Internet Engineering Task Force MMUSIC WG Internet Draft H. Schulzrinne/Columbia U. draft-ietf-mmusic-rfc2326bis-06.txt A. Rao/Cisco February 16, 2004 R. Lanphier/RealNetworks Expires: August, 2004 M. Westerlund/Ericsson A. Narasimhan/Sun Real Time Streaming Protocol (RTSP) STATUS OF THIS MEMO This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress". The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt To view the list Internet-Draft Shadow Directories, see http://www.ietf.org/shadow.html. Abstract This memorandum is a revision of RFC 2326, which is currently a Proposed Standard. The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP (RFC 3550). H. Schulzrinne et. al. [Page 1]
Internet Draft RTSP February 16, 2004 H. Schulzrinne et. al. [Page 2]
Internet Draft RTSP February 16, 2004 Table of Contents 1 Introduction ........................................ 8 1.1 The Update of the RTSP Specification ................ 8 1.2 Purpose ............................................. 9 1.3 Notational Conventions .............................. 11 1.4 Terminology ......................................... 11 1.5 Protocol Properties ................................. 14 1.6 Extending RTSP ...................................... 16 1.7 Overall Operation ................................... 17 1.8 RTSP States ......................................... 18 1.9 Relationship with Other Protocols ................... 19 2 RTSP Use Cases ...................................... 19 2.1 On-demand Playback of Stored Content ................ 20 2.2 Unicast distribution of Live Content ................ 20 2.3 Inviting RTSP on-demand servers into a multicast group ............................................... 20 2.4 On-demand Playback using Multicast .................. 20 3 Protocol Parameters ................................. 20 3.1 RTSP Version ........................................ 20 3.2 RTSP URL ............................................ 20 3.3 Session Identifiers ................................. 22 3.4 SMPTE Relative Timestamps ........................... 22 3.5 Normal Play Time .................................... 22 3.6 Absolute Time ....................................... 23 3.7 Feature-tags ........................................ 23 4 RTSP Message ........................................ 24 4.1 Message Types ....................................... 24 4.2 Message Headers ..................................... 24 4.3 Message Body ........................................ 25 4.4 Message Length ...................................... 25 5 General Header Fields ............................... 25 6 Request ............................................. 25 6.1 Request Line ........................................ 26 6.2 Request Header Fields ............................... 27 7 Response ............................................ 27 7.1 Status-Line ......................................... 28 7.1.1 Status Code and Reason Phrase ....................... 28 7.1.2 Response Header Fields .............................. 29 8 Entity .............................................. 30 8.1 Entity Header Fields ................................ 32 8.2 Entity Body ......................................... 32 9 Connections ......................................... 32 H. Schulzrinne et. al. [Page 3]
Internet Draft RTSP February 16, 2004 9.1 Pipelining .......................................... 33 9.2 Reliability and Acknowledgements .................... 33 9.3 Unreliable Transport ................................ 34 9.4 The usage of connections ............................ 34 9.5 Timing Out RTSP messages ............................ 36 9.6 Use of IPv6 ......................................... 36 10 Capability Handling ................................. 37 11 Method Definitions .................................. 38 11.1 OPTIONS ............................................. 39 11.2 DESCRIBE ............................................ 40 11.3 SETUP ............................................... 42 11.4 PLAY ................................................ 44 11.5 PAUSE ............................................... 48 11.6 TEARDOWN ............................................ 51 11.7 GET_PARAMETER ....................................... 52 11.8 SET_PARAMETER ....................................... 53 11.9 REDIRECT ............................................ 55 11.10 PING ................................................ 56 12 Embedded (Interleaved) Binary Data .................. 57 13 Status Code Definitions ............................. 59 13.1 Success 1xx ......................................... 59 13.1.1 100 Continue ........................................ 59 13.2 Success 2xx ......................................... 59 13.3 Redirection 3xx ..................................... 59 13.3.1 TBW ................................................. 60 13.3.2 301 Moved Permanently ............................... 60 13.3.3 302 Found ........................................... 60 13.3.4 303 See Other ....................................... 60 13.3.5 304 Not Modified .................................... 60 13.3.6 305 Use Proxy ....................................... 61 13.4 Client Error 4xx .................................... 61 13.4.1 400 Bad Request ..................................... 61 13.4.2 405 Method Not Allowed .............................. 61 13.4.3 451 Parameter Not Understood ........................ 61 13.4.4 452 reserved ........................................ 61 13.4.5 453 Not Enough Bandwidth ............................ 61 13.4.6 454 Session Not Found ............................... 62 13.4.7 455 Method Not Valid in This State .................. 62 13.4.8 456 Header Field Not Valid for Resource ............. 62 13.4.9 457 Invalid Range ................................... 62 13.4.10 458 Parameter Is Read-Only .......................... 62 13.4.11 459 Aggregate Operation Not Allowed ................. 62 13.4.12 460 Only Aggregate Operation Allowed ................ 62 13.4.13 461 Unsupported Transport ........................... 62 13.4.14 462 Destination Unreachable ......................... 63 13.5 Server Error 5xx .................................... 63 13.5.1 551 Option not supported ............................ 63 14 Header Field Definitions ............................ 63 H. Schulzrinne et. al. [Page 4]
Internet Draft RTSP February 16, 2004 14.1 Accept .............................................. 65 14.2 Accept-Encoding ..................................... 65 14.3 Accept-Language ..................................... 66 14.4 Accept-Ranges ....................................... 66 14.5 Allow ............................................... 66 14.6 Authorization ....................................... 66 14.7 Bandwidth ........................................... 66 14.8 Blocksize ........................................... 68 14.9 Cache-Control ....................................... 70 14.10 Connection .......................................... 72 14.11 Content-Base ........................................ 73 14.12 Content-Encoding .................................... 73 14.13 Content-Language .................................... 73 14.14 Content-Length ...................................... 73 14.15 Content-Location .................................... 73 14.16 Content-Type ........................................ 73 14.17 CSeq ................................................ 73 14.18 Date ................................................ 74 14.19 Expires ............................................. 74 14.20 From ................................................ 75 14.21 Host ................................................ 75 14.22 If-Match ............................................ 75 14.23 If-Modified-Since ................................... 75 14.24 Last-Modified ....................................... 76 14.25 Location ............................................ 76 14.26 Proxy-Authenticate .................................. 76 14.27 Proxy-Require ....................................... 76 14.28 Public .............................................. 77 14.29 Range ............................................... 77 14.30 Referer ............................................. 79 14.31 Retry-After ......................................... 79 14.32 Require ............................................. 79 14.33 RTP-Info ............................................ 80 14.34 Scale ............................................... 82 14.35 Speed ............................................... 82 14.36 Server .............................................. 83 14.37 Session ............................................. 83 14.38 Supported ........................................... 85 14.39 Timestamp ........................................... 86 14.40 Transport ........................................... 86 14.41 Unsupported ......................................... 92 14.42 User-Agent .......................................... 92 14.43 Vary ................................................ 93 14.44 Via ................................................. 93 14.45 WWW-Authenticate .................................... 93 15 Caching ............................................. 93 16 Examples ............................................ 94 16.1 Media on Demand (Unicast) ........................... 94 H. Schulzrinne et. al. [Page 5]
Internet Draft RTSP February 16, 2004 16.2 Streaming of a Container file ....................... 97 16.3 Single Stream Container Files ....................... 100 16.4 Live Media Presentation Using Multicast ............. 101 16.5 Capability Negotiation .............................. 103 17 Syntax .............................................. 104 17.1 Base Syntax ......................................... 104 17.2 RTSP Protocol Definition ............................ 105 17.2.1 Generic Protocol elements ........................... 105 17.2.2 Message Syntax ...................................... 106 17.2.3 Header Syntax ....................................... 110 18 Security Considerations ............................. 112 19 IANA Considerations ................................. 115 19.1 Feature-tags ........................................ 115 19.1.1 Description ......................................... 115 19.1.2 Registering New Feature-tags with IANA .............. 116 19.1.3 Registered entries .................................. 116 19.2 RTSP Methods ........................................ 116 19.2.1 Description ......................................... 116 19.2.2 Registering New Methods with IANA ................... 116 19.2.3 Registered Entries .................................. 117 19.3 RTSP Status Codes ................................... 117 19.3.1 Description ......................................... 117 19.3.2 Registering New Status Codes with IANA .............. 117 19.3.3 Registered Entries .................................. 117 19.4 RTSP Headers ........................................ 117 19.4.1 Description ......................................... 117 19.4.2 Registering New Headers with IANA ................... 118 19.4.3 Registered entries .................................. 118 19.5 Transport Header registries ......................... 118 19.5.1 Transport Protocols ................................. 119 19.5.2 Profile ............................................. 119 19.5.3 Lower Transport ..................................... 119 19.5.4 Transport modes ..................................... 120 19.6 Cache Directive Extensions .......................... 120 19.7 SDP attributes ...................................... 121 A RTSP Protocol State Machine ......................... 122 A.1 States .............................................. 122 A.2 State variables ..................................... 122 A.3 Abbreviations ....................................... 122 A.4 State Tables ........................................ 123 B Media Transport Alternatives ........................ 125 B.1 RTP ................................................. 126 B.1.1 AVP ................................................. 126 B.1.2 AVP/UDP ............................................. 127 B.1.3 AVP/TCP ............................................. 128 B.1.4 Handling NPT Jumps in the RTP Media Layer ........... 129 B.1.5 Handling RTP Timestamps after PAUSE ................. 131 B.1.6 RTSP / RTP Integration .............................. 133 H. Schulzrinne et. al. [Page 6]
Internet Draft RTSP February 16, 2004 B.1.7 Scaling with RTP .................................... 133 B.1.8 Maintaining NPT synchronization with RTP timesatmps .......................................... 133 B.1.9 Continuous Audio .................................... 134 B.1.10 Multiple Sources in an RTP Session .................. 134 B.1.11 Usage of SSRCs and the RTCP BYE Message During a RTSP Session ........................................ 134 B.2 Future Additions .................................... 134 C Use of SDP for RTSP Session Descriptions ............ 135 C.1 Definitions ......................................... 135 C.1.1 Control URL ......................................... 135 C.1.2 Media Streams ....................................... 136 C.1.3 Payload Type(s) ..................................... 137 C.1.4 Format-Specific Parameters .......................... 137 C.1.5 Range of Presentation ............................... 137 C.1.6 Time of Availability ................................ 138 C.1.7 Connection Information .............................. 138 C.1.8 Entity Tag .......................................... 139 C.2 Aggregate Control Not Available ..................... 139 C.3 Aggregate Control Available ......................... 140 C.4 RTSP external SDP delivery .......................... 141 D Minimal RTSP implementation ......................... 141 D.1 Client .............................................. 141 D.1.1 Basic Playback ...................................... 142 D.1.2 Authentication-enabled .............................. 142 D.2 Server .............................................. 143 D.2.1 Basic Playback ...................................... 143 D.2.2 Authentication-enabled .............................. 144 E Open Issues ......................................... 144 F Changes ............................................. 145 G Author Addresses .................................... 151 H Contributors ........................................ 152 I Acknowledgements .................................... 152 J Normative References ................................ 152 K Informative References .............................. 154 H. Schulzrinne et. al. [Page 7]
Internet Draft RTSP February 16, 2004 1 Introduction 1.1 The Update of the RTSP Specification This is the draft to an update of RTSP which is currently a proposed | standard defined in RFC 2326 [1]. Many flaws have been found in RTSP | since it was published. While this draft tries to address the flaws, | not all known issues have been resolved. However in this version only | a few remains, please see Open Issues in section E. The goal of the current work on RTSP is to progress it to draft standard status. Whether this is possible without first republishing RTSP as a proposed standard depends on the changes necessary to make the protocol work. The list of changes in appendix F indicates the issues that have already been addressed. The currently open issues are listed in appendix E. There is also a list of reported bugs available at "http://rtspspec.sourceforge.net". These bugs should be taken into account when reading this specification. While a lot of these bugs are addressed, not all are yet accounted for in this specification. Input on the unresolved bugs and other issues can be sent via e-mail to the MMUSIC WG's mailing list mmusic@ietf.org and the authors. Not all of the contents of RFC 2326 are part of this draft. In an | attempt to prevent the draft from exploding in size, the | specification has been reduced and split. The content of this draft | is the core specification of the protocol. It contains the general | idea behind RTSP and the basic functionality necessary to establish | an on-demand play-back session. It also contains the mechanisms for | extending the protocol. Any other functionality will be published as | extension documents. The Working group currently is working on: | o NAT and FW traversal mechanisms for RTSP are described in a | document called "How to make Real-Time Streaming Protocol | (RTSP) traverse Network Address Translators (NAT) and interact | with Firewalls." [20]. | There have also been discussion or proposals about the following | extensions to RTSP: | o Mute and Unmute Extension [21]. | o RTSP Stream Switching [22]. | o Live Streaming Relays [23]. | o Transport security for RTSP messages (rtsps). | H. Schulzrinne et. al. [Page 8]
Internet Draft RTSP February 16, 2004 o Unreliable transport of RTSP messages (rtspu). | o The Record functionality. | o A text body type with suitable syntax for basic parameters to | be used in SET_PARAMETER, and GET_PARAMETER. Including IANA | registry within the defined name space. | o A RTSP MIB. | 1.2 Purpose The Real-Time Streaming Protocol (RTSP) establishes and controls single or several time-synchronized streams of continuous media such as audio and video. Put simply, RTSP acts as a "network remote control" for multimedia servers. There is no notion of a RTSP connection in the protocol. Instead, a RTSP server maintains a session labelled by an identifier to associate groups of media streams and their states. A RTSP session is not tied to a transport-level connection such as a TCP connection. During a session, a client may open and close many reliable transport connections to the server to issue RTSP requests for that session. This memorandum describes the use of RTSP over a reliable connection based transport level protocol such as TCP. RTSP may be implemented over an unreliable connectionless transport protocol such as UDP. While nothing in RTSP precludes this, additional definition of this problem area must be handled as an extension to the core specification. The mechanisms of RTSP's operation over UDP were left out of this spec. because they were poorly defined in RFC 2326 [1] and the tradeoff in size and complexity of this spec. for a small gain in a targeted problem space was not deemed justifiable. The set of streams to be controlled is defined by a presentation | description. This memorandum does not define a format for the | presentation description. However appendix C defines how SDP [2] is | used for this purpose. The streams controlled by RTSP may use RTP [3] | for their data transport, but the operation of RTSP does not depend | on the transport mechanism used to carry continuous media. The | protocol is intentionally similar in syntax and operation to HTTP/1.1 | [4] so that extension mechanisms to HTTP can in most cases also be | added to RTSP. However, RTSP differs in a number of important aspects | H. Schulzrinne et. al. [Page 9]
Internet Draft RTSP February 16, 2004 from HTTP: o RTSP introduces a number of new methods and has a different protocol identifier. o RTSP has the notion of a session built into the protocol. o A RTSP server needs to maintain state by default in almost all cases, as opposed to the stateless nature of HTTP. o Both a RTSP server and client can issue requests. o Data is usually carried out-of-band by a different protocol. Session descriptions returned in a DESCRIBE response (see Section 11.2) and interleaving of RTP with RTSP over TCP are exceptions to this rule (see Section 12). | o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO | 8859-1, consistent with HTML internationalization efforts | [24]. o The Request-URL always contains the absolute URL. Because of backward compatibility with a historical blunder, HTTP/1.1 [4] carries only the absolute path in the request and puts the host name in a separate header field. This makes "virtual hosting" easier, where a single host with one IP address hosts several document trees. The protocol supports the following operations: Retrieval of media from media server: The client can either | request a presentation description via RTSP DESCRIBE, HTTP | or some other method. If the presentation is being | multicast, the presentation description contains the | multicast addresses and ports to be used for the continuous | media. If the presentation is to be sent only to the client | via unicast, the client provides the destination for | security reasons. Invitation of a media server to a conference: A media server can be "invited" to join an existing conference to play back media into the presentation. This mode is useful for distributed teaching applications. Several parties in the conference may take turns "pushing the remote control buttons". H. Schulzrinne et. al. [Page 10]
Internet Draft RTSP February 16, 2004 RTSP requests may be handled by proxies, tunnels and caches as in HTTP/1.1 [4]. 1.3 Notational Conventions | Since many of the definitions and syntax are identical to HTTP/1.1, | this specification only points to the section where they are defined | rather than copying it. For brevity, [HX.Y] is to be taken to refer | to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [4]). | All the mechanisms specified in this document are described in both | prose and the augmented Backus-Naur form (BNF) described in detail in | RFC 2234 [5]. | In this specification, we use indented and smaller-type paragraphs to | provide 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 that they are in RTSP. | The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", | "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this | document are to be interpreted as described in RFC 2119 [6]. | This specification uses the word "Unspecified" to indicate | functionality or features that are not defined in this specification, | and therefore can't be used. Any such functionality or feature can in | the future be evaluated and if technical sound be defined in | specification extending RTSP. 1.4 Terminology Some of the terminology has been adopted from HTTP/1.1 [4]. Terms not listed here are defined as in HTTP/1.1. Aggregate control: The concept of controlling multiple streams using a single timeline, generally maintained by the server. A client, for example, uses aggregate control when it issues a single play or pause message to simultaneously control both the audio and video in a movie. Aggregate control URL: The URL used in a RTSP request to refer | to and control an aggregated session. It normally, but not | always, corresponds to the presentation URL specified in | the session description. See Section 11.3 for more | information. H. Schulzrinne et. al. [Page 11]
Internet Draft RTSP February 16, 2004 Conference: a multiparty, multimedia presentation, where "multi" implies greater than or equal to one. Client: The client requests media service from the media server. Connection: A transport layer virtual circuit established between two programs for the purpose of communication. Container file: A file which may contain multiple media streams which often comprise a presentation when played together. RTSP servers may offer aggregate control on these files, though the concept of a container file is not embedded in the protocol. Continuous media: Data where there is a timing relationship between source and sink; that is, the sink must 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), where there is a "tight" timing relationship between source and sink, or streaming (playback), where the relationship is less strict. Entity: The information transferred as the payload of a request or response. An entity consists of meta-information in the form of entity-header fields and content in the form of an entity-body, as described in Section 8. Feature-tag: A tag representing a certain set of functionality, i.e. a feature. Live: Normally used to describe a presentation or session with | media coming from ongoing event. This generally results in | that the session has a unbound or only loosely defined | duration, and that no seek operations are possible. Media initialization: Datatype/codec specific initialization. This includes such things as clockrates, color tables, etc. Any transport-independent information which is required by a client for playback of a media stream occurs in the media initialization phase of stream setup. Media parameter: Parameter specific to a media type that may be changed before or during stream playback. Media server: The server providing playback services for one or more media streams. Different media streams within a H. Schulzrinne et. al. [Page 12]
Internet Draft RTSP February 16, 2004 presentation may originate from different media servers. A media server may reside on the same or a different host as the web server the presentation is invoked from. Media server indirection: Redirection of a media client to a different media server. (Media) stream: A single media instance, e.g., an audio stream or a video stream as well as a single whiteboard or shared application group. When using RTP, a stream consists of all RTP and RTCP packets created by a source within an RTP session. This is equivalent to the definition of a DSM-CC stream([25]). Message: The basic unit of RTSP communication, consisting of a structured sequence of octets matching the syntax defined in Section 17 and transmitted via a connection or a connectionless protocol. Non-Aggregated Control: Control of a single media stream. Only possible in RTSP sessions with a single media. Participant: Member of a conference. A participant may be a machine, e.g., a playback server. Presentation: A set of one or more streams presented to the client as a complete media feed, using a presentation description as defined below. In most cases in the RTSP context, this implies aggregate control of those streams, but does not have to. 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 (RFC 2327 [2]) use the term "session" for a presentation. The presentation description may take several different formats, including but not limited to the session description format SDP. Response: A RTSP response. If an HTTP response is meant, that is indicated explicitly. Request: A RTSP request. If an HTTP request is meant, that is indicated explicitly. Request URL: The URL used in a request to indicate the resource | H. Schulzrinne et. al. [Page 13]
Internet Draft RTSP February 16, 2004 which the request shall be performed on. RTSP session: A stateful abstraction upon which the main control methods of RTSP operate. A RTSP session is a server entity; it is created, maintained and destroyed by the server. It is established by a RTSP server upon the completion of a successful SETUP request (when 200 OK response is sent) and is labelled by a session identifier at that time. The session exists until timed out by the server or explicitly removed by a TEARDOWN request. A RTSP session is also a stateful entity; a RTSP server maintains an explicit session state machine (see Appendix A) where most state transitions are triggered by client requests. The existence of a session implies the existence of state about the session's media streams and their respective transport mechanisms. A given session can have zero or more media streams associated with it. A RTSP server uses the session to aggregate control over multiple media streams. Transport initialization: The negotiation of transport information (e.g., port numbers, transport protocols) between the client and the server. URI: Universal Resource Identifier, see RFC 2396 [12]. In RTSP | the used URIs are as general rule in fact URL's as they | gives an location for the resource. Therefore although RTSP | URLs are a subset of URIs, they will be refered as URLs. | URL: Universal Resource Locator, is an URI which identifies the | resource through its primary access mechanism, rather than | identifying the resource by name or by some other | attribute(s) of that resource. 1.5 Protocol Properties RTSP has the following properties: Extendable: New methods and parameters can be easily added to RTSP. Easy to parse: RTSP can be parsed by standard HTTP or MIME parsers. Secure: RTSP re-uses web security mechanisms, either at the transport level (TLS, RFC 2246 [26]) or within the protocol itself. All HTTP authentication mechanisms such as basic (RFC 2616 [4]) and digest authentication (RFC 2617 [7]) are H. Schulzrinne et. al. [Page 14]
Internet Draft RTSP February 16, 2004 directly applicable. Transport-independent: RTSP does not preclude the use of an | unreliable datagram protocol (UDP) (RFC 768 [8]), a | reliable datagram protocol (RDP, RFC 1151, not widely used | [27]) as it would be possible to implement application- | level reliability. The use of a connectionless datagram | protocol such as UDP or RDP requires additional definition | that may be provided as extensions to the core RTSP | specification. The usage of the reliable stream protocol | TCP (RFC 793 [9]) is what is currently defined as transport | protocol of RTSP messages. Multi-server capable: Each media stream within a presentation can reside on a different server. The client automatically establishes several concurrent control sessions with the different media servers. Media synchronization is performed at the transport level. Separation of stream control and conference initiation: Stream control is divorced from inviting a media server to a conference. In particular, SIP [28] or H.323 [29] may be used to invite a server to a conference. Suitable for professional applications: RTSP supports frame- level accuracy through SMPTE time stamps to allow remote digital editing. Presentation description neutral: The protocol does not impose a particular presentation description or metafile format and can convey the type of format to be used. However, the presentation description must contain at least one RTSP URL. Proxy and firewall friendly: The protocol should be readily | handled by both application and transport-layer (SOCKS | [30]) firewalls. A firewall may need to understand the | SETUP method to open a "hole" for the media stream. HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so that the existing infrastructure can be reused. This infrastructure includes PICS (Platform for Internet Content Selection [31,32]) for associating labels with content. However, RTSP does not just add methods to HTTP since the controlling continuous media requires server state in most cases. H. Schulzrinne et. al. [Page 15]
Internet Draft RTSP February 16, 2004 Appropriate server control: If a client can start a stream, it must be able to stop a stream. Servers should not start streaming to clients in such a way that clients cannot stop the stream. Transport negotiation: The client can negotiate the transport method prior to actually needing to process a continuous media stream. Capability negotiation: If basic features are disabled, there must be some clean mechanism for the client to determine which methods are not going to be implemented. This allows clients to present the appropriate user interface. For example, if seeking is not allowed, the user interface must be able to disallow moving a sliding position indicator. An earlier requirement in RTSP was multi-client capability. However, it was determined that a better approach was to make sure that the protocol is easily extensible to the multi-client scenario. Stream identifiers can be used by several control streams, so that "passing the remote" would be possible. The protocol would not address how several clients negotiate access; this is left to either a "social protocol" or some other floor control mechanism. 1.6 Extending RTSP Since not all media servers have the same functionality, media servers by necessity will support different sets of requests. For example: o A server may not be capable of seeking (absolute positioning) if it is to support live events only. o Some servers may not support setting stream parameters and thus not support GET_PARAMETER and SET_PARAMETER. A server SHOULD implement all header fields described in Section 14. It is up to the creators of presentation descriptions not to ask the impossible of a server. This situation is similar in HTTP/1.1 [4], where the methods described in [H19.5] are not likely to be supported across all servers. RTSP can be extended in three ways, listed here in order of the magnitude of changes supported: H. Schulzrinne et. al. [Page 16]
Internet Draft RTSP February 16, 2004 o Existing methods can be extended with new parameters, e.g. | headers, as long as these parameters can be safely ignored by | the recipient. (This is equivalent to adding new parameters to | an HTML tag.) If the client needs negative acknowledgement | when a method extension is not supported, a tag corresponding | to the extension may be added in the Require: field (see | Section 14.32). o New methods can be added. If the recipient of the message does not understand the request, it responds with error code 501 (Not Implemented) and the sender should not attempt to use this method again. A client may also use the OPTIONS method to inquire about methods supported by the server. The server SHOULD 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. The basic capability discovery mechanism can be used to both discover support for a certain feature and to ensure that a feature is available when performing a request. For detailed explanation of this see chapter 10. 1.7 Overall Operation Each presentation and media stream is identified by an RTSP URL. The overall presentation and the properties of the media the presentation is made up of are defined by a presentation description file, the format of which is outside the scope of this specification. The presentation description file may be obtained by the client using HTTP or other means such as email and may not necessarily be stored on the media server. For the purposes of this specification, a presentation description is assumed to describe one or more presentations, each of which maintains a common time axis. For simplicity of exposition and without loss of generality, it is assumed that the presentation description contains exactly one such presentation. A presentation may contain several media streams. The presentation description file contains a description of the media streams making up the presentation, including their encodings, language, and other parameters that enable the client to choose the most appropriate combination of media. In this presentation description, each media stream that is individually controllable by RTSP is identified by a RTSP URL, which points to the media server H. Schulzrinne et. al. [Page 17]
Internet Draft RTSP February 16, 2004 handling that particular media stream and names the stream stored on that server. Several media streams can be located on different servers; for example, audio and video streams can be split across servers for load sharing. The description also enumerates which transport methods the server is capable of. Besides the media parameters, the network destination address and port need to be determined. Several modes of operation can be distinguished: Unicast: The media is transmitted to the source of the RTSP request, with the port number chosen by the client. Alternatively, the media is transmitted on the same reliable stream as RTSP. Multicast, server chooses address: The media server picks the multicast address and port. This is the typical case for a live or near-media-on-demand transmission. Multicast, client chooses address: If the server is to participate in an existing multicast conference, the multicast address, port and encryption key are given by the conference description, established by means outside the scope of this specification. 1.8 RTSP States RTSP controls a stream which may be sent via a separate protocol, independent of the control channel. For example, RTSP control may occur on a TCP connection while the data flows via UDP. Thus, data delivery continues even if no RTSP requests are received by the media server. Also, during its lifetime, a single media stream may be controlled by RTSP requests issued sequentially on different TCP connections. Therefore, the server needs to maintain "session state" to be able to correlate RTSP requests with a stream. The state transitions are described in Appendix A. Many methods in RTSP do not contribute to state. However, the following play a central role in defining the allocation and usage of stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, PING and TEARDOWN. SETUP: Causes the server to allocate resources for a stream and create a RTSP session. PLAY: Starts data transmission on a stream allocated via SETUP. PAUSE: Temporarily halts a stream without freeing server H. Schulzrinne et. al. [Page 18]
Internet Draft RTSP February 16, 2004 resources. REDIRECT: Indicates that the session should be moved to new server / location PING: Prevents the identified session from being timed out. TEARDOWN: Frees resources associated with the stream. The RTSP session ceases to exist on the server. RTSP methods that contribute to state use the Session header field (Section 14.37) to identify the RTSP session whose state is being manipulated. The server generates session identifiers in response to SETUP requests (Section 11.3). 1.9 Relationship with Other Protocols RTSP has some overlap in functionality with HTTP. It also may interact with HTTP in that the initial contact with streaming content is often to be made through a web page. The current protocol specification aims to allow different hand-off points between a web server and the media server implementing RTSP. For example, the presentation description can be retrieved using HTTP or RTSP, which reduces roundtrips in web-browser-based scenarios, yet also allows for standalone RTSP servers and clients which do not rely on HTTP at all. However, RTSP differs fundamentally from HTTP in that most data delivery takes place out-of-band in a different protocol. HTTP is an asymmetric protocol where the client issues requests and the server responds. In RTSP, both the media client and media server can issue requests. RTSP requests are also stateful; they may set parameters and continue to control a media stream long after the request has been acknowledged. Re-using HTTP functionality has advantages in at least two areas, namely security and proxies. The requirements are very similar, so having the ability to adopt HTTP work on caches, proxies and authentication is valuable. RTSP assumes the existence of a presentation description format that can express both static and temporal properties of a presentation containing several media streams. Session Description Protocol (SDP) [2] is generally the format of choice; however, RTSP is not bound to it. For data delivery, most real-time media will use RTP as a transport protocol. While RTSP works well with RTP, it is not tied to RTP. 2 RTSP Use Cases H. Schulzrinne et. al. [Page 19]
Internet Draft RTSP February 16, 2004 This section describes some of the use cases RTSP can be used for. | They are listed in descending order of importance in regards to | ensuring that all necessary functionality is present. | TODO: Fill this headings with descriptions of the use cases. | 2.1 On-demand Playback of Stored Content | 2.2 Unicast distribution of Live Content | 2.3 Inviting RTSP on-demand servers into a multicast group | 2.4 On-demand Playback using Multicast | 3 Protocol Parameters 3.1 RTSP Version HTTP Specification Section [H3.1] applies, with HTTP replaced by RTSP. This specification defines version 1.0 of RTSP. 3.2 RTSP URL The "rtsp", "rtsps" and "rtspu" schemes are used to refer to network resources via the RTSP protocol. This section defines the scheme- specific syntax and semantics for RTSP URLs. The RTSP URL is case sensitive. Informative RTSP URL syntax: rtsp[u|s]://host[:port]/abspath[?query]#fragment See section 17.2.1 for the formal definition of the RTSP URL syntax. Note that fragment and query identifiers do not have a well-defined meaning at this time, i.e. their usage is unspecified, with the interpretation left to the RTSP server. The URL scheme rtsp requires that commands are issued via a reliable protocol (within the Internet, TCP), while the scheme rtspu is intended to identify RTSP over an unreliable protocol (within the Internet, UDP). The scheme rtsps is intended to identify a reliable transport using secure transport, perhaps TLS [26]. The rtspu and rtsps is not defined in this specification, and are for future H. Schulzrinne et. al. [Page 20]
Internet Draft RTSP February 16, 2004 extensions of the protocol to define how to use. If the port is empty or not given, port 554 SHALL be assumed. The | semantics are that the identified resource can be controlled by RTSP | at the server listening for TCP (scheme "rtsp") connections or UDP | (scheme "rtspu") packets on that port of host, and the Request-URL | for the resource is rtsp_URL. For the scheme rtsps the TCP and UDP | port 322 is registered and SHALL be assumed. The use of IP addresses in URLs SHOULD be avoided whenever possible (see RFC 1924 [10]). Note: Using qualified domain names in any URL is one requirement for making it possible for RFC 2326 implementations of RTSP to use IPv6. This specification is updated to allow for literal IPv6 addresses in RTSP URLs using the host specification in RFC 2732 [11]. A presentation or a stream is identified by a textual media identifier, using the character set and escape conventions [H3.2] of URLs (RFC 2396 [12]). URLs may refer to a stream or an aggregate of streams, i.e., a presentation. Accordingly, requests described in Section 11 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 URL: rtsp://media.example.com:554/twister/audiotrack identifies 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 URL: rtsp://media.example.com:554/twister identifies the presentation "twister", which may be composed of audio and video streams. This does not imply a standard way to reference streams in URLs. The presentation description defines the hierarchical relationships in the presentation and the URLs for the individual streams. A presentation description may name a stream "a.mov" and the whole presentation "b.mov". H. Schulzrinne et. al. [Page 21]
Internet Draft RTSP February 16, 2004 The path components of the RTSP URL are opaque to the client and do not imply any particular file system structure for the server. This decoupling also allows presentation descriptions to be used with non-RTSP media control protocols simply by replacing the scheme in the URL. 3.3 Session Identifiers Session identifiers are strings of any arbitrary length. A session identifier MUST be chosen randomly and MUST be at least eight characters long to make guessing it more difficult. (See Section 18.) 3.4 SMPTE Relative Timestamps A SMPTE relative timestamp expresses time relative to the start of the clip. Relative timestamps are expressed as SMPTE time codes for frame-level access accuracy. The time code has the format hours:minutes:seconds:frames.subframes, with the origin at the start of the clip. The default smpte format is"SMPTE 30 drop" format, with frame rate is 29.97 frames per second. Other SMPTE codes MAY be supported (such as "SMPTE 25") through the use of alternative use of "smpte time". For 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 value is zero, it may be omitted. Subframes are measured in one-hundredth of a frame. Examples: smpte=10:12:33:20- smpte=10:07:33- smpte=10:07:00-10:07:33:05.01 smpte-25=10:07:00-10:07:33:05.01 3.5 Normal Play Time Normal play time (NPT) indicates the stream absolute position relative to the beginning of the presentation, not to be confused with the Network Time Protocol (NTP). The timestamp consists of a decimal fraction. The part left of the decimal may be expressed in either seconds or hours, minutes, and seconds. The part right of the decimal point measures fractions of a second. H. Schulzrinne et. al. [Page 22]
Internet Draft RTSP February 16, 2004 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 type event. It MAY only be used for live type events, and SHALL NOT be used for on-demand content. NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the viewer associates with a program. It is often digitally displayed on a VCR. NPT advances normally when in normal play mode (scale = 1), advances at a faster rate when in fast scan forward (high positive scale ratio), decrements when in scan reverse (high negative scale ratio) and is fixed in pause mode. NPT is (logically) equivalent to SMPTE time codes." [25] Examples: npt=123.45-125 npt=12:05:35.3- npt=now- The syntax conforms to ISO 8601. The npt-sec notation is optimized for automatic generation, the ntp-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. 3.6 Absolute Time Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT). Fractions of a second may be indicated. Example for November 8, 1996 at 14h37 and 20 and a quarter seconds UTC: 19961108T143720.25Z 3.7 Feature-tags Feature-tags are unique identifiers used to designate features in RTSP. These tags are used in Require (Section 14.32), Proxy-Require (Section 14.27), Unsupported (Section 14.41), and Supported (Section 14.38) header fields. H. Schulzrinne et. al. [Page 23]
Internet Draft RTSP February 16, 2004 Feature tag needs to indicate if they apply to servers only, proxies only, or both server and proxies. 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 19. 4 RTSP Message RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding (RFC 2279 [13]). Lines are terminated by CRLF, but receivers should be prepared to also interpret CR and LF by themselves as line terminators. 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 10646 character set avoids tricky character set switching, but is invisible to the application as long as US-ASCII is being used. This is also the encoding used for RTCP. ISO 8859-1 translates directly into Unicode with a high-order octet of zero. ISO 8859-1 characters with the most-significant bit set are represented as 1100001x 10xxxxxx. (See RFC 2279 [13]) RTSP messages can be carried over any lower-layer transport protocol that is 8-bit clean. RTSP messages are vulnerable to bit errors and SHOULD NOT be subjected to them. 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. 4.1 Message Types See [H4.1]. 4.2 Message Headers H. Schulzrinne et. al. [Page 24]
Internet Draft RTSP February 16, 2004 See [H4.2]. 4.3 Message Body See [H4.3] 4.4 Message Length When a message body is included with a message, the length of that body is determined by one of the following (in order of precedence): 1. Any response message which MUST NOT include a message body (such as the 1xx, 204, and 304 responses) is always terminated by the first empty line after the header fields, regardless of the entity-header fields present in the message. (Note: An empty line consists of only CRLF.) 2. If a Content-Length header field (section 14.14) is present, its value in bytes represents the length of the message-body. If this header field is not present, a value of zero is assumed. Note that RTSP does not (at present) support the HTTP/1.1 "chunked" transfer coding(see [H3.6.1]) and requires the presence of the Content-Length header field. 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. 5 General Header Fields See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade, and Warning headers are not defined. RTSP further defines the CSeq, and Timestamp. The general headers are listed in table 1: 6 Request A request message from a client to a server or vice versa includes, | within the first line (Request Line) of that message, the method to | be applied to the resource, the identifier of the resource, and the | protocol version in use. Then follows zero or more headers that can | be of general (Section 5), request (Section 6.2), or entity (Section | 8.1) type. Then an empty line, i.e. a line with only the two | characters Carriage Return (CR) and Line Feed (LF), indicates the end | H. Schulzrinne et. al. [Page 25]
Internet Draft RTSP February 16, 2004 Header Name Comment _________________________________ Cache-Control See section 14.9 Connection See section 14.10 CSeq See section 14.17 Date See section 14.18 Supported See section 14.38 Timestamp See section 14.39 Via See section 14.44 Table 1: The General headers used in RTSP. of the header part. Optionally a message body (entity) follows to the | end of the message. The length of the message body is indicated | through the entity headers. | 6.1 Request Line | The request line, provides the most important things about the | request: What method, on what resources and using which RTSP version. | The methods that is defined by this specification can be seen in | Table 6.1. The resource is identified through an absolute RTSP URL | (see section 3.2. <Method> SP <Request-URL> SP <RTSP-Version> CRLF Please note: The request line's syntax can't be changed in future versions of RTSP, as this line indicates the version of the messages and need to be parsable also by older versions. Note that in contrast to HTTP/1.1 [4], RTSP requests always contain the absolute URL (that is, including the scheme, host and port) rather than just the absolute path. HTTP/1.1 requires servers to understand the absolute URL, 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. The asterisk "*" in the Request-URL means that the request does not apply to a particular resource, but to the server or proxy itself, and is only allowed when the method used does not necessarily apply H. Schulzrinne et. al. [Page 26]
Internet Draft RTSP February 16, 2004 Method Defined In Section _________________________________ DESCRIBE Section 11.2 GET_PARAMETER Section 11.7 OPTIONS Section 11.1 PAUSE Section 11.5 PLAY Section 11.4 PING Section 11.10 REDIRECT Section 11.9 SETUP Section 11.3 SET_PARAMETER Section 11.8 TEARDOWN Section 11.6 Table 2: The RTSP Methods to a resource. One example would be as follows: OPTIONS * RTSP/1.0 An OPTIONS in this form will determine the capabilities of the server or the proxy that first receives the request. If one needs to address the server explicitly, then one should use an absolute URL with the server's address. OPTIONS rtsp://example.com RTSP/1.0 6.2 Request Header Fields The RTSP headers in Table 6.2 can be included in a request with the purpose to give further define how the request should be fulfilled. A request header MAY also be response header, see section 7.1.2. 7 Response [H6] applies except that HTTP-Version is replaced by RTSP-Version. | Also, RTSP defines additional status codes and does not define some | H. Schulzrinne et. al. [Page 27]
Internet Draft RTSP February 16, 2004 Header Defined in Section _____________________________________ Accept Section 14.1 Accept-Encoding Section 14.2 Accept-Language Section 14.3 Authorization Section 14.6 Bandwidth Section 14.7 Blocksize Section 14.8 From Section 14.20 If-Match Section 14.22 If-Modified-Since Section 14.23 Proxy-Require Section 14.27 Range Section 14.29 Referer Section 14.30 Require Section 14.32 Scale Section 14.34 Session Section 14.37 Speed Section 14.35 Supported Section 14.38 Transport Section 14.40 User-Agent Section 14.42 Table 3: The RTSP request headers HTTP codes. The valid response codes and the methods they can be used | with are defined in Table 4. | After receiving and interpreting a request message, the recipient | responds with an RTSP response message. | 7.1 Status-Line | The first line of a Response message is the Status-Line, consisting | of the protocol version followed by a numeric status code, and the | textual phrase associated with the status code, with each element | separated by SP characters. No CR or LF is allowed except in the | final CRLF sequence. | <RTSP-Version> SP <Status-Code> SP <Reason-Phrase> CRLF | 7.1.1 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 13. The Reason-Phrase is intended to give a short | H. Schulzrinne et. al. [Page 28]
Internet Draft RTSP February 16, 2004 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: | o 1xx: Informational - Request received, continuing process | o 2xx: Success - The action was successfully received, | understood, and accepted | o 3rr: Redirection - Further action must be taken in order to | complete the request | o 4xx: Client Error - The request contains bad syntax or cannot | be fulfilled | o 5xx: Server Error - The server failed to fulfill an apparently | valid request | The individual values of the numeric status codes defined for | RTSP/1.0, and an example set of corresponding Reason-Phrase's, 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 [4] | status codes and adds RTSP-specific status codes starting at x50 to | avoid conflicts with newly defined HTTP status codes. | RTSP status codes are extensible. RTSP applications are not required | to understand the meaning of all registered status codes, though such | understanding is obviously desirable. However, applications MUST | understand the class of any status code, as indicated by the first | digit, and treat any unrecognized response as being equivalent to the | x00 status code of that class, with the exception that an | unrecognized response MUST NOT be cached. For example, if an | unrecognized status code of 431 is received by the client, it can | safely assume that there was something wrong with its request and | treat the response as if it had received a 400 status code. In such | cases, user agents SHOULD present to the user the entity returned | with the response, since that entity is likely to include human- | readable information which will explain the unusual status. | 7.1.2 Response Header Fields | H. Schulzrinne et. al. [Page 29]
Internet Draft RTSP February 16, 2004 The response-header fields allow the request recipient to pass | additional information about the response which cannot be placed in | the Status-Line. These header fields give information about the | server and about further access to the resource identified by the | Request-URL. All headers currently being classified as response | headers are listed in table 7.1.2. Header Defined in Section ______________________________________ Accept-Ranges Section 14.4 Location Section 14.25 Proxy-Authenticate Section 14.26 Public Section 14.28 Range Section 14.29 Retry-After Section 14.31 RTP-Info Section 14.33 Scale Section 14.34 Session Section 14.37 Server Section 14.36 Speed Section 14.35 Transport Section 14.40 Unsupported Section 14.41 Vary Section 14.43 WWW-Authenticate Section 14.45 Table 5: The RTSP response headers Response-header field names can be extended reliably only in combination with a change in the protocol version. However, new or experimental header fields MAY be given the semantics of response- header fields if all parties in the communication recognize them to be response-header fields. Unrecognized header fields are treated as entity-header fields. 8 Entity Request and Response messages MAY transfer an entity if not otherwise | restricted by the request method or response status code. An entity | consists of entity-header fields and an entity-body, although some | responses will only include the entity-headers. | The SET_PARAMETER, and GET_PARAMETER request and response, and | DESCRIBE response MAY have an entity. All 4xx and 5xx responses MAY | also have an entity. | H. Schulzrinne et. al. [Page 30]
Internet Draft RTSP February 16, 2004 Code Reason Method _______________________________________________________ 100 Continue all _______________________________________________________ 200 OK all 201 Created RECORD 250 Low on Storage Space RECORD _______________________________________________________ 300 Multiple Choices all 301 Moved Permanently all 302 Found all 303 See Other all 305 Use Proxy all 350 Going Away all 351 Load Balancing all _______________________________________________________ 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 411 Length Required all 412 Precondition Failed DESCRIBE, SETUP 413 Request Entity Too Large all 414 Request-URL Too Long all 415 Unsupported Media Type all 451 Parameter Not Understood SET_PARAMETER 452 reserved n/a 453 Not Enough Bandwidth SETUP 454 Session Not Found all 455 Method Not Valid In This State all 456 Header Field Not Valid all 457 Invalid Range PLAY, PAUSE 458 Parameter Is Read-Only SET_PARAMETER 459 Aggregate Operation Not Allowed all 460 Only Aggregate Operation Allowed all 461 Unsupported Transport all 462 Destination Unreachable all _______________________________________________________ 500 Internal Server Error all 501 Not Implemented all 502 Bad Gateway all 503 Service Unavailable all 504 Gateway Timeout all 505 RTSP Version Not Supported all H. Schulzrinne et. al. [Page 31]
Internet Draft RTSP February 16, 2004 Table 4: Status codes and their usage with RTSP methods In this section, both sender and recipient refer to either the client | or the server, depending on who sends and who receives the entity. | 8.1 Entity Header Fields | Entity-header fields define optional meta-information about the | entity-body or, if no body is present, about the resource identified | by the request. The entity header fields are listed in table 8.1. | Header Defined in Section ____________________________________ Allow Section 14.5 Content-Base Section 14.11 Content-Encoding Section 14.12 Content-Language Section 14.13 Content-Length Section 14.14 Content-Location Section 14.15 Content-Type Section 14.16 Expires Section 14.19 Last-Modified Section 14.24 Table 6: The RTSP entity headers The extension-header mechanism allows additional entity-header fields | to be defined without changing the protocol, but these fields cannot | be assumed to be recognizable by the recipient. Unrecognized header | fields SHOULD be ignored by the recipient and forwarded by proxies. | 8.2 Entity Body | See [H7.2] with the addition that a RTSP message with an entity body | MUST include the Content-Type and Content-Length headers. | 9 Connections RTSP requests can be transmitted in several different ways: o persistent transport connections used for several request- response transactions; o one connection per request/response transaction; H. Schulzrinne et. al. [Page 32]
Internet Draft RTSP February 16, 2004 o connectionless mode. The type of transport is defined by the RTSP URL (Section 3.2). For | the scheme "rtsp", a connection is assumed, while the scheme "rtspu" | calls for RTSP requests to be sent without setting up a connection. Unlike HTTP, RTSP allows the media server to send requests to the media client. However, this is only supported for persistent connections, as the media server otherwise has no reliable way of reaching the client. Also, this is the only way that requests from media server to client are likely to traverse firewalls. 9.1 Pipelining A client that supports persistent connections or connectionless mode MAY "pipeline" its requests (i.e., send multiple requests without waiting for each response). A server MUST send its responses to those requests in the same order that the requests were received. 9.2 Reliability and Acknowledgements The transmission of RTSP over UDP was optionally to implement and specified in RFC 2326. However that definition was not satisfactory for interoperable implementations. Due to lack of interest, this specification does not specify how RTSP over UDP shall be implemented. However to maintain backwards compatibility in the message format certain RTSP headers must be maintained. These mechanism are described below. The next section Unreliable Transport (section 9.3) provides documentation of certain features that are necessary for transport protocols like UDP. Any RTSP request according to this specification SHALL NOT be sent to a multicast address. Any RTSP request SHALL be acknowledged. If a reliable transport protocol is used to carry RTSP, requests MUST NOT be retransmitted; the RTSP application MUST instead rely on the underlying transport to provide reliability. If both the underlying reliable transport such as TCP and the RTSP application retransmit requests, it is possible that each packet loss results in two retransmissions. The receiver cannot typically 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. H. Schulzrinne et. al. [Page 33]
Internet Draft RTSP February 16, 2004 Each request carries a sequence number in the CSeq header (Section 14.17), which MUST be incremented by one for each distinct request transmitted to the destination end-point. The initial sequence number MAY be chosen arbitrary, but is RECOMMENDED to begin with 0. If a request is repeated because of lack of acknowledgement, the request MUST carry the original sequence number (i.e., the sequence number is not incremented). 9.3 Unreliable Transport This section provides some information to future specification of RTSP over unreliable transport. Requests 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). The round-trip time is | estimated as in TCP (RFC 1123) [14], with an initial round-trip value | of 500 ms. An implementation MAY cache the last RTT measurement as | the initial value for future connections. If RTSP is used over a small-RTT LAN, standard procedures for optimizing initial TCP round trip estimates, such as those used in T/TCP (RFC 1644) [33], can be beneficial. The Timestamp header (Section 14.39) is used to avoid the retransmission ambiguity problem [34] and obviates the need for Karn's algorithm. If a request is repeated because of lack of acknowledgement, the request must carry the original sequence number (i.e., the sequence number is not incremented). A number of RTSP messages destined for the same control end point may | be packed into a single lower-layer PDU. The default port for the RTSP server is 554 for UDP. 9.4 The usage of connections Systems implementing RTSP MUST support carrying RTSP over TCP. The | default port for the RTSP server is 554 for TCP. A number of RTSP | packets destined for the same control end point may be encapsulated | into a TCP stream. RTSP data MAY be interleaved with RTP and RTCP | packets, see section 12. Unlike HTTP, an RTSP message MUST contain a | Content-Length header field whenever that message contains a payload | (entity). Otherwise, an RTSP packet is terminated with an empty line | immediately following the last message header. H. Schulzrinne et. al. [Page 34]
Internet Draft RTSP February 16, 2004 TCP can be used for both persistent connections and for one message exchange per connection, as presented above. This section gives further rules and recommendations on how to handle these connections so maximum interoperability and flexibility can be achieved. A server SHALL handle both persistent connections and one request/response transaction per connection. A persistent connection MAY be used for all transactions between the server and client, including messages to multiple RTSP sessions. However the persistent connection MAY also be closed after a few message exchanges, e.g. the initial setup and play command in a session. Later when the client wishes to send a new request, e.g. pause, to the session a new connection is opened. This connection may either be for a single message exchange or can be kept open for several messages, i.e. persistent. A major motivation for allowing non-persistent connections are that they ensure fault tolerance. A second one is to allow for application layer mobility. A server and client supporting non-persistent connection can survive a 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. The client MAY close the connection at any time when no outstanding | request/response transactions exist. The server SHOULD NOT close the | connection unless at least one RTSP session timeout period has passed | without data traffic. A server SHOULD NOT initiate a close of a | connection directly after responding to a TEARDOWN request for the | whole session. A server SHOULD NOT close the connection as a result | of responding to a request with an error code. Doing this would | prevent or result in extra overhead for the client when testing | advanced or special types of requests. The client SHOULD NOT have more than one connection to the server at any given point. If a client or proxy handles multiple RTSP sessions on the same server, it is RECOMMENDED to use only a single connection. Older services which was implemented according to RFC 2326 sometimes requires the client to use persistent connection. The client closing the connection may result in that the server removes the session. To achieve interoperability with old servers any client is strongly RECOMMENDED to use persistent connections. A Client is also strongly RECOMMENDED to use persistent connections as it allows the server to send request to the client. In cases where no connection exist between the server and the client, this may cause the server to be forced to drop the RTSP session without H. Schulzrinne et. al. [Page 35]
Internet Draft RTSP February 16, 2004 notifying the client why, due to the lack of signalling channel. An example of such a case is when the server desires to send a REDIRECT request for a RTSP session to the client. A server implemented according to this specification MUST respond that it supports the "play.basic" feature-tag above. A client MAY send a request including the Supported header in a request to determine support of non-persistent connections. A server supporting non-persistent connections will return the "play.basic" feature-tag in its response. If the client receives the feature-tag in the response, it can be certain that the server handles non-persistent connections. 9.5 Timing Out RTSP messages | Receivers of a request (responder) SHOULD respond to requests in a | timely manner even when a reliable transport such as TCP is used. | Similarly, the sender of a request (requestor) SHOULD wait for a | sufficient time for a response before concluding that the responder | will not be acting upon its request. | A responder SHOULD respond to all requests within 5 seconds. If the | responder recognizes that processing of a request will take longer | than 5 seconds, it SHOULD send a 100 response as soon as possible. It | SHOULD continue sending a 100 response every 5 seconds thereafter | until it is ready to send the final response to the requestor. After | sending a 100 response, the receiver MUST send a final response | indicating the success or failure of the request. | A requestor SHOULD wait at least 10 seconds for a response before | concluding that the responder will not be responding to its request. | After receiving a 100 response, the requestor SHOULD continue waiting | for further responses. If more than 10 seconds elapses without | receiving any response, the requestor MAY assume the responder is | unresponsive and abort the connection. | A requestor SHOULD wait longer than 10 seconds for a response if it | is experiencing significant transport delays on its connection to the | responder. The requestor is capable of determining the RTT using the | Timestamp header (section 14.39) in any RTSP request. 9.6 Use of IPv6 This specification has been updated so that it supports IPv6. However this support was not present in RFC 2326 therefore some interoperability issues exist. A RFC 2326 implementation can support IPv6 as long as no explicit IPv6 addresses are used within RTSP H. Schulzrinne et. al. [Page 36]
Internet Draft RTSP February 16, 2004 messages. This require that any RTSP URL pointing at a IPv6 host must use fully qualified domain name and not a IPv6 address. Further the Transport header must not use the parameters source and destination. Implementations according to this specification MUST understand IPv6 addresses in URLs, and headers. By this requirement the feature-tag "play.basic" can be used to determine that a server or client is capable of handling IPv6 within RTSP. 10 Capability Handling This chapter describes the capability handling mechanism available in RTSP 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 these two cases. When a method is added the involved parties can use the OPTIONS method to discover if it is supported. This is done by issuing a OPTIONS request to the other party. Depending on the URL it will either apply in regards to a certain media resource, the whole server in general, or simply the next hop. The OPTIONS response will contain a Public header which declares all methods supported for the indicated resource. It is not necessary to use OPTIONS to discover support of a method, the client could simply try the method. If the receiver of the request does not support the method it will respond with an error code indicating the the method is either not implemented (501) or does not apply for the resource (405). The choice between the two discovery methods depends on the requirements of the service. To handle functionality additions that are not new methods feature- tags are defined. Each feature-tag represents a certain block of functionality. The amount of functionality that a feature-tag represents can vary significantly. A simple feature-tag can simple represent the functionality a single header gives. Another feature- tag is "play.basic" which represents the minimal playback implementation according to the updated specification. The feature-tags are then used to determine if the client, server or proxy supports the functionality that is necessary to achieve the desired service. To determine support of a feature-tag several different headers can be used, each explained below: Supported: The supported header is used to determine the complete set of functionality that both client and server has. The intended usage is to determine before one needs to H. Schulzrinne et. al. [Page 37]
Internet Draft RTSP February 16, 2004 use a functionality that it is supported. It can be used in any method however OPTIONS is the most suitable as one at the same time determines all methods that are implemented. When sending a request the requestor declares all its capabilities by including all supported feature-tags. This results in that the receiver learns the requestors feature support. The receiver then includes its set of features in the response. Require: The Require header can be included in any request where the end point, i.e. the client or server, is required to understand the feature to correctly perform the request. This can for example be a SETUP request where the server must 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 understood are 551 (Option Not Supported). Also the features that are not understood are given in the Unsupported header in the response. Proxy-Require: This method has the same purpose and workings as Require except that it only applies to proxies and not the end point. Features that needs to be supported by both proxies and end-point needs to be included in both the Require and Proxy-Require header. Unsupported: This header is used in 551 error response to tell which feature(s) that was not supported. Such a response is only the result of the usage of the Require and/or Proxy- Require header where one or more feature where not supported. This information allows the requestor to make the best of situations as it knows which features that was not supported. 11 Method Definitions The method indicates what is to be performed on the resource | identified by the Request-URL. The method name is case-sensitive. | New methods may be defined in the future. Method names may not start | with a $ character (decimal 24) and must be a token as defined by the | ABNF. Methods are summarized in Table 7. Notes on Table 7: PAUSE is recommended, but not required in that a | H. Schulzrinne et. al. [Page 38]
Internet Draft RTSP February 16, 2004 method direction object Server req. Client req. ___________________________________________________________________ DESCRIBE C -> S P,S recommended recommended GET_PARAMETER C -> S, S -> C P,S optional optional OPTIONS C -> S, S -> C P,S R=Req, Sd=Opt Sd=Req, R=Opt PAUSE C -> S P,S recommended recommended PING C -> S, S -> C P,S recommended optional PLAY C -> S P,S required required REDIRECT S -> C P,S optional optional SETUP C -> S S required required SET_PARAMETER C -> S, S -> C P,S optional optional TEARDOWN C -> S P,S required required Table 7: Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on. Legend: R=Responde to, Sd=Send, Opt: Optional, Req: Required, Rec: Recommended fully functional server can be built that does not support this | method, for example, for live feeds. If a server does not support a | particular method, it MUST return 501 (Not Implemented) and a client | SHOULD NOT try this method again for this server and resource. 11.1 OPTIONS The behavior is equivalent to that described in [H9.2]. An OPTIONS request may be issued at any time, e.g., if the client is about to try a nonstandard request. It does not influence the session state. The Public header MUST be included in responses to indicate which methods that are supported by the server. To specify which methods that are possible to use for the specified resource, the Allow MAY be used. By including in the OPTIONS request a Supported header, the requester can determine which features the other part supports. The request URL determines which scope the OPTIONS request has. By giving the URL of a certain media the capabilities regarding this media will be responded. By using the "*" URL the request regards the next hop only, while having a URL with only the host address regards the server without any media relevance. The OPTIONS method can be used for RTSP session keep alive | signalling, however this method is not the most recommended one, see | section 14.37 for a preference list. A keep alive OPTIONS request | SHOULD use the media or aggregated control URL. For options to | function as session state keep-alive, it is REQUIRED that the session | ID is included in the Session header. H. Schulzrinne et. al. [Page 39]
Internet Draft RTSP February 16, 2004 Example: C->S: OPTIONS * RTSP/1.0 CSeq: 1 User-Agent: PhonyClient/1.2 Require: Proxy-Require: gzipped-messages Supported: play-basic S->C: RTSP/1.0 200 OK CSeq: 1 Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE Supported: play-basic, implicit-play, gzipped-messages Server: PhonyServer/1.0 Note that some of the feature-tags in Require and Proxy-Require are necessarily fictional features (one would hope that we would not purposefully overlook a truly useful feature just so that we could have a strong example in this section). 11.2 DESCRIBE The DESCRIBE method retrieves the description of a presentation or | media object identified by the request URL from a server. The request | MAY use the Accept header to specify the description formats that the | client understands. The server responds with a description of the | requested resource. The DESCRIBE reply-response pair constitutes the | media initialization phase of RTSP. Example: C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 CSeq: 312 User-Agent: PhonyClient 1.2 Accept: application/sdp, application/rtsl, application/mheg S->C: RTSP/1.0 200 OK CSeq: 312 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 Content-Type: application/sdp Content-Length: 376 v=0 H. Schulzrinne et. al. [Page 40]
Internet Draft RTSP February 16, 2004 o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4 s=SDP Seminar i=A Seminar on the session description protocol u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps e=mjh@isi.edu (Mark Handley) c=IN IP4 224.2.17.12/127 t=2873397496 2873404696 a=recvonly m=audio 3456 RTP/AVP 0 m=video 2232 RTP/AVP 31 m=application 32416 UDP WB a=orient:portrait The DESCRIBE response MUST contain all media initialization information for the resource(s) that it describes. If a media client obtains a presentation description from a source other than DESCRIBE and that description contains a complete set of media initialization parameters, the client SHOULD use those parameters and not then request a description for the same media via RTSP. Additionally, servers SHOULD NOT use the DESCRIBE response as a means of media indirection. By forcing a DESCRIBE response to contain all media initialization for the set of streams that it describes, and discouraging use of DESCRIBE for media indirection, we avoid looping problems that might result from other approaches. Media initialization is a requirement for any RTSP-based system, but the RTSP specification does not dictate that this must be done via the DESCRIBE method. There are three ways that an RTSP client may receive initialization information: o via RTSP's DESCRIBE method; o via some other protocol (HTTP, email attachment, etc.); o via the command line or standard input (thus working as a browser helper application launched with an SDP file or other media initialization format). It is RECOMMENDED that minimal servers support the DESCRIBE method, and highly recommended that minimal clients support the ability to act as a "helper application" that accepts a media initialization H. Schulzrinne et. al. [Page 41]
Internet Draft RTSP February 16, 2004 file from standard input, command line, and/or other means that are appropriate to the operating environment of the client. 11.3 SETUP The SETUP request for a URL specifies the transport mechanism to be used for the streamed media. The SETUP method may be used in two different cases; Create a RTSP session or add a media to a session, and change the transport parameters of already set up media stream. Using SETUP to create or add media to a session when in PLAY state is unspecified. Otherwise SETUP can be used in all three states; INIT, and READY, for both purposes and in PLAY to change the transport parameters. The Transport header, see section 14.40, specifies the transport | parameters acceptable to the client for data transmission; the | response will contain the transport parameters selected by the | server. This allows the client to enumerate in priority order the | transport mechanisms and parameters acceptable to it, while the | server can select the most appropriate. It is expected that the | session description format used will enable the client to select a | limited number possible configurations that are offered to the server | to choose from. All transport parameters SHOULD be included in the | Transport header, the use of other headers for this purpose is | discouraged due to middle boxes. For the benefit of any intervening firewalls, a client SHOULD indicate the transport parameters even if it has no influence over these parameters, for example, where the server advertises a fixed multicast address. 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. In a SETUP response the server SHOULD include the Accept-Ranges header (see section 14.4 to indicate which time formats that are acceptable to use for this media resource. C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 CSeq: 302 Transport: RTP/AVP;unicast;client_port=4588-4589, RTP/AVP/TCP;unicast;interleave=0-1 H. Schulzrinne et. al. [Page 42]
Internet Draft RTSP February 16, 2004 S->C: RTSP/1.0 200 OK CSeq: 302 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 Session: 47112344 Transport: RTP/AVP;unicast;client_port=4588-4589; server_port=6256-6257;ssrc=2A3F93ED Accept-Ranges: NPT In the above example the client want to create a RTSP session containing the media resource "rtsp://example.com/foo/bar/baz.rm". The transport parameters acceptable to the client is either RTP/AVP/UDP (UDP per default) to be received on client port 4588 and 4589 or RTP/AVP interleaved on the RTSP control channel. The server selects the RTP/AVP/UDP transport and adds the ports it will send and received RTP and RTCP from, and the RTP SSRC that will be used by the server. The server MUST generate a session identifier in response to a successful SETUP request, unless a SETUP request to a server includes a session identifier, in which case the server MUST bundle this setup request into the existing session (aggregated session) or return error 459 (Aggregate Operation Not Allowed) (see Section 13.4.11). An Aggregate control URL MUST be used to control an aggregated session. This URL MUST be different from the stream control URLs of the individual media streams included in the aggregate. The Aggregate control URL 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 URL is not specified in the session description, it is normally an indication that non-aggregated control should be | used. The SETUP of media streams in an aggregate which has not been | given an aggregated control URL is unspecified. While the session ID sometimes has enough information for aggregate control of a session, the Aggregate control URL is still important for some methods such as SET_PARAMETER where the control URL enables the resource in question to be easily identified. The Aggregate control URL 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. Finally, presence of the Aggregate control URL allows for backwards compatibility with RFC 2326 [1]. H. Schulzrinne et. al. [Page 43]
Internet Draft RTSP February 16, 2004 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 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 chapter 14.37. Signs of liveness for a 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 for any of the media streams in that RTSP session. If a SETUP request on a session fails for any reason, the session state, as well as transport and other parameters for associated streams SHALL remain unchanged from their values as if the SETUP request had never been received by the server. 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 this, it MUST respond with error 455 (Method Not Valid In This State). Reasons to support changing transport parameters, is to allow for application layer mobility and flexibility to utilize the best available transport as it becomes available. In a SETUP response for a request to change the transport parameters while in Play state, the server SHOULD include the Range to indicate from what point the new transport parameters are used. Further if RTP is used for delivery the server SHOULD also include the RTP-Info header to indicate from what timestamp and RTP sequence number the change has taken place. If both RTP-Info and Range is included in the response the "rtp_time" parameter and range MUST be for the corresponding time, i.e. be used in the same way as for PLAY to ensure the correct synchronization information is present. If the transport parameter 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. 11.4 PLAY The PLAY method tells the server to start sending data via the H. Schulzrinne et. al. [Page 44]
Internet Draft RTSP February 16, 2004 mechanism specified in SETUP. A client MUST NOT issue a PLAY request until any outstanding SETUP requests have been acknowledged as successful. PLAY requests are valid when the session is in READY state; the use of PLAY requests when the session is in PLAY state is deprecated. A PLAY request MUST include a Session header to indicate which session the request applies to. In an aggregated session the PLAY request MUST contain an aggregated control URL. A server SHALL responde with error 460 (Only Aggregate Operation Allowed) if the client PLAY request URL is for one of the media. The media in an aggregate SHALL be played in sync. If a client want individual control of the media it must use separate RTSP sessions for each media. The PLAY request SHALL position the normal play time to the beginning of the range specified by the Range header and delivers stream data until the end of the range if given, else to the end of the media is reached. To allow for precise composition multiple ranges MAY be specified in one PLAY Request. The range values are valid if all given ranges are part of any media within the aggregate. If a given range value points outside of the media, the response SHALL be the 457 (Invalid Range) error code. The below example will first play seconds 10 through 15, then, immediately following, seconds 20 to 25, and finally seconds 30 through the end. C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 CSeq: 835 Session: 12345678 Range: npt=10-15, npt=20-25, npt=30- See the description of the PAUSE request for further examples. A PLAY request without a Range header is legal. It SHALL start playing a stream from the beginning (npt=0-) unless the stream has been paused. If a stream has been paused via PAUSE, stream delivery resumes at the pause point. The stream SHALL play until the end of the media. The Range header MUST NOT contain a time parameter. The usage of time | in PLAY method has been deprecated. If a request with time parameter | is received the server SHOULD respond with a 457 (Invalid Range) to | indicate that the time parameter is not supported. | H. Schulzrinne et. al. [Page 45]
Internet Draft RTSP February 16, 2004 Server MUST include a "Range" header in any PLAY response. The | response MUST use the same format as the request's range header | contained. If no Range header was in the request, the NPT time format | SHOULD be used unless the client showed support for an other format | more appropriate. Also for a session with live media streams the | Range header MUST indicate a valid time. It is RECOMMENDED that | normal play time is used, either the "now" indicator, for example | "npt=now-", or the time since session start as an open interval, e.g. | "npt=96.23-". An absolute time value (clock) for the corresponding | time MAY be given, i.e. "clock=20030213T143205Z-". The UTC clock | format SHOULD only be used if client has shown support for it. A media server only supporting playback MUST support the npt format and MAY support the clock and smpte formats. For a on-demand stream, the server MUST reply with the actual range that will be played back. This may differ from the requested range if alignment of the requested range to valid frame boundaries is required for the media source. If no range is specified in the request, the start position SHALL still be returned in the reply. If the medias that are part of an aggregate has different lengths, the PLAY request SHALL be performed as long as the given range is valid for any media, for example the longest media. Media will be sent whenever it is available for the given play-out point. After playing the desired range, the presentation is NOT | automatically paused, media delivery simply stops. A PAUSE request | MUST be issued before another PLAY request can be issued. Note: This | is one change resulting in a non-operability with RFC 2326 | implementations. A client not issuing a PAUSE request before a new | PLAY will be stuck in PLAY state. To mitigate this backwards | compatibility issue the following behavior is recommended. If a | server receives a PLAY request when in play state and all media has | finished the requested playout, the server MAY interpret this as a | PLAY request received in ready state. However the server SHALL NOT do | this if the client has shown any support for this specification, for | example by sending a Supported header with the play.basic feature | tag. A client desiring to play the media from the beginning MUST send a PLAY request with a Range header pointing at the beginning, e.g. npt=0-. If a PLAY request is received without a Range header when media delivery has stopped at the end, the server SHOULD respond with a 457 "Invalid Range" error response. In that response the current pause point in a Range header SHALL be included. The following example plays the whole presentation starting at SMPTE time code 0:10:20 until the end of the clip. Note: The RTP-Info H. Schulzrinne et. al. [Page 46]
Internet Draft RTSP February 16, 2004 headers has been broken into several lines to fit the page. C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 | CSeq: 833 | Session: 12345678 | Range: smpte=0:10:20- | S->C: RTSP/1.0 200 OK | CSeq: 833 | Date: 23 Jan 1997 15:35:06 GMT | Server: PhonyServer 1.0 | Range: smpte=0:10:22-0:15:45 | RTP-Info:url=rtsp://example.com/twister.en; | 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/1.0 CSeq: 835 Session: 12345678 Range: clock=19961108T142300Z-19961108T143520Z S->C: RTSP/1.0 200 OK CSeq: 835 Date: 23 Jan 1997 15:35:06 GMT Server:PhonyServer 1.0 Range: clock=19961108T142300Z-19961108T143520Z RTP-Info:url=rtsp://example.com/meeting.en; seq=53745;rtptime=484589019 All range specifiers in this specification allow for ranges with unspecified begin times (e.g. "npt=-30"). When used in a PLAY request, the server treats this as a request to start/resume playback from the current pause point, ending at the end time specified in the Range header. If the pause point is located later than the given end value, a 457 (Invalid Range) response SHALL be given. The queued play functionality described in RFC 2326 [1] is removed | and multiple ranges can be used to achieve a similar functionality. | If a server receives a PLAY request while in the PLAY state, the | H. Schulzrinne et. al. [Page 47]
Internet Draft RTSP February 16, 2004 server SHALL responde using the error code 455 (Method Not Valid In | This State). This will signal the client that queued play are not | supported. The use of PLAY for keep-alive signaling, i.e. PLAY request without a range header in PLAY state, has also been depreciated. Instead a client can use, PING, SET_PARAMETER or OPTIONS for keep alive. A server receiving a PLAY keep alive SHALL respond with the 455 error code. 11.5 PAUSE The PAUSE request causes the stream delivery to be interrupted (halted) temporarily. A PAUSE request MUST be done with the aggregated control URL for aggregated sessions, resulting in all media being halted, or the media URL for non-aggregated sessions. Any attempt to do muting of a single media with an PAUSE request in an aggregated session SHALL be responded with error 460 (Only Aggregate Operation Allowed). After resuming playback, synchronization of the tracks MUST be maintained. Any server resources are kept, though servers MAY close the session and free resources after being paused for the duration specified with the timeout parameter of the Session header in the SETUP message. Example: C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 834 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 834 Date: 23 Jan 1997 15:35:06 GMT Range: npt=45.76- The PAUSE request MAY contain a Range header specifying when the stream or presentation is to be halted. We refer to this point as the "pause point". The time parameter in the Range MUST NOT be used. The Range header MUST contain a single value, expressed as the beginning value an open range. For example, the following clip will be played from 10 seconds through 21 seconds of the clip's normal play time, under the assumption that the PAUSE request reaches the server within 11 seconds of the PLAY request. Note that some lines has been broken in an non-correct way to fit the page: H. Schulzrinne et. al. [Page 48]
Internet Draft RTSP February 16, 2004 C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 834 Session: 12345678 Range: npt=10-30 S->C: RTSP/1.0 200 OK CSeq: 834 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 Range: npt=10-30 RTP-Info:url=rtsp://example.com/fizzle/audiotrack; seq=5712;rtptime=934207921, url=rtsp://example.com/fizzle/videotrack; seq=57654;rtptime=2792482193 Session: 12345678 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 835 Session: 12345678 Range: npt=21- S->C: RTSP/1.0 200 OK CSeq: 835 Date: 23 Jan 1997 15:35:09 GMT Server: PhonyServer 1.0 Range: npt=21- Session: 12345678 The pause request becomes effective the first time the server is encountering the time point specified in any of the multiple ranges. If the Range header specifies a time outside any range from the PLAY request, the error 457 (Invalid Range) SHALL be returned. If a media unit (such as an audio or video frame) starts presentation at exactly the pause point, it is not played. If the Range header is missing, stream delivery is interrupted immediately on receipt of the message and the pause point is set to the current normal play time. However, the pause point in the media stream MUST be maintained. A subsequent PLAY request without Range header SHALL resume from the pause point and play until media end. If the server has already sent data beyond the time specified in the PAUSE request's Range header, a PLAY without range SHALL resume at the point in time specified by the PAUSE request's Range header, as it is assumed that the client has discarded data after that point. This ensures continuous pause/play cycling without gaps. H. Schulzrinne et. al. [Page 49]
Internet Draft RTSP February 16, 2004 The pause point after any PAUSE request SHALL be returned to the client by adding a Range header with what remains unplayed of the PLAY request's ranges, i.e. including all the remaining ranges part of multiple range specification. If one desires to resume playing a ranged request, one simply includes the Range header from the PAUSE response. Note that this server behavior was not mandated previously and servers implementing according to RFC 2326 will probably not return the range header. For example, if the server have a play request for ranges 10 to 15 and 20 to 29 pending and then receives a pause request for NPT 21, it would start playing the second range and stop at NPT 21. If the pause request is for NPT 12 and the server is playing at NPT 13 serving the first play request, the server stops immediately. If the pause request is for NPT 16, the server returns a 457 error message. To prevent that the second range is played and the server stops after completing the first range, a PAUSE request for 20 must be issued. As another example, if a server has received requests to play ranges 10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE request for NPT=14 would take effect while the server plays the first range, with the second range effectively being ignored, assuming the PAUSE request arrives before the server has started playing the second, overlapping range. Regardless of when the PAUSE request arrives, it sets the pause point to 14. The below example messages is for the above case when the PAUSE request arrives before the first occurrence of NPT=14. C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 834 Session: 12345678 Range: npt=10-15, npt=13-20 S->C: RTSP/1.0 200 OK CSeq: 834 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 Range: npt=10-15, npt=13-20 RTP-Info:url=rtsp://example.com/fizzle/audiotrack; seq=5712;rtptime=934207921, url=rtsp://example.com/fizzle/videotrack; seq=57654;rtptime=2792482193 Session: 12345678 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 835 Session: 12345678 H. Schulzrinne et. al. [Page 50]
Internet Draft RTSP February 16, 2004 Range: npt=14- S->C: RTSP/1.0 200 OK CSeq: 835 Date: 23 Jan 1997 15:35:09 GMT Server: PhonyServer 1.0 Range: npt=14-15, npt=13-20 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, even if the PAUSE request is asking for some other pause point. See examples below: Examples: | C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 | CSeq: 834 | Session: 12345678 | S->C: RTSP/1.0 200 OK | CSeq: 834 | Session: 12345678 | Date: 23 Jan 1997 15:35:06 GMT | Range: npt=45.76-98.36 | C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 | CSeq: 835 | Session: 12345678 | Range: 86- | S->C: RTSP/1.0 200 OK | CSeq: 835 | Session: 12345678 | Date: 23 Jan 1997 15:35:07 GMT | Range: npt=45.76-98.36 | 11.6 TEARDOWN The TEARDOWN client to server request stops the stream delivery for the given URL, freeing the resources associated with it. TEARDOWN MAY be done using either an aggregated or a media control URL. H. Schulzrinne et. al. [Page 51]
Internet Draft RTSP February 16, 2004 However some restrictions apply depending on the current state. The TEARDOWN request SHALL contain a Session header indicating what session the request applies to. A TEARDOWN using the aggregated control URL or the media URL in a session under non-aggregated control MAY be done in any state (Ready, and Play). A successful request SHALL result in that media delivery is immediately halted and the session state is destroyed. This SHALL be indicated through the lack of a Session header in the response. A TEARDOWN using a media URL in an aggregated session MAY only be | done in Ready state. Such a request only removes the indicated media | stream and associated resources from the session. This may result in | that a session returns to non-aggregated control, due to that it only | contains a single media. In the response to TEARDOWN request | resulting in that the session still exist SHALL contain a Session | header to indicate this. Note, the indication with the session header if sessions state remain may not be done correctly by a RFC 2326 client, but will be for any server signalling the "play.basic" tag. Example: C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 892 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 892 Server: PhonyServer 1.0 11.7 GET_PARAMETER The GET_PARAMETER request retrieves the value of a parameter or | parameters for a presentation or stream specified in the URL. If the | Session header is present in a request, the value of a parameter MUST | be retrieved in the specified session context. The content of the | reply and response is left to the implementation. | The method MAY also be used without a body (entity). If the this | request is successful, i.e. a 200 OK response is received, then the | keep-alive time has been updated. Any non-required header present in | such a request, may or may not been processed. The allow a client to | determine if any such header has been processed, it is necessary to | H. Schulzrinne et. al. [Page 52]
Internet Draft RTSP February 16, 2004 use a feature tag and the Require header. Due to this reason it is | RECOMMENDED that any parameters to be retrieved are sent in the body, | rather than using any header. Example: S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 431 Content-Type: text/parameters Session: 12345678 Content-Length: 15 packets_received jitter C->S: RTSP/1.0 200 OK CSeq: 431 Content-Length: 46 Content-Type: text/parameters packets_received: 10 jitter: 0.3838 The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation. 11.8 SET_PARAMETER This method requests to set the value of a parameter or a set of | parameters for a presentation or stream specified by the URL. The | method MAY also be used without a body (entity). If the this request | is successful, i.e. a 200 OK response is received, then the keep- | alive time has been updated. Any non-required header present in such | a request, may or may not been processed. The allow a client to | determine if any such header has been processed, it is necessary to | use a feature tag and the Require header. Due to this reason it is | RECOMMENDED that any parameters are sent in the body, rather than | using any header. A request is RECOMMENDED to only contain a single parameter to allow the client to determine why a particular request failed. If the request contains several parameters, the server MUST only act on the H. Schulzrinne et. al. [Page 53]
Internet Draft RTSP February 16, 2004 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 can locate a parameter error 451 (Parameter Not Understood) SHALL be used. In the case a parameter is not allowed to change the error code 458 (Parameter Is Read-Only). The response body SHOULD contain only the parameters that has errors. Otherwise no body SHALL be returned. Note: transport parameters for the media stream MUST only be set with the SETUP command. Restricting setting transport parameters to SETUP is for the benefit of firewalls. The parameters are split in a fine-grained fashion so that there can be more meaningful error indications. However, it may make sense to allow the setting of several parameters if an atomic setting is desirable. Imagine device control where the client does not want the camera to pan unless it can also tilt to the right angle at the same time. Example: C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 421 Content-length: 20 Content-type: text/parameters barparam: barstuff S->C: RTSP/1.0 451 Parameter Not Understood CSeq: 421 Content-length: 10 Content-type: text/parameters barparam The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation. H. Schulzrinne et. al. [Page 54]
Internet Draft RTSP February 16, 2004 11.9 REDIRECT A redirect request informs the client that it MUST connect to another server location. The REDIRECT request MAY contain the header Location, which indicates that the client should issue requests for that URL. The lack of a Location header in the response SHALL be interpreted as that the server can't any longer fulfill the current request, but has no alternative at the present where the client can continue. If a REDIRECT request contains a Session header, it is end-to-end and applies only to the given session. If there are proxies in the request chain, they SHOULD NOT disconnect the control channel unless there are no remaining sessions. If the Location header is included it SHALL contain a full absolute URL pointing out the resource to reconnect too, i.e. the Location SHALL NOT contain only host and port. If a REDIRECT request does not contain a Session header, it is next- hop and applies also to the control connection. If the Location header is included it SHOULD only contain an absolute URL containing the host address and OPTIONAL the port number. If there are proxies in the request chain, they SHOULD do all of the following: (1) respond to the REDIRECT request, (2) disconnect the control channel from the requestor, (3) reconnect to the given host address, and (4) pass the request to each applicable client (typically those clients with an active session or unanswered request from the requestor). Note that the proxy is responsible for accepting the REDIRECT response from its clients and these responses MUST NOT be passed on to either the requesting or the destination server. A REDIRECT request with a Session header MAY only be received by a | client when it has the established session. A REDIRECT request | without a Session MAY be received at any time communication is | established with the server. The redirect request MAY contain the header Range, which indicates when the redirection takes effect. If the Range contains a "time=" value that is the wall clock time that the redirection MUST at the latest take place. When the "time=" parameter is present the range value MUST be ignored. However the range entered MUST be syntactical correct and SHALL point at the beginning of any on-demand content. If no time parameter is part of the Range header then redirection SHALL take place when the media playout from the server reaches the given time. The range value MUST be a single value in the open ended form, e.g. npt=59-. A server upon receiving a successful (2xx) response for a REDIRECT H. Schulzrinne et. al. [Page 55]
Internet Draft RTSP February 16, 2004 request without any Range header SHALL consider the session as removed and can free any session state. For this type of requests the rest of this paragraph applies. The server MAY close the signalling connection upon receiving the response for REDIRECT requests without a Session header. The client SHOULD close the signaling connection after having given the 2xx response to a REDIRECT response, unless it has several sessions on the server. If the client has multiple session on the server it SHOULD close the connection when it has received and responded to REDIRECT requests for all sessions. A client receiving a REDIRECT request with a Range header SHALL issue a TEARDOWN request when the in indicated redirect point is reached. The client SHOULD for REDIRECT requests with Range header close the signalling connection after a 2xx response on its TEARDOWN request. The normal connection considerations apply for the server. This differentiation from REDIRECT requests without range headers is to allow clear an explicit state handling. As the state in the server needs to be kept until the point of redirection, the handling becomes more clear if the client is required to tear down the session at that point. If the client wants to continue to send or receive media for this resource, the client will have to establish a new session with the designated host. A client SHOULD issue a new DESCRIBE request with the URL given in the Location header, unless the URL only contains a host address. In the cases the Location only contains a host address the client MAY assume that the media on the server it is redirected to is identical. Identical media means that all media configuration information from the old session still is valid except for the host address. In the case of absolute URLs in the location header the media redirected to can be either identical, slightly different or totally different. This is the reason why a new DESCRIBE request SHOULD be issued. 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/1.0 CSeq: 732 Location: rtsp://s2.example.com:8001 Range: npt=0- ;time=19960213T143205Z Session: uZ3ci0K+Ld-M 11.10 PING H. Schulzrinne et. al. [Page 56]
Internet Draft RTSP February 16, 2004 This method is a bi-directional mechanism for server or client liveness checking. It has no side effects. The issuer of the request MUST include a session header with the session ID of the session that is being checked for liveness. Prior to using this method, an OPTIONS method is RECOMMENDED to be issued in the direction which the PING method would be used. This method MUST NOT be used if support is not indicated by the Public header. Note: That an 501 (Not Implemented) response means that the keep-alive timer has not been updated. When a proxy is in use, PING with a * indicates a single-hop liveness check, whereas PING with a URL including an host address indicates an end-to-end liveness check. Example: C->S: PING * RTSP/1.0 CSeq: 123 Session:12345678 S->C: RTSP/1.0 200 OK CSeq: 123 Session:12345678 12 Embedded (Interleaved) Binary Data Certain firewall designs and other circumstances may force a server to interleave RTSP messages and media stream data. This interleaving should generally be avoided unless necessary since it complicates client and server operation and imposes additional overhead. Also head of line blocking may cause problems. Interleaved binary data SHOULD only be used if RTSP is carried over TCP. Stream data such as RTP packets is encapsulated by an ASCII dollar sign (24 decimal), followed by a one-byte channel identifier, followed by the length of the encapsulated binary data as a binary, two-byte integer in network byte order. The stream data follows immediately afterwards, without a CRLF, but including the upper-layer protocol headers. Each $ block SHALL contain exactly one upper-layer protocol data unit, e.g., one RTP packet. 0 1 2 3 H. Schulzrinne et. al. [Page 57]
Internet Draft RTSP February 16, 2004 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | "$" = 24 | Channel ID | Length in bytes | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : Length number of bytes of binary data : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The channel identifier is defined in the Transport header with the interleaved parameter(Section 14.40). When the transport choice is RTP, RTCP messages are also interleaved by the server over the TCP connection. The usage of RTCP messages is indicated by including a range containing a second channel in the interleaved parameter of the Transport header, see section 14.40. If RTCP is used, packets SHALL be sent on the first available channel higher than the RTP channel. The channels are bi-directional and therefore RTCP traffic are sent on the second channel in both directions. RTCP is needed for synchronization when two or more streams are interleaved in such a fashion. Also, this provides a convenient way to tunnel RTP/RTCP packets through the TCP control connection when required by the network configuration and transfer them onto UDP when possible. C->S: SETUP rtsp://example.com/bar.file RTSP/1.0 | CSeq: 2 | Transport: RTP/AVP/TCP;unicast;interleaved=0-1 | S->C: RTSP/1.0 200 OK | CSeq: 2 | Date: 05 Jun 1997 18:57:18 GMT | Transport: RTP/AVP/TCP;unicast;interleaved=5-6 | Session: 12345678 | C->S: PLAY rtsp://example.com/bar.file RTSP/1.0 | CSeq: 3 | Session: 12345678 | S->C: RTSP/1.0 200 OK | CSeq: 3 | H. Schulzrinne et. al. [Page 58]
Internet Draft RTSP February 16, 2004 Session: 12345678 | Date: 05 Jun 1997 18:59:15 GMT | RTP-Info: url=rtsp://example.com/bar.file; | seq=232433;rtptime=972948234 | S->C: $005{2 byte length}{"length" bytes data, w/RTP header} | S->C: $005{2 byte length}{"length" bytes data, w/RTP header} | S->C: $006{2 byte length}{"length" bytes RTCP packet} | 13 Status Code Definitions Where applicable, HTTP status [H10] codes are reused. Status codes that have the same meaning are not repeated here. See Table 4 for a listing of which status codes may be returned by which requests. All error messages, 4xx and 5xx MAY return a body containing further information about the error. 13.1 Success 1xx 13.1.1 100 Continue See, [H10.1.1]. 13.2 Success 2xx 13.3 Redirection 3xx The notation "3rr" indicates response codes from 300 to 399 inclusive which are meant for redirection. The response code 304 is excluded from this set, as it is not used for redirection. See [H10.3] for definition of status code 300 to 305. However comments are given for some to how they apply to RTSP. Within RTSP, redirection may be used for load balancing or redirecting stream requests to a server topologically closer to the client. Mechanisms to determine topological proximity are beyond the scope of this specification. A 3rr code MAY be used to respond to any request. It is RECOMMENDED | that they are used if necessary before a session is established, i.e. | in response to DESCRIBE or SETUP. However in cases where a server is | not able to send a REDIRECT request to the client, the server MAY | need to resort to using 3rr responses to inform a client with a | established session about the need for redirecting the session. If an | 3rr response is received for an request in relation to a established | H. Schulzrinne et. al. [Page 59]
Internet Draft RTSP February 16, 2004 session, the client SHOULD send a TEARDOWN request for the session, | and MAY reestablish the session using the resource indicated by the | Location. If the the Location header is used in a response it SHALL contain an absolute URL pointing out the media resource the client is redirected to, the URL SHALL NOT only contain the host name. 13.3.1 300 Multiple Choices 13.3.2 301 Moved Permanently The request resource are moved permanently and resides now at the URL given by the location header. The user client SHOULD redirect automatically to the given URL. This response MUST NOT contain a message-body. 13.3.3 302 Found The requested resource reside temporarily at the URL given by the Location header. The Location header MUST be included in the response. Is intended to be used for many types of temporary redirects, e.g. load balancing. It is RECOMMENDED that one set the reason phrase to something more meaningful than "Found" in these cases. The user client SHOULD redirect automatically to the given URL. This response MUST NOT contain a message-body. 13.3.4 303 See Other This status code SHALL NOT be used in RTSP. However as it was allowed | to use in RFC 2326 it is possible that such response may be received, | in which case the behavior is undefined. 13.3.5 304 Not Modified If the client has performed a conditional DESCRIBE or SETUP (see 12.23) 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 ETag and/or Content-Location, if the header would have been sent in a 200 response to the same request. o Expires, Cache-Control, and/or Vary, if the field-value might H. Schulzrinne et. al. [Page 60]
Internet Draft RTSP February 16, 2004 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 and a 304 response to SETUP does NOT imply that the resource description is unchanged. The ETag and If-Match headers may be used to link the DESCRIBE and SETUP in this manner. 13.3.6 305 Use Proxy See [H10.3.6]. 13.4 Client Error 4xx 13.4.1 400 Bad Request The request could not be understood by the server due to malformed syntax. The client SHOULD NOT repeat the request without modifications [H10.4.1]. If the request does not have a CSeq header, the server MUST NOT include a CSeq in the response. 13.4.2 405 Method Not Allowed The method specified in the request is not allowed for the resource identified by the request URL. 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, e.g., if a RECORD request is issued even though the mode parameter in the Transport header only specified PLAY. 13.4.3 451 Parameter Not Understood The recipient of the request does not support one or more parameters contained in the request.When returning this error message the sender SHOULD return a entity body containing the offending parameter(s). 13.4.4 452 reserved This error code was removed from RFC 2326 [1] and is obsolete. 13.4.5 453 Not Enough Bandwidth The request was refused because there was insufficient bandwidth. This may, for example, be the result of a resource reservation failure. H. Schulzrinne et. al. [Page 61]
Internet Draft RTSP February 16, 2004 13.4.6 454 Session Not Found The RTSP session identifier in the Session header is missing, invalid, or has timed out. 13.4.7 455 Method Not Valid in This State The client or server cannot process this request in its current state. The response SHOULD contain an Allow header to make error recovery easier. 13.4.8 456 Header Field Not Valid for Resource The server could not act on a required request header. For example, if PLAY contains the Range header field but the stream does not allow seeking. This error message may also be used for specifying when the time format in Range is impossible for the resource. In that case the Accept-Ranges header SHOULD be returned to inform the client of which format(s) that are allowed. 13.4.9 457 Invalid Range The Range value given is out of bounds, e.g., beyond the end of the presentation. 13.4.10 458 Parameter Is Read-Only The parameter to be set by SET_PARAMETER can be read but not modified. When returning this error message the sender SHOULD return a entity body containing the offending parameter(s). 13.4.11 459 Aggregate Operation Not Allowed The requested method may not be applied on the URL in question since it is an aggregate (presentation) URL. The method may be applied on a media URL. 13.4.12 460 Only Aggregate Operation Allowed The requested method may not be applied on the URL in question since it is not an aggregate control (presentation) URL. The method may be applied on the aggregate control URL. 13.4.13 461 Unsupported Transport The Transport field did not contain a supported transport specification. H. Schulzrinne et. al. [Page 62]
Internet Draft RTSP February 16, 2004 13.4.14 462 Destination Unreachable The data transmission channel could not be established because the client address could not be reached. This error will most likely be the result of a client attempt to place an invalid Destination parameter in the Transport field. 13.5 Server Error 5xx 13.5.1 551 Option not supported An feature-tag given in the Require or the Proxy-Require fields was not supported. The Unsupported header SHOULD be returned stating the feature for which there is no support. 14 Header Field Definitions method direction object acronym Body _________________________________________________ DESCRIBE C -> S P,S DES r GET_PARAMETER C -> S, S -> C P,S GPR R,r OPTIONS C -> S P,S OPT S -> C PAUSE C -> S P,S PSE PING C -> S, S -> C P,S PNG PLAY C -> S P,S PLY REDIRECT S -> C P,S RDR SETUP C -> S S STP SET_PARAMETER C -> S, S -> C P,S SPR R,r TEARDOWN C -> S P,S TRD Table 8: Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on. Body notes if a method is allowed to carry body and in which direction, R = Request, r=response. Note: It is allowed for all error messages 4xx and 5xx to have a body The general syntax for header fields is covered in Section 4.2 This section lists the full set of header fields along with notes on meaning, and usage. The syntax definition for headers are present in section 17.2.3. Throughout this section, we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1 specification RFC 2616 [4]. Examples of each header field are given. Information about header fields in relation to methods and proxy processing is summarized in Table 9 and Table 10. H. Schulzrinne et. al. [Page 63]
Internet Draft RTSP February 16, 2004 The "where" column describes the request and response types in which the header field can be used. Values in this column are: 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 all requests and responses. The "proxy" column describes the operations a proxy may perform on a header field: 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 must be able to read the header field, and thus this header field cannot be encrypted. The rest of the columns relate to the presence of a header field in a method. The method names when abbreviated, are according to table 8: c: Conditional; requirements on the header field depend on the context of the message. m: The header field is mandatory. m*: The header field SHOULD be sent, but clients/servers need to be prepared to receive messages without that header field. o: The header field is optional. *: The header field is required if the message body is not empty. See sections 14.14, 14.16 and 4.3 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 | H. Schulzrinne et. al. [Page 64]
Internet Draft RTSP February 16, 2004 field, in other case it is request to process the header. This | is regulated by the method and header descriptions. Example of | such headers that require processing are the Require and Proxy- | Require header fields discussed in 14.32 and 14.27. 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. A Client/Server SHOULD ignore extension header parameters that are not understood. The From, Location, and RTP-Info header fields contain a URL. If the URL contains a comma, or semicolon, the URL MUST be enclosed in double quotas ("). Any URL parameters are contained within these quotas. If the URL is not enclosed in double quotas, any semicolon- delimited parameters are header-parameters, not URL parameters. 14.1 Accept The Accept request-header field can be used to specify certain presentation description content types which are acceptable for the response. The "level" parameter for presentation descriptions is properly defined as part of the MIME type registration, not here. See [H14.1] for syntax. Example of use: | Accept: application/rtsl q=1.0, application/sdp | 14.2 Accept-Encoding H. Schulzrinne et. al. [Page 65]
Internet Draft RTSP February 16, 2004 See [H14.3] 14.3 Accept-Language See [H14.4]. Note that the language specified applies to the presentation description and any reason phrases, not the media content. 14.4 Accept-Ranges The Accept-Ranges response-header field allows the server to indicate its acceptance of range requests and possible formats for a resource: Accept-Ranges: NPT, SMPTE This header has the same syntax as [H14.5] and the syntax is defined in 17.2.3. However new range-units are defined. Inclusion of any of the time formats indicates acceptance by the server for PLAY and PAUSE requests with this format. The headers value is valid for the resource specified by the URL in the request, this response corresponds to. A server SHOULD use this header in SETUP responses to indicate to the client which range time formats the media supports. The header SHOULD also be included in "456" responses which is a result of use of unsupported range formats. 14.5 Allow The Allow entity-header field lists the methods supported by the resource identified by the request-URL. The purpose of this field is to strictly inform the recipient of valid methods associated with the resource. An Allow header field MUST be present in a 405 (Method Not Allowed) response. See [H14.7] for syntax definition. Example of use: Allow: SETUP, PLAY, SET_PARAMETER 14.6 Authorization See [H14.8] 14.7 Bandwidth H. Schulzrinne et. al. [Page 66]
Internet Draft RTSP February 16, 2004 Header Where Proxy DES OPT SETUP PLAY PAUSE TRD _____________________________________________________________ Accept R o - - - - - Accept-Encoding R r o - - - - - Accept-Language R r o - - - - - Accept-Ranges r r - - o - - - Accept-Ranges 456 r - - - o o - Allow r - o - - - - Allow 405 m m m m m m Authorization R o o o o o o Bandwidth R o o o o - - Blocksize R o - o o - - Cache-Control r - - o - - - Connection o o o o o o Content-Base r o - - - - - Content-Base 4xx o o o o o o Content-Encoding R r - - - - - - Content-Encoding r r o - - - - - Content-Encoding 4xx r o o o o o o Content-Language R r - - - - - - Content-Language r r o - - - - - Content-Language 4xx r o o o o o o Content-Length r r * - - - - - Content-Length 4xx r * * * * * * Content-Location r o - - - - - Content-Location 4xx o o o o o o Content-Type r * - - - - - Content-Type 4xx * * * * * * CSeq Rc m m m m m m Date am o o o o o o Expires r r o - - - - - From R r o o o o o o Host - - - - - - If-Match R r - - o - - - If-Modified-Since R r o - o - - - Last-Modified r r o - - - - - Location 3rr o o o o o o Proxy-Authenticate 407 amr m m m m m m Proxy-Require R ar o o o o o o Public r admr - m* - - - - Public 501 admr m* m* m* m* m* m* Range R - - - o o - Range r - - c m* m* - Referer R o o o o o o Require R o o o o o o Retry-After 3rr,503 o o o - - - H. Schulzrinne et. al. [Page 67]
Internet Draft RTSP February 16, 2004 Header Where Proxy DES OPT SETUP PLAY PAUSE TRD _________________________________________________________ Scale - - - o - - Session R - o o m m m Session r - c m m m o Server R - o - - - - Server r o o o o o o Speed - - - o - - Supported R o o o o o o Supported r c c c c c c Timestamp R o o o o o o Timestamp c m m m m m m Transport - - m - - - Unsupported r c c c c c c User-Agent R m* m* m* m* m* m* Vary r c c c c c c Via R amr o o o o o o Via c dr m m m m m m WWW-Authenticate 401 m m m m m m _________________________________________________________ Header Where Proxy DES OPT SETUP PLAY PAUSE TRD Table 9: Overview of RTSP header fields related to methods DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN. The Bandwidth request-header field describes the estimated bandwidth available to the client, expressed as a positive integer and measured in bits per second. The bandwidth available to the client may change during an RTSP session, e.g., due to modem retraining. Example: Bandwidth: 4000 14.8 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 H. Schulzrinne et. al. [Page 68]
Internet Draft RTSP February 16, 2004 Header Where Proxy GPR SPR RDR PNG __________________________________________________ Allow 405 m m m m Authorization R o o o o Bandwidth R - o - - Blocksize R - o - - Connection o o o - Content-Base R o o - - Content-Base r o o - - Content-Base 4xx o o o - Content-Encoding R r o o - - Content-Encoding r r o o - - Content-Encoding 4xx r o o o - Content-Language R r o o - - Content-Language r r o o - - Content-Language 4xx r o o o - Content-Length R r * * - - Content-Length r r * * - - Content-Length 4xx r * * * - Content-Location R o o - - Content-Location r o o - - Content-Location 4xx o o o - Content-Type R * * - - Content-Type r * * - - Content-Type 4xx * * * - CSeq Rc m m m m Date am o o o o From R r o o o o Host - - - - Last-Modified R r - - - - Last-Modified r r o - - - Location 3rr o o o o Location R - - m - Proxy-Authenticate 407 amr m m m m Proxy-Require R ar o o o o Public 501 admr m* m* m* m* Range R - - o - Referer R o o o - Require R o o o o Retry-After 3rr,503 o o - - Scale - - - - Session R o o o m Session r c c o m Server R o o o o Server r o o - o Supported R o o o o H. Schulzrinne et. al. [Page 69]
Internet Draft RTSP February 16, 2004 Timestamp c m m m m Unsupported r c c c c User-Agent R m* m* - m* User-Agent r - - m* - Vary r c c - - Via R amr o o o o Via c dr m m m m WWW-Authenticate 401 m m m m __________________________________________________ Header Where Proxy GPR SPR RDR PNG Table 10: Overview of RTSP header fields related to methods GET_PARAMETER, SET_PARAMETER, REDIRECT, and PING. only returns an error (4xx) if the value is syntactically invalid. 14.9 Cache-Control The Cache-Control general-header field is used to specify directives that MUST be obeyed by all caching mechanisms along the request/response chain. Cache directives must be passed through by a proxy or gateway application, regardless of their significance to that application, since the directives may be applicable to all recipients along the request/response chain. It is not possible to specify a cache- directive for a specific cache. Cache-Control should only be specified in a SETUP request and its | response. Note: Cache-Control does not govern the caching of | responses as for HTTP, instead it applies to the media stream | identified by the SETUP request. The caching of RTSP requests are | generally not cacheable, for further information see 15. Below is the | description of the cache directives that can be included in the | Cache-Control header. no-cache: Indicates that the media stream MUST NOT be cached anywhere. This allows an origin server to prevent caching even by caches that have been configured to return stale responses to client requests. public: Indicates that the media stream is cacheable by any cache. private: Indicates that the media stream is intended for a single user and MUST NOT be cached by a shared cache. A H. Schulzrinne et. al. [Page 70]
Internet Draft RTSP February 16, 2004 private (non-shared) cache may cache the media stream. no-transform: An intermediate cache (proxy) may find it useful to convert the media type of a certain stream. A proxy might, for example, convert between video formats to save cache space or to reduce the amount of traffic on a slow link. Serious operational problems may occur, however, when these transformations have been applied to streams intended for certain kinds of applications. For example, applications for medical imaging, scientific data analysis and those using end-to-end authentication all depend on receiving a stream that is bit-for-bit identical to the original media stream. Therefore, if a response includes the no-transform directive, an intermediate cache or proxy MUST NOT change the encoding of the stream. Unlike HTTP, RTSP does not provide for partial transformation at this point, e.g., allowing translation into a different language. only-if-cached: In some cases, such as times of extremely poor network connectivity, a client may want a cache to return only those media streams that it currently has stored, and not to receive these from the origin server. To do this, the client may include the only-if-cached directive in a request. If it receives this directive, a cache SHOULD either respond using a cached media stream that is consistent with the other constraints of the request, or respond with a 504 (Gateway Timeout) status. However, if a group of caches is being operated as a unified system with good internal connectivity, such a request MAY be forwarded within that group of caches. max-stale: Indicates that the client is willing to accept a media stream that has exceeded its expiration time. If max-stale is assigned a value, then the client is willing to accept a response that has exceeded its expiration time by no more than the specified number of seconds. If no value is assigned to max-stale, then the client is willing to accept a stale response of any age. min-fresh: Indicates that the client is willing to accept a media stream whose freshness lifetime is no less than its current age plus the specified time in seconds. That is, the client wants a response that will still be fresh for at least the specified number of seconds. must-revalidate: When the must-revalidate directive is present in a SETUP response received by a cache, that cache MUST H. Schulzrinne et. al. [Page 71]
Internet Draft RTSP February 16, 2004 NOT use the entry after it becomes stale to respond to a subsequent request without first revalidating it with the origin server. That is, the cache must do an end-to-end revalidation every time, if, based solely on the origin server's Expires, the cached response is stale.) proxy-revalidate: The proxy-revalidate directive has the same meaning as the must-revalidate directive, except that it does not apply to non-shared user agent caches. It can be used on a response to an authenticated request to permit the user's cache to store and later return the response without needing to revalidate it (since it has already been authenticated once by that user), while still requiring proxies that service many users to revalidate each time (in order to make sure that each user has been authenticated). Note that such authenticated responses also need the public cache control directive in order to allow them to be cached at all. max-age: When an intermediate cache is forced, by means of a max-age=0 directive, to revalidate its own cache entry, and the client has supplied its own validator in the request, the supplied validator might differ from the validator currently stored with the cache entry. In this case, the cache MAY use either validator in making its own request without affecting semantic transparency. However, the choice of validator might affect performance. The best approach is for the intermediate cache to use its own validator when making its request. If the server replies with 304 (Not Modified), then the cache can return its now validated copy to the client with a 200 (OK) response. If the server replies with a new entity and cache validator, however, the intermediate cache can compare the returned validator with the one provided in the client's request, using the strong comparison function. If the client's validator is equal to the origin server's, then the intermediate cache simply returns 304 (Not Modified). Otherwise, it returns the new entity with a 200 (OK) response. 14.10 Connection See [H14.10]. The use of the connection option "close" in RTSP messages SHOULD be limited to error messages when the server is unable to recover and therefore see it necessary to close the connection. The reason is that the client shall have the choice of continue using a connection indefinitely as long as it sends valid H. Schulzrinne et. al. [Page 72]
Internet Draft RTSP February 16, 2004 messages. 14.11 Content-Base The Content-Base entity-header field may be used to specify the base URL for resolving relative URLs within the entity. Content-Base: rtsp://media.example.com/movie/twister If no Content-Base field is present, the base URL of an entity is defined either by its Content-Location (if that Content-Location URL is an absolute URL) or the URL used to initiate the request, in that order of precedence. Note, however, that the base URL of the contents within the entity-body may be redefined within that entity-body. 14.12 Content-Encoding See [H14.11] 14.13 Content-Language See [H14.12] 14.14 Content-Length The Content-Length general-header field contains the length of the content of the method (i.e. after the double CRLF following the last header). Unlike HTTP, it MUST be included in all messages that carry content beyond the header portion of the message. If it is missing, a default value of zero is assumed. It is interpreted according to [H14.13]. 14.15 Content-Location See [H14.14] 14.16 Content-Type See [H14.17]. Note that the content types suitable for RTSP are likely to be restricted in practice to presentation descriptions and parameter-value types. 14.17 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 | H. Schulzrinne et. al. [Page 73]
Internet Draft RTSP February 16, 2004 sequence number, the corresponding response will have the same | number. Any retransmitted request must contain the same sequence | number as the original (i.e. the sequence number is not incremented | for retransmissions of the same request). For each new RTSP request | the CSeq value SHALL be incremented by one. The initial sequence | number MAY be any number, however it is RECOMMENDED to start at 1. | Each sequence number series is unique between each requester and | responder, i.e. the client has one series for its request to a server | and the server has another when sending request to the client. Each | requester and responder is identified with its network address. Example: CSeq: 239 14.18 Date See [H14.18]. An RTSP message containing a body MUST include a Date header if the sending host has a clock. Servers SHOULD include a Date header in all other RTSP messages. 14.19 Expires The Expires entity-header field gives a date and time after which the description or media-stream should be considered stale. The interpretation depends on the method: DESCRIBE response: The Expires header indicates a date and time after which the description SHOULD be considered stale. SETUP response: The Expires header indicate a date and time after which the media stream SHOULD be considered stale. A stale cache entry may not normally be returned by a cache (either a proxy cache or an user agent cache) unless it is first validated with the origin server (or with an intermediate cache that has a fresh copy of the entity). See section 15 for further discussion of the expiration model. The presence of an Expires field does not imply that the original resource will change or cease to exist at, before, or after that time. The format is an absolute date and time as defined by HTTP-date in [H3.3]; it MUST be in RFC1123-date format: H. Schulzrinne et. al. [Page 74]
Internet Draft RTSP February 16, 2004 An example of its use is Expires: Thu, 01 Dec 1994 16:00:00 GMT RTSP/1.0 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/1.0 servers SHOULD NOT send Expires dates more than one year in the future. The presence of an Expires header field with a date value of some time in the future on a media stream that otherwise would by default be non-cacheable indicates that the media stream is cacheable, unless indicated otherwise by a Cache-Control header field (Section 14.9). 14.20 From See [H14.22]. 14.21 Host The Host HTTP request header field [H14.23] is not needed for RTSP, and SHALL NOT be sent. It SHALL be silently ignored if received. 14.22 If-Match See [H14.24]. The If-Match request-header field is especially useful for ensuring the integrity of the presentation description, in both the case where it is fetched via means external to RTSP (such as HTTP), or in the case where the server implementation is guaranteeing the integrity of the description between the time of the DESCRIBE message and the SETUP message. The identifier is an opaque identifier, and thus is not specific to any particular session description language. 14.23 If-Modified-Since The If-Modified-Since request-header field is used with the DESCRIBE H. Schulzrinne et. al. [Page 75]
Internet Draft RTSP February 16, 2004 and SETUP methods to make them conditional. If the requested variant has not been modified since the time specified in this field, a description will not be returned from the server (DESCRIBE) or a stream will not be set up (SETUP). Instead, a 304 (Not Modified) response SHALL be returned without any message-body. An example of the field is: If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT 14.24 Last-Modified The Last-Modified entity-header field indicates the date and time at which the origin server believes the presentation description or media stream was last modified. See [H14.29]. For the methods DESCRIBE, the header field indicates the last modification date and time of the description, for SETUP that of the media stream. 14.25 Location See [H14.30]. 14.26 Proxy-Authenticate See [H14.33]. 14.27 Proxy-Require The Proxy-Require request-header field is used to indicate proxy- sensitive features that MUST be supported by the proxy. Any Proxy- Require header features that are not supported by the proxy MUST be negatively acknowledged by the proxy to the client using the Unsupported header. The proxy SHALL use the 551 (Option Not Supported) status code in the response. Any feature tag included in the Proxy-Require does not apply to the server. To ensure that a feature is supported by both proxies and servers the tag must be included in also a Require header. See Section 14.32 for more details on the mechanics of this message and a usage example. Example of use: Proxy-Require: play.basic H. Schulzrinne et. al. [Page 76]
Internet Draft RTSP February 16, 2004 14.28 Public The Public response-header field lists the set of methods supported by the server. The purpose of this field is strictly to inform the recipient of the capabilities of the server regarding unusual methods. The methods listed may or may not be applicable to the Request-URL; the Allow header field (section 14.7) MAY be used to indicate methods allowed for a particular URL. Example of use: Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN This header field applies only to the server directly connected to the client (i.e., the nearest neighbor in a chain of connections). If the response passes through a proxy, the proxy MUST either remove the Public header field or replace it with one applicable to its own capabilities. 14.29 Range The Range request and response header field specifies a range of | time. The header is used in PLAY request to indicate the range of | time the client desires the server to play back. The Range header in | a PLAY indicates what range of time that is actually being played. In | a SETUP response the header MAY be used, to ensure correct | synchronization information after changes of transport parameters. | The range can be specified in a number of units. This specification | defines the smpte (Section 3.4), npt (Section 3.5), and clock | (Section 3.6) range units. Within RTSP, byte ranges [H14.35.1] are | normally not meaningful, and the behavior is unspecified, but they | and other extended units MAY be used. Servers supporting the Range | header MUST understand the NPT range format and SHOULD understand the | SMPTE range format. If the Range header is given in a time format | that is not understood, the recipient should return 456 (Header Field | Not Valid for Resource) and include a Accept-Ranges header indicating | which time format that is supported for this resource. The header MAY contain a time parameter in UTC, specifying the time at which the operation is to be made effective. This functionality SHALL only be used with the REDIRECT method. Ranges are half-open intervals, including the first point, but excluding the second point. In other words, a range of A-B starts exactly at time A, but stops just before B. Only the start time of a H. Schulzrinne et. al. [Page 77]
Internet Draft RTSP February 16, 2004 media unit such as a video or audio frame is relevant. As an 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. Example: Range: clock=19960213T143205Z-;time=19970123T143720Z The notation is similar to that used for the HTTP/1.1 [4] byte-range header. It allows clients to select an excerpt from the media object, and to play from a given point to the end as well as from the current location to a given point. The start of playback can be scheduled for any time in the future, although a server may refuse to keep server resources for extended idle periods. 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 14.34) indicates a negative scale value. For example, this would be the case when a playback in reverse is desired. Example: Scale: -1 Range: npt=15-10 Decreasing ranges are still half open intervals as described above. Thus, For range A-B, A is closed and B is open. In the above example, 15 is closed and 10 is open. An exception to this rule is the case when B=0 in a decreasing range. In this case, the range is closed on both ends, as otherwise there would be no way to reach 0 on a reverse playback. H. Schulzrinne et. al. [Page 78]
Internet Draft RTSP February 16, 2004 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. 14.30 Referer See [H14.36]. The URL refers to that of the presentation description, typically retrieved via HTTP. 14.31 Retry-After See [H14.37]. 14.32 Require The Require request-header field is used by clients or servers to ensure that the other end-point supports features that are required in respect to this request. It can also be used to query if the other end-point supports certain features, however the use of the Supported (Section 14.38) is much more effective in this purpose. The server MUST respond to this header by using the Unsupported header to negatively acknowledge those feature-tags which are NOT supported. The response SHALL use the error code 551 (Option Not Supported). This header does not apply to proxies, for the same functionality in respect to proxies see, header Proxy-Require (Section 14.27). 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: C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0 H. Schulzrinne et. al. [Page 79]
Internet Draft RTSP February 16, 2004 CSeq: 302 Require: funky-feature Funky-Parameter: funkystuff S->C: RTSP/1.0 551 Option not supported CSeq: 302 Unsupported: funky-feature In this example, "funky-feature" is the feature-tag which indicates to the client that the fictional Funky-Parameter field is required. The relationship between "funky-feature" and Funky-Parameter is not communicated via the RTSP exchange, since that relationship is an immutable property of "funky-feature" and thus should not be transmitted with every exchange. Proxies and other intermediary devices SHOULD ignore features that are not understood in this field. If a particular extension requires that intermediate devices support it, the extension should be tagged in the Proxy-Require field instead (see Section 14.27). 14.33 RTP-Info The RTP-Info response-header field is used to set RTP-specific | parameters in the PLAY response. These parameters correspond to the | synchronization source identified by the ssrc parameter of the | Transport response header in the SETUP reponse. For streams using RTP | as transport protocol the RTP-Info header SHOULD be part of a 200 | response to PLAY. | The exclusion of the RTP-Info in a PLAY response for RTP | transported media will result in that a client needs to | synchronize the media streams using RTCP. This may have | negative impact as the RTCP can be lost, and does not need | to be particulary timely in their arrival. Also | functionality as informing the client from which packet a | seek has occurred is affected. | The RTP-Info MAY also be included in SETUP responses to provide | synchronization information when changing transport parameters, see | 11.3. | The header can carry the following parameters: url: Indicates the stream URL which for which the following RTP H. Schulzrinne et. al. [Page 80]
Internet Draft RTSP February 16, 2004 parameters correspond, this URL MUST be the same used in the SETUP request for this media stream. Any relative URL SHALL use the request URL as base URL. seq: Indicates the sequence number of the first packet of the stream. 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. rtptime: Indicates the RTP timestamp corresponding to the time value in the Range response header. (Note: For aggregate control, a particular stream may not actually generate a packet for the Range time value returned or implied. Thus, there is no guarantee that the packet with the sequence number indicated by seq actually has the timestamp indicated by rtptime.) The client uses this value to calculate the mapping of RTP time to NPT. A mapping from RTP timestamps to NTP timestamps (wall clock) is available via RTCP. However, this information is not sufficient to generate a mapping from RTP timestamps to NPT. 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. Additionally, the RTP-Info header parameter fields only apply to a | single SSRC within a stream (the SSRC reported in the transport | response header; see section 14.40). If there are multiple | synchronization sources (SSRCs) present within a RTP session | transmitting media, RTCP must be used to map RTP and NTP timestamps | for those sources, for both synchronization and drift-checking. Due | to backwards compatibility reasons these shortcomings can't be fixed | without defining a new header, which is for future work if needed. Additional constraint: The syntax element "safe-url" (see section 17.2.3) MUST NOT contain the semicolon (";") or comma (",") characters. The quoted-url form SHOULD only be used when a URL does not meet the safe-url constraint, in order to ensure compatibility with implementations conformant to RFC 2326 [1]. H. Schulzrinne et. al. [Page 81]
Internet Draft RTSP February 16, 2004 Example: | RTP-Info: url=rtsp://example.com/bar.avi/streamid=0;seq=45102, | url=rtsp://example.com/bar.avi/streamid=1;seq=30211 | 14.34 Scale A scale value of 1 indicates normal play at the normal forward | viewing rate. If not 1, the value corresponds to the rate with | respect to normal viewing rate. For example, a ratio of 2 indicates | twice the normal viewing rate ("fast forward") and a ratio of 0.5 | indicates half the normal viewing rate. In other words, a ratio of 2 | has normal play time increase at twice the wallclock rate. For every | second of elapsed (wallclock) time, 2 seconds of content will be | delivered. A negative value indicates reverse direction. For certain | media transports this may require certain considerations to work | consistent, see section B.1 for description on how RTP handles this. Unless requested otherwise by the Speed parameter, the data rate SHOULD not be changed. Implementation of scale changes depends on the server and media type. For video, a server may, for example, deliver only key frames or selected key frames. For audio, it may time-scale the audio while preserving pitch or, less desirably, deliver fragments of audio. The server should try to approximate the viewing rate, but may restrict the range of scale values that it supports. The response MUST contain the actual scale value chosen by the server. If the server does not implement the possibility to scale, it will not return a Scale header. A server supporting Scale operations for PLAY SHALL indicate this with the use of the "play.scale" feature- tags. When indicating a negative scale for a reverse playback, the Range header must indicate a decreasing range as described in section 14.29. Example of playing in reverse at 3.5 times normal rate: Scale: -3.5 Range: npt=15-10 14.35 Speed H. Schulzrinne et. al. [Page 82]
Internet Draft RTSP February 16, 2004 The Speed request-header field requests the server to deliver data to the client at a particular speed, contingent on the server's ability and desire to serve the media stream at the given speed. Implementation by the server is OPTIONAL. The default is the bit rate of the stream. The parameter value is expressed as a decimal ratio, e.g., a value of 2.0 indicates that data is to be delivered twice as fast as normal. A speed of zero is invalid. All speeds may not be possible to support. Therefore the actual used speed MUST be included in the response. The lack of a response header is indication of lack of support from the server of this functionality. Support of the speed functionality are indicated by the "play.speed" featuretag. Example: Speed: 2.5 Use of this field changes the bandwidth used for data delivery. It is meant for use in specific circumstances where preview of the presentation at a higher or lower rate is necessary. Implementors should keep in mind that bandwidth for the session may be negotiated beforehand (by means other than RTSP), and therefore re-negotiation may be necessary. When data is delivered over UDP, it is highly recommended that means such as RTCP be used to track packet loss rates. If the data transport is performed over public best-effort networks the sender SHOULD perform congestion control of the stream(s). This can result in that the communicated speed is impossible to maintain. 14.36 Server See [H14.38], however the header syntax is corrected in section 17.2.3. 14.37 Session The Session request-header and response-header field identifies an RTSP session. An RTSP session is created by the server as a result of a successful SETUP request and in the response the session identifier is given to the client. The RTSP session exist until destroyed by a TEARDOWN or timed out by the server. The session identifier is chosen by the server (see Section 3.3) and MUST be returned in the SETUP response. Once a client receives a session identifier, it SHALL be included in any request related to H. Schulzrinne et. al. [Page 83]
Internet Draft RTSP February 16, 2004 that session. This means that the Session header MUST be included in a request using the following methods: PLAY, PAUSE, PING, and TEARDOWN, and MAY be included in SETUP, OPTIONS, SET_PARAMETER, GET_PARAMETER, and REDIRECT, and SHALL NOT be included in DESCRIBE. In a RTSP response the session header SHALL be included in methods, SETUP, PING, PLAY, and PAUSE, and MAY be included in methods, TEARDOWN, and REDIRECT, and if included in the request of the following methods it SHALL also be included in the response, OPTIONS, GET_PARAMETER, and SET_PARAMETER, and SHALL NOT be included in DESCRIBE. Note that RFC 2326 servers and client may in some cases not include or return a Session header when expected according to the above text. Any client or server is RECOMMENDED to be forgiving of this error if possible (which it is in many cases). The timeout parameter MAY be included in a SETUP response, and SHALL | NOT be included in requests. The server uses it to indicate to the | client how long the server is prepared to wait between RTSP commands | or other signs of life before closing the session due to lack of | activity (see below and Section A). The timeout is measured in | seconds, with a default of 60 seconds (1 minute). The length of the | session timeout SHALL NOT be changed in a established session. The mechanisms for showing liveness of the client is, any RTSP request with a Session header, if RTP & RTCP is used an RTCP message, or through any other used media protocol capable of indicating liveness of the RTSP client. It is RECOMMENDED that a client does not wait to the last second of the timeout before trying to send a liveness message. The RTSP message may be lost or when using reliable protocols, such as TCP, the message may take some time to arrive safely at the receiver. To show liveness between RTSP request issued to accomplish other things, the following mechanisms can be used, in descending order of preference: RTCP: If RTP is used for media transport RTCP SHOULD be used. If RTCP is used to report transport statistics, it SHALL also work as keep alive. The server can determine the client by used network address and port together with the fact that the client is reporting on the servers SSRC(s). A downside of using RTCP is that it only gives statistical guarantees to reach the server. However that probability is so low that it can be ignored in most cases. For example, a session with 60 seconds timeout and enough bitrate assigned to RTCP messages to send a message from client to server on average every 5 seconds. That client have for a network with 5 % packet loss, the probability to fail showing liveness sign in that session within the timeout interval H. Schulzrinne et. al. [Page 84]
Internet Draft RTSP February 16, 2004 of 2.4*E-16. In sessions with shorter timeout times, or much higher packet loss, or small RTCP bandwidths SHOULD also use any of the mechanisms below. PING: The use of the PING method is the best of the RTSP based methods. It has no other effects than updating the timeout timer. In that way it will be a minimal message, that also does not cause any extra processing for the server. The downside is that it may not be implemented. A client SHOULD use a OPTIONS request to verify support of the PING at the server. It is also possible to detect support by sending a PING to the server. If a 200 (OK) message is received the server supports it. In case a 501 (Not Implemented) is received it does not support PING and there is no meaning in continue trying. Also the reception of a error message will also mean that the liveness timer has not been updated. SET_PARAMETER: When using SET_PARAMETER for keep alive, no body SHOULD be included. This method is basically as good as PING, however the implementation support of the method is today limited. The same considerations as for PING apply regarding checking of support in server and proxies. OPTIONS: This method does also work. However it causes the server to perform unnecessary processing and result in bigger responses than necessary for the task. The reason for this is that the Public is always included creating overhead. Note that a session identifier identifies an RTSP session across transport sessions or connections. RTSP requests for a given session can use different URLs (Presentation and media URLs). Note, that there are restrictions depending on the session which URLs that are acceptable for a given method. However, multiple "user" sessions for the same URL from the same client will require use of different session identifiers. The session identifier is needed to distinguish several delivery requests for the same URL coming from the same client. The response 454 (Session Not Found) SHALL be returned if the session identifier is invalid. 14.38 Supported The Supported header field enumerates all the extensions supported by H. Schulzrinne et. al. [Page 85]
Internet Draft RTSP February 16, 2004 the client or server. When offered in a request, the receiver MUST respond with its corresponding Supported header. The Supported header field contains a list of feature-tags, described in Section 3.7, that are understood by the client or server. Example: C->S: OPTIONS rtsp://example.com/ RTSP/1.0 Supported: foo, bar, blech S->C: RTSP/1.0 200 OK Supported: bar, blech, baz 14.39 Timestamp The Timestamp general-header field describes when the client sent the request to the server. The value of the timestamp is of significance only to the client and may use any timescale. The server MUST echo the exact same value and MAY, if it has accurate information about this, add a floating point number indicating the number of seconds that has elapsed since it has received the request. The timestamp is used by the client to compute the round-trip time to the server so that it can adjust the timeout value for retransmissions. It also resolves retransmission ambiguities for unreliable transport of RTSP. 14.40 Transport The Transport request and response header field indicates which transport protocol is to be used and configures its parameters such as destination address, compression, multicast time-to-live and destination port for a single stream. It sets those values not already determined by a presentation description. Transports are comma separated, listed in order of preference. | Parameters may be added to each transport, separated by a semicolon. | The server SHOULD return a Transport response-header field in the | response to indicate the values actually chosen. The Transport header | field MAY also be used to change certain transport parameters. A | server MAY refuse to change parameters of an existing stream. | A Transport request header field MAY contain a list of transport | options acceptable to the client, in the form of multiple | transportspec entries. In that case, the server MUST return the | single option (transport-spec) which was actually chosen. The number | of transportspec entries is expected to be limited as the client will | H. Schulzrinne et. al. [Page 86]
Internet Draft RTSP February 16, 2004 get guidance on what configurations that are possible from the | presentation description. A transport-spec transport option may only contain one of any given | parameter within it. Parameters may be given in any order. | Additionally, it may only contain the unicast or multicast transport | parameter. Unknown transport parameters SHALL be ignored. The | requester need to ensure that the responder understands the | parameters through the use of feature tags and the Require header. | The usage of any parameter that was not defined in RFC 2326 or in an | extended way requires that request or response contains a Require | header with the "play.basic" feature tag. The Transport header field is restricted to describing a single media stream. (RTSP can also control multiple streams as a single entity.) Making it part of RTSP rather than relying on a multitude of session description formats greatly simplifies designs of firewalls. The syntax for the transport specifier is transport/profile/lower-transport. The default value for the "lower-transport" parameters is specific to the profile. For RTP/AVP, the default is UDP. There is three different methods for how to specify where the media | should be delivered: | o The presence of this parameter and its values indicates | address and port pairs for one or more IP flow necessary for | the media transport. This is an improved version of the | Destination parameter. | o The presence of this parameter and its value indicates what IP | address the media shall be delivered to. This method is kept | for backwards compatibility reasons, dest_addr is a better | choice. | o The lack of of both of the above parameters indicates that the | server SHALL send media to same address for which the RTSP | messages originates. | The choice of method for indicating where the media shall be | delivered depends on the use case. In many case the only allowed | H. Schulzrinne et. al. [Page 87]
Internet Draft RTSP February 16, 2004 method will be to use no explicit indication and have the server | deliver media to the source of the RTSP messages. | An RTSP proxy will also 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 MUST indicate such capability by including two full transport-specs with separate parameters for each. destination: The address of the stream recipient to which a stream will be sent. The client originating the RTSP request may specify the destination address of the stream recipient with the destination parameter. When the destination field 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 SHOULD authenticate the client originating the request and SHOULD log such attempts before allowing the client to direct a media stream to a recipient address not chosen by the server. While, this is particularly important if RTSP commands are issued via UDP, implementations cannot rely on TCP as reliable means of client identification by itself either. 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 recipient itself and the credentials passed along to the server. However, in certain cases, such as when recipient address is a multicast group, or when the recipient is unable to communicate with the server in an out-of-band manner, this may not be possible. In these cases server may chose another method such as a server-resident authorization list to ensure that the request originator has the proper credentials to request stream delivery to the recipient. H. Schulzrinne et. al. [Page 88]
Internet Draft RTSP February 16, 2004 This parameter SHALL NOT be used when src_addr and dest_addr is used in a transport declaration. For IPv6 addresses it is RECOMMENDED that they be given as fully qualified domain to make it backwards compatible with RFC 2326 implementations. source: If the source address for the stream is different than can be derived from the RTSP endpoint address (the server in playback), the source address SHOULD be specified. To maintain backwards compatibility with RFC 2326, any IPv6 host's address must be given as a fully qualified domain name. This parameter SHALL NOT be used when src_addr and dest_addr is used in a transport declaration. 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. layers: The number of multicast layers to be used for this media stream. The layers are sent to consecutive addresses starting at the destination address. dest_addr: A general destination address parameter that can | contain one or more address and port pair. For each | combination of Protocol/Profile/Lower Transport the | interpretation of the address or addresses needs to be | defined. The host address part of the tuple MAY be empty, | for example ":8000", in cases when only destination port is | desired to be specified. The client or server SHALL NOT use this parameter unless both client and server has shown support. This parameter MUST be supported by client and servers that implements this specification. Support is indicated by the use of the feature-tag "play.basic". This parameter SHALL NOT be used in the same transport specification as any of the parameters "destination", "source", "port", "client_port", and "server_port". The same security consideration that are given for the "Destination" parameter does also applies to this parameter. This parameter can be used for redirecting traffic to recipient not desiring the media traffic. src_addr: A General source address parameter that can contain H. Schulzrinne et. al. [Page 89]
Internet Draft RTSP February 16, 2004 one or more address and port pair. For each combination of Protocol/Profile/Lower Transport the interpretation of the address or addresses needs to be defined. The client or server SHALL NOT use this parameter unless both client and server has shown support. This parameter MUST be supported by client and servers that implements this specification. Support is indicated by the use the feature-tag "play.basic". This parameter SHALL NOT be used in the same transport specification as any of the parameters "destination", "source", "port", "client_port", and "server_port". This parameter MUST be specified by the server if it | transmits media packets from another address than the one | RTSP messages are sent to. This will allow the client to | verify source address and give it a destination address for | its RTCP feedback packets if RTP is used. The address or | addresses indicated in the src_addr parameter SHOULD be | used both for sending and receiving of the media streams | data packets. The main reasons are three: First by sending | from the indicated ports the source address will be known | by the receiver of the packet. Secondly, in the presence of | NATs some traversal mechanism requires either knowledge | from which address and port a packet flow is coming, or | having the possibility to send data to the sender port. mode: The mode parameter indicates the methods to be supported for this session. Valid values are PLAY and RECORD. If not provided, the default is PLAY. The RECORD value was defined in RFC 2326 and is deprecated in this specification. append: The append parameter was used together with RECORD and is now deprecated. 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 12. The argument provides the channel number to be used in the $ statement and MUST be present. This parameter MAY be specified as a range, e.g., interleaved=4-5 in cases where the transport choice for the media stream requires it, e.g. for RTP with RTCP. The channel number given in the request are only a guidance from the client to the server on what channel number(s) to use. The server MAY set any valid channel number in the response. The declared channel(s) are bi-directional, so both end-parties MAY send data on the given channel. One H. Schulzrinne et. al. [Page 90]
Internet Draft RTSP February 16, 2004 example of such usage is the second channel used for RTCP, where both server and client sends RTCP packets on the same channel. This allows RTP/RTCP to be handled similarly to the way that it is done with UDP, i.e., one channel for RTP and the other for RTCP. Multicast-specific: ttl: multicast time-to-live. RTP-specific: These parameters are MAY only be used if the media transport protocol is RTP. port: This parameter provides the RTP/RTCP port pair for a multicast session. It is should be specified as a range, e.g., port=3456-3457 client_port: This parameter provides the unicast RTP/RTCP port pair on the client where media data and control information is to be sent. It is specified as a range, e.g., port=3456-3457. This parameter SHALL NOT be used when src_addr and dest_addr is used in a transport declaration. server_port: This parameter provides the unicast RTP/RTCP port pair on the server where media data and control information is to be sent. It is specified as a range, e.g., port=3456-3457. This parameter SHALL NOT be used when src_addr and dest_addr is used in a transport declaration. ssrc: The ssrc parameter, if included in a SETUP response, indicates the RTP SSRC [15] value that will be used by the media server for RTP packets within the stream. It is expressed as an eight digit hexadecimal value. If the server does not act as a synchronization source for stream data (for instance, server is a translator, reflector, etc.) the value will be the "packet sender's SSRC" that would have been used in the RTCP Receiver Reports generated by the server, regardless of whether the server actually generates RTCP RRs. If there are multiple sources within the stream, the ssrc parameter only indicates the value for a single synchronization source. Other sources must be deduced from the actual RTP/RTCP stream. H. Schulzrinne et. al. [Page 91]
Internet Draft RTSP February 16, 2004 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 [15]. If the parameter is included in the transport header of a SETUP request, the server MAY ignore it, and choose an appropriate SSRC for the stream. The server MAY set the ssrc parameter in the transport header of the response. The combination of transport protocol, profile and lower transport needs to be defined. A number of combinations are defined in the appendix B. 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/1.0 CSeq: 302 Transport: RTP/AVP;multicast;mode="PLAY", RTP/AVP;unicast;client_port=3456-3457;mode="PLAY" S->C: RTSP/1.0 200 OK CSeq: 302 Date: 23 Jan 1997 15:35:06 GMT Session: 47112344 Transport: RTP/AVP;unicast;client_port=3456-3457; server_port=6256-6257;mode="PLAY" 14.41 Unsupported The Unsupported response-header field lists the features not supported by the server. In the case where the feature was specified via the Proxy-Require field (Section 14.27), if there is a proxy on the path between the client and the server, the proxy MUST send a response message with a status code of 551 (Option Not Supported). The request SHALL NOT be forwarded. See Section 14.32 for a usage example. 14.42 User-Agent See [H14.43] for explanation, however the syntax is clarified due to an error in RFC 2616. A Client SHOULD include this header in all RTSP messages it sends. H. Schulzrinne et. al. [Page 92]
Internet Draft RTSP February 16, 2004 14.43 Vary See [H14.44] 14.44 Via See [H14.45]. 14.45 WWW-Authenticate See [H14.47]. 15 Caching In HTTP, response-request pairs are cached. RTSP differs significantly in that respect. Responses are not cacheable, with the exception of the presentation description returned by DESCRIBE. (Since the responses for anything but DESCRIBE and GET_PARAMETER do not return any data, caching is not really an issue for these requests.) However, it is desirable for the continuous media data, typically delivered out-of-band with respect to RTSP, to be cached, as well as the session description. On receiving a SETUP or PLAY request, a proxy ascertains whether it has an up-to-date copy of the continuous media content and its description. It can determine whether the copy is up-to-date by issuing a SETUP or DESCRIBE request, respectively, and comparing the Last-Modified header with that of the cached copy. If the copy is not up-to-date, it modifies the SETUP transport parameters as appropriate and forwards the request to the origin server. Subsequent control commands such as PLAY or PAUSE then pass the proxy unmodified. The proxy delivers the continuous media data to the client, while possibly making a local copy for later reuse. The exact behavior allowed to the cache is given by the cache-response directives described in Section 14.9. A cache MUST answer any DESCRIBE requests if it is currently serving the stream to the requestor, as it is possible that low-level details of the stream description may have changed on the origin-server. Note that an RTSP cache, unlike the HTTP cache, is of the "cut- through" variety. Rather than retrieving the whole resource from the origin server, the cache simply copies the streaming data as it passes by on its way to the client. Thus, it does not introduce additional latency. To the client, an RTSP proxy cache appears like a regular media server, to the media origin server like a client. Just as an HTTP cache has to store the content type, content language, and so on for H. Schulzrinne et. al. [Page 93]
Internet Draft RTSP February 16, 2004 the objects it caches, a media cache has to store the presentation description. Typically, a cache eliminates all transport-references (that is, 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 Examples This section contains several different examples trying to illustrate | possible ways of using RTSP. The examples can also help with the | understanding of how functions of RTSP work. However remember that | this is examples and the normative and syntax description in the | other chapters takes precedence. Please also note that many of the | example MAY contain syntax illegal line breaks to accommodate the | formatting restriction that the RFC series impose. | 16.1 Media on Demand (Unicast) | 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. | C->W: GET /twister.sdp HTTP/1.1 | Host: www.example.com | Accept: application/sdp | W->C: HTTP/1.0 200 OK | Date: 23 Jan 1997 15:35:06 GMT | Content-Type: application/sdp | Content-Length: 255 | Expires: 23 Jan 1998 15:35:06 GMT | v=0 | o=- 2890844526 2890842807 IN IP4 192.16.24.202 | s=RTSP Session | H. Schulzrinne et. al. [Page 94]
Internet Draft RTSP February 16, 2004 e=adm@example.com | a=range:npt=0-1:49:34 | t=0 0 | m=audio 0 RTP/AVP 0 | a=control:rtsp://audio.example.com/twister/audio.en | m=video 0 RTP/AVP 31 | a=control:rtsp://video.example.com/twister/video | C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 | CSeq: 1 | User-Agent: PhonyClient/1.2 | Transport: RTP/AVP/UDP;unicast;client_port=3056-3057, | RTP/AVP/TCP;unicast;interleave=0-1 | A->C: RTSP/1.0 200 OK | CSeq: 1 | Session: 12345678 | Transport: RTP/AVP/UDP;unicast;client_port=3056-3057; | server_port=5000-5001 | 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 | C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 | CSeq: 1 | User-Agent: PhonyClient/1.2 | Transport: RTP/AVP/UDP;unicast;client_port=3058-3059, | RTP/AVP/TCP;unicast;interleave=0-1 | V->C: RTSP/1.0 200 OK | CSeq: 1 | Session: 23456789 | Transport: RTP/AVP/UDP;unicast;client_port=3058-3059; | server_port=5002-5003 | Date: 23 Jan 1997 15:35:12 GMT | Server: PhonyServer/1.0 | Cache-Control: public | Expires: 24 Jan 1997 15:35:12 GMT | Accept-Ranges: NPT, SMPTE | C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 | CSeq: 2 | User-Agent: PhonyClient/1.2 | Session: 23456789 | Range: smpte=0:10:00- | H. Schulzrinne et. al. [Page 95]
Internet Draft RTSP February 16, 2004 V->C: RTSP/1.0 200 OK | CSeq: 2 | Session: 23456789 | Range: smpte=0:10:00-1:49:23 | RTP-Info: url=rtsp://video.example.com/twister/video; | seq=12312232;rtptime=78712811 | Server: PhonyServer/2.0 | Date: 23 Jan 1997 15:35:13 GMT | C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 | CSeq: 2 | User-Agent: PhonyClient/1.2 | Session: 12345678 | Range: smpte=0:10:00- | A->C: RTSP/1.0 200 OK | CSeq: 2 | Session: 12345678 | Range: smpte=0:10:00-1:49:23 | RTP-Info: url=rtsp://audio.example.com/twister/audio.en; | 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/1.0 | CSeq: 3 | User-Agent: PhonyClient/1.2 | Session: 12345678 | A->C: RTSP/1.0 200 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/1.0 | CSeq: 3 | User-Agent: PhonyClient/1.2 | Session: 23456789 | V->C: RTSP/1.0 200 OK | CSeq: 3 | Server: PhonyServer/2.0 | Date: 23 Jan 1997 15:36:52 GMT | H. Schulzrinne et. al. [Page 96]
Internet Draft RTSP February 16, 2004 Even though the audio and video track are on two different servers, | and may start at slightly different times and may drift with respect | to each other, the client can synchronize the two using standard RTP | methods, in particular the time scale contained in the RTCP sender | reports. Initial synchronization is achieved through the RTP-Info and | Range headers information in the PLAY response. | 16.2 Streaming of 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 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 URL. | The following is an example of using a single RTSP session to control | multiple streams. It also illustrates the use of aggregate URLs. In a | container file it is also desirable to not write any URL parts which | is not kept, when the container is distributed, like the host and | most of the path element. Therefore this example also uses the "*" | and relative URL in the delivered SDP. | Client C requests a presentation from media server M. The movie is | stored in a container file. The client has obtained an RTSP URL to | the container file. | C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/1.0 | CSeq: 1 | User-Agent: PhonyClient/1.2 | M->C: RTSP/1.0 200 OK | H. Schulzrinne et. al. [Page 97]
Internet Draft RTSP February 16, 2004 CSeq: 1 | Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:35:06 GMT | Content-Type: application/sdp | Content-Length: 257 | Content-Base: rtsp://example.com/twister.3gp/ | Expires: 24 Jan 1997 15:35:06 GMT | v=0 | o=- 2890844256 2890842807 IN IP4 172.16.2.93 | s=RTSP Session | i=An Example of RTSP Session Usage | e=adm@example.com | a=control: * | a=range: npt=0-0:10:34.10 | t=0 0 | m=audio 0 RTP/AVP 0 | a=control: trackID=1 | m=video 0 RTP/AVP 26 | a=control: trackID=4 | C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/1.0 | CSeq: 2 | User-Agent: PhonyClient/1.2 | Require: play.basic | Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001" | M->C: RTSP/1.0 200 OK | CSeq: 2 | Server: PhonyServer/1.0 | Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001; | src_addr="172.16.2.93:9000"/"172.16.2.93:9001" | ssrc=93CB001E | Session: 12345678 | Expires: 24 Jan 1997 15:35:12 GMT | Date: 23 Jan 1997 15:35:12 GMT | Accept-Ranges: NPT | C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/1.0 | CSeq: 3 | User-Agent: PhonyClient/1.2 | Require: play.basic | Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003" | Session: 12345678 | M->C: RTSP/1.0 200 OK | CSeq: 3 | Server: PhonyServer/1.0 | H. Schulzrinne et. al. [Page 98]
Internet Draft RTSP February 16, 2004 Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003; | src_addr="172.16.2.93:9002"/"172.16.2.93:9003"; | ssrc=A813FC13 | Session: 12345678 | Expires: 24 Jan 1997 15:35:13 GMT | Date: 23 Jan 1997 15:35:13 GMT | Accept-Range: NPT | C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/1.0 | CSeq: 4 | User-Agent: PhonyClient/1.2 | Range: npt=0-10, npt=30- | Session: 12345678 | M->C: RTSP/1.0 200 OK | CSeq: 4 | Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:35:14 GMT | Session: 12345678 | Range: npt=0-10, npt=30-623.10 | RTP-Info: url=rtsp://example.com/twister.3gp/trackID=4; | seq=12345;rtptime=3450012, | url=rtsp://example.com/twister.3gp/trackID=1; | seq=54321;rtptime=2876889 | C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/1.0 | CSeq: 5 | User-Agent: PhonyClient/1.2 | Session: 12345678 | M->C: RTSP/1.0 200 OK | CSeq: 5 | Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:36:01 GMT | Session: 12345678 | Range: npt=34.57-623.10 | C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/1.0 | CSeq: 6 | User-Agent: PhonyClient/1.2 | Range: npt=34.57-623.10 | Session: 12345678 | M->C: RTSP/1.0 200 OK | CSeq: 6 | Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:36:01 GMT | Session: 12345678 | H. Schulzrinne et. al. [Page 99]
Internet Draft RTSP February 16, 2004 Range: npt=34.57-623.10 | RTP-Info: url=rtsp://example.com/twister.3gp/trackID=4; | seq=12555;rtptime=6330012, | url=rtsp://example.com/twister.3gp/trackID=1; | seq=55021;rtptime=3132889 | 16.3 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 SHOULD use the rules set forth in the session | description for request URLs, rather than assuming that a consistent | URL may always be used throughout. Here's an example of how a multi- | stream server might expect a single-stream file to be served: | C->S: DESCRIBE rtsp://foo.com/test.wav RTSP/1.0 | Accept: application/x-rtsp-mh, application/sdp | CSeq: 1 | User-Agent: PhonyClient/1.2 | S->C: RTSP/1.0 200 OK | CSeq: 1 | Content-base: rtsp://foo.com/test.wav/ | Content-type: application/sdp | Content-length: 48 | Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:35:06 GMT | Expires: 23 Jan 1997 17:00:00 GMT | v=0 | o=- 872653257 872653257 IN IP4 172.16.2.187 | s=mu-law wave file | i=audio test | t=0 0 | a=control: * | m=audio 0 RTP/AVP 0 | a=control:streamid=0 | C->S: SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0 | Transport: RTP/AVP/UDP;unicast; | client_port=6970-6971;mode="PLAY" | CSeq: 2 | User-Agent: PhonyClient/1.2 | H. Schulzrinne et. al. [Page 100]
Internet Draft RTSP February 16, 2004 S->C: RTSP/1.0 200 OK | Transport: RTP/AVP/UDP;unicast;client_port=6970-6971; | server_port=6970-6971;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 | C->S: PLAY rtsp://foo.com/test.wav RTSP/1.0 | CSeq: 3 | User-Agent: PhonyClient/1.2 | Session: 2034820394 | S->C: RTSP/1.0 200 OK | CSeq: 3 | Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:35:08 GMT | Session: 2034820394 | Range: npt=0-600 | RTP-Info: url=rtsp://foo.com/test.wav/streamid=0; | seq=981888;rtptime=3781123 | Note the different URL in the SETUP command, and then the switch back | to the aggregate URL in the PLAY command. This makes complete sense | when there are multiple streams with aggregate control, but is less | than intuitive in the special case where the number of streams is | one. However the server has declared that the aggregated control URL | in the SDP and therefore this is fine. | If however the server had not declared an aggregated control URL it | would be another question, in which the client should consider it | lucky if it works. | In this case, it is also required that servers accept implementations | that uses the non-aggregated interpretation and uses the individual | media URL, like this: | C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/1.0 | CSeq: 3 | User-Agent: PhonyClient/1.2 | 16.4 Live Media Presentation Using Multicast | H. Schulzrinne et. al. [Page 101]
Internet Draft RTSP February 16, 2004 The media server M chooses the multicast address and port. Here, we | assume that the web server only contains a pointer to the full | description, while the media server M maintains the full description. | Editors note: Is this example really valid? In what situations does | it make sense to do a setup to a multicast distribution channel, and | also issue PLAY requests? | C->W: GET /sessions.html HTTP/1.1 | Host: www.example.com | W->C: HTTP/1.1 200 OK | Content-Type: text/html | <html> | ... | <href "Stremed Live Music performance" | src="rtsp://live.example.com/concert/audio"> | ... | </html> | C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0 | CSeq: 1 | Supported: play.basic, play.scale | M->C: RTSP/1.0 200 OK | CSeq: 1 | Content-Type: application/sdp | Content-Length: 181 | Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:35:06 GMT | Supported: play.basic | v=0 | o=- 2890844526 2890842807 IN IP4 192.16.24.202 | s=RTSP Session | m=audio 3456 RTP/AVP 0 | c=IN IP4 224.2.0.1/16 | a=control: rtsp://live.example.com/concert/audio | a=range:npt=0- | C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0 | CSeq: 2 | Transport: RTP/AVP;multicast | M->C: RTSP/1.0 200 OK | CSeq: 2 | H. Schulzrinne et. al. [Page 102]
Internet Draft RTSP February 16, 2004 Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:35:06 GMT | Transport: RTP/AVP;multicast;destination=224.2.0.1; | port=3456-3457;ttl=16 | Session: 0456804596 | Accept-Ranges: NPT, UTC | C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0 | CSeq: 3 | Session: 0456804596 | M->C: RTSP/1.0 200 OK | CSeq: 3 | Server: PhonyServer/1.0 | Date: 23 Jan 1997 15:35:07 GMT | Session: 0456804596 | Range:npt=1256- | RTP-Info: url=rtsp://live.example.com/concert/audio; | seq=1473; rtptime=80000 | 16.5 Capability Negotiation | This examples illustrate how the client and server determines there | capability to support a special feature, in this case "play.scale". | The server through the clients request, and included Supported header | learns that the client is supporting this updated specification, and | also support the playback time scaling feature of RTSP. The server's | response declares that it is also an updated specification minimal | implementation and supports the extra features, of client requested | time scaling and faster than normal transmission rates, plus one | "example.com" proprietary feature "flight". The client also learns | what methods that are possible to use in regards to the indicated | resource. | C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/1.0 | CSeq: 1 | Supported: play.basic, play.scale | User-Agent: PhonyClient/1.2 | S->C: RTSP/1.0 200 OK | CSeq: 1 | Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN | Server: PhonyServer/2.0 | Supported: play.basic, play.scale, play.speed, example.com.flight| H. Schulzrinne et. al. [Page 103]
Internet Draft RTSP February 16, 2004 When the client sends its SETUP request it tells the server that it | is must support the play.scale feature for this session by including | the Require header. | C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/1.0 | CSeq: 3 | User-Agent: PhonyClient/1.2 | Transport: RTP/AVP/UDP;unicast;client_port=3056-3057, | RTP/AVP/TCP;unicast;interleave=0-1 | Require: play.scale | S->C: RTSP/1.0 200 OK | CSeq: 3 | Session: 12345678 | Transport: RTP/AVP/UDP;unicast;client_port=3056-3057; | server_port=5000-5001 | Server: PhonyServer/2.0 | Accept-Ranges: NPT, SMPTE | 17 Syntax The RTSP syntax is described in an augmented Backus-Naur Form (BNF) | as defined in RFC 2234 [5]. | 17.1 Base Syntax | 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)> | DQUOTE = %x22 <US-ASCII double-quote mark (34)> | BACKSLASH = %x5C <US-ASCII backslash (92)> | CRLF = CR LF | H. Schulzrinne et. al. [Page 104]
Internet Draft RTSP February 16, 2004 LWS = [CRLF] 1*( SP / HT ) | TEXT = %x20-7D / %x80-FF <any OCTET except CTLs> | tspecials = "(" / ")" / "<" / ">" / "@" | / "," / ";" / ":" / BACKSLASH / DQUOTE | / "/" / "[" / "]" / "?" / "=" | / "{" / "}" / SP / HT | token = %x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39 | / %x41-5A / %x5E-7A / %x7C / %x7E | 1*<any CHAR except CTLs or tspecials> | quoted-string = ( DQUOTE *(qdtext) DQUOTE ) | qdtext = %x20-21 / %x23-7D / %x80-FF <any TEXT except <">> | quoted-pair = BACKSLASH CHAR | safe = "$" / "-" / "_" / "." / "+" | extra = "!" / "*" / "'" / "(" / ")" / "," | hex = DIGIT / "A" / "B" / "C" / "D" / "E" / "F" / | "a" / "b" / "c" / "d" / "e" / "f" | escape = "%" hex hex | reserved = ";" / "/" / "?" / ":" / "@" / "&" / "=" | unreserved = alpha / digit / safe / extra | xchar = unreserved / reserved / escape | 17.2 RTSP Protocol Definition | 17.2.1 Generic Protocol elements | absoluteURL = as defined in RFC 2396 [12] and RFC2732 [11] | relativeURL = as defined in RFC 2396 [12] and RFC2732 [11] | rtsp_URL = rtsp-scheme "//" host [":" port] | [abs_path ["?" query]] ["#" fragment] | rtsp-scheme = ( "rtsp:" / "rtspu:" / "rtsps:" ) | host = As defined by RFC 2732 [11] | abs_path = As defined by RFC 2396 [12] | port = *DIGIT | query = As defined by RFC 2396 [12] | fragment = As defined by RFC 2396 [12] | smpte-range = smpte-type "=" smpte-range-spec | ;Section 3.4 | smpte-range-spec = ( smpte-time "-" [ smpte-time ] ) | / ( "-" smpte-time ) | smpte-type = "smpte" / "smpte-30-drop" | H. Schulzrinne et. al. [Page 105]
Internet Draft RTSP February 16, 2004 / "smpte-25" / smpte-type-extension | ; other timecodes may be added | smpte-type-extension = token | smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT | [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ] | npt-range = ["npt" "="] npt-range-spec ; Section 3.5 | ; implementations SHOULD use npt= prefix, | ;but SHOULD be prepared to interoperate with | ; RFC 2326 implementations which don't use it. | npt-range-spec = ( npt-time "-" [ npt-time ] ) / ( "-" npt-time ) | npt-time = "now" / npt-sec / npt-hhmmss | npt-sec = 1*DIGIT [ "." *DIGIT ] | npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ] | npt-hh = 1*DIGIT ; any positive number | npt-mm = 1*2DIGIT ; 0-59 | npt-ss = 1*2DIGIT ; 0-59 | utc-range = "clock" "=" utc-range-spec ; Section 3.6 | utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time ) | utc-time = utc-date "T" utc-clock "Z" | utc-date = 8DIGIT ; < YYYYMMDD > | utc-clock = 6DIGIT [ "." fraction ]; < HHMMSS.fraction > | fraction = 1*DIGIT | feature-tag = token | session-id = 8*( ALPHA / DIGIT / safe ) | message-header = field-name ":" [ field-value ] CRLF | field-name = token | field-value = *( field-content / LWS ) | field-content = <the OCTETs making up the field-value and | consisting of either *TEXT or combinations | of token, tspecials, and quoted-string> | 17.2.2 Message Syntax | RTSP-message = Request / Response ; RTSP/1.0 messages | Request = Request-Line ; Section 6.1 | *( general-header ; Section 5 | H. Schulzrinne et. al. [Page 106]
Internet Draft RTSP February 16, 2004 / request-header ; Section 6.2 | / entity-header ) ; Section 8.1 | CRLF | [message-body ] ; Section 4.3 | Response = Status-Line ; Section 7.1 | *(general-header ; Section 5 | / response-header ; Section 7.1.2 | / entity-header ) ; Section 8.1 | CRLF | [ message-body ] ; Section 4.3 | Request-Line = Method SP Request-URI SP RTSP-Version CRLF | Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF | Method = "DESCRIBE" ; Section 11.2 | / "GET_PARAMETER" ; Section 11.7 | / "OPTIONS" ; Section 11.1 | / "PAUSE" ; Section 11.5 | / "PLAY" ; Section 11.4 | / "PING" ; Section 11.10 | / "REDIRECT" ; Section 11.9 | / "SETUP" ; Section 11.3 | / "SET_PARAMETER" ; Section 11.8 | / "TEARDOWN" ; Section 11.6 | / extension-method | extension-method = token | Request-URI = "*" / absolute_URL | RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT | Status-Code = "100" ; Continue | / "200" ; OK | / "201" ; Created | / "250" ; Low on Storage Space | / "300" ; Multiple Choices | / "301" ; Moved Permanently | / "302" ; Moved Temporarily | / "303" ; See Other | H. Schulzrinne et. al. [Page 107]
Internet Draft RTSP February 16, 2004 / "304" ; Not Modified | / "305" ; Use Proxy | / "400" ; Bad Request | / "401" ; Unauthorized | / "402" ; Payment Required | / "403" ; Forbidden | / "404" ; Not Found | / "405" ; Method Not Allowed | / "406" ; Not Acceptable | / "407" ; Proxy Authentication Required | / "408" ; Request Time-out | / "410" ; Gone | / "411" ; Length Required | / "412" ; Precondition Failed | / "413" ; Request Entity Too Large | / "414" ; Request-URI Too Large | / "415" ; Unsupported Media Type | / "451" ; Parameter Not Understood | / "452" ; reserved | / "453" ; Not Enough Bandwidth | / "454" ; Session Not Found | / "455" ; Method Not Valid in This State | / "456" ; Header Field Not Valid for Resource | / "457" ; Invalid Range | / "458" ; Parameter Is Read-Only | / "459" ; Aggregate operation not allowed | / "460" ; Only aggregate operation allowed | / "461" ; Unsupported transport | / "462" ; Destination unreachable | / "500" ; Internal Server Error | / "501" ; Not Implemented | / "502" ; Bad Gateway | / "503" ; Service Unavailable | / "504" ; Gateway Time-out | / "505" ; RTSP Version not supported | / "551" ; Option not supported | / extension-code | extension-code = 3DIGIT | Reason-Phrase = *<TEXT, excluding CR, LF> | general-header = Cache-Control ; Section 14.9 | / Connection ; Section 14.10 | / CSeq ; Section 14.17 | H. Schulzrinne et. al. [Page 108]
Internet Draft RTSP February 16, 2004 / Date ; Section 14.18 | / Supported ; Section 14.38 | / Timestamp ; Section 14.39 | / Via ; Section 14.44 | / extension-header | request-header = Accept ; Section 14.1 and [H14.1] | / Accept-Encoding ; Section 14.2 and [H14.3] | / Accept-Language ; Section 14.3 and [H14.4] | / Authorization ; Section 14.6 and [H14.8] | / Bandwidth ; Section 14.7 | / Blocksize ; Section 14.8 | / From ; Section 14.20 | / If-Match ; Section 14.22 | / If-Modified-Since ; Section 14.23 and [H14.25] | / Proxy-Require ; Section 14.27 | / Range ; Section 14.29 | / Referer ; Section 14.30 | / Require ; Section 14.32 | / Scale ; Section 14.34 | / Session ; Section 14.37 | / Speed ; Section 14.35 | / Supported ; Section 14.38 | / Transport ; Section 14.40 | / User-Agent ; Section 14.42 | / extension-header | response-header = Accept-Ranges ; Section 14.4 | / Location ; Section 14.25 | / Proxy-Authenticate ; Section 14.26 | / Public ; Section 14.28 | / Range ; Section 14.29 | / Retry-After ; Section 14.31 | / RTP-Info ; Section 14.33 | / Scale ; Section 14.34 | / Session ; Section 14.37 | / Server ; Section 14.36 | / Speed ; Section 14.35 | / Transport ; Section 14.40 | / Unsupported ; Section 14.41 | / Vary ; Section 14.43 | / WWW-Authenticate ; Section 14.45 | / extension-header | H. Schulzrinne et. al. [Page 109]
Internet Draft RTSP February 16, 2004 entity-header = Allow ; Section 14.5 | / Content-Base ; Section 14.11 | / Content-Encoding ; Section 14.12 | / Content-Language ; Section 14.13 | / Content-Length ; Section 14.14 | / Content-Location ; Section 14.15 | / Content-Type ; Section 14.16 | / Expires ; Section 14.19 and [H14.21] | / Last-Modified ; Section 14.24 | / extension-header | extension-header = message-header | 17.2.3 Header Syntax | All header syntaxes not defined in this section are defined in | chapter 14 of the HTTP 1.1 specification [4]. | Accept-Ranges = "Accept-Ranges" ":" acceptable-ranges | acceptable-ranges = (range-unit *("," LWS range-unit)) | / "none" | range-unit = NPT / SMPTE / UTC / extension-format | extension-format = token | Bandwidth = "Bandwidth" ":" 1*DIGIT | Blocksize = "Blocksize" ":" 1*DIGIT | Cache-Control = "Cache-Control" ":" cache-directive | *("," LWS cache-directive) | cache-directive = cache-request-directive | / cache-response-directive | cache-request-directive = "no-cache" | / "max-stale" ["=" delta-seconds] | / "min-fresh" "=" delta-seconds | / "only-if-cached" | / cache-extension | cache-response-directive = "public" | / "private" | / "no-cache" | / "no-transform" | / "must-revalidate" | / "proxy-revalidate" | / "max-age" "=" delta-seconds | / cache-extension | cache-extension = token ["=" (token / quoted-string)] | delta-seconds = 1*DIGIT | H. Schulzrinne et. al. [Page 110]
Internet Draft RTSP February 16, 2004 Content-Base = "Content-Base" ":" absoluteURL | CSeq = "Cseq" ":" 1*DIGIT | Proxy-Require = "Proxy-Require" ":" feature-tag | *("," LWS feature-tag) | Public = "Public" ":" method *("," LWS method) | Range = "Range" ":" ranges-spec *("," LWS ranges-spec) | [ ";" "time" "=" utc-time ] | ranges-spec = npt-range / utc-range / smpte-range | Require = "Require" ":" feature-tag *("," LWS feature-tag) | RTP-Info = "RTP-Info" ":" rtsp-info-spec | *("," LWS rtsp-info-spec) | rtsp-info-spec = stream-url 1*parameter | stream-url = quoted-url / unquoted-url | unquoted-url = "url" "=" safe-url | quoted-url = "url" "=" DQUOTE needquote-url DQUOTE | safe-url = url | needquote-url = url //That contains ; or , | url = ( absoluteURL / relativeURL ) | parameter = ";" "seq" "=" 1*DIGIT | / ";" "rtptime" "=" 1*DIGIT | Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ] | Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ] | Server = "Server" ":" ( product / comment ) | *(SP (product / comment)) | Session = "Session" ":" session-id | [ ";" "timeout" "=" delta-seconds ] | Supported = "Supported" ":" [feature-tag *("," LWS feature-tag)] | Timestamp = "Timestamp" ":" *(DIGIT) ["." *(DIGIT)] [delay] | delay = *(DIGIT) [ "." *(DIGIT) ] | Transport = "Transport" ":" transport-spec | *("," LWS transport-spec) | transport-spec = transport-id *parameter | transport-id = transport-prot "/" profile ["/" lower-transport] | ; no LWS is allowed inside transport-id | transport-prot = "RTP" / token | profile = "AVP" / token | H. Schulzrinne et. al. [Page 111]
Internet Draft RTSP February 16, 2004 lower-transport = "TCP" / "UDP" / token | parameter = ";" ( "unicast" / "multicast" ) | / ";" "source" "=" host | / ";" "destination" [ "=" host ] | / ";" "interleaved" "=" channel [ "-" channel ] | / ";" "append" | / ";" "ttl" "=" ttl | / ";" "layers" "=" 1*DIGIT | / ";" "port" "=" port-spec | / ";" "client_port" "=" port-spec | / ";" "server_port" "=" port-spec | / ";" "ssrc" "=" ssrc | / ";" "client_ssrc" "=" ssrc | / ";" "mode" "=" mode-spec | / ";" "dest_addr" "=" addr-list | / ";" "src_addr" "=" addr-list | / ";" trn-param-ext | port-spec = port [ "-" port ] | trn-param-ext = par-name "=" trn-par-value | par-name = token | trn-par-value = *(unreserved / DQUOTE *TEXT DQUOTE) | ttl = 1*3(DIGIT) | ssrc = 8*8(HEX) | channel = 1*3(DIGIT) | mode-spec = ( DQUOTE mode *("," *SP mode) DQUOTE ) / mode | mode = "PLAY" / "RECORD" / token | addr-list = quoted-host-port *("/" quoted-host-port) | quoted-host-port = DQUOTE host [":" port] DQUOTE | Unsupported = "Unsupported" ":" feature-tag *("," feature-tag) | User-Agent = "User-Agent" ":" ( product / comment ) | 0*(SP (product / comment) | 18 Security Considerations Because of the similarity in syntax and usage between RTSP servers and HTTP servers, the security considerations outlined in [H15] apply. Specifically, please note the following: Authentication Mechanisms: RTSP and HTTP share common authentication schemes, and thus should follow the same prescriptions with regards to authentication . See chapter 15.1 of [16] for client authentication issues, and chapter 15.2 of [16] for issues regarding support for multiple H. Schulzrinne et. al. [Page 112]
Internet Draft RTSP February 16, 2004 authentication mechanisms. Also see [H15.6]. Abuse of Server Log Information: RTSP and HTTP servers will presumably have similar logging mechanisms, and thus should be equally guarded in protecting the contents of those logs, thus protecting the privacy of the users of the servers. See [H15.1.1] for HTTP server recommendations regarding server logs. Transfer of Sensitive Information: There is no reason to believe that information transferred via RTSP may be any less sensitive than that normally transmitted via HTTP. Therefore, all of the precautions regarding the protection of data privacy and user privacy apply to implementors of RTSP clients, servers, and proxies. See [H15.1.2] for further details. Attacks Based On File and Path Names: Though RTSP URLs are opaque handles that do not necessarily have file system semantics, it is anticipated that many implementations will translate portions of the request URLs directly to file system calls. In such cases, file systems SHOULD follow the precautions outlined in [H15.5], such as checking for ".." in path components. Personal Information: RTSP clients are often privy to the same information that HTTP clients are (user name, location, etc.) and thus should be equally. 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 one 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 H. Schulzrinne et. al. [Page 113]
Internet Draft RTSP February 16, 2004 to invalidate resources over which they have no authority. ([H15.4]) In addition to the recommendations in the current HTTP specification (RFC 2616 [4], as of this writing) and also of the previous RFC2068 [16], future HTTP specifications may provide additional guidance on security issues. The following are added considerations for RTSP implementations. Concentrated denial-of-service attack: The protocol offers the opportunity for a remote-controlled denial-of-service attack. The attacker may initiate traffic flows to one or more IP addresses by specifying them as the destination in SETUP requests. While the attacker's IP address may be known in this case, this is not always useful in prevention of more attacks or ascertaining the attackers identity. Thus, an RTSP server SHOULD only allow client-specified destinations for RTSP-initiated traffic flows if the server has verified the client's identity, either against a database of known users using RTSP authentication mechanisms (preferably digest authentication or stronger), or other secure means. Session hijacking: Since there is no or little relation between a transport layer connection and an RTSP session, it is possible for a malicious client to issue requests with random session identifiers which would affect unsuspecting clients. The server SHOULD use a large, random and non- sequential session identifier to minimize the possibility of this kind of attack. Authentication: Servers SHOULD implement both basic and digest [7] authentication. In environments requiring tighter security for the control messages, transport layer mechanisms such as TLS (RFC 2246 [26]) SHOULD be used. Stream issues: RTSP only provides for stream control. Stream delivery issues are not covered in this section, nor in the rest of this draft. RTSP implementations will most likely rely on other protocols such as RTP, IP multicast, RSVP and IGMP, and should address security considerations brought up in those and other applicable specifications. Persistently suspicious behavior: RTSP servers SHOULD return error code 403 (Forbidden) upon receiving a single instance of behavior which is deemed a security risk. RTSP servers H. Schulzrinne et. al. [Page 114]
Internet Draft RTSP February 16, 2004 SHOULD also be aware of attempts to probe the server for weaknesses and entry points and MAY arbitrarily disconnect and ignore further requests clients which are deemed to be in violation of local security policy. 19 IANA Considerations This section set up a number of registers for RTSP that should be maintained by IANA. For each registry there is a description on what it shall contain, what specification is needed when adding a entry with IANA, and finally the entries that this document needs to register. See also the section 1.6 "Extending RTSP". There is also an IANA registration of two SDP attributes. The sections describing how to register an item uses some of the requirements level described in RFC 2434 [17], namely " First Come, First Served", "Specification Required", and "Standards Action". A registration request to IANA MUST contain the following information: o A name of the item to register according to the rules specified by the intended registry. o Indication of who has change control over the feature (for example, IETF, ISO, ITU-T, other international standardization bodies, a consortium, a particular company or group of companies, or an individual); o A reference to a further description, if available, for example (in order of preference) an RFC, a published standard, a published paper, a patent filing, a technical report, documented source code or a computer manual; o For proprietary features, contact information (postal and email address); 19.1 Feature-tags 19.1.1 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 10 and 11.1. H. Schulzrinne et. al. [Page 115]
Internet Draft RTSP February 16, 2004 19.1.2 Registering New Feature-tags with IANA The registering of feature-tags is done on a first come, first served basis. The name of the feature MUST follow these rules: The name may be of any length, but SHOULD be no more than twenty characters long. The name MUST not contain any spaces, or control characters. The registration SHALL indicate if the feature tag applies to servers only, proxies only or both server and proxies. Any proprietary feature SHALL have as the first part of the name a vendor tag, which identifies the organization. 19.1.3 Registered entries The following feature-tags are in this specification defined and hereby registered. The change control belongs to the Authors and the IETF MMUSIC WG. play.basic: The minimal implementation for playback operations according to section D. Applies for both servers and proxies. play.scale: Support of scale operations for media playback. Applies only for servers. play.speed: Support of the speed functionality for playback. Applies only for servers 19.2 RTSP Methods 19.2.1 Description What a method is, is described in section 11. Extending the protocol with new methods allow for totally new functionality. 19.2.2 Registering New Methods with IANA A new method MUST be registered through an IETF standard track document. The reason is that new methods may radically change the protocols behavior and purpose. A specification for a new RTSP method MUST consist of the following items: o A method name which follows the BNF rules for methods. o A clear specification on what action and response a request H. Schulzrinne et. al. [Page 116]
Internet Draft RTSP February 16, 2004 with the method will result in. 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 registered headers that are allowed to use with the method in request or/and response. o Describe how the method relates to network proxies. 19.2.3 Registered Entries This specification, RFCXXXX, registers 10 methods: DESCRIBE, GET_PARAMETER, OPTIONS, PAUSE, PING, PLAY, REDIRECT, SETUP, SET_PARAMETER, and TEARDOWN. 19.3 RTSP Status Codes 19.3.1 Description A status code is the three digit numbers used to convey information in RTSP response messages, see 7. The number space is limited and care should be taken not to fill the space. 19.3.2 Registering New Status Codes with IANA A new status code can only be registered by an IETF standards track document. A specification for a new status code MUST specify the following: o The requested number. o A description what the status code means and the expected behavior of the sender and receiver of the code. 19.3.3 Registered Entries RFCXXX, registers the numbered status code defined in the BNF entry "Status-Code" except "extension-code" in section 17.2.2. 19.4 RTSP Headers 19.4.1 Description By specifying new headers a method(s) can be enhanced in many different ways. An unknown header will be ignored by the receiving entity. If the new header is vital for a certain functionality, a feature-tag for the functionality can be created and demanded to be H. Schulzrinne et. al. [Page 117]
Internet Draft RTSP February 16, 2004 used by the counter-part with the inclusion of a Require header carrying the feature-tag. 19.4.2 Registering New Headers with IANA A public available specification is required to register a header. The specification SHOULD be a standards document, preferable an IETF RFC. The specification MUST contain the following information: o The name of the header. o A BNF 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 shall be handled by proxies. o A description of the purpose of the header. 19.4.3 Registered entries All headers specified in section 14 in RFCXXXX are to be registered. Furthermore the following RTSP headers defined in other specifications are registered: o x-wap-profile defined in [35]. o x-wap-profile-diff defined in [35]. o x-wap-profile-warning defined in [35]. o x-predecbufsize defined in [35]. o x-initpredecbufperiod defined in [35]. o x-initpostdecbufperiod defined in [35]. Note: The use of "X-" is NOT RECOMMENDED but the above headers in the register list was defined prior to the clarification. 19.5 Transport Header registries The transport header contains a number of parameters which have H. Schulzrinne et. al. [Page 118]
Internet Draft RTSP February 16, 2004 possibilities for future extensions. Therefore registries for these must be defined. 19.5.1 Transport Protocols A registry for the parameter transport-protocol shall be defined with the following rules: o Registering requires public available standards specification. o A contact person or organization with address and email. o A value definition that are following the BNF token definition. o A describing text that explains how the registered value are used in RTSP. This specification registers 1 value: o Use of the RTP [15] protocol for media transport. The usage is explained in RFC XXXX, appendix B.1. 19.5.2 Profile A registry for the parameter profile shall be defined with the following rules: o Registering requires public available standards specification. o A contact person or organization with address and email. o A value definition that are following the BNF token definition. o A definition of which Transport protocol(s) that this profile is valid for. o A describing text that explains how the registered value are used in RTSP. This specification registers 1 value: o The "RTP profile for audio and video conferences with minimal control" [3] MUST only be used when the transport specification's transport-protocol is "RTP". 19.5.3 Lower Transport H. Schulzrinne et. al. [Page 119]
Internet Draft RTSP February 16, 2004 A registry for the parameter lower-transport shall be defined with the following rules: o Registering requires public available standards specification. o A contact person or organization with address and email. o A value definition that are following the BNF token definition. o A text describing how the registered value are used in RTSP. This specification registers 2 values: | o Indicates the use of the "User datagram protocol" [8] for media transport. o Indicates the use Transmission control protocol [9] for media transport. 19.5.4 Transport modes A registry for the transport parameter mode shall be defined with the following rules: o Registering requires a IETF standard tracks document. o A contact person or organization with address and email. o A value definition that are following the BNF token definition. o A describing text that explains how the registered value are used in RTSP. This specification registers 2 values: | o See RFC XXXX. o See RFC XXXX. 19.6 Cache Directive Extensions There exist a number of cache directives which can be sent in the Cache-Control header. A registry for this cache directives shall be defined with the following rules: o Registering requires a IETF standard tracks document. H. Schulzrinne et. al. [Page 120]
Internet Draft RTSP February 16, 2004 o A registration shall name a contact person. o Name of the directive and a definition of the value, if any. o Specification if it is an request or response directive. o A describing text that explains how the cache directive is used for RTSP controlled media streams. This specification registers the following values: | 19.7 SDP attributes This specification defines two SDP [2] attributes that it is requested that IANA register. SDP Attribute ("att-field"): | Attribute name: range | Long form: Media Range Attribute | Type of name: att-field | Type of attribute: Media and session level | Subject to charset: No | Purpose: RFC XXXX | Reference: RFC XXXX | Values: See ABNF definition. | Attribute name: control | Long form: RTSP control URL | Type of name: att-field | Type of attribute: Media and session level | Subject to charset: No | Purpose: RFC XXXX | Reference: RFC XXXX | Values: Absolute or Relative URLs. | Attribute name: etag | Long form: Entity Tag | Type of name: att-field | Type of attribute: Media and session level | Subject to charset: No | Purpose: RFC XXXX | Reference: RFC XXXX | Values: See ABNF definition | H. Schulzrinne et. al. [Page 121]
Internet Draft RTSP February 16, 2004 A RTSP Protocol State Machine The RTSP session state machine describe the behavior of the protocol from RTSP session initialization through RTSP session termination. 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. This state machine is one possible representation that helps explain how the protocol works and when different requests are allowed. We find it a reasonable representation but does not mandate it, and other representations can be created. A.1 States The state machine contains three states, described below. For each state there exist a table which shows which requests and events that is allowed and if they will result in a state change. Init: Initial state no session exist. Ready: Session is ready to start playing. Play: Session is playing, i.e. sending media stream data in the direction S -> C. A.2 State variables This representation of the state machine needs more than its state to work. A small number of variables are also needed and is explained below. NRM: The number of media streams part of this session. RP: Resume point, the point in the presentation time line at which a request to continue will resume from. A time format for the variable is not mandated. A.3 Abbreviations To make the state tables more compact a number of abbreviations are used, which are explained below. IFI: IF Implemented. H. Schulzrinne et. al. [Page 122]
Internet Draft RTSP February 16, 2004 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. A.4 State Tables This section contains a table for each state. The table contains all the requests and events that this state is allowed to act on. The events which is method names are, unless noted, requests with the given method in the direction client to server (C -> S). In some cases there exist one or more requisite. The response column tells what type of response actions should be performed. Possible actions that is requested for an event includes: response codes, e.g. 200, headers that MUST be included in the response, setting of state variables, or setting of other session related parameters. The new state column tells which state the state machine shall change to. The response to valid request meeting the requisites is normally a | 2xx (SUCCESS) unless other noted in the response column. The | exceptions shall be given a response according to the response | column. If the request does not meet the requisite, is erroneous or | some other type of error occur the appropriate response code MUST be | sent. If the response code is a 4xx the session state is unchanged. A | response code of 3rr will result in that the session is ended and its | state is changed to Init. A response code of 304 results in no state | change. However there exist restrictions to when a 3xx response may | be used. A 5xx response SHALL not result in any change of the session | state, except if the error is not possible to recover from. A | unrecoverable error SHALL result the ending of the session. As it in | the general case can't be determined if it was a unrecoverable error | or not the client will be required to test. In the case that the next | request after a 5xx is responded with 454 (Session Not Found) the | client shall assume that the session has been ended. The server will timeout the session after the period of time specified in the SETUP response, if no activity from the client is detected. Therefore there exist a timeout event for all states except Init. In the case that NRM=1 the presentation URL is equal to the media H. Schulzrinne et. al. [Page 123]
Internet Draft RTSP February 16, 2004 URL. For NRM>1 the presentation URL MUST be other than any of the medias that are part of the session. This applies to all states. Event Prerequisite Response ______________________________________________________________ DESCRIBE Needs REDIRECT 3rr Redirect DESCRIBE 200, Session description OPTIONS Session ID 200, Reset session timeout timer OPTIONS 200 SET_PARAMETER Valid parameter 200, change value of parameter GET_PARAMETER Valid parameter 200, return value of parameter Table 11: None state-machine changing events The methods in Table 11 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. 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 12: State: Init The initial state of the state machine, see Table 12 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 URL and/or server by a 3rr response. In the Ready state, see Table 13, 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 URL) already is part of the session change the transport parameters. TEARDOWN is depending on both the request URL and the H. Schulzrinne et. al. [Page 124]
Internet Draft RTSP February 16, 2004 Action Requisite New State Response ______________________________________________________________________ SETUP New URL Ready NRM+=1 SETUP Setten up URL Ready Change transport param. TEARDOWN Prs URL,NRM>1 Init No session hdr TEARDOWN md URL,NRM=1 Init No Session hdr, NRM=0 TEARDOWN md URL,NRM>1 Ready Session hdr, NRM-=1 PLAY Prs URL, No range Play Play from RP PLAY Prs URL, Range Play according to range PAUSE Prs URL Ready Return PP S -> C:REDIRECT Range hdr Ready Set RedP S -> C:REDIRECT no range hdr Init Session is removed Timeout Init RedP reached Ready TEARDOWN of session Table 13: State: Ready number of media stream within the session. If the request URL is the presentations URL the whole session is torn down. If a media URL is used in the TEARDOWN request and more than one media exist in the session, the session will remain and a session header MUST be returned in the response. If only a single media stream remains in the session when performing a TEARDOWN with a media URL the session is removed. The number of media streams remaining after tearing down a media stream determines the new state. The Play state table, see Table 14, is the largest. The table contains an number of request that has presentation URL as a prerequisite on the request URL, this is due to the exclusion of non-aggregated stream control in sessions with more than one media stream. To avoid inconsistencies between the client and server, automatic state transitions are avoided. This can be seen at for example "End of media" event when all media has finished playing, the session still remain in Play state. An explicit PAUSE request must be sent to change the state to Ready. It may appear that there exist two automatic transitions in "RedP reached" and "PP reached", however they are requested and acknowledge before they take place. The time at which the transition will happen is known by looking at the range header. If the client sends request close in time to these transitions it must be prepared for getting error message as the state may or may not have changed. B Media Transport Alternatives H. Schulzrinne et. al. [Page 125]
Internet Draft RTSP February 16, 2004 Action Requisite New State Response ________________________________________________________________________ PAUSE PrsURL,No range Ready Set RP to present point PAUSE PrsURL,Range>now Play Set RP & PP to given point PAUSE PrsURL,Range<now Ready Set RP to Range Hdr. PP reached Ready RP = PP End of media All media Play No action, RP = Invalid End of media >1 Media plays Play No action End of range Play Set RP = End of range SETUP New URL Play 455 SETUP Setuped URL Play 455 SETUP Setuped URL, IFI Play Change transport param. TEARDOWN Prs URL,NRM>1 Init No session hdr TEARDOWN md URL,NRM=1 Init No Session hdr, NRM=0 TEARDOWN md URL Play 455 S -> C:REDIRECT Range hdr Play Set RedP S -> C:REDIRECT no range hdr Init Session is removed RedP reached Play TEARDOWN of session Timeout Init Stop Media playout Table 14: State: Play This chapter defines how certain combinations of protocols, profiles and lower transports are used. This includes the usage of the Transport header's general source and destination parameters "src_addr" and "dest_addr". B.1 RTP This section defines the interaction of RTSP with respect to the RTP protocol [15]. It also defines any necessary media transport signalling with regards to RTP. The available RTP profiles and lower layer transports are described below along with rules on signalling the available combinations. B.1.1 AVP The usage of the "RTP Profile for Audio and Video Conferences with Minimal Control" [3] when using RTP for media transport over different lower layer transport protocols are defined below in regards to RTSP. On such case is defined within this document, the use of embedded (interleaved) binary data as defined in section 12. The usage of this method is indicated by include the "interleaved" parameter. H. Schulzrinne et. al. [Page 126]
Internet Draft RTSP February 16, 2004 When using embedded binary data the "src_addr" and "dest_addresses" SHALL NOT be used. This addressing and multiplexing is used as defined with use of channel numbers and the interleaved parameter. B.1.2 AVP/UDP This part descibes sending of RTP [15] over lower transport layer UDP [8] according to the profile "RTP Profile for Audio and Video Conferences with Minimal Control" defined in RFC 3551 [3]. This profiles requires that 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. Embedded (interleaved) data when RTSP messages is transported over UDP SHOULD NOT be performed. The RTP/UDP and RTCP/UDP flows can be established in two ways using the Transport header's parameters. The way provided in RFC 2326 was to use the necessary parameters from the set of "source", "destination", "client_port", and "server_port". This has the advantage of being compatible with all RTP capable RTSP servers and clients. However this method does not provide a possibility to specify non-continues port ranges for RTP and RTCP. The other way is to use the parameters "src_addr", and "dest_addr". This method provides total flexibility in specifying address and port number for each transport flow. However the disadvantage is that it is not supported by non-updated clients, i.e. clients not supporting the "play.basic" feature-tag. When using the "source", "destination", "client_port", and "server_port" the packets are be addressed in the following way for media playback: o RTP/UDP packet from the server to the client SHALL be sent to the address specified in the "destination" parameter and first even port number given in client_port range. If there is only a single port number given that MUST be given. o The server SHOULD send its RTP/UDP packets from the address specified in "source" parameter and from the first even port number specified in "server_port" parameter. o If there is specified a range in "client_port" parameter that contains at least two port numbers, the RTCP/UDP packets from server to client SHALL be sent to address specified in the "destination" parameter and first odd port number part of the range specified in the client_port parameter. o The Server SHOULD send its RTCP/UDP packets from the address | H. Schulzrinne et. al. [Page 127]
Internet Draft RTSP February 16, 2004 specified in "source" parameter and from the first odd port | number greater than the RTP port number specified in | "server_port" parameter. | o RTCP/UDP packets from the client to the server SHALL be sent | to the address specified in the "source" parameter and first | odd port number greater than the RTP port number given in | client_port range. | o The client SHOULD send its RTCP/UDP packets from the address | specified in "destination" parameter and from the first odd | port number specified in "server_port" parameter. | The usage of "src_addr" and "dest_addr" parameters to specify the address and port numbers are done in the following way for media playback, i.e. Mode=PLAY: o The "src_addr" and "dest_addr" parameters MUST contain either 1 or 2 address and port pairs. o Each address and port pair MUST contain both and address and a port number. o The first address and port pair given in either of the parameters applies to the RTP stream. The second address and port pair if present applies to the RTCP stream. o The RTP/UDP packets from the server to the client SHALL be sent to the address and port given by first address and port pair of the "dest_addr" parameter. o The RTCP/UDP packets from the server to the client SHALL be sent to the address and port given by the second address and port pair of the "dest_addr" parameter. If no second pair is given RTCP SHALL NOT be sent. o The RTCP/UDP packets from the client to the server SHALL be sent to the address and port given by the second address and port pair of the "dest_addr" parameter. If no second pair is given RTCP SHALL NOT be sent. o RTP and RTCP Packets SHOULD be sent from the corresponding receiver port, i.e. RTCP packets from server should be sent from the "src_addr" parameters second address port pair. B.1.3 AVP/TCP H. Schulzrinne et. al. [Page 128]
Internet Draft RTSP February 16, 2004 Note that this combination is not yet defined using sperate TCP connections. However the use of embedded (interleaved) binary data transported on the RTSP connection is possible as specified in section 12. When using this declared combination of interleaved binary data the RTSP messages MUST be transported over TCP. A possible future for this profile would be to define the use of a combination of the two drafts "Connection-Oriented Media Transport in SDP" [36] and "Framing RTP and RTCP Packets over Connection-Oriented Transport" [37]. However as this work is not finished, this functionality is unspecified. B.1.4 Handling NPT Jumps in the RTP Media Layer | RTSP allows media clients to control selected, non-contiguous | sections of media presentations, rendering those streams with an RTP | media layer[15]. Such control allows jumps to be created in NPT | timeline of the RTSP session. For example, jumps in NPT can be caused | by multiple ranges in the range specifier of a PLAY request or | through a "seek" opertaion on an RTSP session which involves a PLAY, | PAUSE, PLAY scenario where a new NPT is set for the session. The | media layer rendering the RTP stream should not be affected by jumps | in NPT. Thus, both RTP sequence numbers and RTP timestamps MUST be | continuous and monotonic across jumps of NPT. | We cannot assume that the RTSP client can communicate with | the RTP media agent, as the two may be independent | processes. If the RTP timestamp shows the same gap as the | NPT, the media agent will assume that there is a pause in | the presentation. If the jump in NPT is large enough, the | RTP timestamp may roll over and the media agent may believe | later packets to be duplicates of packets just played out. | As an example, assume a clock frequency of 8000 Hz, a packetization | interval of 100 ms and an initial sequence number and timestamp of | zero. | C->S: PLAY rtsp://xyz/fizzle RTSP/1.0 | CSeq: 4 | Session: abcdefg | Range: npt=10-15; | S->C: RTSP/1.0 200 OK | CSeq: 4 | Session: abcdefg | H. Schulzrinne et. al. [Page 129]
Internet Draft RTSP February 16, 2004 Range: npt=10-15 | RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=0; | rtptime=0 | The ensuing RTP data stream is depicted below: | S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s | S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s | . . . | S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s | Immediately after the end of the play range, the client follows up | with a request to PLAY from a new NPT. | C->S: PAUSE rtsp://xyz/fizzle RTSP/1.0 | CSeq: 5 | Session: abcdefg | S->C: RTSP/1.0 200 OK | CSeq: 5 | Session: abcdefg | Range: npt=15-15 | C->S: PLAY rtsp://xyz/fizzle RTSP/1.0 | CSeq: 6 | Session: abcdefg | Range: npt=18-20; | S->C: RTSP/1.0 200 OK | CSeq: 6 | Session: abcdefg | Range: npt=18-20 | RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=50; | rtptime=40000 | | The ensuing RTP data stream is depicted below: | S->C: RTP packet - seq = 50, rtptime = 40000, NPT time = 18s | S->C: RTP packet - seq = 51, rtptime = 40800, NPT time = 18.1s | . . . | H. Schulzrinne et. al. [Page 130]
Internet Draft RTSP February 16, 2004 S->C: RTP packet - seq = 69, rtptime = 55200, NPT time = 19.9s | First we play NPT 10 through 15, then 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 40,000 through 55,200. While there is a | gap in the NPT, there is no gap in the sequence number or timestamp | space of the RTP data stream. | B.1.5 Handling RTP Timestamps after PAUSE | During a PAUSE / PLAY interaction in a 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 thesubsequent PLAY request was received. The | duration includes all latency incurred and processing time required | to complete the request. | The RTP RFC [15] 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 [15], the RTP timestamp space | must increase continously with real time. While this is not optimal | for stored media, it is required for RTP and RTCP to function as | intended. Using a continous 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://xyz/fizzle RTSP/1.0 | CSeq: 4 | Session: abcdefg | Range: npt=10-15; | S->C: RTSP/1.0 200 OK | CSeq: 4 | H. Schulzrinne et. al. [Page 131]
Internet Draft RTSP February 16, 2004 Session: abcdefg | Range: npt=10-15 | RTP-Info: url= rtsp://xyz/fizzle/audiotrack;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: | C->S: PAUSE rtsp://xyz/fizzle RTSP/1.0 | CSeq: 5 | Session: abdcdefg | S->C: RTSP/1.0 200 OK | CSeq: 5 | Session: abcdefg | Range: npt=10.4-15 | 20 seconds elapse and then the client sends a PLAY request. In | addtion the server requires 15 ms to process the request: | C->S: PLAY rtsp://xyz/fizzle RTSP/1.0 | CSeq: 6 | Session: abcdefg | S->C: RTSP/1.0 200 OK | CSeq: 6 | Session: abcdefg | Range: npt=10.4-15 | RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=4; | rtptime=164400 | H. Schulzrinne et. al. [Page 132]
Internet Draft RTSP February 16, 2004 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, we play NPT 10 through 10.3, when a PAUSE is received by the | server. After 20 seconds a PLAY is recieved by the server which take | 15ms to process. The duration of time for which the session was | paused is reflected in the RTP timestamp of the RTP packets sent | after this PLAY request. | A client can use the RTSP range header and RTP-Info header to map NPT | time of a presentation with the RTP timestamp. | Note: In RFC 2326 [1], this matter was not clearly defined and was | misunderstood commonly. Therefore, clients SHOULD expect servers to | break the continuity of the RTP timestamp space in various arbitrary | manners after a PAUSE request. In these cases, it is RECOMMENDED that | clients accept the RTP stream after the pause with appropriate | mappings provided by the RTP-Info and Range headers. | B.1.6 RTSP / RTP Integration | For certain datatypes, tight integration between the RTSP layer and | the RTP layer will be necessary. This by no means precludes the above | restrictions. Combined RTSP/RTP media clients should use the RTP-Info | field to determine whether incoming RTP packets were sent before or | after a seek or before or after a PAUSE. | B.1.7 Scaling with RTP | For scaling (see Section 14.34), RTP timestamps should correspond to | the playback timing. For example, when playing video recorded at 30 | frames/second at a scale of two and speed (Section 14.35) 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. | B.1.8 Maintaining NPT synchronization with RTP timesatmps | The client can maintain a correct display of NPT by noting the RTP | timestamp value of the first packet arriving after repositioning. | The sequence parameter of the RTP-Info (Section 14.33) header | provides the first sequence number of the next segment. | H. Schulzrinne et. al. [Page 133]
Internet Draft RTSP February 16, 2004 B.1.9 Continuous Audio | For continuous audio, the server SHOULD set the RTP marker bit at the | beginning of serving a new PLAY request. This allows the client to | perform playout delay adaptation. | B.1.10 Multiple Sources in an RTP Session | Note that more than one SSRC MAY be sent in the media stream. | However, without further extensions RTSP can't synchronize more than | the single one indicated in the Transport header. In these cases RTCP | needs to be used for synchronization. | B.1.11 Usage of SSRCs and the RTCP BYE Message During a RTSP Session | The RTCP BYE message indicates the end of a RTP session as well as | the end of use of a given SSRC. Therefore, a client or server SHALL | NOT send a RTCP BYE message until it has finished using a SSRC. A | server SHOULD keep using a SSRC until the RTP session is terminated. | Prologing the use of a SSRC allows the established synchronization | context associated with that SSRC to be used to sychronize subsequent | PLAY requests even if the PLAY response is late. Additionally, | changing the server side SSRC will prevent the server from | synchronizing the new SSRC within RTSP as it is connected to the one | declared in the ssrc parameter in the Transport header. | B.2 Future Additions It is the intention that any future protocol or profile regarding both for media delivery and lower transport should be easy to add to RTSP. This chapter provides the necessary steps that needs to be meet. The following things needs to be considered when adding a new protocol of profile for use with RTSP: o The protocol or profile needs to define a name tag representing it. This tag is required to be a ABNF "token" to be possible to use in the Transport header specification. o The useful combinations of protocol/profile/lower-layer needs to be defined and for each combination declare the necessary parameters to use in the Transport header. o For new media protocols the interaction with RTSP needs to be addressed. One important factor will be the media synchronization. H. Schulzrinne et. al. [Page 134]
Internet Draft RTSP February 16, 2004 See the IANA section ( 19) on how to register the necessary attributes. C Use of SDP for RTSP Session Descriptions The Session Description Protocol (SDP, RFC 2327 [2]) may be used to describe streams or presentations in RTSP. This description is typically returned in reply to a DESCRIBE request on a URL from a server to a client, or received via HTTP from a server to a client. This appendix describes how an SDP file determines the operation of an RTSP session. SDP as is provides no mechanism by which a client can distinguish, without human guidance, between several media streams to be rendered simultaneously and a set of alternatives (e.g., two audio streams spoken in different languages). However the SDP extension "Grouping of Media Lines in the Session Description Protocol (SDP)" [38] may provide such functionality depending on need. Also future grouping semantics may in the future be developed. C.1 Definitions The terms "session-level", "media-level" and other key/attribute names and values used in this appendix are to be used as defined in SDP (RFC 2327 [2]): C.1.1 Control URL The "a=control:" attribute is used to convey the control URL. This attribute is used both for the session and media descriptions. If used for individual media, it indicates the URL to be used for controlling that particular media stream. If found at the session level, the attribute indicates the URL for aggregate control (presentation URL). The session level URL SHALL be different from any media level URL. The presence of a session level control attribute SHALL be interpreted as support for aggregated control. The control attribute SHALL be present on media level unless the presentation only contains a single media stream, in which case the attribute MAY only be present on the session level. control-attribute = "a=" "control" ":" url Example: a=control:rtsp://example.com/foo H. Schulzrinne et. al. [Page 135]
Internet Draft RTSP February 16, 2004 This attribute MAY contain either relative and absolute URLs, following the rules and conventions set out in RFC 2396 [12]. Implementations SHALL look for a base URL in the following order: 1. the RTSP Content-Base field; 2. the RTSP Content-Location field; 3. the RTSP request URL. If this attribute contains only an asterisk (*), then the URL SHALL be treated as if it were an empty embedded URL, and thus inherit the entire base URL. For SDP retrieved from a container file, there are certain things to consider. Lets say that the container file has the following URL: "rtsp://example.com/container.mp4". A media level relative URL needs to contain the file name container.mp4 in the beginning to be resolved correctly relative to the before given URL. An alternative if one does not desire to enter the container files name is to ensure that the base URL for the SDP document becomes: "rtsp://example.com/container.mp4/", i.e. an extra trailing slash. When using the URL resolution rules in RFC 2396 that will resolve correctly. However, please note that if the session level control URL is a *, that control URL will be equal to "rtsp://example.com/container.mp4/" and include the slash. C.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 a multicast, the port number indicated SHOULD be used for reception. The client MAY try to override the destination port, through the Transport header. The servers MAY allow this, the response will indicate if allowed or not. If the session is unicast, the port number is the ones RECOMMENDED by the server to the client, about which receiver ports to use; the client MUST still include its receiver ports in its SETUP request. The client MAY ignore this recommendation. If the server has no preference, it SHOULD set the port number value to zero. The "m=" lines contain information about what transport protocol, profile, and possibly lower-layer shall 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 section B, further combinations MAY be specified. H. Schulzrinne et. al. [Page 136]
Internet Draft RTSP February 16, 2004 TODO: Write something about the usage of Grouping of media line, RFC 3388 [38]. Example: m=audio 0 RTP/AVP 31 C.1.3 Payload Type(s) The payload type(s) are specified in the "m=" field. In case the | payload type is a static payload type from RFC 3551 [3], no other | information is required. In case it is a dynamic payload type, the | media attribute "rtpmap" is used to specify what the media is. The | "encoding name" within the "rtpmap" attribute may be one of those | specified in RFC 3551 (Sections 5 and 6), or an MIME type registered | with IANA, or an experimental encoding as specified in SDP (RFC 2327 | [2]). Codec-specific parameters are not specified in this field, but | rather in the "fmtp" attribute described below. C.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. C.1.5 Range of Presentation The "a=range" attribute defines the total time range of the stored session or an individual media. Non-seekable live sessions can be indicated, while the length of live sessions can be deduced from the "t" and "r" SDP parameters. The attribute is both a session and a media level attribute. For presentations that contains media streams of the same durations, the range attribute SHOULD only be used at session-level. In case of different length the range attribute MUST be given at media level for all media, and SHOULD NOT be given at session level. If the attribute is present at both media level and session level the media level values SHALL be used. The unit is specified first, followed by the value range. The units | and their values are as defined in Section 3.4, 3.5 and 3.6 and MAY | be extended with further formats. Any open ended range (start-), i.e. | H. Schulzrinne et. al. [Page 137]
Internet Draft RTSP February 16, 2004 without stop range, is of unspecified duration and SHALL be | considered as non-seekable content unless this property is | overridden. This attribute is defined in ABNF [5] as: a-range-def = "a" "=" "range" ":" ranges-specifier CRLF Examples: a=range:npt=0-34.4368 a=range:clock=19971113T2115-19971113T2203 Non seekable stream of unknown duration: a=range:npt=0- C.1.6 Time of Availability The "t=" field MUST contain suitable values for the start and stop times for both aggregate and non-aggregate stream control. The server SHOULD indicate a stop time value for which it guarantees the description to be valid, and a start time that is equal to or before the time at which the DESCRIBE request was received. It MAY also indicate start and stop times of 0, meaning that the session is always available. For sessions that are of live type, i.e. specific start time, unknown | stop time, likely unseekable, the "t=" and "r=" field SHOULD be used | to indicate the start time of the event. The stop time SHOULD be | given so that the live event will with high probability have ended at | that time, while still not be unnecessary long into the future. C.1.7 Connection Information In SDP, the "c=" field contains the destination address for the media stream. For a media destination address that is a IPv6 one, the SDP extension defined in [18] needs to be used. 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 must specify a destination address, the "c=" field SHOULD be set to a null value. For addresses of type "IP4", this value SHALL be "0.0.0.0", and for type "IP6", this value SHALL be "0:0:0:0:0:0:0:0", i.e. the unspecified address according to RFC 3513 [19]. H. Schulzrinne et. al. [Page 138]
Internet Draft RTSP February 16, 2004 C.1.8 Entity Tag The optional "a=etag" attribute identifies a version of the session description. It is opaque to the client. SETUP requests may include this identifier in the If-Match field (see section 14.22) 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. a-etag-def = "a" "=" "etag" ":" etag-string CRLF etag-string = 1*(%x01-09/%x0B-0C/%x0E-FF) Example: a=etag:158bb3e7c7fd62ce67f12b533f06b83a One could argue that the "o=" field provides identical functionality. However, it does so in a manner that would put constraints on servers that need to support multiple session description types other than SDP for the same piece of media content. C.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 URL specified via the "a=control:" attribute. Example: v=0 o=- 2890844256 2890842807 IN IP4 204.34.34.32 s=I came from a web page e=adm@example.com c=IN IP4 0.0.0.0 t=0 0 m=video 8002 RTP/AVP 31 a=control:rtsp://audio.com/movie.aud m=audio 8004 RTP/AVP 3 a=control:rtsp://video.com/movie.vid H. Schulzrinne et. al. [Page 139]
Internet Draft RTSP February 16, 2004 Note that the position of the control URL in the description implies that the client establishes separate RTSP control sessions to the servers audio.com and video.com It is recommended that an SDP file contains the complete media initialization information even if it is delivered to the media client through non-RTSP means. This is necessary as there is no mechanism to indicate that the client should request more detailed media stream information via DESCRIBE. C.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 URLs, and a session-level "a=control:" attribute which is used as the request URL for aggregate control. If the media-level URL is relative, it is resolved to absolute URLs according to Section C.1.1 above. Example: C->M: DESCRIBE rtsp://example.com/movie RTSP/1.0 CSeq: 1 M->C: RTSP/1.0 200 OK CSeq: 1 Date: 23 Jan 1997 15:35:06 GMT Content-Type: application/sdp Content-Base: rtsp://example.com/movie/ Content-Length: 164 v=0 o=- 2890844256 2890842807 IN IP4 204.34.34.32 s=I contain i=<more info> e=adm@example.com c=IN IP4 0.0.0.0 t=0 0 a=control:* m=video 8002 RTP/AVP 31 a=control:trackID=1 m=audio 8004 RTP/AVP 3 a=control:trackID=2 H. Schulzrinne et. al. [Page 140]
Internet Draft RTSP February 16, 2004 In this example, the client is required to establish a single RTSP session to the server, and uses the URLs rtsp://example.com/movie/trackID=1 and rtsp://example.com/movie/trackID=2 to set up the video and audio streams, respectively. The URL rtsp://example.com/movie/ , which is resolved from the "*", controls the whole presentation (movie). A client is not required to issues SETUP requests for all streams within an aggregate object. Servers should allow the client to ask for only a subset of the streams. C.4 RTSP external SDP delivery There are some considerations that needs to be made when the session description is delivered to client outside of RTSP, for example in HTTP or email. First of all the SDP needs to contain absolute URLs, relative will in most cases not work as the delivery will not correctly forward the base URL. And as SDP might be temporarily stored on file system before being loaded into a RTSP capable client, thus if possible to transport the base URL it still would need to be merged into the file. The writing of the SDP session availability information, i.e. "t=" and "r=", needs to be carefully considered. When the SDP is fetched by the DESCRIBE method it is with very high probability that the it is valid. However the same are much less certain for SDPs distributed using other methods. Therefore the publisher of the SDP should take care to follow the recommendations about availability in the SDP specification [2]. D Minimal RTSP implementation D.1 Client A client implementation MUST be able to do the following : o Generate the following requests: SETUP, TEARDOWN, PLAY. o Include the following headers in requests: CSeq, Connection, Session, Transport. o Parse and understand the following headers in responses: CSeq, Connection, Session, Transport, Content-Language, Content-Encoding, Content-Length, Content-Type. o Understand the class of each error code received and notify H. Schulzrinne et. al. [Page 141]
Internet Draft RTSP February 16, 2004 the end-user, if one is present, of error codes in classes 4xx and 5xx. The notification requirement may be relaxed if the end-user explicitly does not want it for one or all status codes. o Expect and respond to asynchronous requests from the server, such as REDIRECT. This does not necessarily mean that it should implement the REDIRECT method, merely that it MUST respond positively or negatively to any request received from the server. Though not required, the following are RECOMMENDED. o Implement RTP/AVP/UDP as a valid transport. o Inclusion of the User-Agent header. o Understand SDP session descriptions as defined in Appendix C o Accept media initialization formats (such as SDP) from standard input, command line, or other means appropriate to the operating environment to act as a "helper application" for other applications (such as web browsers). There may be RTSP applications different from those initially envisioned by the contributors to the RTSP specification for which the requirements above do not make sense. Therefore, the recommendations above serve only as guidelines instead of strict requirements. D.1.1 Basic Playback To support on-demand playback of media streams, the client MUST additionally be able to do the following: o generate the PAUSE request; o implement the REDIRECT method, and the Location header. D.1.2 Authentication-enabled In order to access media presentations from RTSP servers that require authentication, the client MUST additionally be able to do the following: o recognize the 401 (Unauthorized) status code; H. Schulzrinne et. al. [Page 142]
Internet Draft RTSP February 16, 2004 o parse and include the WWW-Authenticate header; o implement Basic Authentication and Digest Authentication. D.2 Server A minimal server implementation MUST be able to do the following: o Implement the following methods: SETUP, TEARDOWN, OPTIONS and PLAY. o Include the following headers in responses: Connection, Content-Length, Content-Type, Content-Language, Content- Encoding, Timestamp, Transport, Public, and Via, and Unsupported. RTP-compliant implementations MUST also implement the RTP-Info field. o Parse and respond appropriately to the following headers in requests: Connection, Proxy-Require, Session, Transport, and Require. Though not required, the following are highly recommended at the time of publication for practical interoperability with initial implementations and/or to be a "good citizen". o Implement RTP/AVP/UDP as a valid transport. | o Inclusion of the Server, Cache-Control Date, and Expires | headers. | o Implement the DESCRIBE method. | o Generate SDP session descriptions as defined in Appendix C | There may be RTSP applications different from those initially envisioned by the contributors to the RTSP specification for which the requirements above do not make sense. Therefore, the recommendations above serve only as guidelines instead of strict requirements. D.2.1 Basic Playback To support on-demand playback of media streams, the server MUST additionally be able to do the following: H. Schulzrinne et. al. [Page 143]
Internet Draft RTSP February 16, 2004 o Recognize the Range header, and return an error if seeking is not supported. o Implement the PAUSE method. In addition, in order to support commonly-accepted user interface features, the following are highly recommended for on-demand media servers: o Include and parse the Range header, with NPT units. Implementation of SMPTE units is recommended. o Include the length of the media presentation in the media initialization information. o Include mappings from data-specific timestamps to NPT. When RTP is used, the rtptime portion of the RTP-Info field may be used to map RTP timestamps to NPT. Client implementations may use the presence of length information to determine if the clip is seekable, and visably disable seeking features for clips for which the length information is unavailable. A common use of the presentation length is to implement a "slider bar" which serves as both a progress indicator and a timeline positioning tool. Mappings from RTP timestamps to NPT are necessary to ensure correct positioning of the slider bar. D.2.2 Authentication-enabled In order to correctly handle client authentication, the server MUST additionally be able to do the following: o Generate the 401 (Unauthorized) status code when authentication is required for the resource. o Parse and include the WWW-Authenticate header o Implement Basic Authentication and Digest Authentication E Open Issues 1. The proxy indications in the two header tables in chapter | 14 needs review. | H. Schulzrinne et. al. [Page 144]
Internet Draft RTSP February 16, 2004 2. Should the Allow header be possible to use optional in | request or responses besides the now specified 405 error | code? | 3. What text should be written on use of authorization in this | spec? | 4. How does entity tags relate to the If-Match header? The | usage in SDP must also be clarified related to syntax, etc. | 5. The minimal implementation must be looked over to see if it | complies with the specification. All must and should shall | be included in the minimal. Feature-tags for these needs to | be defined. Further feature-tags needs to be discussed. | 6. The list specifying which status codes are allowed on which | request methods seem to be in error and need review. | 7. The capability negotiation statement in section 1.5 does | not declare something that RTSP really supports fully. | 8. Should we define a 463 status code that informs the client | that the tried media stream destination was not allowed? | 9. There is need for clearer rule in regards to Transport | parameters changes in mid session. | 10. Can fragment be included in a request URL? Or should it as | for HTTP only be handled on the User-Agent side? | 11. Write the use case descriptions. | F Changes Compared to RFC 2326, the following issues has been addressed: | o The Transport header has been changed in the following way: | - The ABNF has been changed to define that extensions are | possible, and that unknown extension parameters shall be | ignored. | - To prevent backwards compatibility issues, any extension or | new parameter requires the usage of a feature tag combined | with the Require header. | - Syntax unclarities with the Mode parameter has been | resolved. | H. Schulzrinne et. al. [Page 145]
Internet Draft RTSP February 16, 2004 - Syntax error with ";" for multicast and unicast has been | resolved. | - Two new addressing parameters has been defined, src_addr and | dest_addr. These allow one to specify more than one complete | address and port tuple if needed. | - Support for IPv6 explicit addresses in all address fields | has been included. | - To handle URI definitions that contain ";" or "," a quoted | URL format has been introduced. | - Defined IANA registries for the transport headers | parameters, transport-protocol, profile, lower-transport, | and mode. | - The transport headers interleave parameter's text was made | more strict and use formal requirements levels. However no | change on how it is used was made. | - 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 defintion for SSRC has been clarified to require 8*8 | HEX. | - Updated the text on the transport headers "destination" and | "dest_addr" parameters regarding what security precautions | the server shall perform. | - The embedded (interleaved) binary data and its transport | parameter was clarified to being symmetric and that it is | the server that sets the channel numbers. | o The Range formats has been changed in the following way: | - The NPT format has been given a initial NPT identifier that | should be used, if missing NPT is assumed. | - All formats now support initial open ended formats of type | "npt=-10". | o RTSP message handling has been changed in the following way: | - It has been clarified that a 4xx message due to missing CSeq | header shall be returned without a CSeq header. | H. Schulzrinne et. al. [Page 146]
Internet Draft RTSP February 16, 2004 - Rules for how to handle timing out RTSP messages has been | added. | o The HTTP references has been updated to RFC 2616 and RFC 2617. | This has required that public, and content-base header are now | 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 chapter has completely been rewritten. It | includes now more details and are also more clear about the | model used. | o A IANA section has been included with contains a number of | registries and their rules. This will allow us to use IANA to | keep track of all RTSP extensions. | o Than transport of RTSP messages has seen the following | changes: | - The use of UDP has been deprecated due to missing interest | and to broken specification. | - The rules for how TCP connections shall be handled has been | clarified. Now it is made clear that servers should not | close the TCP connection unless they have been unused for | significant time. | - Strong recommendations why server and clients should use | persistent connections has also been added. | - There is now a requirement to handle non-persistent | connections as this provides great fault tolerance. | - Added wording on the usage of Connection:Close for RTSP. | o The following header related changes has been made: | - Accept-Ranges response header is added. This header | clarifies which range formats that can be used for a | resource. | - Clarified that Range header allows multiple ranges to allow | for creating editing list. | - Fixed the missing definitions for the Cache-Control header. | H. Schulzrinne et. al. [Page 147]
Internet Draft RTSP February 16, 2004 Also added to the syntax definition the missing delta- | seconds for max-stale and min-fresh parameters. | - Put requirement on CSeq header that the value is increased | by one for each new RTSP request. A Recommendation to start | at 1 has also been added. | - Added requirement that the Date header must be used for all | messages with entity. Also the Server should always include | it. | - Removed possibility to use Range header combined with Scale | header to indicate when it shall be activated, due to that | it can't work as defined. Also added rule that lack of scale | header in response indicate lack of support. Feature-tags | for scaled playback defined. | - The Speed header must now be responded to indicate support | and the actual speed going to be used. A feature-tag is | defined. Notes on congestion control was also added. | - The Supported header was borrowed from SIP to help with the | feature negotiation in RTSP. | - Clarified that the Timestamp header can be used to resolve | retransmission ambiguities. | - The Session header text has been expanded with a explanation | on keep alive and which methods to use. | - It has been clarified how the Range header formats shall be | used to indicate pause points. | - Clarified that RTP-Info URLs that are relative, uses the | request URL as base URL. Also clarified that the URL that | must be used is the SETUP. | - Added text that requires the Range to always be present in | PLAY responses. Clarified what should be sent in case of | live streams. | - The quoted URL format may also be used with the RTP-Info | header. Backwards compatibility issues exist with such | usage, thus it can only be used for implementations | following this specification. | - The Headers table has been updated using a structured | borrowed from SIP. This table carries much more information | H. Schulzrinne et. al. [Page 148]
Internet Draft RTSP February 16, 2004 and should provide a good overview of the available headers. | - It has been is clarified that any message with a message | body is required to have a Content-Length header. This was | the case in RFC 2326 but could be misinterpreted. | o The minimal implementation specification has been changed: | - Added Headers Timestamp, Via, Unsupported as required for a | minimal server implementation. | - Added headers Cache-Control, Expires and Date as recommended | headers to support by server implementations. | o The Protocol Syntax has been changed in the following way: | - All BNF definitions are updated according to the rules | defined in RFC 2234 [5] and has been gathered in a separate | chapter 17. | - The BNF for the User-Agent and Server headers has been | corrected so now only the description is in the HTTP | specification. | - The definition in the introduction of the RTSP session has | been changed. | - The protocol has been made fully IPv6 capable. Certain of | the functionality, like using explicit IPv6 addresses in | fields requires that the protocol support this updated | specification. | - Added a fragment part to the RTSP URL. This seem to be | indicated by the note below the definition however it was | not part of the BNF. | o The Status codes has been changed in the following way: | - The use of status code 303 "See Other" has been decapitated | as it does not make sense to use in RTSP. | - Added status code 350, 351 and updated usage of the other | redirect status codes, see chapter 13.3. | - When sending response 451 and 458 the response body should | contain the offending parameters. | - Clarification on when a 3rr redirect status code can be | H. Schulzrinne et. al. [Page 149]
Internet Draft RTSP February 16, 2004 received has been added. This includes receiving 3rr as a | result of request within a established session. This | provides clarification to a previous unspecified behavior. | - Removed the 250 (Low On Storage Space) status code as it | only is relevant to recording which is deprecated. | o The following functionality has been deprecated from the | protocol: | - The use of Queued Play. | - The use of PLAY method for keep-alive in play state. | - The RECORD and ANNOUNCE methods and all related | functionality. Some of the syntax has been removed. | - The possibility to use timed execution of methods with the | time parameter in the Range header. | - The description on how rtspu and rtsps works is not part of | the core specification and will require external | description. Only that they exist is defined here. | o Text specifying the special behavior of PLAY for live content. | o The following changes has been made in relation to methods: | - The OPTIONS method has been clarified on how to use the | Public and Allow headers. | - The RECORD and ANNOUNCE methods are removed as they are | lacking implementation and not considered necessary in the | core specification. Any work on these methods should be done | as a extension document to RTSP. | - Added text clarifying the usage of SET_PARAMETER for keep- | alive and usage without any body. | - Added a backwards compatibility resolution for how to handle | the new state machine without automatic state transition, | for example for returning to ready when finished playing. | o Wrote a new chapter about how to setup different media | transport alternatives and their profiles, and lower layer | protocols. This resulted that the appendix on RTP interaction | was moved there instead in the part describing RTP. The | chapter also includes guidelines what to think of when writing | H. Schulzrinne et. al. [Page 150]
Internet Draft RTSP February 16, 2004 usage guidelines for new protocols and profiles. | o Added a new chapter describing the available mechanisms to | determine if functionality is supported, called "Capability | Handling". Renamed option-tags to feature-tags. | o Added a contributors chapter with people who has contribute | actual text to the specification. | o Added a chapter 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. | Note that this list does not reflect minor changes in wording or | correction of typographical errors. | A word-by-word diff from RFC 2326 can be found at http://rtsp.org/ G Author Addresses Henning Schulzrinne Dept. of Computer Science Columbia University 1214 Amsterdam Avenue New York, NY 10027 USA electronic mail: schulzrinne@cs.columbia.edu Anup Rao Cisco USA electronic mail: anrao@cisco.com Robert Lanphier RealNetworks P.O. Box 91123 Seattle, WA 98111-9223 USA electronic mail: robla@real.com Magnus Westerlund Ericsson AB, EAB/TVA/A Torshamsgatan 23 SE-164 80 STOCKHOLM SWEDEN electronic mail: magnus.westerlund@ericsson.com H. Schulzrinne et. al. [Page 151]
Internet Draft RTSP February 16, 2004 Aravind Narasimhan Sun Microsystems, Inc. 101 Park Avenue, 3rd & 4th Floor New York, NY USA electronic mail: aravind.narasimhan@sun.com H Contributors The following people has made written contribution included in the specification: o Tom Marshall has contributed with text about the usage of 3rr status codes. o Thomas Zheng has contributed with text regarding the usage of the Range in PLAY responses. o Sean Sheedy has contributed with text regarding timing out RTSP messages. I Acknowledgements This draft is based on the functionality of the original RTSP draft submitted in October 1996. It also borrows format and descriptions from HTTP/1.1. This document has benefited greatly from the comments of all those participating in the MMUSIC-WG. In addition to those already mentioned, the following individuals have contributed to this specification: Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning, Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari, Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt, John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov, Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith, Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen Chesire, David Walker, and Geetha Srikantan. J Normative References [1] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr. H. Schulzrinne et. al. [Page 152]
Internet Draft RTSP February 16, 2004 1998. [2] M. Handley and V. Jacobson, "SDP: session description protocol," RFC 2327, Internet Engineering Task Force, Apr. 1998. [3] S. C. H. Schulzrinne, "RTP profile for audio and video conferences with minimal control," RFC 3551, Internet Engineering Task Force, July 2003. [4] e. a. R. Fielding, "Hypertext transfer protocol -- http/1.1," RFC 2616, Internet Engineering Task Force, June 1999. [5] D. Crocker and P. Overell, "Augmented BNF for syntax specifications: ABNF," RFC 2234, Internet Engineering Task Force, Nov. 1997. [6] S. Bradner, "Key words for use in RFCs to indicate requirement levels," RFC 2119, Internet Engineering Task Force, Mar. 1997. [7] J. F. et. al., "Http authentication: Basic and digest access authentication," RFC 2617, Internet Engineering Task Force, June 1999. [8] J. Postel, "User datagram protocol," RFC STD 6, 768, Internet Engineering Task Force, Aug. 1980. [9] J. Postel, "Transmission control protocol," RFC STD 7, 793, Internet Engineering Task Force, Sept. 1981. [10] R. Elz, "A compact representation of IPv6 addresses," RFC 1924, Internet Engineering Task Force, Apr. 1996. [11] L. M. R. Hinden, B. Carpenter, "Format for literal ipv6 addresses in url's," RFC 2732, Internet Engineering Task Force, Dec. 1999. [12] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource identifiers (URI): generic syntax," RFC 2396, Internet Engineering Task Force, Aug. 1998. [13] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC 2279, Internet Engineering Task Force, Jan. 1998. [14] B. Braden, "Requirements for internet hosts - application and support," RFC STD 3, 1123, Internet Engineering Task Force, Oct. 1989. [15] e. a. H. Schulzrinne, "RTP: a transport protocol for real-time H. Schulzrinne et. al. [Page 153]
Internet Draft RTSP February 16, 2004 applications," RFC 3550, Internet Engineering Task Force, July 2003. [16] R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T. Berners- Lee, "Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet Engineering Task Force, Jan. 1997. [17] H. Alvestrand and T. Narten, "Guidelines for writing an IANA considerations section in RFCs," RFC 2434, Internet Engineering Task Force, Oct. 1998. [18] A. B. R. S. Olson, G. Camarill, "Support for ipv6 in session description protocol (sdp)," RFC 3266, Internet Engineering Task Force, June 2002. [19] S. D. R. Hinden, "Internet protocol version 6 (ipv6) addressing architecture," RFC 3513, Internet Engineering Task Force, Apr. 2003. K Informative References [20] T. Z. M. Westerlund, "How to make real-time streaming protocol (rtsp) traverse network address translators (nat) and interact with firewalls.," internet draft, Internet Engineering Task Force, Feb. 2004. Work in progress. [21] A. Narasimhan, "Mute and unmute extension to rtsp," internet draft, Internet Engineering Task Force, Feb. 2002. Work in progress. [22] P. Gentric, "Rtsp stream switching," internet draft, Internet Engineering Task Force, Jan. 2004. Work in progress. [23] A. L. G. Srikantan, J. Murata, "Streaming relays," internet draft, Internet Engineering Task Force, Dec. 2003. Work in progress. [24] F. Yergeau, G. Nicol, G. Adams, and M. Duerst, "Internationalization of the hypertext markup language," RFC 2070, Internet Engineering Task Force, Jan. 1997. [25] ISO/IEC, "Information technology -- generic coding of moving pictures and associated audio informaiton -- part 6: extension for digital storage media and control," Draft International Standard ISO 13818-6, International Organization for Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995. [26] C. A. T. Dierks, "The tls protocol, version 1.0," RFC 2246, Internet Engineering Task Force, Jan. 1999. [27] B. Hinden and C. Partridge, "Version 2 of the reliable data protocol (RDP)," RFC 1151, Internet Engineering Task Force, Apr. H. Schulzrinne et. al. [Page 154]
Internet Draft RTSP February 16, 2004 1990. [28] H. Schulzrinne, "A comprehensive multimedia control architecture for the Internet," in Proc. International Workshop on Network and Operating System Support for Digital Audio and Video (NOSSDAV), (St. Louis, Missouri), May 1997. [29] International Telecommunication Union, "Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service," Recommendation H.323, Telecommunication Standardization Sector of ITU, Geneva, Switzerland, May 1996. [30] P. McMahon, "GSS-API authentication method for SOCKS version 5," RFC 1961, Internet Engineering Task Force, June 1996. [31] J. Miller, P. Resnick, and D. Singer, "Rating services and rating systems (and their machine readable descriptions)," Recommendation REC-PICS-services-961031, W3C (World Wide Web Consortium), Boston, Massachusetts, Oct. 1996. [32] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label distribution label syntax and communication protocols," Recommendation REC-PICS-labels-961031, W3C (World Wide Web Consortium), Boston, Massachusetts, Oct. 1996. [33] B. Braden, "T/TCP -- TCP extensions for transactions functional specification," RFC 1644, Internet Engineering Task Force, July 1994. [34] W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2. Reading, Massachusetts: Addison-Wesley, 1994. [35] Third Generation Partnership Project (3GPP), "Transparent end- to-end packet-switched streaming service (pss); protocols and codecs," Technical Specification 26.234, Third Generation Partnership Project (3GPP), Dec. 2002. [36] D. Yon, "Connection-oriented media transport in sdp," internet draft, Internet Engineering Task Force, Mar. 2003. Work in progress. [37] J. Lazzaro, "Framing rtp and rtcp packets over connection- oriented transport," internet draft, Internet Engineering Task Force, Oct. 2003. Work in progress. [38] e. a. G. Camarillo, "Grouping of media lines in the session description protocol (sdp)," RFC 3388, Internet Engineering Task Force, Dec. 2002. H. Schulzrinne et. al. [Page 155]
Internet Draft RTSP February 16, 2004 IPR Notice The IETF takes no position regarding the validity or scope of any intellectual property or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; neither does it represent that it has made any effort to identify any such rights. Information on the IETF's procedures with respect to rights in standards-track and standards-related documentation can be found in BCP-11. Copies of claims of rights made available for publication and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementors or users of this specification can be obtained from the IETF Secretariat. The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights which may cover technology that may be required to practice this standard. Please address the information to the IETF Executive Director. Full Copyright Statement Copyright (C) The Internet Society (2004). All Rights Reserved. This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implmentation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English. The limited permissions granted above are perpetual and will not be revoked by the Internet Society or its successors or assigns. This document and the information contained herein is provided on an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION H. Schulzrinne et. al. [Page 156]
Internet Draft RTSP February 16, 2004 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. H. Schulzrinne et. al. [Page 157]
