Internet Engineering Task Force MMUSIC WG Internet Draft H. Schulzrinne Columbia U. A. Rao Cisco R. Lanphier RealNetworks M. Westerlund Ericsson A. Narasimhan Sun draft-ietf-mmusic-rfc2326bis-04.txt June 30, 2003 Expires: December, 2003 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 mate- rial 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 Pro- posed Standard. The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time H. Schulzrinne et. al. [Page 1] Internet Draft RTSP June 30, 2003 properties. RTSP provides an extensible framework to enable con- trolled, 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 1889). H. Schulzrinne et. al. [Page 2] Internet Draft RTSP June 30, 2003 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 [21]. 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. The goal of the current work on RTSP is to progress it to draft stan- dard status. Whether this is possible without first publishing RTSP as a proposed standard depends on the changes necessary to make the protocol work. The list of changes in chapter F indicates the issues that have already been addressed. The currently open issues are listed in chapter E. There is also a list of reported bugs available at "http://rtsp- spec.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. Take special notice of the following: + The example section 15 has not yet been revised since the changes to protocol have not been completed. + The BNF chapter 16 has not been compiled completely. + 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 spec- ification has been reduced and split. The content of this draft is the core specification of the protocol. It contains the gen- eral 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. Two proposals exist at this time: + NAT and FW traversal mechanisms for RTSP are described in a docu- ment called "How to make Real-Time Streaming Protocol (RTSP) tra- verse Network Address Translators (NAT) and interact with Fire- walls." [33]. + The MUTE extension [34] contains a proposal on adding functional- ity to mute and unmute media streams in an aggregated media ses- sion without affecting the time-line of the playback. H. Schulzrinne et. al. [Page 3] Internet Draft RTSP June 30, 2003 There have also been discussions about the following extensions to RTSP: + Transport security for RTSP messages (rtsps). + Unreliable transport of RTSP messages (rtspu). + The Record functionality. + 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. + A RTSP MIB. However, so far, they have not become concrete proposals. 1.2 Purpose The Real-Time Streaming Protocol (RTSP) establishes and controls sin- gle 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 associ- ate groups of media streams and their states. A RTSP session is nor- mally not tied to a transport-level connection such as a TCP connec- tion. 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 specifica- tion. The mechanisms of RTSP's operation over UDP were left out of this spec. because they were poorly defined in RFC 2336 [21] and the tradeoff in size and complexity of this spec. for a small gain in a targeted problem space was not deemed justifi- able. The set of streams to be controlled is defined by a presentation description. This memorandum does not define a format for the H. Schulzrinne et. al. [Page 4] Internet Draft RTSP June 30, 2003 presentation description. The streams controlled by RTSP may use RTP [1] 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 [26] so that extension mechanisms to HTTP can in most cases also be added to RTSP. However, RTSP differs in a number of important aspects from HTTP: + RTSP introduces a number of new methods and has a different pro- tocol identifier. + RTSP has the notion of a session built into the protocol. + A RTSP server needs to maintain state by default in almost all cases, as opposed to the stateless nature of HTTP. + Both a RTSP server and client can issue requests. + Data is usually carried out-of-band by a different protocol. Session descriptions returned in a DESCRIBE response (see Section 11.2) and interleaving of RTP with RTSP over TCP are exceptions to this rule (see Section 11.11). + RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, consistent with current HTML internationalization efforts [3]. + The Request-URI always contains the absolute URI. Because of backward compatibility with a historical blunder, HTTP/1.1 [26] 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 request a pre- sentation description via 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 | H. Schulzrinne et. al. [Page 5] Internet Draft RTSP June 30, 2003 into the presentation. This mode is useful for distributed | teaching applications. Several parties in the conference may | take turns "pushing the remote control buttons". Addition of media to an existing presentation: Particularly for live presentations, it is useful if the server can tell the client about additional media becoming available. RTSP requests may be handled by proxies, tunnels and caches as in HTTP/1.1 [26]. 1.3 Requirements 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 [4]. 1.4 Terminology Some of the terminology has been adopted from HTTP/1.1 [26]. 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 URI: The URI used in a RTSP request to refer to and control an aggregated session. It normally, but not always, corresponds to the presentation URI specified in the session description. See Section 11.3 for more information. Conference: a multiparty, multimedia presentation, where "multi" implies greater than or equal to one. Client: The client requests media service from the media server. Connection: A transport layer virtual circuit established between two programs for the purpose of communication. Container file: A file which may contain multiple media streams which often 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 H. Schulzrinne et. al. [Page 6] Internet Draft RTSP June 30, 2003 relationship that existed at the source. The most common exam- ples of continuous media are audio and motion video. Continu- ous 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. 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 ini- tialization 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 presenta- | tion 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 dif- ferent 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 applica- tion 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([5]). Message: The basic unit of RTSP communication, consisting of a structured sequence of octets matching the syntax defined in Section 16 and transmitted via a connection or a connection- less protocol. Non-Aggregated Control: Control of a single media stream. Only possible in RTSP sessions with a single media. H. Schulzrinne et. al. [Page 7] Internet Draft RTSP June 30, 2003 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 presenta- tion, such as the set of encodings, network addresses and information about the content. Other IETF protocols such as SDP (RFC 2327 [24]) use the term "session" for a live presen- tation. The presentation description may take several differ- ent formats, including but not limited to the session descrip- tion 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 indi- cated explicitly. 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 success- ful 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 exis- tence 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. 1.5 Protocol Properties RTSP has the following properties: H. Schulzrinne et. al. [Page 8] Internet Draft RTSP June 30, 2003 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 trans- port level (TLS, RFC 2246 [27]) or within the protocol itself. All HTTP authentication mechanisms such as basic (RFC 2616 [26]) and digest authentication (RFC 2069 [6]) are directly applicable. Transport-independent: RTSP does not preclude the use of an unreli- able datagram protocol (UDP) (RFC 768 [7]), a reliable data- gram protocol (RDP, RFC 1151, not widely used [8]) or a reli- able stream protocol such as TCP (RFC 793 [9]) as it imple- ments application-level reliability. The use of a connection- less datagram protocol such as UDP or RDP requires additional definition that may be provided as extensions to the core RTSP specification. Multi-server capable: Each media stream within a presentation can reside on a different server. The client automatically estab- lishes 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 con- trol is divorced from inviting a media server to a conference. In particular, SIP [10] or H.323 [28] 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 presenta- tion description must contain at least one RTSP URI. Proxy and firewall friendly: The protocol should be readily handled by both application and transport-layer (SOCKS [11]) fire- walls. A firewall may need to understand the SETUP method to open a "hole" for the UDP 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 [12,13]) for associating labels with content. However, RTSP H. Schulzrinne et. al. [Page 9] Internet Draft RTSP June 30, 2003 does not just add methods to HTTP since the controlling con- tinuous media requires server state in most cases. 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 possi- ble. The protocol would not address how several clients nego- tiate 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: + A server may not be capable of seeking (absolute positioning) if it is to support live events only. + 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 13. 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 [26], where the methods described in [H19.5] are not likely to be supported across all servers. H. Schulzrinne et. al. [Page 10] Internet Draft RTSP June 30, 2003 RTSP can be extended in three ways, listed here in order of the mag- nitude of changes supported: + Existing methods can be extended with new parameters, 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 13.32). + 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 meth- ods it supports using the Public response header. + 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 avail- able when performing a request. For detailed explanation of this see chapter 10. 1.7 Overall Operation Each presentation and media stream may be identified by a RTSP URL. The overall presentation and the properties of the media the presen- tation 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 main- tains 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, lan- guage, 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 H. Schulzrinne et. al. [Page 11] Internet Draft RTSP June 30, 2003 identified by a RTSP URL, which points to the media server handling that particular media stream and names the stream stored on that server. Several media streams can be located on different servers; for example, audio and video streams can be split across servers for load sharing. The description also enumerates which transport methods the server is capable of. Besides the media parameters, the network destination address and port need to be determined. Several modes of operation can be distin- guished: Unicast: The media is transmitted to the source of the RTSP request, with the port number chosen by the client. Alterna- tively, the media is transmitted on the same reliable stream as RTSP. Multicast, server chooses address: The media server picks the mul- ticast 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 descrip- tion, established by means outside the scope of this specifi- cation. 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 con- trolled by RTSP requests issued sequentially on different TCP connec- tions. 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 follow- | ing 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. H. Schulzrinne et. al. [Page 12] Internet Draft RTSP June 30, 2003 PLAY: Starts data transmission on a stream allocated via SETUP. | PAUSE: Temporarily halts a stream without freeing server resources. REDIRECT: Indicates that the session should be moved to 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 13.37) 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 inter- act with HTTP in that the initial contact with streaming content is often to be made through a web page. The current protocol specifica- tion 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 stan- dalone 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) [24] 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 trans- port protocol. While RTSP works well with RTP, it is not tied to RTP. H. Schulzrinne et. al. [Page 13] Internet Draft RTSP June 30, 2003 2 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 [26]). All the mechanisms specified in this document are described in both prose and an augmented Backus-Naur form (BNF) similar to that used in [H2.1]. It is described in detail in RFC 2234 [14], with the differ- ence that this RTSP specification maintains the "#" notation for comma-separated lists from [H2.1]. In this draft, 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 understand- ing of why things are the way that they are in RTSP. b 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-spe- cific syntax and semantics for RTSP URLs. The RTSP URL is case sensi- tive. | rtsp_URL = ( "rtsp:" / "rtspu:" / "rtsps:" ) || "//" host [ ":" port ] [ abs_path [ "?" query ]] || host = As defined by RFC 2732 [30] || abs_path = As defined by RFC 2396 [22] || port = *DIGIT || query = As defined by RFC 2396 [22] || Note that fragment and query identifiers do not have a well- defined meaning at this time, with the interpretation left to the RTSP server. H. Schulzrinne et. al. [Page 14] Internet Draft RTSP June 30, 2003 The scheme rtsp requires that commands are issued via a reliable pro- | tocol (within the Internet, TCP), while the scheme rtspu identifies | an unreliable protocol (within the Internet, UDP). The scheme rtsps | identifies a reliable transport using secure transport, perhaps TLS | [27]. The rtspu and rtsps is not defined in this specification and if | for future extensions of the protocol. 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-URI for the resource is rtsp_URL. The use of IP addresses in URLs SHOULD be avoided whenever possible (see RFC 1924 [16]). 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 lit- eral IPv6 addresses in RTSP URLs using the host specification in RFC 2732 [30]. A presentation or a stream is identified by a textual media identi- fier, using the character set and escape conventions [H3.2] of URLs (RFC 2396 [22]). 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 individ- ual 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 H. Schulzrinne et. al. [Page 15] Internet Draft RTSP June 30, 2003 relationships in the presentation and the URLs for the indi- vidual streams. A presentation description may name a stream "a.mov" and the whole presentation "b.mov". 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 charac- | ters long to make guessing it more difficult. (See Section 17.) session-id = 8*( ALPHA / DIGIT / safe ) 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. smpte-range = smpte-type "=" smpte-range-spec smpte-range-spec = ( smpte-time "-" [ smpte-time ] ) / ( "-" smpte-time ) smpte-type = "smpte" / "smpte-30-drop" / "smpte-25" ; other timecodes may be added smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ] H. Schulzrinne et. al. [Page 16] Internet Draft RTSP June 30, 2003 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 rela- tive 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. The beginning of a presentation corresponds to 0.0 seconds. Negative values are not defined. The special constant now is defined as the current instant of a live event. It MAY only be used for live events, and SHALL NOT be used for on-demand 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." [5] npt-range = ["npt" "="] npt-range-spec ; 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 Examples: npt=123.45-125 npt=12:05:35.3- H. Schulzrinne et. al. [Page 17] Internet Draft RTSP June 30, 2003 npt=now- The syntax conforms to ISO 8601. The npt-sec notation is opti- mized 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. utc-range = "clock" "=" utc-range-spec utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time ) utc-time = utc-date "T" utc-time "Z" utc-date = 8DIGIT ; < YYYYMMDD > utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction > fraction = 1*DIGIT 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 new features in RTSP. These tags are used in in Require (Section 13.32), Proxy- Require (Section 13.27), Unsupported (Section 13.41), and Supported (Section 13.38) header fields. Syntax: feature-tag = token The creator of a new RTSP feature-tag should either prefix the fea- ture-tag with a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name for a feature whose inventor can be reached at H. Schulzrinne et. al. [Page 18] Internet Draft RTSP June 30, 2003 "foo.com"), or register the new feature-tag with the Internet Assigned Numbers Authority (IANA), see IANA Section 18. 4 RTSP Message RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding (RFC 2279 [18]). Lines are terminated by CRLF, but receivers should be prepared to also interpret CR and LF by them- selves 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 [18]) 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 See [H4.2]. 4.3 Message Body See [H4.3] 4.4 Message Length H. Schulzrinne et. al. [Page 19] Internet Draft RTSP June 30, 2003 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 13.14) 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 Con- tent-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 chun- ked 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: general-header = Cache-Control ; Section 13.9 / Connection ; Section 13.10 / CSeq ; Section 13.17 / Date ; Section 13.18 / Timestamp ; Section 13.39 / Via ; Section 13.44 6 Request A request message from a client to a server or vice versa includes, within the first line of that message, the method to be applied to the resource, the identifier of the resource, and the protocol ver- sion in use. H. Schulzrinne et. al. [Page 20] Internet Draft RTSP June 30, 2003 Request = Request-Line ; Section 6.1 *( general-header ; Section 5 / request-header ; Section 6.2 / entity-header ) ; Section 8.1 CRLF [ message-body ] ; Section 4.3 6.1 Request Line Request-Line = Method SP Request-URI SP RTSP-Version 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_URI RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT 6.2 Request Header Fields request-header = Accept ; Section 13.1 / Accept-Encoding ; Section 13.2 / Accept-Language ; Section 13.3 / Authorization ; Section 13.6 / Bandwidth ; Section 13.7 / Blocksize ; Section 13.8 / From ; Section 13.20 / If-Modified-Since ; Section 13.23 / Proxy-Require ; Section 13.27 / Range ; Section 13.29 H. Schulzrinne et. al. [Page 21] Internet Draft RTSP June 30, 2003 / Referer ; Section 13.30 / Require ; Section 13.32 / Scale ; Section 13.34 / Session ; Section 13.37 / Speed ; Section 13.35 / Supported ; Section 13.38 / Transport ; Section 13.40 / User-Agent ; Section 13.42 Note that in contrast to HTTP/1.1 [26], 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-URI 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 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 7 Response [H6] applies except that HTTP-Version is replaced by RTSP-Version. Also, RTSP defines additional status codes and does not define some HTTP codes. The valid response codes and the methods they can be used with are defined in Table 1. H. Schulzrinne et. al. [Page 22] Internet Draft RTSP June 30, 2003 After receiving and interpreting a request message, the recipient responds with an RTSP response message. 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 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. Status-Line = 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 12. The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata and the Reason-Phrase is intended for the human user. The client is not required to examine or display the Reason- Phrase. The first digit of the Status-Code defines the class of response. The last two digits do not have any categorization role. There are 5 values for the first digit: + 1xx: Informational - Request received, continuing process + 2xx: Success - The action was successfully received, understood, and accepted + 3rr: Redirection - Further action must be taken in order to com- plete the request + 4xx: Client Error - The request contains bad syntax or cannot be fulfilled H. Schulzrinne et. al. [Page 23] Internet Draft RTSP June 30, 2003 + 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 below. 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 [26] status codes and adds RTSP-specific status codes starting at x50 to avoid conflicts with newly defined HTTP status codes. 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 / "304" ; Not Modified / "305" ; Use Proxy / "350" ; Going Away / "351" ; Load Balancing / "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 H. Schulzrinne et. al. [Page 24] Internet Draft RTSP June 30, 2003 / "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 = * 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 unrecog- nized response MUST NOT be cached. For example, if an unrecognized status code of 431 is received by the client, it can safely assume that there was something wrong with its request and treat the response as if it had received a 400 status code. In such cases, user agents SHOULD present to the user the entity returned with the response, since that entity is likely to include human-readable information which will explain the unusual status. Code Reason Method -------------------------------------------------------- 100 Continue all -------------------------------------------------------- 200 OK all 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 -------------------------------------------------------- H. Schulzrinne et. al. [Page 25] Internet Draft RTSP June 30, 2003 Code Reason Method -------------------------------------------------------- 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-URI Too Long all 415 Unsupported Media Type all 451 Parameter Not Understood SET_PARAMETER 452 reserved n/a 453 Not Enough Bandwidth SETUP 454 Session Not Found all 455 Method Not Valid In This State all 456 Header Field Not Valid all 457 Invalid Range PLAY, PAUSE 458 Parameter Is Read-Only SET_PARAMETER 459 Aggregate Operation Not Allowed all 460 Only Aggregate Operation Allowed all 461 Unsupported Transport all 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 551 Option not support all Table 1: Status codes and their usage with RTSP methods 7.1.2 Response Header Fields The response-header fields allow the request recipient to pass addi- tional information about the response which cannot be placed in the Status-Line. These header fields give information about the server and about further access to the resource identified by the Request- URI. H. Schulzrinne et. al. [Page 26] Internet Draft RTSP June 30, 2003 response-header = Accept-Ranges ; Section 13.4 / Location ; Section 13.25 / Proxy-Authenticate ; Section 13.26 / Public ; Section 13.28 / Range ; Section 13.29 / Retry-After ; Section 13.31 / RTP-Info ; Section 13.33 / Scale ; Section 13.34 / Session ; Section 13.37 / Server ; Section 13.36 / Speed ; Section 13.35 / Transport ; Section 13.40 / Unsupported ; Section 13.41 / Vary ; Section 13.43 / WWW-Authenticate ; Section 13.45 Response-header field names can be extended reliably only in combina- tion with a change in the protocol version. However, new or experi- mental 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. 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. entity-header = Allow ; Section 13.5 H. Schulzrinne et. al. [Page 27] Internet Draft RTSP June 30, 2003 / Content-Base ; Section 13.11 / Content-Encoding ; Section 13.12 / Content-Language ; Section 13.13 / Content-Length ; Section 13.14 / Content-Location ; Section 13.15 / Content-Type ; Section 13.16 / Expires ; Section 13.19 / Last-Modified ; Section 13.24 / extension-header extension-header = message-header 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 a Content-Type header. 9 Connections RTSP requests can be transmitted in several different ways: + persistent transport connections used for several request- response transactions; + one connection per request/response transaction; + connectionless mode. The type of transport connection is defined by the RTSP URI (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 connec- tions, 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 H. Schulzrinne et. al. [Page 28] Internet Draft RTSP June 30, 2003 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 Requests are acknowledged by the receiver unless they are sent to a multicast group. 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) [15], 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 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 can- not typically take advantage of the application-layer retrans- mission since the transport stack will not deliver the appli- cation-layer retransmission before the first attempt has reached the receiver. If the packet loss is caused by conges- tion, multiple retransmissions at different layers will exac- erbate the congestion. If RTSP is used over a small-RTT LAN, standard procedures for opti- mizing initial TCP round trip estimates, such as those used in T/TCP (RFC 1644) [19], can be beneficial. The Timestamp header (Section 13.39) is used to avoid the retransmis- sion ambiguity problem [20] and obviates the need for Karn's algo- rithm. Each request carries a sequence number in the CSeq header (Section 13.17), which MUST be incremented by one for each distinct request transmitted. If a request is repeated because of lack of acknowledge- ment, the request MUST carry the original sequence number (i.e., the sequence number is not incremented). Systems implementing RTSP MUST support carrying RTSP over TCP and MAY support UDP. The default port for the RTSP server is 554 for both UDP and TCP. A number of RTSP packets destined for the same control end point may be packed into a single lower-layer PDU or encapsulated into a TCP stream. RTSP data MAY be interleaved with RTP and RTCP packets. H. Schulzrinne et. al. [Page 29] Internet Draft RTSP June 30, 2003 Unlike HTTP, an RTSP message MUST contain a Content-Length header field whenever that message contains a payload. Otherwise, an RTSP packet is terminated with an empty line immediately following the last message header. 9.3 The usage of connections TCP can be used for both persistent connections and for one message exchange per connection, as presented above. This section gives fur- ther 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 con- nection 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 server and client supporting non-per- sistent connection can survive a loss of a TCP connection, e.g. due to a NAT timeout. When the it is discovered that the TCP connection has been lost one sets up a new one. 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 MUST NOT initiate a close of a connec- | tion directly after responding to a TEARDOWN request for the whole | session. A server MUST 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 connec- tion. 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 REC- OMMENDED to use persistent connections. H. Schulzrinne et. al. [Page 30] Internet Draft RTSP June 30, 2003 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 noti- fying the client why,due to the lack of signalling channel. An exam- ple of such a case is when the server desires to send a REDIRECT request for a RTSP session to the client. If a service requires the use of persistent connection an feature-tag is specified for usage in the Require and Proxy-Require headers. con.persistent 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 deter- mine 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. 9.4 Use of IPv6 This specification has been updated so that it supports IPv6. How- ever this support was not present in RFC 2326 therefore some interop- erability issues exist. A RFC 2326 implementation can support IPv6 as long as no explicit IPv6 addresses are used within RTSP 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 too 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 H. Schulzrinne et. al. [Page 31] Internet Draft RTSP June 30, 2003 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 which declares all methods supported for the indicated resource. It is not necessary to use OPTIONS to discover support of a method, it is possible to simple try it. If the receiver of the request does not support the method it will respond with an error code indicating the the method are 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 repre- sents can vary significant. 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 dif- ferent headers can be used, each explained below: Supported: The supported header are used to determine the complete set of functionality that both client and server has. The intended usage is to determine before one needs to use a func- tionality that it is supported. If 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 includ- ing all supported feature-tags. The 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 H. Schulzrinne et. al. [Page 32] Internet Draft RTSP June 30, 2003 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 sit- uations as it knows which features that was not supported. 11 Method Definitions The method token indicates the method to be performed on the resource identified by the Request-URI 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 2. 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 2: 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 Notes on Table 2: PAUSE is recommended, but not required in that a 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 H. Schulzrinne et. al. [Page 33] Internet Draft RTSP June 30, 2003 SHOULD not try this method again for this server. 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 URI determines which scope the OPTIONS request has. By giving the URI of a certain media the capabilities regarding this media will be responded. By using the "*" URI the request regards the next hop only, while having a URL with only the host address regards the server without any media relevance. 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 pur- posefully 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. It may use the Accept header to specify the description formats that the client understands. The server responds with a description of the requested H. Schulzrinne et. al. [Page 34] Internet Draft RTSP June 30, 2003 resource. The DESCRIBE reply-response pair constitutes the media ini- tialization 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 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 informa- tion 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 parame- ters, 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 initial- ization for the set of streams that it describes, and discour- aging use of DESCRIBE for media indirection, we avoid looping H. Schulzrinne et. al. [Page 35] Internet Draft RTSP June 30, 2003 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: + via RTSP's DESCRIBE method; + via some other protocol (HTTP, email attachment, etc.); + via the command line or standard input (thus working as a browser helper application launched with an SDP file or other media ini- tialization 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 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 URI specifies the transport mechanism to be used for the streamed media. A client can 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). A server MAY allow a client to do SETUP while in playing state to add additional media streams. If not supported, the server SHALL respond with error 455 (Method Not Allowed In This State). If supported, the added media SHALL then start to play in sync with the already playing media. To be able to sync the media with the already playing streams the SETUP response MUST include a RTP-Info header with the timestamp value, and a Range header with the corresponding normal play time. To indicate support for this optional feature the feature-tag: "setup.playing" is defined. The Transport header specifies the transport parameters acceptable to the client for data transmission; the response will contain the transport parameters selected by the server. C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 CSeq: 302 Transport: RTP/AVP;unicast;client_port=4588-4589 H. Schulzrinne et. al. [Page 36] Internet Draft RTSP June 30, 2003 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 For the benefit of any intervening firewalls, a client must indicate the transport parameters even if it has no influence over these parameters, for example, where the server advertises a fixed multi- cast 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 ini- tialization. The server generates a session identifier in response to a SETUP request. If a SETUP request to a server includes a session identi- fier, the server MUST bundle this setup request into the existing session (aggregated session) or return error 459 (Aggregate Operation Not Allowed) (see Section 12.4.11). An Aggregate control URI MUST be used to control an aggregated session. This URI MUST be different from the stream control URIs of the individual media streams included in the aggregate. The Aggregate control URI SHOULD be specified by the session description since there is no general rule for deriving it from the various stream control URIs in the session. If an Aggre- gate control URI is not specified in the session description, a client MUST create an URI for aggregate control of the session. This URI MUST contain the servers host address and MUST contain the port, if applicable. Once an URI is used to refer to an aggregation for a given session, that URI MUST be used to refer to that aggregation for the duration of the session. If the contents of the aggregation change, then a different aggregate control URI SHOULD be used. While the session ID has enough information for aggregate con- trol of a session, the Aggregate control URI is still impor- tant for some methods such as SET_PARAMETER where the control URI enables the resource in question to be easily identified. The Aggregate control URI is also useful for proxies, enabling them to route the request to the appropriate server, and for logging, where it is useful to note the actual resource that a H. Schulzrinne et. al. [Page 37] Internet Draft RTSP June 30, 2003 request was operating on. Finally, presence of the Aggregate control URI allows for backwards compatibility with RFC 2326 [21]. 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 dis- cussion see chapter 13.37. Signs of liveness for a RTSP session are: + Any RTSP request from a client which includes a Session header with that session's ID. + 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. 11.4 PLAY The PLAY method tells the server to start sending data via the mecha- nism specified in SETUP. A client MUST NOT issue a PLAY request until any outstanding SETUP requests have been acknowledged as successful. 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 URI is for one of the media. The media in an aggregate SHALL be played in sync. If a client want individual control of the media it must use separate RTSP ses- sions for each media. The PLAY request positions 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 is reached. To allow for precise composi- tion multiple ranges MAY be specified. The range values are valid if all given ranges are part of any media. 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, imme- diately following, seconds 20 to 25, and finally seconds 30 through the end. H. Schulzrinne et. al. [Page 38] Internet Draft RTSP June 30, 2003 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 starts playing a stream from the beginning unless the stream has been paused. If a stream has been paused via PAUSE, stream delivery resumes at the pause point. The Range header MUST NOT contain a time parameter. The usage of time | has been deprecated. Server MUST include a "Range" header in any PLAY response. The | response MUST use the same format as the request's range header con- | tained. If no Range header was in the request, the NPT time format | SHOULD be used unless the client showed support for other formats. | For a session with live media streams the Range header MUST also be | given, containing a valid time indication. It is RECOMMENDED that | either "npt=now-" or a absolute time value (clock) for the corre- | sponding time is given, i.e. "clock=20030213T143205Z-". The UTC | clock format SHOULD only be used if client has shown support for it. 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. The unit of the range in the reply is the same as that in the request. If the medias part of an aggregate has different lengths the PLAY request and any Range SHALL be performed as long it is valid for 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 automati- cally paused, media deliver simply stops. A PAUSE request MUST be issued before another PLAY request can 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 PLAYING state. A client desiring to play the media from the begin- ning MUST send a PLAY request with a Range header pointing at the beginning, e.g. npt=0-. H. Schulzrinne et. al. [Page 39] Internet Draft RTSP June 30, 2003 The following example plays the whole presentation starting at SMPTE time code 0:10:20 until the end of the clip. The playback is to start at 15:36 on 23 Jan 1997. Note: The RTP-Info headers has been broken into several lines to fit the page. C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 CSeq: 833 Session: 12345678 Range: smpte=0:10:20-;time=19970123T153600Z 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-;time=19970123T153600Z RTP-Info:url=rtsp://example.com/twister.en; seq=14783;rtptime=2345962545 For playing back a recording of a live presentation, it may be desir- able 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 A media server only supporting playback MUST support the npt format and MAY support the clock and smpte formats. 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 H. Schulzrinne et. al. [Page 40] Internet Draft RTSP June 30, 2003 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 [21] is removed and multiple ranges can be used to achieve a similar performance. If a server receives a PLAY request while in the PLAY state, the server SHALL responde using the error code 455 (Method Not Valid In This State). This will signal the client that queued play are not sup- ported. The use of PLAY for keep-alive signaling, i.e. PLAY request without a range header, 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. When playing live media, indicated by the Accept-Ranges header the | session are in a live state. This live state will put some restric- | tions on the action available for a client. A PLAY request without a | Range header will start media deliver at the current point in the | live presentation, i.e. now. Any seeking in the media will be impos- | sible. The only allowed usage of the Range header is npt=now-, and | certain clock units. The usage of npt=now- is unnecessary as it has | the exact same meaning as a request without Range header. The clock | format can be used to specify start and stop times for media delivery | in a live session. 11.5 PAUSE The PAUSE request causes the stream delivery to be interrupted | (halted) temporarily. A PAUSE request MUST be done with the aggre- | gated control URI for aggregated sessions, resulting in all media | being halted, or the media URI for non-aggregated sessions. Any | attempt to do muting of a single media with an PAUSE request in an | aggregated session SHALL be responded with error 460 (Only Aggregate | Operation Allowed). After resuming playback, synchronization of the | tracks MUST be maintained. Any server resources are kept, though | servers MAY close the session and free resources after being paused | for the duration specified with the timeout parameter of the Session | header in the SETUP message. Example: C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 834 Session: 12345678 S->C: RTSP/1.0 200 OK H. Schulzrinne et. al. [Page 41] Internet Draft RTSP June 30, 2003 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 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: 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 H. Schulzrinne et. al. [Page 42] Internet Draft RTSP June 30, 2003 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 resumes from the pause point and | play until media end. The actual 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 simple included the Range header from the PAUSE response. 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 sec- ond, overlapping range. Regardless of when the PAUSE request arrives, it sets the pause point to 14. If the server has already sent data beyond the time specified in the the PAUSE request Range header, a PLAY without range would still resume at that 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. 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 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: H. Schulzrinne et. al. [Page 43] Internet Draft RTSP June 30, 2003 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- 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- 11.6 TEARDOWN The TEARDOWN request stops the stream delivery for the given URI, freeing the resources associated with it. If the URI is the aggre- gated control URI for this presentation, any RTSP session identifier associated with the session is no longer valid. The use of "*" as URI in TEARDOWN will also result in that the session is removed indepen- dent of the number of medias that was part of it. If the URI in the request was for a media within an aggregated session that media is removed from the aggregate. However the session and any other media stream yet not torn down remains, and any valid request, e.g. PLAY or SETUP, can be issued. As an optional feature a server MAY keep the session in case the last remaining media is torn down with a TEARDOWN request with an URI equal to the media URI. To Indicate what has been performed, a server that after any TEARDOWN request, still has a valid session MUST in the response return a session header. A server MAY choose to allow TEARDOWN of individual media while in PLAY state. When this is not allowed the response SHALL be 455 (Method Not Valid In This State). If a server implements TEARDOWN and SETUP in PLAY state it MUST signal this using the "setup.playing" feature-tag. H. Schulzrinne et. al. [Page 44] Internet Draft RTSP June 30, 2003 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 of a presentation or stream specified in the URI. If the Session header is present in a request, the value of a parameter MUST be retrieved in the sessions context. The content of the reply and response is left to the implementation. GET_PARAMETER with no entity body may be used to test client or server liveness ("ping"). 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 H. Schulzrinne et. al. [Page 45] Internet Draft RTSP June 30, 2003 be defined after further experimentation. 11.8 SET_PARAMETER This method requests to set the value of a parameter for a presenta- tion or stream specified by the URI. A request is RECOMMENDED to only contain a single parameter to allow the client to determine why a particular request failed. If the request contains several parameters, the server MUST only act on the request if all of the parameters can be set successfully. A server MUST allow a parameter to be set repeatedly to the same value, but it MAY disallow changing parameter values. If the receiver of the request does not understand or 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 H. Schulzrinne et. al. [Page 46] Internet Draft RTSP June 30, 2003 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. 11.9 REDIRECT A redirect request informs the client that it MUST connect to another | server location. The REDIRECT request MAY contain the header Loca- | tion, which indicates that the client should issue requests for that | URL. If the Location URL only contains a host address the client | shall connect to the given host, while using the path from the URL on | the current server. | 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 a REDIRECT request does not contain a Session header, it is next- | hop and applies to the control connection. The Location header SHOULD | only contain a host address. If there are proxies in the request | chain, they SHOULD do all of the following: (1) respond to the REDI- | RECT 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 ses- | sion 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. 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-. H. Schulzrinne et. al. [Page 47] Internet Draft RTSP June 30, 2003 If the client wants to continue to send or receive media for this resource, the client MUST issue a TEARDOWN request for the current session. A new session must be established 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://bigserver.com:8001 Range: npt=0- ;time=19960213T143205Z Session: uZ3ci0K+Ld-M 11.10 PING This method is a bi-directional mechanism for server or client live- ness 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 H. Schulzrinne et. al. [Page 48] Internet Draft RTSP June 30, 2003 S->C: RTSP/1.0 200 OK CSeq: 123 Session:12345678 11.11 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, fol- lowed by the length of the encapsulated binary data as a binary, two- byte integer in network byte order. The stream data follows immedi- ately afterwards, without a CRLF, but including the upper-layer pro- tocol headers. Each $ block contains exactly one upper-layer protocol data unit, e.g., one RTP packet. 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | "$" = 24 | Channel ID | Length in bytes | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : Length number of bytes of binary data : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The channel identifier is defined in the Transport header with the interleaved parameter(Section 13.40). 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 13.40. 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 direc- tions. H. Schulzrinne et. al. [Page 49] Internet Draft RTSP June 30, 2003 RTCP is needed for synchronization when two or more streams are interleaved in such a fashion. Also, this provides a con- venient 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://foo.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://foo.com/bar.file RTSP/1.0 CSeq: 3 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 3 Session: 12345678 Date: 05 Jun 1997 18:59:15 GMT RTP-Info: url=rtsp://foo.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} 12 Status Code Definitions Where applicable, HTTP status [H10] codes are reused. Status codes that have the same meaning are not repeated here. See Table 1 for a listing of which status codes may be returned by which requests. All error messages, 4xx and 5xx MAY return a body containing further information about the error. 12.1 Success 1xx 12.1.1 100 Continue See, [H10.1.1]. H. Schulzrinne et. al. [Page 50] Internet Draft RTSP June 30, 2003 12.2 Success 2xx 12.2.1 250 Low on Storage Space The server returns this warning after receiving a RECORD request that it may not be able to fulfill completely due to insufficient storage space. If possible, the server should use the Range header to indi- cate what time period it may still be able to record. Since other processes on the server may be consuming storage space simultane- ously, a client should take this only as an estimate. 12.3 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 com- ments are given for some to how they apply to RTSP. Within RTSP, redirection may be used for load balancing or redirect- ing stream requests to a server topologically closer to the client. Mechanisms to determine topological proximity are beyond the scope of this specification. 12.3.1 300 Multiple Choices 12.3.2 301 Moved Permanently The request resource are moved permanently and resides now at the URI given by the location header. The user client SHOULD redirect auto- matically to the given URI. This response MUST NOT contain a message- body. 12.3.3 302 Found The requested resource reside temporarily at the URI given by the Location header. The Location header MUST be included in the response. Is intended to be used for many types of temporary redi- rects, 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 URI. This response MUST NOT contain a message-body. 12.3.4 303 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. H. Schulzrinne et. al. [Page 51] Internet Draft RTSP June 30, 2003 12.3.5 304 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: + Date + ETag and/or Content-Location, if the header would have been sent in a 200 response to the same request. + Expires, Cache-Control, and/or Vary, if the field-value might differ from that sent in any previous response for the same vari- ant. 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. 12.3.6 305 Use Proxy See [H10.3.6]. 12.4 Client Error 4xx 12.4.1 400 Bad Request The request could not be understood by the server due to malformed syntax. The client SHOULD NOT repeat the request without modifica- tions [H10.4.1]. If the request does not have a CSeq header, the server MUST NOT include a CSeq in the response. 12.4.2 405 Method Not Allowed The method specified in the request is not allowed for the resource identified by the request URI. The response MUST include an Allow header containing a list of valid methods for the requested resource. This status code is also to be used if a request attempts to use a method not indicated during SETUP, e.g., if a RECORD request is issued even though the mode parameter in the Transport header only specified PLAY. H. Schulzrinne et. al. [Page 52] Internet Draft RTSP June 30, 2003 12.4.3 451 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). 12.4.4 452 reserved This error code was removed from RFC 2326 [21] and is obsolete. 12.4.5 453 Not Enough Bandwidth The request was refused because there was insufficient bandwidth. This may, for example, be the result of a resource reservation fail- ure. 12.4.6 454 Session Not Found The RTSP session identifier in the Session header is missing, invalid, or has timed out. 12.4.7 455 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. 12.4.8 456 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. 12.4.9 457 Invalid Range The Range value given is out of bounds, e.g., beyond the end of the presentation. 12.4.10 458 Parameter Is Read-Only The parameter to be set by SET_PARAMETER can be read but not modi- fied. When returning this error message the sender SHOULD return a entity body containing the offending parameter(s). H. Schulzrinne et. al. [Page 53] Internet Draft RTSP June 30, 2003 12.4.11 459 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. 12.4.12 460 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. 12.4.13 461 Unsupported Transport The Transport field did not contain a supported transport specifica- tion. 12.4.14 462 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 param- eter in the Transport field. 12.5 Server Error 5xx 12.5.1 551 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. 13 Header Field Definitions 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 syn- tax, meaning, and usage. Throughout this section, we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1 specification RFC 2616 [26]. Examples of each header field are given. Information about header fields in relation to methods and proxy pro- cessing is summarized in Table 4 and Table 5. The "where" column describes the request and response types in which the header field can be used. Values in this column are: H. Schulzrinne et. al. [Page 54] Internet Draft RTSP June 30, 2003 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 3: 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 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 3: H. Schulzrinne et. al. [Page 55] Internet Draft RTSP June 30, 2003 c: Conditional; requirements on the header field depend on the con- text 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 13.14, 13.16 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, and a Client/Server MAY ignore the header field if present in the request or response (The exception to this rule is the Require header field discussed in 13.32). A "mandatory" header field MUST be present in a request, and MUST be understood by the Client/Server receiving the request. A mandatory response header field MUST be present in the response, and the header field MUST be understood by the Client/Server processing the response. "Not appli- cable" 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 URI. If the URI contains a comma, or semicolon, the URI MUST be enclosed in dou- ble quotas ("). Any URI parameters are contained within these quotas. If the URI is not enclosed in double quotas, any semicolon- delimited parameters are header-parameters, not URI parameters. 13.1 Accept The Accept request-header field can be used to specify certain pre- sentation description content types which are acceptable for the response. H. Schulzrinne et. al. [Page 56] Internet Draft RTSP June 30, 2003 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 - 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 o o o o o o 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 - - - RTP-Info r - - o m - - H. Schulzrinne et. al. [Page 57] Internet Draft RTSP June 30, 2003 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 4: Overview of RTSP header fields related to methods DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN. The "level" parameter for presentation descriptions is prop- erly 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;level=2 H. Schulzrinne et. al. [Page 58] Internet Draft RTSP June 30, 2003 Header Where Proxy GPR SPR RDR PNG ----------------------------------------------------- Allow 405 - - - - 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 o o o o Last-Modified R r - - - - Last-Modified r r o - - - Location 3rr