . . . . . . . . . 122 17.4.23. 459 Aggregate Operation Not Allowed . . . . . . . . 123 17.4.24. 460 Only Aggregate Operation Allowed . . . . . . . . 123 17.4.25. 461 Unsupported Transport . . . . . . . . . . . . . 123 17.4.26. 462 Destination Unreachable . . . . . . . . . . . . 123 17.4.27. 463 Destination Prohibited . . . . . . . . . . . . . 123 17.4.28. 464 Data Transport Not Ready Yet . . . . . . . . . . 123 17.4.29. 465 Notification Reason Unknown . . . . . . . . . . 124 17.4.30. 466 Key Management Error . . . . . . . . . . . . . . 124 17.4.31. 470 Connection Authorization Required . . . . . . . 124 17.4.32. 471 Connection Credentials not accepted . . . . . . 124 17.4.33. 472 Failure to establish secure connection . . . . . 124 17.5. Server Error 5xx . . . . . . . . . . . . . . . . . . . . 124 Schulzrinne, et al. Expires March 15, 2014 [Page 5] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 17.5.1. 500 Internal Server Error . . . . . . . . . . . . . 124 17.5.2. 501 Not Implemented . . . . . . . . . . . . . . . . 125 17.5.3. 502 Bad Gateway . . . . . . . . . . . . . . . . . . 125 17.5.4. 503 Service Unavailable . . . . . . . . . . . . . . 125 17.5.5. 504 Gateway Timeout . . . . . . . . . . . . . . . . 125 17.5.6. 505 RTSP Version Not Supported . . . . . . . . . . . 125 17.5.7. 551 Option not supported . . . . . . . . . . . . . . 126 17.5.8. 553 Proxy Unavailable . . . . . . . . . . . . . . . 126 18. Header Field Definitions . . . . . . . . . . . . . . . . . . 127 18.1. Accept . . . . . . . . . . . . . . . . . . . . . . . . . 137 18.2. Accept-Credentials . . . . . . . . . . . . . . . . . . . 138 18.3. Accept-Encoding . . . . . . . . . . . . . . . . . . . . 138 18.4. Accept-Language . . . . . . . . . . . . . . . . . . . . 139 18.5. Accept-Ranges . . . . . . . . . . . . . . . . . . . . . 140 18.6. Allow . . . . . . . . . . . . . . . . . . . . . . . . . 141 18.7. Authentication-Info . . . . . . . . . . . . . . . . . . 141 18.8. Authorization . . . . . . . . . . . . . . . . . . . . . 141 18.9. Bandwidth . . . . . . . . . . . . . . . . . . . . . . . 142 18.10. Blocksize . . . . . . . . . . . . . . . . . . . . . . . 143 18.11. Cache-Control . . . . . . . . . . . . . . . . . . . . . 143 18.12. Connection . . . . . . . . . . . . . . . . . . . . . . . 146 18.13. Connection-Credentials . . . . . . . . . . . . . . . . . 146 18.14. Content-Base . . . . . . . . . . . . . . . . . . . . . . 147 18.15. Content-Encoding . . . . . . . . . . . . . . . . . . . . 148 18.16. Content-Language . . . . . . . . . . . . . . . . . . . . 148 18.17. Content-Length . . . . . . . . . . . . . . . . . . . . . 149 18.18. Content-Location . . . . . . . . . . . . . . . . . . . . 149 18.19. Content-Type . . . . . . . . . . . . . . . . . . . . . . 150 18.20. CSeq . . . . . . . . . . . . . . . . . . . . . . . . . . 151 18.21. Date . . . . . . . . . . . . . . . . . . . . . . . . . . 152 18.22. Expires . . . . . . . . . . . . . . . . . . . . . . . . 153 18.23. From . . . . . . . . . . . . . . . . . . . . . . . . . . 154 18.24. If-Match . . . . . . . . . . . . . . . . . . . . . . . . 155 18.25. If-Modified-Since . . . . . . . . . . . . . . . . . . . 155 18.26. If-None-Match . . . . . . . . . . . . . . . . . . . . . 155 18.27. Last-Modified . . . . . . . . . . . . . . . . . . . . . 156 18.28. Location . . . . . . . . . . . . . . . . . . . . . . . . 157 18.29. Media-Properties . . . . . . . . . . . . . . . . . . . . 157 18.30. Media-Range . . . . . . . . . . . . . . . . . . . . . . 159 18.31. MTag . . . . . . . . . . . . . . . . . . . . . . . . . . 160 18.32. Notify-Reason . . . . . . . . . . . . . . . . . . . . . 160 18.33. Pipelined-Requests . . . . . . . . . . . . . . . . . . . 160 18.34. Proxy-Authenticate . . . . . . . . . . . . . . . . . . . 161 18.35. Proxy-Authentication-Info . . . . . . . . . . . . . . . 162 18.36. Proxy-Authorization . . . . . . . . . . . . . . . . . . 162 18.37. Proxy-Require . . . . . . . . . . . . . . . . . . . . . 162 18.38. Proxy-Supported . . . . . . . . . . . . . . . . . . . . 162 18.39. Public . . . . . . . . . . . . . . . . . . . . . . . . . 163 Schulzrinne, et al. Expires March 15, 2014 [Page 6] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18.40. Range . . . . . . . . . . . . . . . . . . . . . . . . . 164 18.41. Referrer . . . . . . . . . . . . . . . . . . . . . . . . 166 18.42. Request-Status . . . . . . . . . . . . . . . . . . . . . 166 18.43. Require . . . . . . . . . . . . . . . . . . . . . . . . 167 18.44. Retry-After . . . . . . . . . . . . . . . . . . . . . . 168 18.45. RTP-Info . . . . . . . . . . . . . . . . . . . . . . . . 168 18.46. Scale . . . . . . . . . . . . . . . . . . . . . . . . . 170 18.47. Seek-Style . . . . . . . . . . . . . . . . . . . . . . . 171 18.48. Server . . . . . . . . . . . . . . . . . . . . . . . . . 173 18.49. Session . . . . . . . . . . . . . . . . . . . . . . . . 173 18.50. Speed . . . . . . . . . . . . . . . . . . . . . . . . . 175 18.51. Supported . . . . . . . . . . . . . . . . . . . . . . . 176 18.52. Terminate-Reason . . . . . . . . . . . . . . . . . . . . 176 18.53. Timestamp . . . . . . . . . . . . . . . . . . . . . . . 177 18.54. Transport . . . . . . . . . . . . . . . . . . . . . . . 177 18.55. Unsupported . . . . . . . . . . . . . . . . . . . . . . 184 18.56. User-Agent . . . . . . . . . . . . . . . . . . . . . . . 185 18.57. Via . . . . . . . . . . . . . . . . . . . . . . . . . . 185 18.58. WWW-Authenticate . . . . . . . . . . . . . . . . . . . . 186 19. Security Framework . . . . . . . . . . . . . . . . . . . . . 187 19.1. RTSP and HTTP Authentication . . . . . . . . . . . . . . 187 19.1.1. Digest Authentication . . . . . . . . . . . . . . . 188 19.2. RTSP over TLS . . . . . . . . . . . . . . . . . . . . . 188 19.3. Security and Proxies . . . . . . . . . . . . . . . . . . 189 19.3.1. Accept-Credentials . . . . . . . . . . . . . . . . . 191 19.3.2. User approved TLS procedure . . . . . . . . . . . . 192 20. Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . . 194 20.1. Base Syntax . . . . . . . . . . . . . . . . . . . . . . 194 20.2. RTSP Protocol Definition . . . . . . . . . . . . . . . . 196 20.2.1. Generic Protocol elements . . . . . . . . . . . . . 196 20.2.2. Message Syntax . . . . . . . . . . . . . . . . . . . 199 20.2.3. Header Syntax . . . . . . . . . . . . . . . . . . . 203 20.3. SDP extension Syntax . . . . . . . . . . . . . . . . . . 212 21. Security Considerations . . . . . . . . . . . . . . . . . . . 213 21.1. Signaling Protocol Threats . . . . . . . . . . . . . . . 213 21.2. Media Stream Delivery Threats . . . . . . . . . . . . . 216 21.2.1. Remote Denial of Service Attack . . . . . . . . . . 217 21.2.2. RTP Security analysis . . . . . . . . . . . . . . . 218 22. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 220 22.1. Feature-tags . . . . . . . . . . . . . . . . . . . . . . 221 22.1.1. Description . . . . . . . . . . . . . . . . . . . . 221 22.1.2. Registering New Feature-tags with IANA . . . . . . . 221 22.1.3. Registered entries . . . . . . . . . . . . . . . . . 221 22.2. RTSP Methods . . . . . . . . . . . . . . . . . . . . . . 222 22.2.1. Description . . . . . . . . . . . . . . . . . . . . 222 22.2.2. Registering New Methods with IANA . . . . . . . . . 222 22.2.3. Registered Entries . . . . . . . . . . . . . . . . . 222 22.3. RTSP Status Codes . . . . . . . . . . . . . . . . . . . 223 Schulzrinne, et al. Expires March 15, 2014 [Page 7] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.3.1. Description . . . . . . . . . . . . . . . . . . . . 223 22.3.2. Registering New Status Codes with IANA . . . . . . . 223 22.3.3. Registered Entries . . . . . . . . . . . . . . . . . 223 22.4. RTSP Headers . . . . . . . . . . . . . . . . . . . . . . 223 22.4.1. Description . . . . . . . . . . . . . . . . . . . . 224 22.4.2. Registering New Headers with IANA . . . . . . . . . 224 22.4.3. Registered entries . . . . . . . . . . . . . . . . . 224 22.5. Accept-Credentials . . . . . . . . . . . . . . . . . . . 226 22.5.1. Accept-Credentials policies . . . . . . . . . . . . 226 22.5.2. Accept-Credentials hash algorithms . . . . . . . . . 226 22.6. Cache-Control Cache Directive Extensions . . . . . . . . 227 22.7. Media Properties . . . . . . . . . . . . . . . . . . . . 227 22.7.1. Description . . . . . . . . . . . . . . . . . . . . 228 22.7.2. Registration Rules . . . . . . . . . . . . . . . . . 228 22.7.3. Registered Values . . . . . . . . . . . . . . . . . 228 22.8. Notify-Reason header . . . . . . . . . . . . . . . . . . 228 22.8.1. Description . . . . . . . . . . . . . . . . . . . . 228 22.8.2. Registration Rules . . . . . . . . . . . . . . . . . 229 22.8.3. Registered Values . . . . . . . . . . . . . . . . . 229 22.9. Range Header Formats . . . . . . . . . . . . . . . . . . 229 22.9.1. Description . . . . . . . . . . . . . . . . . . . . 229 22.9.2. Registration Rules . . . . . . . . . . . . . . . . . 230 22.9.3. Registered Values . . . . . . . . . . . . . . . . . 230 22.10. Terminate-Reason Header . . . . . . . . . . . . . . . . 230 22.10.1. Redirect Reasons . . . . . . . . . . . . . . . . . . 230 22.10.2. Terminate-Reason Header Parameters . . . . . . . . . 231 22.11. RTP-Info header parameters . . . . . . . . . . . . . . . 231 22.11.1. Description . . . . . . . . . . . . . . . . . . . . 231 22.11.2. Registration Rules . . . . . . . . . . . . . . . . . 231 22.11.3. Registered Values . . . . . . . . . . . . . . . . . 232 22.12. Seek-Style Policies . . . . . . . . . . . . . . . . . . 232 22.12.1. Description . . . . . . . . . . . . . . . . . . . . 232 22.12.2. Registration Rules . . . . . . . . . . . . . . . . . 232 22.12.3. Registered Values . . . . . . . . . . . . . . . . . 232 22.13. Transport Header Registries . . . . . . . . . . . . . . 233 22.13.1. Transport Protocol Identifier . . . . . . . . . . . 233 22.13.2. Transport modes . . . . . . . . . . . . . . . . . . 235 22.13.3. Transport Parameters . . . . . . . . . . . . . . . . 235 22.14. URI Schemes . . . . . . . . . . . . . . . . . . . . . . 236 22.14.1. The rtsp URI Scheme . . . . . . . . . . . . . . . . 236 22.14.2. The rtsps URI Scheme . . . . . . . . . . . . . . . . 237 22.14.3. The rtspu URI Scheme . . . . . . . . . . . . . . . . 239 22.15. SDP attributes . . . . . . . . . . . . . . . . . . . . . 239 22.16. Media Type Registration for text/parameters . . . . . . 240 23. References . . . . . . . . . . . . . . . . . . . . . . . . . 242 23.1. Normative References . . . . . . . . . . . . . . . . . . 242 23.2. Informative References . . . . . . . . . . . . . . . . . 244 Appendix A. Examples . . . . . . . . . . . . . . . . . . . . . . 247 Schulzrinne, et al. Expires March 15, 2014 [Page 8] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 A.1. Media on Demand (Unicast) . . . . . . . . . . . . . . . 247 A.2. Media on Demand using Pipelining . . . . . . . . . . . . 251 A.3. Secured Media Session for on Demand Content . . . . . . 253 A.4. Media on Demand (Unicast) . . . . . . . . . . . . . . . 256 A.5. Single Stream Container Files . . . . . . . . . . . . . 260 A.6. Live Media Presentation Using Multicast . . . . . . . . 262 A.7. Capability Negotiation . . . . . . . . . . . . . . . . . 263 Appendix B. RTSP Protocol State Machine . . . . . . . . . . . . 265 B.1. States . . . . . . . . . . . . . . . . . . . . . . . . . 265 B.2. State variables . . . . . . . . . . . . . . . . . . . . 265 B.3. Abbreviations . . . . . . . . . . . . . . . . . . . . . 266 B.4. State Tables . . . . . . . . . . . . . . . . . . . . . . 266 Appendix C. Media Transport Alternatives . . . . . . . . . . . . 273 C.1. RTP . . . . . . . . . . . . . . . . . . . . . . . . . . 273 C.1.1. AVP . . . . . . . . . . . . . . . . . . . . . . . . 273 C.1.2. AVP/UDP . . . . . . . . . . . . . . . . . . . . . . 273 C.1.3. AVPF/UDP . . . . . . . . . . . . . . . . . . . . . . 275 C.1.4. SAVP/UDP . . . . . . . . . . . . . . . . . . . . . . 275 C.1.5. SAVPF/UDP . . . . . . . . . . . . . . . . . . . . . 278 C.1.6. RTCP usage with RTSP . . . . . . . . . . . . . . . . 278 C.2. RTP over TCP . . . . . . . . . . . . . . . . . . . . . . 280 C.2.1. Interleaved RTP over TCP . . . . . . . . . . . . . . 280 C.2.2. RTP over independent TCP . . . . . . . . . . . . . . 280 C.3. Handling Media Clock Time Jumps in the RTP Media Layer . 284 C.4. Handling RTP Timestamps after PAUSE . . . . . . . . . . 288 C.5. RTSP / RTP Integration . . . . . . . . . . . . . . . . . 290 C.6. Scaling with RTP . . . . . . . . . . . . . . . . . . . . 290 C.7. Maintaining NPT synchronization with RTP timestamps . . 290 C.8. Continuous Audio . . . . . . . . . . . . . . . . . . . . 290 C.9. Multiple Sources in an RTP Session . . . . . . . . . . . 290 C.10. Usage of SSRCs and the RTCP BYE Message During an RTSP Session . . . . . . . . . . . . . . . . . . . . . . 290 C.11. Future Additions . . . . . . . . . . . . . . . . . . . . 291 Appendix D. Use of SDP for RTSP Session Descriptions . . . . . . 292 D.1. Definitions . . . . . . . . . . . . . . . . . . . . . . 292 D.1.1. Control URI . . . . . . . . . . . . . . . . . . . . 292 D.1.2. Media Streams . . . . . . . . . . . . . . . . . . . 293 D.1.3. Payload Type(s) . . . . . . . . . . . . . . . . . . 294 D.1.4. Format-Specific Parameters . . . . . . . . . . . . . 294 D.1.5. Directionality of media stream . . . . . . . . . . . 294 D.1.6. Range of Presentation . . . . . . . . . . . . . . . 295 D.1.7. Time of Availability . . . . . . . . . . . . . . . . 296 D.1.8. Connection Information . . . . . . . . . . . . . . . 296 D.1.9. Message Body Tag . . . . . . . . . . . . . . . . . . 297 D.2. Aggregate Control Not Available . . . . . . . . . . . . 297 D.3. Aggregate Control Available . . . . . . . . . . . . . . 298 D.4. Grouping of Media Lines in SDP . . . . . . . . . . . . . 299 D.5. RTSP external SDP delivery . . . . . . . . . . . . . . . 299 Schulzrinne, et al. Expires March 15, 2014 [Page 9] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix E. RTSP Use Cases . . . . . . . . . . . . . . . . . . . 300 E.1. On-demand Playback of Stored Content . . . . . . . . . . 300 E.2. Unicast Distribution of Live Content . . . . . . . . . . 301 E.3. On-demand Playback using Multicast . . . . . . . . . . . 302 E.4. Inviting an RTSP server into a conference . . . . . . . 302 E.5. Live Content using Multicast . . . . . . . . . . . . . . 303 Appendix F. Text format for Parameters . . . . . . . . . . . . . 305 Appendix G. Requirements for Unreliable Transport of RTSP . . . 306 Appendix H. Backwards Compatibility Considerations . . . . . . . 308 H.1. Play Request in Play State . . . . . . . . . . . . . . . 308 H.2. Using Persistent Connections . . . . . . . . . . . . . . 308 Appendix I. Changes . . . . . . . . . . . . . . . . . . . . . . 309 I.1. Brief Overview . . . . . . . . . . . . . . . . . . . . . 309 I.2. Detailed List of Changes . . . . . . . . . . . . . . . . 310 Appendix J. Acknowledgements . . . . . . . . . . . . . . . . . . 317 J.1. Contributors . . . . . . . . . . . . . . . . . . . . . . 317 Appendix K. RFC Editor Consideration . . . . . . . . . . . . . . 319 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 320 Schulzrinne, et al. Expires March 15, 2014 [Page 10] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 1. Introduction This memo defines version 2.0 of the Real Time Streaming Protocol (RTSP 2.0). RTSP 2.0 is an application-level protocol for setup and control over the delivery of data with real-time properties, typically streaming media. Streaming media is, for instance, video on demand or audio live streaming. Put simply, RTSP acts as a "network remote control" for multimedia servers, similar to the remote control for a DVD player. The protocol operates between RTSP 2.0 clients and servers, but also supports the usage of proxies placed between clients and servers. Clients can request information about streaming media from servers by asking for a description of the media or use media description provided externally. The media delivery protocol is used to establish the media streams described by the media description. Clients can then request to play out the media, pause it, or stop it completely, as known from DVD players remote control or media players. The requested media can consist of multiple audio and video streams that are delivered as time-synchronized streams from servers to clients. RTSP 2.0 is a replacement of RTSP 1.0 [RFC2326] and obsoletes that specification. This protocol is based on RTSP 1.0 but is not backwards compatible other than in the basic version negotiation mechanism. The changes are documented in Appendix I. There are many reasons why RTSP 2.0 can't be backwards compatible with RTSP 1.0 but some of the main ones are: o Most headers that needed to be extensible did not define the allowed syntax, preventing safe deployment of extensions; o The changed behavior of the PLAY method when received in Play state; o Changed behavior of the extensibility model and its mechanism; o The change of syntax for some headers. In summary, there are so many small details that changing version became necessary to enable clarification and consistent behavior. This document is structured as follows. It begins with an overview of the protocol operations and its functions in an informal way. Then a set of definitions of terms used and document conventions is introduced. It is followed by the actual RTSP 2.0 core protocol specification. The appendixes describe and define some functionalities that are not part of the core RTSP specification, but Schulzrinne, et al. Expires March 15, 2014 [Page 11] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 which are still important to enable some usages. Among them, the RTP usage is defined in Appendix C, the Session Description Protocol (SDP) usage with RTSP is defined in Appendix D, and the text/ parameters file format Appendix F, are three normative specification appendixes. Others include a number of informational parts discussing the changes, use cases, different considerations or motivations. Schulzrinne, et al. Expires March 15, 2014 [Page 12] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 2. Protocol Overview This section provides an informative overview of the different mechanisms in the RTSP 2.0 protocol, to give the reader a high level understanding before getting into all the different details. In case of conflict with this description and the later sections, the later sections take precedence. For more information about use cases considered for RTSP see Appendix E. RTSP 2.0 is a bi-directional request and response protocol that first establishes a context including content resources (the media) and then controls the delivery of these content resources from the provider to the consumer. RTSP has three fundamental parts: Session Establishment, Media Delivery Control, and an extensibility model described below. The protocol is based on some assumptions about existing functionality to provide a complete solution for client controlled real-time media delivery. RTSP uses text-based messages, requests and responses, that may contain a binary message body. An RTSP request starts with a method line that identifies the method, the protocol and version and the resource to act on. The resource is identified by an URI and the hostname part of the URI is used by RTSP client to resolve the IPv4 or IPv6 address of the RTSP server. Following the method line are a number of RTSP headers. This part is ended by two consecutive carriage return line feed (CRLF) character pairs. The message body if present follows the two CRLF and the body's length is described by a message header. RTSP responses are similar, but start with a response line with the protocol and version, followed by a status code and a reason phrase. RTSP messages are sent over a reliable transport protocol between the client and server. RTSP 2.0 requires clients and servers to implement TCP, and TLS over TCP, as mandatory transports for RTSP messages. 2.1. Presentation Description RTSP exists to provide access to multi-media presentations and content, but tries to be agnostic about the media type or the actual media delivery protocol that is used. To enable a client to implement a complete system, an RTSP-external mechanism for describing the presentation and the delivery protocol(s) is used. RTSP assumes that this description is either delivered completely out of band or as a data object in the response to a client's request using the DESCRIBE method (Section 13.2). Parameters that commonly have to be included in the Presentation Description are the following: Schulzrinne, et al. Expires March 15, 2014 [Page 13] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o Number of media streams; o The resource identifier for each media stream/resource that is to be controlled by RTSP; o The protocol that each media stream is to be delivered over; o Transport protocol parameters that are not negotiated or vary with each client; o Media encoding information enabling a client to correctly decode the media upon reception; o An aggregate control resource identifier. RTSP uses its own URI schemes ("rtsp" and "rtsps") to reference media resources and aggregates under common control (See Section 4.2). This specification describes in Appendix D how one uses SDP [RFC4566] for Presentation Description 2.2. Session Establishment The RTSP client can request the establishment of an RTSP session after having used the presentation description to determine which media streams are available, and also which media delivery protocol is used and their particular resource identifiers. The RTSP session is a common context between the client and the server that consists of one or more media resources that are to be under common media delivery control. The client creates an RTSP session by sending a request using the SETUP method (Section 13.3) to the server. In the SETUP request the client also includes all the transport parameters necessary to enable the media delivery protocol to function in the "Transport" header (Section 18.54). This includes parameters that are pre-established by the presentation description but necessary for any middlebox to correctly handle the media delivery protocols. The Transport header in a request may contain multiple alternatives for media delivery in a prioritized list, which the server can select from. These alternatives are typically based on information in the presentation description. The server determines if the media resource is available upon receiving a SETUP request and if any of the transport parameter specifications are acceptable. If that is successful, an RTSP session context is created and the relevant parameters and state is stored. An identifier is created for the RTSP session and included Schulzrinne, et al. Expires March 15, 2014 [Page 14] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 in the response in the Session header (Section 18.49). The SETUP response includes a Transport header that specifies which of the alternatives has been selected and relevant parameters. A SETUP request that references an existing RTSP session but identifies a new media resource is a request to add that media resource under common control with the already present media resources in an aggregated session. A client can expect this to work for all media resources under RTSP control within a multi-media content. However, aggregating resources from different content are likely to be refused by the server. The RTSP session as aggregate is referenced by the aggregate control URI, even if the RTSP session only contains a single media. To avoid an extra round trip in the session establishment of aggregated RTSP sessions, RTSP 2.0 supports pipelined requests; i.e., the client can send multiple requests back-to-back without waiting first for the completion of any of them. The client uses a client- selected identifier in the Pipelined-Requests header (Section 18.33) to instruct the server to bind multiple requests together as if they included the session identifier. The SETUP response also provides additional information about the established sessions in a couple of different headers. The Media- Properties header (Section 18.29) includes a number of properties that apply for the aggregate that is valuable when doing media delivery control and configuring user interface. The Accept-Ranges header (Section 18.5) informs the client about which range formats that the server supports with these media resources. The Media-Range header (Section 18.30) informs the client about the time range of the media currently available. 2.3. Media Delivery Control After having established an RTSP session, the client can start controlling the media delivery. The basic operations are Start by using the PLAY method (Section 13.4) and Halt by using the PAUSE method (Section 13.6). PLAY also allows for choosing the starting media position from which the server should deliver the media. The positioning is done by using the Range header (Section 18.40) that supports several different time formats: Normal Play Time (NPT) (Section 4.4.2), Society of Motion Picture and Television Engineers (SMPTE) Timestamps (Section 4.4.1) and absolute time (Section 4.4.3). The Range header does further allow the client to specify a position where delivery should end, thus allowing a specific interval to be delivered. The support for positioning/searching within a content depends on the Schulzrinne, et al. Expires March 15, 2014 [Page 15] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 content's media properties. Content exists in a number of different types, such as: on-demand, live, and live with simultaneous recording. Even within these categories there are differences in how the content is generated and distributed, which affect how it can be accessed for playback. The properties applicable for the RTSP session are provided by the server in the SETUP response using the Media-Properties header (Section 18.29). These are expressed using one or several independent attributes. A first attribute is Random Access, which expresses if positioning can be done, and with what granularity. Another aspect is whether the content will change during the lifetime of the session. While on-demand content will be provided in full from the beginning, a live stream being recorded results in the length of the accessible content growing as the session goes on. There also exists content that is dynamically built by another protocol than RTSP and thus also changes in steps during the session, but maybe not continuously. Furthermore, when content is recorded, there are cases where not the complete content is maintained, but, for example, only the last hour. All these properties result in the need for mechanisms that will be discussed below. When the client accesses on-demand content that allows random access, the client can issue the PLAY request for any point in the content between the start and the end. The server will deliver media from the closest random access point prior to the requested point and indicate that in its PLAY response. If the client issues a PAUSE, the delivery will be halted and the point at which the server stopped will be reported back in the response. The client can later resume by sending a PLAY request without a range header. When the server is about to complete the PLAY request by delivering the end of the content or the requested range, the server will send a PLAY_NOTIFY request (Section 13.5) indicating this. When playing live content with no extra functions, such as recording, the client will receive the live media from the server after having sent a PLAY request. Seeking in such content is not possible as the server does not store it, but only forwards it from the source of the session. Thus delivery continues until the client sends a PAUSE request, tears down the session, or the content ends. For live sessions that are being recorded the client will need to keep track of how the recording progresses. Upon session establishment the client will learn the current duration of the recording from the Media-Range header. As the recording is ongoing the content grows in direct relation to the passed time. Therefore, each server's response to a PLAY request will contain the current Media-Range header. The server should also regularly send approximately every 5 minutes the current media range in a Schulzrinne, et al. Expires March 15, 2014 [Page 16] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 PLAY_NOTIFY request (Section 13.5.2). If the live transmission ends, the server must send a PLAY_NOTIFY request with the updated Media- Properties indicating that the content stopped being a recorded live session and instead became on-demand content; the request also contains the final media range. While the live delivery continues the client can request to play the current live point by using the NPT timescale symbol "now", or it can request a specific point in the available content by an explicit range request for that point. If the requested point is outside of the available interval the server will adjust the position to the closest available point, i.e., either at the beginning or the end. A special case of recording is that where the recording is not retained longer than a specific time period, thus as the live delivery continues the client can access any media within a moving window that covers, for example, "now" to "now" minus 1 hour. A client that pauses on a specific point within the content may not be able to retrieve the content anymore. If the client waits too long before resuming the pause point, the content may no longer be available. In this case the pause point will be adjusted to the closest point in the available media. 2.4. Session Parameter Manipulations A session may have additional state or functionality that affects how the server or client treats the session, content, how it functions, or feedback on how well the session works. Such extensions are not defined in this specification, but may be done in various extensions. RTSP has two methods for retrieving and setting parameter values on either the client or the server: GET_PARAMETER (Section 13.8) and SET_PARAMETER (Section 13.9). These methods carry the parameters in a message body of the appropriate format. One can also use headers to query state with the GET_PARAMETER method. As an example, clients needing to know the current media-range for a time-progressing session can use the GET_PARAMETER method and include the media-range. Furthermore, synchronization information can be requested by using a combination of RTP-Info (Section 18.45) and Range (Section 18.40). RTSP 2.0 does not have a strong mechanism for providing negotiation of the headers, or parameters and their formats, which can be used. However, responses will indicate request headers or parameters that are not supported. A priori determination of what features are available needs to be done through out-of-band mechanisms, like the session description, or through the usage of feature tags (Section 4.5). Schulzrinne, et al. Expires March 15, 2014 [Page 17] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 2.5. Media Delivery The delivery of media to the RTSP client is done with a protocol outside of RTSP and this protocol is determined during the session establishment. This document specifies how media is delivered with RTP [RFC3550] over UDP [RFC0768], TCP [RFC0793] or the RTSP connection. Additional protocols may be specified in the future based on demand. The usage of RTP as media delivery protocol requires some additional information to function well. The PLAY response contains information to enable reliable and timely delivery of how a client should synchronize different sources in the different RTP sessions. It also provides a mapping between RTP timestamps and the content time scale. When the server wants to notify the client about the completion of the media delivery, it sends a PLAY_NOTIFY request to the client. The PLAY_NOTIFY request includes information about the stream end, including the last RTP sequence number for each stream, thus enabling the client to empty the buffer smoothly. 2.5.1. Media Delivery Manipulations The basic playback functionality of RTSP enables delivery of a range of requested content to the client at the pace intended by the content's creator. However, RTSP can also manipulate the delivery to the client in two ways. Scale: The ratio of media content time delivered per unit playback time. Speed: The ratio of playback time delivered per unit of wallclock time. Both affect the media delivery per time unit. However, they manipulate two independent time scales and the effects are possible to combine. Scale (Section 18.46) is used for fast forward or slow motion control as it changes the amount of content timescale that should be played back per time unit. Scale > 1.0, means fast forward, e.g., Scale=2.0 results in that 2 seconds of content is played back every second of playback. Scale = 1.0 is the default value that is used if no Scale is specified, i.e., playback at the content's original rate. Scale values between 0 and 1.0 is providing for slow motion. Scale can be negative to allow for reverse playback in either regular pace (Scale = -1.0) or fast backwards (Scale < -1.0) or slow motion backwards (-1.0 < Scale < 0). Scale = 0 is equal to pause and is not allowed. Schulzrinne, et al. Expires March 15, 2014 [Page 18] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 In most cases the realization of scale means server side manipulation of the media to ensure that the client can actually play it back. The nature of these media manipulations and when they are needed is highly media-type dependent. Let's consider an example with two common media types audio and video. It is very difficult to modify the playback rate of audio. A maximum of 10-30% is possible by changing the pitch-rate of speech. Music goes out of tune if one tries to manipulate the playback rate by resampling it. This is a well known problem and audio is commonly muted or played back in short segments with skips to keep up with the current playback point. For video it is possible to manipulate the frame rate, although the rendering capabilities are often limited to certain frame rates. Also the allowed bitrates in decoding, the structure used in the encoding and the dependency between frames and other capabilities of the rendering device limits the possible manipulations. Therefore, the basic fast forward capabilities often are implemented by selecting certain subsets of frames. Due to the media restrictions, the possible scale values are commonly restricted to the set of realizable scale ratios. To enable the clients to select from the possible scale values, RTSP can signal the supported Scale ratios for the content. To support aggregated or dynamic content, where this may change during the ongoing session and dependent on the location within the content, a mechanism for updating the media properties and the scale factor currently in use, exists. Speed (Section 18.50) affects how much of the playback timeline is delivered in a given wallclock period. The default is Speed = 1 which means to deliver at the same rate the media is consumed. Speed > 1 means that the receiver will get content faster than it regularly would consume it. Speed < 1 means that delivery is slower than the regular media rate. Speed values of 0 or lower have no meaning and are not allowed. This mechanism enables two general functionalities. One is client side scale operations, i.e., the client receives all the frames and makes the adjustment to the playback locally. The second is delivery control for buffering of media. By specifying a speed over 1.0 the client can build up the amount of playback time it has present in its buffers to a level that is sufficient for its needs. A naive implementation of Speed would only affect the transmission schedule of the media and has a clear impact on the needed bandwidth. This would result in the data rate being proportional to the speed factor. Speed = 1.5, i.e., 50% faster than normal delivery, would Schulzrinne, et al. Expires March 15, 2014 [Page 19] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 result in a 50% increase in the data transport rate. If that can be supported or not depends solely on the underlying network path. Scale may also have some impact on the required bandwidth due to the manipulation of the content in the new playback schedule. An example is fast forward where only the independently decodable intra frames are included in the media stream. This usage of solely intra frames increases the data rate significantly compared to a normal sequence with the same number of frames, where most frames are encoded using prediction. This potential increase of the data rate needs to be handled by the media sender. The client has requested that the media will be delivered in a specific way, which should be honored. However, the media sender cannot ignore if the network path between the sender and the receiver can't handle the resulting media stream. In that case the media stream needs to be adapted to fit the available resources of the path. This can result in a reduced media quality. The need for bitrate adaptation becomes especially problematic in connection with the Speed semantics. If the goal is to fill up the buffer, the client may not want to do that at the cost of reduced quality. If the client wants to make local playout changes then it may actually require that the requested speed be honored. To resolve this issue, Speed uses a range so that both cases can be supported. The server is requested to use the highest possible speed value within the range which is compatible with the available bandwidth. As long as the server can maintain a speed value within the range it shall not change the media quality, but instead modify the actual delivery rate in response to available bandwidth and reflect this in the Speed value in the response. However, if this is not possible, the server should instead modify the media quality to respect the lowest speed value and the available bandwidth. This functionality enables the local scaling implementation to use a tight range, or even a range where the lower bound equals the upper bound, to identify that it requires the server to deliver the requested amount of media time per delivery time independent of how much it needs to adapt the media quality to fit within the available path bandwidth. For buffer filling, it is suitable to use a range with a reasonable span and with a lower bound at the nominal media rate 1.0, such as 1.0 - 2.5. If the client wants to reduce the buffer, it can specify an upper bound that is below 1.0 to force the server to deliver slower than the nominal media rate. 2.6. Session Maintenance and Termination The session context that has been established is kept alive by having the client show liveness. This is done in two main ways: Schulzrinne, et al. Expires March 15, 2014 [Page 20] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o Media transport protocol keep-alive. RTP Control Protocol (RTCP) may be used when using RTP. o Any RTSP request referencing the session context. Section 10.5 discusses the methods for showing liveness in more depth. If the client fails to show liveness for more than the established session timeout value (normally 60 seconds), the server may terminate the context. Other values may be selected by the server through the inclusion of the timeout parameter in the session header. The session context is normally terminated by the client sending a TEARDOWN request (Section 13.7) to the server referencing the aggregated control URI. An individual media resource can be removed from a session context by a TEARDOWN request referencing that particular media resource. If all media resources are removed from a session context, the session context is terminated. A client may keep the session alive indefinitely if allowed by the server; however, it is recommended to release the session context when an extended period of time without media delivery activity has passed. The client can re-establish the session context if required later. What constitutes an extended period of time is dependent on the server and its usage. It is recommended that the client terminates the session before ten times the session timeout value has passed. A server may terminate the session after one session timeout period without any client activity beyond keep-alive. When a server terminates the session context, it does that by sending a TEARDOWN request indicating the reason. A server can also request that the client tear down the session and re-establish it at an alternative server, as may be needed for maintenance. This is done by using the REDIRECT method (Section 13.10). The Terminate-Reason header (Section 18.52) is used to indicate when and why. The Location header indicates where it should connect if there is an alternative server available. When the deadline expires, the server simply stops providing the service. To achieve a clean closure, the client needs to initiate session termination prior to the deadline. In case the server has no other server to redirect to, and wants to close the session for maintenance, it shall use the TEARDOWN method with a Terminate-Reason header. 2.7. Extending RTSP RTSP is quite a versatile protocol which supports extensions in many different directions. Even this core specification contains several Schulzrinne, et al. Expires March 15, 2014 [Page 21] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 blocks of functionality that are optional to implement. The use case and need for the protocol deployment should determine what parts are implemented. Allowing for extensions makes it possible for RTSP to reach out to additional use cases. However, extensions will affect the interoperability of the protocol and therefore it is important that they can be added in a structured way. The client can learn the capability of a server by using the OPTIONS method (Section 13.1) and the Supported header (Section 18.51). It can also try and possibly fail using new methods, or require that particular features are supported using the Require (Section 18.43) or Proxy-Require (Section 18.37) header. The RTSP protocol in itself can be extended in three ways, listed here in increasing order of the magnitude of changes supported: o Existing methods can be extended with new parameters, for example, headers, as long as these parameters can be safely ignored by the recipient. If the client needs negative acknowledgment when a method extension is not supported, a tag corresponding to the extension may be added in the field of the Require or Proxy- Require headers. o New methods can be added. If the recipient of the message does not understand the request, it must respond with error code 501 (Not Implemented) so that the sender can avoid using this method again. A client may also use the OPTIONS method to inquire about methods supported by the server. The server must list the methods it supports using the Public response header. o A new version of the protocol can be defined, allowing almost all aspects (except the position of the protocol version number) to change. A new version of the protocol must be registered through an IETF standards track document. The basic capability discovery mechanism can be used to both discover support for a certain feature and to ensure that a feature is available when performing a request. For a detailed explanation of this see Section 11. New media delivery protocols may be added and negotiated at session establishment, in addition to extensions to the core protocol. Certain types of protocol manipulations can be done through parameter formats using SET_PARAMETER and GET_PARAMETER. Schulzrinne, et al. Expires March 15, 2014 [Page 22] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 3. Document Conventions 3.1. Notational Conventions Since a few of the definitions are identical to HTTP/1.1, this specification only points to the section where they are defined rather than copying it. For brevity, [HX.Y] is to be taken to refer to Section X.Y of the current HTTP/1.1 specification ([RFC2616]). All the mechanisms specified in this document are described in both prose and the Augmented Backus-Naur form (ABNF) described in detail in [RFC5234]. Indented paragraphs are used to provide informative background and motivation. This is intended to give readers who were not involved with the formulation of the specification an understanding of why things are the way they are in RTSP. The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119]. The word, "unspecified" is used to indicate functionality or features that are not defined in this specification. Such functionality cannot be used in a standardized manner without further definition in an extension specification to RTSP. 3.2. Terminology Aggregate control: The concept of controlling multiple streams using a single timeline, generally maintained by the server. A client, for example, uses aggregate control when it issues a single play or pause message to simultaneously control both the audio and video in a movie. A session which is under aggregate control is referred to as an aggregated session. Aggregate control URI: The URI used in an RTSP request to refer to and control an aggregated session. It normally, but not always, corresponds to the presentation URI specified in the session description. See Section 13.3 for more information. Client: The client requests media service from the media server. Connection: A transport layer virtual circuit established between two programs for the purpose of communication. Schulzrinne, et al. Expires March 15, 2014 [Page 23] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Container file: A file which may contain multiple media streams which often constitutes a presentation when played together. The concept of a container file is not embedded in the protocol. However, RTSP servers may offer aggregate control on the media streams within these files. Continuous media: Data where there is a timing relationship between source and sink; that is, the sink needs to reproduce the timing relationship that existed at the source. The most common examples of continuous media are audio and motion video. Continuous media can be real-time (interactive or conversational), where there is a "tight" timing relationship between source and sink, or streaming where the relationship is less strict. Feature-tag: A tag representing a certain set of functionality, i.e., a feature. IRI: Internationalized Resource Identifier, is 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 delivery. Media server: The server providing media delivery services for one or more media streams. Different media streams within a presentation may originate from different media servers. A media server may reside on the same host or on a different host from which the presentation is invoked. (Media) stream: A single media instance, e.g., an audio stream or a video stream as well as a single whiteboard or shared application group. When using RTP, a stream consists of all RTP and RTCP packets created by a source within an RTP session. Schulzrinne, et al. Expires March 15, 2014 [Page 24] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Message: The basic unit of RTSP communication, consisting of a structured sequence of octets matching the syntax defined in Section 20 and transmitted over a connection-based transport. A message is either a Request or a Response. Message Body: The information transferred as the payload of a message (Request or response). A message body consists of meta- information in the form of message-body headers and content in the form of a message-body, as described in Section 9. Non-Aggregated Control: Control of a single media stream. Presentation: A set of one or more streams presented to the client as a complete media feed and described by a presentation description as defined below. Presentations with more than one media stream are often handled in RTSP under aggregate control. Presentation description: A presentation description contains information about one or more media streams within a presentation, such as the set of encodings, network addresses and information about the content. Other IETF protocols such as SDP ([RFC4566]) use the term "session" for a presentation. The presentation description may take several different formats, including but not limited to the session description protocol format, SDP. Response: An RTSP response to a Request. One type of RTSP message. If an HTTP response is meant, it is indicated explicitly. Request: An RTSP request. One type of RTSP message. If an HTTP request is meant, it is indicated explicitly. Request-URI: The URI used in a request to indicate the resource on which the request is to be performed. RTSP agent: Refers to either an RTSP client, an RTSP server, or an RTSP proxy. In this specification, there are many capabilities that are common to these three entities such as the capability to send requests or receive responses. This term will be used when describing functionality that is applicable to all three of these entities. RTSP session: A stateful abstraction upon which the main control methods of RTSP operate. An RTSP session is a common context; it is created and maintained on client's request and can be destroyed by either the client or server. It is established by an RTSP server upon the completion of a successful SETUP request (when a 200 OK response is sent) and is labeled with a session identifier at that time. The session exists until timed out by the server or Schulzrinne, et al. Expires March 15, 2014 [Page 25] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 explicitly removed by a TEARDOWN request. An RTSP session is a stateful entity; an RTSP server maintains an explicit session state machine (see Appendix B) where most state transitions are triggered by client requests. The existence of a session implies the existence of state about the session's media streams and their respective transport mechanisms. A given session can have one or more media streams associated with it. An RTSP server uses the session to aggregate control over multiple media streams. Origin Server: The server on which a given resource resides. Transport initialization: The negotiation of transport information (e.g., port numbers, transport protocols) between the client and the server. URI: Universal Resource Identifier, see [RFC3986]. The URIs used in RTSP are generally URLs as they give a location for the resource. As URLs are a subset of URIs, they will be referred to as URIs to cover also the cases when an RTSP URI would not be an URL. URL: Universal Resource Locator, is 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 March 15, 2014 [Page 26] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 4. Protocol Parameters 4.1. RTSP Version This specification defines version 2.0 of RTSP. RTSP uses a "." numbering scheme to indicate versions of the protocol. The protocol versioning policy is intended to allow the sender to indicate the format of a message and its capacity for understanding further RTSP communication, rather than the features obtained via that communication. No change is made to the version number for the addition of message components which do not affect communication behavior or which only add to extensible field values. The number is incremented when the changes made to the protocol add features which do not change the general message parsing algorithm, but which may add to the message semantics and imply additional capabilities of the sender. The number is incremented when the format of a message within the protocol is changed. The version of an RTSP message is indicated by an RTSP- Version field in the first line of the message. Note that the major and minor numbers MUST be treated as separate integers and that each MAY be incremented higher than a single digit. Thus, RTSP/2.4 is a lower version than RTSP/2.13, which in turn is lower than RTSP/12.3. Leading zeros MUST be ignored by recipients and MUST NOT be sent. 4.2. RTSP IRI and URI RTSP 2.0 defines and registers or updates three URI schemes "rtsp", "rtsps" and "rtspu". The usage of the last, "rtspu", is unspecified in RTSP 2.0, and is defined here to register the URI scheme that was defined in RTSP 1.0. The "rtspu" scheme indicates unspecified transport of the RTSP messages over unreliable transport (UDP in RTSP 1.0). An RTSP server MUST respond with an error code indicating the "rtspu" scheme is not implemented (501) to a request that carries a "rtspu" URI scheme. The details of the syntax of "rtsp" and "rtsps" URIs has been changed from RTSP 1.0. These changes are: o Support for IPV6 literal in host part and future IP literals through RFC 3986 defined mechanism. o A new relative format to use in the RTSP protocol elements that is not required to start with "/". Neither should have any significant impact on interoperability. If one is required to use IPv6 literals to reach an RTSP server, then Schulzrinne, et al. Expires March 15, 2014 [Page 27] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 that RTSP server must be IPv6 capable, and RTSP 1.0 is not a fully IPv6 capable protocol. If an RTSP 1.0 client attempts to process the URI it will not match the allowed syntax and be considered invalid and processing will be stopped. This is clearly a failure to reach the resource, however it is not a signification issue as RTSP 2.0 support was needed anyway in both server and client. Thus failure will only occur in a later step when there is a RTSP version mismatch between client and server. The second change will only occur inside RTSP message headers, as the request URI must be an absolute URI. Thus such usages are occurring after agents has accepted processing RTSP 2.0 messages, and an RTSP 1.0 only agent will not be required to parse such types of relative URIs. This specification also defines the format of the RTSP IRI [RFC3987] that can be used as RTSP resource identifiers and locators, in web pages, user interfaces, on paper, etc. However, the RTSP request message format only allows usage of the absolute URI format. The RTSP IRI format MUST use the rules and transformation for IRIs to URIs, as defined in [RFC3987]. This allows a URI that matches the RTSP 2.0 specification, and so is suitable for use in a request, to be created from an RTSP IRI. The RTSP IRI and URI are both syntax restricted compared to the generic syntax defined in [RFC3986] and [RFC3987]: o An absolute URI requires the authority part; i.e., a host identity MUST be provided. o Parameters in the path element are prefixed with the reserved separator ";". The RTSP URI and IRI are case sensitive, with the exception of those parts that [RFC3986] and [RFC3987] define as case-insensitive; for example, the scheme and host part. The fragment identifier is used as defined in sections 3.5 and 4.3 of [RFC3986], i.e., the fragment is to be stripped from the IRI by the requester and not included in the request URI. The user agent needs to interpret the value of the fragment based on the media type the request relates to; i.e., the media type indicated in Content-Type header in the response to DESCRIBE. The syntax of any URI query string is unspecified and responder (usually the server) specific. The query is, from the requester's perspective, an opaque string and needs to be handled as such. Please note that relative URI with queries are difficult to handle due to the RFC 3986 relative URI handling rules. Any change of the path element using a relative URI results in the stripping of the Schulzrinne, et al. Expires March 15, 2014 [Page 28] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 query, which means the relative part needs to contain the query. The URI scheme "rtsp" requires that commands are issued via a reliable protocol (within the Internet, TCP), while the scheme "rtsps" identifies a reliable transport using secure transport (TLS [RFC5246], see (Section 19). For the scheme "rtsp", if no port number is provided in the authority part of the URI, the port number 554 MUST be used. For the scheme "rtsps", if no port number is provided in the authority part of the URI port number, the TCP port 322 MUST be used. A presentation or a stream is identified by a textual media identifier, using the character set and escape conventions of URIs [RFC3986]. URIs may refer to a stream or an aggregate of streams; i.e., a presentation. Accordingly, requests described in (Section 13) can apply to either the whole presentation or an individual stream within the presentation. Note that some request methods can only be applied to streams, not presentations, and vice versa. For example, the RTSP URI: 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. Schulzrinne, et al. Expires March 15, 2014 [Page 29] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 This decoupling also allows presentation descriptions to be used with non-RTSP media control protocols simply by replacing the scheme in the URI. 4.3. Session Identifiers Session identifiers are strings of length 8-128 characters. A session identifier MUST be chosen cryptographically random (see [RFC4086]). It is RECOMMENDED that it contains 128 bits of entropy, i.e., approximately 22 characters from a high quality generator (see Section 21). However, note that the session identifier does not provide any security against session hijacking unless it is kept confidential by the client, server and trusted proxies. 4.4. Media Time Formats RTSP currently supports three different media time formats defined below. Additional time formats may be specified in the future. These time formats can be used with the Range header (Section 18.40) to request playback and specify at which media position protocol requests actually will or has taken place. They are also used in description of the media's properties using the Media-Range header (Section 18.30). The format identifier only are used in Accept- Ranges header (Section 18.5) to declare supported time formats and also in the Range header (Section 18.40) to request the time format used in the response. 4.4.1. SMPTE Relative Timestamps A Society of Motion Picture and Television Engineers (SMPTE) relative timestamp expresses time relative to the start of the clip. Relative timestamps are expressed as SMPTE time codes [SMPTE_TC] for frame- level access accuracy. The time code has the format hours:minutes:seconds:frames.subframes, with the origin at the start of the clip. The default SMPTE format is "SMPTE 30 drop" format, with frame rate is 29.97 frames per second. Other SMPTE codes MAY be supported (such as "SMPTE 25") through the use of "smpte-type". For SMPTE 30, the "frames" field in the time value can assume the values 0 through 29. The difference between 30 and 29.97 frames per second is handled by dropping the first two frame indices (values 00 and 01) of every minute, except every tenth minute. If the frame and the subframe values are zero, they may be omitted. Subframes are measured in one-hundredth of a frame. Examples: Schulzrinne, et al. Expires March 15, 2014 [Page 30] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 smpte=10:12:33:20- smpte=10:07:33- smpte=10:07:00-10:07:33:05.01 smpte-25=10:07:00-10:07:33:05.01 4.4.2. Normal Play Time Normal play time (NPT) indicates the stream absolute position relative to the beginning of the presentation, not to be confused with the Network Time Protocol (NTP) [RFC5905]. The timestamp consists of two parts: the mandatory first part may be expressed in either seconds or hours, minutes, and seconds. The optional second part consists of a decimal point and decimal figures and indicates fractions of a second. The beginning of a presentation corresponds to 0.0 seconds. Negative values are not defined. The special constant "now" is defined as the current instant of a live event. It MAY only be used for live events, and MUST NOT be used for on-demand (i.e., non-live) content. NPT is defined as in DSM-CC [ISO.13818-6.1995]: "Intuitively, NPT is the clock the viewer associates with a program. It is often digitally displayed on a VCR. NPT advances normally when in normal play mode (scale = 1), advances at a faster rate when in fast scan forward (high positive scale ratio), decrements when in scan reverse (negative scale ratio) and is fixed in pause mode. NPT is (logically) equivalent to SMPTE time codes." Examples: npt=123.45-125 npt=12:05:35.3- npt=now- The syntax 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. 4.4.3. Absolute Time Absolute time is expressed as ISO 8601 [ISO.8601.2000] timestamps, using UTC (GMT). Fractions of a second may be indicated. Schulzrinne, et al. Expires March 15, 2014 [Page 31] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Example for clock format range request for a starting time of November 8, 1996 at 14h 37 min and 20 and a quarter seconds UTC playing for 10 min and 5 seconds, a Media-Properties header's "Time- Limited UTC property for 24th of December 2014 at 15 hours and 00 mins, and a Terminate-Readon headers "time" property for 18th of June 2013 at 16 hours, 12 minutes and 56 seconds: clock=19961108T143720.25Z-19961108T144725.25Z Time-Limited=20141224T1500Z time=20130618T161256Z 4.5. Feature-Tags Feature-tags are unique identifiers used to designate features in RTSP. These tags are used in Require (Section 18.43), Proxy-Require (Section 18.37), Proxy-Supported (Section 18.38), Supported (Section 18.51) and Unsupported (Section 18.55) header fields. A feature-tag definition MUST indicate which combination of clients, servers or proxies it applies to. The creator of a new RTSP feature-tag should either prefix the feature-tag with a reverse domain name (e.g., "com.example.mynewfeature" is an apt name for a feature whose inventor can be reached at "example.com"), or register the new feature-tag with the Internet Assigned Numbers Authority (IANA) (see IANA Section 22). The usage of feature-tags is further described in Section 11 that deals with capability handling. 4.6. Message Body Tags Message body tags are opaque strings that are used to compare two message bodies from the same resource, for example in caches or to optimize setup after a redirect. Message body tags can be carried in the MTag header (see Section 18.31) or in SDP (see Appendix D.1.9). MTag is similar to ETag in HTTP/1.1 (see Section 3.11 of [RFC2068]). A message body tag MUST be unique across all versions of all message bodies associated with a particular resource. A given message body tag value MAY be used for message bodies obtained by requests on different URIs. The use of the same message body tag value in conjunction with message bodies obtained by requests on different URIs does not imply the equivalence of those message bodies Message body tags are used in RTSP to make some methods conditional. The methods are made conditional through the inclusion of headers; Schulzrinne, et al. Expires March 15, 2014 [Page 32] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 see "If-Match" (Section 18.24) and "If-None-Match" (Section 18.26). Note that RTSP message body tags apply to the complete presentation; i.e., both the presentation description and the individual media streams. Thus message body tags can be used to verify at setup time after a redirect that the same session description applies to the media at the new location using the If-Match header. 4.7. Media Properties When an RTSP server handles media, it is important to consider the different properties a media instance for delivery and playback can have. This specification considers the media properties listed below in its protocol operations. They are derived from the differences between a number of supported usages. On-demand: Media that has a fixed (given) duration that doesn't change during the life time of the RTSP session and is known at the time of the creation of the session. It is expected that the content of the media will not change, even if the representation, i.e encoding, quality, etc, may change. Generally one can seek, i.e., request any range, within the media. Dynamic On-demand: This is a variation of the on-demand case where external methods are used to manipulate the actual content of the media setup for the RTSP session. The main example is a content defined by a playlist. Live: Live media represents a progressing content stream (such as broadcast TV) where the duration may or may not be known. It is not seekable, only the content presently being delivered can be accessed. Live with Recording: A Live stream that is combined with a server- side capability to store and retain the content of the live session, and allow for random access delivery within the part of the already recorded content. The actual behavior of the media stream is very much dependent on the retention policy for the media stream; either the server will be able to capture the complete media stream, or it will have a limitation in how much will be retained. The media range will dynamically change as the session progress. For servers with a limited amount of storage available for recording, there will typically be a sliding window that moves forward while new data is made available and older data is discarded. To cover the above usages, the following media properties with appropriate values are specified: Schulzrinne, et al. Expires March 15, 2014 [Page 33] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 4.7.1. Random Access and Seeking Random Access is the ability to specify and get media delivered starting from any time instant within the content, an operation called seeking. The Media-Properties header will indicate the general capability for a media resource to perform random access: Random-Access: The media is seekable to any out of a large number of points within the media. Due to media encoding limitations, a particular point may not be reachable, but seeking to a point close by is enabled. A floating point number of seconds may be provided to express the worst case distance between random access points. Beginning-Only: Seeking is only possible to the beginning of the content. No-seeking: Seeking is not possible at all. If random access is possible, as indicated by Media-Properties header, the actual behavior policy when seeking can be controlled using the Seek-Style header (Section 18.47). 4.7.2. Retention Media may have different retention policies in place that affect the operation on media. The following different media retention policies are envisioned and taken into consideration where applicable: Unlimited: The media will not be removed as long as the RTSP session is in existence. Time-Limited: The media will not be removed before the given wallclock time. After that time it may or may not be available any more. Time-Duration: Each individual unit of the media will be retained for the specified duration. 4.7.3. Content Modifications There is also the question of how the content may change over time for a given media resource: Schulzrinne, et al. Expires March 15, 2014 [Page 34] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Immutable: The content of the media will not change, even if the representation, i.e., encoding, quality, etc., may change. Dynamic: Between explicit updates the media content will not change, but the content may change due to external methods or triggers, such as playlists. Time-Progressing: As time progresses new content will become available. If the content also is retained it will become longer as everything between the start point and the point currently being made available can be accessed. If the media server uses a sliding window policy for retention, the start point will also change as time progresses. 4.7.4. Supported Scale Factors Content often supports only a limited set or range of scales when delivering the media.. To enable the client to know what values or ranges of scale operations that the whole content or the current position supports, a media properties attribute for this is defined which contains a list with the values and/or ranges that are supported. The attribute is named "Scales". It may be updated at any point in the content due to content consisting of spliced pieces or content being dynamically updated by out-of-band mechanisms. 4.7.5. Mapping to the Attributes This section shows examples of how one would map the above usages to the properties and their values. On-demand: Random Access: Random-Access=5.0, Content Modifications: Immutable, Retention: Unlimited or Time-Limited. Dynamic On-demand: Random Access: Random-Access=3.0, Content Modifications: Dynamic, Retention: Unlimited or Time-Limited. Live: Random Access: No-seeking, Content Modifications: Time- Progressing, Retention: Time-Duration=0.0 Live with Recording: Random Access: Random-Access=3.0, Content Modifications: Time-Progressing, Retention: Time-Duration=7200.0 Schulzrinne, et al. Expires March 15, 2014 [Page 35] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 5. RTSP Message RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding RFC 3629 [RFC3629]. Lines MUST be terminated by CRLF. Text-based protocols make it easier to add optional parameters in a self-describing manner. Since the number of parameters and the frequency of commands is low, processing efficiency is not a concern. Text-based protocols, if done carefully, also allow easy implementation of research prototypes in scripting languages such as TCL, Visual Basic and Perl. The ISO 10646 character set avoids character set switching, but is invisible to the application as long as US-ASCII is being used. This is also the encoding used for RTCP [RFC3550]. Requests contain methods, the object the method is operating upon and parameters to further describe the method. Methods are idempotent unless otherwise noted. Methods are also designed to require little or no state maintenance at the media server. 5.1. Message Types RTSP messages consist of requests from client to server, or server to client, and responses in the reverse direction. Request (Section 7) and Response (Section 8) messages use a format based on the generic message format of RFC 2822 [RFC2822] for transferring bodies (the payload of the message). Both types of messages consist of a start- line, zero or more header fields (also known as "headers"), an empty line (i.e., a line with nothing preceding the CRLF) indicating the end of the headers, and possibly the data of the message body. The below ABNF [RFC5234] is for information and the formal message specification is present in Section 20.2.2. generic-message = start-line *(message-header CRLF) CRLF [ message-body-data ] start-line = Request-Line | Status-Line In the interest of robustness, agents MUST ignore any empty line(s) received where a Request-Line or Response-Line is expected. In other words, if the agent is reading the protocol stream at the beginning of a message and receives a CRLF first, it MUST ignore the CRLF. Schulzrinne, et al. Expires March 15, 2014 [Page 36] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 5.2. Message Headers RTSP header fields (see Section 18) include general-header, request- header, response-header, and message-body header fields. The order in which header fields with differing field names are received is not significant. However, it is "good practice" to send general-header fields first, followed by request-header or response- header fields, and ending with the Message-body header fields. Multiple message-header fields with the same field-name MAY be present in a message if and only if the entire field-value for that header field is defined as a comma-separated list. It MUST be possible to combine the multiple header fields into one "field-name: field-value" pair, without changing the semantics of the message, by appending each subsequent field-value to the first, each separated by a comma. The order in which header fields with the same field-name are received is therefore significant to the interpretation of the combined field value, and thus a proxy MUST NOT change the order of these field values when a message is forwarded. Unknown message headers MUST be ignored (skipping over the header to the next protocol element, and not causing an error) by a RTSP server or client. An RTSP Proxy MUST forward unknown message headers. Message headers defined outside of this specification that are required to be interpreted by the RTSP agent will need to use feature tags (Section 4.5) and include them in the appropriate Require (Section 18.43) or Proxy-Require (Section 18.37) header. 5.3. Message Body The message body (if any) of an RTSP message is used to carry further information for a particular resource associated with the request or response. An example of a message body is a Session Description Protocol (SDP) message. The presence of a message body in either a request or a response MUST be signaled by the inclusion of a Content-Length header (see Section 18.17) and Content-Type (see Section 18.19). A message body MUST NOT be included in a request or response if the specification of the particular method (see Method Definitions (Section 13)) does not allow sending a message body. In case a message body is received in a message when not expected the message body data SHOULD be discarded. This is to allow future extensions to define optional use of message body. Schulzrinne, et al. Expires March 15, 2014 [Page 37] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 5.4. Message Length An RTSP Message that does not contain any message body is terminated by the first empty line after the header fields (Note: An empty line is a line with nothing preceding the CRLF.). In RTSP messages that contain message bodies the empty line is followed by the message body. The length of that body is determined by the value of the Content-Length header (Section 18.17). The value in the header represents the length of the message-body in octets. If this header field is not present, a value of zero is assumed, i.e., no message body present in the message. Unlike an HTTP message, an RTSP message MUST contain a Content-Length header whenever it contains a message body. Note that RTSP does not support the HTTP/1.1 "chunked" transfer coding (see [H3.6.1]). Given the moderate length of presentation descriptions returned, the server should always be able to determine its length, even if it is generated dynamically, making the chunked transfer encoding unnecessary. Schulzrinne, et al. Expires March 15, 2014 [Page 38] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 6. General Header Fields General headers are headers that may be used in both requests and responses. The general headers are listed in Table 1: +--------------------+--------------------+ | Header Name | Defined in Section | +--------------------+--------------------+ | Accept-Ranges | Section 18.5 | | | | | Cache-Control | Section 18.11 | | | | | Connection | Section 18.12 | | | | | CSeq | Section 18.20 | | | | | Date | Section 18.21 | | | | | Media-Properties | Section 18.29 | | | | | Media-Range | Section 18.30 | | | | | Pipelined-Requests | Section 18.33 | | | | | Proxy-Supported | Section 18.38 | | | | | Range | Section 18.40 | | | | | RTP-Info | Section 18.45 | | | | | Scale | Section 18.46 | | | | | Seek-Style | Section 18.47 | | | | | Server | Section 18.48 | | | | | Session | Section 18.49 | | | | | Speed | Section 18.50 | | | | | Supported | Section 18.51 | | | | | Timestamp | Section 18.53 | | | | | Transport | Section 18.54 | | | | | User-Agent | Section 18.56 | | | | Schulzrinne, et al. Expires March 15, 2014 [Page 39] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | Via | Section 18.57 | +--------------------+--------------------+ Table 1: The general headers used in RTSP Schulzrinne, et al. Expires March 15, 2014 [Page 40] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 7. Request A request message uses the format outlined below regardless of the direction of a request, client to server or server to client: o Request line, containing the method to be applied to the resource, the identifier of the resource, and the protocol version in use; o Zero or more Header lines, that can be of the following types: general headers (Section 6), request headers (Section 7.2), or message body headers (Section 9.1); o One empty line (CRLF) to indicate the end of the header section; o Optionally a message-body, consisting of one or more lines. The length of the message body in octets is indicated by the Content- Length message header. 7.1. Request Line The request line provides the key information about the request: what method, on what resources and using which RTSP version. The methods that are defined by this specification are listed in Table 2. Schulzrinne, et al. Expires March 15, 2014 [Page 41] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 +---------------+--------------------+ | Method | Defined in Section | +---------------+--------------------+ | DESCRIBE | Section 13.2 | | | | | GET_PARAMETER | Section 13.8 | | | | | OPTIONS | Section 13.1 | | | | | PAUSE | Section 13.6 | | | | | PLAY | Section 13.4 | | | | | PLAY_NOTIFY | Section 13.5 | | | | | REDIRECT | Section 13.10 | | | | | SETUP | Section 13.3 | | | | | SET_PARAMETER | Section 13.9 | | | | | TEARDOWN | Section 13.7 | +---------------+--------------------+ Table 2: The RTSP Methods The syntax of the RTSP request line is the following: SP SP CRLF Note: This syntax cannot be freely changed in future versions of RTSP. This line needs to remain parsable by older RTSP implementations since it indicates the RTSP version of the message. In contrast to HTTP/1.1 [RFC2616], RTSP requests identify the resource through an absolute RTSP URI (including scheme, host, and port) (see Section 4.2) rather than just the absolute path. HTTP/1.1 requires servers to understand the absolute URI, but clients are supposed to use the Host request header. This is purely needed for backward-compatibility with HTTP/1.0 servers, a consideration that does not apply to RTSP. An asterisk "*" can be used instead of an absolute URI in the Request-URI part to indicate that the request does not apply to a particular resource, but to the server or proxy itself, and is only allowed when the request method does not necessarily apply to a resource. Schulzrinne, et al. Expires March 15, 2014 [Page 42] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 For example: OPTIONS * RTSP/2.0 An OPTIONS in this form will determine the capabilities of the server or the proxy that first receives the request. If the capability of the specific server needs to be determined, without regard to the capability of an intervening proxy, the server should be addressed explicitly with an absolute URI that contains the server's address. For example: OPTIONS rtsp://example.com RTSP/2.0 7.2. Request Header Fields The RTSP headers in Table 3 can be included in a request, as request headers, to modify the specifics of the request. Schulzrinne, et al. Expires March 15, 2014 [Page 43] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 +---------------------+--------------------+ | Header | Defined in Section | +---------------------+--------------------+ | Accept | Section 18.1 | | | | | Accept-Credentials | Section 18.2 | | | | | Accept-Encoding | Section 18.3 | | | | | Accept-Language | Section 18.4 | | | | | Authorization | Section 18.8 | | | | | Bandwidth | Section 18.9 | | | | | Blocksize | Section 18.10 | | | | | From | Section 18.23 | | | | | If-Match | Section 18.24 | | | | | If-Modified-Since | Section 18.25 | | | | | If-None-Match | Section 18.26 | | | | | Notify-Reason | Section 18.32 | | | | | Proxy-Authorization | Section 18.36 | | | | | Proxy-Require | Section 18.37 | | | | | Referrer | Section 18.41 | | | | | Request-Status | Section 18.42 | | | | | Require | Section 18.43 | | | | | Terminate-Reason | Section 18.52 | +---------------------+--------------------+ Table 3: The RTSP request headers Detailed header definitions are provided in Section 18. New request headers may be defined. If the receiver of the request is required to understand the request header, the request MUST include a corresponding feature tag in a Require or Proxy-Require header to ensure the processing of the header. Schulzrinne, et al. Expires March 15, 2014 [Page 44] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 8. Response After receiving and interpreting a request message, the recipient responds with an RTSP response message. Normally, there is only one, final, response. Only responses using the response code class 1xx, are allowed to send one or more 1xx response messages prior to the final response message. The valid response codes and the methods they can be used with are listed in Table 4. 8.1. Status-Line The first line of a Response message is the Status-Line, consisting of the protocol version followed by a numeric status code and the textual phrase associated with the status code, with each element separated by SP characters. No CR or LF is allowed except in the final CRLF sequence. SP SP CRLF 8.1.1. Status Code and Reason Phrase The Status-Code element is a 3-digit integer result code of the attempt to understand and satisfy the request. These codes are fully defined in Section 17. The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata and the Reason-Phrase is intended for the human user. The client is not required to examine or display the Reason- Phrase. The first digit of the Status-Code defines the class of response. The last two digits do not have any categorization role. There are 5 values for the first digit: 1xx: Informational - Request received, continuing process 2xx: Success - The action was successfully received, understood, and accepted 3rr: Redirection - Further action needs to be taken in order to complete the request (3rr rather than 3xx is used as 304 is excluded, see Section 17.3) 4xx: Client Error - The request contains bad syntax or cannot be fulfilled Schulzrinne, et al. Expires March 15, 2014 [Page 45] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 5xx: Server Error - The server failed to fulfill an apparently valid request The individual values of the numeric status codes defined for RTSP/2.0, and an example set of corresponding Reason-Phrases, are presented in Table 4. The reason phrases listed here are only recommended; they may be replaced by local equivalents without affecting the protocol. Note that RTSP adopts most HTTP/1.1 [RFC2616] status codes and adds RTSP-specific status codes starting at x50 to avoid conflicts with future HTTP status codes that are desirable to import into RTSP. All these codes are RTSP specific and RTSP has its own registry separate from HTTP for status codes. RTSP status codes are extensible. RTSP applications are not required to understand the meaning of all registered status codes, though such understanding is obviously desirable. However, applications MUST understand the class of any status code, as indicated by the first digit, and treat any unrecognized response as being equivalent to the x00 status code of that class, with an exception for unknown 3xx codes, which MUST be treated as a 302 (Found). The reason being that no 300 (Multiple Choices in HTTP) is defined for RTSP. An response with unrecognized status code MUST NOT be cached. For example, if an unrecognized status code of 431 is received by the client, it can safely assume that there was something wrong with its request and treat the response as if it had received a 400 status code. In such cases, user agents SHOULD present to the user the message body returned with the response, since that message body is likely to include human-readable information which will explain the unusual status. +------+---------------------------------+--------------------------+ | Code | Reason | Method | +------+---------------------------------+--------------------------+ | 100 | Continue | all | | | | | | | | | | 200 | OK | all | | | | | | | | | | 301 | Moved Permanently | all | | | | | | 302 | Found | all | | | | | | 303 | reserved | n/a | | | | | | 304 | Not Modified | all | | | | | | 305 | Use Proxy | all | Schulzrinne, et al. Expires March 15, 2014 [Page 46] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | 400 | Bad Request | all | | | | | | 401 | Unauthorized | all | | | | | | 402 | Payment Required | all | | | | | | 403 | Forbidden | all | | | | | | 404 | Not Found | all | | | | | | 405 | Method Not Allowed | all | | | | | | 406 | Not Acceptable | all | | | | | | 407 | Proxy Authentication Required | all | | | | | | 408 | Request Timeout | all | | | | | | 410 | Gone | all | | | | | | 412 | Precondition Failed | DESCRIBE, SETUP | | | | | | 413 | Request Message Body Too Large | all | | | | | | 414 | Request-URI Too Long | all | | | | | | 415 | Unsupported Media Type | all | | | | | | 451 | Parameter Not Understood | SET_PARAMETER, | | | | GET_PARAMETER | | | | | | 452 | reserved | n/a | | | | | | 453 | Not Enough Bandwidth | SETUP | | | | | | 454 | Session Not Found | all | | | | | | 455 | Method Not Valid In This State | all | | | | | | 456 | Header Field Not Valid | all | | | | | | 457 | Invalid Range | PLAY, PAUSE | | | | | | 458 | Parameter Is Read-Only | SET_PARAMETER | | | | | | 459 | Aggregate Operation Not Allowed | all | | | | | Schulzrinne, et al. Expires March 15, 2014 [Page 47] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | 460 | Only Aggregate Operation | all | | | Allowed | | | | | | | 461 | Unsupported Transport | all | | | | | | 462 | Destination Unreachable | all | | | | | | 463 | Destination Prohibited | SETUP | | | | | | 464 | Data Transport Not Ready Yet | PLAY | | | | | | 465 | Notification Reason Unknown | PLAY_NOTIFY | | | | | | 466 | Key Management Error | all | | | | | | 470 | Connection Authorization | all | | | Required | | | | | | | 471 | Connection Credentials not | all | | | accepted | | | | | | | 472 | Failure to establish secure | all | | | connection | | | | | | | | | | | 500 | Internal Server Error | all | | | | | | 501 | Not Implemented | all | | | | | | 502 | Bad Gateway | all | | | | | | 503 | Service Unavailable | all | | | | | | 504 | Gateway Timeout | all | | | | | | 505 | RTSP Version Not Supported | all | | | | | | 551 | Option Not Supported | all | | | | | | 553 | Proxy Unavailable | all | +------+---------------------------------+--------------------------+ Table 4: Status codes and their usage with RTSP methods Schulzrinne, et al. Expires March 15, 2014 [Page 48] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 8.2. Response Headers The response-headers allow the request recipient to pass additional information about the response which cannot be placed in the Status- Line. This header gives information about the server and about further access to the resource identified by the Request-URI. All headers currently classified as response headers are listed in Table 5. +------------------------+--------------------+ | Header | Defined in Section | +------------------------+--------------------+ | Authentication-Info | Section 18.7 | | | | | Connection-Credentials | Section 18.13 | | | | | Location | Section 18.28 | | | | | MTag | Section 18.31 | | | | | Proxy-Authenticate | Section 18.34 | | | | | Public | Section 18.39 | | | | | Retry-After | Section 18.44 | | | | | Unsupported | Section 18.55 | | | | | WWW-Authenticate | Section 18.58 | +------------------------+--------------------+ Table 5: The RTSP response headers Response-header names can be extended reliably only in combination with a change in the protocol version. However, the usage of feature-tags in the request allows the responding party to learn the capability of the receiver of the response. A new or experimental header MAY be given the semantics of response-header if all parties in the communication recognize them to be a response-header. Unrecognized headers in responses are treated as message-headers and hence MUST be ignored. Schulzrinne, et al. Expires March 15, 2014 [Page 49] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 9. Message Body Request and Response messages MAY transfer a message body, if not otherwise restricted by the request method or response status code. The message body consists of the content data itself (see also Section 5.3). The SET_PARAMETER and GET_PARAMETER requests and responses, and the DESCRIBE response as defined by this specification MAY have a message body; the purpose of the message body is defined in each case. All 4xx and 5xx responses MAY also have a message body to carry additional response information. Generally a message body MAY be attached to any RTSP 2.0 request or response, but the content of the message body MAY be ignored by the receiver. Extensions to this specification can specify the purpose and content of message bodies, including requiring their inclusion. In this section, both sender and recipient refer to either the client or the server, depending on who sends and who receives the message body. 9.1. Message-Body Header Fields Message-body header fields define meta-information about the content data in the message body. The message-body header fields are listed in Table 6. Schulzrinne, et al. Expires March 15, 2014 [Page 50] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 +------------------+--------------------+ | Header | Defined in Section | +------------------+--------------------+ | Allow | Section 18.6 | | | | | Content-Base | Section 18.14 | | | | | Content-Encoding | Section 18.15 | | | | | Content-Language | Section 18.16 | | | | | Content-Length | Section 18.17 | | | | | Content-Location | Section 18.18 | | | | | Content-Type | Section 18.19 | | | | | Expires | Section 18.22 | | | | | Last-Modified | Section 18.27 | +------------------+--------------------+ Table 6: The RTSP message-body headers The extension-header mechanism allows additional message-body header fields to be defined without changing the protocol, but these fields cannot be assumed to be recognizable by the recipient. Unrecognized header fields MUST be ignored by the recipient and forwarded by proxies. 9.2. Message Body An RTSP message with a message body MUST include the Content-Type and Content-Length headers. When a message body is included with a message, the data type of that content data is determined via the header fields Content-Type and Content-Encoding. Content-Type specifies the media type of the underlying data. Content-Encoding may be used to indicate any additional content codings applied to the data, usually for the purpose of data compression, that are a property of the requested resource. There is no default encoding. The Content-Length of a message is the length of the content, measured in octets. Schulzrinne, et al. Expires March 15, 2014 [Page 51] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 9.3. Message Body Format Negotiation The content format of the message body is provided using the Content- Type header (Section 18.19). To enable the responder of a request to determine which media type it should use, the requestor may include the Accept header (Section 18.1) in a request to identify supported media types or media type ranges suitable to the response. In cases the responder is not supporting any of the specified formats, then the request response will be a 406 (Not Acceptable) error code. The media types that may be used on requests with message bodies needs to be determined through the use of feature-tags, specification requirement or trial and error. Trial and error works in the regards that in case the responder is not supporting the media type of the message body it will respond with a 415 (Unsupported Media Type). The formats supported and their negotiation is done individually on a per method and direction (request or response body) direction. Requirements on supporting particular media types for use as message bodies in requests and response SHALL also be specified on per method and direction basis. Schulzrinne, et al. Expires March 15, 2014 [Page 52] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 10. Connections RTSP Messages are transferred between RTSP agents and proxies using a transport connection. This transport connection uses TCP or TCP/TLS. This transport connection is referred to as the connection or possibly RTSP connection within this document. RTSP requests can be transmitted using the two different connection scenarios listed below: o persistent - a transport connection is used for several request/ response transactions; o transient - a transport connection is used for a single request/ response transaction. RFC 2326 attempted to specify an optional mechanism for transmitting RTSP messages in connectionless mode over a transport protocol such as UDP. However, it was not specified in sufficient detail to allow for interoperable implementations. In an attempt to reduce complexity and scope, and due to lack of interest, RTSP 2.0 does not attempt to define a mechanism for supporting RTSP over UDP or other connectionless transport protocols. A side-effect of this is that RTSP requests MUST NOT be sent to multicast groups since no connection can be established with a specific receiver in multicast environments. Certain RTSP headers, such as the CSeq header (Section 18.20), which may appear to be relevant only to connectionless transport scenarios are still retained and MUST be implemented according to the specification. In the case of CSeq, it is quite useful for matching responses to requests if the requests are pipelined (see Section 12). It is also useful in proxies for keeping track of the different requests when aggregating several client requests on a single TCP connection. 10.1. Reliability and Acknowledgements Since RTSP messages are transmitted using reliable transport protocols, they MUST NOT be retransmitted at the RTSP protocol level. Instead, the implementation must rely on the underlying transport to provide reliability. The RTSP implementation may use any indication of reception acknowledgment of the message from the underlying transport protocols to optimize the RTSP behavior. Schulzrinne, et al. Expires March 15, 2014 [Page 53] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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 congestion. Lack of acknowledgment of an RTSP request should be handled within the constraints of the connection timeout considerations described below (Section 10.4). 10.2. Using Connections A TCP transport can be used for both persistent connections (for several message exchanges) and transient connections (for a single message exchange). Implementations of this specification MUST support RTSP over TCP. The scheme of the RTSP URI (Section 4.2) indicates the default port that the server will listen on if the port is not explicitly given. In addition to the registered default ports, i.e., 554 (rtsp) and 322 (rtsps), there is an alternative port 8554 registered. This port may provide some benefits from non-registered ports if a RTSP server is unable to use the default ports. The benefits may include pre- configured security policies as well as classifiers in network monitoring tools. A RTSP client opening a TCP connection for accessing a particular resource as identified by a URI uses the IP address and port derived from the host and port parts of the URI. The IP address is either the explicit address provided in the URI or any of the addresses provided when performing A and AAAA record DNS lookups of the host name in the URI. A server MUST handle both persistent and transient connections. Transient connections facilitate mechanisms for fault tolerance. They also allow for application layer mobility. A server and client pair that support transient connections can survive the loss of a TCP connection; e.g., due to a NAT timeout. When the client has discovered that the TCP connection has been lost, it can set up a new one when there is need to communicate again. A persistent connection is RECOMMENDED to be used for all transactions between the server and client, including messages for Schulzrinne, et al. Expires March 15, 2014 [Page 54] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 multiple RTSP sessions. However, a persistent connection MAY be closed after a few message exchanges. For example, a client may use a persistent connection for the initial SETUP and PLAY message exchanges in a session and then close the connection. Later, when the client wishes to send a new request, such as a PAUSE for the session, a new connection would be opened. This connection may either be transient or persistent. An RTSP agent MAY use one connection to handle multiple RTSP sessions on the same server. The RTSP agent SHALL NOT use more than one connection per RTSP session at any given point. Using a single connection for multiple RTSP session saves connection resources on the server. Not using more than one connection at a time for a particular RTSP session avoids wasting connection resources and allows the server to track only for the latest in client to server used connection for each RTSP session as being the currently valid server to client connection. RTSP allows a server to send requests to a client. However, this can be supported only if a client establishes a persistent connection with the server. In cases where a persistent connection does not exist between a server and its client, due to the lack of a signaling channel the server may be forced to silently discard RTSP messages, and may even drop an RTSP session without notifying the client. An example of such a case is when the server desires to send a REDIRECT request for an RTSP session to the client but is not able to do so because it cannot reach the client. A server that attempts to send a request to a client that has no connection currently to the server SHALL discard the request directly. Without a persistent connection between the client and the server, the media server has no reliable way of reaching the client. Because the likely failure of server to client established connections the server will not even attempt establishing any connection. Queuing of server to client requests has been considered. However a security issue exist in how one authorizes a client establishing a new connection as being a legit receiver of request related to a particular RTSP session without the client first issuing requests related to the request. Thus, likely making any such requests even more delayed and less useful. The sending of client and server requests can be asynchronous events. To avoid deadlock situations both client and server MUST be able to send and receive requests simultaneously. As an RTSP response may be queued up for transmission, reception or processing behind the peer Schulzrinne, et al. Expires March 15, 2014 [Page 55] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTSP agent's own requests, all RTSP agents are required to have a certain capability of handling outstanding messages. A potential issue is that outstanding requests may timeout despite them being processed by the peer due to the response being caught in the queue behind a number of requests that the RTSP agent is processing but that take some time to complete. To avoid this problem an RTSP agent is recommended to buffer incoming messages locally so that any response messages can be processed immediately upon reception. If responses are separated from requests and directly forwarded for processing, not only can the result be used immediately, the state associated with that outstanding request can also be released. However, buffering a number of requests on the receiving RTSP agent consumes resources and enables a resource exhaustion attack on the agent. Therefore this buffer should be limited so that an unreasonable number of requests or total message size is not allowed to consume the receiving agent's resources. In most APIs, having the receiving agent stop reading from the TCP socket will result in TCP's window being clamped. Thus forcing the buffering onto the sending agent when the load is larger than expected. However, as both RTSP message sizes and frequency may be changed in the future by protocol extensions, an agent should be careful against taking harsher measurements against a potential attack. When under attack an RTSP agent can close TCP connections and release state associated with that TCP connection. To provide some guidance on what is reasonable the following guidelines are given. It is RECOMMENDED that: o an RTSP agent should not have more than 10 outstanding requests per RTSP session; o an RTSP agent should not have more than 10 outstanding requests that are not related to an RTSP session or that are requesting to create an RTSP session. In light of the above, it is RECOMMENDED that clients use persistent connections whenever possible. A client that supports persistent connections MAY "pipeline" its requests (see Section 12). RTSP Agents can send requests to multiple different destinations, either servers or client contexts over the same connection to a proxy. Then the proxy forks the message to the different destinations over proxy to agent connections. In these cases when multiple requests are outstanding the requesting agent MUST be ready to receive the responses out of order compared to the order they where sent on the connection. The order between multiple messages for each destination will be maintained, however, the order between response from different destinations can be different. Schulzrinne, et al. Expires March 15, 2014 [Page 56] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 The reason for this is to avoid a head of line blocking. In a sequence of request an early outstanding request may take time to be processed at one destination. Simultaneously a responses from any other destinations, that was later in the sequence of requests may have arrived at the proxy. Thus allowing out-of-order responses avoid forcing the proxy to buffer this response and instead deliver it as soon as possible. Note, this will not affect the order they where processed at request destination. This scenario can occur in two cases involving proxies. The first is a client issuing requests for sessions on different servers using a common client to proxy connection. The second is for server to client requests, like REDIRECT being sent by the server over a common transport connection the proxy created for its different connecting clients. 10.3. Closing Connections The client MAY close a connection at any point when no outstanding request/response transactions exist for any RTSP session being managed through the connection. The server, however, SHOULD NOT close a connection until all RTSP sessions being managed through the connection have been timed out (Section 18.49). A server SHOULD NOT close a connection immediately after responding to a session-level TEARDOWN request for the last RTSP session being controlled through the connection. Instead, the server should wait for a reasonable amount of time for the client to receive and act upon the TEARDOWN response, and initiate the connection closing. The server SHOULD wait at least 10 seconds after sending the TEARDOWN response before closing the connection. This is to ensure that the client has time to issue a SETUP for a new session on the existing connection after having torn the last one down. 10 seconds should give the client ample opportunity to get its message to the server. A server SHOULD NOT close the connection directly as a result of responding to a request with an error code. Certain error responses such as "460 Only Aggregate Operation Allowed" (Section 17.4.24) 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 other hand, keeping connections open after sending an error response poses a Denial of Service security risk (Section 21). Schulzrinne, et al. Expires March 15, 2014 [Page 57] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 The server MAY close a connection if it receives an incomplete message and if the message is not completed within a reasonable amount of time. It is RECOMMENDED that the server waits at least 10 seconds for the completion of a message or for the next part of the message to arrive (which is an indication that the transport and the client are still alive). Servers believing they are under attack or otherwise starved for resources during that event MAY consider using a shorter timeout. If a server closes a connection while the client is attempting to send a new request, the client will have to close its current connection, establish a new connection and send its request over the new connection. An RTSP message SHOULD NOT be terminated by closing the connection. Such a message MAY be considered to be incomplete by the receiver and discarded. An RTSP message is properly terminated as defined in Section 5. 10.4. Timing Out Connections and RTSP Messages Receivers of a request (responder) SHOULD respond to requests in a timely manner even when a reliable transport such as TCP is used. Similarly, the sender of a request (requester) SHOULD wait for a sufficient time for a response before concluding that the responder will not be acting upon its request. A responder SHOULD respond to all requests within 5 seconds. If the responder recognizes that processing of a request will take longer than 5 seconds, it SHOULD send a 100 (Continue) response as soon as possible. It SHOULD continue sending a 100 response every 5 seconds thereafter until it is ready to send the final response to the requester. After sending a 100 response, the receiver MUST send a final response indicating the success or failure of the request. A requester SHOULD wait at least 10 seconds for a response before concluding that the responder will not be responding to its request. After receiving a 100 response, the requester SHOULD continue waiting for further responses. If more than 10 seconds elapses without receiving any response, the requester MAY assume that the responder is unresponsive and abort the connection by closing the TCP connection. Note: In cases multiple RTSP sessions share the same transport connection, abandoning a request closing the connection may have significant impact on those other sessions. First of all, other RTSP requests may have become queued up due to the request taking long time. Secondly also those sessions loose the possibility to Schulzrinne, et al. Expires March 15, 2014 [Page 58] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 receive server to client requests. To mitigate that situation the RTSP agent should establish a new connection and send any queued up and non-responded requests on this new connection. Secondly, to ensure that the RTSP server knows which connection that is valid for a particular RTSP session, the RTSP agent should send a keep-alive request, if no other request will be sent immediately for that RTSP session, for each RTSP session on the old connection. The keep-alive request will normally be a GET_PARAMETER with a session header to inform the server that this agent cares about this RTSP session. A requester SHOULD wait longer than 10 seconds for a response if it is experiencing significant transport delays on its connection to the responder. The requester is capable of determining the round trip time (RTT) of the request/response cycle using the Timestamp header (Section 18.53) in any RTSP request. 10 seconds was chosen for the following reasons. It gives TCP time to perform a couple of retransmissions, even if operating on default values. It is short enough that users may not abandon the process themselves. However, it should be noted that 10 seconds can be aggressive on certain type of networks. The 5 seconds value for 1xx messages is half the timeout giving a reasonable chance of successful delivery before timeout happens on the requester side. 10.5. Showing Liveness 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 requests being issued to accomplish other things, the following mechanisms can be used, in descending order of preference: RTCP: If RTP is used for media transport RTCP SHOULD be used. If RTCP is used to report transport statistics, it will necessarily also function as a keep-alive. The server can determine the client by network address and port together with the fact that the client is reporting on the server's RTP sender sources (SSRCs). A downside of using RTCP is that it only gives statistical guarantees of reaching the server. However, the probability of a false client timeout is so low that it can be ignored in most cases. For example, assume a Schulzrinne, et al. Expires March 15, 2014 [Page 59] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 session with 60 seconds timeout and enough bitrate assigned to RTCP messages to send a message from client to server on average every 5 seconds. That client has, for a network with 5% packet loss, a probability of failing to confirm liveness within the timeout interval for that session of 2.4*E-16. Sessions with shorter timeouts, or much higher packet loss, or small RTCP bandwidths SHOULD also implement one or more of the mechanisms below. SET_PARAMETER: When using SET_PARAMETER for keep-alive, a body SHOULD NOT be included. This method is the RECOMMENDED RTSP method to use for a request intended only to perform keep- alive. Support of SET_PARAMETER is mandatory for RTSP Servers to ensure clients can use this method. GET_PARAMETER: When using GET_PARAMETER for keep-alive, no body SHOULD be included. Dependent on implementation support in server. OPTIONS: This method is also usable, but it causes the server to perform more unnecessary processing and results in bigger responses than necessary for the task. The reason is that the server needs to determine the capabilities associated with the media resource to correctly populate the Public and Allow headers. The timeout parameter of the Session header (Section 18.49) MAY be included in a SETUP response, and MUST NOT be included in requests. The server uses it to indicate to the client how long the server is prepared to wait between RTSP commands or other signs of life before closing the session due to lack of activity (see Appendix B). The timeout is measured in seconds, with a default of 60 seconds. The length of the session timeout MUST NOT be changed in an established session. 10.6. Use of IPv6 Explicit IPv6 [RFC2460] support was not present in RTSP 1.0 (RFC 2326). RTSP 2.0 has been updated for explicit IPv6 support. Implementations of RTSP 2.0 MUST understand literal IPv6 addresses in URIs and RTSP headers. Although the general URI format envisages potential future new versions of the literal IP address, usage of any such new version would require other modifications to the RTSP specification (e.g. address fields in the Transport header (Section 18.54)). Schulzrinne, et al. Expires March 15, 2014 [Page 60] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 10.7. Overload Control Overload in RTSP can occur when servers and proxies have insufficient resources to complete the processing of a request. An improper handling of such an overload situation at proxies and servers can impact the operation of the RTSP deployment, and probably worsen the situation. RTSP defines the 503 (Service Unavailable) response (Section 17.5.4) to let servers and proxies notify requesting proxies and RTSP clients about an overload situation. In conjunction with the Retry-After header (Section 18.44) the server or proxy can indicate the time after the requesting entity can send another request to the proxy or server. There are two scopes of such 503 answers, one for established RTSP sessions, where the request resulting in the 503 response as well as the response carries a Session header identifying the session which is suffering overload. This response only applies to this particular session. The other scope is the general RTSP server as identified by the host in the request URL. Such a 503 answer with any Retyr-After header applies to all non-session specific requests to that server, including SETUP request intended to create a new RTSP session. Another scope for overload situation exists, which is the RTSP proxy. To enable an RTSP proxy to signal that it is overloaded, or otherwise unavailable and can't handle the request, a 553 response code has been defined with the meaning "Proxy Unavailable". Also for proxies there is a separation in response scopes between requests associated with existing RTSP sessions, and requests to create new sessions or general proxy requests. Simply implementing and using the 503 (Service Unavailable) and 553 (Proxy Unavailable) is not sufficient for properly handling overload situations. For instance, a simplistic approach would be to send the 503 response with a Retry-After header set to a fixed value. However, this can cause the situation where multiple RTSP clients again send requests to a proxy or server at roughly the same time which may again cause an overload situation, or if the "old" overload situation is not yet solved, i.e., the length indicated in the Retry- After header was too short. An RTSP server or proxy in an overload situation must select the value of the Retry-After header carefully and bearing in mind its current load situation. It is REQUIRED to increase the timeout period in proportion to the current load on the server, i.e., an increasing workload should result in an increased length of the indicated unavailability. It is REQUIRED to not send the same value in the Retry-After header to all requesting proxies and clients, but to add a variation to the mean value of the Retry-After header. Schulzrinne, et al. Expires March 15, 2014 [Page 61] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 A more complex case may arise when a load balancing RTSP proxy is in use, i.e., where an RTSP proxy is used to select amongst a set of RTSP servers to handle the requests, or when multiple server addresses are available for a given server name. The proxy or client may receive a 503 (Service Unavailable) or 553 (Proxy Unavailable) from one of its RTSP servers or proxies, or a TCP timeout (if the server is even unable to handle the request message). The proxy or client simply retries the other addresses or configured proxies, but may also receive a 503 (Service Unavailable) or 553 (Proxy Unavailable) response or TCP timeouts from those addresses. In such a situation, where none of the RTSP servers/proxies/addresses can handle the request, the RTSP agent has to wait before it can send any new requests to the RTSP server. Any additional request to a specific address MUST be delayed according to the Retry-After headers received. For addresses where no response was received or TCP timeout occurred, an initial wait timer SHOULD be set to 5 seconds. That timer MUST be doubled for each additional failure to connect or receive response until the value exceeds 30 minutes when the timers mean value may be set to 30 minutes. It is REQUIRED to not set the same value in the timer for each scheduling, but instead to add a variation to the mean value, resulting in picking a random value within the range of 0.5 to 1.5 of the mean value. Schulzrinne, et al. Expires March 15, 2014 [Page 62] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 11. Capability Handling This section describes the available capability handling mechanism which allows RTSP to be extended. Extensions to this version of the protocol are basically done in two ways. First, new headers can be added. Secondly, new methods can be added. The capability handling mechanism is designed to handle both cases. When a method is added, the involved parties can use the OPTIONS method to discover whether it is supported. This is done by issuing an OPTIONS request to the other party. Depending on the URI it will either apply in regards to a certain media resource, the whole server in general, or simply the next hop. The OPTIONS response MUST contain a Public header which declares all methods supported for the indicated resource. It is not necessary to use OPTIONS to discover support of a method, as the client could simply try the method. If the receiver of the request does not support the method it will respond with an error code indicating the method is either not implemented (501) or does not apply for the resource (405). The choice between the two discovery methods depends on the requirements of the service. Feature-tags are defined to handle functionality additions that are not new methods. Each feature-tag represents a certain block of functionality. The amount of functionality that a feature-tag represents can vary significantly. A feature-tag can for example represent the functionality a single RTSP header provides. Another feature-tag can represent much more functionality, such as the "play.basic" feature-tag (Section 11.1) which represents the minimal media delivery for playback implementation. Feature-tags are used to determine whether the client, server or proxy supports the functionality that is necessary to achieve the desired service. To determine support of a feature-tag, several different headers can be used, each explained below: Supported: This header is used to determine the complete set of functionality that both client and server have in general and is not dependent on specific resource. The intended usage is to determine before one needs to use a functionality that it is supported. It can be used in any method, but OPTIONS is the most suitable one as it at the same time determines all methods that are implemented. When sending a request the requester declares all its capabilities by including all supported feature-tags. This results in the receiver is learning the requester's feature support. The receiver then includes its set of features in the response. Schulzrinne, et al. Expires March 15, 2014 [Page 63] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Proxy-Supported: This header is used similarly to the Supported header, but instead of giving the supported functionality of the client or server it provides both the requester and the responder a view of the common functionality supported in general by all members of the proxy chain between the two supports and not dependent on the resource. Proxies are required to add this header whenever the Supported header is present, but proxies may also add it independently of the requester. Require: This header can be included in any request where the end- point, i.e., the client or server, is required to understand the feature to correctly perform the request. This can, for example, be a SETUP request where the server is required to understand a certain parameter to be able to set up the media delivery correctly. Ignoring this parameter would not have the desired effect and is not acceptable. Therefore the end-point receiving a request containing a Require MUST negatively acknowledge any feature that it does not understand and not perform the request. The response in cases where features are not supported are 551 (Option Not Supported). Also the features that are not supported are given in the Unsupported header in the response. Proxy-Require: This header has the same purpose and workings as Require except that it only applies to proxies and not the end- point. Features that need to be supported by both proxies and end-points need to be included in both the Require and Proxy- Require header. Unsupported: This header is used in a 551 error response, to indicate which features were not supported. Such a response is only the result of the usage of the Require and/or Proxy- Require header where one or more features where not supported. This information allows the requester to make the best of situations as it knows which features are not supported. 11.1. Feature Tag: play.basic The play.basic feature tag represents an RTSP implementation according to all normative RTSP protocol features specified in this specification. This specification is both a RTSP core specification as well intended to enable setup and control of playback of media. Thus following all normative parts in the main sections (the ones with numbers), not the appendices (starting with letters), unless explicitly specified in a main section for being a required appendix. Schulzrinne, et al. Expires March 15, 2014 [Page 64] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Note: This feature tag does not mandate any media delivery protocol, such as RTP. In RTSP 1.0 there was a minimal implementation section. However, that was not consistent with the rest of the specification. So rather than making an attempt to explicitly enumerate the features for play.basic this specification have to be read in completeness and the necessary features normatively defined as being required are included. Schulzrinne, et al. Expires March 15, 2014 [Page 65] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 12. Pipelining Support Pipelining is a general method to improve performance of request response protocols by allowing the requesting agent to have more than one request outstanding and send them over the same persistent connection. For RTSP, where the relative order of requests will matter, it is important to maintain the order of the requests. Because of this, the responding agent MUST process the incoming requests in their sending order. The sending order can be determined by the CSeq header and its sequence number. For TCP the delivery order will be the same between two agents, as the sending order. The processing of the request MUST also have been finished before processing the next request from the same agent. The responses MUST be sent in the order the requests were processed. RTSP 2.0 has extended support for pipelining compared to RTSP 1.0. The major improvement is to allow all requests involved in setting up and initiating media delivery to be pipelined after each other. This is accomplished by the utilization of the Pipelined-Requests header (see Section 18.33). This header allows a client to request that two or more requests are processed in the same RTSP session context which the first request creates. In other words, a client can request that two or more media streams are set-up and then played without needing to wait for a single response. This speeds up the initial startup time for an RTSP session by at least one RTT. If a pipelined request builds on the successful completion of one or more prior requests the requester must verify that all requests were executed as expected. A common example will be two SETUP requests and a PLAY request. In case one of the SETUP fails unexpectedly, the PLAY request can still be successfully executed. However, the resulting presentation will not be as expected by the requesting client, as only a single media instead of two will be played. In this case the client can send a PAUSE request, correct the failing SETUP request and then request it to be played. Schulzrinne, et al. Expires March 15, 2014 [Page 66] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 13. Method Definitions The method indicates what is to be performed on the resource identified by the Request-URI. The method name is case-sensitive. New methods may be defined in the future. Method names MUST NOT start with a $ character (decimal 36) and MUST be a token as defined by the ABNF [RFC5234] in the syntax chapter Section 20. The methods are summarized in Table 7. +---------------+-----------+--------+-------------+-------------+ | method | direction | object | Server req. | Client req. | +---------------+-----------+--------+-------------+-------------+ | DESCRIBE | C -> S | P,S | recommended | recommended | | | | | | | | GET_PARAMETER | C -> S | P,S | optional | optional | | | | | | | | | S -> C | P,S | optional | optional | | | | | | | | OPTIONS | C -> S | P,S | required | required | | | | | | | | | S -> C | P,S | optional | optional | | | | | | | | PAUSE | C -> S | P,S | required | required | | | | | | | | PLAY | C -> S | P,S | required | required | | | | | | | | PLAY_NOTIFY | S -> C | P,S | required | required | | | | | | | | REDIRECT | S -> C | P,S | optional | required | | | | | | | | SETUP | C -> S | S | required | required | | | | | | | | SET_PARAMETER | C -> S | P,S | required | optional | | | | | | | | | S -> C | P,S | optional | optional | | | | | | | | TEARDOWN | C -> S | P,S | required | required | | | | | | | | | S -> C | P | required | required | +---------------+-----------+--------+-------------+-------------+ Table 7: Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on. Further it indicates if a server or a client implementation are required (mandatory), recommended or if it is optional to implement the method. Schulzrinne, et al. Expires March 15, 2014 [Page 67] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Note on Table 7: GET_PARAMETER is optional. For example, a fully functional server can be built to deliver media without any parameters. However, SET_PARAMETER is required, i.e. mandatory to implement for the server, this is due to its usage for keep-alive. PAUSE is required because it is the only way of leaving the Play state without terminating the whole session. If an RTSP agent does not support a particular method, it MUST return 501 (Not Implemented) and the requesting RTSP agent, in turn, SHOULD NOT try this method again for the given agent / resource combination. An RTSP proxy whose main function is to log or audit and not modify transport or media handling in any way MAY forward RTSP messages with unknown methods. Note that the proxy still needs to perform the minimal required processing, like adding the Via header. 13.1. OPTIONS The semantics of the RTSP OPTIONS method is similar to that of the HTTP OPTIONS method described in [H9.2]. In RTSP however, OPTIONS is bi-directional, in that a client can send the request to a server and vice versa. A client MUST implement the capability to send an OPTIONS request and a server or a proxy MUST implement the capability to respond to an OPTIONS request. In addition to this "MUST implement" functionality, clients, servers and proxies MAY provide support both for sending OPTIONS requests and generating responses to the requests. An OPTIONS request may be issued at any time. Such a request does not modify the session state. However, it may prolong the session lifespan (see below). The URI in an OPTIONS request determines the scope of the request and the corresponding response. If the Request- URI refers to a specific media resource on a given host, the scope is limited to the set of methods supported for that media resource by the indicated RTSP agent. A Request-URI with only the host address limits the scope to the specified RTSP agent's general capabilities without regard to any specific media. If the Request-URI is an asterisk ("*"), the scope is limited to the general capabilities of the next hop (i.e., the RTSP agent in direct communication with the request sender). Regardless of the scope of the request, the Public header MUST always be included in the OPTIONS response listing the methods that are supported by the responding RTSP agent. In addition, if the scope of the request is limited to a media resource, the Allow header MUST be included in the response to enumerate the set of methods that are allowed for that resource unless the set of methods completely matches the set in the Public header. If the given resource is not available, the RTSP agent SHOULD return an appropriate response code Schulzrinne, et al. Expires March 15, 2014 [Page 68] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 such as 3rr or 4xx. The Supported header MAY be included in the request to query the set of features that are supported by the responding RTSP agent. The OPTIONS method can be used to keep an RTSP session alive. However, this is not the preferred way of session keep-alive signaling, see Section 18.49. An OPTIONS request intended for keeping alive an RTSP session MUST include the Session header with the associated session identifier. Such a request SHOULD also use the media or the aggregated control URI as the Request-URI. Example: C->S: OPTIONS rtsp://server.example.com RTSP/2.0 CSeq: 1 User-Agent: PhonyClient/1.2 Proxy-Require: gzipped-messages Supported: play.basic S->C: RTSP/2.0 200 OK CSeq: 1 Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS Supported: play.basic, setup.rtp.rtcp.mux, play.scale Server: PhonyServer/1.1 Note that some of the feature-tags in Supported and Proxy-Require are fictitious features. 13.2. DESCRIBE The DESCRIBE method is used to retrieve the description of a presentation or media object from a server. The Request-URI of the DESCRIBE request identifies the media resource of interest. The client MAY include the Accept header in the request to list the description formats that it understands. The server MUST respond with a description of the requested resource and return the description in the message body of the response, if the DESCRIBE method request can be successfully fulfilled. The DESCRIBE reply- response pair constitutes the media initialization phase of RTSP. The DESCRIBE response SHOULD contain all media initialization information for the resource(s) that it describes. Servers SHOULD NOT use the DESCRIBE response as a means of media indirection by having the description point at another server; instead, using the 3rr responses is RECOMMENDED. Schulzrinne, et al. Expires March 15, 2014 [Page 69] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 By forcing a DESCRIBE response to contain all media initialization information for the set of streams that it describes, and discouraging the use of DESCRIBE for media indirection, any looping problems can be avoided that might have resulted from other approaches. Example: C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/2.0 CSeq: 312 User-Agent: PhonyClient/1.2 Accept: application/sdp, application/example S->C: RTSP/2.0 200 OK CSeq: 312 Date: Thu, 23 Jan 1997 15:35:06 GMT Server: PhonyServer/1.1 Content-Base: rtsp://server.example.com/fizzle/foo/ Content-Type: application/sdp Content-Length: 358 v=0 o=MNobody 2890844526 2890842807 IN IP4 192.0.2.46 s=SDP Seminar i=A Seminar on the session description protocol u=http://www.example.com/lectures/sdp.ps e=seminar@example.com (Seminar Management) c=IN IP4 0.0.0.0 a=control:* t=2873397496 2873404696 m=audio 3456 RTP/AVP 0 a=control:audio m=video 2232 RTP/AVP 31 a=control:video Media initialization is a requirement for any RTSP-based system, but the RTSP specification does not dictate that this is required to be done via the DESCRIBE method. There are three ways that an RTSP client may receive initialization information: o via an RTSP DESCRIBE request o via some other protocol (HTTP, email attachment, etc.) o via some form of user interface If a client obtains a valid description from an alternate source, the client MAY use this description for initialization purposes without Schulzrinne, et al. Expires March 15, 2014 [Page 70] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 issuing a DESCRIBE request for the same media. The client should use any MTag to either validate the presentation description or make the session establishment conditional on being valid. It is RECOMMENDED that minimal servers support the DESCRIBE method, and highly recommended that minimal clients support the ability to act as "helper applications" that accept a media initialization file from a user interface, and/or other means that are appropriate to the operating environment of the clients. 13.3. SETUP The below description uses the following states in a protocol state machine that are related to a specific session when that session has been created. The state transitions are driven by protocol interactions. For additional information about the state machine see Appendix B. Init: Initial state: no session exists. Ready: Session is ready to start playing. Play: Session is playing, i.e., sending media stream data in the direction S->C. The SETUP request for an URI specifies the transport mechanism to be used for the streamed media. The SETUP method may be used in two different cases; Create an RTSP session and change the transport parameters of already set up media stream(s). SETUP can be used in all three states; Init, and Ready, for both purposes and in PLAY to change the transport parameters. There is also a third possible usage for the SETUP method which is not specified in this memo: adding a media to a session. Using SETUP to add media to an existing session, when the session is in Play state, is unspecified. The Transport header, see Section 18.54, specifies the media transport parameters acceptable to the client for data transmission; the response will contain the transport parameters selected by the server. This allows the client to enumerate in descending order of preference the transport mechanisms and parameters acceptable to it, while the server can select the most appropriate. It is expected that the session description format used will enable the client to select a limited number of possible configurations that are offered to the server to choose from. All transport related parameters SHALL be included in the Transport header; the use of other headers for this purpose is NOT RECOMMENDED due to middleboxes, such as firewalls or NATs. Schulzrinne, et al. Expires March 15, 2014 [Page 71] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 For the benefit of any intervening firewalls, a client MUST indicate the known transport parameters, even if it has no influence over these parameters, for example, where the server advertises a fixed multicast address as destination. Since SETUP includes all transport initialization information, firewalls and other intermediate network devices (which need this information) are spared the more arduous task of parsing the DESCRIBE response, which has been reserved for media initialization. The client MUST include the Accept-Ranges header in the request indicating all supported unit formats in the Range header. This allows the server to know which formats it may use in future session related responses, such as a PLAY response without any range in the request. If the client does not support a time format necessary for the presentation, the server MUST respond using 456 (Header Field Not Valid for Resource) and include the Accept-Ranges header with the range unit formats supported for the resource. In a SETUP response the server MUST include the Accept-Ranges header (see Section 18.5) to indicate which time formats are acceptable to use for this media resource. The SETUP response 200 OK MUST include the Media-Properties header (see Section 18.29 ). The combination of the parameters of the Media-Properties header indicates the nature of the content present in the session (see also Section 4.7). For example, a live stream with time shifting is indicated by o Random Access set to Random-Access, o Content Modifications set to Time Progressing, o Retention set to Time-Duration (with specific recording window time value). The SETUP response 200 OK MUST include the Media-Range header (see Section 18.30) if the media is Time-Progressing. A basic example for SETUP: Schulzrinne, et al. Expires March 15, 2014 [Page 72] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0 CSeq: 302 Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589", RTP/AVP/TCP;unicast;interleaved=0-1 Accept-Ranges: npt, clock User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 302 Date: Thu, 23 Jan 1997 15:35:06 GMT Server: PhonyServer/1.1 Session: 47112344;timeout=60 Transport: RTP/AVP;unicast;dest_addr="192.0.2.53:4588"/ "192.0.2.53:4589"; src_addr="198.51.100.241:6256"/ "198.51.100.241:6257"; ssrc=2A3F93ED Accept-Ranges: npt Media-Properties: Random-Access=3.2, Time-Progressing, Time-Duration=3600.0 Media-Range: npt=0-2893.23 In the above example the client wants to create an RTSP session containing the media resource "rtsp://example.com/foo/bar/baz.rm". The transport parameters acceptable to the client are either RTP/AVP/ UDP (UDP per default) to be received on client port 4588 and 4589 at the address the RTSP setup connection comes from or RTP/AVP interleaved on the RTSP control channel. The server selects the RTP/ AVP/UDP transport and adds the address and ports it will send and receive RTP and RTCP from, and the RTP SSRC that will be used by the server. The server MUST generate a session identifier in response to a successful SETUP request, unless a SETUP request to a server includes a session identifier or a Pipelined-Requests header referencing an existing session context, in which case the server MUST bundle this SETUP request into the existing session (aggregated session) or return error 459 (Aggregate Operation Not Allowed) (see Section 17.4.23). An Aggregate control URI MUST be used to control an aggregated session. This URI MUST be different from the stream control URIs of the individual media streams included in the aggregate (see Section 13.4.2 for aggregated sessions and for the particular URIs see Appendix D.1.1). The Aggregate control URI is to be specified by the session description if the server supports aggregated control and aggregated control is desired for the session. However, even if aggregated control is offered the client MAY chose to not set up the session in aggregated control. If an Aggregate control URI is not specified in the session description, it is normally an indication that non-aggregated control should be used. The SETUP of media streams in an aggregate which has not been given Schulzrinne, et al. Expires March 15, 2014 [Page 73] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 an aggregated control URI is unspecified. While the session ID sometimes carries enough information for aggregate control of a session, the Aggregate control URI is still important for some methods such as SET_PARAMETER where the control URI enables the resource in question to be easily identified. The Aggregate control URI is also useful for proxies, enabling them to route the request to the appropriate server, and for logging, where it is useful to note the actual resource that a request was operating on. A session will exist until it is either removed by a TEARDOWN request or is timed-out by the server. The server MAY remove a session that has not demonstrated liveness signs from the client(s) within a certain timeout period. The default timeout value is 60 seconds; the server MAY set this to a different value and indicate so in the timeout field of the Session header in the SETUP response. For further discussion see Section 18.49. Signs of liveness for an RTSP session are: o Any RTSP request from a client which includes a Session header with that session's ID. o If RTP is used as a transport for the underlying media streams, an RTCP sender or receiver report from the client(s) for any of the media streams in that RTSP session. RTCP Sender Reports may for example be received in sessions where the server is invited into a conference session and is valid for keep-alive. If a SETUP request on a session fails for any reason, the session state, as well as transport and other parameters for associated streams MUST remain unchanged from their values as if the SETUP request had never been received by the server. 13.3.1. Changing Transport Parameters A client MAY issue a SETUP request for a stream that is already set up or playing in the session to change transport parameters, which a server MAY allow. If it does not allow changing of parameters, it MUST respond with error 455 (Method Not Valid In This State). The reasons to support changing transport parameters include allowing application layer mobility and flexibility to utilize the best available transport as it becomes available. If a client receives a 455 when trying to change transport parameters while the server is in Play state, it MAY try to put the server in Ready state using PAUSE, before issuing the SETUP request again. If that also fails the changing of transport parameters will require that the client performs a TEARDOWN of the affected media and then to set it up Schulzrinne, et al. Expires March 15, 2014 [Page 74] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 again. For an aggregated session avoiding tearing down all the media at the same time will avoid the creation of a new session. All transport parameters MAY be changed. However, the primary usage expected is to either change the transport protocol completely, like switching from Interleaved TCP mode to UDP or vice versa, or to change the delivery address. In a SETUP response for a request to change the transport parameters while in Play state, the server MUST include the Range to indicate at what point the new transport parameters will be used. Further, if RTP is used for delivery, the server MUST also include the RTP-Info header to indicate at what timestamp and RTP sequence number the change will take place. If both RTP-Info and Range are included in the response the "rtp_time" parameter and start point in the Range header MUST be for the corresponding time, i.e., be used in the same way as for PLAY to ensure the correct synchronization information is available. If the transport parameters change while in Play state results in a change of synchronization related information, for example changing RTP SSRC, the server MUST provide in the SETUP response the necessary synchronization information. However, the server is RECOMMENDED to avoid changing the synchronization information if possible. 13.4. PLAY This section describes the usage of the PLAY method in general, for aggregated sessions, and in different usage scenarios. 13.4.1. General Usage The PLAY method tells the server to start sending data via the mechanism specified in SETUP and which part of the media should be played out. PLAY requests are valid when the session is in Ready or Play states. A PLAY request MUST include a Session header to indicate which session the request applies to. Upon receipt of the PLAY request, the server MUST position the normal play time to the beginning of the range specified in the received Range header, within the limits of the media resource and in accordance with the Seek-Style header (Section 18.47) and deliver stream data until the end of the range if given, until a new PLAY request is received, or until the end of the media is reached. If no Range header is present in the PLAY request the server SHALL play from current pause point until the end of media. The pause point defaults at session start to the beginning of the media. For media that is time-progressing and has no retention, the pause point will Schulzrinne, et al. Expires March 15, 2014 [Page 75] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 always be set equal to NPT "now", i.e., the current delivery point. The pause point may also be set to a particular point in the media by the PAUSE method, see Section 13.6. The pause point for media that is currently playing is equal to the current media position. For time-progressing media with time-limited retention, if the pause point represents a position that is older than what is retained by the server, the pause point will be moved to the oldest retained. What range values are valid depends on the type of content. For content that isn't time progressing the range value is valid if the given range is part of any media within the aggregate. In other words the valid media range for the aggregate is the union of all of the media components in the aggregate. If a given range value points outside of the media, the response MUST be the 457 (Invalid Range) error code and include the Media-Range header (Section 18.30) with the valid range for the media. Except for time progressing content where the client requests a start point prior to what is retained, the start point is adjusted to the oldest retained content. For a start point that is beyond the media front edge, i.e., beyond the current value for "now", the server SHALL adjust the start value to the current front edge. The Range header's stop point value may point beyond the current media edge. In that case, the server SHALL deliver media from the requested (and possibly adjusted) start point until the provided stop point, or the end of the media is reached prior to the specified stop point. Please note that if one simply wants to play from a particular start point until the end of media using a Range header with an implicit stop point is RECOMMENDED. If a client requests to start playing at the end of media, either explicitly with a Range header or implicitly with a pause point that is at the end of media, a 457 (Invalid Range) error MUST be sent and include the Media-Range header (Section 18.30). It is specified below that the Range header also must be included in the response and that it will carry the pause point in the media, in the case of the session being in Ready State. Note that this also applies if the pause point or requested start point is at the beginning of the media and a Scale header (Section 18.46) is included with a negative value (playing backwards). For media with random access properties a client may express its preference on which policy for start point selection the server shall use. This is done by including the Seek-Style header (Section 18.47) in the PLAY request. The Seek-Style applied will effect the content of the Range header as it will be adjusted to indicate from what point the media actually is delivered. 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., Schulzrinne, et al. Expires March 15, 2014 [Page 76] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 "npt=0-". If a PLAY request is received without a Range header and media delivery has stopped at the end, the server SHOULD respond with a 457 "Invalid Range" error response. In that response, the current pause point MUST be included in a Range header. All range specifiers in this specification allow for ranges with an implicit start point (e.g., "npt=-30"). When used in a PLAY request, the server treats this as a request to start or resume delivery from the current pause point, ending at the end time specified in the Range header. If the pause point is located later than the given end value, a 457 (Invalid Range) response MUST be given. The example below will play seconds 10 through 25. It also requests the server to deliver media from the first Random Access Point prior to the indicated start point. C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0 CSeq: 835 Session: 12345678 Range: npt=10-25 Seek-Style: RAP User-Agent: PhonyClient/1.2 Servers MUST include a "Range" header in any PLAY response, even if no Range header was present in the request. The response MUST use the same format as the request's range header contained. If no Range header was in the request, the format used in any previous PLAY request within the session SHOULD be used. If no format has been indicated in a previous request the server MAY use any time format supported by the media and indicated in the Accept-Ranges header in the SETUP request. It is RECOMMENDED that NPT is used if supported by the media. For any error response to a PLAY request, the server's response depends on the current session state. If the session is in Ready state, the current pause-point is returned using Range header with the pause point as the explicit start-point and an implicit stop- point. For time-progressing content where the pause-point moves with real-time due to limited retention, the current pause point is returned. For sessions in Play state, the current playout point and the remaining parts of the range request is returned. For any media with retention longer than 0 seconds the currently valid Media-Range header SHALL also be included in the response. A PLAY response MAY include a header carrying synchronization information. As the information necessary is dependent on the media transport format, further rules specifying the header and its usage are needed. For RTP the RTP-Info header is specified, see Schulzrinne, et al. Expires March 15, 2014 [Page 77] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Section 18.45, and used in the following example. Here is a simple example for a single audio stream where the client requests the media starting from 3.52 seconds and to the end. The server sends a 200 OK response with the actual play time which is 10 ms prior (3.51) and the RTP-Info header that contains the necessary parameters for the RTP stack. C->S: PLAY rtsp://example.com/audio RTSP/2.0 CSeq: 836 Session: 12345678 Range: npt=3.52- User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 836 Date: Thu, 23 Jan 1997 15:35:06 GMT Server: PhonyServer/1.0 Range: npt=3.51-324.39 Seek-Style: First-Prior RTP-Info:url="rtsp://example.com/audio" ssrc=0D12F123:seq=14783;rtptime=2345962545 S->C: RTP Packet TS=2345962545 => NPT=3.51 Media duration=0.16 seconds The server replies with the actual start point that will be delivered. This may differ from the requested range if alignment of the requested range to valid frame boundaries is required for the media source. Note that some media streams in an aggregate may need to be delivered from even earlier points. Also, some media formats have a very long duration per individual data unit, therefore it might be necessary for the client to parse the data unit, and select where to start. The server SHALL also indicate which policy it uses for selecting the actual start point by including a Seek-Style header. In the following example the client receives the first media packet that stretches all the way up and past the requested playtime. Thus, it is the client's decision whether to render to the user the time between 3.52 and 7.05, or to skip it. In most cases it is probably most suitable not to render that time period. Schulzrinne, et al. Expires March 15, 2014 [Page 78] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: PLAY rtsp://example.com/audio RTSP/2.0 CSeq: 836 Session: 12345678 Range: npt=7.05- User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 836 Date: Thu, 23 Jan 1997 15:35:06 GMT Server: PhonyServer/1.0 Range: npt=3.52- Seek-Style: First-Prior RTP-Info:url="rtsp://example.com/audio" ssrc=0D12F123:seq=14783;rtptime=2345962545 S->C: RTP Packet TS=2345962545 => NPT=3.52 Duration=4.15 seconds After playing the desired range, the presentation does NOT change to the Ready state, media delivery simply stops. A PAUSE request MUST be issued to make the stream enter the Ready state. A PLAY request while the stream is still in the Play state is legal, and can be issued without an intervening PAUSE request. Such a request MUST replace the current PLAY action with the new one requested, i.e., being handled in the same way as if as the request was received in Ready state. In the case that the range in Range header has an implicit start time ("-endtime"), the server MUST continue to play from where it currently was until the specified end point. This is useful to change the end to at another point than in the previous request. The following example plays the whole presentation starting at SMPTE time code 0:10:20 until the end of the clip. Note: The RTP-Info headers has been broken into several lines, where following lines start with whitespace as allowed by the syntax. Schulzrinne, et al. Expires March 15, 2014 [Page 79] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: PLAY rtsp://audio.example.com/twister.en RTSP/2.0 CSeq: 833 Session: 12345678 Range: smpte=0:10:20- User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 833 Date: Thu, 23 Jan 1997 15:35:06 GMT Session: 12345678 Server: PhonyServer/1.0 Range: smpte=0:10:22-0:15:45 Seek-Style: Next RTP-Info:url="rtsp://example.com/twister.en" ssrc=0D12F123:seq=14783;rtptime=2345962545 For playing back a recording of a live presentation, it may be desirable to use clock units: C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/2.0 CSeq: 835 Session: 12345678 Range: clock=19961108T142300Z-19961108T143520Z User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 835 Date: Thu, 23 Jan 1997 15:35:06 GMT Session: 12345678 Server: PhonyServer/1.0 Range: clock=19961108T142300Z-19961108T143520Z Seek-Style: Next RTP-Info:url="rtsp://example.com/meeting.en" ssrc=0D12F123:seq=53745;rtptime=484589019 13.4.2. Aggregated Sessions PLAY requests can operate on sessions controlling a single media and on aggregated sessions controlling multiple media. In an aggregated session the PLAY request MUST contain an aggregated control URI. A server MUST respond with error 460 (Only Aggregate Operation Allowed) if the client PLAY Request-URI is for a single media. The media in an aggregate MUST be played in sync. If a client wants individual control of the media, it needs to use separate RTSP sessions for each media. For aggregated sessions where the initial SETUP request (creating a Schulzrinne, et al. Expires March 15, 2014 [Page 80] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 session) is followed by one or more additional SETUP requests, a PLAY request MAY be pipelined after those additional SETUP requests without awaiting their responses. This procedure can reduce the delay from start of session establishment until media play-out has started with one round trip time. However, a client needs to be aware that using this procedure will result in the playout of the server state established at the time of processing the PLAY, i.e., after the processing of all the requests prior to the PLAY request in the pipeline. This state may not be the intended one due to failure of any of the prior requests. A client can easily determine this based on the responses from those requests. In case of failure, the client can halt the media playout using PAUSE and try to establish the intended state again before issuing another PLAY request. 13.4.3. Updating current PLAY Requests Clients can issue PLAY requests while the stream is in Play state and thus updating their request. The important difference compared to a PLAY request in Ready state is the handling of the current play point and how the Range header in the request is constructed. The session is actively playing media and the play point will be moving, making the exact time a request will take effect hard to predict. Depending on how the PLAY header appears two different cases exist: total replacement or continuation. A total replacement is signaled by having the first range specification have an explicit start value, e.g., "npt=45-" or "npt=45-60", in which case the server stops playout at the current playout point and then starts delivering media according to the Range header. This is equivalent to having the client first send a PAUSE and then a new PLAY request that isn't based on the pause point. In the case of continuation the first range specifier has an implicit start point and an explicit stop value (Z), e.g., "npt=-60", which indicate that it MUST convert the range specifier being played prior to this PLAY request (X to Y) into (X to Z) and continue as this was the request originally played. If the current delivery point is beyond the stop point, the server SHALL immediately pause delivery. As the request has been completed successfully it shall be responded with 200 OK. A PLAY_NOTIFY with end-of-stream is also sent to indicate the actual stop point. The pause point is set to the requested stop point. Following is an example of this behavior: The server has received requests to play ranges 10 to 15. If the new PLAY request arrives at the server 4 seconds after the previous one, it will take effect while the server still plays the first range (10-15). The server changes the current play to continue to 25 seconds, i.e., the equivalent single request would be PLAY with "range: npt=10-25". Schulzrinne, et al. Expires March 15, 2014 [Page 81] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 834 Session: 12345678 Range: npt=10-15 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 834 Date: Thu, 23 Jan 1997 15:35:06 GMT Session: 12345678 Server: PhonyServer/1.0 Range: npt=10-15 Seek-Style: Next RTP-Info:url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=5712;rtptime=934207921, url="rtsp://example.com/fizzle/videotrack" ssrc=789DAF12:seq=57654;rtptime=2792482193 Session: 12345678 C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 835 Session: 12345678 Range: npt=-25 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 835 Date: Thu, 23 Jan 1997 15:35:09 GMT Session: 12345678 Server: PhonyServer/1.0 Range: npt=14-25 Seek-Style: Next RTP-Info:url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=5712;rtptime=934239921, url="rtsp://example.com/fizzle/videotrack" ssrc=789DAF12:seq=57654;rtptime=2792842193 A common use of a PLAY request while in Play state is changing the scale of the media, i.e., entering or leaving fast forward or fast rewind. The client can issue an updating PLAY request that is either a continuation or a complete replacement, as discussed above this section. We give an example of a client that is requesting a fast forward (scale=2) without giving a stop point and then change from fast forward to regular playout (scale = 1). In the second PLAY request the time is set explicitly to be where ever the server currently plays out (npt=now-) and the server responds with the actual playback point where the new scale actually takes effect (npt=2:17:27.144-). Schulzrinne, et al. Expires March 15, 2014 [Page 82] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 2034 Session: 12345678 Range: npt=now- Scale: 2.0 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 2034 Date: Thu, 23 Jan 1997 15:35:06 GMT Session: 12345678 Server: PhonyServer/1.0 Range: npt=2:17:21.394- Seek-Style: Next RTP-Info:url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=5712;rtptime=934207921, url="rtsp://example.com/fizzle/videotrack" ssrc=789DAF12:seq=57654;rtptime=2792482193 [playing in fast forward and now returning to scale = 1] C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 2035 Session: 12345678 Range: npt=now- Scale: 1.0 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 2035 Date: Thu, 23 Jan 1997 15:35:09 GMT Session: 12345678 Server: PhonyServer/1.0 Range: npt=2:17:27.144- Seek-Style: Next RTP-Info:url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=5712;rtptime=934239921, url="rtsp://example.com/fizzle/videotrack" ssrc=789DAF12:seq=57654;rtptime=2792842193 13.4.4. Playing On-Demand Media On-demand media is indicated by the content of the Media-Properties header in the SETUP response by (see also Section 18.29): o Random Access property is set to Random-Access; Schulzrinne, et al. Expires March 15, 2014 [Page 83] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o Content Modifications set to Immutable; o Retention set to Unlimited or Time-Limited. Playing on-demand media follows the general usage as described in Section 13.4.1. 13.4.5. Playing Dynamic On-Demand Media Dynamic on-demand media is indicated by the content of the Media- Properties header in the SETUP response by (see also Section 18.29): o Random Access set to Random-Access; o Content Modifications set to Dynamic; o Retention set to Unlimited or Time-Limited. Playing on-demand media follows the general usage as described in Section 13.4.1 as long as the media has not been changed. There are two ways for the client to be informed about changes of media resources in Play state. The client will receive a PLAY_NOTIFY request with Notify-Reason header set to media-properties-update (see Section 13.5.2. The client can use the value of the Media-Range to decide further actions, if the Media-Range header is present in the PLAY_NOTIFY request. The second way is that the client issues a GET_PARAMETER request without a body but including a Media-Range header. The 200 OK response MUST include the current Media-Range header (see Section 18.30). 13.4.6. Playing Live Media Live media is indicated by the content of the Media-Properties header in the SETUP response by (see also Section 18.29): o Random-Access set to No-Seeking; o Content Modifications set to Time-Progressing; o Retention with Time-Duration set to 0.0. For live media, the SETUP response 200 OK MUST include the Media- Range header (see Section 18.30). A client MAY send PLAY requests without the Range header. If the request includes the Range header it MUST use a symbolic value representing "now". For NPT that range specification is "npt=now-". Schulzrinne, et al. Expires March 15, 2014 [Page 84] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 The server MUST include the Range header in the response and it MUST indicate an explicit time value and not a symbolic value. In other words, "npt=now-" is not valid to be used in the response. Instead the time since session start is recommended expressed as an open interval, e.g., "npt=96.23-". An absolute time value (clock) for the corresponding time MAY be given, i.e., "clock=20030213T143205Z-". The Absolute Time format can only be used if client has shown support for it using the Accept-Ranges header. 13.4.7. Playing Live with Recording Certain media servers may offer recording services of live sessions to their clients. This recording would normally be from the beginning of the media session. Clients can randomly access the media between now and the beginning of the media session. This live media with recording is indicated by the content of the Media- Properties header in the SETUP response by (see also Section 18.29): o Random Access set to Random-Access; o Content Modifications set to Time-Progressing; o Retention set to Time-Limited or Unlimited The SETUP response 200 OK MUST include the Media-Range header (see Section 18.30) for this type of media. For live media with recording, the Range header indicates the current delivery point in the media and the Media-Range header indicates the currently available media window around the current time. This window can cover recorded content in the past (seen from current time in the media) or recorded content in the future (seen from current time in the media). The server adjusts the delivery point to the requested border of the window. If the client requests a delivery point that is located outside the recording window, e.g., if the requested point is too far in the past, the server selects the oldest point in the recording. The considerations in Section 13.5.3 apply if a client requests delivery with Scale (Section 18.46) values other than 1.0 (Normal playback rate) while delivering live media with recording. 13.4.8. Playing Live with Time-Shift Certain media servers may offer time-shift services to their clients. This time shift records a fixed interval in the past, i.e., a sliding window recording mechanism, but not past this interval. Clients can randomly access the media between now and the interval. This live media with recording is indicated by the content of the Media- Properties header in the SETUP response by (see also Section 18.29): Schulzrinne, et al. Expires March 15, 2014 [Page 85] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o Random Access set to Random-Access; o Content Modifications set to Time-Progressing; o Retention set to Time-Duration and a value indicating the recording interval (>0). The SETUP response 200 OK MUST include the Media-Range header (see Section 18.30) for this type of media. For live media with recording the Range header indicates the current time in the media and the Media Range indicates a window around the current time. This window can cover recorded content in the past (seen from current time in the media) or recorded content in the future (seen from current time in the media). The server adjusts the play point to the requested border of the window, if the client requests a play point that is located outside the recording windows, e.g., if requested too far in the past, the server selects the oldest range in the recording. The considerations in Section 13.5.3 apply, if a client requests delivery using a Scale (Section 18.46) value other than 1.0 (Normal playback rate) while delivering live media with time-shift. 13.5. PLAY_NOTIFY The PLAY_NOTIFY method is issued by a server to inform a client about an asynchronous event for a session in Play state. The Session header MUST be presented in a PLAY_NOTIFY request and indicates the scope of the request. Sending of PLAY_NOTIFY requests requires a persistent connection between server and client, otherwise there is no way for the server to send this request method to the client. PLAY_NOTIFY requests have an end-to-end (i.e., server to client) scope, as they carry the Session header, and apply only to the given session. The client SHOULD immediately return a response to the server. PLAY_NOTIFY requests MAY use both aggregate control URI and individual media resource URIs depending on scope of the notification. This scope may have important distinctions for aggregated sessions, and each reason for a PLAY_NOTIFY request needs to specify the interpretation and if aggregated control URIs or individual URIs may be used in requests. PLAY_NOTIFY requests MAY be used with a message body, depending on the value of the Notify-Reason header. It is described in the particular section for each Notify-Reason if a message body is used. However, currently there is no Notify-Reason that allows using a message body. In this case, there is a need to obey some limitations when adding new Notify-Reasons that intend to use a message body: the Schulzrinne, et al. Expires March 15, 2014 [Page 86] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 server can send any type of message body, but it is not ensured that the client can understand the received message body. This is related to DESCRIBE (see Section 13.2 ), but in this particular case the client can state its acceptable message bodies by using the Accept header. In the case of PLAY_NOTIFY, the server does not know which message bodies are understood by the client. The Notify-Reason header (see Section 18.32) specifies the reason why the server sends the PLAY_NOTIFY request. This is extensible and new reasons MAY be added in the future (see Section 22.8). In case the client does not understand the reason for the notification it MUST respond with a 465 (Notification Reason Unknown) (Section 17.4.29) error code. Servers can send PLAY_NOTIFY with these types: o end-of-stream (see Section 13.5.1); o media-properties-update (see Section 13.5.2); o scale-change (see Section 13.5.3). 13.5.1. End-of-Stream A PLAY_NOTIFY request with Notify-Reason header set to end-of-stream indicates the completion or near completion of the PLAY request and the ending delivery of the media stream(s). The request MUST NOT be issued unless the server is in the Play state. The end of the media stream delivery notification may be used to indicate either a successful completion of the PLAY request currently being served, or to indicate some error resulting in failure to complete the request. The Request-Status header (Section 18.42) MUST be included to indicate which request the notification is for and its completion status. The message response status codes (Section 8.1.1) are used to indicate how the PLAY request concluded. The sender of a PLAY_NOTIFY can issue an updated PLAY_NOTIFY, in the case of a PLAY_NOTIFY sent with wrong information. For instance, a PLAY_NOTIFY was issued before reaching the end-of-stream, but some error occurred resulting in that the previously sent PLAY_NOTIFY contained a wrong time when the stream will end. In this case a new PLAY_NOTIFY MUST be sent including the correct status for the completion and all additional information. PLAY_NOTIFY requests with Notify-Reason header set to end-of-stream MUST include a Range header and the Scale header if the scale value is not 1. The Range header indicates the point in the stream or streams where delivery is ending with the timescale that was used by the server in the PLAY response for the request being fulfilled. The server MUST NOT use the "now" constant in the Range header; it MUST use the actual numeric end position in the proper timescale. When Schulzrinne, et al. Expires March 15, 2014 [Page 87] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 end-of-stream notifications are issued prior to having sent the last media packets, this is evident as the end time in the Range header is beyond the current time in the media being received by the client, e.g., "npt=-15", if npt is currently at 14.2 seconds. The Scale header is to be included so that it is evident if the media time scale is moving backwards and/or have a non-default pace. The end- of-stream notification does not prevent the client from sending a new PLAY request. If RTP is used as media transport, a RTP-Info header MUST be included, and the RTP-Info header MUST indicate the last sequence number in the seq parameter. For RTSP Session where media resources under aggregated control the media resources will normally end at approximately the same time, although some small differences may exist, on the scale of a few hundred of milliseconds. In those cases a RTSP session under aggregated control SHOULD send only a single PLAY_NOTIFY request. By using the aggregate control URI in the PLAY_NOTIFY request the RTSP server indicates that this applies to all media resources within the session. In cases RTP is used for media delivery corresponding RTP- Info needs to be included for all media resources. In cases where one or more media resource has significantly shorter duration than some other resources in the aggregated session the server MAY send end-of-stream notifications using individual media resource URIs to indicate to agents that there will be no more media for this particular media resource related to the current active PLAY request. In such cases when the remaining media resources comes to end-of- stream they MUST send a PLAY_NOTIFY request using the aggregate control URI to indicate that no more resources remains. A PLAY_NOTIFY request with Notify-Reason header set to end-of-stream MUST NOT carry a message body. This example request notifies the client about a future end-of-stream event: Schulzrinne, et al. Expires March 15, 2014 [Page 88] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 854 Notify-Reason: end-of-stream Request-Status: cseq=853 status=200 reason="OK" Range: npt=-145 RTP-Info:url="rtsp://example.com/fizzle/foo/audio" ssrc=0D12F123:seq=14783;rtptime=2345962545, url="rtsp://example.com/fizzle/video" ssrc=789DAF12:seq=57654;rtptime=2792482193 Session: uZ3ci0K+Ld-M Date: Mon, 08 Mar 2010 13:37:16 GMT C->S: RTSP/2.0 200 OK CSeq: 854 User-Agent: PhonyClient/1.2 Session: uZ3ci0K+Ld-M 13.5.2. Media-Properties-Update A PLAY_NOTIFY request with Notify-Reason header set to media- properties-update indicates an update of the media properties for the given session (see Section 18.29) and/or the available media range that can be played as indicated by Media-Range (Section 18.30). PLAY_NOTIFY requests with Notify-Reason header set to media- properties-update MUST include a Media-Properties and Date header and SHOULD include a Media-Range header. The Media-Properties header has session scope, thus for aggregated sessions the PLAY_NOTIFY request MUST be using the aggregated control URI. This notification MUST be sent for media that are Time-Progressing every time an event happens that changes the basis for making estimates on how the available for play-back media range will progress with wall clock time. In addition it is RECOMMENDED that the server sends these notifications approximately every 5 minutes for time-progressing content to ensure the long-term stability of the client estimation and allowing for clock skew detection by the client. The time between notifications should be greater than 1 minute and less than 2 hours. Requests for the just mentioned reasons MUST include Media-Range header to provide current Media duration and the Range header to indicate the current playing point and any remaining parts of the requested range. The recommendation for sending updates every 5 minutes is due to any clock skew issues. In 5 minutes the clock skew should not become too significant as this is not used for media playback and synchronization, only for determining which content is available to the user. Schulzrinne, et al. Expires March 15, 2014 [Page 89] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 A PLAY_NOTIFY request with Notify-Reason header set to media- properties-update MUST NOT carry a message body. S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0 Date: Tue, 14 Apr 2008 15:48:06 GMT CSeq: 854 Notify-Reason: media-properties-update Session: uZ3ci0K+Ld-M Media-Properties: Time-Progressing, Time-Limited=20080415T153919.36Z, Random-Access=5.0 Media-Range: npt=0-1:37:21.394 Range: npt=1:15:49.873- C->S: RTSP/2.0 200 OK CSeq: 854 User-Agent: PhonyClient/1.2 Session: uZ3ci0K+Ld-M 13.5.3. Scale-Change The server may be forced to change the rate of media time per playback time when a client requests delivery using a Scale (Section 18.46) value other than 1.0 (normal playback rate). For time progressing media with some retention, i.e., the server stores already sent content, a client requesting to play with Scale values larger than 1 may catch up with the front end of the media. The server will then be unable to continue to provide content at Scale larger than 1 as content is only made available by the server at Scale=1. Another case is when Scale < 1 and the media retention is time-duration limited. In this case the delivery point can reach the oldest media unit available, and further playback at this scale becomes impossible as there will be no media available. To avoid having the client lose any media, the scale will need to be adjusted to the same rate at which the media is removed from the storage buffer, commonly Scale = 1.0. Another case is when the content itself consists of spliced pieces or is dynamically updated. In these cases the server may be required to change from one supported scale value (different than Scale=1.0) to another. In this case the server will pick the closest value and inform the client of what it has picked. In these cases the media properties will also be sent updating the supported Scale values. This enables a client to adjust the Scale value used. To minimize impact on playback in any of the above cases the server MUST modify the playback properties and set Scale to a supportable value and continue delivery of the media. When doing this modification it MUST send a PLAY_NOTIFY message with the Notify- Schulzrinne, et al. Expires March 15, 2014 [Page 90] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Reason header set to "scale-change". The request MUST contain a Range header with the media time when the change took effect, a Scale header with the new value in use, Session header with the identifier for the session it applies to and a Date header with the server wallclock time of the change. For time progressing content also the Media-Range and the Media-Properties at this point in time MUST be included. The Media-Properties header MUST be included if the scale change was due to the content changing what scale values that is supported. For media streams being delivered using RTP also a RTP-Info header MUST be included. It MUST contain the rtptime parameter with a value corresponding to the point of change in that media and optionally also the sequence number. PLAY_NOTIFY requests for aggregated sessions MUST use the aggregated control URI in the request. The scale change for any aggregated session do apply to all media streams part of the aggregate. A PLAY_NOTIFY request with Notify-Reason header set to "Scale-Change" MUST NOT carry a message body. S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0 Date: Tue, 14 Apr 2008 15:48:06 GMT CSeq: 854 Notify-Reason: scale-change Session: uZ3ci0K+Ld-M Media-Properties: Time-Progressing, Time-Limited=20080415T153919.36Z, Random-Access=5.0 Media-Range: npt=0-1:37:21.394 Range: npt=1:37:21.394- Scale: 1 RTP-Info: url="rtsp://example.com/fizzle/foo/audio" ssrc=0D12F123:rtptime=2345962545, url="rtsp://example.com/fizzle/videotrack" ssrc=789DAF12:seq=57654;rtptime=2792482193 C->S: RTSP/2.0 200 OK CSeq: 854 User-Agent: PhonyClient/1.2 Session: uZ3ci0K+Ld-M 13.6. PAUSE The PAUSE request causes the stream delivery to immediately be interrupted (halted). A PAUSE request MUST be done either with the aggregated control URI for aggregated sessions, resulting in all media being halted, or the media URI for non-aggregated sessions. Schulzrinne, et al. Expires March 15, 2014 [Page 91] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Any attempt to do muting of a single media with a PAUSE request in an aggregated session MUST be responded to with error 460 (Only Aggregate Operation Allowed). After resuming playback, synchronization of the tracks MUST be maintained. Any server resources are kept, though servers MAY close the session and free resources after being paused for the duration specified with the timeout parameter of the Session header in the SETUP message. Example: C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 834 Session: 12345678 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 834 Date: Thu, 23 Jan 1997 15:35:06 GMT Range: npt=45.76-75.00 The PAUSE request causes stream delivery to be interrupted immediately on receipt of the message and the pause point is set to the current point in the presentation. That pause point in the media stream needs to be maintained. A subsequent PLAY request without Range header resumes from the pause point and plays until media end. The pause point after any PAUSE request MUST be returned to the client by adding a Range header with what remains unplayed of the PLAY request's range. For media with random access properties, if one desires to resume playing a ranged request, one simply includes the Range header from the PAUSE response and includes the Seek-Style header with the Next policy in the PLAY request. For media that is time-progressing and has retention duration=0 the follow-up PLAY request to start media delivery again, MUST use "npt=now-" and not the answer given in the response to PAUSE. Schulzrinne, et al. Expires March 15, 2014 [Page 92] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 834 Session: 12345678 Range: npt=10-30 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 834 Date: Thu, 23 Jan 1997 15:35:06 GMT Server: PhonyServer/1.0 Range: npt=10-30 Seek-Style: First-Prior RTP-Info:url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=5712;rtptime=934207921, url="rtsp://example.com/fizzle/videotrack" ssrc=4FAD8726:seq=57654;rtptime=2792482193 Session: 12345678 After 11 seconds, i.e., at 21 seconds into the presentation: C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 835 Session: 12345678 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 835 Date: 23 Jan 1997 15:35:17 GMT Server: PhonyServer/1.0 Range: npt=21-30 Session: 12345678 If a client issues a PAUSE request and the server acknowledges and enters the Ready state, the proper server response, if the player issues another PAUSE, is still 200 OK. The 200 OK response MUST include the Range header with the current pause point. See examples below: Schulzrinne, et al. Expires March 15, 2014 [Page 93] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 834 Session: 12345678 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 834 Session: 12345678 Date: Thu, 23 Jan 1997 15:35:06 GMT Range: npt=45.76-98.36 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 835 Session: 12345678 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 835 Session: 12345678 Date: 23 Jan 1997 15:35:07 GMT Range: npt=45.76-98.36 13.7. TEARDOWN 13.7.1. Client to Server The TEARDOWN client to server request stops the stream delivery for the given URI, freeing the resources associated with it. A TEARDOWN request MAY be performed on either an aggregated or a media control URI. However, some restrictions apply depending on the current state. The TEARDOWN request MUST contain a Session header indicating what session the request applies to. The TEARDOWN request MUST NOT include a Terminate-Reason header. A TEARDOWN using the aggregated control URI or the media URI in a session under non-aggregated control (single media session) MAY be done in any state (Ready and Play). A successful request MUST result in that media delivery being immediately halted and the session state being destroyed. This MUST be indicated through the lack of a Session header in the response. A TEARDOWN using a media URI in an aggregated session 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 a session returning to non-aggregated control, because it only contains a single media after the request's completion. A session that will exist after the processing of the TEARDOWN request MUST in the response to that TEARDOWN request contain a Session header. Thus Schulzrinne, et al. Expires March 15, 2014 [Page 94] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 the presence of the Session header indicates to the receiver of the response if the session is still existing or has been removed. Example: C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 892 Session: 12345678 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 892 Server: PhonyServer/1.0 13.7.2. Server to Client The server can send TEARDOWN requests in the server to client direction to indicate that the server has been forced to terminate the ongoing session. This may happen for several reasons, such as server maintenance without available backup, or that the session has been inactive for extended periods of time. The reason is provided in the Terminate-Reason header (Section 18.52). When a RTSP client has maintained a RTSP session that otherwise is inactive for an extended period of time the server may reclaim the resources. That is done by issuing a TEARDOWN request with the Terminate-Reason set to "Session-Timeout". This MAY be done when the client has been inactive in the RTSP session for more than one Session Timeout period (Section 18.49). However, the server is RECOMMENDED to not perform this operation until an extended period of inactivity of 10 times the Session Timeout period has passed. It is to the operator of the RTSP server to actually configure how long this extended period of inactivity is. An operator should take into account when doing this configuration what the served content is and what this means for the extended period of inactivity. In case the server needs to stop providing service to the established sessions and there is no server to point at in a REDIRECT request, then TEARDOWN SHALL be used to terminate the session. This method can also be used when non-recoverable internal errors have happened and the server has no other option then to terminate the sessions. The TEARDOWN request MUST be done only on the session aggregate control URI (i.e., it is not allowed to terminate individual media streams, if it is a session aggregate) and MUST include the following headers; Session and Terminate-Reason headers. The request only applies to the session identified in the Session header. The server may include a message to the client's user with the "user-msg" Schulzrinne, et al. Expires March 15, 2014 [Page 95] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 parameter. The TEARDOWN request may alternatively be done on the wild card URI * and without any session header. The scope of such a request is limited to the next-hop (i.e., the RTSP agent in direct communication with the server) and applies, as well, to the RTSP connection between the next-hop RTSP agent and the server. This request indicates that all sessions and pending requests being managed via the connection are terminated. Any intervening proxies SHOULD do all of the following in the order listed: 1. respond to the TEARDOWN request 2. disconnect the control channel from the requesting server 3. pass the TEARDOWN request to each applicable client (typically those clients with an active session or an unanswered request) Note: The proxy is responsible for accepting TEARDOWN responses from its clients; these responses MUST NOT be passed on to either the original server or the target server in the redirect. 13.8. GET_PARAMETER The GET_PARAMETER request retrieves the value of any specified parameter or parameters for a presentation or stream specified in the URI. If the Session header is present in a request, the value of a parameter MUST be retrieved in the specified session context. There are two ways of specifying the parameters to be retrieved. The first is by including headers which have been defined such that you can use them for this purpose. Headers for this purpose should allow empty, or stripped value parts to avoid having to specify bogus data when indicating the desire to retrieve a value. The successful completion of the request should also be evident from any filled out values in the response. The headers in this specification that MAY be used for retrieving their current value using GET_PARAMETER below. Additional headers MAY be specified in the future: o Accept-Ranges o Media-Range o Media-Properties o Range Schulzrinne, et al. Expires March 15, 2014 [Page 96] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o RTP-Info The other way is to specify a message body that lists the parameter(s) that are desired to be retrieved. The Content-Type header (Section 18.19) is used to specify which format the message body has. If the receiver of the request is not supporting the media type used for the message body, it SHALL respond using the error code 415 (Unsupported Media Type). The responder to a GET_PARAMETER request MUST use the media type of the request for the response. For additional considerations regarding message body negotiation see Section 9.3. RTSP Agents implementing support for responding to GET_PARAMETER requests SHALL implement the text/parameters format (Appendix F). This to ensure that at least one known format for parameter are implemented and thus prevent parameter format negotiation failure. Parameters specified within the body of the message must all be understood by the request receiving agent. If one or more parameters are not understood a 451 (Parameter Not Understood) MUST be sent including a body listing these parameters that weren't understood. If all parameters are understood their values are filled in and returned in the response message body. The method MAY also be used without a message body or any header that requests parameters for keep-alive purpose. The keep-alive timer has been updated for any request that is successful, i.e., a 200 OK response is received. Any non-required header present in such a request may or may not have been processed. Normally the presence of filled out values in the header will be indication that the header has been processed. However, for cases when this is difficult to determine, it is recommended to use a feature-tag and the Require header. For this reason it is usually easier if any parameters to be retrieved are sent in the body, rather than using any header. Example: Schulzrinne, et al. Expires March 15, 2014 [Page 97] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 431 User-Agent: PhonyClient/1.2 Session: 12345678 Content-Length: 26 Content-Type: text/parameters packets_received jitter C->S: RTSP/2.0 200 OK CSeq: 431 Session: 12345678 Server: PhonyServer/1.1 Date: Mon, 08 Mar 2010 13:43:23 GMT Content-Length: 38 Content-Type: text/parameters packets_received: 10 jitter: 0.3838 13.9. SET_PARAMETER This method requests to set the value of a parameter or a set of parameters for a presentation or stream specified by the URI. The method MAY also be used without a message body. It is the RECOMMENDED method to be used in a request sent for the sole purpose of updating the keep-alive timer. If this request is successful, i.e., a 200 OK response is received, then the keep-alive timer has been updated. Any non-required header present in such a request may or may not have been processed. To allow a client to determine if any such header has been processed, it is necessary to use a feature tag and the Require header. Due to this reason it is RECOMMENDED that any parameters are sent in the body, rather than using any header. When using a message body to list the parameter(s) that are desired to be set the Content-Type header (Section 18.19) is used to specify which format the message body has. If the receiver of the request is not supporting the media type used for the message body, it SHALL respond using the error code 415 (Unsupported Media Type). For additional considerations regarding message body negotiation see Section 9.3. RTSP Agents implementing support for responding to SET_PARAMETER requests SHALL implement the text/parameters format (Appendix F). This to ensure that at least one known format for parameters are implemented and thus prevent parameter format negotiation failure. Schulzrinne, et al. Expires March 15, 2014 [Page 98] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 A request is RECOMMENDED to only contain a single parameter to allow the client to determine why a particular request failed. If the request contains several parameters, the server MUST only act on the request if all of the parameters can be set successfully. A server MUST allow a parameter to be set repeatedly to the same value, but it MAY disallow changing parameter values. If the receiver of the request does not understand or cannot locate a parameter, error 451 (Parameter Not Understood) MUST be used. When a parameter is not allowed to change, the error code is 458 (Parameter Is Read-Only). The response body MUST contain only the parameters that have errors. Otherwise no body MUST be returned. The response body MUST use the media type of the request for the response. Note: transport parameters for the media stream MUST only be set with the SETUP command. Restricting setting transport parameters to SETUP is for the benefit of firewalls. The parameters are split in a fine-grained fashion so that there can be more meaningful error indications. However, it may make sense to allow the setting of several parameters if an atomic setting is desirable. Imagine device control where the client does not want the camera to pan unless it can also tilt to the right angle at the same time. Example: C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 421 User-Agent: PhonyClient/1.2 Session: iixT43KLc Date: Mon, 08 Mar 2010 14:45:04 GMT Content-length: 20 Content-type: text/parameters barparam: barstuff S->C: RTSP/2.0 451 Parameter Not Understood CSeq: 421 Session: iixT43KLc Server: PhonyServer/1.0 Date: Mon, 08 Mar 2010 14:44:56 GMT Content-length: 20 Content-type: text/parameters barparam: barstuff Schulzrinne, et al. Expires March 15, 2014 [Page 99] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 13.10. REDIRECT The REDIRECT method is issued by a server to inform a client that the service provided will be terminated and where a corresponding service can be provided instead. This may happen for different reasons. One is that the server is being administered such that it must stop providing service. Thus the client is required to connect to another server location to access the resource indicated by the Request-URI. The REDIRECT request SHALL contain a Terminate-Reason header (Section 18.52) to inform the client of the reason for the request. Additional parameters related to the reason may also be included. The intention here is to allow a server administrator to do a controlled shutdown of the RTSP server. That requires sufficient time to inform all entities having associated state with the server and for them to perform a controlled migration from this server to a fall back server. A REDIRECT request with a Session header has end-to-end (i.e., server to client) scope and applies only to the given session. Any intervening proxies SHOULD NOT disconnect the control channel while there are other remaining end-to-end sessions. The REQUIRED Location header MUST contain a complete absolute URI pointing to the resource to which the client SHOULD reconnect. Specifically, the Location MUST NOT contain just the host and port. A client may receive a REDIRECT request with a Session header, if and only if, an end-to-end session has been established. A client may receive a REDIRECT request without a Session header at any time when it has communication or a connection established with a server. The scope of such a request is limited to the next-hop (i.e., the RTSP agent in direct communication with the server) and applies to all sessions controlled, as well as the connection between the next-hop RTSP agent and the server. A REDIRECT request without a Session header indicates that all sessions and pending requests being managed via the connection MUST be redirected. The Location header, if included in such a request, SHOULD contain an absolute URI with only the host address and the OPTIONAL port number of the server to which the RTSP agent SHOULD reconnect. Any intervening proxies SHOULD do all of the following in the order listed: 1. respond to the REDIRECT request 2. disconnect the control channel from the requesting server 3. connect to the server at the given host address Schulzrinne, et al. Expires March 15, 2014 [Page 100] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 4. pass the REDIRECT request to each applicable client (typically those clients with an active session or an unanswered request) Note: The proxy is responsible for accepting REDIRECT responses from its clients; these responses MUST NOT be passed on to either the original server or the redirected server. When the server lacks any alternative server and needs to terminate a session or all sessions the TEARDOWN request SHALL be used instead. When no Terminate-Reason "time" parameter is included in a REDIRECT request, the client SHALL perform the redirection immediately and return a response to the server. The server shall consider the session as terminated and can free any associated state after it receives the successful (2xx) response. The server MAY close the signaling connection upon receiving the response and the client SHOULD close the signaling connection after sending the 2xx response. The exception to this is when the client has several sessions on the server being managed by the given signaling connection. In this case, the client SHOULD close the connection when it has received and responded to REDIRECT requests for all the sessions managed by the signaling connection. The Terminate-Reason header "time" parameter MAY be used to indicate the wallclock time by when the redirection MUST have taken place. To allow a client to determine that redirect time without being time synchronized with the server, the server MUST include a Date header in the request. The client should have terminated the session and closed the connection before the redirection time-line terminated. The server MAY simply cease to provide service when the deadline time has been reached, or it may issue TEARDOWN requests to the remaining sessions. If the REDIRECT request times out following the rules in Section 10.4 the server MAY terminate the session or transport connection that would be redirected by the request. This is a safeguard against misbehaving clients that refuse to respond to a REDIRECT request. Thus, removing any incentive to not acknowledge the reception of a REDIRECT request. After a REDIRECT request has been processed, a client that wants to continue to receive media for the resource identified by the Request- URI will have to establish a new session with the designated host. If the URI given in the Location header is a valid resource URI, a client SHOULD issue a DESCRIBE request for the URI. Schulzrinne, et al. Expires March 15, 2014 [Page 101] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Note: The media resource indicated by the Location header can be identical, slightly different or totally different. This is the reason why a new DESCRIBE request SHOULD be issued. If the Location header contains only a host address, the client MAY assume that the media on the new server is identical to the media on the old server, i.e., all media configuration information from the old session is still valid except for the host address. However, the usage of conditional SETUP using MTag identifiers is RECOMMENDED as a means to verify the assumption. This example request redirects traffic for this session to the new server at the given absolute time: S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0 CSeq: 732 Location: rtsp://s2.example.com:8001 Terminate-Reason: Server-Admin ;time=19960213T143205Z Session: uZ3ci0K+Ld-M Date: Thu, 13 Feb 1996 14:30:43 GMT C->S: RTSP/2.0 200 OK CSeq: 732 User-Agent: PhonyClient/1.2 Session: uZ3ci0K+Ld-M Schulzrinne, et al. Expires March 15, 2014 [Page 102] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 14. Embedded (Interleaved) Binary Data In order to fulfill certain requirements on the network side, e.g., in conjunction with network address translators that block RTP traffic over UDP, it may be necessary to interleave RTSP messages and media stream data. This interleaving should generally be avoided unless necessary since it complicates client and server operation and imposes additional overhead. Also, head of line blocking may cause problems. Interleaved binary data SHOULD only be used if RTSP is carried over TCP. Interleaved data is not allowed inside RTSP messages. Stream data such as RTP packets is encapsulated by an ASCII dollar sign (36 decimal), followed by a one-octet channel identifier, followed by the length of the encapsulated binary data as a binary, two-octet unsigned integer in network byte order. The stream data follows immediately afterwards, without a CRLF, but including the upper-layer protocol headers. Each $ block MUST contain exactly one upper-layer protocol data unit, e.g., one RTP packet. Note that this mechanism does not support larger PDUs than 65535 bytes, which is the same that regular IPv4 and IPv6 is capable. If the media delivery protocol intended to be used has larger PDUs than that, the definition of such mechanisms usage of this mechanism will require the definition of a PDU fragmentation mechanism. 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | "$" = 36 | Channel ID | Length in octets | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : Binary data (Length according to Length field) : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 1: Embedded Interleaved Binary Data Format The channel identifier is defined in the Transport header with the interleaved parameter (Section 18.54). When the transport choice is RTP, RTCP messages are also interleaved by the server over the TCP connection. The usage of RTCP messages is indicated by including an interval containing a second channel in the interleaved parameter of the Transport header, see Section 18.54. If RTCP is used, packets MUST be sent on the first available channel higher than the RTP channel. The channels are bi-directional, using the same Channel ID in both directions, and therefore RTCP traffic is Schulzrinne, et al. Expires March 15, 2014 [Page 103] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 sent on the second channel in both directions. RTCP is sometimes needed for synchronization when two or more streams are interleaved in such a fashion. Also, this provides a convenient way to tunnel RTP/RTCP packets through the RTSP connection (TCP or TCP/TLS) when required by the network configuration and transfer them onto UDP when possible. C->S: SETUP rtsp://example.com/bar.file RTSP/2.0 CSeq: 2 Transport: RTP/AVP/TCP;unicast;interleaved=0-1 Accept-Ranges: npt, smpte, clock User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 2 Date: Thu, 05 Jun 1997 18:57:18 GMT Transport: RTP/AVP/TCP;unicast;interleaved=5-6 Session: 12345678 Accept-Ranges: npt Media-Properties: Random-Access=0.2, Immutable, Unlimited C->S: PLAY rtsp://example.com/bar.file RTSP/2.0 CSeq: 3 Session: 12345678 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 3 Session: 12345678 Date: Thu, 05 Jun 1997 18:57:19 GMT RTP-Info: url="rtsp://example.com/bar.file" ssrc=0D12F123:seq=232433;rtptime=972948234 Range: npt=0-56.8 Seek-Style: RAP S->C: $005{2 octet length}{"length" octets data, w/RTP header} S->C: $005{2 octet length}{"length" octets data, w/RTP header} S->C: $006{2 octet length}{"length" octets RTCP packet} Schulzrinne, et al. Expires March 15, 2014 [Page 104] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 15. Proxies RTSP Proxies are RTSP agents that are located in between a client and a server. A proxy can take on both the role as a client and as server depending on what it tries to accomplish. RTSP proxies use two transport layer connections, one from the RTSP client to the RTSP proxy and a second from the RTSP proxy to the RTSP server. Proxies are introduced for several different reasons and the below listed are often combined. There are these types of RTSP proxies: Caching Proxy: This type of proxy is used to reduce the workload on servers and connections. By caching the description and media streams, i.e., the presentation, the proxy can serve a client with content, but without requesting it from the server once it has been cached and has not become stale. See the caching Section 16. This type of proxy is also expected to understand RTSP end-point functionality, i.e., functionality identified in the Require header in addition to what Proxy-Require demands. Translator Proxy: This type of proxy is used to ensure that an RTSP client gets access to servers and content on an external network or using content encodings not supported by the client. The proxy performs the necessary translation of addresses, protocols or encodings. This type of proxy is expected to also understand RTSP end-point functionality, i.e., functionality identified in the Require header in addition to what Proxy- Require demands. Access Proxy: This type of proxy is used to ensure that an RTSP client gets access to servers on an external network. Thus this proxy is placed on the border between two domains, e.g., a private address space and the public Internet. The proxy performs the necessary translation, usually addresses. This type of proxy is required to redirect the media to itself or a controlled gateway that performs the translation before the media can reach the client. Security Proxy: This type of proxy is used to help facilitate security functions around RTSP. For example when having a firewalled network, the security proxy requests that the necessary pinholes in the firewall are opened when a client in the protected network wants to access media streams on the external side. This proxy can also limit the client's access to certain types of content. This proxy can perform its function without redirecting the media between the server and client. However, in deployments with private address spaces Schulzrinne, et al. Expires March 15, 2014 [Page 105] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 this proxy is likely to be combined with the access proxy. Anyway, the functionality of this proxy is usually closely tied into understanding all aspects of the media transport. Auditing Proxy: RTSP proxies can also provide network owners with a logging and audit point for RTSP sessions, e.g., for corporations that track their employees usage of the network. This type of proxy can perform its function without inserting itself or any other node in the media transport. This proxy type can also accept unknown methods as it doesn't interfere with the clients' requests. All types of proxies can 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 types of deployment. In those cases, the secured sessions do by-pass 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. This would impede RTSP's future development and usage. 15.1. Proxies and Protocol Extensions The existence of proxies must always be considered when developing new RTSP extensions. Most types of proxies will need to implement any new method to operate correctly in the presence of that extension. New headers can be introduced and will not be blocked by older proxies. However, it is important to consider if this header and its function is required to be understood by the proxy or can be forwarded. If the header needs to be understood, a feature-tag representing the functionality MUST be included in the Proxy-Require header. Below are guidelines for analysis if the header needs to be understood. The transport header and its parameters also shows that headers that are extensible and require correct interpretation in the proxy also require handling rules. Whether a proxy needs to understand a header is not easy to determine, as they serve a broad variety of functions. When evaluating if a header needs to be understood, one can divide the functionality into three main categories: Schulzrinne, et al. Expires March 15, 2014 [Page 106] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Media modifying: The caching and translator proxies are modifying the actual media and therefore need to understand also the request directed to the server that affects how the media is rendered. Thus, this type of proxy needs to also understand the server side functionality. Transport modifying: The access and the security proxy both need to understand how the transport is performed, either for opening pinholes or to translate the outer headers, e.g., IP and UDP. Non-modifying: The audit proxy is special in that it does not modify the messages in other ways than to insert the Via header. That makes it possible for this type to forward RTSP messages that contain different types of unknown methods, headers or header parameters. Based on the above classification, one should evaluate if the new functionality requires the Transport modifying type of proxies to understand it or not. 15.2. Multiplexing and Demultiplexing of Messages RTSP proxies may have to multiplex multiple RTSP sessions from their clients towards RTSP servers. This requires that RTSP requests from multiple clients are multiplexed onto a common connection for requests outgoing to an RTSP server and on the way back the responses are demultiplexed from the server to per client responses. On the protocol level this requires that request and response messages are handled in both ways, requiring that there is a mechanism to correlate what request/response pair exchanged between proxy and server is mapped to what client (or client request). This multiplexing of requests and demultiplexing of responses is done by using the CSeq header field. The proxy has to rewrite the CSeq in requests to the server and responses from the server and remember what CSeq is mapped to what client. The proxy also needs to ensure that the order of the message related to each client is maintained. Section 18.20 is defining the handling of how requests and responses are rewritten. Schulzrinne, et al. Expires March 15, 2014 [Page 107] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 16. Caching In HTTP, request-response pairs are cached. RTSP differs significantly in that respect. Responses are not cacheable, with the exception of the presentation description returned by DESCRIBE. (Since the responses for anything but DESCRIBE and GET_PARAMETER do not return any data, caching is not really an issue for these requests.) However, it is desirable for the continuous media data, typically delivered out-of-band with respect to RTSP, to be cached, as well as the session description. On receiving a SETUP or PLAY request, a proxy ascertains whether it has an up-to-date copy of the continuous media content and its description. It can determine whether the copy is up-to-date by issuing a SETUP or DESCRIBE request, respectively, and comparing the Last-Modified header with that of the cached copy. If the copy is not up-to-date, it modifies the SETUP transport parameters as appropriate and forwards the request to the origin server. Subsequent control commands such as PLAY or PAUSE then pass the proxy unmodified. The proxy delivers the continuous media data to the client, while possibly making a local copy for later reuse. The exact allowed behavior of the cache is given by the cache-response directives described in Section 18.11. A cache MUST answer any DESCRIBE requests if it is currently serving the stream to the requester, as it is possible that low-level details of the stream description may have changed on the origin-server. Note that an RTSP cache, is of the "cut-through" variety. Rather than retrieving the whole resource from the origin server, the cache simply copies the streaming data as it passes by on its way to the client. Thus, it does not introduce additional latency. To the client, an RTSP proxy cache appears like a regular media server. To the media origin server an RTSP proxy cache appears like a client. Just as an HTTP cache has to store the content type, content language, and so on for the objects it caches, a media cache has to store the presentation description. Typically, a cache eliminates all transport-references (e.g., multicast information) from the presentation description, since these are independent of the data delivery from the cache to the client. Information on the encodings remains the same. If the cache is able to translate the cached media data, it would create a new presentation description with all the encoding possibilities it can offer. 16.1. Validation Model When a cache has a stale entry that it would like to use as a response to a client's request, it first has to check with the origin Schulzrinne, et al. Expires March 15, 2014 [Page 108] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 server (or possibly an intermediate cache with a fresh response) to see if its cached entry is still usable. We call this "validating" the cache entry. Since we do not want to have to pay the overhead of retransmitting the full response if the cached entry is good, and we do not want to pay the overhead of an extra round trip if the cached entry is invalid, the RTSP protocol supports the use of conditional methods. The key protocol features for supporting conditional methods are those concerned with "cache validators." When an origin server generates a full response, it attaches some sort of validator to it, which is kept with the cache entry. When a client (user agent or proxy cache) makes a conditional request for a resource for which it has a cache entry, it includes the associated validator in the request. The server then checks that validator against the current validator for the requested resource, and, if they match (see Section 16.1.3), it responds with a special status code (usually, 304 (Not Modified)) and no message body. Otherwise, it returns a full response (including message body). Thus, we avoid transmitting the full response if the validator matches, and we avoid an extra round trip if it does not match. In RTSP, a conditional request looks exactly the same as a normal request for the same resource, except that it carries a special header (which includes the validator) that implicitly turns the method (usually DESCRIBE or SETUP) into a conditional. The protocol includes both positive and negative senses of cache- validating conditions. That is, it is possible to request either that a method be performed if and only if a validator matches or if and only if no validators match. Note: a response that lacks a validator may still be cached, and served from cache until it expires, unless this is explicitly prohibited by a cache-control directive (see Section 18.11). However, a cache cannot do a conditional retrieval if it does not have a validator for the resource, which means it will not be refreshable after it expires. Media streams that are being adapted based on the transport capacity between the server and the cache makes caching more difficult. A server needs to consider how it views caching of media streams that it adapts and potentially instruct any caches to not cache such streams. Schulzrinne, et al. Expires March 15, 2014 [Page 109] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 16.1.1. Last-Modified Dates The Last-Modified header (Section 18.27) value is often used as a cache validator. In simple terms, a cache entry is considered to be valid if the content has not been modified since the Last-Modified value. 16.1.2. Message Body Tag Cache Validators The MTag response-header field value, a message body tag, provides for an "opaque" cache validator. This might allow more reliable validation in situations where it is inconvenient to store modification dates, where the one-second resolution of RTSP-date values is not sufficient, or where the origin server wishes to avoid certain paradoxes that might arise from the use of modification dates. Message body tags are described in Section 4.6 16.1.3. Weak and Strong Validators Since both origin servers and caches will compare two validators to decide if they represent the same or different entities, one normally would expect that if the message body (i.e., the presentation description) or any associated message body headers changes in any way, then the associated validator would change as well. If this is true, then we call this validator a "strong validator." We call message body (i.e., the presentation description) or any associated message body headers an entity for a better understanding. However, there might be cases when a server prefers to change the validator only on semantically significant changes, and not when insignificant aspects of the entity change. A validator that does not always change when the resource changes is a "weak validator." Message body tags are normally "strong validators," but the protocol provides a mechanism to tag a message body tag as "weak." One can think of a strong validator as one that changes whenever the bits of an entity changes, while a weak value changes whenever the meaning of an entity changes. Alternatively, one can think of a strong validator as part of an identifier for a specific entity, while a weak validator is part of an identifier for a set of semantically equivalent entities. Schulzrinne, et al. Expires March 15, 2014 [Page 110] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Note: One example of a strong validator is an integer that is incremented in stable storage every time an entity is changed. An entity's modification time, if represented with one-second resolution, could be a weak validator, since it is possible that the resource might be modified twice during a single second. Support for weak validators is optional. However, weak validators allow for more efficient caching of equivalent objects. A "use" of a validator is either when a client generates a request and includes the validator in a validating header field, or when a server compares two validators. Strong validators are usable in any context. Weak validators are only usable in contexts that do not depend on exact equality of an entity. For example, either kind is usable for a conditional DESCRIBE of a full entity. However, only a strong validator is usable for a sub-range retrieval, since otherwise the client might end up with an internally inconsistent entity. Clients MAY issue DESCRIBE requests with either weak validators or strong validators. Clients MUST NOT use weak validators in other forms of requests. The only function that the RTSP protocol defines on validators is comparison. There are two validator comparison functions, depending on whether the comparison context allows the use of weak validators or not: o The strong comparison function: in order to be considered equal, both validators MUST be identical in every way, and both MUST NOT be weak. o The weak comparison function: in order to be considered equal, both validators MUST be identical in every way, but either or both of them MAY be tagged as "weak" without affecting the result. A message body tag is strong unless it is explicitly tagged as weak. A Last-Modified time, when used as a validator in a request, is implicitly weak unless it is possible to deduce that it is strong, using the following rules: o The validator is being compared by an origin server to the actual current validator for the entity and, Schulzrinne, et al. Expires March 15, 2014 [Page 111] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o That origin server reliably knows that the associated entity did not change more than once during the second covered by the presented validator. OR o The validator is about to be used by a client in an If-Modified- Since, because the client has a cache entry for the associated entity, and o That cache entry includes a Date value, which gives the time when the origin server sent the original response, and o The presented Last-Modified time is at least 60 seconds before the Date value. OR o The validator is being compared by an intermediate cache to the validator stored in its cache entry for the entity, and o That cache entry includes a Date value, which gives the time when the origin server sent the original response, and o The presented Last-Modified time is at least 60 seconds before the Date value. This method relies on the fact that if two different responses were sent by the origin server during the same second, but both had the same Last-Modified time, then at least one of those responses would have a Date value equal to its Last-Modified time. The arbitrary 60- second limit guards against the possibility that the Date and Last- Modified values are generated from different clocks, or at somewhat different times during the preparation of the response. An implementation MAY use a value larger than 60 seconds, if it is believed that 60 seconds is too short. If a client wishes to perform a sub-range retrieval on a value for which it has only a Last-Modified time and no opaque validator, it MAY do this only if the Last-Modified time is strong in the sense described here. 16.1.4. Rules for When to Use Message Body Tags and Last-Modified Dates We adopt a set of rules and recommendations for origin servers, clients, and caches regarding when various validator types ought to be used, and for what purposes. Schulzrinne, et al. Expires March 15, 2014 [Page 112] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTSP origin servers: o SHOULD send a message body tag validator unless it is not feasible to generate one. o MAY send a weak message body tag instead of a strong message body tag, if performance considerations support the use of weak message body tags, or if it is unfeasible to send a strong message body tag. o SHOULD send a Last-Modified value if it is feasible to send one, unless the risk of a breakdown in semantic transparency that could result from using this date in an If-Modified-Since header would lead to serious problems. In other words, the preferred behavior for an RTSP origin server is to send both a strong message body tag and a Last-Modified value. In order to be legal, a strong message body tag MUST change whenever the associated entity value changes in any way. A weak message body tag SHOULD change whenever the associated entity changes in a semantically significant way. Note: in order to provide semantically transparent caching, an origin server MUST avoid reusing a specific strong message body tag value for two different entities, or reusing a specific weak message body tag value for two semantically different entities. Cache entries might persist for arbitrarily long periods, regardless of expiration times, so it might be inappropriate to expect that a cache will never again attempt to validate an entry using a validator that it obtained at some point in the past. RTSP clients: o If a message body tag has been provided by the origin server, MUST use that message body tag in any cache-conditional request (using If-Match or If-None-Match). o If only a Last-Modified value has been provided by the origin server, SHOULD use that value in non-subrange cache-conditional requests (using If-Modified-Since). o If both a message body tag and a Last-Modified value have been provided by the origin server, SHOULD use both validators in cache-conditional requests. An RTSP origin server, upon receiving a conditional request that includes both a Last-Modified date (e.g., in an If-Modified-Since Schulzrinne, et al. Expires March 15, 2014 [Page 113] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 header) and one or more message body tags (e.g., in an If-Match, If- None-Match, or If-Range header field) as cache validators, MUST NOT return a response status of 304 (Not Modified) unless doing so is consistent with all of the conditional header fields in the request. Note: The general principle behind these rules is that RTSP servers and clients should transmit as much non-redundant information as is available in their responses and requests. RTSP systems receiving this information will make the most conservative assumptions about the validators they receive. 16.1.5. Non-validating Conditionals The principle behind message body tags is that only the service author knows the semantics of a resource well enough to select an appropriate cache validation mechanism, and the specification of any validator comparison function more complex than octet-equality would open up a can of worms. Thus, comparisons of any other headers are never used for purposes of validating a cache entry. 16.2. Invalidation After Updates or Deletions The effect of certain methods performed on a resource at the origin server might cause one or more existing cache entries to become non- transparently invalid. That is, although they might continue to be "fresh," they do not accurately reflect what the origin server would return for a new request on that resource. There is no way for the RTSP protocol to guarantee that all such cache entries are marked invalid. For example, the request that caused the change at the origin server might not have gone through the proxy where a cache entry is stored. However, several rules help reduce the likelihood of erroneous behavior. In this section, the phrase "invalidate an entity" means that the cache will either remove all instances of that entity from its storage, or will mark these as "invalid" and in need of a mandatory revalidation before they can be returned in response to a subsequent request. Some RTSP methods MUST cause a cache to invalidate an entity. This is either the entity referred to by the Request-URI, or by the Location or Content-Location headers (if present). These methods are: o DESCRIBE Schulzrinne, et al. Expires March 15, 2014 [Page 114] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o SETUP In order to prevent denial of service attacks, an invalidation based on the URI in a Location or Content-Location header MUST only be performed if the host part is the same as in the Request-URI. A cache that passes through requests for methods it does not understand SHOULD invalidate any entities referred to by the Request- URI. Schulzrinne, et al. Expires March 15, 2014 [Page 115] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 17. Status Code Definitions Where applicable, HTTP status [H10] codes are reused. See Table 4 in Section 8.1 for a listing of which status codes may be returned by which requests. All error messages, 4xx and 5xx MAY return a body containing further information about the error. 17.1. Success 1xx 17.1.1. 100 Continue The client SHOULD continue with its request. This interim response is used to inform the client that the initial part of the request has been received and has not yet been rejected by the server. The client SHOULD continue by sending the remainder of the request or, if the request has already been completed, ignore this response. The server MUST send a final response after the request has been completed. 17.2. Success 2xx This class of status code indicates that the client's request was successfully received, understood, and accepted. 17.2.1. 200 OK The request has succeeded. The information returned with the response is dependent on the method used in the request. 17.3. Redirection 3xx The notation "3xx" indicates response codes from 300 to 399 inclusive which are meant for redirection. The response code 304 is excluded, as it is not used for redirection and instead the "3rr" notation is used. The 304 response code appears here, rather than a 2xx response code which would have been appropriate, this as 304 has been used also in RTSP 1.0 [RFC2326]. Within RTSP, redirection may be used for load balancing or redirecting stream requests to a server topologically closer to the client. Mechanisms to determine topological proximity are beyond the scope of this specification. A 3rr code MAY be used to respond to any request. 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 Schulzrinne, et al. Expires March 15, 2014 [Page 116] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 with an established session about the need for redirecting the session. If a 3rr response is received for a request in relation to an established session, the client SHOULD send a TEARDOWN request for the session, and MAY reestablish the session using the resource indicated by the Location. If the Location header is used in a response it MUST contain an absolute URI pointing out the media resource the client is redirected to, the URI MUST NOT only contain the host name. 17.3.1. 300 This response code is not used in RTSP 2.0. For behavior to use with unknown 3rr status codes, one follows the 302 (Section 17.3.3). 17.3.2. 301 Moved Permanently The requested resource is moved permanently and resides now at the URI given by the Location header. The user client SHOULD redirect automatically to the given URI. This response MUST NOT contain a message-body. The Location header MUST be included in the response. 17.3.3. 302 Found The requested resource resides temporarily at the URI given by the Location header. 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, clock User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 302 Try Other Server CSeq: 2 Location: rtsp://s2.example.com:8001/fizzle/foo Schulzrinne, et al. Expires March 15, 2014 [Page 117] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 17.3.4. 303 See Other This status code MUST NOT be used in RTSP 2.0. However, it was allowed in RTSP 1.0 [RFC2326]. 17.3.5. 304 Not Modified If the client has performed a conditional DESCRIBE or SETUP (see Section 18.25) and the requested resource has not been modified, the server SHOULD send a 304 response. This response MUST NOT contain a message-body. The response MUST include the following header fields: o Date o MTag and/or Content-Location, if the header(s) would have been sent in a 200 response to the same request. o Expires and Cache-Control if the field-value might differ from that sent in any previous response for the same variant. This response is independent for the DESCRIBE and SETUP requests. That is, a 304 response to DESCRIBE does NOT imply that the resource content is unchanged (only the session description) and a 304 response to SETUP does NOT imply that the resource description is unchanged. The MTag and If-Match headers may be used to link the DESCRIBE and SETUP in this manner. 17.3.6. 305 Use Proxy The requested resource MUST be accessed through the proxy given by the Location field. The Location field gives the URI of the proxy. The recipient is expected to repeat this single request via the proxy. 305 responses MUST only be generated by origin servers. 17.4. Client Error 4xx 17.4.1. 400 Bad Request The request could not be understood by the server due to malformed syntax. The client SHOULD NOT repeat the request without modifications. If the request does not have a CSeq header, the server MUST NOT include a CSeq in the response. Schulzrinne, et al. Expires March 15, 2014 [Page 118] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 17.4.2. 401 Unauthorized The request requires user authentication. The response MUST include a WWW-Authenticate header (Section 18.58) field containing a challenge applicable to the requested resource. The client MAY repeat the request with a suitable Authorization header field. If the request already included Authorization credentials, then the 401 response indicates that authorization has been refused for those credentials. If the 401 response contains the same challenge as the prior response, and the user agent has already attempted authentication at least once, then the user SHOULD be presented the message body that was given in the response, since that message body might include relevant diagnostic information. HTTP access authentication is explained in [RFC2617]. 17.4.3. 402 Payment Required This code is reserved for future use. 17.4.4. 403 Forbidden The server understood the request, but is refusing to fulfill it. Authorization will not help and the request SHOULD NOT be repeated. If the server wishes to make public why the request has not been fulfilled, it SHOULD describe the reason for the refusal in the message body. If the server does not wish to make this information available to the client, the status code 404 (Not Found) can be used instead. 17.4.5. 404 Not Found The server has not found anything matching the Request-URI. No indication is given of whether the condition is temporary or permanent. The 410 (Gone) status code SHOULD be used if the server knows, through some internally configurable mechanism, that an old resource is permanently unavailable and has no forwarding address. This status code is commonly used when the server does not wish to reveal exactly why the request has been refused, or when no other response is applicable. 17.4.6. 405 Method Not Allowed The method specified in the request is not allowed for the resource identified by the Request-URI. The response MUST include an Allow header containing a list of valid methods for the requested resource. This status code is also to be used if a request attempts to use a method not indicated during SETUP. Schulzrinne, et al. Expires March 15, 2014 [Page 119] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 17.4.7. 406 Not Acceptable The resource identified by the request is only capable of generating response message bodies which have content characteristics not acceptable according to the Accept headers sent in the request. The response SHOULD include a message body containing a list of available message body characteristics and location(s) from which the user or user agent can choose the one most appropriate. The message body format is specified by the media type given in the Content-Type header field. Depending upon the format and the capabilities of the user agent, selection of the most appropriate choice MAY be performed automatically. However, this specification does not define any standard for such automatic selection. If the response could be unacceptable, a user agent SHOULD temporarily stop receipt of more data and query the user for a decision on further actions. 17.4.8. 407 Proxy Authentication Required This code is similar to 401 (Unauthorized) (Section 17.4.2), but indicates that the client must first authenticate itself with the proxy. The proxy MUST return a Proxy-Authenticate header field (Section 18.34) containing a challenge applicable to the proxy for the requested resource. 17.4.9. 408 Request Timeout The client did not produce a request within the time that the server was prepared to wait. The client MAY repeat the request without modifications at any later time. 17.4.10. 410 Gone The requested resource is no longer available at the server and the forwarding address is not known. This condition is expected to be considered permanent. If the server does not know, or has no facility to determine, whether or not the condition is permanent, the status code 404 (Not Found) SHOULD be used instead. This response is cacheable unless indicated otherwise. The 410 response is primarily intended to assist the task of repository maintenance by notifying the recipient that the resource is intentionally unavailable and that the server owners desire that remote links to that resource be removed. Such an event is common for limited-time, promotional services and for resources belonging to individuals no longer working at the server's site. It is not Schulzrinne, et al. Expires March 15, 2014 [Page 120] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 necessary to mark all permanently unavailable resources as "gone" or to keep the mark for any length of time -- that is left to the discretion of the owner of the server. 17.4.11. 412 Precondition Failed The precondition given in one or more of the 'if-' request-header fields evaluated to false when it was tested on the server. See these sections for the 'if-' headers: If-Match Section 18.24, If- Modified-Since Section 18.25, and If-None-Match Section 18.26. This response code allows the client to place preconditions on the current resource meta information (header field data) and thus prevent the requested method from being applied to a resource other than the one intended. 17.4.12. 413 Request Message Body Too Large The server is refusing to process a request because the request message body is larger than the server is willing or able to process. The server MAY close the connection to prevent the client from continuing the request. If the condition is temporary, the server SHOULD include a Retry- After header field to indicate that it is temporary and after what time the client MAY try again. 17.4.13. 414 Request-URI Too Long The server is refusing to service the request because the Request-URI is longer than the server is willing to interpret. This rare condition is only likely to occur when a client has used a request with long query information, when the client has descended into a URI "black hole" of redirection (e.g., a redirected URI prefix that points to a suffix of itself), or when the server is under attack by a client attempting to exploit security holes present in some servers using fixed-length buffers for reading or manipulating the Request- URI. 17.4.14. 415 Unsupported Media Type The server is refusing to service the request because the message body of the request is in a format not supported by the requested resource for the requested method. 17.4.15. 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 Schulzrinne, et al. Expires March 15, 2014 [Page 121] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 sender SHOULD return a message body containing the offending parameter(s). 17.4.16. 452 reserved This status code MUST NOT be used in RTSP 2.0. However, it was allowed in RTSP 1.0 [RFC2326]. 17.4.17. 453 Not Enough Bandwidth The request was refused because there was insufficient bandwidth. This may, for example, be the result of a resource reservation failure. 17.4.18. 454 Session Not Found The RTSP session identifier in the Session header is missing, invalid, or has timed out. 17.4.19. 455 Method Not Valid in This State The client or server cannot process this request in its current state. The response MUST contain an Allow header to make error recovery possible. 17.4.20. 456 Header Field Not Valid for Resource The server could not act on a required request header. For example, if PLAY contains the Range header field but the stream does not allow seeking. This error message may also be used for specifying when the time format in Range is impossible for the resource. In that case the Accept-Ranges header MUST be returned to inform the client of which format(s) that are allowed. 17.4.21. 457 Invalid Range The Range value given is out of bounds, e.g., beyond the end of the presentation. 17.4.22. 458 Parameter Is Read-Only The parameter to be set by SET_PARAMETER can be read but not modified. When returning this error message the sender SHOULD return a message body containing the offending parameter(s). Schulzrinne, et al. Expires March 15, 2014 [Page 122] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 17.4.23. 459 Aggregate Operation Not Allowed The requested method may not be applied on the URI in question since it is an aggregate (presentation) URI. The method may be applied on a media URI. 17.4.24. 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. 17.4.25. 461 Unsupported Transport The Transport field did not contain a supported transport specification. 17.4.26. 462 Destination Unreachable The data transmission channel could not be established because the client address could not be reached. This error will most likely be the result of a client attempt to place an invalid dest_addr parameter in the Transport field. 17.4.27. 463 Destination Prohibited The data transmission channel was not established because the server prohibited access to the client address. This error is most likely the result of a client attempt to redirect media traffic to another destination with a dest_addr parameter in the Transport header. 17.4.28. 464 Data Transport Not Ready Yet The data transmission channel to the media destination is not yet ready for carrying data. However, the responding agent still expects that the data transmission channel will be established at some point in time. Note, however, that this may result in a permanent failure like 462 "Destination Unreachable". An example when this error may occur is in the case a client sends a PLAY request to a server prior to ensuring that the TCP connections negotiated for carrying media data was successfully established (In violation of this specification). The server would use this error code to indicate that the requested action could not be performed due to the failure of completing the connection establishment. Schulzrinne, et al. Expires March 15, 2014 [Page 123] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 17.4.29. 465 Notification Reason Unknown This indicates that the client has received a PLAY_NOTIFY (Section 13.5) with a Notify-Reason header (Section 18.32) unknown to the client. 17.4.30. 466 Key Management Error This indicates that there has been an error in a Key Management function used in conjunction with a request. For example usage of MIKEY [RFC3830] according to Appendix C.1.4.1 may result in this error. 17.4.31. 470 Connection Authorization Required The secured connection attempt needs user or client authorization before proceeding. The next hop's certificate is included in this response in the Accept-Credentials header. 17.4.32. 471 Connection Credentials not accepted When performing a secure connection over multiple connections, an intermediary has refused to connect to the next hop and carry out the request due to unacceptable credentials for the used policy. 17.4.33. 472 Failure to establish secure connection A proxy fails to establish a secure connection to the next hop RTSP agent. This is primarily caused by a fatal failure at the TLS handshake, for example due to server not accepting any cipher suites. 17.5. Server Error 5xx Response status codes beginning with the digit "5" indicate cases in which the server is aware that it has erred or is incapable of performing the request The server SHOULD include a message body containing an explanation of the error situation, and whether it is a temporary or permanent condition. User agents SHOULD display any included message body to the user. These response codes are applicable to any request method. 17.5.1. 500 Internal Server Error The server encountered an unexpected condition which prevented it from fulfilling the request. Schulzrinne, et al. Expires March 15, 2014 [Page 124] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 17.5.2. 501 Not Implemented The server does not support the functionality required to fulfill the request. This is the appropriate response when the server does not recognize the request method and is not capable of supporting it for any resource. 17.5.3. 502 Bad Gateway The server, while acting as a gateway or proxy, received an invalid response from the upstream server it accessed in attempting to fulfill the request. 17.5.4. 503 Service Unavailable The server is currently unable to handle the request due to a temporary overloading or maintenance of the server. The implication is that this is a temporary condition which will be alleviated after some delay. If known, the length of the delay MAY be indicated in a Retry-After header. If no Retry-After is given, the client SHOULD handle the response as it would for a 500 response. The client MUST honor the length, if given in the Retry-After header. Note: The existence of the 503 status code does not imply that a server must use it when becoming overloaded. Some servers may wish to simply refuse the connection. The response scope is dependent on the Request. If the request was in relation to an existing RTSP session, the scope of the overload response is to this individual RTSP session. If the request was non- session specific or intended to form a RTSP session it applies to the RTSP server identified by the host name in the request URI. 17.5.5. 504 Gateway Timeout The server, while acting as a proxy, did not receive a timely response from the upstream server specified by the URI or some other auxiliary server (e.g., DNS) it needed to access in attempting to complete the request. 17.5.6. 505 RTSP Version Not Supported The server does not support, or refuses to support, the RTSP protocol version that was used in the request message. The server is indicating that it is unable or unwilling to complete the request using the same major version as the client other than with this error message. The response SHOULD contain a message body describing why that version is not supported and what other protocols are supported Schulzrinne, et al. Expires March 15, 2014 [Page 125] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 by that server. 17.5.7. 551 Option not supported A feature-tag given in the Require or the Proxy-Require fields was not supported. The Unsupported header MUST be returned stating the feature for which there is no support. 17.5.8. 553 Proxy Unavailable The proxy is currently unable to handle the request due to a temporary overloading or maintenance of the proxy. The implication is that this is a temporary condition which will be alleviated after some delay. If known, the length of the delay MAY be indicated in a Retry-After header. If no Retry-After is given, the client SHOULD handle the response as it would for a 500 response. The client MUST honor the length, if given in the Retry-After header. Note: The existence of the 553 status code does not imply that a proxy must use it when becoming overloaded. Some proxies may wish to simply refuse the connection. The response scope is dependent on the Request. If the request was in relation to an existing RTSP session, the scope of the overload response is to this individual RTSP session. If the request was non- session specific or intended to form a RTSP session it applies to all such requests to this proxy. Schulzrinne, et al. Expires March 15, 2014 [Page 126] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18. Header Field Definitions +---------------+----------------+--------+---------+------+ | method | direction | object | acronym | Body | +---------------+----------------+--------+---------+------+ | DESCRIBE | C -> S | P,S | DES | r | | | | | | | | GET_PARAMETER | C -> S, S -> C | P,S | GPR | R,r | | | | | | | | OPTIONS | C -> S, S -> C | P,S | OPT | | | | | | | | | PAUSE | C -> S | P,S | PSE | | | | | | | | | PLAY | C -> S | P,S | PLY | | | | | | | | | PLAY_NOTIFY | S -> C | P,S | PNY | R | | | | | | | | REDIRECT | S -> C | P,S | RDR | | | | | | | | | SETUP | C -> S | S | STP | | | | | | | | | SET_PARAMETER | C -> S, S -> C | P,S | SPR | R,r | | | | | | | | TEARDOWN | C -> S | P,S | TRD | | | | | | | | | | S -> C | P | TRD | | +---------------+----------------+--------+---------+------+ Table 8: Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on. Body notes if a method is allowed to carry body and in which direction, R = Request, r=response. Note: All error messages for statuses 4xx and 5xx are allowed to carry a body 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 reference 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 March 15, 2014 [Page 127] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 R: header field may only appear in requests; r: header field may only appear in responses; 2xx, 4xx, etc.: A numerical value or range indicates response codes with which the header field can be used; c: header field is copied from the request to the response. An empty entry in the "where" column indicates that the header field may be present in both requests and responses. The "proxy" column describes the operations a proxy may perform on a header field. An empty proxy column indicates that the proxy MUST NOT do any changes to that header, all allowed operations are explicitly stated: a: A proxy can add or concatenate the header field if not present. m: A proxy can modify an existing header field value. d: A proxy can delete a header field value. r: A proxy needs to be able to read the header field, and thus this header field cannot be encrypted. The rest of the columns relate to the presence of a header field in a method. The method names when abbreviated, are according to Table 8: c: Conditional; requirements on the header field depend on the context of the message. m: The header field is mandatory. m*: The header field SHOULD be sent, but clients/servers need to be prepared to receive messages without that header field. o: The header field is optional. *: The header field MUST be present if the message body is not empty. See Section 18.17, Section 18.19 and Section 5.3 for details. -: The header field is not applicable. "Optional" means that a Client/Server MAY include the header field in a request or response. The Client/Server behavior when receiving such headers varies, for some it may ignore the header field, in Schulzrinne, et al. Expires March 15, 2014 [Page 128] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 other cases it is a request to process the header. This is regulated by the method and header descriptions. Example of headers that require processing are the Require and Proxy-Require header fields discussed in Section 18.43 and Section 18.37. A "mandatory" header field MUST be present in a request, and MUST be understood by the Client/Server receiving the request. A mandatory response header field MUST be present in the response, and the header field MUST be understood by the Client/Server processing the response. "Not applicable" means that the header field MUST NOT be present in a request. If one is placed in a request by mistake, it MUST be ignored by the Client/Server receiving the request. Similarly, a header field labeled "not applicable" for a response means that the Client/Server MUST NOT place the header field in the response, and the Client/Server MUST ignore the header field in the response. An RTSP agent MUST ignore extension headers that are not understood. The From and Location header fields contain an URI. If the URI contains a comma, or semicolon, the URI MUST be enclosed in double quotes ("). Any URI parameters are contained within these quotes. If the URI is not enclosed in double quote, any semicolon-delimited parameters are header-parameters, not URI parameters. +------------------+-------+-----+----+-----+-----+-----+-----+-----+ | Header | Where | Pro | DE | OPT | STP | PLY | PSE | TRD | | | | xy | S | | | | | | +------------------+-------+-----+----+-----+-----+-----+-----+-----+ | Accept | R | | o | - | - | - | - | - | | | | | | | | | | | | Accept-Credentia | R | rm | o | o | o | o | o | o | | ls | | | | | | | | | | | | | | | | | | | | Accept-Encoding | R | r | o | - | - | - | - | - | | | | | | | | | | | | Accept-Language | R | r | o | - | - | - | - | - | | | | | | | | | | | | Accept-Ranges | R | r | - | - | m | - | - | - | | | | | | | | | | | | Accept-Ranges | r | r | - | - | m | - | - | - | | | | | | | | | | | | Accept-Ranges | 456 | r | - | - | - | m | - | - | | | | | | | | | | | | Allow | r | am | c | c | c | - | - | - | | | | | | | | | | | | Allow | 405 | am | m | m | m | m | m | m | | | | | | | | | | | | Authentication-I | r | | o | o | o | o | o | o/- | | nfo | | | | | | | | | Schulzrinne, et al. Expires March 15, 2014 [Page 129] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | 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 | | ad | o | o | o | o | o | o | | | | | | | | | | | | Connection-Crede | 470,4 | ar | o | o | o | o | o | o | | ntials | 07 | | | | | | | | | | | | | | | | | | | Content-Base | r | | o | - | - | - | - | - | | | | | | | | | | | | Content-Base | 4xx,5 | | o | o | o | o | o | o | | | xx | | | | | | | | | | | | | | | | | | | Content-Encoding | R | r | - | - | - | - | - | - | | | | | | | | | | | | Content-Encoding | r | r | o | - | - | - | - | - | | | | | | | | | | | | Content-Encoding | 4xx,5 | r | o | o | o | o | o | o | | | xx | | | | | | | | | | | | | | | | | | | Content-Language | R | r | - | - | - | - | - | - | | | | | | | | | | | | Content-Language | r | r | o | - | - | - | - | - | | | | | | | | | | | | Content-Language | 4xx,5 | r | o | o | o | o | o | o | | | xx | | | | | | | | | | | | | | | | | | | Content-Length | r | r | * | - | - | - | - | - | | | | | | | | | | | | Content-Length | 4xx,5 | r | * | * | * | * | * | * | | | xx | | | | | | | | | | | | | | | | | | | Content-Location | r | r | o | - | - | - | - | - | | | | | | | | | | | | Content-Location | 4xx,5 | r | o | o | o | o | o | o | | | xx | | | | | | | | | | | | | | | | | | | Content-Type | r | r | * | - | - | - | - | - | | | | | | | | | | | | Content-Type | 4xx,5 | ar | * | * | * | * | * | * | | | xx | | | | | | | | | | | | | | | | | | | CSeq | Rc | rm | m | m | m | m | m | m | Schulzrinne, et al. Expires March 15, 2014 [Page 130] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | Date | | am | o/ | o/* | o/* | o/* | o/* | o/* | | | | | * | | | | | | | | | | | | | | | | | Expires | r | r | o | - | o | - | - | - | | | | | | | | | | | | From | R | r | o | o | o | o | o | o | | | | | | | | | | | | If-Match | R | r | - | - | o | - | - | - | | | | | | | | | | | | If-Modified-Sinc | R | r | o | - | o | - | - | - | | e | | | | | | | | | | | | | | | | | | | | If-None-Match | R | r | o | - | o | - | - | - | | | | | | | | | | | | Last-Modified | r | r | o | - | o | - | - | - | | | | | | | | | | | | Location | 3rr | | o | o | o | o | o | o | +------------------+-------+-----+----+-----+-----+-----+-----+-----+ Table 9: Overview of RTSP header fields (A-L) related to methods DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN. +------------------+---------+-----+----+----+----+-----+-----+-----+ | Header | Where | Pro | DE | OP | ST | PLY | PSE | TRD | | | | xy | S | T | P | | | | +------------------+---------+-----+----+----+----+-----+-----+-----+ | Media- | | | - | - | m | m | m | - | | Properties | | | | | | | | | | | | | | | | | | | | Media-Range | | | - | - | m | m | m | - | | | | | | | | | | | | MTag | r | r | o | - | o | - | - | - | | | | | | | | | | | | Pipelined-Reques | | amd | - | o | o | o | o | o | | ts | | r | | | | | | | | | | | | | | | | | | Proxy- | 407 | amr | m | m | m | m | m | m | | Authenticate | | | | | | | | | | | | | | | | | | | | Proxy-Authentica | r | amd | o | o | o | o | o | o/- | | tion-Info | | r | | | | | | | | | | | | | | | | | | Proxy- | R | rd | o | o | o | o | o | o | | Authorization | | | | | | | | | | | | | | | | | | | | Proxy- Require | R | ar | o | o | o | o | o | o | | | | | | | | | | | | Proxy- Require | r | r | c | c | c | c | c | c | Schulzrinne, et al. Expires March 15, 2014 [Page 131] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | Proxy- Supported | R | amr | c | c | c | c | c | c | | | | | | | | | | | | Proxy- Supported | r | | c | c | c | c | c | c | | | | | | | | | | | | Public | r | amr | - | m | - | - | - | - | | | | | | | | | | | | Public | 501 | amr | m | m | m | m | m | m | | | | | | | | | | | | Range | R | | - | - | - | o | - | - | | | | | | | | | | | | Range | r | | - | - | c | m | m | - | | | | | | | | | | | | Referrer | R | | o | o | o | o | o | o | | | | | | | | | | | | Request- Status | R | | - | - | - | - | - | - | | | | | | | | | | | | Require | R | | o | o | o | o | o | o | | | | | | | | | | | | Retry-After | 3rr,503 | | o | o | o | o | o | - | | | ,553 | | | | | | | | | | | | | | | | | | | Retry-After | 413 | | o | - | - | - | - | - | | | | | | | | | | | | RTP-Info | r | | - | - | c | c | - | - | | | | | | | | | | | | Scale | R | r | - | - | - | o | - | - | | | | | | | | | | | | Scale | r | amr | - | - | - | c | - | - | | | | | | | | | | | | Seek-Style | R | | - | - | - | o | - | - | | | | | | | | | | | | Seek-Style | r | | - | - | - | m | - | - | | | | | | | | | | | | Server | R | r | - | o | - | - | - | o | | | | | | | | | | | | Server | r | r | o | o | o | o | o | o | | | | | | | | | | | | Session | R | r | - | o | o | m | m | m | | | | | | | | | | | | Session | r | r | - | c | m | m | m | o | | | | | | | | | | | | Speed | R | adm | - | - | - | o | - | - | | | | r | | | | | | | | | | | | | | | | | | Speed | r | adm | - | - | - | c | - | - | | | | r | | | | | | | | | | | | | | | | | | Supported | R | amr | o | o | o | o | o | o | Schulzrinne, et al. Expires March 15, 2014 [Page 132] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | Supported | r | amr | c | c | c | c | c | c | | | | | | | | | | | | Terminate-Reason | R | r | - | - | - | - | - | - | | | | | | | | | | | | Timestamp | R | adm | o | o | o | o | o | o | | | | r | | | | | | | | | | | | | | | | | | Timestamp | c | adm | m | m | m | m | m | m | | | | r | | | | | | | | | | | | | | | | | | Transport | | mr | - | - | m | - | - | - | | | | | | | | | | | | Unsupported | r | | c | c | c | c | c | c | | | | | | | | | | | | User-Agent | R | | m* | m* | m* | m* | m* | m* | | | | | | | | | | | | Via | R | amr | o | o | o | o | o | o | | | | | | | | | | | | Via | c | dr | m | m | m | m | m | m | | | | | | | | | | | | WWW- | 401 | | m | m | m | m | m | m | | Authenticate | | | | | | | | | +------------------+---------+-----+----+----+----+-----+-----+-----+ Table 10: Overview of RTSP header fields (M-W) related to methods DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN. +-------------------------+---------+-------+-----+-----+-----+-----+ | Header | Where | Proxy | GPR | SPR | RDR | PNY | +-------------------------+---------+-------+-----+-----+-----+-----+ | Accept | R | arm | o | o | - | - | | | | | | | | | | Accept-Credentials | R | rm | o | o | o | - | | | | | | | | | | Accept-Encoding | R | r | o | o | o | - | | | | | | | | | | Accept-Language | R | r | o | o | o | - | | | | | | | | | | Accept-Ranges | | rm | o | - | - | - | | | | | | | | | | Allow | 405 | amr | m | m | m | - | | | | | | | | | | Authentication-Info | r | | o/- | o/- | - | - | | | | | | | | | | Authorization | R | | o | o | o | - | | | | | | | | | | Bandwidth | R | | - | o | - | - | | | | | | | | | Schulzrinne, et al. Expires March 15, 2014 [Page 133] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | Blocksize | R | | - | o | - | - | | | | | | | | | | Cache-Control | | r | o | o | - | - | | | | | | | | | | Connection | | | o | 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 | o | | | | | | | | | | Content-Encoding | R | r | o | o | - | - | | | | | | | | | | Content-Encoding | r | r | o | o | - | - | | | | | | | | | | Content-Encoding | 4xx,5xx | r | o | o | o | o | | | | | | | | | | Content-Language | R | r | o | o | - | - | | | | | | | | | | Content-Language | r | r | o | o | - | - | | | | | | | | | | Content-Language | 4xx,5xx | r | o | 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 | o | | | | | | | | | | Content-Type | R | | * | * | - | - | | | | | | | | | | Content-Type | r | | * | * | - | - | | | | | | | | | | Content-Type | 4xx,5xx | | * | * | * | * | | | | | | | | | | CSeq | R,c | mr | m | m | m | m | | | | | | | | | | Date | R | a | o | o | m | o | | | | | | | | | Schulzrinne, et al. Expires March 15, 2014 [Page 134] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | Date | r | am | o | o | o | o | | | | | | | | | | Expires | r | r | - | - | - | - | | | | | | | | | | From | R | r | o | o | o | - | | | | | | | | | | If-Match | R | r | - | - | - | - | | | | | | | | | | If-Modified-Since | R | am | o | - | - | - | | | | | | | | | | If-None-Match | R | am | o | - | - | - | | | | | | | | | | Last-Modified | R | r | - | - | - | - | | | | | | | | | | Last-Modified | r | r | o | - | - | - | | | | | | | | | | Location | 3rr | | o | o | o | - | | | | | | | | | | Location | R | | - | - | m | - | | | | | | | | | | Media-Properties | R | amr | o | - | - | c | | | | | | | | | | Media-Properties | r | mr | c | - | - | - | | | | | | | | | | Media-Range | R | | o | - | - | c | | | | | | | | | | Media-Range | r | | c | - | - | - | | | | | | | | | | MTag | r | r | o | - | - | - | | | | | | | | | | Notify-Reason | R | | - | - | - | m | | | | | | | | | | Pipelined-Requests | R | amdr | o | o | - | - | | | | | | | | | | Proxy-Authenticate | 407 | amdr | m | m | m | - | | | | | | | | | | Proxy-Authentication-In | r | amdr | o/- | o/- | - | - | | fo | | | | | | | | | | | | | | | | Proxy-Authorization | R | amdr | o | o | o | - | | | | | | | | | | Proxy-Require | R | ar | o | o | o | - | | | | | | | | | | Proxy-Supported | R | amr | c | c | c | - | | | | | | | | | | Proxy-Supported | r | | c | c | c | - | | | | | | | | | Schulzrinne, et al. Expires March 15, 2014 [Page 135] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | Public | 501 | admr | m | m | m | - | +-------------------------+---------+-------+-----+-----+-----+-----+ Table 11: Overview of RTSP header fields (A-P) related to methods GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY. +------------------+---------+-------+-----+-----+-----+-----+ | Header | Where | Proxy | GPR | SPR | RDR | PNY | +------------------+---------+-------+-----+-----+-----+-----+ | Range | R | | o | - | o | m | | | | | | | | | | Referrer | R | | o | o | o | - | | | | | | | | | | Request-Status | R | | - | - | - | c | | | | | | | | | | Require | R | r | o | o | o | - | | | | | | | | | | Retry-After | 3rr,503 | | o | o | - | - | | | | | | | | | | Retry-After | 413 | | o | o | - | - | | | | | | | | | | RTP-Info | R | r | o | - | - | C | | | | | | | | | | RTP-Info | r | r | c | - | - | - | | | | | | | | | | Scale | | | - | - | - | c | | | | | | | | | | Seek-Style | | | - | - | - | - | | | | | | | | | | Server | R | r | o | o | o | o | | | | | | | | | | Server | r | r | o | o | - | - | | | | | | | | | | Session | R | r | o | o | o | m | | | | | | | | | | Session | r | r | c | c | o | m | | | | | | | | | | Speed | | | - | - | - | - | | | | | | | | | | Supported | R | adrm | o | o | o | - | | | | | | | | | | Supported | r | adrm | c | c | c | - | | | | | | | | | | Terminate-Reason | R | r | - | - | m | - | | | | | | | | | | Timestamp | R | adrm | o | o | o | - | | | | | | | | | | Timestamp | c | adrm | m | m | m | - | Schulzrinne, et al. Expires March 15, 2014 [Page 136] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 | Transport | | mr | - | - | - | - | | | | | | | | | | Unsupported | r | arm | c | c | c | - | | | | | | | | | | User-Agent | R | r | m* | m* | - | - | | | | | | | | | | User-Agent | r | r | m* | m* | m* | m* | | | | | | | | | | Via | R | amr | o | o | o | - | | | | | | | | | | Via | c | dr | m | m | m | - | | | | | | | | | | WWW-Authenticate | 401 | | m | m | m | - | +------------------+---------+-------+-----+-----+-----+-----+ Table 12: Overview of RTSP header fields (R-W) related to methods GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY. 18.1. Accept The Accept request-header field can be used to specify certain presentation description and parameter media types [RFC6838] which are acceptable for the response to DESCRIBE and GET_PARAMETER requests. See Section 20.2.3 for the syntax. The asterisk "*" character is used to group media types into ranges, with "*/*" indicating all media types and "type/*" indicating all subtypes of that type. The media-range MAY include media type parameters that are applicable to that range. Each media-range MAY be followed by one or more accept-params, beginning with the "q" parameter for indicating a relative quality factor. The first "q" parameter (if any) separates the media-range parameter(s) from the accept-params. Quality factors allow the user or user agent to indicate the relative degree of preference for that media-range, using the qvalue scale from 0 to 1 (section 3.9). The default value is q=1. Example of use: Accept: application/example ;q=0.7, application/sdp Indicates that the requesting agent prefers the media type application/sdp through the default 1.0 rating but also accepts the application/example media type with a 0.7 quality rating. Schulzrinne, et al. Expires March 15, 2014 [Page 137] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 If no Accept header field is present, then it is assumed that the client accepts all media types. If an Accept header field is present, and if the server cannot send a response which is acceptable according to the combined Accept field value, then the server SHOULD send a 406 (not acceptable) response. 18.2. Accept-Credentials The Accept-Credentials header is a request header used to indicate to any trusted intermediary how to handle further secured connections to proxies or servers. See Section 19 for the usage of this header. It MUST NOT be included in server to client requests. In a request the header MUST contain the method (User, Proxy, or Any) for approving credentials selected by the requester. The method MUST NOT be changed by any proxy, unless it is "Proxy" when a proxy MAY change it to "user" to take the role of user approving each further hop. If the method is "User" the header contains zero or more of credentials that the client accepts. The header may contain zero credentials in the first RTSP request to a RTSP server when using the "User" method. This is because the client has not yet received any credentials to accept. Each credential MUST consist of one URI identifying the proxy or server, the hash algorithm identifier, and the hash over that agent's ASN.1 distinguished encoding rules (DER) encoded certificate [RFC5280] in Base64 [RFC4648]. All RTSP clients and proxies MUST implement the SHA-256[FIPS-pub-180-2] algorithm for computation of the hash of the DER encoded certificate. The SHA-256 algorithm is identified by the token "sha-256". The intention with allowing for other hash algorithms is to enable the future retirement of algorithms that are not implemented somewhere else than here. Thus the definition of future algorithms for this purpose is intended to be extremely limited. A feature tag can be used to ensure that support for the replacement algorithm exists. Example: Accept-Credentials:User "rtsps://proxy2.example.com/";sha-256;exaIl9VMbQMOFGClx5rXnPJKVNI=, "rtsps://server.example.com/";sha-256;lurbjj5khhB0NhIuOXtt4bBRH1M= 18.3. Accept-Encoding The Accept-Encoding request-header field is similar to Accept, but restricts the content-codings (see Section 18.15),i.e., transformation codings of the message body, such as gzip compression, that are acceptable in the response. Schulzrinne, et al. Expires March 15, 2014 [Page 138] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 A server tests whether a content-coding is acceptable, according to an Accept-Encoding field, using these rules: 1. If the content-coding is one of the content-codings listed in the Accept-Encoding field, then it is acceptable, unless it is accompanied by a qvalue of 0. (As defined in [H3.9], a qvalue of 0 means "not acceptable.") 2. The special "*" symbol in an Accept-Encoding field matches any available content-coding not explicitly listed in the header field. 3. If multiple content-codings are acceptable, then the acceptable content-coding with the highest non-zero qvalue is preferred. 4. The "identity" content-coding is always acceptable, i.e., no transformation at all, unless specifically refused because the Accept-Encoding field includes "identity;q=0", or because the field includes "*;q=0" and does not explicitly include the "identity" content-coding. If the Accept-Encoding field-value is empty, then only the "identity" encoding is acceptable. If an Accept-Encoding field is present in a request, and if the server cannot send a response which is acceptable according to the Accept-Encoding header, then the server SHOULD send an error response with the 406 (Not Acceptable) status code. If no Accept-Encoding field is present in a request, the server MAY assume that the client will accept any content coding. In this case, if "identity" is one of the available content-codings, then the server SHOULD use the "identity" content-coding, unless it has additional information that a different content-coding is meaningful to the client. 18.4. Accept-Language The Accept-Language request-header field is similar to Accept, but restricts the set of natural languages that are preferred as a response to the request. Note that the language specified applies to the presentation description and any reason phrases, but not the media content. A language tag identifies a natural language spoken, written, or otherwise conveyed by human beings for communication of information to other human beings. Computer languages are explicitly excluded. The syntax and registry of RTSP 2.0 language tags is the same as that defined by [RFC5646]. Schulzrinne, et al. Expires March 15, 2014 [Page 139] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Each language-range MAY be given an associated quality value which represents an estimate of the user's preference for the languages specified by that range. The quality value defaults to "q=1". For example: Accept-Language: da, en-gb;q=0.8, en;q=0.7 would mean: "I prefer Danish, but will accept British English and other types of English." A language-range matches a language-tag if it exactly equals the full tag, or if it exactly equals a prefix of the tag, i.e., the primary-tag in the ABNF, such that the character following primary-tag is "-". The special range "*", if present in the Accept-Language field, matches every tag not matched by any other range present in the Accept-Language field. Note: This use of a prefix matching rule does not imply that language tags are assigned to languages in such a way that it is always true that if a user understands a language with a certain tag, then this user will also understand all languages with tags for which this tag is a prefix. The prefix rule simply allows the use of prefix tags if this is the case. In the process of selecting a language, each language-tag is assigned a qualification factor, i.e., if a language being supported by the client is actually supported by the server and what "preference" level the language achieves. The quality value (q-value) of the longest language-range in the field that matches the language-tag is assigned as the qualification factor for a particular language-tag. If no language-range in the field matches the tag, the language qualification factor assigned is 0. If no Accept-Language header is present in the request, the server SHOULD assume that all languages are equally acceptable. If an Accept-Language header is present, then all languages which are assigned a qualification factor greater than 0 are acceptable. 18.5. Accept-Ranges The Accept-Ranges general-header field allows indication of the format supported in the Range header. The client MUST include the header in SETUP requests to indicate which formats are acceptable when received in PLAY and PAUSE responses, and REDIRECT requests. The server MUST include the header in SETUP and 456 error responses to indicate the formats supported for the resource indicated by the request URI. The header MAY be included in GET_PARAMETER request and response pairs. The GET_PARAMETER request MUST contain a Session header to identify the session context the request is related to. The requester and responder will indicate their capabilities regarding Range formats respectively. Schulzrinne, et al. Expires March 15, 2014 [Page 140] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Accept-Ranges: npt, smpte, clock The syntax is defined in Section 20.2.3. 18.6. Allow The Allow message-header field lists the methods supported by the resource identified by the Request-URI. The purpose of this field is to inform the recipient of the complete set of valid methods associated with the resource. An Allow header field MUST be present in a 405 (Method Not Allowed) response. The Allow header MUST also be present in all OPTIONS responses where the content of the header will not include exactly the same methods as listed in the Public header. The Allow message-header MUST also be included in SETUP and DESCRIBE responses, if the methods allowed for the resource is different from the complete set of methods defined in this memo. Example of use: Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE 18.7. Authentication-Info The Authentication-Info header is used by the server to communicate some information regarding the successful authentication in the response message. This usage of this header is specified in [RFC2617] with some RTSP clarification in Section 19.1. This header MUST only be used in response messages related to client to server requests. 18.8. Authorization An RTSP client that wishes to authenticate itself with a server using authentication mechanism from HTTP [RFC2617] , usually, but not necessarily, after receiving a 401 response, does so by including an Authorization request-header field with the request. The Authorization field value consists of credentials containing the authentication information of the user agent for the realm of the resource being requested. This header MUST only be used in client to server requests. If a request is authenticated and a realm specified, the same credentials SHOULD be valid for all other requests within this realm (assuming that the authentication scheme itself does not require otherwise, such as credentials that vary according to a challenge value or using synchronized clocks). Schulzrinne, et al. Expires March 15, 2014 [Page 141] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 When a shared cache (see Section 16) receives a request containing an Authorization field, it MUST NOT return the corresponding response as a reply to any other request, unless one of the following specific exceptions holds: 1. If the response includes the "max-age" cache-control directive, the cache MAY use that response in replying to a subsequent request. But (if the specified maximum age has passed) a proxy cache MUST first revalidate it with the origin server, using the request-headers from the new request to allow the origin server to authenticate the new request. (This is the defined behavior for max-age.) If the response includes "max-age=0", the proxy MUST always revalidate it before re-using it. 2. If the response includes the "must-revalidate" cache-control directive, the cache MAY use that response in replying to a subsequent request. But if the response is stale, all caches MUST first revalidate it with the origin server, using the request-headers from the new request to allow the origin server to authenticate the new request. 3. If the response includes the "public" cache-control directive, it MAY be returned in reply to any subsequent request. 18.9. Bandwidth The Bandwidth request-header field describes the estimated bandwidth available to the client, expressed as a positive integer and measured in kilobits per second. The bandwidth available to the client may change during an RTSP session, e.g., due to mobility, congestion, etc. Clients may not be able to accurately determine the available bandwidth, for example because first hop is not a bottleneck. For example most local area networks (LAN) will not be a bottleneck if the server is not in the same LAN. Thus link speeds of WLAN or Ethernet networks are normally not a basis for estimating the available bandwidth. Cellular devices or other devices directly connected to a modem or connection enabling device may more accurately estimate the bottleneck bandwidth and what is a reasonable share of it for RTSP controlled media. The client will also need to take into account other traffic sharing the bottleneck. For example by only assigning a certain fraction to RTSP and its media streams. It is RECOMMENDED that only clients that have accurate and explicit information about bandwidth bottlenecks uses this header. This header is not a substitute for proper congestion control. It is only a method providing an initial estimate and coarsely determines Schulzrinne, et al. Expires March 15, 2014 [Page 142] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 if the selected content can be delivered at all. Example: Bandwidth: 62360 18.10. Blocksize The Blocksize request-header field is sent from the client to the media server asking the server for a particular media packet size. This packet size does not include lower-layer headers such as IP, UDP, or RTP. The server is free to use a blocksize which is lower than the one requested. The server MAY truncate this packet size to the closest multiple of the minimum, media-specific block size, or override it with the media-specific size if necessary. The block size MUST be a positive decimal number, measured in octets. The server only returns an error (4xx) if the value is syntactically invalid. 18.11. Cache-Control The Cache-Control general-header field is used to specify directives that MUST be obeyed by all caching mechanisms along the request/ response chain. Cache directives MUST be passed through by a proxy or gateway application, regardless of their significance to that application, since the directives may be applicable to all recipients along the request/response chain. It is not possible to specify a cache- directive for a specific cache. Cache-Control should only be specified in a DESCRIBE, GET_PARAMETER, SET_PARAMETER and SETUP request and its response. Note: Cache- Control does not govern just the caching of responses as for HTTP, instead it also applies to the media stream identified by the SETUP request. The RTSP requests are generally not cacheable, for further information see Section 16. Below are the descriptions of the cache directives that can be included in the Cache-Control header. no-cache: Indicates that the media stream or RTSP response MUST NOT be cached anywhere. This allows an origin server to prevent caching even by caches that have been configured to return stale responses to client requests. Note, there is no security function preventing the caching of content. Schulzrinne, et al. Expires March 15, 2014 [Page 143] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 public: Indicates that the media stream or RTSP response is cacheable by any cache. private: Indicates that the media stream or RTSP response is intended for a single user and MUST NOT be cached by a shared cache. A private (non-shared) cache may cache the media streams. no-transform: An intermediate cache (proxy) may find it useful to convert the media type of a certain stream. A proxy might, for example, convert between video formats to save cache space or to reduce the amount of traffic on a slow link. Serious operational problems may occur, however, when these transformations have been applied to streams intended for certain kinds of applications. For example, applications for medical imaging, scientific data analysis and those using end- to-end authentication all depend on receiving a stream that is bit-for-bit identical to the original media stream or RTSP response. Therefore, if a response includes the no-transform directive, an intermediate cache or proxy MUST NOT change the encoding of the stream or response. Unlike HTTP, RTSP does not provide for partial transformation at this point, e.g., allowing translation into a different language. only-if-cached: In some cases, such as times of extremely poor network connectivity, a client may want a cache to return only those media streams or RTSP responses that it currently has stored, and not to receive these from the origin server. To do this, the client may include the only-if-cached directive in a request. If it receives this directive, a cache SHOULD either respond using a cached media stream or response that is consistent with the other constraints of the request, or respond with a 504 (Gateway Timeout) status. However, if a group of caches is being operated as a unified system with good internal connectivity, such a request MAY be forwarded within that group of caches. max-stale: Indicates that the client is willing to accept a media stream or RTSP response that has exceeded its expiration time. If max-stale is assigned a value, then the client is willing to accept a response that has exceeded its expiration time by no more than the specified number of seconds. If no value is assigned to max-stale, then the client is willing to accept a stale response of any age. Schulzrinne, et al. Expires March 15, 2014 [Page 144] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 min-fresh: Indicates that the client is willing to accept a media stream or RTSP response whose freshness lifetime is no less than its current age plus the specified time in seconds. That is, the client wants a response that will still be fresh for at least the specified number of seconds. must-revalidate: When the must-revalidate directive is present in a SETUP response received by a cache, that cache MUST NOT use the cache entry after it becomes stale to respond to a subsequent request without first revalidating it with the origin server. That is, the cache is required to do an end-to-end revalidation every time, if, based solely on the origin server's Expires, the cached response is stale. proxy-revalidate: The proxy-revalidate directive has the same meaning as the must-revalidate directive, except that it does not apply to non-shared user agent caches. It can be used on a response to an authenticated request to permit the user's cache to store and later return the response without needing to revalidate it (since it has already been authenticated once by that user), while still requiring proxies that service many users to revalidate each time (in order to make sure that each user has been authenticated). Note that such authenticated responses also need the public cache control directive in order to allow them to be cached at all. max-age: When an intermediate cache is forced, by means of a max- age=0 directive, to revalidate its own cache entry, and the client has supplied its own validator in the request, the supplied validator might differ from the validator currently stored with the cache entry. In this case, the cache MAY use either validator in making its own request without affecting semantic transparency. However, the choice of validator might affect performance. The best approach is for the intermediate cache to use its own validator when making its request. If the server replies with 304 (Not Modified), then the cache can return its now validated copy to the client with a 200 (OK) response. If the server replies with a new message body and cache validator, however, the intermediate cache can compare the returned validator with the one provided in the client's request, using the strong comparison function. If the client's validator is equal to the origin server's, then the intermediate cache simply returns 304 (Not Modified). Otherwise, it returns the new message body with a 200 (OK) response. Schulzrinne, et al. Expires March 15, 2014 [Page 145] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18.12. Connection The Connection general-header field allows the sender to specify options that are desired for that particular connection. It MUST NOT be communicated by proxies over further connections. RTSP 2.0 proxies MUST parse the Connection header field before a message is forwarded and, for each connection-token in this field, remove any header field(s) from the message with the same name as the connection-token. Connection options are signaled by the presence of a connection-token in the Connection header field, not by any corresponding additional header field(s), since the additional header field may not be sent if there are no parameters associated with that connection option. Message headers listed in the Connection header MUST NOT include end- to-end headers, such as Cache-Control. RTSP 2.0 defines the "close" connection option for the sender to signal that the connection will be closed after completion of the response. For example, Connection: close in either the request or the response header fields indicates that the connection SHOULD NOT be considered `persistent' (Section 10.2) after the current request/ response is complete. The use of the connection option "close" in RTSP messages SHOULD be limited to error messages when the server is unable to recover and therefore sees it necessary to close the connection. The reason is that the client has the choice of continuing using a connection indefinitely, as long as it sends valid messages. 18.13. Connection-Credentials The Connection-Credentials response header is used to carry the chain of credentials for any next hop that needs to be approved by the requester. It MUST only be used in server to client responses. The Connection-Credentials header in an RTSP response MUST, if included, contain the credential information (in form of a list of certificates providing the chain of certification) of the next hop that an intermediary needs to securely connect to. The header MUST include the URI of the next hop (proxy or server) and a base64 [RFC4648] encoded binary structure containing a sequence of DER encoded X.509v3 certificates[RFC5280] . The binary structure starts with the number of certificates (NR_CERTS) included as a 16 bit unsigned integer. This is followed by NR_CERTS number of 16 bit unsigned integers providing the size in Schulzrinne, et al. Expires March 15, 2014 [Page 146] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 octets of each DER encoded certificate. This is followed by NR_CERTS number of DER encoded X.509v3 certificates in a sequence (chain). This format is exemplified in Figure 2. The proxy or server's certificate must come first in the structure. Each following certificate must directly certify the one preceding it. Because certificate validation requires that root keys be distributed independently, the self-signed certificate which specifies the root certificate authority may optionally be omitted from the chain, under the assumption that the remote end must already possess it in order to validate it in any case. Example: Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC... Where MIIDNTCC... is a Base64 encoding of the following structure: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Number of certificates | Size of certificate #1 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Size of certificate #2 | Size of certificate #3 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : DER Encoding of Certificate #1 : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : DER Encoding of Certificate #2 : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : DER Encoding of Certificate #3 : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 2: Connection-Credentials header's Certificate Format Example 18.14. Content-Base The Content-Base message-header field may be used to specify the base URI for resolving relative URIs within the message body. Content-Base: rtsp://media.example.com/movie/twister/ If no Content-Base field is present, the base URI of an message body is defined either by its Content-Location (if that Content-Location URI is an absolute URI) or the URI used to initiate the request, in that order of precedence. Note, however, that the base URI of the contents within the message-body may be redefined within that message-body. Schulzrinne, et al. Expires March 15, 2014 [Page 147] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18.15. Content-Encoding The Content-Encoding message-header field is used as a modifier to the media-type. When present, its value indicates what additional content codings have been applied to the message body, and thus what decoding mechanisms must be applied in order to obtain the media-type referenced by the Content-Type header field. Content-Encoding is primarily used to allow a document to be compressed without losing the identity of its underlying media type. The content-coding is a characteristic of the message body identified by the Request-URI. Typically, the message body is stored with this encoding and is only decoded before rendering or analogous usage. However, an RTSP proxy MAY modify the content-coding if the new coding is known to be acceptable to the recipient, unless the "no- transform" cache-control directive is present in the message. If the content-coding of a message body is not "identity", then the response MUST include a Content-Encoding Message-body header that lists the non-identity content-coding(s) used. If the content-coding of a message body in a request message is not acceptable to the origin server, the server SHOULD respond with a status code of 415 (Unsupported Media Type). If multiple encodings have been applied to a message body, the content codings MUST be listed in the order in which they were applied, first to last from left to right. Additional information about the encoding parameters MAY be provided by other header fields not defined by this specification. 18.16. Content-Language The Content-Language message-header field describes the natural language(s) of the intended audience for the enclosed message body. Note that this might not be equivalent to all the languages used within the message body. Language tags are mentioned in Section 18.4. The primary purpose of Content-Language is to allow a user to identify and differentiate entities according to the user's own preferred language. Thus, if the body content is intended only for a Danish-literate audience, the appropriate field is Schulzrinne, et al. Expires March 15, 2014 [Page 148] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Content-Language: da If no Content-Language is specified, the default is that the content is intended for all language audiences. This might mean that the sender does not consider it to be specific to any natural language, or that the sender does not know for which language it is intended. Multiple languages MAY be listed for content that is intended for multiple audiences. For example, a rendition of the "Treaty of Waitangi," presented simultaneously in the original Maori and English versions, would call for Content-Language: mi, en However, just because multiple languages are present within a message body does not mean that it is intended for multiple linguistic audiences. An example would be a beginner's language primer, such as "A First Lesson in Latin," which is clearly intended to be used by an English-literate audience. In this case, the Content-Language would properly only include "en". Content-Language MAY be applied to any media type -- it is not limited to textual documents. 18.17. Content-Length The Content-Length message-header field contains the length of the message body of the RTSP message (i.e., after the double CRLF following the last header). Unlike HTTP, it MUST be included in all messages that carry a message body beyond the header portion of the RTSP message. If it is missing, a default value of zero is assumed. Any Content-Length greater than or equal to zero is a valid value. 18.18. Content-Location The Content-Location message-header field MAY be used to supply the resource location for the message body enclosed in the message when that body is accessible from a location separate from the requested resource's URI. A server SHOULD provide a Content-Location for the variant corresponding to the response message body; especially in the case where a resource has multiple variants associated with it, and those entities actually have separate locations by which they might be individually accessed, the server SHOULD provide a Content- Location for the particular variant which is returned. As example, if an RTSP client performs a DESCRIBE request on a given resource, e.g., "rtsp://a.example.com/movie/Plan9FromOuterSpace", then the server may use additional information, such as the User- Schulzrinne, et al. Expires March 15, 2014 [Page 149] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Agent header, to determine the capabilities of the agent. The server will then return a media description tailored to that class of RTSP agents. To indicate which specific description the agent receives the resource identifier ("rtsp://a.example.com/movie/Plan9FromOuterSpace/FullHD.sdp") is provided in Content-Location, while the description is still a valid response for the generic resource identifier. Thus enabling both debugging and cache operation as discussed below. The Content-Location value is not a replacement for the original requested URI; it is only a statement of the location of the resource corresponding to this particular variant at the time of the request. Future requests MAY specify the Content-Location URI as the request URI if the desire is to identify the source of that particular variant. This is useful if the RTSP agent desires to verify if the resource variant is current through a conditional request. A cache cannot assume that a message body with a Content-Location different from the URI used to retrieve it can be used to respond to later requests on that Content-Location URI. However, the Content- Location can be used to differentiate between multiple variants retrieved from a single requested resource. If the Content-Location is a relative URI, the relative URI is interpreted relative to the Request-URI. Note, that Content-Location can be used in some cases to derive the base-URI for relative URI(s) present in session description formats. This needs to be taken into account when Content-Location is used. The easiest way to avoid needing to consider that issue is to include the Content-Base whenever the Content-Location is included. Note also, when using Media Tags in conjunction with Content-Location it is important that the different versions have different MTags, even if provided under different Content-Location URIs. This as they have still been provided under the same request URI. Note also, as in most cases the URI used in the DESCRIBE and the SETUP requests are different, the URI provided in a DESCRIBE Content- Location response can't directly be used in a SETUP request. Instead the extra step of resolving URIs combined with the media descriptions indication, like with SDP's a=control attribute. 18.19. Content-Type The Content-Type message-header indicates the media type of the message body sent to the recipient. Note that the content types suitable for RTSP are likely to be restricted in practice to Schulzrinne, et al. Expires March 15, 2014 [Page 150] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 presentation descriptions and parameter-value types. 18.20. 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. For each new RTSP request an agent creates the CSeq value MUST be incremented by one. This primarily allows for associating requests with responses. It also enables detecting loss of a request and await a retransmission prior to processing a sub-sequent request when using unreliable transport. 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. Agents receiving a request over a reliable transport with an in-order delivery MUST ignore how the sequence value increments, i.e. it can increment with other values than 1. 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 socket address (IP address and port number), i.e., per direction of a TCP connection. The above rules may appear unnecessary loose. However, they are allowing for a behavior which is not uncommon. When using multiple connections in sequence it may still be easiest to use a single sequence number series for a client connecting with a particular server. Thus, the initial sequence number may be arbitrary depending on the number of previous requests. Proxies that aggregate several client sessions on the same transport will have to ensure that the requests sent towards a particular server have a joint sequence number space. A proxy having one client with concurrent sessions with two different servers using the same client proxy connection can avoid rewriting on the proxy to server connection. The latest equally applies to server to client requests, where one server may have multiple clients over the same proxy. The proxy can use only one joint sequence number space for a given transport connection to a particular server for sending request, as the identification of the RTSP agents, i.e., the proxy and the server is based on the IP address and port number. This requires that the proxy renumbers the CSeq header field in both requests and responses to fulfill the rules for the header. An RTSP proxy MUST renumber the RTSP request from RTSP agents that Schulzrinne, et al. Expires March 15, 2014 [Page 151] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 are sent to a particular RTSP agent in order to preserve the joint sequence number space on the connection between the proxy and the agent. The RTSP proxy MUST increase the CSeq for each request it transmits over a particular transport connection or transport flow, without regard of different sessions. An RTSP proxy MUST renumber RTSP responses back to the sequence number that the corresponding request had when originally received by the proxy before forwarding it to the RTSP agent. A proxy that forwards responses from multiple RTSP agents towards a specific agent MUST maintain the order between request/responses on a per incoming connection basis. This means that different RTSP sessions are handled over the same same transport connection between proxy and the specific agent. Given that the RTSP proxy and the agents are using reliable transport connections, the proxy MAY forward received responses without considering the response's relation to responses from other connections which will share the same outgoing transport connection from the proxy. Note: This exception is to avoid responses being blocked by other agents being slow to respond. This can result in out-of- order delivery of responses arriving at the RTSP client in relation to the transport connection, but that delivery is in- order with respect to the RTSP agent and any session. 18.21. Date The Date general-header field represents the date and time at which the message was originated. The inclusion of the Date header in RTSP message follows these rules: o An RTSP message, sent either by the client or the server, containing a body MUST include a Date header, if the sending host has a clock; o Clients and servers are RECOMMENDED to include a Date header in all other RTSP messages, if the sending host has a clock; o If the server does not have a clock that can provide a reasonable approximation of the current time, its responses MUST NOT include a Date header field. In this case, this rule MUST be followed: Some origin server implementations might not have a clock available. An origin server without a clock MUST NOT assign Expires or Last-Modified values to a response, unless these values were associated with the resource by a system or user with a Schulzrinne, et al. Expires March 15, 2014 [Page 152] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 reliable clock. It MAY assign an Expires value that is known, at or before server configuration time, to be in the past (this allows "pre-expiration" of responses without storing separate Expires values for each resource). A received message that does not have a Date header field MUST be assigned one by the recipient if the message will be cached by that recipient. An RTSP implementation without a clock MUST NOT cache responses without revalidating them on every use. An RTSP cache, especially a shared cache, SHOULD use a mechanism, such as NTP, to synchronize its clock with a reliable external standard. The RTSP-date, a full date as specified by Section 3.3 of [RFC2822], sent in a Date header SHOULD NOT represent a date and time subsequent to the generation of the message. It SHOULD represent the best available approximation of the date and time of message generation, unless the implementation has no means of generating a reasonably accurate date and time. In theory, the date ought to represent the moment just before the message body is generated. In practice, the date can be generated at any time during the message origination without affecting its semantic value. Note: The RTSP 2.0 date is defined as RFC 2822 format date. This format is more allowing than the RTSP 1.0 and earlier draft versions using RFC 1123 date format. Thus implementations should use single spaces as recommended as separators and support receiving the obsoleted identifiers. 18.22. Expires The Expires message-header field gives a date and time after which the description or media-stream should be considered stale. The interpretation depends on the method: DESCRIBE response: The Expires header indicates a date and time after which the presentation description (body) SHOULD be considered stale. SETUP response: The Expires header indicate a date and time after which the media stream SHOULD be considered stale. A stale cache entry may not normally be returned by a cache (either a proxy cache or an user agent cache) unless it is first validated with the origin server (or with an intermediate cache that has a fresh copy of the message body). See Section 16 for further discussion of the expiration model. The presence of an Expires field does not imply that the original Schulzrinne, et al. Expires March 15, 2014 [Page 153] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 resource will change or cease to exist at, before, or after that time. The format is an absolute date and time as defined by RTSP-date. An example of its use is Expires: Thu, 01 Dec 1994 16:00:00 GMT RTSP/2.0 clients and caches MUST treat other invalid date formats, especially including the value "0", as having occurred in the past (i.e., already expired). To mark a response as "already expired," an origin server should use an Expires date that is equal to the Date header value. To mark a response as "never expires," an origin server SHOULD use an Expires date approximately one year from the time the response is sent. RTSP/2.0 servers SHOULD NOT send Expires dates more than one year in the future. 18.23. From The From request-header field, if given, SHOULD contain an Internet e-mail address for the human user who controls the requesting user agent. The address SHOULD be machine-usable, as defined by "mailbox" in [RFC1123]. This header field MAY be used for logging purposes and as a means for identifying the source of invalid or unwanted requests. It SHOULD NOT be used as an insecure form of access protection. The interpretation of this field is that the request is being performed on behalf of the person given, who accepts responsibility for the method performed. In particular, robot agents SHOULD include this header so that the person responsible for running the robot can be contacted if problems occur on the receiving end. The Internet e-mail address in this field MAY be separate from the Internet host which issued the request. For example, when a request is passed through a proxy the original issuer's address SHOULD be used. The client SHOULD NOT send the From header field without the user's approval, as it might conflict with the user's privacy interests or their site's security policy. It is strongly recommended that the user be able to disable, enable, and modify the value of this field at any time prior to a request. Schulzrinne, et al. Expires March 15, 2014 [Page 154] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18.24. If-Match The If-Match request-header field is especially useful for ensuring the integrity of the presentation description, independent of how the presentation description was received. The presentation description can be fetched via means external to RTSP (such as HTTP) or via the DESCRIBE message. In the case of retrieving the presentation description via RTSP, the server implementation is guaranteeing the integrity of the description between the time of the DESCRIBE message and the SETUP message. By including the MTag given in or with the session description in an If-Match header part of the SETUP request, the client ensures that resources set up are matching the description. A SETUP request with the If-Match header for which the MTag validation check fails, MUST generate a response using 412 (Precondition Failed). This validation check is also very useful if a session has been redirected from one server to another. 18.25. If-Modified-Since The If-Modified-Since request-header field is used with the DESCRIBE and SETUP methods to make them conditional. If the requested variant has not been modified since the time specified in this field, a description will not be returned from the server (DESCRIBE) or a stream will not be set up (SETUP). Instead, a 304 (Not Modified) response MUST be returned without any message-body. An example of the field is: If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT 18.26. If-None-Match This request header can be used with one or several message body tags to make DESCRIBE requests conditional. A client that has one or more message bodies previously obtained from the resource, can verify that none of those entities is current by including a list of their associated message body tags in the If-None-Match header field. The purpose of this feature is to allow efficient updates of cached information with a minimum amount of transaction overhead. As a special case, the value "*" matches any current entity of the resource. If any of the message body tags match the message body tag of the message body that would have been returned in the response to a similar DESCRIBE request (without the If-None-Match header) on that resource, or if "*" is given and any current entity exists for that Schulzrinne, et al. Expires March 15, 2014 [Page 155] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 resource, then the server MUST NOT perform the requested method, unless required to do so because the resource's modification date fails to match that supplied in an If-Modified-Since header field in the request. Instead, if the request method was DESCRIBE, the server SHOULD respond with a 304 (Not Modified) response, including the cache-related header fields (particularly MTag) of one of the message bodies that matched. For all other request methods, the server MUST respond with a status of 412 (Precondition Failed). See Section 16.1.3 for rules on how to determine if two message body tags match. If none of the message body tags match, then the server MAY perform the requested method as if the If-None-Match header field did not exist, but MUST also ignore any If-Modified-Since header field(s) in the request. That is, if no message body tags match, then the server MUST NOT return a 304 (Not Modified) response. If the request would, without the If-None-Match header field, result in anything other than a 2xx or 304 status, then the If-None-Match header MUST be ignored. (See Section 16.1.4 for a discussion of server behavior when both If-Modified-Since and If-None-Match appear in the same request.) The result of a request having both an If-None-Match header field and an If-Match header field is unspecified and MUST be considered an illegal request. 18.27. Last-Modified The Last-Modified message-header field indicates the date and time at which the origin server believes the presentation description or media stream was last modified. For the method DESCRIBE, the header field indicates the last modification date and time of the description, for SETUP that of the media stream. An origin server MUST NOT send a Last-Modified date which is later than the server's time of message origination. In such cases, where the resource's last modification would indicate some time in the future, the server MUST replace that date with the message origination date. An origin server SHOULD obtain the Last-Modified value of the message body as close as possible to the time that it generates the Date value of its response. This allows a recipient to make an accurate assessment of the message body's modification time, especially if the message body changes near the time that the response is generated. Schulzrinne, et al. Expires March 15, 2014 [Page 156] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTSP servers SHOULD send Last-Modified whenever feasible. 18.28. Location The Location response-header field is used to redirect the recipient to a location other than the Request-URI for completion of the request or identification of a new resource. For 3rr responses, the location SHOULD indicate the server's preferred URI for automatic redirection to the resource. The field value consists of a single absolute URI. Note: The Content-Location header field (Section 18.18) differs from Location in that the Content-Location identifies the original location of the message body enclosed in the request. It is therefore possible for a response to contain header fields for both Location and Content-Location. Also, see Section 16.2 for cache requirements of some methods. 18.29. Media-Properties This general header is used in SETUP response or PLAY_NOTIFY requests to indicate the media's properties that currently are applicable to the RTSP session. PLAY_NOTIFY MAY be used to modify these properties at any point. However, the client SHOULD have received the update prior to any action related to the new media properties taking effect. For aggregated sessions, the Media-Properties header will be returned in each SETUP response. The header received in the latest response is the one that applies on the whole session from this point until any future update. The header MAY be included without value in GET_PARAMETER requests to the server with a Session header included to query the current Media-Properties for the session. The responder MUST include the current session's media properties. The media properties expressed by this header is the one applicable to all media in the RTSP session. For aggregated sessions, the header expressed the combined media-properties. As a result, aggregation of media MAY result in a change of the media properties, and thus the content of the Media-Properties header contained in subsequent SETUP responses. The header contains a list of property values that are applicable to the currently setup media or aggregate of media as indicated by the RTSP URI in the request. No ordering is enforced within the header. Property values should be grouped into a single group that handles a particular orthogonal property. Values or groups that express multiple properties SHOULD NOT be used. The list of properties that can be expressed MAY be extended at any time. Unknown property values MUST be ignored. Schulzrinne, et al. Expires March 15, 2014 [Page 157] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 This specification defines the following 4 groups and their property values: Random Access: Random-Access: Indicates that random access is possible. May optionally include a floating point value in seconds indicating the longest duration between any two random access points in the media. Beginning-Only: Seeking is limited to the beginning only. No-Seeking: No seeking is possible. Content Modifications: Immutable: The content will not be changed during the life-time of the RTSP session. Dynamic: The content may be changed based on external methods or triggers Time-Progressing: The media accessible progresses as wallclock time progresses. Retention: Unlimited: Content will be retained for the duration of the life- time of the RTSP session. Time-Limited: Content will be retained at least until the specified wallclock time. The time must be provided in the absolute time format specified in Section 4.4.3. Time-Duration: Each individual media unit is retained for at least the specified time duration. This definition allows for retaining data with a time based sliding window. The time duration is expressed as floating point number in seconds. 0.0 is a valid value as this indicates that no data is retained in a time-progressing session. Supported Scale: Scales: A quoted comma separated list of one or more decimal values or ranges of scale values supported by the content in arbitrary order. A range has a start and stop value separated by a colon. A range indicates that the content supports fine grained selection of scale values. Fine grained allows for Schulzrinne, et al. Expires March 15, 2014 [Page 158] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 steps at least as small as one tenth of a scale value. A content is considered to support fine grained selection when the server in response to a given scale value can produce content with an actual scale that is less than 1 tenth of scale unit, i.e., 0.1, from the requested value. Negative values are supported. The value 0 has no meaning and MUST NOT be used. Examples of this header for on-demand content and a live stream without recording are: On-demand: Media-Properties: Random-Access=2.5, Unlimited, Immutable, Scales="-20, -10, -4, 0.5:1.5, 4, 8, 10, 15, 20" Live stream without recording/timeshifting: Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0.0 18.30. Media-Range The Media-Range general header is used to give the range of the media at the time of sending the RTSP message. This header MUST be included in SETUP response, and PLAY and PAUSE response for media that are Time-Progressing, and PLAY and PAUSE response after any change for media that are Dynamic, and in PLAY_NOTIFY request that are sent due to Media-Property-Update. Media-Range header without any range specifications MAY be included in GET_PARAMETER requests to the server to request the current range. The server MUST in this case include the current range at the time of sending the response. The header MUST include range specifications for all time formats supported for the media, as indicated in Accept-Ranges header (Section 18.5) when setting up the media. The server MAY include more than one range specification of any given time format to indicate media that has non-continuous range. The range specifications shall be ordered with the range with the lowest value or earliest start time first, followed by ranges with increasingly higher values or later start time. For media that has the Time-Progressing property, the Media-Range values will only be valid for the particular point in time when it was issued. As wallclock progresses so will also the media range. However, it shall be assumed that media time progresses in direct relationship to wallclock time (with the exception of clock skew) so that a reasonably accurate estimation of the media range can be calculated. Schulzrinne, et al. Expires March 15, 2014 [Page 159] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18.31. MTag The MTag response header MAY be included in DESCRIBE, GET_PARAMETER or SETUP responses. The message body tags (Section 4.6) returned in a DESCRIBE response, and the one in SETUP refers to the presentation, i.e., both the returned session description and the media stream. This allows for verification that one has the right session description to a media resource at the time of the SETUP request. However, it has the disadvantage that a change in any of the parts results in invalidation of all the parts. If the MTag is provided both inside the message body, e.g., within the "a=mtag" attribute in SDP, and in the response message, then both tags MUST be identical. It is RECOMMENDED that the MTag is primarily given in the RTSP response message, to ensure that caches can use the MTag without requiring content inspection. However, for session descriptions that are distributed outside of RTSP, for example using HTTP, etc. it will be necessary to include the message body tag in the session description as specified in Appendix D.1.9. SETUP and DESCRIBE requests can be made conditional upon the MTag using the headers If-Match (Section 18.24) and If-None-Match ( Section 18.26). 18.32. Notify-Reason The Notify Reason header is solely used in the PLAY_NOTIFY method. It indicates the reason why the server has sent the asynchronous PLAY_NOTIFY request (see Section 13.5). 18.33. Pipelined-Requests The Pipelined-Requests general header is used to indicate that a request is to be executed in the context created by a previous request(s). The primary usage of this header is to allow pipelining of SETUP requests so that any additional SETUP request after the first one does not need to wait for the session ID to be sent back to the requesting agent. The header contains a unique identifier that is scoped by the persistent connection used to send the requests. Upon receiving a request with the Pipelined-Requests the responding agent MUST look up if there exists a binding between this Pipelined- Requests identifier for the current persistent connection and an RTSP session ID. If that exists then the received request is processed the same way as if it contained the Session header with the found session ID. If there does not exist a mapping and no Session header is included in the request, the responding agent MUST create a binding upon the successful completion of a session creating request, Schulzrinne, et al. Expires March 15, 2014 [Page 160] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 i.e., SETUP. A binding MUST NOT be created, if the request failed to create an RTSP session. In case the request contains both a Session header and the Pipelined-Requests header the Pipelined-Requests MUST be ignored. Note: Based on the above definition at least the first request containing a new unique Pipelined-Requests will be required to be a SETUP request (unless the protocol is extended with new methods of creating a session). After that first one, additional SETUP requests or requests of any type using the RTSP session context may include the Pipelined-Requests header. When responding to any request that contained the Pipelined-Requests header the server MUST also include the Session header when a binding to a session context exists. An RTSP agent that knows the session identifier SHOULD NOT use the Pipelined-Requests header in any request and only use the Session header. This as the Session identifier is persistent across transport contexts, like TCP connections, which the Pipelined-Requests identifier is not. The RTSP agent sending the request with a Pipelined-Requests header has the responsibility for using a unique and previously unused identifier within the transport context. Currently only a TCP connection is defined as such transport context. A server MUST delete the Pipelined-Requests identifier and its binding to a session upon the termination of that session. Despite the previous mandate, RTSP agents are RECOMMENDED to not reuse identifiers to allow for better error handling and logging. RTSP Proxies may need to translate Pipelined-Requests identifier values from incoming requests to outgoing to allow for aggregation of requests onto a persistent connection. 18.34. Proxy-Authenticate The Proxy-Authenticate response-header field MUST be included as part of a 407 (Proxy Authentication Required) response. The field value consists of a challenge that indicates the authentication scheme and parameters applicable to the proxy for this Request-URI. The HTTP access authentication process is described in [RFC2617]. Unlike WWW-Authenticate, the Proxy-Authenticate header field applies only to the current connection and SHOULD NOT be passed on to downstream agents. This header MUST only be used in response messages related to client to server requests. Schulzrinne, et al. Expires March 15, 2014 [Page 161] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18.35. Proxy-Authentication-Info The Proxy-Authentication-Info header is used by the proxy to communicate some information regarding the successful authentication to the proxy in the message response. The content and usage of this header is described in the HTTP access authentication [RFC2617] that is also used by RTSP and clarified in Section 19.1. This header MUST only be used in response messages related to client to server requests. This header has hop by hop scope. 18.36. Proxy-Authorization The Proxy-Authorization request-header field allows the client to identify itself (or its user) to a proxy which requires authentication. The Proxy-Authorization field value consists of credentials containing the authentication information of the user agent for the proxy and/or realm of the resource being requested. The HTTP access authentication process is described in [RFC2617]. Unlike Authorization, the Proxy-Authorization header field applies only to the next hop proxy. This header MUST only be used in client to server requests. 18.37. Proxy-Require The Proxy-Require request-header field is used to indicate proxy- sensitive features that MUST be supported by the proxy. Any Proxy- Require header features that are not supported by the proxy MUST be negatively acknowledged by the proxy to the client using the Unsupported header. The proxy MUST use the 551 (Option Not Supported) status code in the response. Any feature-tag included in the Proxy-Require does not apply to the end-point (server or client). To ensure that a feature is supported by both proxies and servers the tag needs to be included in also a Require header. See Section 18.43 for more details on the mechanics of this message and a usage example. See discussion in the proxies section (Section 15.1) about when to consider that a feature requires proxy support. Example of use: Proxy-Require: play.basic 18.38. Proxy-Supported The Proxy-Supported general-header field enumerates all the extensions supported by the proxy using feature-tags. The header Schulzrinne, et al. Expires March 15, 2014 [Page 162] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 carries the intersection of extensions supported by the forwarding proxies. The Proxy-Supported header MAY be included in any request by a proxy. It MUST be added by any proxy if the Supported header is present in a request. When present in a request, the receiver MUST in the response copy the received Proxy-Supported header. The Proxy-Supported header field contains a list of feature-tags applicable to proxies, as described in Section 4.5. The list is the intersection of all feature-tags understood by the proxies. To achieve an intersection, the proxy adding the Proxy-Supported header includes all proxy feature-tags it understands. Any proxy receiving a request with the header, MUST check the list and removes any feature-tag(s) it does not support. A Proxy-Supported header present in the response MUST NOT be modified by the proxies. These feature tags are the ones the proxy chain support in general, and is not specific to the request resource. Example: C->P1: OPTIONS rtsp://example.com/ RTSP/2.0 Supported: foo, bar, blech User-Agent: PhonyClient/1.2 P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0 Supported: foo, bar, blech Proxy-Supported: proxy-foo, proxy-bar, proxy-blech Via: 2.0 pro.example.com P2->S: OPTIONS rtsp://example.com/ RTSP/2.0 Supported: foo, bar, blech Proxy-Supported: proxy-foo, proxy-blech Via: 2.0 pro.example.com, 2.0 prox2.example.com S->C: RTSP/2.0 200 OK Supported: foo, bar, baz Proxy-Supported: proxy-foo, proxy-blech Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN Via: 2.0 pro.example.com, 2.0 prox2.example.com 18.39. Public The Public response header field lists the set of methods supported by the response sender. This header applies to the general capabilities of the sender and its only purpose is to indicate the sender's capabilities to the recipient. The methods listed may or may not be applicable to the Request-URI; the Allow header field (Section 18.6) MAY be used to indicate methods allowed for a particular URI. Schulzrinne, et al. Expires March 15, 2014 [Page 163] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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. 18.40. Range The Range general-header specifies a time range in PLAY (Section 13.4), PAUSE (Section 13.6), SETUP (Section 13.3), REDIRECT (Section 13.10), and PLAY_NOTIFY (Section 13.5) requests and responses. It MAY be included in GET_PARAMETER requests from the client to the server with only a Range format and no value to request the current media position, whether the session is in Play or Ready state in the included format. The server SHALL, if supporting the range format, respond with the current playing point or pause point as the start of the range. If an explicit stop point was used in the previous PLAY request, then that value shall be included as stop point. Note that if the server is currently under any type of media playback manipulation affecting the interpretation of Range, like Scale, that is also required to be included in any GET_PARAMETER response to provide complete information. The range can be specified in a number of units. This specification defines smpte (Section 4.4.1), npt (Section 4.4.2), and clock (Section 4.4.3) range units. While byte ranges [H14.35.1] and other extended units MAY be used, their behavior is unspecified since they are not normally meaningful in RTSP. Servers supporting the Range header MUST understand the NPT range format and SHOULD understand the SMPTE range format. If the Range header is sent in a time format that is not understood, the recipient SHOULD return 456 (Header Field Not Valid for Resource) and include an Accept-Ranges header indicating the supported time formats for the given resource. Example: Schulzrinne, et al. Expires March 15, 2014 [Page 164] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Range: clock=19960213T143205Z- The Range header contains a range of one single range format. A range is a half-open interval with a start and an end point, including the start point, but excluding the end point. A range may either be fully specified with explicit values for start point and end point, or have either start or end point be implicit. An implicit start point indicates the session's pause point, and if no pause point is set the start of the content. An implicit end point indicates the end of the content. The usage of both implicit start and end point is not allowed in the same range header, however, the exclusion of the range header has that meaning, i.e., from pause point (or start) until end of content. Regarding the half-open intervals; a range of A-B starts exactly at time A, but ends just before B. Only the start time of a media unit such as a video or audio frame is relevant. For example, assume that video frames are generated every 40 ms. A range of 10.0-10.1 would include a video frame starting at 10.0 or later time and would include a video frame starting at 10.08, even though it lasted beyond the interval. A range of 10.0-10.08, on the other hand, would exclude the frame at 10.08. Please note the difference between NPT time scales' "now" and an implicit start value. Implicit value reference the current pause- point. While "now" is the currently ongoing time. In a time- progressing session with recording (retention for some or full time) the pause point may be 2 min into the session while now could be 1 hour into the session. By default, range intervals increase, where the second point is larger than the first point. Example: Range: npt=10-15 However, range intervals can also decrease if the Scale header (see Section 18.46) indicates a negative scale value. For example, this would be the case when a playback in reverse is desired. Example: Scale: -1 Range: npt=15-10 Decreasing ranges are still half open intervals as described above. Thus, for range A-B, A is closed and B is open. In the above Schulzrinne, et al. Expires March 15, 2014 [Page 165] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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. 18.41. Referrer The Referrer request-header field allows the client to specify, for the server's benefit, the address (URI) of the resource from which the Request-URI was obtained. The URI refers to that of the presentation description, typically retrieved via HTTP. The Referrer request-header allows a server to generate lists of back-links to resources for interest, logging, optimized caching, etc. It also allows obsolete or mistyped links to be traced for maintenance. The Referrer field MUST NOT be sent if the Request-URI was obtained from a source that does not have its own URI, such as input from the user keyboard. If the field value is a relative URI, it SHOULD be interpreted relative to the Request-URI. The URI MUST NOT include a fragment identifier. Because the source of a link might be private information or might reveal an otherwise private information source, it is strongly recommended that the user be able to select whether or not the Referrer field is sent. For example, a streaming client could have a toggle switch for openly/anonymously, which would respectively enable/disable the sending of Referrer and From information. Clients SHOULD NOT include a Referrer header field in a (non-secure) RTSP request if the referring page was transferred with a secure protocol. 18.42. Request-Status This request header is used to indicate the end result for requests that take time to complete, such as PLAY (Section 13.4). It is sent Schulzrinne, et al. Expires March 15, 2014 [Page 166] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 in PLAY_NOTIFY (Section 13.5) with the end-of-stream reason to report how the PLAY request concluded, either in success or in failure. The header carries a reference to the request it reports on using the CSeq number for the session indicated by the Session header in the request. It provides both a numerical status code (according to Section 8.1.1) and a human readable reason phrase. Example: Request-Status: cseq=63 status=500 reason="Media data unavailable" 18.43. Require The Require request-header field is used by agents to ensure that the other end-point supports features that are required in respect to this request. It can also be used to query if the other end-point supports certain features, however, the use of the Supported message- header (Section 18.51) is much more effective in this purpose. In case any of the feature-tags listed by the Require header are not supported by the server or client receiving the request, it MUST respond to the request using the error code 551 (Option Not Supported) and include the Unsupported header listing those feature- tags which are NOT supported. This header does not apply to proxies, for the same functionality in respect to proxies see Proxy-Require header (Section 18.37) with the exception of media modifying proxies. Media modifying proxies, due to their nature of handling media in a way that is very similar to a server, do need to understand also the server's features to correctly serve the client. This is to make sure that the client-server interaction will proceed without delay when all features are understood by both sides, and only slow down if features are not understood (as in the example below). For a well-matched client-server pair, the interaction proceeds quickly, saving a round-trip often required by negotiation mechanisms. In addition, it also removes state ambiguity when the client requires features that the server does not understand. Example (Not complete): C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0 CSeq: 302 Require: funky-feature Funky-Parameter: funkystuff S->C: RTSP/2.0 551 Option not supported CSeq: 302 Unsupported: funky-feature Schulzrinne, et al. Expires March 15, 2014 [Page 167] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 In this example, "funky-feature" is the feature-tag which indicates to the client that the fictional Funky-Parameter field is required. The relationship between "funky-feature" and Funky-Parameter is not communicated via the RTSP exchange, since that relationship is an immutable property of "funky-feature" and thus should not be transmitted with every exchange. Proxies and other intermediary devices MUST ignore this header. If a particular extension requires that intermediate devices support it, the extension should be tagged in the Proxy-Require field instead (see Section 18.37). See discussion in the proxies section (Section 15.1) about when to consider that a feature requires proxy support. 18.44. Retry-After The Retry-After response-header field can be used with a 503 (Service Unavailable) or 553 (Proxy Unavailable) response to indicate how long the service is expected to be unavailable to the requesting client. This field MAY also be used with any 3rr (Redirection) response to indicate the minimum time the user-agent is asked to wait before issuing the redirected request. The value of this field can be either an RTSP-date or an integer number of seconds (in decimal) after the time of the response. Example: Retry-After: Fri, 31 Dec 1999 23:59:59 GMT Retry-After: 120 In the latter example, the delay is 2 minutes. 18.45. RTP-Info The RTP-Info general header field is used to set RTP-specific parameters in the PLAY and GET_PARAMETER responses or a PLAY_NOTIFY and GET_PARAMETER requests. For streams using RTP as transport protocol the RTP-Info header SHOULD be part of a 200 response to PLAY. The exclusion of the RTP-Info in a PLAY response for RTP transported media will result in a client needing to synchronize the media streams using RTCP. This may have negative impact as the RTCP can be lost, and does not need to be particularly timely in its arrival. Also functionality that informs the client from which packet a seek has occurred is affected. The RTP-Info MAY be included in SETUP responses to provide Schulzrinne, et al. Expires March 15, 2014 [Page 168] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 synchronization information when changing transport parameters, see Section 13.3. The RTP-Info header and the Range header MAY be included in a GET_PARAMETER request from client to server without any values to request the current playback point and corresponding RTP synchronization information. When the RTP-Info header is included in a Request the Range header MUST also be included (Note, Range header only MAY be used). The server response SHALL include both the Range header and the RTP-Info header. If the session is in Play state, then the value of the Range header SHALL be filled in with the current playback point and with the corresponding RTP-Info values. If the server is another state, no values are included in the RTP- Info header. The header is included in PLAY_NOTIFY requests with the Notify-Reason of end-of-stream to provide RTP information about the end of the stream. The header can carry the following parameters: url: Indicates the stream URI for which the following RTP parameters correspond, this URI MUST be the same as used in the SETUP request for this media stream. Any relative URI MUST use the Request-URI as base URI. This parameter MUST be present. ssrc: The Synchronization source (SSRC) that the RTP timestamp and sequence number provided applies to. This parameter MUST be present. seq: Indicates the sequence number of the first packet of the stream that is direct result of the request. This allows clients to gracefully deal with packets when seeking. The client uses this value to differentiate packets that originated before the seek from packets that originated after the seek. Note that a client may not receive the packet with the expressed sequence number, and instead packets with a higher sequence number, due to packet loss or reordering. This parameter is RECOMMENDED to be present. rtptime: MUST indicate the RTP timestamp value corresponding to the start time value in the Range response header, or if not explicitly given the implied start point. The client uses this value to calculate the mapping of RTP time to NPT or other media timescale. This parameter SHOULD be present to ensure inter-media synchronization is achieved. There exists no requirement that any received RTP packet will have the same RTP timestamp value as the one in the parameter used to establish synchronization. Schulzrinne, et al. Expires March 15, 2014 [Page 169] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 A mapping from RTP timestamps to Network Time Protocol (NTP) format timestamps (wallclock) is available via RTCP. However, this information is not sufficient to generate a mapping from RTP timestamps to media clock time (NPT, etc.). Furthermore, in order to ensure that this information is available at the necessary time (immediately at startup or after a seek), and that it is delivered reliably, this mapping is placed in the RTSP control channel. In order to compensate for drift for long, uninterrupted presentations, RTSP clients should additionally map NPT to NTP, using initial RTCP sender reports to do the mapping, and later reports to check drift against the mapping. Example: Range:npt=3.25-15 RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102; rtptime=12345678,url="rtsp://example.com/foo/video" ssrc=9A9DE123:seq=30211;rtptime=29567112 Lets assume that Audio uses a 16kHz RTP timestamp clock and Video a 90kHz RTP timestamp clock. Then the media synchronization is depicted in the following way. NPT 3.0---3.1---3.2-X-3.3---3.4---3.5---3.6 Audio PA A Video V PV X: NPT time value = 3.25, from Range header. A: RTP timestamp value for Audio from RTP-Info header (12345678). V: RTP timestamp value for Video from RTP-Info header (29567112). PA: RTP audio packet carrying an RTP timestamp of 12344878. Which corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2 PV: RTP video packet carrying an RTP timestamp of 29573412. Which corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32 18.46. Scale The Scale general-header indicates the requested or used view rate for the media resource being played back. A scale value of 1 indicates normal play at the normal forward viewing rate. If not 1, the value corresponds to the rate with respect to normal viewing rate. For example, a ratio of 2 indicates twice the normal viewing rate ("fast forward") and a ratio of 0.5 indicates half the normal viewing rate. In other words, a ratio of 2 has content time increase at twice the playback time. For every second of elapsed (wallclock) time, 2 seconds of content time will be delivered. A negative value indicates reverse direction. For certain media transports this may Schulzrinne, et al. Expires March 15, 2014 [Page 170] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 require certain considerations to work consistent, see Appendix C.1 for description on how RTP handles this. The transmitted data rate SHOULD NOT be changed by selection of a different scale value. The resulting bit-rate should be reasonably close to the nominal bit-rate of the content for Scale = 1. The server has to actively manipulate the data when needed to meet the bitrate constraints. Implementation of scale changes depends on the server and media type. For video, a server may, for example, deliver only key frames or selected frames. For audio, it may time-scale the audio while preserving pitch or, less desirably, deliver fragments of audio, or completely mute the audio. The server and content may restrict the range of scale values that it supports. The supported values are indicated by the Media-Properties header (Section 18.29). The client SHOULD only indicate request values to be supported. However, as the values may change as the content progresses a requested value may no longer be valid when the request arrives. Thus, a non-supported value in a request does not generate an error, only forces the server to choose the closest value. The response MUST always contain the actual scale value chosen by the server. If the server does not implement the possibility to scale, it will not return a Scale header. A server supporting Scale operations for PLAY MUST indicate this with the use of the "play.scale" feature-tag. When indicating a negative scale for a reverse playback, the Range header MUST indicate a decreasing range as described in Section 18.40. Example of playing in reverse at 3.5 times normal rate: Scale: -3.5 Range: npt=15-10 18.47. Seek-Style When a client sends a PLAY request with a Range header to perform a random access to the media, the client does not know if the server will pick the first media samples or the first random access point prior to the request range. Depending on use case, the client may have a strong preference. To express this preference and provide the client with information on how the server actually acted on that preference the Seek-Style header is defined. Seek-Style is a general header that MAY be included in any PLAY request to indicate the client's preference for any media stream that Schulzrinne, et al. Expires March 15, 2014 [Page 171] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 has random access properties. The server MUST always include the header in any PLAY response for media with random access properties to indicate what policy was applied. A server that receives an unknown Seek-Style policy MUST ignore it and select the server default policy. A client receiving an unknown policy MUST ignore it and use the Range header and any media synchronization information as basis to determine what the server did. This specification defines the following seek policies that may be requested (see also Section 4.7.1): RAP: Random Access Point (RAP) is the behavior of requesting the server to locate the closest previous random access point that exists in the media aggregate and deliver from that. By requesting a RAP, media quality will be the best possible as all media will be delivered from a point where full media state can be established in the media decoder. CoRAP: Conditional Random Access Point (CoRAP) is a variant of the above RAP behavior. This policy is primarily intended for cases where there is larger distance between the random access points in the media. CoRAP is conditioned on that there is a Random Access Point closer to the requested start point than to the current pause point. This policy assumes that the media state existing prior to the pause is usable if delivery is continued. If the client or server knows that this is not the fact the RAP policy should be used. In other words: in most cases when the client requests a start point prior to the current pause point, a valid decoding dependency chain from the media delivered prior to the pause and to the requested media unit will not exist. If the server searched to a random access point the server MUST return the CoRAP policy in the Seek-Style header and adjust the Range header to reflect the position of the picked RAP. In case the random access point is further away and the server selects to continue from the current pause point it MUST include the "Next" policy in the Seek-Style header and adjust the Range header start point to the current pause point. First-Prior: The first-prior policy will start delivery with the media unit that has a playout time first prior to the requested time. For discrete media that would only include media units that would still be rendered at the request time. For continuous media that is media that will be rendered during the requested start time of the range. Schulzrinne, et al. Expires March 15, 2014 [Page 172] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Next: The next media units after the provided start time of the range. For continuous framed media that would mean the first next frame after the provided time. For discrete media the first unit that is to be rendered after the provided time. The main usage for this case is when the client knows it has all media up to a certain point and would like to continue delivery so that a complete non-interrupted media playback can be achieved. Example of such scenarios include switching from a broadcast/multicast delivery to a unicast based delivery. This policy MUST only be used on the client's explicit request. Please note that these expressed preferences exist for optimizing the startup time or the media quality. The "Next" policy breaks the normal definition of the Range header to enable a client to request media with minimal overlap, although some may still occur for aggregated sessions. RAP and First-Prior both fulfill the requirement of providing media from the requested range and forward. However, unless RAP is used, the media quality for many media codecs using predictive methods can be severely degraded unless additional data is available as, for example, already buffered, or through other side channels. 18.48. Server The Server general-header field contains information about the software used by the origin server to create or handle the request. The field can contain multiple product tokens and comments identifying the server and any significant subproducts. The product tokens are listed in order of their significance for identifying the application. Example: Server: PhonyServer/1.0 If the response is being forwarded through a proxy, the proxy application MUST NOT modify the Server response-header. Instead, it SHOULD include a Via field (Section 18.57). If the response is generated by the proxy, the proxy application MUST return the Server response-header as previously returned by the server. 18.49. Session The Session general-header field identifies an RTSP session. An RTSP session is created by the server as a result of a successful SETUP request and in the response the session identifier is given to the client. The RTSP session exists until destroyed by a TEARDOWN, REDIRECT or timed out by the server. Schulzrinne, et al. Expires March 15, 2014 [Page 173] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 The session identifier is chosen by the server (see Section 4.3) and MUST be returned in the SETUP response. Once a client receives a session identifier, it MUST be included in any request related to that session. This means that the Session header MUST be included in a request, using the following methods: PLAY, PAUSE, and TEARDOWN, and MAY be included in SETUP, OPTIONS, SET_PARAMETER, GET_PARAMETER, and REDIRECT, and MUST NOT be included in DESCRIBE. The Session header MUST NOT be included in the following methods, if these requests are pipelined and if the session identifier is not yet known: PLAY, PAUSE, TEARDOWN, SETUP, OPTIONS SET_PARAMETER, and GET_PARAMETER. In an RTSP response the session header MUST be included in methods, SETUP, PLAY, and PAUSE, and MAY be included in methods, TEARDOWN, and REDIRECT, and if included in the request of the following methods it MUST also be included in the response, OPTIONS, GET_PARAMETER, and SET_PARAMETER, and MUST NOT be included in DESCRIBE responses. Note that a session identifier identifies an RTSP session across transport sessions or connections. RTSP requests for a given session can use different URIs (Presentation and media URIs). Note, that there are restrictions depending on the session which URIs that are acceptable for a given method. However, multiple "user" sessions for the same URI from the same client will require use of different session identifiers. The session identifier is needed to distinguish several delivery requests for the same URI coming from the same client. The response 454 (Session Not Found) MUST be returned if the session identifier is invalid. The header MAY include a parameter for session timeout period. If not explicitly provided this value is set to 60 seconds. As this affects how often session keep-alives are needed values smaller than 30 seconds are not recommended. However, larger than default values can be useful in applications of RTSP that have inactive but established sessions for longer time periods. 60 seconds was chosen as session timeout value due to: Resulting in not too frequent keep-alive messages and having low sensitivity to variations in request response timing. If one reduces the timeout value to below 30 seconds the corresponding request response timeout becomes a significant part of the session timeout. 60 seconds also allows for reasonably rapid recovery of committed server resources in case of client failure. Schulzrinne, et al. Expires March 15, 2014 [Page 174] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18.50. Speed The Speed general-header field requests the server to deliver specific amounts of nominal media time per unit of delivery time, contingent on the server's ability and desire to serve the media stream at the given speed. The client requests the delivery speed to be within a given range with a lower and upper bound. The server SHALL deliver at the highest possible speed within the range, but not faster than the upper-bound, for which the underlying network path can support the resulting transport data rates. As long as any speed value within the given range can be provided the server SHALL NOT modify the media quality. Only if the server is unable to deliver media at the speed value provided by the lower bound shall it reduce the media quality. Implementation of the Speed functionality by the server is OPTIONAL. The server can indicate its support through a feature-tag, play.speed. The lack of a Speed header in the response is an indication of lack of support of this functionality. The speed parameter values are expressed as a positive decimal value, e.g., a value of 2.0 indicates that data is to be delivered twice as fast as normal. A speed value of zero is invalid. The range is specified in the form "lower bound - upper bound". The lower bound value may be smaller or equal to the upper bound. All speeds may not be possible to support. Therefore the server MAY modify the requested values to the closest supported. The actual supported speed MUST be included in the response. Note, however, that the use cases may vary and that Speed value ranges such as 0.7 - 0.8, 0.3-2.0, 1.0-2.5, 2.5-2.5 all have their usage. Example: Speed: 1.0-2.5 Use of this header changes the bandwidth used for data delivery. It is meant for use in specific circumstances where delivery of the presentation at a higher or lower rate is desired. The main use cases are buffer operations or local scale operations. Implementors should keep in mind that bandwidth for the session may be negotiated beforehand (by means other than RTSP), and therefore re-negotiation may be necessary. To perform Speed operations the server needs to ensure that the network path can support the resulting bit-rate. Thus the media transport needs to support feedback so that the server can react and adapt to the available bitrate. Schulzrinne, et al. Expires March 15, 2014 [Page 175] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 18.51. Supported The Supported general header enumerates all the extensions supported by the client or server using feature tags. The header carries the extensions supported by the message sending client or server. The Supported header MAY be included in any request. When present in a request, the receiver MUST respond with its corresponding Supported header. Note that the Supported header is also included in 4xx and 5xx responses. The Supported header contains a list of feature-tags, described in Section 4.5, that are understood by the client or server. These feature tags are the ones the server or client support in general, and is not specific to the request resource. Example: C->S: OPTIONS rtsp://example.com/ RTSP/2.0 Supported: foo, bar, blech User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK Supported: bar, blech, baz 18.52. Terminate-Reason The Terminate-Reason request header allows the server when sending a REDIRECT or TEARDOWN request to provide a reason for the session termination and any additional information. This specification identifies three reasons for Redirections and may be extended in the future: Server-Admin: The server needs to be shutdown for some administrative reason. Session-Timeout: A client's session has been kept alive for extended periods of time and the server has determined that it needs to reclaim the resources associated with this session. Internal-Error An internal error that is impossible to recover from has occurred forcing the server to terminate the session. The Server may provide additional parameters containing information around the redirect. This specification defines the following ones. Schulzrinne, et al. Expires March 15, 2014 [Page 176] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 time: Provides a wallclock time when the server will stop providing any service. user-msg: An UTF-8 text string with a message from the server to the user. This message SHOULD be displayed to the user. 18.53. Timestamp The Timestamp general-header describes when the agent sent the request. The value of the timestamp is of significance only to the agent and may use any timescale. The responding agent MUST echo the exact same value and MAY, if it has accurate information about this, add a floating point number indicating the number of seconds that has elapsed since it has received the request. The timestamp can be used by the agent to compute the round-trip time to the responding agent so that it can adjust the timeout value for retransmissions when running over an unreliable protocol. It also resolves retransmission ambiguities for unreliable transport of RTSP. Note that the present specification provides only for reliable transport of RTSP messages. The Timestamp general-header is specified in case the protocol is extended in the future to use unreliable transport. 18.54. Transport The Transport general-header indicates which transport protocol is to be used and configures its parameters such as destination address, compression, multicast time-to-live and destination port for a single stream. It sets those values not already determined by a presentation description. A Transport request header MAY contain a list of transport options acceptable to the client, in the form of multiple transport specification entries. Transport specifications are comma separated, listed in decreasing order of preference. Each transport specification consist of a transport protocol identifier, followed by any number of parameters, each parameter separated by a semicolon. A Transport request header MAY contain multiple transport specifications using the same transport protocol Identifier. The server MUST return a Transport response-header in the response to indicate the values actually chosen if any. If no transport specification is supported, no transport header is returned and the request MUST be responded using the status code 461 (Unsupported Transport) (Section 17.4.25). In case more than one transport specification was present in the request, the server MUST return the single transport specification (transport-spec) which was actually chosen, if any. The number of transport-spec entries is expected to Schulzrinne, et al. Expires March 15, 2014 [Page 177] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 be limited as the client will get guidance on what configurations that are possible from the presentation description. The Transport header MAY also be used in subsequent SETUP requests to change transport parameters. A server MAY refuse to change parameters of an existing stream. The transport protocol identifier defines for each transport specification which transport protocol to use and any related rules. Each transport protocol identifier defines the parameters that are required to occur, additional optional parameters MAY occur. This as parameters may be different and provide different options to the RTSP Agent. A transport specification may only contain one of any given parameter within it. A parameter consists of a name and optionally a value string. Parameters MAY be given in any order. Additionally, transport specification may only contain either of the unicast or the multicast transport type parameter. The transport protocol identifier and all parameters need to be understood in a transport specification, if not, the transport specification MUST be ignored. An RTSP proxy of any type that uses or modifies the transport specification, e.g., access proxy or security proxy, MUST remove specifications with unknown parameters before forwarding the RTSP message. If that results in no remaining transport specification the proxy SHALL send a 461 (Unsupported Transport) (Section 17.4.25) response without any Transport header. The Transport header is restricted to describing a single media stream. (RTSP can also control multiple streams as a single entity.) Making it part of RTSP rather than relying on a multitude of session description formats greatly simplifies designs of firewalls. The general syntax for the transport protocol identifier is a list of slash separated tokens: Value1/Value2/Value3... Which for RTP transports take the form: RTP/profile/lower-transport. The default value for the "lower-transport" parameters is specific to the profile. For RTP/AVP, the default is UDP. There are two different methods for how to specify where the media should be delivered for unicast transport: Schulzrinne, et al. Expires March 15, 2014 [Page 178] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 dest_addr: The presence of this parameter and its values indicates the destination address or addresses (host address and port pairs for IP flows) necessary for the media transport. No dest_addr: The lack of the dest_addr parameter indicates that the server MUST send media to the same address from which the RTSP messages originates. The choice of method for indicating where the media is to be delivered depends on the use case. In some cases the only allowed method will be to use no explicit address indication and have the server deliver media to the source of the RTSP messages. For Multicast there is several methods for specifying addresses but they are different in how they work compared with unicast: dest_addr with client picked address: The address and relevant parameters, like TTL (scope), for the actual multicast group to deliver the media to. There are security implications (Section 21) with this method that need to be addressed if using this method because a RTSP server can be used as a Denial of Service (DoS) attacker on an existing multicast group. dest_addr using Session Description Information: The information included in the transport header can all be coming from the session description, e.g., the SDP c= and m= line. This mitigates some of the security issues of the previous methods as it is the session provider that picks the multicast group and scope. The client MUST include the information if it is available in the session description. No dest_addr: The behavior when no explicit multicast group is present in a request is not defined. An RTSP proxy will need to take care. If the media is not desired to be routed through the proxy, the proxy will need to introduce the destination indication. Below are the configuration parameters associated with transport: General parameters: unicast / multicast: This parameter is a mutually exclusive indication of whether unicast or multicast delivery will be attempted. One of the two values MUST be specified. Clients that are capable of handling both unicast and multicast transmission need to indicate such capability by including two full transport-specs with separate parameters for each. Schulzrinne, et al. Expires March 15, 2014 [Page 179] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 layers: The number of multicast layers to be used for this media stream. The layers are sent to consecutive addresses starting at the dest_addr address. If the parameter is not included, it defaults to a single layer. dest_addr: A general destination address parameter that can contain one or more address specifications. Each combination of protocol/profile/lower transport needs to have the format and interpretation of its address specification defined. For RTP/ AVP/UDP and RTP/AVP/TCP, the address specification is a tuple containing a host address and port. Note, only a single destination parameter per transport spec is intended. The usage of multiple destinations to distribute a single media to multiple entities is unspecified. The client originating the RTSP request MAY specify the destination address of the stream recipient with the host address part of the tuple. When the destination address is specified, the recipient may be a different party than the originator of the request. To avoid becoming the unwitting perpetrator of a remote-controlled denial-of-service attack, a server MUST perform security checks (see Section 21.2.1) and SHOULD log such attempts before allowing the client to direct a media stream to a recipient address not chosen by the server. Implementations cannot rely on TCP as reliable means of client identification. If the server does not allow the host address part of the tuple to be set, it MUST return 463 (Destination Prohibited). The host address part of the tuple MAY be empty, for example ":58044", in cases when it is desired to specify only the destination port. Responses to requests including the Transport header with a dest_addr parameter SHOULD include the full destination address that is actually used by the server. The server MUST NOT remove address information present already in the request when responding unless the protocol requires it. src_addr: A general source address parameter that can contain one or more address specifications. Each combination of protocol/ profile/lower transport needs to have the format and interpretation of its address specification defined. For RTP/ AVP/UDP and RTP/AVP/TCP, the address specification is a tuple containing a host address and port. This parameter MUST be specified by the server if it transmits media packets from another address than the one RTSP messages are sent to. This will allow the client to verify source address and give it a destination address for its RTCP feedback Schulzrinne, et al. Expires March 15, 2014 [Page 180] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 packets, if RTP is used. The address or addresses indicated in the src_addr parameter SHOULD be used both for sending and receiving of the media stream's data packets. The main reasons are threefold: First, indicating the port and source address(s) lets the receiver know where from the packets is expected to originate. Secondly, traversal of NATs is greatly simplified when traffic is flowing symmetrically over a NAT binding. Thirdly, certain NAT traversal mechanisms, needs to know to which address and port to send so called "binding packets" from the receiver to the sender, thus creating an address binding in the NAT that the sender to receiver packet flow can use. This information may also be available through SDP. However, since this is more a feature of transport than media initialization, the authoritative source for this information should be in the SETUP response. mode: The mode parameter indicates the methods to be supported for this session. Currently defined valid values are "PLAY". If not provided, the default is "PLAY". The "RECORD" value was defined in RFC 2326 and is in this specification unspecified but reserved. RECORD and other values may be specified in the future. interleaved: The interleaved parameter implies mixing the media stream with the control stream in whatever protocol is being used by the control stream, using the mechanism defined in Section 14. The argument provides the channel number to be used in the $ block (see Section 14) and MUST be present. This parameter MAY be specified as an interval, e.g., interleaved=4-5 in cases where the transport choice for the media stream requires it, e.g., for RTP with RTCP. The channel number given in the request is only a guidance from the client to the server on what channel number(s) to use. The server MAY set any valid channel number in the response. The declared channel(s) are bi-directional, so both end-parties MAY send data on the given channel. One example of such usage is the second channel used for RTCP, where both server and client send RTCP packets on the same channel. This allows RTP/RTCP to be handled similarly to the way that it is done with UDP, i.e., one channel for RTP and the other for RTCP. Schulzrinne, et al. Expires March 15, 2014 [Page 181] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 MIKEY: This parameter is used in conjunction with transport specifications that can utilize MIKEY [RFC3830] for security context establishment. So far only the SRTP based RTP profiles SAVP and SAVPF can utilize MIKEY and this is defined in Appendix C.1.4.1. This parameter can be included both in request and response messages. The binary MIKEY message SHALL be Base64 [RFC4648] encoded before being included in the value part of the parameter. Multicast-specific: ttl: multicast time-to-live for IPv4. When included in requests the value indicate the TTL value that the client requests the server to use. In a response, the value actually being used by the server is returned. A server will need to consider what values that are reasonable and also the authority of the user to set this value. Corresponding functions are not needed for IPv6 as the scoping is part of the IPv6 multicast address [RFC4291]. RTP-specific: These parameters MAY only be used if the media transport protocol is RTP. ssrc: The ssrc parameter, if included in a SETUP response, indicates the RTP SSRC [RFC3550] value(s) that will be used by the media server for RTP packets within the stream. It is expressed as an eight digit hexadecimal value. The ssrc parameter MUST NOT be specified in requests. The functionality of specifying the ssrc parameter in a SETUP request is deprecated as it is incompatible with the specification of RTP in RFC 3550[RFC3550]. If the parameter is included in the Transport header of a SETUP request, the server SHOULD ignore it, and choose appropriate SSRCs for the stream. The server SHOULD set the ssrc parameter in the Transport header of the response. RTCP-mux: Use to negotiate the usage of RTP and RTCP multiplexing [RFC5761] on a single underlying transport stream / flow. The presence of this parameter in a SETUP request indicates the client's support and requires the server to use RTP and RTCP multiplexing. The client SHALL only include one transport stream in the Transport header specification. To provide the server with a choice between using RTP/RTCP multiplexing or not, two different transport header specifications must be included. Schulzrinne, et al. Expires March 15, 2014 [Page 182] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 The parameters setup and connection defined below MAY only be used if the media transport protocol of the lower-level transport is connection-oriented (such as TCP). However, these parameters MUST NOT be used when interleaving data over the RTSP connection. setup: Clients use the setup parameter on the Transport line in a SETUP request, to indicate the roles it wishes to play in a TCP connection. This parameter is adapted from [RFC4145]. 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 connection parameter in a transport specification part of the Transport header in a SETUP request, to indicate the client's preference for either reusing an existing connection between client and server (in which case the client sets the "connection" parameter to "existing"), or requesting the creation of a new connection between client and server (in which cast the client sets the "connection" parameter to "new"). Typically, clients use the "new" value Schulzrinne, et al. Expires March 15, 2014 [Page 183] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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, clock User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 302 Date: Thu, 23 Jan 1997 15:35:06 GMT Session: 47112344 Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/ "192.0.2.5:3457";src_addr="192.0.2.224:6256"/ "192.0.2.224:6257";mode="PLAY" Accept-Ranges: npt Media-Properties: Random-Access=0.6, Dynamic, Time-Limited=20081128T165900 18.55. Unsupported The Unsupported response-header lists the features not supported by the responding RTSP agent. In the case where the feature was specified via the Proxy-Require field (Section 18.37), if there is a Schulzrinne, et al. Expires March 15, 2014 [Page 184] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 proxy on the path between the client and the server, the proxy MUST send a response message with a status code of 551 (Option Not Supported). The request MUST NOT be forwarded. See Section 18.43 for a usage example. 18.56. User-Agent The User-Agent general-header field contains information about the user agent originating the request or producing a response. This is for statistical purposes, the tracing of protocol violations, and automated recognition of user agents for the sake of tailoring responses to avoid particular user agent limitations. User agents SHOULD include this field with requests. The field can contain multiple product tokens and comments identifying the agent and any subproducts which form a significant part of the user agent. By convention, the product tokens are listed in order of their significance for identifying the application. Example: User-Agent: PhonyClient/1.2 18.57. Via The Via general-header field MUST be used by proxies to indicate the intermediate protocols and recipients between the user agent and the server on requests, and between the origin server and the client on responses. The field is intended to be used for tracking message forwards, avoiding request loops, and identifying the protocol capabilities of all senders along the request/response chain. Multiple Via field values represents each proxy that has forwarded the message. Each recipient MUST append its information such that the end result is ordered according to the sequence of forwarding applications. Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by default, forward the names and ports of hosts within the private/ protected region. This information SHOULD only be propagated if explicitly enabled. If not enabled, the via-received of any host behind the firewall/NAT SHOULD be replaced by an appropriate pseudonym for that host. For organizations that have strong privacy requirements for hiding internal structures, a proxy MAY combine an ordered subsequence of Via header field entries with identical sent-protocol values into a single such entry. Applications MUST NOT combine entries which have Schulzrinne, et al. Expires March 15, 2014 [Page 185] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 different received-protocol values. 18.58. WWW-Authenticate The WWW-Authenticate response-header field MUST be included in 401 (Unauthorized) response messages. The field value consists of at least one challenge that indicates the authentication scheme(s) and parameters applicable to the Request-URI. This header MUST only be used in response messages related to client to server requests. The HTTP access authentication process is described in [RFC2617] with some clarification in Section 19.1. User agents are advised to take special care in parsing the WWW- Authenticate field value as it might contain more than one challenge, or if more than one WWW-Authenticate header field is provided, the contents of a challenge itself can contain a comma-separated list of authentication parameters. Schulzrinne, et al. Expires March 15, 2014 [Page 186] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 19. Security Framework The RTSP security framework consists of two high level components: the pure authentication mechanisms based on HTTP authentication, and the message transport protection based on TLS, which is independent of RTSP. Because of the similarity in syntax and usage between RTSP servers and HTTP servers, the security for HTTP is re-used to a large extent. 19.1. RTSP and HTTP Authentication RTSP and HTTP share common authentication schemes, and thus follow the same usage guidelines as specified in [RFC2617] with the additions for digest specified below in Section 19.1.1. Servers SHOULD implement both basic and digest [RFC2617] authentication. Clients MUST implement both basic and digest authentication [RFC2617] so that a server that requires the client to authenticate can trust that the capability is present. It should be stressed that using the HTTP authentication alone does not provide full control message security. Therefore, in environments requiring tighter security for the control messages, TLS SHOULD be used, see Section 19.2. Any RTSP message containing an Authorization header using basic authorization MUST be using a TLS connection with confidentiality protection enabled, i.e. no NULL encryption. In cases where there is a chain of proxies between the client and the server, each proxy may individually request the client or previous proxy to authenticate itself. This is done using the Proxy- Authenticate (Section 18.34), the Proxy-Authorization (Section 18.36) and the Proxy-Authentication-Info (Section 18.35) headers. These headers are hop-by-hop headers and have only scope for the current connection. Thus if a chain exist, the proxy connecting to another proxy will have to act as a client to authorize itself towards the next proxy. The WWW-Authenticate (Section 18.58), Authorization (Section 18.8) and Authentication-Info (Section 18.7) headers are end-to-end and must not be modified by proxies. This authentication mechanism works only for client to server requests as currently defined. This leaves server to client request outside of the context of TLS based communication more vulnerable to message injection attacks on the client. Based on the server to client methods that exist, the potential risks are various; hijacking (REDIRECT), denial of service (TEARDOWN and PLAY_NOTIFY) or attacks with uncertain results (SET_PARAMETER). Schulzrinne, et al. Expires March 15, 2014 [Page 187] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 19.1.1. Digest Authentication This section describes the modifications and clarifications required to apply the HTTP Digest authentication scheme to RTSP. The RTSP scheme usage is almost completely identical to that for HTTP [RFC2617]. These are based on the procedures defined for SIP 2.0 [RFC3261]. The rules for Digest authentication follow those defined in [RFC2617], with "HTTP/1.1" replaced by "RTSP/2.0" in addition to the following differences: 1. Use the ABNF specified in this document, rather than the one in [RFC2617]. That way the following is assured: * Using the right RTSP URIs allowed in the challenge as well as in the digest. * Resolved the error in the "uri" parameter of the Authorization header in [RFC2617]. 2. If MTags are used then the example procedure for choosing a nonce based on Etag can work based on replacing ETag with the MTag. 3. As a clarification to the calculation of the A2 value for message integrity assurance in the Digest authentication scheme, implementers should assume, when the entity-body is empty (that is, when the RTSP messages have no message body) that the hash of the message-body resolves to the MD5 hash of an empty string, or: H(entity-body) = MD5("") = "d41d8cd98f00b204e9800998ecf8427e". 4. RFC 2617 notes that a cnonce value MUST NOT be sent in an Authorization (and by extension Proxy-Authorization) header field if no qop directive has been sent. Therefore, any algorithms that have a dependency on the cnonce (including "MD5-Sess") require that the qop directive be sent. Use of the "qop" parameter is optional in RFC 2617 for the purposes of backwards compatibility with RFC 2069; since this specification defines RTSP 2.0 there is no backwards compatibility issue with mandating. Thus, all RTSP agents MUST implement qop-options. 19.2. RTSP over TLS RTSP agents MUST implement RTSP over TLS as defined in this section and the next Section 19.3. RTSP MUST follow the same guidelines with regards to TLS [RFC5246] usage as specified for HTTP, see [RFC2818]. RTSP over TLS is separated from unsecured RTSP both on the URI level and the port level. Instead of using the "rtsp" scheme identifier in Schulzrinne, et al. Expires March 15, 2014 [Page 188] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 the URI, the "rtsps" scheme identifier MUST be used to signal RTSP over TLS. If no port is given in a URI with the "rtsps" scheme, port 322 MUST be used for TLS over TCP/IP. When a client tries to setup an insecure channel to the server (using the "rtsp" URI), and the policy for the resource requires a secure channel, the server MUST redirect the client to the secure service by sending a 301 redirect response code together with the correct Location URI (using the "rtsps" scheme). A user or client MAY upgrade a non secured URI to a secured by changing the scheme from "rtsp" to "rtsps". A server implementing support for "rtsps" MUST allow this. It should be noted that TLS allows for mutual authentication (when using both server and client certificates). Still, one of the more common ways TLS is used is to only provide server side authentication (often to avoid client certificates). TLS is then used in addition to HTTP authentication, providing transport security and server authentication, while HTTP Authentication is used to authenticate the client. RTSP includes the possibility to keep a TCP session up between the client and server, throughout the RTSP session lifetime. It may be convenient to keep the TCP session, not only to save the extra setup time for TCP, but also the extra setup time for TLS (even if TLS uses the resume function, there will be almost two extra round trips). Still, when TLS is used, such behavior introduces extra active state in the server, not only for TCP and RTSP, but also for TLS. This may increase the vulnerability to DoS attacks. There exist a potential security vulnerability when reusing TCP and TLS state for different resources (URIs). If two different host names points at the same IP address it can be desirable to re-use the TCP/TLS connection to that server. In that case the RTSP agent having the TCP/TLS connection MUST verify that the server certificate associated with the connection has a SubjectAltName matching the host name present in the URI for the resource an RTSP request is to be issued for. In addition to these recommendations, Section 19.3 gives further recommendations of TLS usage with proxies. 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 Schulzrinne, et al. Expires March 15, 2014 [Page 189] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 agents. The TLS proxy mechanism allows for server and proxy identification using certificates. However, the client cannot be identified based on certificates. The client needs to select between using the procedure specified below or using a TLS connection directly (by-passing any proxies) to the server. The choice may be dependent on policies. There are in general two categories of proxies, the transparent proxies (of which the client is not aware) and the non-transparent proxies (of which the client is aware). This memo specifies only non-transparent RTSP proxies, i.e., proxies visible to the RTSP client and RTSP server. An infrastructure based on proxies requires that the trust model is such that both client and servers can trust the proxies to handle the RTSP messages correctly. To be able to trust a proxy, the client and server also need to be aware of the proxy. Hence, transparent proxies cannot generally be seen as trusted and will not work well with security (unless they work only at transport layer). In the rest of this section any reference to proxy will be to a non-transparent proxy, which inspects or manipulates the RTSP messages. HTTP Authentication is built on the assumption of proxies and can provide user-proxy authentication and proxy-proxy/server authentication in addition to the client-server authentication. When TLS is applied and a proxy is used, the client will connect to the proxy's address when connecting to any RTSP server. This implies that for TLS, the client will authenticate the proxy server and not the end server. Note that when the client checks the server certificate in TLS, it MUST check the proxy's identity (URI or possibly other known identity) against the proxy's identity as presented in the proxy's Certificate message. The problem is that for a proxy accepted by the client, the proxy needs to be provided information on which grounds it should accept the next-hop certificate. Both the proxy and the user may have rules for this, and the user should have the possibility to select the desired behavior. To handle this case, the Accept-Credentials header (See Section 18.2) is used, where the client can request the proxy/ proxies to relay back the chain of certificates used to authenticate any intermediate proxies as well as the server. The assumption that the proxies are viewed as trusted, gives the user a possibility to enforce policies to each trusted proxy of whether it should accept the next agent in the chain. However, it should be noted that not all deployments will return the chain of certificates used to authenticate any intermediate proxies as well as the server. An operator of such a deployment may want to hide its topology from the client. It should be noted well that the client does not have any Schulzrinne, et al. Expires March 15, 2014 [Page 190] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 insight into the proxy's operation. Even if the proxy is trusted, it can still return an incomplete chain of certificates. A proxy MUST use TLS for the next hop if the RTSP request includes a "rtsps" URI. TLS MAY be applied on intermediate links (e.g., between client and proxy, or between proxy and proxy), even if the resource and the end server are not required to use it. The proxy MUST, when initiating the next hop TLS connection, use the incoming TLS connections cipher suite list, only modified by removing any cipher suites that the proxy does not support. In case a proxy fails to establish a TLS connection due to cipher suite mismatch between proxy and next hop proxy or server, this is indicated using error code 472 (Failure to establish secure connection). 19.3.1. Accept-Credentials The Accept-Credentials header can be used by the client to distribute simple authorization policies to intermediate proxies. The client includes the Accept-Credentials header to dictate how the proxy treats the server/next proxy certificate. There are currently three methods defined: Any, which means that the proxy (or proxies) MUST accept whatever certificate is presented. This is of course not a recommended option to use, but may be useful in certain circumstances (such as testing). Proxy: which means that the proxy (or proxies) MUST use its own policies to validate the certificate and decide whether to accept it or not. This is convenient in cases where the user has a strong trust relation with the proxy. Reasons why a strong trust relation may exist are: personal/company proxy, proxy has a out-of-band policy configuration mechanism. User, which means that the proxy (or proxies) MUST send credential information about the next hop to the client for authorization. The client can then decide whether the proxy should accept the certificate or not. See Section 19.3.2 for further details. If the Accept-Credentials header is not included in the RTSP request from the client, then the "Proxy" method MUST be used as default. If another method than the "Proxy" is to be used, then the Accept- Credentials header MUST be included in all of the RTSP requests from the client. This is because it cannot be assumed that the proxy always keeps the TLS state or the user's previous preference between different RTSP messages (in particular if the time interval between the messages is long). Schulzrinne, et al. Expires March 15, 2014 [Page 191] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 With the "Any" and "Proxy" methods the proxy will apply the policy as defined for each method. If the policy does not accept the credentials of the next hop, the proxy MUST respond with a message using status code 471 (Connection Credentials not accepted). An RTSP request in the direction server to client MUST NOT include the Accept-Credentials header. As for the non-secured communication, the possibility for these requests depends on the presence of a client established connection. However, if the server to client request is in relation to a session established over a TLS secured channel, it MUST be sent in a TLS secured connection. That secured connection MUST also be the one used by the last client to server request. If no such transport connection exists at the time when the server desires to send the request, the server MUST discard the message. Further policies MAY be defined and registered, but should be done so with caution. 19.3.2. User approved TLS procedure For the "User" method, each proxy MUST perform the following procedure for each RTSP request: o Setup the TLS session to the next hop if not already present (i.e., run the TLS handshake, but do not send the RTSP request). o Extract the peer certificate chain for the TLS session. o Check if a matching identity and hash of the peer certificate is present in the Accept-Credentials header. If present, send the message to the next hop, and conclude these procedures. If not, go to the next step. o The proxy responds to the RTSP request with a 470 or 407 response code. The 407 response code MAY be used when the proxy requires both user and connection authorization from user or client. In this message the proxy MUST include a Connection-Credentials header, see Section 18.13 with the next hop's identity and certificate. The client MUST upon receiving a 470 or 407 response with Connection- Credentials header take the decision on whether to accept the certificate or not (if it cannot do so, the user SHOULD be consulted). 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. Schulzrinne, et al. Expires March 15, 2014 [Page 192] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Example: C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0 CSeq: 2 Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/ "192.0.2.5:4589" Accept-Ranges: npt, smpte, clock Accept-Credentials: User P->C: RTSP/2.0 470 Connection Authorization Required CSeq: 2 Connection-Credentials: "rtsps://test.example.org"; MIIDNTCCAp... C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0 CSeq: 3 Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/ "192.0.2.5:4589" Accept-Credentials: User "rtsps://test.example.org";sha-256; dPYD7txpoGTbAqZZQJ+vaeOkyH4= Accept-Ranges: npt, smpte, clock P->S: SETUP rtsps://test.example.org/secret/audio RTSP/2.0 CSeq: 3 Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/ "192.0.2.5:4589" Via: RTSP/2.0 proxy.example.org Accept-Credentials: User "rtsps://test.example.org";sha-256; dPYD7txpoGTbAqZZQJ+vaeOkyH4= Accept-Ranges: npt, smpte, clock One implication of this process is that the connection for secured RTSP messages may take significantly more round-trip times for the first message. A complete extra message exchange between the proxy connecting to the next hop and the client results because of the process for approval for each hop. However, if each message contains the chain of proxies that the requester accepts, the remaining message exchange should not be delayed. The procedure of including the credentials in each request rather than building state in each proxy, avoids the need for revocation procedures. Schulzrinne, et al. Expires March 15, 2014 [Page 193] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 20. Syntax The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF) as defined in RFC 5234 [RFC5234]. It uses the basic definitions present in RFC 5234. Please note that ABNF strings, e.g., "Accept", are case insensitive as specified in section 2.3 of RFC 5234. The RTSP syntax makes use of the ISO 10646 character set in UTF-8 encoding RFC 3629 [RFC3629]. 20.1. Base Syntax RTSP header values can be folded onto multiple lines if the continuation line begins with a space or horizontal tab. All linear white space, including folding, has the same semantics as SP. A recipient MAY replace any linear white space with a single SP before interpreting the field value or forwarding the message downstream. This is intended to behave exactly as HTTP/1.1 as described in RFC 2616 [RFC2616]. The SWS construct is used when linear white space is optional, generally between tokens and separators. To separate the header name from the rest of value, a colon is used, which, by the above rule, allows whitespace before, but no line break, and whitespace after, including a line break. The HCOLON defines this construct. OCTET = %x00-FF ; any 8-bit sequence of data CHAR = %x01-7F ; any US-ASCII character (octets 1 - 127) UPALPHA = %x41-5A ; any US-ASCII uppercase letter "A".."Z" LOALPHA = %x61-7A ;any US-ASCII lowercase letter "a".."z" ALPHA = UPALPHA / LOALPHA DIGIT = %x30-39 ; any US-ASCII digit "0".."9" CTL = %x00-1F / %x7F ; any US-ASCII control character ; (octets 0 - 31) and DEL (127) CR = %x0D ; US-ASCII CR, carriage return (13) LF = %x0A ; US-ASCII LF, linefeed (10) SP = %x20 ; US-ASCII SP, space (32) HT = %x09 ; US-ASCII HT, horizontal-tab (9) BACKSLASH = %x5C ; US-ASCII backslash (92) CRLF = CR LF Schulzrinne, et al. Expires March 15, 2014 [Page 194] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 LWS = [CRLF] 1*( SP / HT ) ; Line-breaking White Space SWS = [LWS] ; Separating White Space HCOLON = *( SP / HT ) ":" SWS TEXT = %x20-7E / %x80-FF ; any OCTET except CTLs tspecials = "(" / ")" / "<" / ">" / "@" / "," / ";" / ":" / BACKSLASH / DQUOTE / "/" / "[" / "]" / "?" / "=" / "{" / "}" / SP / HT token = 1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39 / %x41-5A / %x5E-7A / %x7C / %x7E) ; 1* quoted-string = ( DQUOTE *qdtext DQUOTE ) qdtext = %x20-21 / %x23-5B / %x5D-7E / quoted-pair / UTF8-NONASCII ; No DQUOTE and no "\" quoted-pair = "\\" / ( "\" DQUOTE ) ctext = %x20-27 / %x2A-7E / %x80-FF ; any OCTET except CTLs, "(" and ")" generic-param = token [ EQUAL gen-value ] gen-value = token / host / quoted-string safe = "$" / "-" / "_" / "." / "+" extra = "!" / "*" / "'" / "(" / ")" / "," rtsp-extra = "!" / "*" / "'" / "(" / ")" HEX = DIGIT / "A" / "B" / "C" / "D" / "E" / "F" / "a" / "b" / "c" / "d" / "e" / "f" LHEX = DIGIT / "a" / "b" / "c" / "d" / "e" / "f" ; lowercase "a-f" Hex reserved = ";" / "/" / "?" / ":" / "@" / "&" / "=" unreserved = ALPHA / DIGIT / safe / extra rtsp-unreserved = ALPHA / DIGIT / safe / rtsp-extra base64 = *base64-unit [base64-pad] base64-unit = 4base64-char base64-pad = (2base64-char "==") / (3base64-char "=") base64-char = ALPHA / DIGIT / "+" / "/" Schulzrinne, et al. Expires March 15, 2014 [Page 195] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 SLASH = SWS "/" SWS ; slash EQUAL = SWS "=" SWS ; equal LPAREN = SWS "(" SWS ; left parenthesis RPAREN = SWS ")" SWS ; right parenthesis COMMA = SWS "," SWS ; comma SEMI = SWS ";" SWS ; semicolon COLON = SWS ":" SWS ; colon MINUS = SWS "-" SWS ; minus/dash LDQUOT = SWS DQUOTE ; open double quotation mark RDQUOT = DQUOTE SWS ; close double quotation mark RAQUOT = ">" SWS ; right angle quote LAQUOT = SWS "<" ; left angle quote TEXT-UTF8char = %x21-7E / UTF8-NONASCII UTF8-NONASCII = UTF8-1 / UTF8-2 / UTF8-3 / UTF8-4 UTF8-1 = UTF8-2 = UTF8-3 = UTF8-4 = UTF8-CONT = %x80-BF POS-FLOAT = 1*12DIGIT ["." 1*9DIGIT] FLOAT = ["-"] POS-FLOAT 20.2. RTSP Protocol Definition 20.2.1. Generic Protocol elements Schulzrinne, et al. Expires March 15, 2014 [Page 196] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTSP-IRI = schemes ":" IRI-rest IRI-rest = ihier-part [ "?" iquery ] ihier-part = "//" iauthority ipath-abempty RTSP-IRI-ref = RTSP-IRI / irelative-ref irelative-ref = irelative-part [ "?" iquery ] irelative-part = "//" iauthority ipath-abempty / ipath-absolute / ipath-noscheme / ipath-empty iauthority = < As defined in RFC 3987> ipath = ipath-abempty ; begins with "/" or is empty / ipath-absolute ; begins with "/" but not "//" / ipath-noscheme ; begins with a non-colon segment / ipath-rootless ; begins with a segment / ipath-empty ; zero characters ipath-abempty = *( "/" isegment ) ipath-absolute = "/" [ isegment-nz *( "/" isegment ) ] ipath-noscheme = isegment-nz-nc *( "/" isegment ) ipath-rootless = isegment-nz *( "/" isegment ) ipath-empty = 0 isegment = *ipchar [";" *ipchar] isegment-nz = 1*ipchar [";" *ipchar] / ";" *ipchar isegment-nz-nc = (1*ipchar-nc [";" *ipchar-nc]) / ";" *ipchar-nc ; non-zero-length segment without any colon ":" ; No parameter (; delimited) inside path. ipchar = iunreserved / pct-encoded / sub-delims / ":" / "@" ipchar-nc = iunreserved / pct-encoded / sub-delims / "@" ; sub-delims is different from RFC 3987 ; not including ";" iquery = < As defined in RFC 3987> iunreserved = < As defined in RFC 3987> pct-encoded = < As defined in RFC 3987> Schulzrinne, et al. Expires March 15, 2014 [Page 197] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTSP-URI = schemes ":" URI-rest RTSP-REQ-URI = schemes ":" URI-req-rest RTSP-URI-Ref = RTSP-URI / RTSP-Relative RTSP-REQ-Ref = RTSP-REQ-URI / RTSP-REQ-Rel schemes = "rtsp" / "rtsps" / scheme scheme = < As defined in RFC 3986> URI-rest = hier-part [ "?" query ] URI-req-rest = hier-part [ "?" query ] ; Note fragment part not allowed in requests hier-part = "//" authority path-abempty RTSP-Relative = relative-part [ "?" query ] RTSP-REQ-Rel = relative-part [ "?" query ] relative-part = "//" authority path-abempty / path-absolute / path-noscheme / path-empty authority = < As defined in RFC 3986> query = < As defined in RFC 3986> path = path-abempty ; begins with "/" or is empty / path-absolute ; begins with "/" but not "//" / path-noscheme ; begins with a non-colon segment / path-rootless ; begins with a segment / path-empty ; zero characters path-abempty = *( "/" segment ) path-absolute = "/" [ segment-nz *( "/" segment ) ] path-noscheme = segment-nz-nc *( "/" segment ) path-rootless = segment-nz *( "/" segment ) path-empty = 0 segment = *pchar [";" *pchar] segment-nz = ( 1*pchar [";" *pchar]) / (";" *pchar) segment-nz-nc = ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc) ; non-zero-length segment without any colon ":" ; No parameter (; delimited) inside path. pchar = unreserved / pct-encoded / sub-delims / ":" / "@" pchar-nc = unreserved / pct-encoded / sub-delims / "@" sub-delims = "!" / "$" / "&" / "'" / "(" / ")" / "*" / "+" / "," / "=" ; sub-delims is different from RFC 3986/3987 ; not including ";" Schulzrinne, et al. Expires March 15, 2014 [Page 198] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 smpte-range = smpte-type [EQUAL smpte-range-spec] ; See section 4.4 smpte-range-spec = ( smpte-time "-" [ smpte-time ] ) / ( "-" smpte-time ) smpte-type = "smpte" / "smpte-30-drop" / "smpte-25" / smpte-type-extension ; other timecodes may be added smpte-type-extension = "smpte" token smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ] npt-range = "npt" [EQUAL npt-range-spec] npt-range-spec = ( npt-time "-" [ npt-time ] ) / ( "-" npt-time ) npt-time = "now" / npt-sec / npt-hhmmss npt-sec = 1*19DIGIT [ "." 1*9DIGIT ] npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." 1*9DIGIT ] npt-hh = 1*19DIGIT ; any positive number npt-mm = 1*2DIGIT ; 0-59 npt-ss = 1*2DIGIT ; 0-59 utc-range = "clock" [EQUAL utc-range-spec] utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time ) utc-time = utc-date "T" utc-clock "Z" utc-date = 8DIGIT utc-clock = 6DIGIT [ "." 1*9DIGIT ] feature-tag = token session-id = 1*256( ALPHA / DIGIT / safe ) extension-header = header-name HCOLON header-value header-name = token header-value = *(TEXT-UTF8char / UTF8-CONT / LWS) 20.2.2. Message Syntax Schulzrinne, et al. Expires March 15, 2014 [Page 199] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTSP-message = Request / Response ; RTSP/2.0 messages Request = Request-Line *((general-header / request-header / message-header) CRLF) CRLF [ message-body-data ] Response = Status-Line *((general-header / response-header / message-header) CRLF) CRLF [ message-body-data ] Request-Line = Method SP Request-URI SP RTSP-Version CRLF Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF Method = "DESCRIBE" / "GET_PARAMETER" / "OPTIONS" / "PAUSE" / "PLAY" / "PLAY_NOTIFY" / "REDIRECT" / "SETUP" / "SET_PARAMETER" / "TEARDOWN" / extension-method extension-method = token Request-URI = "*" / RTSP-REQ-URI RTSP-Version = "RTSP/" 1*DIGIT "." 1*DIGIT message-body-data = 1*OCTET Status-Code = "100" ; Continue / "200" ; OK / "301" ; Moved Permanently / "302" ; Found / "303" ; See Other / "304" ; Not Modified / "305" ; Use Proxy / "400" ; Bad Request / "401" ; Unauthorized / "402" ; Payment Required Schulzrinne, et al. Expires March 15, 2014 [Page 200] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 / "403" ; Forbidden / "404" ; Not Found / "405" ; Method Not Allowed / "406" ; Not Acceptable / "407" ; Proxy Authentication Required / "408" ; Request Time-out / "410" ; Gone / "412" ; Precondition Failed / "413" ; Request Message Body Too Large / "414" ; Request-URI Too Large / "415" ; Unsupported Media Type / "451" ; Parameter Not Understood / "452" ; reserved / "453" ; Not Enough Bandwidth / "454" ; Session Not Found / "455" ; Method Not Valid in This State / "456" ; Header Field Not Valid for Resource / "457" ; Invalid Range / "458" ; Parameter Is Read-Only / "459" ; Aggregate operation not allowed / "460" ; Only aggregate operation allowed / "461" ; Unsupported Transport / "462" ; Destination Unreachable / "463" ; Destination Prohibited / "464" ; Data Transport Not Ready Yet / "465" ; Notification Reason Unknown / "466" ; Key Management Error / "470" ; Connection Authorization Required / "471" ; Connection Credentials not accepted / "472" ; Failure to establish secure connection / "500" ; Internal Server Error / "501" ; Not Implemented / "502" ; Bad Gateway / "503" ; Service Unavailable / "504" ; Gateway Time-out / "505" ; RTSP Version not supported / "551" ; Option not supported / extension-code extension-code = 3DIGIT Reason-Phrase = 1*(TEXT-UTF8char / HT / SP) Schulzrinne, et al. Expires March 15, 2014 [Page 201] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 general-header = Accept-Ranges / Cache-Control / Connection / CSeq / Date / Media-Properties / Media-Range / Pipelined-Requests / Proxy-Supported / Range / RTP-Info / Scale / Seek-Style / Server / Session / Speed / Supported / Timestamp / Transport / User-Agent / Via / extension-header request-header = Accept / Accept-Credentials / Accept-Encoding / Accept-Language / Authorization / Bandwidth / Blocksize / From / If-Match / If-Modified-Since / If-None-Match / Notify-Reason / Proxy-Authorization / Proxy-Require / Referrer / Request-Status / Require / Terminate-Reason / extension-header Schulzrinne, et al. Expires March 15, 2014 [Page 202] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 response-header = Authentication-Info / Connection-Credentials / Location / MTag / Proxy-Authenticate / Proxy-Authentication-Info / Public / Retry-After / Unsupported / WWW-Authenticate / extension-header message-header = Allow / Content-Base / Content-Encoding / Content-Language / Content-Length / Content-Location / Content-Type / Expires / Last-Modified / extension-header 20.2.3. Header Syntax Accept = "Accept" HCOLON [ accept-range *(COMMA accept-range) ] accept-range = media-type-range [SEMI accept-params] media-type-range = ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH m-subtype ) ) *( SEMI m-parameter ) accept-params = "q" EQUAL qvalue *(SEMI generic-param ) qvalue = ( "0" [ "." *3DIGIT ] ) / ( "1" [ "." *3("0") ] ) Accept-Credentials = "Accept-Credentials" HCOLON cred-decision cred-decision = ("User" [LWS cred-info]) / "Proxy" / "Any" / (token [LWS 1*header-value]) ; For future extensions cred-info = cred-info-data *(COMMA cred-info-data) cred-info-data = DQUOTE RTSP-REQ-URI DQUOTE SEMI hash-alg SEMI base64 hash-alg = "sha-256" / extension-alg extension-alg = token Schulzrinne, et al. Expires March 15, 2014 [Page 203] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Accept-Encoding = "Accept-Encoding" HCOLON [ encoding *(COMMA encoding) ] encoding = codings [SEMI accept-params] codings = content-coding / "*" content-coding = "identity" / token Accept-Language = "Accept-Language" HCOLON language *(COMMA language) language = language-range [SEMI accept-params] language-range = language-tag / "*" language-tag = primary-tag *( "-" subtag ) primary-tag = 1*8ALPHA subtag = 1*8ALPHA Accept-Ranges = "Accept-Ranges" HCOLON acceptable-ranges acceptable-ranges = (range-unit *(COMMA range-unit)) range-unit = "npt" / "smpte" / "smpte-30-drop" / "smpte-25" / "clock" / extension-format extension-format = token Allow = "Allow" HCOLON Method *(COMMA Method) Authentication-Info = "Authentication-Info" HCOLON auth-info auth-info = auth-info-entry *(COMMA auth-info-entry) auth-info-entry = nextnonce / message-qop / response-auth / cnonce / nonce-count nextnonce = "nextnonce" EQUAL nonce-value response-auth = "rspauth" EQUAL response-digest response-digest = DQUOTE *LHEX DQUOTE Authorization = "Authorization" HCOLON credentials credentials = basic-credential / digest-credential / other-response basic-credential = "Basic" LWS basic-credentials basic-credentials = base64 ; Base64 encoding of user-password user-password = basic-username ":" password basic-username = *CF-TEXT CF-TEXT = %x20-39 / %x3B-7E / %x80-FF ; TEXT without : password = *TEXT digest-credential = ("Digest" LWS digest-response) digest-response = dig-resp *(COMMA dig-resp) dig-resp = username / realm / nonce / digest-uri / dresponse / algorithm / cnonce / opaque / message-qop / nonce-count / auth-param username = "username" EQUAL username-value username-value = quoted-string digest-uri = "uri" EQUAL LDQUOT digest-uri-value RDQUOT digest-uri-value = RTSP-REQ-URI Schulzrinne, et al. Expires March 15, 2014 [Page 204] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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*19DIGIT Blocksize = "Blocksize" HCOLON 1*9DIGIT Cache-Control = "Cache-Control" HCOLON cache-directive *(COMMA cache-directive) cache-directive = cache-rqst-directive / cache-rspns-directive cache-rqst-directive = "no-cache" / "max-stale" [EQUAL delta-seconds] / "min-fresh" EQUAL delta-seconds / "only-if-cached" / cache-extension cache-rspns-directive = "public" / "private" / "no-cache" / "no-transform" / "must-revalidate" / "proxy-revalidate" / "max-age" EQUAL delta-seconds / cache-extension cache-extension = token [EQUAL (token / quoted-string)] delta-seconds = 1*19DIGIT Connection = "Connection" HCOLON connection-token *(COMMA connection-token) connection-token = "close" / token Connection-Credentials = "Connection-Credentials" HCOLON cred-chain cred-chain = DQUOTE RTSP-REQ-URI DQUOTE SEMI base64 Schulzrinne, et al. Expires March 15, 2014 [Page 205] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Content-Base = "Content-Base" HCOLON RTSP-URI Content-Encoding = "Content-Encoding" HCOLON content-coding *(COMMA content-coding) Content-Language = "Content-Language" HCOLON language-tag *(COMMA language-tag) Content-Length = "Content-Length" HCOLON 1*19DIGIT Content-Location = "Content-Location" HCOLON RTSP-REQ-Ref Content-Type = "Content-Type" HCOLON media-type media-type = m-type SLASH m-subtype *(SEMI m-parameter) m-type = discrete-type / composite-type discrete-type = "text" / "image" / "audio" / "video" / "application" / extension-token composite-type = "message" / "multipart" / extension-token extension-token = ietf-token / x-token ietf-token = token x-token = "x-" token m-subtype = extension-token / iana-token iana-token = token m-parameter = m-attribute EQUAL m-value m-attribute = token m-value = token / quoted-string CSeq = "CSeq" HCOLON cseq-nr cseq-nr = 1*9DIGIT Date = "Date" HCOLON RTSP-date RTSP-date = date-time ; date-time = Expires = "Expires" HCOLON RTSP-date From = "From" HCOLON from-spec from-spec = ( name-addr / addr-spec ) *( SEMI from-param ) name-addr = [ display-name ] LAQUOT addr-spec RAQUOT addr-spec = RTSP-REQ-URI / absolute-URI absolute-URI = < As defined in RFC 3986> display-name = *(token LWS) / quoted-string from-param = tag-param / generic-param tag-param = "tag" EQUAL token If-Match = "If-Match" HCOLON ("*" / message-tag-list) message-tag-list = message-tag *(COMMA message-tag) message-tag = [ weak ] opaque-tag weak = "W/" opaque-tag = quoted-string If-Modified-Since = "If-Modified-Since" HCOLON RTSP-date If-None-Match = "If-None-Match" HCOLON ("*" / message-tag-list) Last-Modified = "Last-Modified" HCOLON RTSP-date Location = "Location" HCOLON RTSP-REQ-URI Media-Properties = "Media-Properties" HCOLON [media-prop-list] media-prop-list = media-prop-value *(COMMA media-prop-value) media-prop-value = ("Random-Access" [EQUAL POS-FLOAT]) Schulzrinne, et al. Expires March 15, 2014 [Page 206] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 / "Beginning-Only" / "No-Seeking" / "Immutable" / "Dynamic" / "Time-Progressing" / "Unlimited" / ("Time-Limited" EQUAL utc-time) / ("Time-Duration" EQUAL POS-FLOAT) / ("Scales" EQUAL scale-value-list) / media-prop-ext media-prop-ext = token [EQUAL (1*rtsp-unreserved / quoted-string)] scale-value-list = DQUOTE scale-entry *(COMMA scale-entry) DQUOTE scale-entry = scale-value / (scale-value COLON scale-value) scale-value = FLOAT Media-Range = "Media-Range" HCOLON [ranges-list] ranges-list = ranges-spec *(COMMA ranges-spec) MTag = "MTag" HCOLON message-tag Notify-Reason = "Notify-Reason" HCOLON Notify-Reas-val Notify-Reas-val = "end-of-stream" / "media-properties-update" / "scale-change" / Notify-Reason-extension Notify-Reason-extension = token Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id startup-id = 1*8DIGIT Schulzrinne, et al. Expires March 15, 2014 [Page 207] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Proxy-Authenticate = "Proxy-Authenticate" HCOLON challenge-list challenge-list = challenge *(COMMA challenge) challenge = ("Digest" LWS digest-cln *(COMMA digest-cln)) / ("Basic" LWS realm) / other-challenge other-challenge = auth-scheme LWS auth-param *(COMMA auth-param) digest-cln = realm / domain / nonce / opaque / stale / algorithm / qop-options / auth-param realm = "realm" EQUAL realm-value realm-value = quoted-string domain = "domain" EQUAL LDQUOT RTSP-REQ-Ref *(1*SP RTSP-REQ-Ref ) RDQUOT nonce = "nonce" EQUAL nonce-value nonce-value = quoted-string opaque = "opaque" EQUAL quoted-string stale = "stale" EQUAL ( "true" / "false" ) algorithm = "algorithm" EQUAL ("MD5" / "MD5-sess" / token) qop-options = "qop" EQUAL LDQUOT qop-value *("," qop-value) RDQUOT qop-value = "auth" / "auth-int" / token Proxy-Authentication-Info = "Proxy-Authentication-Info" HCOLON auth-info Proxy-Authorization = "Proxy-Authorization" HCOLON credentials Proxy-Require = "Proxy-Require" HCOLON feature-tag-list feature-tag-list = feature-tag *(COMMA feature-tag) Proxy-Supported = "Proxy-Supported" HCOLON [feature-tag-list] Public = "Public" HCOLON Method *(COMMA Method) Range = "Range" HCOLON ranges-spec ranges-spec = npt-range / utc-range / smpte-range / range-ext range-ext = extension-format [EQUAL range-value] range-value = 1*(rtsp-unreserved / quoted-string / ":" ) Referrer = "Referrer" HCOLON (absolute-URI / RTSP-URI-Ref) Request-Status = "Request-Status" HCOLON req-status-info req-status-info = cseq-info LWS status-info LWS reason-info cseq-info = "cseq" EQUAL cseq-nr status-info = "status" EQUAL Status-Code reason-info = "reason" EQUAL DQUOTE Reason-Phrase DQUOTE Require = "Require" HCOLON feature-tag-list Schulzrinne, et al. Expires March 15, 2014 [Page 208] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTP-Info = "RTP-Info" HCOLON [rtsp-info-spec *(COMMA rtsp-info-spec)] rtsp-info-spec = stream-url 1*ssrc-parameter stream-url = "url" EQUAL DQUOTE RTSP-REQ-Ref DQUOTE ssrc-parameter = LWS "ssrc" EQUAL ssrc HCOLON ri-parameter *(SEMI ri-parameter) ri-parameter = ("seq" EQUAL 1*5DIGIT) / ("rtptime" EQUAL 1*10DIGIT) / generic-param Retry-After = "Retry-After" HCOLON (RTSP-date / delta-seconds) Scale = "Scale" HCOLON scale-value Seek-Style = "Seek-Style" HCOLON Seek-S-values Seek-S-values = "RAP" / "CoRAP" / "First-Prior" / "Next" / Seek-S-value-ext Seek-S-value-ext = token Server = "Server" HCOLON ( product / comment ) *(LWS (product / comment)) product = token [SLASH product-version] product-version = token comment = LPAREN *( ctext / quoted-pair) RPAREN Session = "Session" HCOLON session-id [ SEMI "timeout" EQUAL delta-seconds ] Speed = "Speed" HCOLON lower-bound MINUS upper-bound lower-bound = POS-FLOAT upper-bound = POS-FLOAT Supported = "Supported" HCOLON [feature-tag-list] Schulzrinne, et al. Expires March 15, 2014 [Page 209] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Terminate-Reason = "Terminate-Reason" HCOLON TR-Info TR-Info = TR-Reason *(SEMI TR-Parameter) TR-Reason = "Session-Timeout" / "Server-Admin" / "Internal-Error" / token TR-Parameter = TR-time / TR-user-msg / generic-param TR-time = "time" EQUAL utc-time TR-user-msg = "user-msg" EQUAL quoted-string Timestamp = "Timestamp" HCOLON timestamp-value [LWS delay] timestamp-value = *19DIGIT [ "." *9DIGIT ] delay = *9DIGIT [ "." *9DIGIT ] Transport = "Transport" HCOLON transport-spec *(COMMA transport-spec) transport-spec = transport-id *trns-parameter transport-id = trans-id-rtp / other-trans trans-id-rtp = "RTP/" profile ["/" lower-transport] ; no LWS is allowed inside transport-id other-trans = token *("/" token) Schulzrinne, et al. Expires March 15, 2014 [Page 210] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 profile = "AVP" / "SAVP" / "AVPF" / "SAVPF" / token lower-transport = "TCP" / "UDP" / token trns-parameter = (SEMI ( "unicast" / "multicast" )) / (SEMI "interleaved" EQUAL channel ["-" channel]) / (SEMI "ttl" EQUAL ttl) / (SEMI "layers" EQUAL 1*DIGIT) / (SEMI "ssrc" EQUAL ssrc *(SLASH ssrc)) / (SEMI "mode" EQUAL mode-spec) / (SEMI "dest_addr" EQUAL addr-list) / (SEMI "src_addr" EQUAL addr-list) / (SEMI "setup" EQUAL contrans-setup) / (SEMI "connection" EQUAL contrans-con) / (SEMI "RTCP-mux") / (SEMI "MIKEY" EQUAL MIKEY-Value) / (SEMI trn-param-ext) contrans-setup = "active" / "passive" / "actpass" contrans-con = "new" / "existing" trn-param-ext = par-name [EQUAL trn-par-value] par-name = token trn-par-value = *(rtsp-unreserved / quoted-string) ttl = 1*3DIGIT ; 0 to 255 ssrc = 8HEX channel = 1*3DIGIT ; 0 to 255 MIKEY-Value = base64 mode-spec = ( DQUOTE mode *(COMMA mode) DQUOTE ) mode = "PLAY" / token addr-list = quoted-addr *(SLASH quoted-addr) quoted-addr = DQUOTE (host-port / extension-addr) DQUOTE host-port = ( host [":" port] ) / ( ":" port ) extension-addr = 1*qdtext host = < As defined in RFC 3986> port = < As defined in RFC 3986> Schulzrinne, et al. Expires March 15, 2014 [Page 211] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Unsupported = "Unsupported" HCOLON feature-tag-list User-Agent = "User-Agent" HCOLON ( product / comment ) *(LWS (product / comment)) Via = "Via" HCOLON via-parm *(COMMA via-parm) via-parm = sent-protocol LWS sent-by *( SEMI via-params ) via-params = via-ttl / via-maddr / via-received / via-extension via-ttl = "ttl" EQUAL ttl via-maddr = "maddr" EQUAL host via-received = "received" EQUAL (IPv4address / IPv6address) IPv4address = < As defined in RFC 3986> IPv6address = < As defined in RFC 3986> via-extension = generic-param sent-protocol = protocol-name SLASH protocol-version SLASH transport-prot protocol-name = "RTSP" / token protocol-version = token transport-prot = "UDP" / "TCP" / "TLS" / other-transport other-transport = token sent-by = host [ COLON port ] WWW-Authenticate = "WWW-Authenticate" HCOLON challenge-list 20.3. SDP extension Syntax This section defines in ABNF the SDP extensions defined for RTSP. See Appendix D for the definition of the extensions in text. control-attribute = "a=control:" *SP RTSP-REQ-Ref CRLF a-range-def = "a=range:" ranges-spec CRLF a-mtag-def = "a=mtag:" message-tag CRLF Schulzrinne, et al. Expires March 15, 2014 [Page 212] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 21. Security Considerations The security considerations and threats around RTSP and its usage can be divided into considerations around the signaling protocol itself and the issues related to the media stream delivery. However, when it comes to mitigations of security threats, a threat depending on the media stream delivery may in fact be mitigated by a mechanism in the signaling protocol. There are several chapters and an appendix in this document that define security solutions for the protocol. We will reference them when discussing the threats below. But the reader should take special notice of the Security Framework (Section 19) and the specification of how to use SRTP and its key-mangement (Appendix C.1.4) to achieve certain aspects of the media security. 21.1. Signaling Protocol Threats This section focuses on issues related to the signaling protocol. Because of the similarity in syntax and usage between RTSP servers and HTTP servers, the security considerations outlined in [H15] apply also. Specifically, please note the following: Abuse of Server Log Information: A server is in the position to save personal data about a user's requests which might identify their media consumption patterns or subjects of interest. This information is clearly confidential in nature and its handling can be constrained by law in certain countries. RTSP servers will presumably have similar logging mechanisms to HTTP, and thus should be equally guarded in protecting the contents of those logs, thus protecting the privacy of the users of the servers. People using the RTSP protocol to provide media are responsible for ensuring that logging material is not distributed without the permission of any individuals that are identifiable by the published results. Transfer of Sensitive Information: There is no reason to believe that information transferred 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. Schulzrinne, et al. Expires March 15, 2014 [Page 213] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Attacks Based On File and Path Names: Though RTSP URIs are opaque handles that do not necessarily have file system semantics, it is anticipated that many implementations will translate portions of the Request-URIs directly to file system calls. In such cases, file systems SHOULD follow the precautions outlined in [H15.2], such as checking for ".." in path components. Personal Information: RTSP clients are often privy to the same information that HTTP clients are (user name, location, etc.) and thus should be equally sensitive. See [H15.1] for further recommendations. Privacy Issues Connected to Accept Headers: Since 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. Location Headers and Spoofing: If a single server supports multiple organizations that do not trust each another, then it MUST 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 RFC 2068 [RFC2068], future HTTP specifications may provide additional guidance on security issues. The following are added considerations for RTSP implementations. Session hijacking: Since there is no or little relation between a transport layer connection and an RTSP session, it is possible for a malicious client to issue requests with random session identifiers which could affect other clients of an unsuspecting server. To mitigate this the server SHALL use a large, random and non-sequential session identifier to minimize the possibility of this kind of attack. However, unless the RTSP signaling is always confidentiality protected, e.g., using TLS, Schulzrinne, et al. Expires March 15, 2014 [Page 214] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 an on-path attacker will be able to hijack a session. Another choice for preventing session hijacking is to use client authentication and only allow the authenticated client creating the session to access that session. Authentication: Servers SHOULD implement both basic and digest [RFC2617] authentication. In environments requiring tighter security for the control messages, the transport layer mechanism TLS [RFC5246] SHOULD be used. Suspicious behavior: RTSP servers upon detecting instances of behavior which is deemed a security risk SHOULD return error code 403 (Forbidden). RTSP servers SHOULD also be aware of attempts to probe the server for weaknesses and entry points and MAY arbitrarily disconnect and ignore further requests from clients which are deemed to be in violation of local security policy. TLS through proxies: If one uses the possibility to connect TLS in multiple legs (Section 19.3) one really needs to be aware of the trust model. That procedure requires full faith and trust in all proxies, which will be identified, that one allows to connect through. They are men in the middle and have access to all that goes on over the TLS connection. Thus it is important to consider if that trust model is acceptable in the actual application. Further discussion of the actual trust model is in Section 19.3. Resource Exhaustion: As RTSP is a stateful protocol and establishes resource usage on the server there is a clear possibility to attack the server by trying to overbook these resources to perform a denial of service attack. This attack can be both against ongoing sessions and to prevent others from establishing sessions. RTSP agents will need to have mechanisms to prevent single peers from consuming extensive amounts of resources. The methods for guarding against this are varied and depends on the agent's role and capabilities and policies. Each implementation has to carefully consider their methods and policies to mitigate this threat. For example regarding handling of connections there are recommendations in Section 10.7. The above threats and considerations have resulted in a set of security functions and mechanisms built into or used by the protocol. The signaling protocol relies on two security features defined in the Security Framework (Section 19) namely client authentication using HTTP authentication and TLS based transport protection of the signaling messages. Both of these mechanisms are required to be Schulzrinne, et al. Expires March 15, 2014 [Page 215] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 implemented by any RTSP agent. A number of different security mitigations have been designed into the protocol and will be instantiated if the specification is implemented written, for example by ensuring sufficient amount of entropy in the randomly generated session identifiers when not using client authentication to minimize the risk of session hijacking. When client authentication is used the protection against hijacking will be greatly improved by scoping the accessible sessions to the one this client identity has created. Some of the above threats are such that the implementation of the RTSP functionality itself needs to consider which policy and strategy it uses to mitigate them. 21.2. Media Stream Delivery Threats The fact that RTSP establishes and controls a media stream delivery results in a set of security issues related to the media streams. This section will attempt to analyze general threats, however the choice of media stream transport protocol, such as RTP will result in some differences in threats and what mechanisms that exist to mitigate them. Thus it becomes important that each specification of a new media stream transport and delivery protocol usable by RTSP requires its own security analysis. This section includes one for RTP. The set of general threats from or by the media stream delivery itself are: Concentrated denial-of-service attack: The protocol offers the opportunity for a remote-controlled denial-of-service (DoS) attack, where the media stream is the hammer in that DoS attack. See Section 21.2.1. Media Confidentiality: The media delivery may contain content of any type and it is not possible in general to determine how sensitive this content is from a confidentiality point. Thus it is a strong requirement that any media delivery protocol provides a method for providing confidentiality of the actual media content. In addition to the media level confidentiality it becomes critical that no resource identifiers used in the signaling are exposed to an attacker as they may have human understandable names, or may be also available to the attacker so they can determine the content the user was delivered. Thus the signaling protocol must also provide confidentiality protection of any information related to the media resource. Schulzrinne, et al. Expires March 15, 2014 [Page 216] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Media Integrity and Authentication: There are several reasons, such as discrediting the target, misinformation of the target, why an attacker will be interested in substituting the media stream sent out from the RTSP server with one of the attacker's creation or selection. Therefore it is important that the media protocol provides mechanisms to verify the source authentication, integrity and prevent replay attacks on the media stream. Scope of Multicast: If RTSP is used to control the transmission of media onto a multicast network the scope of the delivery must be considered. RTSP supports the TTL Transport header parameter to indicate this scope for IPv4. IPv6 has a different mechanism for scope boundary. However, such scope control has risks, as it may be set too large and distribute media beyond the intended scope. Below (Section 21.2.2) we do a protocol specific analysis of security considerations for RTP based media transport. In that section we also make clear the requirements on implementing security functions for RTSP agents supporting media delivery over RTP. 21.2.1. Remote Denial of Service Attack The attacker may initiate traffic flows to one or more IP addresses by specifying them as the destination in SETUP requests. While the attacker's IP address may be known in this case, this is not always useful in prevention of more attacks or ascertaining the attacker's identity. Thus, an RTSP server MUST only allow client-specified destinations for RTSP-initiated traffic flows if the server has ensured that the specified destination address accepts receiving media through different security mechanisms. Security mechanisms that are acceptable in order of increasing generality are: o Verification of the client's identity against a database of known users using RTSP authentication mechanisms (preferably digest authentication or stronger) o A list of addresses that have consented to be media destinations, especially considering user identity o Media path based verification The server SHOULD NOT allow the destination field to be set unless a mechanism exists in the system to authorize the request originator to direct streams to the recipient. It is preferred that this authorization be performed by the media recipient (destination) itself and the credentials passed along to the server. However, in certain cases, such as when the recipient address is a multicast group, or when the recipient is unable to communicate with the server Schulzrinne, et al. Expires March 15, 2014 [Page 217] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 in an out-of-band manner, this may not be possible. In these cases the server may chose another method such as a server-resident authorization list to ensure that the request originator has the proper credentials to request stream delivery to the recipient. One solution that performs the necessary verification of acceptance of media suitable for unicast based delivery is the Interactive Connectivity Establishment (ICE) [RFC5245] based NAT traversal method described in [I-D.ietf-mmusic-rtsp-nat]. This mechanism uses random passwords and a username so that the probability of unintended indication as a valid media destination is very low. In addition the server includes in its Session Traversal Utilities for NAT (STUN) [RFC5389] requests a cookie (consisting of random material) that the destination echoes back, thus the solution also safe-guards against having an off-path attacker being able to spoof the STUN checks. This leaves this solution vulnerable only to on-path attackers that can see the STUN requests go to the target of attack and thus forge a response. For delivery to multicast addresses there is a need for another solution which is not specified in this memo. 21.2.2. RTP Security analysis RTP is a commonly used media transport protocol and has been the most common choice for RTSP 1.0 implementations. The core RTP protocol has been in use for a long time and it has well-known security properties and the RTP security consideration (Section 9 of [RFC3550]) needs to be reviewed. In perspective of the usage of RTP in context of RTSP the following properties should be noted: Stream Additions: RTP has support for multiple simultaneous media streams in each RTP session. As some use cases require support for non-synchronized adding and removal of media streams and their identifiers an attacker can easily insert additional media streams into a session context that according to protocol design is intended to be played out. Another threat vector is one of denial of service by exhausting the resources of the RTP session receiver, for example by using a large number of SSRC identifiers simultaneously. The strong mitigation of this is to ensure that one cryptographically authenticates any incoming packet flow to the RTP session. Weak mitigations like blocking additional media streams in session contexts easily lead to a denial of service vulnerability in addition to preventing certain RTP extensions or use cases which rely on multiple media streams, such as RTP retransmission [RFC4588] to function. Schulzrinne, et al. Expires March 15, 2014 [Page 218] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Forged Feedback: The built in RTP control Protocol (RTCP) also offers a large attack surface for a couple of different types of attacks. One venue is to send RTCP feedback to the media sender indicating large amounts of packet loss and thus trigger a media bit-rate adaptation response from the sender resulting in lowered media quality and potentially shut down of the media stream. Another attack is to perform a resource exhaustion attack on the receiver by using many SSRC identifiers to create large state tables and increase the RTCP related processing demands. RTP/RTCP Extensions: RTP and RTCP extensions generally provide additional and sometimes extremely powerful tools to do denial of service or service disruption. For example the Code Control Message [RFC5104] RTCP extensions enables both locking down the bit-rate to low values and disruption of video quality by requesting Intra frames. Taking into account the above general discussion in Section 21.2 and the RTP specific discussion in this section it is clear that it is necessary that a strong security mechanism is supported to protect RTP. Therefore this specification has the following requirements on RTP security functions for all RTSP agents that handles media streams and where the media stream transport is done using RTP. RTSP agents supporting RTP MUST implement Secure RTP (SRTP) [RFC3711] and thus the SAVP profile. In addition the secure AVP profile (SAVPF) [RFC5124] MUST also be supported if the AVPF profile is implemented. This specification requires no additional crypto transforms or configuration values beyond those > mandatory to implement in RFC3711, i.e., AES-CM and HMAC-SHA1. The default key- management mechanism which MUST be implemented is the one defined in the MIKEY Key Establishment (Appendix C.1.4.1). The MIKEY implementation MUST implement the necessary functions for MIKEY-RSA-R mode [RFC4738] and in addition the SRTP parameter negotiation necessary to negotiate the supported SRTP transforms and parameters. Schulzrinne, et al. Expires March 15, 2014 [Page 219] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22. IANA Considerations This section sets up a number of registries for RTSP 2.0 that should be maintained by IANA. These registries are separate from any registries existing for RTSP 1.0. For each registry there is a description of what it is required to contain, what specification is needed when adding an entry with IANA, and finally the entries that this document needs to register. See also the Section 2.7 "Extending RTSP". There is also an IANA registration of three SDP attributes. Registries or entries in registries which have been made for RTSP 1.0 are not moved to RTSP 2.0. The registries and entries in registries of RTSP 1.0 and RTSP 2.0 are independent. If any registry or entry in a registry is also required in RTSP 2.0, it MUST follow the procedure defined below to allocate the registry or entry in a registry. The sections describing how to register an item uses some of the registration policies described in RFC 5226 [RFC5226], namely "First Come, First Served", "Expert Review, "Specification Required", and "Standards Action". RFC-Editor Note: Please replace all occurrences of RFCXXXX with the RFC number this specification receives when published. In case a registry requires a contact person, the authors, with Magnus Westerlund (magnus.westerlund@ericsson.com) as primary, are the contact persons for any entries created by this document. A registration request to IANA MUST contain the following information: o A name of the item to register according to the rules specified by the intended registry. o Indication of who has change control over the feature (for example, IETF, ISO, ITU-T, other international standardization bodies, a consortium, a particular company or group of companies, or an individual); o A reference to a further description, if available, for example (in decreasing order of preference) an RFC, a published standard, a published paper, a patent filing, a technical report, documented source code or a computer manual; o For proprietary features, contact information (postal and email address); Schulzrinne, et al. Expires March 15, 2014 [Page 220] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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. 22.1.2. Registering New Feature-tags with IANA The registering of feature-tags is done on a first come, first served basis. The name of the feature MUST follow these rules: The name may be of any length, but SHOULD be no more than twenty characters long. The name MUST NOT contain any spaces, or control characters. The registration MUST indicate if the feature-tag applies to clients, servers, or proxies only or any combinations of these. Any proprietary feature MUST have as the first part of the name a vendor tag, which identifies the organization. The registry entries consist of the feature tag, a one paragraph description of what it represents, its applicability (server, client, proxy, any combination) and a reference to its specification where applicable. Examples for a vendor tag describing a proprietary feature are: vendorA.specfeat01 vendorA.specfeat02 22.1.3. Registered entries The following feature-tags are defined in this specification and hereby registered. The change control belongs to the IETF. play.basic: The implementation for delivery and playback operations according to the core RTSP specification, as defined in this memo. Applies for both clients, servers and proxies. See Section 11.1. play.scale: Support of scale operations for media playback. Applies only for servers. See Section 18.46. Schulzrinne, et al. Expires March 15, 2014 [Page 221] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 play.speed: Support of the speed functionality for media delivery. Applies only for servers. See Section 18.50. setup.rtp.rtcp.mux Support of the RTP and RTCP multiplexing as discussed in Appendix C.1.6.4. Applies for both client and servers and any media caching proxy. This should be represented by IANA as a table with the feature tags, contact person and their references. 22.2. RTSP Methods 22.2.1. Description Methods are described in Section 13. Extending the protocol with new methods allow for totally new functionality. 22.2.2. Registering New Methods with IANA A new method MUST be registered through an IETF Standards Action. The reason is that new methods may radically change the protocol's 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 what a request using the method does and what responses are expected. Which directions the method is used, C->S or S->C or both. How the use of headers, if any, modifies the behavior and effect of the method. o A list or table specifying which of the IANA registered headers that are allowed to be used with the method in request or/and response. The list or table SHOULD follow the format of tables in Section 18. o Describe how the method relates to network proxies. 22.2.3. Registered Entries This specification, RFCXXXX, registers 10 methods: DESCRIBE, GET_PARAMETER, OPTIONS, PAUSE, PLAY, PLAY_NOTIFY, REDIRECT, SETUP, SET_PARAMETER, and TEARDOWN. The initial table of the registry is provided below. Schulzrinne, et al. Expires March 15, 2014 [Page 222] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Method Directionality Reference ----------------------------------------------------- DESCRIBE C->S [RFCXXXX] GET_PARAMETER C->S, S->C [RFCXXXX] OPTIONS C->S, S->C [RFCXXXX] PAUSE C->S [RFCXXXX] PLAY C->S [RFCXXXX] PLAY_NOTIFY S->C [RFCXXXX] REDIRECT S->C [RFCXXXX] SETUP C->S [RFCXXXX] SET_PARAMETER C->S, S->C [RFCXXXX] TEARDOWN C->S, S->C [RFCXXXX] 22.3. RTSP Status Codes 22.3.1. Description A status code is the three digit number used to convey information in RTSP response messages, see Section 8. The number space is limited and care should be taken not to fill the space. 22.3.2. Registering New Status Codes with IANA A new status code registration follows the policy of IETF Review. New RTSP functionality requiring Status Codes should first be registered in the range x50-x99. Only when the range is full should registrations be done in the x00-x49 range, unless it is to adopt an HTTP extension also to RTSP. The reason is to enable any HTTP extension to be adopted to RTSP without needing to renumber any related status codes. A specification for a new status code MUST specify the following: o The registered number. o A description of what the status code means and the expected behavior of the sender and receiver of the code. 22.3.3. Registered Entries RFCXXXX, registers the numbered status code defined in the ABNF entry "Status-Code" except "extension-code" (that defines the syntax allowed for future extensions) in Section 20.2.2. 22.4. RTSP Headers Schulzrinne, et al. Expires March 15, 2014 [Page 223] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.4.1. Description By specifying new headers a method(s) can be enhanced in many different ways. An unknown header will be ignored by the receiving agent. If the new header is vital for a certain functionality, a feature-tag for the functionality can be created and demanded to be used by the counter-part with the inclusion of a Require header carrying the feature-tag. 22.4.2. Registering New Headers with IANA Registrations in the registry can be done following the Expert Review policy. A specification SHOULD be provided, preferably an IETF RFC or other Standards Developing Organization specification. The minimal information in a registration request is the header name and the contact information. The specification SHOULD contain the following information: o The name of the header. o An ABNF specification of the header syntax. o A list or table specifying when the header may be used, encompassing all methods, their request or response, the direction (C->S or S->C). o How the header is to be handled by proxies. o A description of the purpose of the header. 22.4.3. Registered entries All headers specified in Section 18 in RFCXXXX are to be registered. The Registry is to include header name and reference. Furthermore the following legacy RTSP headers defined in other specifications are registered with header name, reference and description according to below list. Note: These references may not fulfill all of the above rules for registrations due to their legacy status. o x-wap-profile defined in [TS-26234]. The x-wap-profile request header contains one or more absolute URLs to the requesting agent's device capability profile. o x-wap-profile-diff defined in [TS-26234]. The x-wap-profile-diff request header contains a subset of a device capability profile. Schulzrinne, et al. Expires March 15, 2014 [Page 224] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o x-wap-profile-warning defined in [TS-26234]. The x-wap-profile- warning is a response header that contains error codes explaining to what extent the server has been able to match the terminal request in regards to device capability profile as described using x-wap-profile and x-wap-profile-diff headers. o x-predecbufsize defined in [TS-26234]. This response header provides an RTSP agent with the TS 26.234 Annex G hypothetical pre-decoder buffer size. o x-initpredecbufperiod defined in [TS-26234]. This response header provides an RTSP agent with the TS 26.234 Annex G hypothetical pre-decoder buffering period. o x-initpostdecbufperiod defined in [TS-26234]. This response header provides an RTSP agent with the TS 26.234 Annex G post- decoder buffering period. o 3gpp-videopostdecbufsize defined in [TS-26234]. This response header provides an RTSP agent with the TS 26.234 defined post- decoder buffer size usable for H.264 (AVC) video streams. o 3GPP-Link-Char defined in [TS-26234]. This request header provides the RTSP server with the RTSP client's link characteristics as determined from the radio interface. The information that can be provided are guaranteed bit-rate, maximum bit-rate and maximum transfer delay. o 3GPP-Adaptation defined in [TS-26234]. This general header is part of the bit-rate adaptation solution specified for PSS. It provides the RTSP client's buffer sizes and target buffer levels to the server and responses are used to acknowledge the support and values. o 3GPP-QoE-Metrics defined in [TS-26234]. This general header is used by PSS RTSP agents to negotiate the quality of experience metrics that a client should gather and report to the server. o 3GPP-QoE-Feedback defined in [TS-26234]. This request header is used by RTSP clients supporting PSS to report the actual values of the metrics gathered in its quality of experience metering. The use of "x-" is NOT RECOMMENDED but the above headers in the register list were defined prior to the clarification. Schulzrinne, et al. Expires March 15, 2014 [Page 225] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.5. Accept-Credentials The security framework's TLS connection mechanism has two registerable entities. 22.5.1. Accept-Credentials policies In Section 19.3.1 three policies for how to handle certificates are specified. Further policies may be defined and MUST be registered with IANA using the following rules: o Registering requires an IETF Standards Action o A registration is required to name a contact person. o Name of the policy. o A describing text that explains how the policy works for handling the certificates. This specification registers the following values: Any Proxy User 22.5.2. Accept-Credentials hash algorithms The Accept-Credentials header (See Section 18.2) allows for the usage of other algorithms for hashing the DER records of accepted entities. The registration of any future algorithm is expected to be extremely rare and could also cause interoperability problems. Therefore the bar for registering new algorithms is intentionally placed high. Any registration of a new hash algorithm MUST fulfill the following requirement: o Follow the IETF Standards Action policy. o A definition of the algorithm and its identifier meeting the "token" ABNF requirement. The registered value is: Hash Alg. Id Reference ------------------------ sha-256 [RFCXXXX] Schulzrinne, et al. Expires March 15, 2014 [Page 226] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.6. Cache-Control Cache Directive Extensions There exists a number of cache directives which can be sent in the Cache-Control header. A registry for these cache directives MUST be defined with the following rules: o Registering requires an IETF Standards Action or IESG Approval. 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 a request or response directive. o A describing text that explains how the cache directive is used for RTSP controlled media streams. This specification registers the following values: no-cache: public: private: no-transform: only-if-cached: max-stale: min-fresh: must-revalidate: proxy-revalidate: max-age: The registry should be represented as: Name of the directive, contact person and reference. 22.7. Media Properties Schulzrinne, et al. Expires March 15, 2014 [Page 227] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.7.1. Description The media streams being controlled by RTSP can have many different properties. The media properties required to cover the use cases that were in mind when writing the specification are defined. However, it can be expected that further innovation will result in new use cases or media streams with properties not covered by the ones specified here. Thus new media properties can be specified. As new media properties may need a substantial amount of new definitions to correctly specify behavior for this property the bar is intended to be high. 22.7.2. Registration Rules Registering a new media property MUST fulfill the following requirements o Follow the Specification Required policy and get the approval of the designated Expert. o Have an ABNF definition of the media property value name that meets "media-prop-ext" definition o Define which media property group it belongs to or define a new group. o A Contact Person for the Registration o Description of all changes to the behavior of the RTSP protocol as result of these changes. 22.7.3. Registered Values This specification registers the ten values listed in Section 18.29. The registry should be represented as: Name of the media property, property group, contact person and reference. 22.8. Notify-Reason header 22.8.1. Description Notify-Reason values are used for indicating the reason the notification was sent. Each reason has its associated rules on what headers and information that may or must be included in the notification. New notification behaviors need to be specified to enable interoperable usage, thus a specification of each new value is required. Schulzrinne, et al. Expires March 15, 2014 [Page 228] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.8.2. Registration Rules Registrations for new Notify-Reason value MUST fulfill the following requirements o Follow the Specification Required policy and get the approval of the designated Expert. o An ABNF definition of the Notify reason value name that meets "Notify-Reason-extension" definition o A Contact Person for the Registration o Description of which headers shall be included in the request and response, when it should be sent, and any effect it has on the server client state. 22.8.3. Registered Values This specification registers 3 values defined in the Notify-Reas-val ABNF, Section 20.2.3: end-of-stream: This Notify-Reason value indicates the end of a media stream. media-properties-update: This Notify-Reason value allows the server to indicate that the properties of the media has changed during the playout. scale-change: This Notify-Reason value allows the server to notify the client about a change in the Scale of the media. The registry entries should be represented in the registry as: Name, short description, contact and reference. 22.9. Range Header Formats 22.9.1. Description The Range header (Section 18.40) allows for different range formats. These range formats also needs an identifier to be used the Accept- Ranges header (Section 18.5). New range formats may be registered, but moderation should be applied as it makes interoperability more difficult. Schulzrinne, et al. Expires March 15, 2014 [Page 229] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.9.2. Registration Rules A registration MUST fulfill the following requirements: o Follow the Specification Required policy. o An ABNF definition of the range format that fulfills the "range- ext" definition. o Define the range format identifier used in Accept-Ranges header according to the "extension-format" definition. o A Contact person for the registration. o Rules for how one handles the range when using a negative Scale. 22.9.3. Registered Values The registry should be represented as: Range header format identifier, Name of the range format, contact person and reference. This specification registers the following values. npt: Normal Play Time clock: UTC Absolute Time format smpte: SMPTE Timestamps smpte-30-drop: SMPTE Timestamps 29.97 Frames/sec (30 Hz with Drop) smpte-25: SMPTE Timestamps 25 Frames/sec 22.10. Terminate-Reason Header The Terminate-Reason header (Section 18.52) has two registries for extensions. 22.10.1. Redirect Reasons Registrations are done under the policy of Expert Review. The registered value needs to follow the Terminate-Reason ABNF, i.e., be a token. The specification needs to provide a definition of what procedures are to be followed when a client receives this redirect reason. This specification registers three values: o Session-Timeout Schulzrinne, et al. Expires March 15, 2014 [Page 230] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o Server-Admin o Internal-Error The registry should be represented as: Name of the Redirect Reason, contact person and reference. 22.10.2. Terminate-Reason Header Parameters Registrations are done under the policy of Specification Required. The registrations must define a syntax for the parameter that also follows the syntax allowed by the RTSP 2.0 specification. A contact person is also required. This specification registers: o time o user-msg The registry should be represented as: Name of the Terminate Reason, contact person and reference. 22.11. RTP-Info header parameters 22.11.1. Description The RTP-Info header (Section 18.45) carries one or more parameter value pairs with information about a particular point in the RTP stream. RTP extensions or new usages may need new types of information. As RTP information that could be needed is likely to be generic enough and to maximize the interoperability, new registration requires Specification Required. 22.11.2. Registration Rules Registrations for new RTP-Info value MUST fulfill the following requirements o Follow the Specification Required policy and get the approval of the designated Expert. o Have an ABNF definition that meets the "generic-param" definition o A Contact Person for the Registration Schulzrinne, et al. Expires March 15, 2014 [Page 231] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.11.3. Registered Values This specification registers the following parameter value pairs: o url o ssrc o seq o rtptime The registry should be represented as: Name of the parameter, contact person and reference. 22.12. Seek-Style Policies 22.12.1. Description New seek policies may be registered, however, a large number of these will complicate implementation substantially. The impact of unknown policies is that the server will not honor the unknown and use the server default policy instead. 22.12.2. Registration Rules Registrations of new Seek-Style polices MUST fulfill the following requirements o Follow the Specification Required policy. o Have an ABNF definition of the Seek-Style policy name that meets "Seek-S-value-ext" definition o A Contact Person for the Registration o Description of which headers shall be included in the request and response, when it should be sent, and any affect it has on the server client state. 22.12.3. Registered Values This specification registers 4 values: o RAP o CoRAP Schulzrinne, et al. Expires March 15, 2014 [Page 232] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o First-Prior o Next The registry should be represented as: Name of the Seek-Style Policy, short description, contact person and reference. 22.13. Transport Header Registries The transport header (Section 18.54) contains a number of parameters which have possibilities for future extensions. Therefore registries for these need to be defined. 22.13.1. Transport Protocol Identifier A Transport Protocol Specification consists of a Transport Protocol Identifier, representing some combination of transport protocols, and any number of transport header parameters required or optional to use with the identified protocol specification. This registry contains the identifiers used by registered Transport Protocol Identifiers. A registry for the parameter transport protocol identifier MUST be defined with the following rules: o Registering uses the policy of Specification Required. o A contact person or organization with address and email. o A value definition that are following the ABNF syntax definition of "transport-id" Section 20.2.3. o A describing text that explains how the registered value are used in RTSP, which underlying transport protocols that are used, and any required Transport header parameters. The registry should be represented as: The protocol ID string, contact person and reference. This specification registers the following values: RTP/AVP: Use of the RTP [RFC3550] protocol for media transport in combination with the "RTP profile for audio and video conferences with minimal control" [RFC3551] over UDP. The usage is explained in RFC XXXX, Appendix C.1. Schulzrinne, et al. Expires March 15, 2014 [Page 233] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTP/AVP/UDP: the same as RTP/AVP. RTP/AVPF: Use of the RTP [RFC3550] protocol for media transport in combination with the "Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)" [RFC4585] over UDP. The usage is explained in RFC XXXX, Appendix C.1. RTP/AVPF/UDP: the same as RTP/AVPF. RTP/SAVP: Use of the RTP [RFC3550] protocol for media transport in combination with the "The Secure Real-time Transport Protocol (SRTP)" [RFC3711] over UDP. The usage is explained in RFC XXXX, Appendix C.1. RTP/SAVP/UDP: the same as RTP/SAVP. RTP/SAVPF: Use of the RTP[RFC3550] protocol for media transport in combination with the Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF) [RFC5124] over UDP. The usage is explained in RFC XXXX, Appendix C.1. RTP/SAVPF/UDP: the same as RTP/SAVPF. RTP/AVP/TCP: Use of the RTP [RFC3550] protocol for media transport in combination with the "RTP profile for audio and video conferences with minimal control" [RFC3551] over TCP. The usage is explained in RFC XXXX, Appendix C.2.2. RTP/AVPF/TCP: Use of the RTP [RFC3550] protocol for media transport in combination with the "Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)" [RFC4585] over TCP. The usage is explained in RFC XXXX, Appendix C.2.2. RTP/SAVP/TCP: Use of the RTP [RFC3550] protocol for media transport in combination with the "The Secure Real-time Transport Protocol (SRTP)" [RFC3711] over TCP. The usage is explained in RFC XXXX, Appendix C.2.2. RTP/SAVPF/TCP: Use of the RTP [RFC3550] protocol for media transport in combination with the "Extended Secure RTP Profile for Real- time Transport Control Protocol (RTCP)-Based Feedback (RTP/ SAVPF)" [RFC5124] over TCP. The usage is explained in RFC XXXX, Appendix C.2.2. Schulzrinne, et al. Expires March 15, 2014 [Page 234] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 22.13.2. Transport modes The Transport Mode is a Transport header (Section 18.54) parameter, it is used to identify the general mode of media transport. The PLAY value registered defines a PLAYBACK mode, where media flows from Server to Client. A registry for the transport parameter mode MUST be defined with the following rules: o Registering requires an IETF Standards Action. o A contact person or organization with address and email. o A value definition that are following the ABNF "token" definition Section 20.2.3. o A describing text that explains how the registered value are used in RTSP. This specification registers 1 value: PLAY: See RFC XXXX. The registry should be represented as: The Transport Mode value, contact person and reference. 22.13.3. Transport Parameters Transport Parameters are different parameters used in a Transport Header's transport specification (Section 18.54) to provide additional information required beyond the transport protocol identifier to establish a functioning transport. A registry for parameters that may be included in the Transport header MUST be defined with the following rules: o Registering uses the Specification Required policy. o A Transport Parameter Name following the "token" ABNF definition. o A value definition, if the parameter takes a value, that are following the ABNF definition "trn-par-value" Section 20.2.3. o A describing text that explains how the registered value are used in RTSP. This specification registers all the transport parameters defined in Schulzrinne, et al. Expires March 15, 2014 [Page 235] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Section 18.54. This is a copy of this list: o unicast o multicast o interleaved o ttl o layers o ssrc o mode o dest_addr o src_addr o setup o connection o RTCP-mux o MIKEY The registry should be represented as: The transport parameter name, contact person and reference. 22.14. URI Schemes This specification updates two URI schemes, one previously registered "rtsp", and one missing in the registry "rtspu", previously only defined in the RTSP 1.0 [RFC2326], one new URI scheme "rtsps" is also registered. These URI schemes are registered in an existing registry (Uniform Resource Identifier (URI) Schemes) which are not created by this memo. Registrations are following RFC 4395[RFC4395]. 22.14.1. The rtsp URI Scheme URI scheme name: rtsp Status: Permanent Schulzrinne, et al. Expires March 15, 2014 [Page 236] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 URI scheme syntax: See Section 20.2.1 of RFC XXXX. URI scheme semantics: The rtsp scheme is used to indicate resources accessible through the usage of the Real-time Streaming Protocol (RTSP). RTSP allows different operations on the resource identified by the URI, but the primary purpose is the streaming delivery of the resource to a client. However, the operations that are currently defined are: DESCRIBE, GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT, SETUP, SET_PARAMETER, and TEARDOWN. Encoding considerations: IRIs in this scheme are defined and needs to be encoded as RTSP URIs when used within the RTSP protocol. That encoding is done according to RFC 3987. Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC 2326), RTSP 2.0 (RFC XXXX) Interoperability considerations: The extensions in the URI syntax performed between RTSP 1.0 and 2.0 can create interoperability issues. The changes are: Support for IPV6 literal in host part and future IP literals through RFC 3986 defined mechanism. A new relative format to use in the RTSP protocol elements that is not required to start with "/". The above changes should have no impact on interoperability as in detail discussed in Section 4.2 of RFCXXXX. Security considerations: All the security threats identified in Section 7 of RFC 3986 apply also to this scheme. They need to be reviewed and considered in any implementation utilizing this scheme. Contact: Magnus Westerlund, magnus.westerlund@ericsson.com Author/Change controller: IETF References: RFC 2326, RFC 3986, RFC 3987, RFC XXXX 22.14.2. The rtsps URI Scheme Schulzrinne, et al. Expires March 15, 2014 [Page 237] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 URI scheme name: rtsps Status: Permanent URI scheme syntax: See Section 20.2.1 of RFC XXXX. URI scheme semantics: The rtsps scheme is used to indicate resources accessible through the usage of the Real-time Streaming Protocol (RTSP) over TLS. RTSP allows different operations on the resource identified by the URI, but the primary purpose is the streaming delivery of the resource to a client. However, the operations that are currently defined are: DESCRIBE, GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT, SETUP, SET_PARAMETER, and TEARDOWN. Encoding considerations: IRIs in this scheme are defined and needs to be encoded as RTSP URIs when used within the RTSP protocol. That encoding is done according to RFC 3987. Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC 2326), RTSP 2.0 (RFC XXXX) Interoperability considerations: The "rtsps" scheme was never officially defined for RTSP 1.0, however it has seen widespread use in actual deployments of RTSP 1.0. Therefore this section discusses the believed changes between the unspecified RTSP 1.0 "rtsps" scheme and RTSP 2.0 definition. The extensions in the URI syntax performed between RTSP 1.0 and 2.0 can create interoperability issues. The changes are: Support for IPV6 literal in host part and future IP literals through RFC 3986 defined mechanism. A new relative format to use in the RTSP protocol elements that is not required to start with "/". The above changes should have no impact on interoperability as in detail discussed in Section 4.2 of RFCXXXX. Security considerations: All the security threats identified in Section 7 of RFC 3986 apply also to this scheme. They need to be reviewed and considered in any implementation utilizing this scheme. Contact: Magnus Westerlund, magnus.westerlund@ericsson.com Schulzrinne, et al. Expires March 15, 2014 [Page 238] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Author/Change controller: IETF References: RFC 2326, RFC 3986, RFC 3987, RFC XXXX 22.14.3. The rtspu URI Scheme URI scheme name: rtspu Status: Permanent URI scheme syntax: See Section 3.2 of RFC 2326. URI scheme semantics: The rtspu scheme is used to indicate resources accessible through the usage of the Real-time Streaming Protocol (RTSP) over unreliable datagram transport. RTSP allows different operations on the resource identified by the URI, but the primary purpose is the streaming delivery of the resource to a client. However, the operations that are currently defined are: DESCRIBE, GET_PARAMETER, OPTIONS, REDIRECT,PLAY, PLAY_NOTIFY, PAUSE, SETUP, SET_PARAMETER, and TEARDOWN. Encoding considerations: This scheme is not intended to be used with characters outside the US-ASCII repertoire. Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC 2326) Interoperability considerations: The definition of the transport mechanism of RTSP over UDP has interoperability issues. That makes the usage of this scheme problematic. Security considerations: All the security threats identified in Section 7 of RFC 3986 apply also to this scheme. They needs to be reviewed and considered in any implementation utilizing this scheme. Contact: Magnus Westerlund, magnus.westerlund@ericsson.com Author/Change controller: IETF References: RFC 2326 22.15. SDP attributes This specification defines three SDP [RFC4566] attributes that it is requested that IANA register. Schulzrinne, et al. Expires March 15, 2014 [Page 239] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 SDP Attribute ("att-field"): Attribute name: range Long form: Media Range Attribute Type of name: att-field Type of attribute: Media and session level Subject to charset: No Purpose: RFC XXXX Reference: RFC XXXX, RFC 2326 Values: See ABNF definition. Attribute name: control Long form: RTSP control URI Type of name: att-field Type of attribute: Media and session level Subject to charset: No Purpose: RFC XXXX Reference: RFC XXXX, RFC 2326 Values: Absolute or Relative URIs. Attribute name: mtag Long form: Message Tag Type of name: att-field Type of attribute: Media and session level Subject to charset: No Purpose: RFC XXXX Reference: RFC XXXX Values: See ABNF definition 22.16. Media Type Registration for text/parameters Type name: text Subtype name: parameters Required parameters: Optional parameters: charset: The charset parameter is applicable to the encoding of the parameter values. The default charset is UTF-8, if the 'charset' parameter is not present. Encoding considerations: 8bit Security considerations: This format may carry any type of parameters. Some can have security requirements, like privacy, confidentiality or integrity requirements. The format has no built in security protection. For the usage it was defined the Schulzrinne, et al. Expires March 15, 2014 [Page 240] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 transport can be protected between server and client using TLS. However, care must be taken to consider if also the proxies are trusted with the parameters in case hop-by-hop security is used. If stored as a file in file system, the necessary precautions need to be taken in relation to the parameters requirements including object security such as S/MIME [RFC5751]. Interoperability considerations: This media type was mentioned as a fictional example in [RFC2326], but was not formally specified. This has resulted in usage of this media type which may not match its formal definition. Published specification: RFC XXXX, Appendix F. Applications that use this media type: Applications that use RTSP and have additional parameters they like to read and set using the RTSP GET_PARAMETER and SET_PARAMETER methods. Additional information: Magic number(s): File extension(s): Macintosh file type code(s): Person & email address to contact for further information: Magnus Westerlund (magnus.westerlund@ericsson.com) Intended usage: Common Restrictions on usage: None Author: Magnus Westerlund (magnus.westerlund@ericsson.com) Change controller: IETF Addition Notes: Schulzrinne, et al. Expires March 15, 2014 [Page 241] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 23. References 23.1. Normative References [FIPS-pub-180-2] National Institute of Standards and Technology (NIST), "Federal Information Processing Standards Publications (FIPS PUBS) 180-2: Secure Hash Standard", August 2002. [I-D.ietf-avtcore-rtp-circuit-breakers] Perkins, C. and V. Singh, "Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions", draft-ietf-avtcore-rtp-circuit-breakers-03 (work in progress), July 2013. [RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, August 1980. [RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC 793, September 1981. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6) Specification", RFC 2460, December 1998. [RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H., Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999. [RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach, P., Luotonen, A., and L. Stewart, "HTTP Authentication: Basic and Digest Access Authentication", RFC 2617, June 1999. [RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, July 2003. [RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD 63, RFC 3629, November 2003. Schulzrinne, et al. Expires March 15, 2014 [Page 242] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004. [RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, August 2004. [RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform Resource Identifier (URI): Generic Syntax", STD 66, RFC 3986, January 2005. [RFC3987] Duerst, M. and M. Suignard, "Internationalized Resource Identifiers (IRIs)", RFC 3987, January 2005. [RFC4086] Eastlake, D., Schiller, J., and S. Crocker, "Randomness Requirements for Security", BCP 106, RFC 4086, June 2005. [RFC4291] Hinden, R. and S. Deering, "IP Version 6 Addressing Architecture", RFC 4291, February 2006. [RFC4395] Hansen, T., Hardie, T., and L. Masinter, "Guidelines and Registration Procedures for New URI Schemes", BCP 35, RFC 4395, February 2006. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection- Oriented Transport", RFC 4571, July 2006. [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 2006. [RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data Encodings", RFC 4648, October 2006. [RFC4738] Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY- RSA-R: An Additional Mode of Key Distribution in Multimedia Internet KEYing (MIKEY)", RFC 4738, November 2006. [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, February 2008. Schulzrinne, et al. Expires March 15, 2014 [Page 243] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 [RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 5226, May 2008. [RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, January 2008. [RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS) Protocol Version 1.2", RFC 5246, August 2008. [RFC5280] Cooper, D., Santesson, S., Farrell, S., Boeyen, S., Housley, R., and W. Polk, "Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile", RFC 5280, May 2008. [RFC5646] Phillips, A. and M. Davis, "Tags for Identifying Languages", BCP 47, RFC 5646, September 2009. [RFC5751] Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet Mail Extensions (S/MIME) Version 3.2 Message Specification", RFC 5751, January 2010. [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and Control Packets on a Single Port", RFC 5761, April 2010. [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description Protocol (SDP) Grouping Framework", RFC 5888, June 2010. [RFC6838] Freed, N., Klensin, J., and T. Hansen, "Media Type Specifications and Registration Procedures", BCP 13, RFC 6838, January 2013. [SMPTE_TC] Society of Motion Picture and Television Engineers, "SMPTE Standard for Television -- Time and Control Code, ST 12M- 1-2008". [TS-26234] Third Generation Partnership Project (3GPP), "Transparent end-to-end Packet-switched Streaming Service (PSS); Protocols and codecs; Technical Specification 26.234", December 2002. 23.2. Informative References [I-D.ietf-mmusic-rtsp-nat] Goldberg, J., Westerlund, M., and T. Zeng, "A Network Address Translator (NAT) Traversal mechanism for media Schulzrinne, et al. Expires March 15, 2014 [Page 244] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 controlled by Real-Time Streaming Protocol (RTSP)", draft-ietf-mmusic-rtsp-nat-16 (work in progress), May 2013. [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. [RFC1123] Braden, R., "Requirements for Internet Hosts - Application and Support", STD 3, RFC 1123, October 1989. [RFC2068] Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T. Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1", RFC 2068, January 1997. [RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998. [RFC2663] Srisuresh, P. and M. Holdrege, "IP Network Address Translator (NAT) Terminology and Considerations", RFC 2663, August 1999. [RFC2822] Resnick, P., "Internet Message Format", RFC 2822, April 2001. [RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session Announcement Protocol", RFC 2974, October 2000. [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002. [RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the Session Description Protocol (SDP)", RFC 4145, Schulzrinne, et al. Expires March 15, 2014 [Page 245] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 September 2005. [RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E. Carrara, "Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)", RFC 4567, July 2006. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, July 2006. [RFC4855] Casner, S., "Media Type Registration of RTP Payload Formats", RFC 4855, February 2007. [RFC4856] Casner, S., "Media Type Registration of Payload Formats in the RTP Profile for Audio and Video Conferences", RFC 4856, February 2007. [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC 5104, February 2008. [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", RFC 5245, April 2010. [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session Traversal Utilities for NAT (STUN)", RFC 5389, October 2008. [RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding Dependency in the Session Description Protocol (SDP)", RFC 5583, July 2009. [RFC5905] Mills, D., Martin, J., Burbank, J., and W. Kasch, "Network Time Protocol Version 4: Protocol and Algorithms Specification", RFC 5905, June 2010. [RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent, "Computing TCP's Retransmission Timer", RFC 6298, June 2011. [Stevens98] Stevens, W., "Unix Networking Programming - Volume 1, second edition", 1998. Schulzrinne, et al. Expires March 15, 2014 [Page 246] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix A. Examples This section contains several different examples trying to illustrate possible ways of using RTSP. The examples can also help with the understanding of how functions of RTSP work. However, remember that these are examples and the normative and syntax description in the other sections take precedence. Please also note that many of the examples contain syntax illegal line breaks to accommodate the formatting restriction that the RFC series impose. A.1. Media on Demand (Unicast) This is an example of media on demand streaming of a media stored in a container file. For purposes of this example, a container file is a storage entity in which multiple continuous media types pertaining to the same end-user presentation are present. In effect, the container file represents an RTSP presentation, with each of its components being RTSP controlled media streams. Container files are a widely used means to store such presentations. While the components are transported as independent streams, it is desirable to maintain a common context for those streams at the server end. This enables the server to keep a single storage handle open easily. It also allows treating all the streams equally in case of any prioritization of streams by the server. It is also possible that the presentation author may wish to prevent selective retrieval of the streams by the client in order to preserve the artistic effect of the combined media presentation. Similarly, in such a tightly bound presentation, it is desirable to be able to control all the streams via a single control message using an aggregate URI. The following is an example of using a single RTSP session to control multiple streams. It also illustrates the use of aggregate URIs. In a container file it is also desirable to not write any URI parts which are not kept, when the container is distributed, like the host and most of the path element. Therefore this example also uses the "*" and relative URI in the delivered SDP. Also this presentation description (SDP) is not cacheble, as the Expires header is set to an equal value with date indicating immediate expiration of its validity. Client C requests a presentation from media server M. The movie is stored in a container file. The client has obtained an RTSP URI to the container file. Schulzrinne, et al. Expires March 15, 2014 [Page 247] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0 CSeq: 1 User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 200 OK CSeq: 1 Server: PhonyServer/1.0 Date: Thu, 24 Jan 1997 15:35:06 GMT Content-Type: application/sdp Content-Length: 271 Content-Base: rtsp://example.com/twister.3gp/ Expires: 24 Jan 1997 15:35:06 GMT v=0 o=- 2890844256 2890842807 IN IP4 198.51.100.5 s=RTSP Session i=An Example of RTSP Session Usage e=adm@example.com c=IN IP4 0.0.0.0 a=control: * a=range:npt=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 March 15, 2014 [Page 248] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0 CSeq: 2 User-Agent: PhonyClient/1.2 Require: play.basic Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001" Accept-Ranges: npt, smpte, clock M->C: RTSP/2.0 200 OK CSeq: 2 Server: PhonyServer/1.0 Transport: RTP/AVP;unicast; ssrc=93CB001E; dest_addr="192.0.2.53:8000"/"192.0.2.53:8001"; src_addr="198.51.100.5:9000"/"198.51.100.5:9001" Session: 12345678 Expires: 24 Jan 1997 15:35:12 GMT Date: 24 Jan 1997 15:35:12 GMT Accept-Ranges: npt Media-Properties: Random-Access=0.02, Immutable, Unlimited C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0 CSeq: 3 User-Agent: PhonyClient/1.2 Require: play.basic Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003" Session: 12345678 Accept-Ranges: npt, smpte, clock M->C: RTSP/2.0 200 OK CSeq: 3 Server: PhonyServer/1.0 Transport: RTP/AVP;unicast; ssrc=A813FC13; dest_addr="192.0.2.53:8002"/"192.0.2.53:8003"; src_addr="198.51.100.5:9002"/"198.51.100.5:9003"; Session: 12345678 Expires: 24 Jan 1997 15:35:13 GMT Date: 24 Jan 1997 15:35:13 GMT Accept-Range: NPT Media-Properties: Random-Access=0.8, Immutable, Unlimited C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0 CSeq: 4 User-Agent: PhonyClient/1.2 Range: npt=30- Seek-Style: RAP Session: 12345678 Schulzrinne, et al. Expires March 15, 2014 [Page 249] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 M->C: RTSP/2.0 200 OK CSeq: 4 Server: PhonyServer/1.0 Date: 24 Jan 1997 15:35:14 GMT Session: 12345678 Range: npt=30-634.10 Seek-Style: RAP RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4" ssrc=0D12F123:seq=12345;rtptime=3450012, url="rtsp://example.com/twister.3gp/trackID=1" ssrc=4F312DD8:seq=54321;rtptime=2876889 C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0 CSeq: 5 User-Agent: PhonyClient/1.2 Session: 12345678 # Pause happens 0.87 seconds after starting to play M->C: RTSP/2.0 200 OK CSeq: 5 Server: PhonyServer/1.0 Date: 24 Jan 1997 15:36:01 GMT Session: 12345678 Range: npt=30.87-634.10 C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0 CSeq: 6 User-Agent: PhonyClient/1.2 Range: npt=30.87-634.10 Seek-Style: Next Session: 12345678 M->C: RTSP/2.0 200 OK CSeq: 6 Server: PhonyServer/1.0 Date: 24 Jan 1997 15:36:01 GMT Session: 12345678 Range: npt=30.87-634.10 Seek-Style: Next RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4" ssrc=0D12F123:seq=12555;rtptime=6330012, url="rtsp://example.com/twister.3gp/trackID=1" ssrc=4F312DD8:seq=55021;rtptime=3132889 C->M: TEARDOWN rtsp://example.com/twister.3gp/ RTSP/2.0 CSeq: 7 Schulzrinne, et al. Expires March 15, 2014 [Page 250] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 User-Agent: PhonyClient/1.2 Session: 12345678 M->C: RTSP/2.0 200 OK CSeq: 7 Server: PhonyServer/1.0 Date: 24 Jan 1997 15:49:34 GMT A.2. Media on Demand using Pipelining This example is basically the example above (Appendix A.1), but now utilizing pipelining to speed up the setup. It requires only two round trip times until the media starts flowing. First of all, the session description is retrieved to determine what media resources need to be setup. In the second step, one sends the necessary SETUP requests and the PLAY request to initiate media delivery. Client C requests a presentation from media server M. The movie is stored in a container file. The client has obtained an RTSP URI to the container file. C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0 CSeq: 1 User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 200 OK CSeq: 1 Server: PhonyServer/1.0 Date: Thu, 23 Jan 1997 15:35:06 GMT Content-Type: application/sdp Content-Length: 271 Content-Base: rtsp://example.com/twister.3gp/ Expires: 24 Jan 1997 15:35:06 GMT v=0 o=- 2890844256 2890842807 IN IP4 192.0.2.5 s=RTSP Session i=An Example of RTSP Session Usage e=adm@example.com c=IN IP4 0.0.0.0 a=control: * a=range:npt=0-0:10:34.10 t=0 0 m=audio 0 RTP/AVP 0 a=control: trackID=1 m=video 0 RTP/AVP 26 a=control: trackID=4 Schulzrinne, et al. Expires March 15, 2014 [Page 251] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0 CSeq: 2 User-Agent: PhonyClient/1.2 Require: play.basic Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001" Accept-Ranges: npt, smpte, clock Pipelined-Requests: 7654 C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0 CSeq: 3 User-Agent: PhonyClient/1.2 Require: play.basic Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003" Accept-Ranges: npt, smpte, clock Pipelined-Requests: 7654 C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0 CSeq: 4 User-Agent: PhonyClient/1.2 Range: npt=0- Seek-Style: RAP Pipelined-Requests: 7654 M->C: RTSP/2.0 200 OK CSeq: 2 Server: PhonyServer/1.0 Transport: RTP/AVP;unicast; dest_addr="192.0.2.53:8000"/"192.0.2.53:8001"; src_addr="198.51.100.5:9000"/"198.51.100.5:9001"; ssrc=93CB001E Session: 12345678 Expires: 24 Jan 1997 15:35:12 GMT Date: 23 Jan 1997 15:35:12 GMT Accept-Ranges: npt Pipelined-Requests: 7654 Media-Properties: Random-Access=0.2, Immutable, Unlimited M->C: RTSP/2.0 200 OK CSeq: 3 Server: PhonyServer/1.0 Transport: RTP/AVP;unicast; dest_addr="192.0.2.53:8002"/"192.0.2.53:8003; src_addr="198.51.100.5:9002"/"198.51.100.5:9003"; ssrc=A813FC13 Session: 12345678 Expires: 24 Jan 1997 15:35:13 GMT Date: 23 Jan 1997 15:35:13 GMT Accept-Range: NPT Schulzrinne, et al. Expires March 15, 2014 [Page 252] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Pipelined-Requests: 7654 Media-Properties: Random-Access=0.8, Immutable, Unlimited M->C: RTSP/2.0 200 OK CSeq: 4 Server: PhonyServer/1.0 Date: 23 Jan 1997 15:35:14 GMT Session: 12345678 Range: npt=0-623.10 Seek-Style: RAP RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4" ssrc=0D12F123:seq=12345;rtptime=3450012, url="rtsp://example.com/twister.3gp/trackID=1" ssrc=4F312DD8:seq=54321;rtptime=2876889 Pipelined-Requests: 7654 A.3. Secured Media Session for on Demand Content This example is basically the above example (Appendix A.2), but now including establishment of SRTP crypto contexts to get a secured media delivery. First of all, the client attempts to fetch this insecurely, but the server redirects to a URI indicating a requirement on using a secure connection for the RTSP messages. The client establish a TCP/TLS connections and the session description is retrieved to determine what media resources need to be setup. In the this session description secure media (SRTP) is indicated. In the next step, the client sends the necessary SETUP requests including MIKEY messages. This is pipeline with a PLAY request to initiate media delivery. Client C requests a presentation from media server M. The movie is stored in a container file. The client has obtained an RTSP URI to the container file. Note: The below MIKEY messages are not valid MIKEY message and are BASE64 encoded random data to represent where the MIKEY messages would go. C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0 CSeq: 1 User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 301 Moved Permanently CSeq: 1 Server: PhonyServer/1.0 Date: Thu, 23 Jan 1997 15:35:06 GMT Location: rtsps://example.com/twister.3gp C->M: Establish TCP/TLS connection and verify server's Schulzrinne, et al. Expires March 15, 2014 [Page 253] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 certificate that is represents example.com. Used for all below RTSP messages. C->M: DESCRIBE rtsps://example.com/twister.3gp RTSP/2.0 CSeq: 2 User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 200 OK CSeq: 2 Server: PhonyServer/1.0 Date: Thu, 23 Jan 1997 15:35:06 GMT Content-Type: application/sdp Content-Length: 271 Content-Base: rtsps://example.com/twister.3gp/ Expires: 24 Jan 1997 15:35:06 GMT v=0 o=- 2890844256 2890842807 IN IP4 192.0.2.5 s=RTSP Session i=An Example of RTSP Session Usage e=adm@example.com c=IN IP4 0.0.0.0 a=control: * a=range:npt=0-0:10:34.10 t=0 0 m=audio 0 RTP/SAVP 0 a=control: trackID=1 m=video 0 RTP/SAVP 26 a=control: trackID=4 C->M: SETUP rtsps://example.com/twister.3gp/trackID=1 RTSP/2.0 CSeq: 3 User-Agent: PhonyClient/1.2 Require: play.basic Transport: RTP/SAVP;unicast;dest_addr=":8000"/":8001"; MIKEY=VGhpcyBpcyB0aGUgZmlyc3Qgc3RyZWFtcyBNSUtFWSBtZXNzYWdl Accept-Ranges: npt, smpte, clock Pipelined-Requests: 7654 C->M: SETUP rtsps://example.com/twister.3gp/trackID=4 RTSP/2.0 CSeq: 4 User-Agent: PhonyClient/1.2 Require: play.basic Transport: RTP/SAVP;unicast;dest_addr=":8002"/":8003"; MIKEY=TUlLRVkgZm9yIHN0cmVhbSB0d2lzdGVyLjNncC90cmFja0lEPTQ= Accept-Ranges: npt, smpte, clock Pipelined-Requests: 7654 Schulzrinne, et al. Expires March 15, 2014 [Page 254] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->M: PLAY rtsps://example.com/twister.3gp/ RTSP/2.0 CSeq: 5 User-Agent: PhonyClient/1.2 Range: npt=0- Seek-Style: RAP Pipelined-Requests: 7654 M->C: RTSP/2.0 200 OK CSeq: 3 Server: PhonyServer/1.0 Transport: RTP/SAVP;unicast; dest_addr="192.0.2.53:8000"/"192.0.2.53:8001"; src_addr="198.51.100.5:9000"/"198.51.100.5:9001"; ssrc=93CB001E; MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD0x Session: 12345678 Expires: 24 Jan 1997 15:35:12 GMT Date: 23 Jan 1997 15:35:12 GMT Accept-Ranges: npt Pipelined-Requests: 7654 Media-Properties: Random-Access=0.2, Immutable, Unlimited M->C: RTSP/2.0 200 OK CSeq: 4 Server: PhonyServer/1.0 Transport: RTP/SAVP;unicast; dest_addr="192.0.2.53:8002"/"192.0.2.53:8003; src_addr="198.51.100.5:9002"/"198.51.100.5:9003"; ssrc=A813FC13; MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD00 Session: 12345678 Expires: 24 Jan 1997 15:35:13 GMT Date: 23 Jan 1997 15:35:13 GMT Accept-Range: NPT Pipelined-Requests: 7654 Media-Properties: Random-Access=0.8, Immutable, Unlimited M->C: RTSP/2.0 200 OK CSeq: 5 Server: PhonyServer/1.0 Date: 23 Jan 1997 15:35:14 GMT Session: 12345678 Range: npt=0-623.10 Seek-Style: RAP RTP-Info: url="rtsps://example.com/twister.3gp/trackID=4" ssrc=0D12F123:seq=12345;rtptime=3450012, url="rtsps://example.com/twister.3gp/trackID=1" ssrc=4F312DD8:seq=54321;rtptime=2876889; Schulzrinne, et al. Expires March 15, 2014 [Page 255] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Pipelined-Requests: 7654 A.4. Media on Demand (Unicast) An alternative example of media on demand with a bit more tweaks is the following. Client C requests a movie distributed from two different media servers A (audio.example.com) and V ( video.example.com). The media description is stored on a web server W. The media description contains descriptions of the presentation and all its streams, including the codecs that are available, dynamic RTP payload types, the protocol stack, and content information such as language or copyright restrictions. It may also give an indication about the timeline of the movie. In this example, the client is only interested in the last part of the movie. Schulzrinne, et al. Expires March 15, 2014 [Page 256] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->W: GET /twister.sdp HTTP/1.1 Host: www.example.com Accept: application/sdp W->C: HTTP/1.1 200 OK Date: Thu, 23 Jan 1997 15:35:06 GMT Content-Type: application/sdp Content-Length: 278 Expires: 23 Jan 1998 15:35:06 GMT v=0 o=- 2890844526 2890842807 IN IP4 198.51.100.5 s=RTSP Session e=adm@example.com c=IN IP4 0.0.0.0 a=range:npt=0-1:49:34 t=0 0 m=audio 0 RTP/AVP 0 a=control:rtsp://audio.example.com/twister/audio.en m=video 0 RTP/AVP 31 a=control:rtsp://video.example.com/twister/video C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/2.0 CSeq: 1 User-Agent: PhonyClient/1.2 Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057", RTP/AVP/TCP;unicast;interleaved=0-1 Accept-Ranges: npt, smpte, clock A->C: RTSP/2.0 200 OK CSeq: 1 Session: 12345678 Transport: RTP/AVP/UDP;unicast; dest_addr="192.0.2.53:3056"/"192.0.2.53:3057"; src_addr="198.51.100.5:5000"/"198.51.100.5:5001" Date: 23 Jan 1997 15:35:12 GMT Server: PhonyServer/1.0 Expires: 24 Jan 1997 15:35:12 GMT Cache-Control: public Accept-Ranges: npt, smpte Media-Properties: Random-Access=0.02, Immutable, Unlimited Schulzrinne, et al. Expires March 15, 2014 [Page 257] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->V: SETUP rtsp://video.example.com/twister/video RTSP/2.0 CSeq: 1 User-Agent: PhonyClient/1.2 Transport: RTP/AVP/UDP;unicast; dest_addr="192.0.2.53:3058"/"192.0.2.53:3059", RTP/AVP/TCP;unicast;interleaved=0-1 Accept-Ranges: npt, smpte, clock V->C: RTSP/2.0 200 OK CSeq: 1 Session: 23456789 Transport: RTP/AVP/UDP;unicast; dest_addr="192.0.2.53:3058"/"192.0.2.53:3059"; src_addr="198.51.100.5:5002"/"198.51.100.5:5003" Date: 23 Jan 1997 15:35:12 GMT Server: PhonyServer/1.0 Cache-Control: public Expires: 24 Jan 1997 15:35:12 GMT Accept-Ranges: npt, smpte Media-Properties: Random-Access=1.2, Immutable, Unlimited C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0 CSeq: 2 User-Agent: PhonyClient/1.2 Session: 23456789 Range: smpte=0:10:00- V->C: RTSP/2.0 200 OK CSeq: 2 Session: 23456789 Range: smpte=0:10:00-1:49:23 Seek-Style: First-Prior RTP-Info: url="rtsp://video.example.com/twister/video" ssrc=A17E189D:seq=12312232;rtptime=78712811 Server: PhonyServer/2.0 Date: 23 Jan 1997 15:35:13 GMT Schulzrinne, et al. Expires March 15, 2014 [Page 258] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0 CSeq: 2 User-Agent: PhonyClient/1.2 Session: 12345678 Range: smpte=0:10:00- A->C: RTSP/2.0 200 OK CSeq: 2 Session: 12345678 Range: smpte=0:10:00-1:49:23 Seek-Style: First-Prior RTP-Info: url="rtsp://audio.example.com/twister/audio.en" ssrc=3D124F01:seq=876655;rtptime=1032181 Server: PhonyServer/1.0 Date: 23 Jan 1997 15:35:13 GMT C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0 CSeq: 3 User-Agent: PhonyClient/1.2 Session: 12345678 A->C: RTSP/2.0 200 OK CSeq: 3 Server: PhonyServer/1.0 Date: 23 Jan 1997 15:36:52 GMT C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0 CSeq: 3 User-Agent: PhonyClient/1.2 Session: 23456789 V->C: RTSP/2.0 200 OK CSeq: 3 Server: PhonyServer/2.0 Date: 23 Jan 1997 15:36:52 GMT Even though the audio and video track are on two different servers that may start at slightly different times and may drift with respect to each other over time, the client can perform initial synchronization of the two media using RTP-Info and Range received in the PLAY responses. If the two servers are time synchronized the RTCP packets can also be used to maintain synchronization. Schulzrinne, et al. Expires March 15, 2014 [Page 259] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 A.5. Single Stream Container Files Some RTSP servers may treat all files as though they are "container files", yet other servers may not support such a concept. Because of this, clients needs to use the rules set forth in the session description for Request-URIs, rather than assuming that a consistent URI may always be used throughout. Below is an example of how a multi-stream server might expect a single-stream file to be served: C->S: DESCRIBE rtsp://foo.example.com/test.wav RTSP/2.0 Accept: application/x-rtsp-mh, application/sdp CSeq: 1 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 1 Content-base: rtsp://foo.example.com/test.wav/ Content-type: application/sdp Content-length: 163 Server: PhonyServer/1.0 Date: Thu, 23 Jan 1997 15:35:06 GMT Expires: 23 Jan 1997 17:00:00 GMT v=0 o=- 872653257 872653257 IN IP4 192.0.2.5 s=mu-law wave file i=audio test c=IN IP4 0.0.0.0 t=0 0 a=control: * m=audio 0 RTP/AVP 0 a=control:streamid=0 Schulzrinne, et al. Expires March 15, 2014 [Page 260] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: SETUP rtsp://foo.example.com/test.wav/streamid=0 RTSP/2.0 Transport: RTP/AVP/UDP;unicast; dest_addr=":6970"/":6971";mode="PLAY" CSeq: 2 User-Agent: PhonyClient/1.2 Accept-Ranges: npt, smpte, clock S->C: RTSP/2.0 200 OK Transport: RTP/AVP/UDP;unicast; dest_addr="192.0.2.53:6970"/"192.0.2.53:6971"; src_addr="198.51.100.5:6970"/"198.51.100.5:6971"; mode="PLAY";ssrc=EAB98712 CSeq: 2 Session: 2034820394 Expires: 23 Jan 1997 16:00:00 GMT Server: PhonyServer/1.0 Date: 23 Jan 1997 15:35:07 GMT Accept-Ranges: npt Media-Properties: Random-Acces=0.5, Immutable, Unlimited C->S: PLAY rtsp://foo.example.com/test.wav/ RTSP/2.0 CSeq: 3 User-Agent: PhonyClient/1.2 Session: 2034820394 S->C: RTSP/2.0 200 OK CSeq: 3 Server: PhonyServer/1.0 Date: 23 Jan 1997 15:35:08 GMT Session: 2034820394 Range: npt=0-600 Seek-Style: RAP RTP-Info: url="rtsp://foo.example.com/test.wav/streamid=0" ssrc=0D12F123:seq=981888;rtptime=3781123 Note the different URI in the SETUP command, and then the switch back to the aggregate URI in the PLAY command. This makes complete sense when there are multiple streams with aggregate control, but is less than intuitive in the special case where the number of streams is one. However, the server has declared the aggregated control URI in the SDP and therefore this is legal. In this case, it is also required that servers accept implementations that use the non-aggregated interpretation and use the individual media URI, like this: Schulzrinne, et al. Expires March 15, 2014 [Page 261] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0 CSeq: 3 User-Agent: PhonyClient/1.2 Session: 2034820394 A.6. Live Media Presentation Using Multicast The media server M chooses the multicast address and port. Here, it is assumed that the web server only contains a pointer to the full description, while the media server M maintains the full description. C->W: GET /sessions.html HTTP/1.1 Host: www.example.com W->C: HTTP/1.1 200 OK Content-Type: text/html ... Streamed Live Music performance ... C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/2.0 CSeq: 1 Supported: play.basic, play.scale User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 200 OK CSeq: 1 Content-Type: application/sdp Content-Length: 183 Server: PhonyServer/1.0 Date: Thu, 23 Jan 1997 15:35:06 GMT Supported: play.basic v=0 o=- 2890844526 2890842807 IN IP4 192.0.2.5 s=RTSP Session t=0 0 m=audio 3456 RTP/AVP 0 c=IN IP4 233.252.0.54/16 a=control: rtsp://live.example.com/concert/audio a=range:npt=0- Schulzrinne, et al. Expires March 15, 2014 [Page 262] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0 CSeq: 2 Transport: RTP/AVP;multicast; dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16 Accept-Ranges: npt, smpte, clock User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 200 OK CSeq: 2 Server: PhonyServer/1.0 Date: Thu, 23 Jan 1997 15:35:06 GMT Transport: RTP/AVP;multicast; dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16 ;ssrc=4D12AB92/0DF876A3 Session: 0456804596 Accept-Ranges: npt, clock Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0 C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0 CSeq: 3 Session: 0456804596 User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 200 OK CSeq: 3 Server: PhonyServer/1.0 Date: 23 Jan 1997 15:35:07 GMT Session: 0456804596 Seek-Style: Next Range:npt=1256- RTP-Info: url="rtsp://live.example.com/concert/audio" ssrc=0D12F123:seq=1473; rtptime=80000 A.7. Capability Negotiation This example illustrates how the client and server determine their capability to support a special feature, in this case "play.scale". The server, through the clients request and the included Supported header, learns the client supports RTSP 2.0, and also supports the playback time scaling feature of RTSP. The server's response contains the following feature related information to the client; it supports the basic media delivery functions (play.basic), the extended functionality of time scaling of content (play.scale), and one "example.com" proprietary feature (com.example.flight). The client also learns the methods supported (Public header) by the server for the indicated resource. Schulzrinne, et al. Expires March 15, 2014 [Page 263] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0 CSeq: 1 Supported: play.basic, play.scale User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 1 Public:OPTIONS,SETUP,PLAY,PAUSE,TEARDOWN,DESCRIBE,GET_PARAMETER Allow: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN, DESCRIBE Server: PhonyServer/2.0 Supported: play.basic, play.scale, com.example.flight When the client sends its SETUP request it tells the server that it requires support of the play.scale feature for this session by including the Require header. C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0 CSeq: 3 User-Agent: PhonyClient/1.2 Transport: RTP/AVP/UDP;unicast; dest_addr="192.0.2.53:3056"/"192.0.2.53:3057", RTP/AVP/TCP;unicast;interleaved=0-1 Require: play.scale Accept-Ranges: npt, smpte, clock User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 3 Session: 12345678 Transport: RTP/AVP/UDP;unicast; dest_addr="192.0.2.53:3056"/"192.0.2.53:3057"; src_addr="198.51.100.5:5000"/"198.51.100.5:5001" Server: PhonyServer/2.0 Accept-Ranges: npt, smpte Media-Properties: Random-Access=0.8, Immutable, Unlimited Schulzrinne, et al. Expires March 15, 2014 [Page 264] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix B. RTSP Protocol State Machine The RTSP session state machine describes the behavior of the protocol from RTSP session initialization through RTSP session termination. It is likely easiest to think of this as the server's state and then the client need to track what it believe the server's state will be based on sent or received RTSP messages. Thus most cases the below state tables can be read as: If the client do X, assuming it fulfills any pre-requisite the state will move to the new state and the indicated response will come back. However, there are also server to client notifications or requests, where the action describes what notification or request that happens, its requisites and what new state will occur after the server has received the response, and describing the clients response to the action. The State machine is defined on a per session basis which is uniquely identified by the RTSP session identifier. The session may contain one or more media streams depending on state. If a single media stream is part of the session it is in non-aggregated control. If two or more is part of the session it is in aggregated control. The below state machine is an informative description of the protocols behavior. In case of ambiguity with the earlier parts of this specification, the description in the earlier parts take precedence. B.1. States The state machine contains three states, described below. For each state there exists a table which shows which requests and events are allowed and whether they will result in a state change. Init: Initial state no session exists. Ready: Session is ready to start playing. Play: Session is playing, i.e., sending media stream data in the direction S->C. B.2. State variables This representation of the state machine needs more than its state to work. A small number of variables are also needed and they are explained below. Schulzrinne, et al. Expires March 15, 2014 [Page 265] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 NRM: The number of media streams part of this session. RP: Resume point, the point in the presentation time line at which a request to continue playing will resume from. A time format for the variable is not mandated. B.3. Abbreviations To make the state tables more compact a number of abbreviations are used, which are explained below. IFI: IF Implemented. md: Media PP: Pause Point, the point in the presentation time line at which the presentation was paused. Prs: Presentation, the complete multimedia presentation. RedP: Redirect Point, the point in the presentation time line at which a REDIRECT was specified to occur. SES: Session. B.4. State Tables This section contains a table for each state. The table contains all the requests and events that this state is allowed to act on. The events which are method names are, unless noted, requests with the given method in the direction client to server (C->S). In some cases there exist one or more requisite. The response column tells what type of response actions should be performed. Possible actions that are requested for an event include: response codes, e.g., 200, headers that need to be included in the response, setting of state variables, or setting of other session related parameters. The new state column tells which state the state machine changes to. The response to a valid request meeting the requisites is normally a 2xx (SUCCESS) unless otherwise noted in the response column. The exceptions need to be given a response according to the response column. If the request does not meet the requisite, is erroneous or some other type of error occurs, the appropriate response code is to be sent. If the response code is a 4xx the session state is unchanged. A response code of 3rr will result in that the session is ended and its state is changed to Init. A response code of 304 results in no state change. However, there are restrictions to when a 3rr response may be used. A 5xx response does not result in any Schulzrinne, et al. Expires March 15, 2014 [Page 266] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 change of the session state, except if the error is not possible to recover from. A unrecoverable error results in the ending of the session. As it in the general case can't be determined if it was a unrecoverable error or not the client will be required to test. In the case that the next request after a 5xx is responded with 454 (Session Not Found) the client knows that the session has ended. For any request message that cannot be responded to within the time defined in Section 10.4, a 100 response must be sent. The server will timeout the session after the period of time specified in the SETUP response, if no activity from the client is detected. Therefore there exists a timeout event for all states except Init. In the case that NRM = 1 the presentation URI is equal to the media URI or a specified presentation URI. For NRM > 1 the presentation URI needs to be other than any of the medias that are part of the session. This applies to all states. +---------------+-----------------+---------------------------------+ | Event | Prerequisite | Response | +---------------+-----------------+---------------------------------+ | DESCRIBE | Needs REDIRECT | 3rr, Redirect | | | | | | DESCRIBE | | 200, Session description | | | | | | OPTIONS | Session ID | 200, Reset session timeout | | | | timer | | | | | | OPTIONS | | 200 | | | | | | SET_PARAMETER | Valid parameter | 200, change value of parameter | | | | | | GET_PARAMETER | Valid parameter | 200, return value of parameter | +---------------+-----------------+---------------------------------+ Table 13: None state-machine changing events The methods in Table 13 do not have any effect on the state machine or the state variables. However, some methods do change other session related parameters, for example SET_PARAMETER which will set the parameter(s) specified in its body. Also all of these methods that allow Session header will also update the keep-alive timer for the session. Schulzrinne, et al. Expires March 15, 2014 [Page 267] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 +------------------+----------------+-----------+-------------------+ | 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 March 15, 2014 [Page 268] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 +-------------+------------------------+---------+------------------+ | Action | Requisite | New | Response | | | | State | | +-------------+------------------------+---------+------------------+ | SETUP | New URI | Ready | NRM +=1 | | | | | | | SETUP | URI Setup prior | Ready | Change transport | | | | | param | | | | | | | TEARDOWN | Prs URI, | Init | No session hdr, | | | | | NRM = 0 | | | | | | | TEARDOWN | md URI,NRM=1 | Init | No Session hdr, | | | | | NRM = 0 | | | | | | | TEARDOWN | md URI,NRM>1 | Ready | Session hdr, NRM | | | | | -= 1 | | | | | | | PLAY | Prs URI, No range | Play | Play from RP | | | | | | | PLAY | Prs URI, Range | Play | According to | | | | | range | | | | | | | PLAY | md URI, NRM=1, Range | Play | According to | | | | | range | | | | | | | PLAY | md URI, NRM=1 | Play | Play from RP | | | | | | | PAUSE | Prs URI | Ready | Return PP | | | | | | | SC:REDIRECT | Terminate-Reason | Ready | Set RedP | | | | | | | SC:REDIRECT | No Terminate-Reason | Init | Session is | | | time parameter | | removed | | | | | | | Timeout | | Init | | | | | | | | RedP | | Init | TEARDOWN of | | reached | | | session | +-------------+------------------------+---------+------------------+ Table 15: State: Ready In the Ready state, see Table 15, some of the actions are depending on the number of media streams (NRM) in the session, i.e., aggregated or non-aggregated control. A SETUP request in the Ready state can either add one more media stream to the session or, if the media stream (same URI) already is part of the session, change the Schulzrinne, et al. Expires March 15, 2014 [Page 269] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 transport parameters. TEARDOWN is depending on both the Request-URI and the number of media streams within the session. If the Request- URI is the presentations URI the whole session is torn down. If a media URI is used in the TEARDOWN request and more than one media exists in the session, the session will remain and a session header is returned in the response. If only a single media stream remains in the session when performing a TEARDOWN with a media URI the session is removed. The number of media streams remaining after tearing down a media stream determines the new state. Schulzrinne, et al. Expires March 15, 2014 [Page 270] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 +----------------+-----------------------+--------+-----------------+ | Action | Requisite | New | Response | | | | State | | +----------------+-----------------------+--------+-----------------+ | PAUSE | Prs URI | Ready | Set RP to | | | | | present point | | | | | | | End of media | All media | Play | Set RP = End of | | | | | media | | | | | | | End of range | | Play | Set RP = End of | | | | | range | | | | | | | PLAY | Prs URI, No range | Play | Play from | | | | | present point | | | | | | | PLAY | Prs URI, Range | Play | According to | | | | | range | | | | | | | SC:PLAY_NOTIFY | | Play | 200 | | | | | | | SETUP | New URI | Play | 455 | | | | | | | SETUP | Setuped URI | Play | 455 | | | | | | | SETUP | Setuped URI, IFI | Play | Change | | | | | transport | | | | | param. | | | | | | | TEARDOWN | Prs URI | Init | No session hdr | | | | | | | TEARDOWN | md URI,NRM=1 | Init | No Session hdr, | | | | | NRM=0 | | | | | | | TEARDOWN | md URI | Play | 455 | | | | | | | SC:REDIRECT | Terminate Reason with | Play | Set RedP | | | Time parameter | | | | | | | | | SC:REDIRECT | | Init | Session is | | | | | removed | | | | | | | RedP reached | | Init | TEARDOWN of | | | | | session | | | | | | | Timeout | | Init | Stop Media | | | | | playout | +----------------+-----------------------+--------+-----------------+ Schulzrinne, et al. Expires March 15, 2014 [Page 271] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Table 16: State: Play The Play state table, see Table 16, contains a number of requests that need a presentation URI (labeled as Prs URI) to work on (i.e., the presentation URI has to be used as the Request-URI). This is due to the exclusion of non-aggregated stream control in sessions with more than one media stream. To avoid inconsistencies between the client and server, automatic state transitions are avoided. This can be seen at for example "End of media" event when all media has finished playing, the session still remains in Play state. An explicit PAUSE request needs to be sent to change the state to Ready. It may appear that there exist automatic transitions in "RedP reached" and "PP reached". However, they are requested and acknowledged before they take place. The time at which the transition will happen is known by looking at the range header. If the client sends a request close in time to these transitions it needs to be prepared for receiving error messages, as the state may or may not have changed. Schulzrinne, et al. Expires March 15, 2014 [Page 272] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix C. Media Transport Alternatives This section defines how certain combinations of protocols, profiles and lower transports are used. This includes the usage of the Transport header's source and destination address parameters "src_addr" and "dest_addr". C.1. RTP This section defines the interaction of RTSP with respect to the RTP protocol [RFC3550]. It also defines any necessary media transport signaling with regards to RTP. The available RTP profiles and lower layer transports are described below along with rules on signaling the available combinations. C.1.1. AVP The usage of the "RTP Profile for Audio and Video Conferences with Minimal Control" [RFC3551] when using RTP for media transport over different lower layer transport protocols is defined below in regards to RTSP. One such case is defined within this document: the use of embedded (interleaved) binary data as defined in Section 14. The usage of this method is indicated by including the "interleaved" parameter. When using embedded binary data the "src_addr" and "dest_addr" MUST NOT be used. This addressing and multiplexing is used as defined with use of channel numbers and the interleaved parameter. C.1.2. AVP/UDP This part describes sending of RTP [RFC3550] over lower transport layer UDP [RFC0768] according to the profile "RTP Profile for Audio and Video Conferences with Minimal Control" defined in RFC 3551 [RFC3551]. Implementations of RTP/AVP/UDP MUST implement RTCP (Appendix C.1.6). This profile requires one or two uni- or bi- directional UDP flows per media stream. The first UDP flow is for RTP and the second is for RTCP. Multiplexing of RTP and RTCP (Appendix C.1.6.4) MAY be used, in which case a single UDP flow is used for both parts. Embedding of RTP data with the RTSP messages, in accordance with Section 14, SHOULD NOT be performed when RTSP messages are transported over unreliable transport protocols, like UDP [RFC0768]. The RTP/UDP and RTCP/UDP flows can be established using the Transport header's "src_addr", and "dest_addr" parameters. Schulzrinne, et al. Expires March 15, 2014 [Page 273] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 In RTSP PLAY mode, the transmission of RTP packets from client to server is unspecified. The behavior in regards to such RTP packets MAY be defined in future. The "src_addr" and "dest_addr" parameters are used in the following way for media delivery and playback mode, i.e., Mode=PLAY: o The "src_addr" and "dest_addr" parameters MUST contain either 1 or 2 address specifications. Note that two address specifications MAY be provided even if RTP and RTCP multiplexing is negotiated. o Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST contain either: * both an address and a port number, or * a port number without an address. o The first address specification given in either of the parameters applies to the RTP stream. The second specification if present applies to the RTCP stream, unless in case RTP and RTCP multiplexing is negotiated where both RTP and RTCP will use the first specification. o The RTP/UDP packets from the server to the client MUST be sent to the address and port given by the first address specification of the "dest_addr" parameter. o The RTCP/UDP packets from the server to the client MUST be sent to the address and port given by the second address specification of the "dest_addr" parameter, unless RTP and RTCP multiplexing has been negotiated, in which case RTCP MUST be sent to the first address specification. If no second pair is specified and RTP and RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent. o The RTCP/UDP packets from the client to the server MUST be sent to the address and port given by the second address specification of the "src_addr" parameter, unless RTP and RTCP multiplexing has been negotiated, in which case RTCP MUST be sent to the first address specification. If no second pair is specified and RTP and RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent. o The RTP/UDP packets from the client to the server MUST be sent to the address and port given by the first address specification of the "src_addr" parameter. o RTP and RTCP Packets SHOULD be sent from the corresponding receiver port, i.e., RTCP packets from the server should be sent Schulzrinne, et al. Expires March 15, 2014 [Page 274] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 from the "src_addr" parameters second address port pair, unless RTP and RTCP multiplexing has been negotiated in which case the first address port pair is used. C.1.3. AVPF/UDP The RTP profile "Extended RTP Profile for RTCP-based Feedback (RTP/ AVPF)" [RFC4585] MAY be used as RTP profiles in sessions using RTP. All that is defined for AVP MUST also apply for AVPF. The usage of AVPF is indicated by the media initialization protocol used. In the case of SDP it is indicated by media lines (m=) containing the profile RTP/AVPF. That SDP MAY also contain further AVPF related SDP attributes configuring the AVPF session regarding reporting interval and feedback messages to be used [RFC4585]. This configuration MUST be followed. C.1.4. SAVP/UDP The RTP profile "The Secure Real-time Transport Protocol (SRTP)" [RFC3711] is an RTP profile (SAVP) that MAY be used in RTSP sessions using RTP. All that is defined for AVP MUST also apply for SAVP. The usage of SRTP requires that a security context is established. The default key-management unless otherwise signalled SHALL be MIKEY in RSA-R mode as defined in Appendix C.1.4.1, and not according to the procedure defined in "Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)" [RFC4567]. The reason is that RFC 4567 sends the initial MIKEY message in SDP, thus both requiring the usage of the DESCRIBE method and forcing the server to keep state for clients performing DESCRIBE in anticipation that they might require key management. MIKEY is selected as default method for establishing SRTP cryptographic context within an RTSP session as it can be embedded in the RTSP messages, while still ensuring confidentiality of content of the keying material, even when using hop-by-hop TLS security for the RTSP messages. This method does also support pipelining of the RTSP messages. C.1.4.1. MIKEY Key Establishment This method for using MIKEY [RFC3830] to establish the SRTP cryptographic context is initiated in the client's SETUP request, and the server's response to the SETUP carries the MIKEY response. This ensures that the crypto context establishment happens simultaneously with the establishment of the media stream being protected. By using MIKEY's RSA-R mode [RFC4738] the client can be the initiator and Schulzrinne, et al. Expires March 15, 2014 [Page 275] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 still allow the server to set the parameters in accordance with the actual media stream. The SRTP cryptographic context establishment is done according to the following process: 1. The client determines that SAVP or SAVPF shall be used from media description format, e.g., SDP. If no other key management method is explicitly signalled, then MIKEY SHALL be used as defined herein. The use of SRTP with RTSP is only defined with MIKEY with keys established as defined in this Section. Future documents may define how an RTSP implementation treats SDP that indicates some other key mechanism to be used. The need for such specification include [RFC4567] that is not defined for use in RTSP 2.0 within this document. 2. The client SHALL establish a TLS connection for RTSP messages, directly or hop by hop with the server. If hop-by-hop TLS security is used, the User method SHALL be indicated in the Accept-Credentials header. We do note that using hop-by-hop does allow the proxy to insert itself as a man in the middle also in the MIKEY exchange by providing one of its certificates, rather than the server's in the Connection-Credentials header. The client SHALL therefore validate the server certificate. 3. The client retrieves the server's certificate from a direct TLS connection, or if hop by hop from Connection-Credentials header. The client then checks that the server certificate is valid and belongs to the server. 4. The client forms the MIKEY Initiator message using RSA-R mode in unicast mode as specified in [RFC4738]. The client SHOULD use the same certificate for TLS and in MIKEY to enable the server to bind the two together. The client's certificate SHALL be included in the MIKEY message. The client SHALL indicate its SRTP capabilities in the message. 5. The MIKEY message from the previous step is base64 [RFC4648] encoded and becomes the value of the MIKEY parameter that is included in the transport specification(s) that specifies a SRTP based profile (SAVP, SAVPF) in the SETUP request. 6. Any proxy encountering the MIKEY parameter SHALL forward it without modification. A proxy requiring to understand transport specification which doesn't support SAVP/SAVPF with MIKEY will discard the whole transport specification. Most types of proxies can easily support SAVP and SAVPF with MIKEY. If possible bypassing the proxy should be tried. Schulzrinne, et al. Expires March 15, 2014 [Page 276] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 7. The server upon receiving the SETUP request, will need to decide upon the transport specification to use, if multiple are included by the client. In the determination of which transport specifications that are supported and preferred, the server SHOULD decode the MIKEY message to take the embedded SRTP parameters into account. If all transport specs require SRTP but no MIKEY parameter or other supported keying method is included, the server SHALL respond with 403. 8. Upon generating a response the following outcomes can occur: * A transport spec not using SRTP and MIKEY is selected. Thus the response will not contain any MIKEY parameter. * A transport spec using SRTP and MIKEY is selected but an error is encountered in the MIKEY processing. In that case an RTSP error response code of 466 "Key Management Error" SHALL be used. A MIKEY message describing the error MAY be included. * A transport spec using SRTP and MIKEY is selected and a MIKEY response message can be created. The server SHOULD use the same certificate for TLS and in MIKEY to enable client to bind the two together. If a different certificate is used it SHALL be included in the MIKEY message. It is RECOMMENDED that the envelope key cache type is set to 'Cache' and that a single envelope key is reused for all MIKEY messages to the client. That message is included in the MIKEY parameter part of the single selected transport specification in the SETUP response. The server will set the SRTP parameters as preferred for this media stream within the supported range by the client. 9. The server transmits the SETUP response back to the client. 10. The client receives the SETUP response and if the response code indicates a successful request it decodes the MIKEY message and establishes the SRTP cryptographic context from the parameters in the MIKEY response. In the above method the client's certificate may be self-signed in cases where the client's identity is not necessary to authenticate and the security goal is only to ensure that the RTSP signaling client is the same as the one receiving the SRTP security context. Schulzrinne, et al. Expires March 15, 2014 [Page 277] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C.1.5. SAVPF/UDP The RTP profile "Extended Secure RTP Profile for RTCP-based Feedback (RTP/SAVPF)" [RFC5124] is an RTP profile (SAVPF) that MAY be used in RTSP sessions using RTP. All that is defined for AVPF MUST also apply for SAVPF. The usage of SRTP requires that a cryptographic context is established. The default mechanism for establishing that security association is to use MIKEY[RFC3830] with RTSP as defined in Appendix C.1.4.1. C.1.6. RTCP usage with RTSP RTCP has several usages when RTP is used for media transport as explained below. Due to that RTCP MUST be supported if an RTSP agent handles RTP. C.1.6.1. Media synchronization RTCP provides media synchronization and clock drift compensation. The initial media synchronization is available from RTP-Info header. However, to be able to handle any clock drift between the media streams, RTCP is needed. C.1.6.2. RTSP Session keep-alive RTCP traffic from the RTSP client to the RTSP server MUST function as keep-alive. This requires an RTSP server supporting RTP to use the received RTCP packets as indications that the client desires the related RTSP session to be kept alive. C.1.6.3. Bit-rate adaption RTCP Receiver reports and any additional feedback from the client MUST be used to adapt the bit-rate used over the transport for all cases when RTP is sent over UDP. An RTP sender without reserved resources MUST NOT use more than its fair share of the available resources. This can be determined by comparing on short to medium term (some seconds) the used bit-rate and adapt it so that the RTP sender sends at a bit-rate comparable to what a TCP sender would achieve on average over the same path. To ensure that the implementation's adaptation mechanism has a well defined outer envelope, all implementations using a non-congestion controlled unicast transport protocol, like UDP, MUST implement Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions [I-D.ietf-avtcore-rtp-circuit-breakers]. Schulzrinne, et al. Expires March 15, 2014 [Page 278] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C.1.6.4. RTP and RTCP Multiplexing RTSP can be used to negotiate the usage of RTP and RTCP multiplexing as described in [RFC5761]. This allows servers and client to reduce the amount of resources required for the session by only requiring one underlying transport stream per media stream instead of two when using RTP and RTCP. This lessens the server port consumption and also the necessary state and keep-alive work when operating across Network and Address Translators [RFC2663]. Content must be prepared with some consideration for RTP and RTCP multiplexing, mainly ensuring that the RTP payload types used do not collide with the ones used for RTCP packet types. This option likely needs explicit support from the content unless the RTP payload types can be remapped by the server and that is correctly reflected in the session description. Beyond that support of this feature should come at little cost and much gain. It is recommended that if the content and server support RTP and RTCP multiplexing that this is indicated in the session description, for example using the SDP attribute "a=rtcp-mux". If the SDP message contains the a=rtcp-mux attribute for a media stream, the server MUST support RTP and RTCP multiplexing. If indicated or otherwise desired by the client it can include the Transport parameter "RTCP-mux" in any transport specification where it desires to use RTCP-mux. The server will indicate if it supports RTCP-mux. Servers and Clients SHOULD support RTP and RTCP multiplexing. For capability exchange, an RTSP feature tag for RTP and RTCP multiplexing is defined: "setup.rtp.rtcp.mux". To minimize the risk of negotiation failure while using RTP and RTCP multiplexing some recommendations are here provided. If the session description includes explicit indication of support (a=rtcp-mux in SDP), then a RTSP agent can safely create a SETUP request with a transport specification with only a single dest_addr parameter address specification. If no such explicit indication is provided, then even if the feature tag "setup.rtp.rtcp.mux" is provided in a Supported header by the RTSP server or the feature tag included in the Required header in the SETUP request, the media resource may not support RTP and RTCP multiplexing. Thus, to maximize the probability of successful negotiation the RTSP agent is recommended to include two dest_addr parameter address specifications in the first or first set (if pipelining is used) of SETUP request(s) for any media resource aggregate. That way the RTSP server can either accept RTP and RTCP multiplexing and only use the first address specification, and if not use both specifications. The RTSP agent after having received the response for a successful negotiation of the usage of Schulzrinne, et al. Expires March 15, 2014 [Page 279] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTP and RTCP multiplexing, can then release the resources associated with the second address specification. C.2. RTP over TCP Transport of RTP over TCP can be done in two ways: over independent TCP connections using RFC 4571 [RFC4571] or interleaved in the RTSP connection. In both cases the protocol MUST be "rtp" and the lower layer MUST be TCP. The profile may be any of the above specified ones; AVP, AVPF, SAVP or SAVPF. C.2.1. Interleaved RTP over TCP The use of embedded (interleaved) binary data transported on the RTSP connection is possible as specified in Section 14. When using this declared combination of interleaved binary data the RTSP messages MUST be transported over TCP. TLS may or may not be used. If TLS is used both RTSP messages and the binary data will be protected by TLS. One should, however, consider that this will result in all media streams go through any proxy. Using independent TCP connections can avoid that issue. C.2.2. RTP over independent TCP In this Appendix, 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 address specifications needs to be included in the dest_addr parameter. If the client wishes to use RTP without RTCP, one address specification is included in the dest_addr parameter. If the client wishes to multiplex RTP and RTCP on a single transport flow (see Appendix C.1.6.4), one or two address specifications are included in the dest_addr parameter in addition to the RTCP-mux transport parameter. Two address specifications are allowed to allow successful negotiation when server or content can't support RTP and RTCP multiplexing. Ordering rules of dest_addr ports follow the rules for RTP/AVP/UDP. Schulzrinne, et al. Expires March 15, 2014 [Page 280] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 If the client wishes to play the active role in initiating the TCP connection, it MAY set the "setup" parameter (See Section 18.54) on the Transport line to be "active", or it MAY omit the setup parameter, as active is the default. If the client signals the active role, the ports in the address specifications in the dest_addr parameter MUST be set to 9 (the discard port). If the client wishes to play the passive role in TCP connection initiation, it MUST set the "setup" parameter on the Transport line to be "passive". If the client is able to assume the active or the passive role, it MUST set the "setup" parameter on the Transport line to be "actpass". In either case, the dest_addr parameter's address specification port value for RTP MUST be set to the TCP port number on which the client is expecting to receive the TCP connection for RTP, and the dest_addr's address specification port value for RTCP MUST be set to the TCP port number on which the client is expecting to receive the TCP connection for RTCP. In the case that the client wishes to multiplex RTP and RTCP on a single transport flow, the RTCP-mux parameter is included and one or two dest_addr parameter address specifications are included, as mentioned earlier in this section. If upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, a server decides to accept this requested option, the 2xx reply MUST contain a Transport option that specifies RTP/AVP/TCP (without using the interleaved parameter, and with using the unicast parameter). The dest_addr parameter value MUST be echoed from the parameter value in the client request unless the destination address (only port) was not provided in which case the server MAY include the source address of the RTSP TCP connection with the port number unchanged. In addition, the server reply MUST set the setup parameter on the Transport line, to indicate the role the server will play in the connection setup. Permissible values are "active" (if a client set "setup" to "passive" or "actpass") and "passive" (if a client set "setup" to "active" or "actpass"). If a server sets "setup" to "passive", the "src_addr" in the reply MUST indicate the ports the server is willing to receive an TCP connection for RTP and (if the client requested an TCP connection for RTCP by specifying two dest_addr address specifications) an TCP/RTCP connection. If a server sets "setup" to "active", the ports specified in "src_addr" address specifications MUST be set to 9. The server MAY use the "ssrc" parameter, following the guidance in Section 18.54. The server sets only one address specification in the case that the client has indicated only a single address specification or in case RTP and RTCP multiplexing was requested and accepted by server. Port ordering for src_addr follows the rules for Schulzrinne, et al. Expires March 15, 2014 [Page 281] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 RTP/AVP/UDP. Servers MUST support taking the passive role and MAY support taking the active role. Servers with a public IP address take the passive role, thus enabling clients behind NATs and Firewalls a better chance of successful connect to the server by actively connecting outwards. Therefore the clients are RECOMMENDED to take the active role. After sending (receiving) a 2xx reply for a SETUP method for a non- interleaved RTP/AVP/TCP media stream, the active party SHOULD initiate the TCP connection as soon as possible. The client MUST NOT send a PLAY request prior to the establishment of all the TCP connections negotiated using SETUP for the session. In case the server receives a PLAY request in a session that has not yet established all the TCP connections, it MUST respond using the 464 "Data Transport Not Ready Yet" (Section 17.4.28) 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. Unless RTP and RTCP multiplexing has been negotiated in which case RTP and RTCP will flow over a common TCP connection. As in the RTP/UDP case, client to server traffic on a RTP only TCP session is unspecified by this memo. The packets that travel on these connections MUST be framed using the protocol defined in [RFC4571], not by the framing defined for interleaving RTP over the RTSP connection defined in Section 14. A successful PAUSE request for a media being transported over RTP/ AVP/TCP pauses the flow of packets over the connections, without closing the connections. A successful TEARDOWN request signals that the TCP connections for RTP and RTCP are to be closed by the RTSP client as soon as possible. Subsequent SETUP requests on an already-SETUP RTP/AVP/TCP URI may be ambiguous in the following way: does the client wish to open up new TCP connection for RTP or RTCP for the URI, or does the client wish to continue using the existing TCP connections? The client SHOULD use the "connection" parameter (defined in Section 18.54) on the Transport line to make its intention clear (by setting "connection" to "new" if new connections are needed, and by setting "connection" to "existing" if the existing connections are to be used). After a 2xx reply for a SETUP request for a new connection, parties should close the pre-existing connections, after waiting a suitable period for any stray RTP or RTCP packets to arrive. The usage of SRTP, i.e., either SAVP or SAVPF profiles, requires that a security association is established. The default mechanism for Schulzrinne, et al. Expires March 15, 2014 [Page 282] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 establishing that security association is to use MIKEY[RFC3830] with RTSP as defined Appendix C.1.4.1. Below, we rewrite part of the example media on demand example shown in Appendix A.1 to use RTP/AVP/TCP non-interleaved: C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0 CSeq: 1 User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 200 OK CSeq: 1 Server: PhonyServer/1.0 Date: Thu, 23 Jan 1997 15:35:06 GMT Content-Type: application/sdp Content-Length: 227 Content-Base: rtsp://example.com/twister.3gp/ Expires: 24 Jan 1997 15:35:06 GMT v=0 o=- 2890844256 2890842807 IN IP4 198.51.100.34 s=RTSP Session i=An Example of RTSP Session Usage e=adm@example.com c=IN IP4 0.0.0.0 a=control: * a=range:npt=0-0:10:34.10 t=0 0 m=audio 0 RTP/AVP 0 a=control: trackID=1 C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0 CSeq: 2 User-Agent: PhonyClient/1.2 Require: play.basic Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9"; setup=active;connection=new Accept-Ranges: npt, smpte, clock Schulzrinne, et al. Expires March 15, 2014 [Page 283] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 M->C: RTSP/2.0 200 OK CSeq: 2 Server: PhonyServer/1.0 Transport: RTP/AVP/TCP;unicast; dest_addr=":9"/":9"; src_addr="198.51.100.5:53478"/"198.51.100:54091"; setup=passive;connection=new;ssrc=93CB001E Session: 12345678 Expires: 24 Jan 1997 15:35:12 GMT Date: 23 Jan 1997 15:35:12 GMT Accept-Ranges: npt Media-Properties: Random-Access=0.8, Immutable, Unlimited C->M: TCP Connection Establishment x2 C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0 CSeq: 4 User-Agent: PhonyClient/1.2 Range: npt=30- Session: 12345678 M->C: RTSP/2.0 200 OK CSeq: 4 Server: PhonyServer/1.0 Date: 23 Jan 1997 15:35:14 GMT Session: 12345678 Range: npt=30-623.10 Seek-Style: First-Prior RTP-Info: url="rtsp://example.com/twister.3gp/trackID=1" ssrc=4F312DD8:seq=54321;rtptime=2876889 C.3. Handling Media Clock Time Jumps in the RTP Media Layer RTSP allows media clients to control selected, non-contiguous sections of media presentations, rendering those streams with an RTP media layer [RFC3550]. Two cases occur, the first is when a new PLAY request replaces an old ongoing request and the new request results in a jump in the media. This should produce in the RTP layer a continuous media stream. A client may also directly following a completed PLAY request perform a new PLAY request. This will result in some gap in the media layer. The below text will look into both cases. A PLAY request that replaces an ongoing request allows the media layer rendering the RTP stream without being affected by jumps in media clock time. The RTP timestamps for the new media range is set so that they become continuous with the previous media range in the previous request. The RTP sequence number for the first packet in Schulzrinne, et al. Expires March 15, 2014 [Page 284] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 the new range will be the next following the last packet in the previous range, i.e., monotonically increasing. The goal is to allow the media rendering layer to work without interruption or reconfiguration across the jumps in media clock. This should be possible in all cases of replaced PLAY requests for media that has random-access properties. In this case care is needed to align frames or similar media dependent structures. In cases where jumps in media clock time are a result of RTSP signaling operations arriving after a completed PLAY operation, the request timing will result in that media becomes non-continuous. The server becomes unable to send the media so that it arrives timely and still carry timestamps to make the media stream continuous. In these cases the server will produce RTP streams where there are gaps in the RTP timeline for the media. In such cases, if the media has frame structure, aligning the timestamp for the next frame with the previous structure reduces the burden to render this media. The gap should represent the time the server hasn't been serving media, e.g., the time between the end of the media stream or a PAUSE request and the new PLAY request. In these cases the RTP sequence number would normally be monotonically increasing across the gap. For RTSP sessions with media that lacks random access properties, such as live streams, any media clock jump is commonly the result of a correspondingly long pause of delivery. The RTP timestamp will have increased in direct proportion to the duration of the paused delivery. Note also that in this case the RTP sequence number should be the next packet number. If not, the RTCP packet loss reporting will indicate as loss all packets not received between the point of pausing and later resuming. This may trigger congestion avoidance mechanisms. An allowed exception from the above recommendation on monotonically increasing RTP sequence number is live media streams, likely being relayed. In this case, when the client resumes delivery, it will get the media that is currently being delivered to the server itself. For this type of basic delivery of live streams to multiple users over unicast, individual rewriting of RTP sequence numbers becomes quite a burden. For solutions that anyway caches media, timeshifts, etc, the rewriting should be a minor issue. The goal when handling jumps in media clock time is that the provided stream is continuous without gaps in RTP timestamp or sequence number. However, when delivery has been halted for some reason the RTP timestamp when resuming MUST represent the duration the delivery was halted. RTP sequence number MUST generally be the next number, i.e., monotonically increasing modulo 65536. For media resources with the properties Time-Progressing and Time-Duration=0.0 the server MAY create RTP media streams with RTP sequence number jumps in them due to the client first halting delivery and later resuming it (PAUSE Schulzrinne, et al. Expires March 15, 2014 [Page 285] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 and then later PLAY). However, servers utilizing this exception must take into consideration the resulting RTCP receiver reports that likely contain loss reports for all the packets part of the discontinuity. A client cannot rely on that a server will align when resuming playing even if it is RECOMMENDED. The RTP-Info header will provide information on how the server acts in each case. We cannot assume that the RTSP client can communicate with the RTP media agent, as the two may be independent processes. If the RTP timestamp shows the same gap as the NPT, the media agent will assume that there is a pause in the presentation. If the jump in NPT is large enough, the RTP timestamp may roll over and the media agent may believe later packets to be duplicates of packets just played out. Having the RTP timestamp jump will also affect the RTCP measurements based on this. As an example, assume an RTP timestamp frequency of 8000 Hz, a packetization interval of 100 ms and an initial sequence number and timestamp of zero. C->S: PLAY rtsp://example.com/fizzle RTSP/2.0 CSeq: 4 Session: abcdefgh Range: npt=10-15 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 4 Session: abcdefgh Range: npt=10-15 RTP-Info: url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=0;rtptime=0 The ensuing RTP data stream is depicted below: S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s . . . S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s Upon the completion of the requested delivery the server sends a PLAY_NOTIFY Schulzrinne, et al. Expires March 15, 2014 [Page 286] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 S->C: PLAY_NOTIFY rtsp://example.com/fizzle RTSP/2.0 CSeq: 5 Notify-Reason: end-of-stream Request-Status: cseq=4 status=200 reason="OK" Range: npt=-15 RTP-Info:url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=49;rtptime=39200 Session: abcdefgh C->S: RTSP/2.0 200 OK CSeq: 5 User-Agent: PhonyClient/1.2 Upon the completion of the play range, the client follows up with a request to PLAY from a new NPT. C->S: PLAY rtsp://example.com/fizzle RTSP/2.0 CSeq: 6 Session: abcdefg Range: npt=18-20 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 6 Session: abcdefg Range: npt=18-20 RTP-Info: url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=50;rtptime=40100 The ensuing RTP data stream is depicted below: S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s . . . S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s In this example, first, NPT 10 through 15 is played, then the client requests the server to skip ahead and play NPT 18 through 20. The first segment is presented as RTP packets with sequence numbers 0 through 49 and timestamp 0 through 39,200. The second segment consists of RTP packets with sequence number 50 through 69, with timestamps 40,100 through 55,200. While there is a gap in the NPT, there is no gap in the sequence number space of the RTP data stream. The RTP timestamp gap is present in the above example due to the time it takes to perform the second play request, in this case 12.5 ms (100/8000). Schulzrinne, et al. Expires March 15, 2014 [Page 287] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C.4. Handling RTP Timestamps after PAUSE During a PAUSE / PLAY interaction in an RTSP session, the duration of time for which the RTP transmission was halted MUST be reflected in the RTP timestamp of each RTP stream. The duration can be calculated for each RTP stream as the time elapsed from when the last RTP packet was sent before the PAUSE request was received and when the first RTP packet was sent after the subsequent PLAY request was received. The duration includes all latency incurred and processing time required to complete the request. The RTP RFC [RFC3550] states that: The RTP timestamp for each unit [packet] would be related to the wallclock time at which the unit becomes current on the virtual presentation timeline. In order to satisfy the requirements of [RFC3550], the RTP timestamp space needs to increase continuously with real time. While this is not optimal for stored media, it is required for RTP and RTCP to function as intended. Using a continuous RTP timestamp space allows the same timestamp model for both stored and live media and allows better opportunity to integrate both types of media under a single control. As an example, assume a clock frequency of 8000 Hz, a packetization interval of 100 ms and an initial sequence number and timestamp of zero. C->S: PLAY rtsp://example.com/fizzle RTSP/2.0 CSeq: 4 Session: abcdefg Range: npt=10-15 User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 4 Session: abcdefg Range: npt=10-15 RTP-Info: url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=0;rtptime=0 The ensuing RTP data stream is depicted below: S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s The client then sends a PAUSE request: Schulzrinne, et al. Expires March 15, 2014 [Page 288] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0 CSeq: 5 Session: abcdefg User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 5 Session: abcdefg Range: npt=10.4-15 20 seconds elapse and then the client sends a PLAY request. In addition the server requires 15 ms to process the request: C->S: PLAY rtsp://example.com/fizzle RTSP/2.0 CSeq: 6 Session: abcdefg User-Agent: PhonyClient/1.2 S->C: RTSP/2.0 200 OK CSeq: 6 Session: abcdefg Range: npt=10.4-15 RTP-Info: url="rtsp://example.com/fizzle/audiotrack" ssrc=0D12F123:seq=4;rtptime=164400 The ensuing RTP data stream is depicted below: S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s First, NPT 10 through 10.3 is played, then a PAUSE is received by the server. After 20 seconds a PLAY is received by the server which takes 15 ms to process. The duration of time for which the session was paused is reflected in the RTP timestamp of the RTP packets sent after this PLAY request. A client can use the RTSP range header and RTP-Info header to map NPT time of a presentation with the RTP timestamp. Note: In RFC 2326 [RFC2326], this matter was not clearly defined and was misunderstood commonly. However, for RTSP 2.0 it is expected that this will be handled correctly and no exception handling will be required. Note further: It may be required to reset some of the state to ensure the correct media decoding and the usual jitter-buffer handling when issuing a PLAY request. Schulzrinne, et al. Expires March 15, 2014 [Page 289] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 C.5. RTSP / RTP Integration For certain data types, tight integration between the RTSP layer and the RTP layer will be necessary. This by no means precludes the above restrictions. Combined RTSP/RTP media clients should use the RTP-Info field to determine whether incoming RTP packets were sent before or after a seek or before or after a PAUSE. C.6. Scaling with RTP For scaling (see Section 18.46), RTP timestamps should correspond to the rendering timing. For example, when playing video recorded at 30 frames/second at a scale of two and speed (Section 18.50) of one, the server would drop every second frame to maintain and deliver video packets with the normal timestamp spacing of 3,000 per frame, but NPT would increase by 1/15 second for each video frame. Note: The above scaling puts requirements on the media codec or a media stream to support it. For example motion JPEG or other non- predictive video coding can easier handle the above example. C.7. Maintaining NPT synchronization with RTP timestamps The client can maintain a correct display of NPT (Normal Play Time) by noting the RTP timestamp value of the first packet arriving after repositioning. The sequence parameter of the RTP-Info (Section 18.45) header provides the first sequence number of the next segment. C.8. Continuous Audio For continuous audio, the server SHOULD set the RTP marker bit at the beginning of serving a new PLAY request or at jumps in timeline. This allows the client to perform playout delay adaptation. C.9. Multiple Sources in an RTP Session Note that more than one SSRC MAY be sent in the media stream. If it happens all sources are expected to be rendered simultaneously. C.10. Usage of SSRCs and the RTCP BYE Message During an RTSP Session The RTCP BYE message indicates the end of use of a given SSRC. If all sources leave an RTP session, it can, in most cases, be assumed to have ended. Therefore, a client or server MUST NOT send an RTCP BYE message until it has finished using a SSRC. A server SHOULD keep using a SSRC until the RTP session is terminated. Prolonging the use of a SSRC allows the established synchronization context associated Schulzrinne, et al. Expires March 15, 2014 [Page 290] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 with that SSRC to be used to synchronize subsequent PLAY requests even if the PLAY response is late. An SSRC collision with the SSRC that transmits media does also have consequences, as it will normally force the media sender to change its SSRC in accordance with the RTP specification [RFC3550]. However, an RTSP server may wait and see if the client changes and thus resolve the conflict to minimize the impact. As media sender SSRC change will result in a loss of synchronization context, and require any receiver to wait for RTCP sender reports for all media requiring synchronization before being able to play out synchronized. Due to these reasons a client joining a session should take care to not select the same SSRC(s) as the server indicates in the ssrc Transport header parameter. Any SSRC signalled in the Transport header MUST be avoided. A client detecting a collision prior to sending any RTP or RTCP messages SHALL also select a new SSRC. C.11. Future Additions It is the intention that any future protocol or profile regarding media delivery and lower transport should be easy to add to RTSP. This section provides the necessary steps that needs to be meet. The following things needs to be considered when adding a new protocol or profile for use with RTSP: o The protocol or profile needs to define a name tag representing it. This tag is required to be an ABNF "token" to be possible to use in the Transport header specification. o The useful combinations of protocol, profiles and lower layer transport for this extension needs to be defined. For each combination declare the necessary parameters to use in the Transport header. o For new media protocols the interaction with RTSP needs to be addressed. One important factor will be the media synchronization. It may be necessary to have new headers similar to RTP info to carry this information. o Discuss congestion control for media, especially if transport without built in congestion control is used. See the IANA section (Section 22) for information how to register new attributes. Schulzrinne, et al. Expires March 15, 2014 [Page 291] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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. Thus, it is worth pointing out that the interpretation of the SDP is done in the context of the SDP receiver, which is the one being configured. This is the same as in SAP [RFC2974]; this differs from SDP Offer/Answer [RFC3264] where each SDP is interpreted in the context of the agent providing it. SDP as is provides no mechanism by which a client can distinguish, without human guidance, between several media streams to be rendered simultaneously and a set of alternatives (e.g., two audio streams spoken in different languages). The SDP extension "Grouping of Media Lines in the Session Description Protocol (SDP)" [RFC5888] provides such functionality to some degree. Appendix D.4 describes the usage of SDP media line grouping for RTSP. D.1. Definitions The terms "session-level", "media-level" and other key/attribute names and values used in this appendix are to be used as defined in SDP[RFC4566]: D.1.1. Control URI The "a=control:" attribute is used to convey the control URI. This attribute is used both for the session and media descriptions. If used for individual media, it indicates the URI to be used for controlling that particular media stream. If found at the session level, the attribute indicates the URI for aggregate control (presentation URI). The session level URI MUST be different from any media level URI. The presence of a session level control attribute MUST be interpreted as support for aggregated control. The control attribute MUST be present on media level unless the presentation only contains a single media stream, in which case the attribute MAY be present on the session level only and then also apply to that single media stream. ABNF for the attribute is defined in Section 20.3. Example: a=control:rtsp://example.com/foo Schulzrinne, et al. Expires March 15, 2014 [Page 292] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 This attribute MAY contain either relative or absolute URIs, following the rules and conventions set out in RFC 3986 [RFC3986]. Implementations MUST look for a base URI in the following order: 1. the RTSP Content-Base field; 2. the RTSP Content-Location field; 3. the RTSP Request-URI. If this attribute contains only an asterisk (*), then the URI MUST be treated as if it were an empty embedded URI, and thus inherit the entire base URI. Note, RFC 2326 was very unclear on the processing of relative URI and several RTSP 1.0 implementations at the point of publishing this document did not perform RFC 3986 processing to determine the resulting URI, instead simple concatenation is common. To avoid this issue completely it is recommended to use absolute URI in the SDP. The URI handling for SDPs from container files need special consideration. For example let's assume that a container file has the URI: "rtsp://example.com/container.mp4". Let's further assume this URI is the base URI, and that there is an absolute media level URI: "rtsp://example.com/container.mp4/trackID=2". A relative media level URI that resolves in accordance with RFC 3986 [RFC3986] to the above given media URI is: "container.mp4/trackID=2". It is usually not desirable to need to include in or modify the SDP stored within the container file with the server local name of the container file. To avoid this, one can modify the base URI used to include a trailing slash, e.g., "rtsp://example.com/container.mp4/". In this case the relative URI for the media will only need to be: "trackID=2". However, this will also mean that using "*" in the SDP will result in control URI including the trailing slash, i.e., "rtsp://example.com/container.mp4/". Note: The usage of TrackID in the above is not a standardized form, but one example out of several similar strings such as TrackID, Track_ID, StreamID that is used by different server vendors to indicate a particular piece of media inside a container file. 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 Schulzrinne, et al. Expires March 15, 2014 [Page 293] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 indicated SHOULD be used for reception. The client MAY try to override the destination port, through the Transport header. The servers MAY allow this, the response will indicate if allowed or not. If the session is unicast, the port numbers are the ones RECOMMENDED by the server to the client, about which receiver ports to use; the client MUST still include its receiver ports in its SETUP request. The client MAY ignore this recommendation. If the server has no preference, it SHOULD set the port number value to zero. The "m=" lines contain information about which transport protocol, profile, and possibly lower-layer is to be used for the media stream. The combination of transport, profile and lower layer, like RTP/AVP/ UDP needs to be defined for how to be used with RTSP. The currently defined combinations are defined in Appendix C, further combinations MAY be specified. Example: m=audio 0 RTP/AVP 31 D.1.3. Payload Type(s) The payload type(s) are specified in the "m=" line. In case the payload type is a static payload type from RFC 3551 [RFC3551], no other information may be required. In case it is a dynamic payload type, the media attribute "rtpmap" is used to specify what the media is. The "encoding name" within the "rtpmap" attribute may be one of those specified in [RFC4856], or a media type registered with IANA according to [RFC4855], or an experimental encoding as specified in SDP [RFC4566]). Codec-specific parameters are not specified in this field, but rather in the "fmtp" attribute described below. The selection of the RTP payload type numbers used may be required to consider RTP and RTCP Multiplexing [RFC5761] if that is to be supported by the server. D.1.4. Format-Specific Parameters Format-specific parameters are conveyed using the "fmtp" media attribute. The syntax of the "fmtp" attribute is specific to the encoding(s) that the attribute refers to. Note that some of the format specific parameters may be specified outside of the fmtp parameters, like for example the "ptime" attribute for most audio encodings. D.1.5. Directionality of media stream The SDP attributes "a=sendrecv", "a=recvonly" and "a=sendonly" provide instructions about the direction the media streams flow Schulzrinne, et al. Expires March 15, 2014 [Page 294] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 within a session. When using RTSP the SDP can be delivered to a client using either RTSP DESCRIBE or a number of RTSP external methods, like HTTP, FTP, and email. Based on this the SDP applies to how the RTSP client will see the complete session. Thus media streams delivered from the RTSP server to the client, would be given the "a=recvonly" attribute. "a=recvonly" in a SDP provided to the RTSP client indicates that media delivery will only occur in the direction from the RTSP server to the client. SDP provided to the RTSP client that lacks any of the directionality attributes (a=recvonly, a=sendonly, a=sendrecv) would be interpreted as having a=sendrecv. At the time of writing there exist no RTSP mode suitable for media traffic in the direction from the RTSP client to the server. Thus all RTSP SDP SHOULD have a=recvonly attribute when using the PLAY mode defined in this document. If future modes are defined for media in client to server direction, then usage of a=sendonly, or a=sendrecv may become suitable to indicate intended media directions. D.1.6. Range of Presentation The "a=range" attribute defines the total time range of the stored session or an individual media. Non-seekable live sessions can be indicated as specified below, while the length of live sessions can be deduced from the "t=" and "r=" SDP parameters. The attribute is both a session and a media level attribute. For presentations that contain media streams of the same duration, the range attribute SHOULD only be used at session-level. In case of different lengths the range attribute MUST be given at media level for all media, and SHOULD NOT be given at session level. If the attribute is present at both media level and session level the media level values MUST be used. Note: Usually one will specify the same length for all media, even if there isn't media available for the full duration on all media. However, that requires that the server accepts PLAY requests within that range. Servers MUST take care to provide RTSP Range (see Section 18.40) values that are consistent with what is presented in the SDP for the content. There is no reason for non dynamic content, like media clips provided on demand to have inconsistent values. Inconsistent values between the SDP and the actual values for the content handled by the server is likely to generate some failure, like 457 "Invalid Range", in case the client uses PLAY requests with a Range header. In case the content is dynamic in length and it is infeasible to provide a correct value in the SDP the server is recommended to Schulzrinne, et al. Expires March 15, 2014 [Page 295] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 describe this as non-seekable content (see below). The server MAY override that property in the response to a PLAY request using the correct values in the Range header. The unit is specified first, followed by the value range. The units and their values are as defined in Section 4.4.1, Section 4.4.2 and Section 4.4.3 and MAY be extended with further formats. Any open ended range (start-), i.e., without stop range, is of unspecified duration and MUST be considered as non-seekable content unless this property is overridden. Multiple instances carrying different clock formats MAY be included at either session or media level. ABNF for the attribute is defined in Section 20.3. Examples: a=range:npt=0-34.4368 a=range:clock=19971113T211503Z-19971113T220300Z Non seekable stream of unknown duration: a=range:npt=0- D.1.7. Time of Availability The "t=" field defines when the SDP is valid. For on-demand content the server SHOULD indicate a stop time value for which it guarantees the description to be valid, and a start time that is equal to or before the time at which the DESCRIBE request was received. It MAY also indicate start and stop times of 0, meaning that the session is always available. For sessions that are of live type, i.e., specific start time, unknown stop time, likely unseekable, the "t=" and "r=" field SHOULD be used to indicate the start time of the event. The stop time SHOULD be given so that the live event will have ended at that time, while still not be unnecessary long into the future. D.1.8. Connection Information In SDP used with RTSP, the "c=" field contains the destination address for the media stream. If a multicast address is specified the client SHOULD use this address in any SETUP request as destination address, including any additional parameters, such as TTL. For on-demand unicast streams and some multicast streams, the destination address MAY be specified by the client via the SETUP request, thus overriding any specified address. To identify streams without a fixed destination address, where the client is required to specify a destination address, the "c=" field SHOULD be set to a null value. For addresses of type "IP4", this value MUST be "0.0.0.0", and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" (can also be Schulzrinne, et al. Expires March 15, 2014 [Page 296] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 written as "::"), i.e., the unspecified address according to RFC 4291 [RFC4291]. D.1.9. Message Body Tag The optional "a=mtag" attribute identifies a version of the session description. It is opaque to the client. SETUP requests may include this identifier in the If-Match field (see Section 18.24) to only allow session establishment if this attribute value still corresponds to that of the current description. The attribute value is opaque and may contain any character allowed within SDP attribute values. ABNF for the attribute is defined in Section 20.3. Example: a=mtag:"158bb3e7c7fd62ce67f12b533f06b83a" One could argue that the "o=" field provides identical functionality. However, it does so in a manner that would put constraints on servers that need to support multiple session description types other than SDP for the same piece of media content. D.2. Aggregate Control Not Available If a presentation does not support aggregate control no session level "a=control:" attribute is specified. For a SDP with multiple media sections specified, each section will have its own control URI specified via the "a=control:" attribute. Example: v=0 o=- 2890844256 2890842807 IN IP4 192.0.2.56 s=I came from a web page e=adm@example.com c=IN IP4 0.0.0.0 t=0 0 m=video 8002 RTP/AVP 31 a=control:rtsp://audio.example.com/movie.aud m=audio 8004 RTP/AVP 3 a=control:rtsp://video.example.com/movie.vid Note that the position of the control URI in the description implies that the client establishes separate RTSP control sessions to the servers audio.example.com and video.example.com. It is recommended that an SDP file contains the complete media initialization information even if it is delivered to the media Schulzrinne, et al. Expires March 15, 2014 [Page 297] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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: C->M: DESCRIBE rtsp://example.com/movie RTSP/2.0 CSeq: 1 User-Agent: PhonyClient/1.2 M->C: RTSP/2.0 200 OK CSeq: 1 Date: Thu, 23 Jan 1997 15:35:06 GMT Expires: Thu, 23 Jan 1997 16:35:06 GMT Content-Type: application/sdp Content-Base: rtsp://example.com/movie/ Content-Length: 227 v=0 o=- 2890844256 2890842807 IN IP4 192.0.2.211 s=I contain i= e=adm@example.com c=IN IP4 0.0.0.0 a=control:* t=0 0 m=video 8002 RTP/AVP 31 a=control:trackID=1 m=audio 8004 RTP/AVP 3 a=control:trackID=2 In this example, the client is recommended to establish a single RTSP session to the server, and uses the URIs rtsp://example.com/movie/trackID=1 and rtsp://example.com/movie/trackID=2 to set up the video and audio streams, respectively. The URI rtsp://example.com/movie/, which is resolved from the "*", controls the whole presentation (movie). A client is not required to issue SETUP requests for all streams Schulzrinne, et al. Expires March 15, 2014 [Page 298] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 within an aggregate object. Servers should allow the client to ask for only a subset of the streams. D.4. Grouping of Media Lines in SDP For some types of media it is desirable to express a relationship between various media components, for instance, for lip synchronization or Scalable Video Codec (SVC) [RFC5583]. This relationship is expressed on the SDP level by grouping of media lines, as described in [RFC5888] and can be exposed to RTSP. For RTSP it is mainly important to know how to handle grouped medias received by means of SDP, i.e., if the media are under aggregate control (see Appendix D.3) or if aggregate control is not available (see Appendix D.2). It is RECOMMENDED that grouped medias are handled by aggregate control, to give the client the ability to control either the whole presentation or single medias. D.5. RTSP external SDP delivery There are some considerations that need to be made when the session description is delivered to the client outside of RTSP, for example via HTTP or email. First of all, the SDP needs to contain absolute URIs, since relative will in most cases not work as the delivery will not correctly forward the base URI. The writing of the SDP session availability information, i.e., "t=" and "r=", needs to be carefully considered. When the SDP is fetched by the DESCRIBE method, the probability that it is valid is very high. However, the same is much less certain for SDPs distributed using other methods. Therefore the publisher of the SDP should take care to follow the recommendations about availability in the SDP specification [RFC4566] in Section 4.2. Schulzrinne, et al. Expires March 15, 2014 [Page 299] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix E. RTSP Use Cases This Appendix describes the most important and considered use cases for RTSP. They are listed in descending order of importance in regards to ensuring that all necessary functionality is present. This specification only fully supports usage of the two first. Also in these first two cases, there are special cases or exceptions that are not supported without extensions, e.g., the redirection of media delivery to another address than the controlling agent's (client's). E.1. On-demand Playback of Stored Content An RTSP capable server stores content suitable for being streamed to a client. A client desiring playback of any of the stored content uses RTSP to set up the media transport required to deliver the desired content. RTSP is then used to initiate, halt and manipulate the actual transmission (playout) of the content. RTSP is also required to provide necessary description and synchronization information for the content. The above high level description can be broken down into a number of functions that RTSP needs to be capable of. Presentation Description: Provide initialization information about the presentation (content); for example, which media codecs are needed for the content. Other information that is important includes the number of media streams the presentation contains, the transport protocols used for the media streams, and identifiers for these media streams. This information is required before setup of the content is possible and to determine if the client is even capable of using the content. This information need not be sent using RTSP; other external protocols can be used to transmit the transport presentation descriptions. Two good examples are the use of HTTP [RFC2616] or email to fetch or receive presentation descriptions like SDP [RFC4566] Setup: Set up some or all of the media streams in a presentation. The setup itself consists of selecting the protocol for media transport and the necessary parameters for the protocol, like addresses and ports. Control of Transmission: After the necessary media streams have been established the client can request the server to start transmitting the content. The client must be allowed to start or stop the transmission of the content at arbitrary times. The client must also be able to start the transmission at any Schulzrinne, et al. Expires March 15, 2014 [Page 300] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 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 agent controlling the traffic. However, allowing this without appropriate mechanisms for checking that the destination approves of this allows for distributed denial of service attacks (DDoS). E.2. Unicast Distribution of Live Content This use case is similar to the above on-demand content case (see Appendix E.1) the difference is the nature of the content itself. Live content is continuously distributed as it becomes available from a source; i.e., the main difference from on-demand is that one starts distributing content before the end of it has become available to the server. In many cases the consumer of live content is only interested in consuming what actually happens "now"; i.e., very similar to broadcast TV. However, in this case it is assumed that there exists no broadcast or multicast channel to the users, and instead the server functions as a distribution node, sending the same content to multiple receivers, using unicast traffic between server and client. This unicast traffic and the transport parameters are individually negotiated for each receiving client. Schulzrinne, et al. Expires March 15, 2014 [Page 301] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Another aspect of live content is that it often has a very limited time of availability, as it is only available for the duration of the event the content covers. An example of such a live content could be a music concert which lasts 2 hour and starts at a predetermined time. Thus there is a need to announce when and for how long the live content is available. In some cases, the server providing live content may be saving some or all of the content to allow clients to pause the stream and resume it from the paused point, or to "rewind" and play continuously from a point earlier than the live point. Hence, this use case does not necessarily exclude playing from other than the live point of the stream, playing with scales other than 1.0, etc. E.3. On-demand Playback using Multicast It is possible to use RTSP to request that media be delivered to a multicast group. The entity setting up the session (the controller) will then control when and what media is delivered to the group. This use case has some potential for denial of service attacks by flooding a multicast group. Therefore, a mechanism is needed to indicate that the group actually accepts the traffic from the RTSP server. An open issue in this use case is how one ensures that all receivers listening to the multicast or broadcast receives the session presentation configuring the receivers. This specification has to rely on an external solution to solve this issue. E.4. Inviting an RTSP server into a conference If one has an established conference or group session, it is possible to have an RTSP server distribute media to the whole group. Transmission to the group is simplest when controlled by a single participant or leader of the conference. Shared control might be possible, but would require further investigation and possibly extensions. This use case assumes that there exists either multicast or a conference focus that redistribute media to all participants. This use case is intended to be able to handle the following scenario: A conference leader or participant (hereafter called the controller) has some pre-stored content on an RTSP server that he wants to share with the group. The controller sets up an RTSP session at the streaming server for this content and retrieves the session description for the content. The destination for the media content is set to the shared multicast group or conference focus. Schulzrinne, et al. Expires March 15, 2014 [Page 302] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 When desired by the controller, he/she can start and stop the transmission of the media to the conference group. There are several issues with this use case that are not solved by this core specification for RTSP: Denial of service: To avoid an RTSP server from being an unknowing participant in a denial of service attack the server needs to be able to verify the destination's acceptance of the media. Such a mechanism to verify the approval of received media does not yet exist; instead, only policies can be used, which can be made to work in controlled environments. Distributing the presentation description to all participants in the group: To enable a media receiver to correctly decode the content the media configuration information needs to be distributed reliably to all participants. This will most likely require support from an external protocol. Passing control of the session: If it is desired to pass control of the RTSP session between the participants, some support will be required by an external protocol to exchange state information and possibly floor control of who is controlling the RTSP session. E.5. Live Content using Multicast This use case in its simplest form does not require any use of RTSP at all; this is what multicast conferences being announced with SAP [RFC2974] and SDP are intended to handle. However, in use cases where more advanced features like access control to the multicast session are desired, RTSP could be used for session establishment. A client desiring to join a live multicasted media session with cryptographic (encryption) access control could use RTSP in the following way. The source of the session announces the session and gives all interested an RTSP URI. The client connects to the server and requests the presentation description, allowing configuration for reception of the media. In this step it is possible for the client to use secured transport and any desired level of authentication; for example, for billing or access control. An RTSP link also allows for load balancing between multiple servers. If these were the only goals, they could be achieved by simply using HTTP. However, for cases where the sender likes to keep track of each individual receiver of a session, and possibly use the session as a side channel for distributing key-updates or other information on a per-receiver basis, and the full set of receivers is not known Schulzrinne, et al. Expires March 15, 2014 [Page 303] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 prior to the session start, the state establishment that RTSP provides can be beneficial. In this case a client would establish an RTSP session for this multicast group with the RTSP server. The RTSP server will not transmit any media, but instead will point to the multicast group. The client and server will be able to keep the session alive for as long as the receiver participates in the session thus enabling, for example, the server to push updates to the client. This use case will most likely not be able to be implemented without some extensions to the server-to-client push mechanism. Here the PLAY_NOTIFY method (see Section 13.5) with a suitable extension could provide clear benefits. Schulzrinne, et al. Expires March 15, 2014 [Page 304] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix F. Text format for Parameters A resource of type "text/parameters" consists of either 1) a list of parameters (for a query) or 2) a list of parameters and associated values (for an response or setting of the parameter). Each entry of the list is a single line of text. Parameters are separated from values by a colon. The parameter name MUST only use US-ASCII visible characters while the values are UTF-8 text strings. The media type registration form is in Section 22.16. There is a potential interoperability issue for this format. It was named in RFC 2326 but never defined, even if used in examples that hint at the syntax. This format matches the purpose and its syntax supports the examples provided. However, it goes further by allowing UTF-8 in the value part, thus usage of UTF-8 strings may not be supported. However, as individual parameters are not defined, the using application anyway needs to have out-of-band agreement or using feature-tag to determine if the end-point supports the parameters. The ABNF [RFC5234] grammar for "text/parameters" content is: file = *((parameter / parameter-value) CRLF) parameter = 1*visible-except-colon parameter-value = parameter *WSP ":" value visible-except-colon = %x21-39 / %x3B-7E ; VCHAR - ":" value = *(TEXT-UTF8char / WSP) TEXT-UTF8char = %x21-7E / UTF8-NONASCII UTF8-NONASCII = %xC0-DF 1UTF8-CONT / %xE0-EF 2UTF8-CONT / %xF0-F7 3UTF8-CONT / %xF8-FB 4UTF8-CONT / %xFC-FD 5UTF8-CONT UTF8-CONT = %x80-BF WSP = ; Space or HTAB VCHAR = CRLF = Schulzrinne, et al. Expires March 15, 2014 [Page 305] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix G. Requirements for Unreliable Transport of RTSP This appendix provides guidance for those who want to implement RTSP messages over unreliable transports as has been defined in RTSP 1.0 [RFC2326]. RFC 2326 defined the "rtspu" URI scheme and provided some basic information for the transport of RTSP messages over UDP. The information is being provided here as there has been at at least one commercial implementation and compatibility with that should be maintained. The following points should be considered for an interoperable implementation: o 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 6298) [RFC6298], with an initial round-trip value of 500 ms. An implementation may cache the last RTT measurement as the initial value for future connections. o The Timestamp header (Section 18.53) is used to avoid the retransmission ambiguity problem [Stevens98]. o The registered default port for RTSP over UDP for the server is 554. o RTSP messages can be carried over any lower-layer transport protocol that is 8-bit clean. o RTSP messages are vulnerable to bit errors and should not be subjected to them. o Source authentication, or at least validation that RTSP messages comes from the same entity becomes extremely important, as session hijacking may be substantially easier for RTSP message transport using an unreliable protocol like UDP than for TCP. There are two RTSP headers that are primarily intended for being used by the unreliable handling of RTSP messages and which will be maintained: o CSeq: See Section 18.20. It should be noted that the CSeq header is also required to match requests and responses independent whether a reliable or unreliable transport is used. Schulzrinne, et al. Expires March 15, 2014 [Page 306] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o Timestamp: See Section 18.53 Schulzrinne, et al. Expires March 15, 2014 [Page 307] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix H. Backwards Compatibility Considerations This section contains notes on issues about backwards compatibility with clients or servers being implemented according to RFC 2326 [RFC2326]. Note that there exists no requirement to implement RTSP 1.0; in fact we recommend against it as it is difficult to do in an interoperable way. A server implementing RTSP/2.0 MUST include an RTSP-Version of RTSP/2.0 in all responses to requests containing RTSP-Version RTSP/2.0. If a server receives an RTSP/1.0 request, it MAY respond with an RTSP/1.0 response if it chooses to support RFC 2326. If the server chooses not to support RFC 2326, it MUST respond with a 505 (RTSP Version not supported) status code. A server MUST NOT respond to an RTSP-Version RTSP/1.0 request with an RTSP-Version RTSP/2.0 response. Clients implementing RTSP/2.0 MAY use an OPTIONS request with a RTSP- Version of 2.0 to determine whether a server supports RTSP/2.0. If the server responds with either an RTSP-Version of 1.0 or a status code of 505 (RTSP Version not supported), the client will have to use RTSP/1.0 requests if it chooses to support RFC 2326. H.1. Play Request in Play State The behavior in the server when a Play is received in Play state has changed (Section 13.4). In RFC 2326, the new PLAY request would be queued until the current Play completed. Any new PLAY request now takes effect immediately replacing the previous request. H.2. Using Persistent Connections Some server implementations of RFC 2326 maintain a one-to-one relationship between a connection and an RTSP session. Such implementations require clients to use a persistent connection to communicate with the server and when a client closes its connection, the server may remove the RTSP session. This is worth noting if a RTSP 2.0 client also supporting 1.0 connects to a 1.0 server. Schulzrinne, et al. Expires March 15, 2014 [Page 308] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix I. Changes This appendix briefly lists the differences between RTSP 1.0 [RFC2326] and RTSP 2.0 for an informational purpose. For implementers of RTSP 2.0 it is recommended to read carefully through this memo and not to rely on the list of changes below to adapt from RTSP 1.0 to RTSP 2.0, as RTSP 2.0 is not intended to be backwards compatible with RTSP 1.0 [RFC2326] other than the version negotiation mechanism. I.1. Brief Overview The following protocol elements were removed in RTSP 2.0 compared to RTSP 1.0: o there is no section on minimal implementation anymore, but more the definition of RTSP 2.0 core; o the RECORD and ANNOUNCE methods and all related functionality (including 201 (Created) and 250 (Low On Storage Space) status codes); o the use of UDP for RTSP message transport was removed due to missing interest and to broken specification; o the use of PLAY method for keep-alive in Play state. The following protocol elements were added or changed in RTSP 2.0 compared to RTSP 1.0: o RTSP session TEARDOWN from the server to the client; o IPv6 support; o extended IANA registries (e.g., transport headers parameters, transport-protocol, profile, lower-transport, and mode); o request pipelining for quick session start-up; o fully reworked state-machine; o RTSP messages now use URIs rather then URLs; o incorporated much of related HTTP text ([RFC2616]) in this memo, compared to just referencing the sections in HTTP, to avoid ambiguities; Schulzrinne, et al. Expires March 15, 2014 [Page 309] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o the REDIRECT method was expanded and diversified for different situations; o Includes a new section about how to setup different media transport alternatives and their profiles, and lower layer protocols. This caused the appendix on RTP interaction to be moved there instead of being in the part which describes RTP. The section also includes guidelines what to consider when writing usage guidelines for new protocols and profiles; o Added an asynchronous notification method PLAY_NOTIFY. This method is used by the RTSP server to asynchronously notify clients about session changes while in Play state. To a limited extent this is comparable with some implementations of ANNOUNCE in RTSP 1.0 not intended for Recording. I.2. Detailed List of Changes Compared to RTSP 1.0 (RFC 2326), the below changes has been made when defining RTSP 2.0. Note that this list does not reflect minor changes in wording or correction of typographical errors. o The section on minimal implementation was deleted without substitution. o The Transport header has been changed in the following way: * The ABNF has been changed to define that extensions are possible, and that unknown parameters result in that servers ignore the transport specification. * To prevent backwards compatibility issues, any extension or new parameter requires the usage of a feature-tag combined with the Require header. * Syntax unclarities with the Mode parameter have been resolved. * Syntax error with ";" for multicast and unicast has been resolved. * Two new addressing parameters have been defined, src_addr and dest_addr. These replace the parameters "port", "client_port", "server_port", "destination", "source". * Support for IPv6 explicit addresses in all address fields has been included. Schulzrinne, et al. Expires March 15, 2014 [Page 310] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 * To handle URI definitions that contain ";" or "," a quoted URI format has been introduced and is required. * Defined IANA registries for the transport headers parameters, transport-protocol, profile, lower-transport, and mode. * The transport headers interleaved parameter's text was made more strict and uses formal requirements levels. It was also clarified that the interleaved channels are symmetric and that it is the server that sets the channel numbers. * It has been clarified that the client can't request of the server to use a certain RTP SSRC, using a request with the transport parameter SSRC. * Syntax definition for SSRC has been clarified to require 8HEX. It has also been extended to allow multiple values for clients supporting this version. * Clarified the text on the transport headers "dest_addr" parameters regarding what security precautions the server is required to perform. o The Range formats has been changed in the following way: * The NPT format has been given an initial NPT identifier that must now be used. * All formats now support initial open ended formats of type "npt=-10" and also format only "Range: smpte" ranges for usage with GET_PARAMETER requests. o RTSP message handling has been changed in the following way: * RTSP messages now use URIs rather then URLs. * It has been clarified that a 4xx message due to missing CSeq header shall be returned without a CSeq header. * The 300 (Multiple Choices) response code has been removed. * Rules for how to handle timing out RTSP messages has been added. * Extended Pipelining rules allowing for quick session startup. * Sequence numbering and proxy handling of sequence number defined, including case when response arrive out of order. Schulzrinne, et al. Expires March 15, 2014 [Page 311] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o The HTTP references have been updated to RFC 2616 and RFC 2617. Most of the text has been copied and then altered to fit RTSP into this specification. Public, and the Content-Base header has also been imported from RFC 2068 so that they are defined in the RTSP specification. Known effects on RTSP due to HTTP clarifications: * Content-Encoding header can include encoding of type "identity". o The state machine section has been completely rewritten. It now includes more details and is also more clear about the model used. o An IANA section has been included which contains a number of registries and their rules. This will allow us to use IANA to keep track of RTSP extensions. o The transport of RTSP messages has seen the following changes: * The use of UDP for RTSP message transport has been deprecated due to missing interest and to broken specification. * The rules for how TCP connections are to be handled has been clarified. Now it is made clear that servers should not close the TCP connection unless they have been unused for significant time. * Strong recommendations why server and clients should use persistent connections have also been added. * There is now a requirement on the servers to handle non- persistent connections as this provides fault tolerance. * Added wording on the usage of Connection:Close for RTSP. * Specified usage of TLS for RTSP messages, including a scheme to approve a proxy's TLS connection to the next hop. o The following header related changes have been made: * Accept-Ranges response header is added. This header clarifies which range formats that can be used for a resource. * Fixed the missing definitions for the Cache-Control header. Also added to the syntax definition the missing delta-seconds for max-stale and min-fresh parameters. * Put requirement on CSeq header that the value is increased by one for each new RTSP request. A Recommendation to start at 0 Schulzrinne, et al. Expires March 15, 2014 [Page 312] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 has also been added. * Added requirement that the Date header must be used for all messages with message body and the Server should always include it. * Removed possibility of using Range header with Scale header to indicate when it is to be activated, since it can't work as defined. Also added rule that lack of Scale header in response indicates lack of support for the header. Feature-tags for scaled playback has been defined. * The Speed header must now be responded to indicate support and the actual speed going to be used. A feature-tag is defined. Notes on congestion control were also added. * The Supported header was borrowed from SIP [RFC3261] to help with the feature negotiation in RTSP. * Clarified that the Timestamp header can be used to resolve retransmission ambiguities. * The Session header text has been expanded with an explanation on keep-alive and which methods to use. SET_PARAMETER is now recommended to use if only keep-alive within RTSP is desired. * It has been clarified how the Range header formats are used to indicate pause points in the PAUSE response. * Clarified that RTP-Info URIs that are relative, use the Request-URI as base URI. Also clarified that the used URI must be the one that was used in the SETUP request. The URIs are now also required to be quoted. The header also expresses the SSRC for the provided RTP timestamp and sequence number values. * Added text that requires the Range to always be present in PLAY responses. Clarified what should be sent in case of live streams. * The headers table has been updated using a structure borrowed from SIP. Those tables convey much more information and should provide a good overview of the available headers. * It has been clarified that any message with a message body is required to have a Content-Length header. This was the case in RFC 2326, but could be misinterpreted. Schulzrinne, et al. Expires March 15, 2014 [Page 313] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 * ETag has changed name to MTag. * To resolve functionality around MTag. The MTag and If-None- Match header have been added from HTTP with necessary clarification in regards to RTSP operation. * Imported the Public header from HTTP RFC 2068 [RFC2068] since it has been removed from HTTP due to lack of use. Public is used quite frequently in RTSP. * Clarified rules for populating the Public header so that it is an intersection of the capabilities of all the RTSP agents in a chain. * Added the Media-Range header for listing the current availability of the media range. * Added the Notify-Reason header for giving the reason when sending PLAY_NOTIFY requests. * A new header Seek-Style has been defined to direct and inform how any seek operation should/have been performed. o The Protocol Syntax has been changed in the following way: * All ABNF definitions are updated according to the rules defined in RFC 5234 [RFC5234] and have been gathered in a separate Section 20. * The ABNF for the User-Agent and Server headers have been corrected. * Some definitions in the introduction regarding the RTSP session have been changed. * The protocol has been made fully IPv6 capable. * The CHAR rule has been changed to exclude NULL. o The Status codes have been changed in the following way: * The use of status code 303 "See Other" has been deprecated as it does not make sense to use in RTSP. * When sending response 451 and 458 the response body should contain the offending parameters. Schulzrinne, et al. Expires March 15, 2014 [Page 314] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 * Clarification on when a 3rr redirect status code can be received has been added. This includes receiving 3rr as a result of a request within a established session. This provides clarification to a previous unspecified behavior. * Removed the 201 (Created) and 250 (Low On Storage Space) status codes as they are only relevant to recording, which is deprecated. * Several new Status codes have been defined: 464 "Data Transport Not Ready Yet", 465 "Notification Reason Unknown", 470 "Connection Authorization Required", 471 "Connection Credentials not accepted", 472 "Failure to establish secure connection". o The following functionality has been deprecated from the protocol: * The use of Queued Play. * The use of PLAY method for keep-alive in Play state. * The RECORD and ANNOUNCE methods and all related functionality. Some of the syntax has been removed. * The possibility to use timed execution of methods with the time parameter in the Range header. * The description on how rtspu works is not part of the core specification and will require external description. Only that it exists is defined here and some requirements for the transport is provided. o The following changes have been made in relation to methods: * The OPTIONS method has been clarified with regards to the use of the Public and Allow headers. * Added text clarifying the usage of SET_PARAMETER for keep-alive and usage without any body. * PLAY method is now allowed to be pipelined with the pipelining of one or more SETUP requests following the initial that generates the session for aggregated control. * REDIRECT has been expanded and diversified for different situations. Schulzrinne, et al. Expires March 15, 2014 [Page 315] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 * Added a new method PLAY_NOTIFY. This method is used by the RTSP server to asynchronously notify clients about session changes. o Wrote a new section about how to setup different media transport alternatives and their profiles, and lower layer protocols. This caused the appendix on RTP interaction to be moved there instead of being in the part which describes RTP. The section also includes guidelines what to consider when writing usage guidelines for new protocols and profiles. o Setup and usage of independent TCP connections for transport of RTP has been specified. o Added a new section describing the available mechanisms to determine if functionality is supported, called "Capability Handling". Renamed option-tags to feature-tags. o Added a contributors section with people who have contributed actual text to the specification. o Added a section Use Cases that describes the major use cases for RTSP. o Clarified the usage of a=range and how to indicate live content that are not seekable with this header. o Text specifying the special behavior of PLAY for live content. o Security features of RTSP has been clarified: * HTTP based authorization has been clarified requring both Basic and DIGEST support * TLS support mandated * IF one implements RTP then SRTP and defined MIKEY based key- exchange must be supported * Various minor mitigations discussed or resulted in protocol changes. Schulzrinne, et al. Expires March 15, 2014 [Page 316] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix J. Acknowledgements This memorandum defines RTSP version 2.0 which is a revision of the Proposed Standard RTSP version 1.0 which is defined in [RFC2326]. The authors of RFC 2326 are Henning Schulzrinne, Anup Rao, and Robert Lanphier. Both RTSP version 1.0 and RTSP version 2.0 borrow format and descriptions from HTTP/1.1. 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, Maureen Chesire, Jinhang Choi, Francisco Cortes, Elwyn Davies, Kelly Djahandari, Martin Dunsmuir, Stephen Farrell, Ross Finlayson, Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt, John K. Ho, Patrick Hoffman, Go Hori, Philipp Hoschka, Anne Jones, Ingemar Johansson, Jae-Hwan Kim, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Chris Lonvick, Xavier Marjou, Thomas Marshall, Rob McCool, Martti Mela, David Oran, Joerg Ott, Joe Pallas, Maria Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Pekka Pessi, Igor Plotnikov, Holger Schmidt, Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith, Alexander Sokolsky, Dale Stammen, John Francis Stracke, Geetha Srikantan, Scott Taylor, David Walker, Stephan Wenger, Dale R. Worley, and Byungjo Yoon , and especially to Flemming Andreasen. J.1. Contributors The following people have made written contributions that were included in the specification: o Tom Marshall contributed text on the usage of 3rr status codes. o Thomas Zheng contributed text on the usage of the Range in PLAY responses and proposed an earlier version of the PLAY_NOTIFY method. o Sean Sheedy contributed text on the timeout behavior of RTSP messages and connections, the 463 status code, and proposed an earlier version of the PLAY_NOTIFY method. Schulzrinne, et al. Expires March 15, 2014 [Page 317] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 o Greg Sherwood proposed an earlier version of the PLAY_NOTIFY method. o Fredrik Lindholm contributed text about the RTSP security framework. o John Lazzaro contributed the text for RTP over Independent TCP. o Aravind Narasimhan contributed by rewriting Media Transport Alternatives (Appendix C) and editorial improvements on a number of places in the specification. o Torbjorn Einarsson has done some editorial improvements of the text. Schulzrinne, et al. Expires March 15, 2014 [Page 318] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Appendix K. RFC Editor Consideration Please replace RFC XXXX with the RFC number this specification receives. Schulzrinne, et al. Expires March 15, 2014 [Page 319] Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) September 2013 Authors' Addresses Henning Schulzrinne Columbia University 1214 Amsterdam Avenue New York, NY 10027 USA Email: schulzrinne@cs.columbia.edu Anup Rao Cisco USA Email: anrao@cisco.com Rob Lanphier Seattle, WA USA Email: robla@robla.net Magnus Westerlund Ericsson AB Faeroegatan 6 STOCKHOLM, SE-164 80 SWEDEN Email: magnus.westerlund@ericsson.com Martin Stiemerling NEC Laboratories Europe, NEC Europe Ltd. Kurfuersten-Anlage 36 Heidelberg 69115 Germany Phone: +49 (0) 6221 4342 113 Email: martin.stiemerling@neclab.eu URI: http://ietf.stiemerling.org Schulzrinne, et al. Expires March 15, 2014 [Page 320]