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-05.txt October 27, 2003 Expires: April, 2004 Real Time Streaming Protocol (RTSP) STATUS OF THIS MEMO This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress". The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt To view the list Internet-Draft Shadow Directories, see http://www.ietf.org/shadow.html. Abstract This memorandum is a revision of RFC 2326, which is currently a Proposed Standard. The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP (RFC 3550). H. Schulzrinne et. al. [Page 1] Internet Draft RTSP October 27, 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 standard 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://rtspspec.sourceforge.net". These bugs should be taken into account when reading this specification. While a lot of these bugs are addressed, not all are yet accounted for in this specification. Input on the unresolved bugs and other issues can be sent via e-mail to the MMUSIC WG's mailing list mmusic@ietf.org and the authors. Take special notice of the following: o The example section 15 has not yet been revised since the changes to protocol have not been completed. o The BNF chapter 16 has not been compiled completely. o Not all of the contents of RFC 2326 are part of this draft. In an attempt to prevent the draft from exploding in size, the specification has been reduced and split. The content of this draft is the core specification of the protocol. It contains the general idea behind RTSP and the basic functionality necessary to establish an on-demand play-back session. It also contains the mechanisms for extending the protocol. Any other functionality will be published as extension documents. Two proposals exist at this time: o NAT and FW traversal mechanisms for RTSP are described in a document called "How to make Real-Time Streaming Protocol (RTSP) traverse Network Address Translators (NAT) and interact with Firewalls." [33]. o The MUTE extension [34] contains a proposal on adding functionality to mute and unmute media streams in an aggregated media session without affecting the time-line of H. Schulzrinne et. al. [Page 2] Internet Draft RTSP October 27, 2003 the playback. There have also been discussions about the following extensions to RTSP: o Transport security for RTSP messages (rtsps). o Unreliable transport of RTSP messages (rtspu). o The Record functionality. o A text body type with suitable syntax for basic parameters to be used in SET_PARAMETER, and GET_PARAMETER. Including IANA registry within the defined name space. o A RTSP MIB. However, so far, they have not become concrete proposals. 1.2 Purpose The Real-Time Streaming Protocol (RTSP) establishes and controls single or several time-synchronized streams of continuous media such as audio and video. Put simply, RTSP acts as a "network remote control" for multimedia servers. There is no notion of a RTSP connection in the protocol. Instead, a | RTSP server maintains a session labelled by an identifier to | associate groups of media streams and their states. A RTSP session is | not tied to a transport-level connection such as a TCP connection. | During a session, a client may open and close many reliable transport | connections to the server to issue RTSP requests for that session. This memorandum describes the use of RTSP over a reliable connection based transport level protocol such as TCP. RTSP may be implemented over an unreliable connectionless transport protocol such as UDP. While nothing in RTSP precludes this, additional definition of this problem area must be handled as an extension to the core specification. The mechanisms of RTSP's operation over UDP were left out of this spec. because they were poorly defined in RFC 2336 [21] and the tradeoff in size and complexity of this spec. for a small gain in a targeted problem space was not deemed justifiable. The set of streams to be controlled is defined by a presentation H. Schulzrinne et. al. [Page 3] Internet Draft RTSP October 27, 2003 description. This memorandum does not define a format for the 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: o RTSP introduces a number of new methods and has a different protocol identifier. o RTSP has the notion of a session built into the protocol. o A RTSP server needs to maintain state by default in almost all cases, as opposed to the stateless nature of HTTP. o Both a RTSP server and client can issue requests. o Data is usually carried out-of-band by a different protocol. Session descriptions returned in a DESCRIBE response (see Section 11.2) and interleaving of RTP with RTSP over TCP are exceptions to this rule (see Section 11.11). o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, consistent with current HTML internationalization efforts [3]. o 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 presentation 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 H. Schulzrinne et. al. [Page 4] Internet Draft RTSP October 27, 2003 be "invited" to join an existing conference to play back media into the presentation. This mode is useful for distributed teaching applications. Several parties in the conference may take turns "pushing the remote control buttons". 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 H. Schulzrinne et. al. [Page 5] Internet Draft RTSP October 27, 2003 the protocol. Continuous media: Data where there is a timing relationship between source and sink; that is, the sink must reproduce the timing relationship that existed at the source. The most common examples of continuous media are audio and motion video. Continuous media can be real-time (interactive), where there is a "tight" timing relationship between source and sink, or streaming (playback), where the relationship is less strict. Entity: The information transferred as the payload of a request or response. An entity consists of meta-information in the form of entity-header fields and content in the form of an entity-body, as described in Section 8. Feature-tag: A tag representing a certain set of functionality, i.e. a feature. Media initialization: Datatype/codec specific initialization. This includes such things as clockrates, color tables, etc. Any transport-independent information which is required by a client for playback of a media stream occurs in the media initialization phase of stream setup. Media parameter: Parameter specific to a media type that may be changed before or during stream playback. Media server: The server providing playback services for one or more media streams. Different media streams within a presentation may originate from different media servers. A media server may reside on the same or a different host as the web server the presentation is invoked from. Media server indirection: Redirection of a media client to a different media server. (Media) stream: A single media instance, e.g., an audio stream or a video stream as well as a single whiteboard or shared application group. When using RTP, a stream consists of all RTP and RTCP packets created by a source within an RTP session. This is equivalent to the definition of a DSM-CC stream([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 connectionless protocol. H. Schulzrinne et. al. [Page 6] Internet Draft RTSP October 27, 2003 Non-Aggregated Control: Control of a single media stream. Only possible in RTSP sessions with a single media. Participant: Member of a conference. A participant may be a machine, e.g., a playback server. Presentation: A set of one or more streams presented to the client as a complete media feed, using a presentation description as defined below. In most cases in the RTSP context, this implies aggregate control of those streams, but does not have to. Presentation description: A presentation description contains information about one or more media streams within a presentation, such as the set of encodings, network addresses and information about the content. Other IETF protocols such as SDP (RFC 2327 [24]) use the term "session" for a live presentation. The presentation description may take several different formats, including but not limited to the session description format SDP. Response: A RTSP response. If an HTTP response is meant, that is indicated explicitly. Request: A RTSP request. If an HTTP request is meant, that is indicated explicitly. RTSP session: A stateful abstraction upon which the main control methods of RTSP operate. A RTSP session is a server entity; it is created, maintained and destroyed by the server. It is established by a RTSP server upon the completion of a successful SETUP request (when 200 OK response is sent) and is labelled by a session identifier at that time. The session exists until timed out by the server or explicitly removed by a TEARDOWN request. A RTSP session is also a stateful entity; a RTSP server maintains an explicit session state machine (see Appendix A) where most state transitions are triggered by client requests. The existence of a session implies the existence of state about the session's media streams and their respective transport mechanisms. A given session can have zero or more media streams associated with it. A RTSP server uses the session to aggregate control over multiple media streams. Transport initialization: The negotiation of transport information (e.g., port numbers, transport protocols) between the client and the server. H. Schulzrinne et. al. [Page 7] Internet Draft RTSP October 27, 2003 1.5 Protocol Properties RTSP has the following properties: Extendable: New methods and parameters can be easily added to RTSP. Easy to parse: RTSP can be parsed by standard HTTP or MIME parsers. Secure: RTSP re-uses web security mechanisms, either at the transport level (TLS, RFC 2246 [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 unreliable datagram protocol (UDP) (RFC 768 [7]), a reliable datagram protocol (RDP, RFC 1151, not widely used [8]) or a reliable stream protocol such as TCP (RFC 793 [9]) as it implements application-level reliability. The use of a connectionless datagram protocol such as UDP or RDP requires additional definition that may be provided as extensions to the core RTSP specification. Multi-server capable: Each media stream within a presentation can reside on a different server. The client automatically establishes several concurrent control sessions with the different media servers. Media synchronization is performed at the transport level. Separation of stream control and conference initiation: Stream control is divorced from inviting a media server to a conference. In particular, SIP [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 presentation 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 H. Schulzrinne et. al. [Page 8] Internet Draft RTSP October 27, 2003 [11]) firewalls. 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 does not just add methods to HTTP since the controlling continuous 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 possible. The protocol would not address how several clients negotiate access; this is left to either a "social protocol" or some other floor control mechanism. 1.6 Extending RTSP Since not all media servers have the same functionality, media servers by necessity will support different sets of requests. For example: o A server may not be capable of seeking (absolute positioning) if it is to support live events only. o Some servers may not support setting stream parameters and thus not support GET_PARAMETER and SET_PARAMETER. H. Schulzrinne et. al. [Page 9] Internet Draft RTSP October 27, 2003 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. RTSP can be extended in three ways, listed here in order of the magnitude of changes supported: o Existing methods can be extended with new parameters, 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). o New methods can be added. If the recipient of the message does not understand the request, it responds with error code 501 (Not Implemented) and the sender should not attempt to use this method again. A client may also use the OPTIONS method to inquire about methods supported by the server. The server SHOULD list the methods it supports using the Public response header. o A new version of the protocol can be defined, allowing almost all aspects (except the position of the protocol version number) to change. The basic capability discovery mechanism can be used to both discover support for a certain feature and to ensure that a feature is available when performing a request. For detailed explanation of this see chapter 10. 1.7 Overall Operation Each presentation and media stream may be identified by a RTSP URL. The overall presentation and the properties of the media the presentation is made up of are defined by a presentation description file, the format of which is outside the scope of this specification. The presentation description file may be obtained by the client using HTTP or other means such as email and may not necessarily be stored on the media server. For the purposes of this specification, a presentation description is assumed to describe one or more presentations, each of which maintains a common time axis. For simplicity of exposition and H. Schulzrinne et. al. [Page 10] Internet Draft RTSP October 27, 2003 without loss of generality, it is assumed that the presentation description contains exactly one such presentation. A presentation may contain several media streams. The presentation description file contains a description of the media streams making up the presentation, including their encodings, language, and other parameters that enable the client to choose the most appropriate combination of media. In this presentation description, each media stream that is individually controllable by RTSP is identified by a RTSP URL, which points to the media server handling that particular media stream and names the stream stored on that server. Several media streams can be located on different servers; for example, audio and video streams can be split across servers for load sharing. The description also enumerates which transport methods the server is capable of. Besides the media parameters, the network destination address and port need to be determined. Several modes of operation can be distinguished: Unicast: The media is transmitted to the source of the RTSP request, with the port number chosen by the client. Alternatively, the media is transmitted on the same reliable stream as RTSP. Multicast, server chooses address: The media server picks the multicast address and port. This is the typical case for a live or near-media-on-demand transmission. Multicast, client chooses address: If the server is to participate in an existing multicast conference, the multicast address, port and encryption key are given by the conference description, established by means outside the scope of this specification. 1.8 RTSP States RTSP controls a stream which may be sent via a separate protocol, independent of the control channel. For example, RTSP control may occur on a TCP connection while the data flows via UDP. Thus, data delivery continues even if no RTSP requests are received by the media server. Also, during its lifetime, a single media stream may be controlled by RTSP requests issued sequentially on different TCP connections. Therefore, the server needs to maintain "session state" to be able to correlate RTSP requests with a stream. The state transitions are described in Appendix A. Many methods in RTSP do not contribute to state. However, the H. Schulzrinne et. al. [Page 11] Internet Draft RTSP October 27, 2003 following play a central role in defining the allocation and usage of stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, PING and TEARDOWN. SETUP: Causes the server to allocate resources for a stream and create a RTSP session. PLAY: Starts data transmission on a stream allocated via SETUP. PAUSE: Temporarily halts a stream without freeing server 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 interact with HTTP in that the initial contact with streaming content is often to be made through a web page. The current protocol specification aims to allow different hand-off points between a web server and the media server implementing RTSP. For example, the presentation description can be retrieved using HTTP or RTSP, which reduces roundtrips in web-browser-based scenarios, yet also allows for standalone RTSP servers and clients which do not rely on HTTP at all. However, RTSP differs fundamentally from HTTP in that most data delivery takes place out-of-band in a different protocol. HTTP is an asymmetric protocol where the client issues requests and the server responds. In RTSP, both the media client and media server can issue requests. RTSP requests are also stateful; they may set parameters and continue to control a media stream long after the request has been acknowledged. Re-using HTTP functionality has advantages in at least two areas, namely security and proxies. The requirements are very similar, so having the ability to adopt HTTP work on caches, proxies and authentication is valuable. H. Schulzrinne et. al. [Page 12] Internet Draft RTSP October 27, 2003 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 transport protocol. While RTSP works well with RTP, it is not tied to RTP. 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 difference 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 understanding of why things are the way that they are in RTSP. 3 Protocol Parameters 3.1 RTSP Version HTTP Specification Section [H3.1] applies, with HTTP replaced by RTSP. This specification defines version 1.0 of RTSP. 3.2 RTSP URL The "rtsp", "rtsps" and "rtspu" schemes are used to refer to network resources via the RTSP protocol. This section defines the scheme- specific syntax and semantics for RTSP URLs. The RTSP URL is case sensitive. 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] H. Schulzrinne et. al. [Page 13] Internet Draft RTSP October 27, 2003 Note that fragment and query identifiers do not have a well-defined meaning at this time, with the interpretation left to the RTSP server. The scheme rtsp requires that commands are issued via a reliable protocol (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 are for future extensions of the protocol to define. 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 literal IPv6 addresses in RTSP URLs using the host specification in RFC 2732 [30]. A presentation or a stream is identified by a textual media identifier, using the character set and escape conventions [H3.2] of URLs (RFC 2396 [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 individual stream within the presentation. Note that some request methods can only be applied to streams, not presentations and vice versa. For example, the RTSP URL: rtsp://media.example.com:554/twister/audiotrack identifies the audio stream within the presentation "twister", which can be controlled via RTSP requests issued over a TCP connection to port 554 of host media.example.com Also, the RTSP URL: rtsp://media.example.com:554/twister H. Schulzrinne et. al. [Page 14] Internet Draft RTSP October 27, 2003 identifies the presentation "twister", which may be composed of audio and video streams. This does not imply a standard way to reference streams in URLs. The presentation description defines the hierarchical relationships in the presentation and the URLs for the individual streams. A presentation description may name a stream "a.mov" and the whole presentation "b.mov". The path components of the RTSP URL are opaque to the client and do not imply any particular file system structure for the server. This decoupling also allows presentation descriptions to be used with non-RTSP media control protocols simply by replacing the scheme in the URL. 3.3 Session Identifiers Session identifiers are strings of any arbitrary length. A session identifier MUST be chosen randomly and MUST be at least eight characters long to make guessing it more difficult. (See Section 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 ) H. Schulzrinne et. al. [Page 15] Internet Draft RTSP October 27, 2003 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 ] ] Examples: smpte=10:12:33:20- smpte=10:07:33- smpte=10:07:00-10:07:33:05.01 smpte-25=10:07:00-10:07:33:05.01 3.5 Normal Play Time Normal play time (NPT) indicates the stream absolute position relative to the beginning of the presentation, not to be confused with the Network Time Protocol (NTP). The timestamp consists of a decimal fraction. The part left of the decimal may be expressed in either seconds or hours, minutes, and seconds. The part right of the decimal point measures fractions of a second. 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 H. Schulzrinne et. al. [Page 16] Internet Draft RTSP October 27, 2003 npt-ss = 1*2DIGIT ; 0-59 Examples: npt=123.45-125 npt=12:05:35.3- npt=now- The syntax conforms to ISO 8601. The npt-sec notation is optimized for automatic generation, the ntp-hhmmss notation for consumption by human readers. The "now" constant allows clients to request to receive the live feed rather than the stored or time-delayed version. This is needed since neither absolute time nor zero time are appropriate for this case. 3.6 Absolute Time Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT). Fractions of a second may be indicated. 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 features in RTSP. These tags are used in Require (Section 13.32), Proxy-Require (Section 13.27), Unsupported (Section 13.41), and Supported (Section H. Schulzrinne et. al. [Page 17] Internet Draft RTSP October 27, 2003 13.38) header fields. Syntax: feature-tag = token Feature tag needs to indicate if they apply to servers only, proxies | only, or both server and proxies. The creator of a new RTSP feature-tag should either prefix the feature-tag with a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name for a feature whose inventor can be reached at "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 themselves as line terminators. Text-based protocols make it easier to add optional parameters in a self-describing manner. Since the number of parameters and the frequency of commands is low, processing efficiency is not a concern. Text-based protocols, if done carefully, also allow easy implementation of research prototypes in scripting languages such as Tcl, Visual Basic and Perl. The 10646 character set avoids tricky character set switching, but is invisible to the application as long as US-ASCII is being used. This is also the encoding used for RTCP. ISO 8859-1 translates directly into Unicode with a high-order octet of zero. ISO 8859-1 characters with the most-significant bit set are represented as 1100001x 10xxxxxx. (See RFC 2279 [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. H. Schulzrinne et. al. [Page 18] Internet Draft RTSP October 27, 2003 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 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 Content-Length header field. Given the moderate length of presentation descriptions returned, the server should always be able to determine its length, even if it is generated dynamically, making the chunked transfer encoding unnecessary. 5 General Header Fields See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade, and Warning headers are not defined. RTSP further defines the CSeq, and Timestamp: general-header = Cache-Control ; Section 13.9 / Connection ; Section 13.10 / CSeq ; Section 13.17 H. Schulzrinne et. al. [Page 19] Internet Draft RTSP October 27, 2003 / 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 version in use. 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 H. Schulzrinne et. al. [Page 20] Internet Draft RTSP October 27, 2003 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 / 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. H. Schulzrinne et. al. [Page 21] Internet Draft RTSP October 27, 2003 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. 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: H. Schulzrinne et. al. [Page 22] Internet Draft RTSP October 27, 2003 o 1xx: Informational - Request received, continuing process o 2xx: Success - The action was successfully received, understood, and accepted o 3rr: Redirection - Further action must be taken in order to complete the request o 4xx: Client Error - The request contains bad syntax or cannot be fulfilled o 5xx: Server Error - The server failed to fulfill an apparently valid request The individual values of the numeric status codes defined for RTSP/1.0, and an example set of corresponding Reason-Phrase's, are presented 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 H. Schulzrinne et. al. [Page 23] Internet Draft RTSP October 27, 2003 / "413" ; Request Entity Too Large / "414" ; Request-URI Too Large / "415" ; Unsupported Media Type / "451" ; Parameter Not Understood / "452" ; reserved / "453" ; Not Enough Bandwidth / "454" ; Session Not Found / "455" ; Method Not Valid in This State / "456" ; Header Field Not Valid for Resource / "457" ; Invalid Range / "458" ; Parameter Is Read-Only / "459" ; Aggregate operation not allowed / "460" ; Only aggregate operation allowed / "461" ; Unsupported transport / "462" ; Destination unreachable / "500" ; Internal Server Error / "501" ; Not Implemented / "502" ; Bad Gateway / "503" ; Service Unavailable / "504" ; Gateway Time-out / "505" ; RTSP Version not supported / "551" ; Option not supported / extension-code extension-code = 3DIGIT Reason-Phrase = * RTSP status codes are extensible. RTSP applications are not required to understand the meaning of all registered status codes, though such understanding is obviously desirable. However, applications MUST understand the class of any status code, as indicated by the first digit, and treat any unrecognized response as being equivalent to the x00 status code of that class, with the exception that an unrecognized response MUST NOT be cached. For example, if an unrecognized status code of 431 is received by the client, it can safely assume that there was something wrong with its request and treat the response as if it had received a 400 status code. In such cases, user agents SHOULD present to the user the entity returned with the response, since that entity is likely to include human- readable information which will explain the unusual status. 7.1.2 Response Header Fields The response-header fields allow the request recipient to pass additional information about the response which cannot be placed in H. Schulzrinne et. al. [Page 24] Internet Draft RTSP October 27, 2003 the Status-Line. These header fields give information about the server and about further access to the resource identified by the Request-URI. 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 combination with a change in the protocol version. However, new or experimental header fields MAY be given the semantics of response- header fields if all parties in the communication recognize them to be response-header fields. Unrecognized header fields are treated as entity-header fields. 8 Entity Request and Response messages MAY transfer an entity if not otherwise restricted by the request method or response status code. An entity consists of entity-header fields and an entity-body, although some responses will only include the entity-headers. 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 25] Internet Draft RTSP October 27, 2003 Code Reason Method _______________________________________________________ 100 Continue all _______________________________________________________ 200 OK all 201 Created RECORD 250 Low on Storage Space RECORD _______________________________________________________ 300 Multiple Choices all 301 Moved Permanently all 302 Found all 303 See Other all 305 Use Proxy all 350 Going Away all 351 Load Balancing all _______________________________________________________ 400 Bad Request all 401 Unauthorized all 402 Payment Required all 403 Forbidden all 404 Not Found all 405 Method Not Allowed all 406 Not Acceptable all 407 Proxy Authentication Required all 408 Request Timeout all 410 Gone all 411 Length Required all 412 Precondition Failed DESCRIBE, SETUP 413 Request Entity Too Large all 414 Request-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 H. Schulzrinne et. al. [Page 26] Internet Draft RTSP October 27, 2003 Table 1: Status codes and their usage with RTSP methods / 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: o persistent transport connections used for several request- response transactions; o one connection per request/response transaction; o 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 connections, as the media server otherwise has no reliable way of reaching the client. Also, this is the only way that requests from media server to client are likely to traverse firewalls. 9.1 Pipelining H. Schulzrinne et. al. [Page 27] Internet Draft RTSP October 27, 2003 A client that supports persistent connections or connectionless mode MAY "pipeline" its requests (i.e., send multiple requests without waiting for each response). A server MUST send its responses to those requests in the same order that the requests were received. 9.2 Reliability and Acknowledgements | The transmission of RTSP over UDP was optionally to implement and | specified in RFC 2326. However that definition was not satisfactory | for interoperable implementations. Due to lack of interest, this | specification does not specify how RTSP over UDP shall be | implemented. However to maintain backwards compatibility in the | message format certain RTSP headers must be maintained. These | mechanism are described below. The next section Unreliable Transport | (section 9.3) provides documentation of certain features that are | necessary for transport protocols like UDP. | Any RTSP request according to this specification SHALL NOT be sent to | a multicast address. Any RTSP request SHALL be acknowledged. If a | reliable transport protocol is used to carry RTSP, requests MUST NOT | be retransmitted; the RTSP application MUST instead rely on the | underlying transport to provide reliability. | If both the underlying reliable transport such as TCP and | the RTSP application retransmit requests, it is possible | that each packet loss results in two retransmissions. The | receiver cannot typically take advantage of the | application-layer retransmission since the transport stack | will not deliver the application-layer retransmission | before the first attempt has reached the receiver. If the | packet loss is caused by congestion, multiple | retransmissions at different layers will exacerbate the | congestion. | Each request carries a sequence number in the CSeq header (Section | 13.17), which MUST be incremented by one for each distinct request | transmitted to the destination end-point. The initial sequence | number MAY be chosen arbitrary, but is RECOMMENDED to begin with 0. | If a request is repeated because of lack of acknowledgement, the | request MUST carry the original sequence number (i.e., the sequence | number is not incremented). | 9.3 Unreliable Transport | This section provides some information to future specification of | RTSP over unreliable transport. | H. Schulzrinne et. al. [Page 28] Internet Draft RTSP October 27, 2003 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 RTSP is used over a small-RTT LAN, standard procedures for | optimizing initial TCP round trip estimates, such as those used in | T/TCP (RFC 1644) [19], can be beneficial. | The Timestamp header (Section 13.39) is used to avoid the | retransmission ambiguity problem [20] and obviates the need for | Karn's algorithm. | If a request is repeated because of lack of acknowledgement, the | request must carry the original sequence number (i.e., the sequence | number is not incremented). | A number of RTSP 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. | The default port for the RTSP server is 554 for UDP. 9.4 The usage of connections Systems implementing RTSP MUST support carrying RTSP over TCP. The | default port for the RTSP server is 554 for TCP. A number of RTSP | packets destined for the same control end point may be encapsulated | into a TCP stream. RTSP data MAY be interleaved with RTP and RTCP | packets. Unlike HTTP, an RTSP message MUST contain a Content-Length | header field whenever that message contains a payload (entity). | Otherwise, an RTSP packet is terminated with an empty line | immediately following the last message header. TCP can be used for both persistent connections and for one message exchange per connection, as presented above. This section gives further rules and recommendations on how to handle these connections so maximum interoperability and flexibility can be achieved. A server SHALL handle both persistent connections and one request/response transaction per connection. A persistent connection MAY be used for all transactions between the server and client, including messages to multiple RTSP sessions. However the persistent connection MAY also be closed after a few message exchanges, e.g. the initial setup and play command in a session. Later when the client wishes to send a new request, e.g. pause, to the session a new H. Schulzrinne et. al. [Page 29] Internet Draft RTSP October 27, 2003 connection is opened. This connection may either be for a single message exchange or can be kept open for several messages, i.e. persistent. A major motivation for allowing non-persistent connections are that they ensure fault tolerance. A second one is to allow for application layer mobility. A server and client supporting non-persistent connection can survive a loss of a TCP connection, e.g. due to a NAT timeout. When the client has discovered that the TCP connection has been lost, it can set up a new one when there is need to communicate. The client MAY close the connection at any time when no outstanding request/response transactions exist. The server SHOULD NOT close the connection unless at least one RTSP session timeout period has passed without data traffic. A server MUST NOT initiate a close of a connection 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 connection. Older services which was implemented according to RFC 2326 sometimes requires the client to use persistent connection. The client closing the connection may result in that the server removes the session. To achieve interoperability with old servers any client is strongly RECOMMENDED to use persistent connections. A Client is also strongly RECOMMENDED to use persistent connections as it allows the server to send request to the client. In cases where no connection exist between the server and the client, this may cause the server to be forced to drop the RTSP session without notifying the client why,due to the lack of signalling channel. An example of such a case is when the server desires to send a REDIRECT request for a RTSP session to the client. A server implemented according to this specification MUST respond that it supports the "play.basic" feature-tag above. A client MAY send a request including the Supported header in a request to determine support of non-persistent connections. A server supporting non-persistent connections will return the "play.basic" feature-tag in its response. If the client receives the feature-tag in the response, it can be certain that the server handles non-persistent connections. H. Schulzrinne et. al. [Page 30] Internet Draft RTSP October 27, 2003 9.5 Use of IPv6 This specification has been updated so that it supports IPv6. However this support was not present in RFC 2326 therefore some interoperability issues exist. A RFC 2326 implementation can support IPv6 as long as no explicit IPv6 addresses are used within RTSP messages. This require that any RTSP URL pointing at a IPv6 host must use fully qualified domain name and not a IPv6 address. Further the Transport header must not use the parameters source and destination. Implementations according to this specification MUST understand IPv6 addresses in URLs, and headers. By this requirement the feature-tag "play.basic" can be used to determine that a server or client is capable of handling IPv6 within RTSP. 10 Capability Handling This chapter describes the capability handling mechanism available in RTSP which allows RTSP to be extended. Extensions to this version of the protocol are basically done in two ways. First, new headers can be added. Secondly, new methods can be added. The capability handling mechanism is designed to handle these two cases. When a method is added the involved parties can use the OPTIONS method to discover if it is supported. This is done by issuing a OPTIONS request to the other party. Depending on the URL it will either apply in regards to a certain media resource, the whole server in general, or simply the next hop. The OPTIONS response will contain a Public header which declares all methods supported for the indicated resource. It is not necessary to use OPTIONS to discover support of a method, | the client could simply try the method. If the receiver of the | request does not support the method it will respond with an error | code indicating the the method is either not implemented (501) or | does not apply for the resource (405). The choice between the two | discovery methods depends on the requirements of the service. To handle functionality additions that are not new methods feature- tags are defined. Each feature-tag represents a certain block of functionality. The amount of functionality that a feature-tag represents can vary significantly. A simple feature-tag can simple represent the functionality a single header gives. Another feature- tag is "play.basic" which represents the minimal playback implementation according to the updated specification. The feature-tags are then used to determine if the client, server or proxy supports the functionality that is necessary to achieve the H. Schulzrinne et. al. [Page 31] Internet Draft RTSP October 27, 2003 desired service. To determine support of a feature-tag several different headers can be used, each explained below: Supported: The supported header is used to determine the complete set of functionality that both client and server has. The intended usage is to determine before one needs to use a functionality 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 including 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 features that are not understood are given in the Unsupported header in the response. Proxy-Require: This method has the same purpose and workings as Require except that it only applies to proxies and not the end point. Features that needs to be supported by both proxies and end-point needs to be included in both the Require and Proxy-Require header. Unsupported: This header is used in 551 error response to tell which feature(s) that was not supported. Such a response is only the result of the usage of the Require and/or Proxy- Require header where one or more feature where not supported. This information allows the requestor to make the best of situations as it knows which features that was not supported. 11 Method Definitions The method token indicates the method to be performed on the resource identified by the Request-URI case-sensitive. New methods may be H. Schulzrinne et. al. [Page 32] Internet Draft RTSP October 27, 2003 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 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. The OPTIONS method can be used for RTSP session keep alive signalling, however this method is not the most recommended one, see H. Schulzrinne et. al. [Page 33] Internet Draft RTSP October 27, 2003 section 13.37 for a preference list. A keep alive OPTIONS request SHOULD use the media or aggregated control URI. Example: C->S: OPTIONS * RTSP/1.0 CSeq: 1 User-Agent: PhonyClient/1.2 Require: Proxy-Require: gzipped-messages Supported: play-basic S->C: RTSP/1.0 200 OK CSeq: 1 Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE Supported: play-basic, implicit-play, gzipped-messages Server: PhonyServer/1.0 Note that some of the feature-tags in Require and Proxy-Require are necessarily fictional features (one would hope that we would not purposefully overlook a truly useful feature just so that we could have a strong example in this section). 11.2 DESCRIBE The DESCRIBE method retrieves the description of a presentation or media object identified by the request URL from a server. It may use the Accept header to specify the description formats that the client understands. The server responds with a description of the requested resource. The DESCRIBE reply-response pair constitutes the media initialization phase of RTSP. Example: C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 CSeq: 312 User-Agent: PhonyClient 1.2 Accept: application/sdp, application/rtsl, application/mheg S->C: RTSP/1.0 200 OK CSeq: 312 Date: 23 Jan 1997 15:35:06 GMT H. Schulzrinne et. al. [Page 34] Internet Draft RTSP October 27, 2003 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 information for the resource(s) that it describes. If a media client obtains a presentation description from a source other than DESCRIBE and that description contains a complete set of media initialization parameters, the client SHOULD use those parameters and not then request a description for the same media via RTSP. Additionally, servers SHOULD NOT use the DESCRIBE response as a means of media indirection. By forcing a DESCRIBE response to contain all media initialization for the set of streams that it describes, and discouraging use of DESCRIBE for media indirection, we avoid looping problems that might result from other approaches. Media initialization is a requirement for any RTSP-based system, but the RTSP specification does not dictate that this must be done via the DESCRIBE method. There are three ways that an RTSP client may receive initialization information: o via RTSP's DESCRIBE method; o via some other protocol (HTTP, email attachment, etc.); o via the command line or standard input (thus working as a browser helper application launched with an SDP file or other H. Schulzrinne et. al. [Page 35] Internet Draft RTSP October 27, 2003 media initialization format). It is RECOMMENDED that minimal servers support the DESCRIBE method, and highly recommended that minimal clients support the ability to act as a "helper application" that accepts a media initialization 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. The SETUP method may be used in two | different cases; Create a RTSP session or add a media to a session, | and change the transport parameters of already set up media stream. | Using SETUP to create or add media to a session when in PLAY state | are not allowed. Otherwise SETUP can be used in all three states; | INIT, and READY, for both purposes and in PLAY to change the | transport parameters. | The Transport header, see section 13.40, specifies the transport | parameters acceptable to the client for data transmission; the | response will contain the transport parameters selected by the | server. This allows the client to enumerate in priority order the | transport mechanisms and parameters acceptable to it, while the | server can select the most appropriate. All transport parameters | SHOULD be included in the Transport header, the use of other headers | for this purpose is discouraged due to middle boxes. | For the benefit of any intervening firewalls, a client SHOULD | indicate the transport parameters even if it has no influence over | these parameters, for example, where the server advertises a fixed | multicast address. | Since SETUP includes all transport initialization | information, firewalls and other intermediate network | devices (which need this information) are spared the more | arduous task of parsing the DESCRIBE response, which has | been reserved for media initialization. | In a SETUP response the server SHOULD include the Accept-Ranges | header (see section 13.4 to indicate which time formats that are | acceptable to use for this media resource. | C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 | CSeq: 302 | H. Schulzrinne et. al. [Page 36] Internet Draft RTSP October 27, 2003 Transport: RTP/AVP;unicast;client_port=4588-4589, | RTP/AVP/TCP;unicast;interleave=0-1 | S->C: RTSP/1.0 200 OK | CSeq: 302 | Date: 23 Jan 1997 15:35:06 GMT | Server: PhonyServer 1.0 | Session: 47112344 | Transport: RTP/AVP;unicast;client_port=4588-4589; | server_port=6256-6257;ssrc=2A3F93ED | Accept-Ranges: NPT | In the above example the client want to create a RTSP session | containing the media resource "rtsp://example.com/foo/bar/baz.rm". | The transport parameters acceptable to the client is either | RTP/AVP/UDP (UDP per default) to be received on client port 4588 and | 4589 or RTP/AVP interleaved on the RTSP control channel. The server | selects the RTP/AVP/UDP transport and adds the ports it will send and | received RTP and RTCP from, and the RTP SSRC that will be used by the | server. | The server MUST generate a session identifier in response to a | successful SETUP request, unless a SETUP request to a server includes | a session identifier, in which case the server MUST bundle this setup | request into the existing session (aggregated session) or return | error 459 (Aggregate Operation Not Allowed) (see Section 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 is to be specified by the session description if the | server supports aggregated control and aggregated control is desired | for the session. However even if aggregated control is offered the | client MAY chose to not set up the session in aggregated control. | If an Aggregate control URI is not specified in the session | description, it is probably a indication that non-aggregated control | should be used. However a client MAY try to SETUP the session in | aggregated control. If the server refuse to aggregate the specified | media, the server SHALL use the 459 error code. If the server allows | the aggregation, then the 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 (e.g. not default | port). 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. | H. Schulzrinne et. al. [Page 37] Internet Draft RTSP October 27, 2003 While the session ID sometimes has enough information for | aggregate control of a session, the Aggregate control URI | is still important for some methods such as SET_PARAMETER | where the control URI enables the resource in question to | be easily identified. The Aggregate control URI is also | useful for proxies, enabling them to route the request to | the appropriate server, and for logging, where it is useful | to note the actual resource that a request was operating | on. 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 discussion see chapter 13.37. Signs of liveness for a RTSP session are: o Any RTSP request from a client which includes a Session header with that session's ID. o If RTP is used as a transport for the underlying media streams, an RTCP sender or receiver report from the client for any of the media streams in that RTSP session. If a SETUP request on a session fails for any reason, the session | state, as well as transport and other parameters for associated | streams SHALL remain unchanged from their values as if the SETUP | request had never been received by the server. | A client MAY issue a SETUP request for a stream that is already set | up or playing in the session to change transport parameters, which a | server MAY allow. If it does not allow this, it MUST respond with | error 455 (Method Not Valid In This State). Reasons to support | changing transport parameters, is to allow for application layer | mobility and flexibility to utilize the best available transport as | it becomes available. | In a SETUP response for a request to change the transport parameters | while in Play state, the server SHOULD include the Range to indicate | from what point the new transport parameters are used. Further if RTP | is used for delivery the server SHOULD also include the RTP-Info | header to indicate from what timestamp and RTP sequence number the | change has taken place. If both RTP-Info and Range is included in the | response the "rtp_time" parameter and range MUST be for the | H. Schulzrinne et. al. [Page 38] Internet Draft RTSP October 27, 2003 corresponding time, i.e. be used in the same way as for PLAY to | ensure the correct synchronization information is present. | If the transport parameter change while in PLAY state results in a | change of synchronization related information, for example changing | RTP SSRC, the server MUST provide in the SETUP response the necessary | synchronization information. However the server is RECOMMENDED to | avoid changing the synchronization information if possible. | 11.4 PLAY The PLAY method tells the server to start sending data via the | mechanism specified in SETUP. A client MUST NOT issue a PLAY request | until any outstanding SETUP requests have been acknowledged as | successful. PLAY requests are valid when the session is in READY | state; the use of PLAY requests when the session is in PLAY state is | deprecated. A PLAY request MUST include a Session header to indicate | which session the request applies to. In an aggregated session the PLAY request MUST contain an aggregated control URL. A server SHALL responde with error 460 (Only Aggregate Operation Allowed) if the client PLAY request 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 sessions for each media. The PLAY request SHALL position the normal play time to the beginning | of the range specified by the Range header and delivers stream data | until the end of the range if given, else to the end of the media is | reached. To allow for precise composition multiple ranges MAY be | specified in one PLAY Request. The range values are valid if all | given ranges are part of any media within the aggregate. If a given | range value points outside of the media, the response SHALL be the | 457 (Invalid Range) error code. The below example will first play seconds 10 through 15, then, immediately following, seconds 20 to 25, and finally seconds 30 through the end. C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 CSeq: 835 Session: 12345678 Range: npt=10-15, npt=20-25, npt=30- H. Schulzrinne et. al. [Page 39] Internet Draft RTSP October 27, 2003 See the description of the PAUSE request for further examples. A PLAY request without a Range header is legal. It SHALL start | playing a stream from the beginning (npt=0-) unless the stream has | been paused. If a stream has been paused via PAUSE, stream delivery | resumes at the pause point. The stream SHALL play until the end of | the media. The Range header MUST NOT contain a time parameter. The usage of time in PLAY method has been deprecated. Server MUST include a "Range" header in any PLAY response. The | response MUST use the same format as the request's range header | contained. If no Range header was in the request, the NPT time format | SHOULD be used unless the client showed support for an other format. | Also for a session with live media streams the Range header MUST | indicate a valid time. It is RECOMMENDED that normal play time is | used, either the "now" indicator, for example "npt=now-", or the time | since session start as an open interval, e.g. "npt=96.23-". An | absolute time value (clock) for the corresponding time MAY be given, | i.e. "clock=20030213T143205Z-". The UTC clock format SHOULD only be | used if client has shown support for it. | A media server only supporting playback MUST support the npt format | and MAY support the clock and smpte formats. | For a on-demand stream, the server MUST reply with the actual range | that will be played back. This may differ from the requested range if | alignment of the requested range to valid frame boundaries is | required for the media source. If no range is specified in the | request, the start position SHALL still be returned in the reply. If | the medias that are part of an aggregate has different lengths, the | PLAY request SHALL be performed as long as the given range is valid | for any media, for example the longest media. Media will be sent | whenever it is available for the given play-out point. | After playing the desired range, the presentation is NOT | automatically paused, media delivery simply stops. A PAUSE request | MUST be issued before another PLAY request can be issued. Note: This | is one change resulting in a non-operability with RFC 2326 | implementations. A client not issuing a PAUSE request before a new | PLAY will be stuck in PLAY state. | A client desiring to play the media from the beginning MUST send a | PLAY request with a Range header pointing at the beginning, e.g. | npt=0-. If a PLAY request is received without a Range header when | media delivery has stopped at the end, the server SHOULD respond with | a 457 "Invalid Range" error response. In that response the current | H. Schulzrinne et. al. [Page 40] Internet Draft RTSP October 27, 2003 pause point in a Range header SHALL be included. The following example plays the whole presentation starting at SMPTE time code 0:10:20 until the end of the clip. Note: The RTP-Info headers has been broken into several lines to fit the page. C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 CSeq: 833 Session: 12345678 Range: smpte=0:10:20- S->C: RTSP/1.0 200 OK CSeq: 833 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 Range: smpte=0:10:22- RTP-Info:url=rtsp://example.com/twister.en; seq=14783;rtptime=2345962545 For playing back a recording of a live presentation, it may be desirable to use clock units: C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0 CSeq: 835 Session: 12345678 Range: clock=19961108T142300Z-19961108T143520Z S->C: RTSP/1.0 200 OK CSeq: 835 Date: 23 Jan 1997 15:35:06 GMT Server:PhonyServer 1.0 Range: clock=19961108T142300Z-19961108T143520Z RTP-Info:url=rtsp://example.com/meeting.en; seq=53745;rtptime=484589019 All range specifiers in this specification allow for ranges with unspecified begin times (e.g. "npt=-30"). When used in a PLAY request, the server treats this as a request to start/resume playback from the current pause point, ending at the end time specified in the H. Schulzrinne et. al. [Page 41] Internet Draft RTSP October 27, 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 supported. The use of PLAY for keep-alive signaling, i.e. PLAY request without a | range header in PLAY state, has also been depreciated. Instead a | client can use, PING, SET_PARAMETER or OPTIONS for keep alive. A | server receiving a PLAY keep alive SHALL respond with the 455 error | code. 11.5 PAUSE The PAUSE request causes the stream delivery to be interrupted (halted) temporarily. A PAUSE request MUST be done with the aggregated control 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 | 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 H. Schulzrinne et. al. [Page 42] Internet Draft RTSP October 27, 2003 "pause point". The time parameter in the Range MUST NOT be used. The Range header MUST contain a single value, expressed as the beginning value an open range. For example, the following clip will be played from 10 seconds through 21 seconds of the clip's normal play time, under the assumption that the PAUSE request reaches the server within 11 seconds of the PLAY request. Note that some lines has been broken in an non-correct way to fit the page: C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 834 Session: 12345678 Range: npt=10-30 S->C: RTSP/1.0 200 OK CSeq: 834 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 Range: npt=10-30 RTP-Info:url=rtsp://example.com/fizzle/audiotrack; seq=5712;rtptime=934207921, url=rtsp://example.com/fizzle/videotrack; seq=57654;rtptime=2792482193 Session: 12345678 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 835 Session: 12345678 Range: npt=21- S->C: RTSP/1.0 200 OK CSeq: 835 Date: 23 Jan 1997 15:35:09 GMT Server: PhonyServer 1.0 Range: npt=21- Session: 12345678 The pause request becomes effective the first time the server is encountering the time point specified in any of the multiple ranges. If the Range header specifies a time outside any range from the PLAY request, the error 457 (Invalid Range) SHALL be returned. If a media unit (such as an audio or video frame) starts presentation at exactly the pause point, it is not played. If the Range header is missing, stream delivery is interrupted immediately on receipt of the message and the pause point is set to the current normal play time. However, H. Schulzrinne et. al. [Page 43] Internet Draft RTSP October 27, 2003 the pause point in the media stream MUST be maintained. A subsequent PLAY request without Range header SHALL resume from the pause point and play until media end. If the server has already sent data beyond the time specified in the | PAUSE request's Range header, a PLAY without range SHALL resume at | the point in time specified by the PAUSE request's Range header, as | it is assumed that the client has discarded data after that point. | This ensures continuous pause/play cycling without gaps. | The pause point after any PAUSE request SHALL be returned to the | client by adding a Range header with what remains unplayed of the | PLAY request's ranges, i.e. including all the remaining ranges part | of multiple range specification. If one desires to resume playing a | ranged request, one simple included the Range header from the PAUSE | response. Note that this server behavior was not mandated previously | and servers implementing according to RFC 2326 will probably not | return the range header. | For example, if the server have a play request for ranges 10 to 15 | and 20 to 29 pending and then receives a pause request for NPT 21, it | would start playing the second range and stop at NPT 21. If the pause | request is for NPT 12 and the server is playing at NPT 13 serving the | first play request, the server stops immediately. If the pause | request is for NPT 16, the server returns a 457 error message. To | prevent that the second range is played and the server stops after | completing the first range, a PAUSE request for 20 must be issued. | As another example, if a server has received requests to play ranges | 10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE | request for NPT=14 would take effect while the server plays the first | range, with the second range effectively being ignored, assuming the | PAUSE request arrives before the server has started playing the | second, overlapping range. Regardless of when the PAUSE request | arrives, it sets the pause point to 14. The below example messages is | for the above case when the PAUSE request arrives before the first | occurrence of NPT=14. | C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0 | CSeq: 834 | Session: 12345678 | Range: npt=10-15, npt=13-20 | S->C: RTSP/1.0 200 OK | CSeq: 834 | Date: 23 Jan 1997 15:35:06 GMT | H. Schulzrinne et. al. [Page 44] Internet Draft RTSP October 27, 2003 Server: PhonyServer 1.0 | Range: npt=10-15, npt=13-20 | RTP-Info:url=rtsp://example.com/fizzle/audiotrack; | seq=5712;rtptime=934207921, | url=rtsp://example.com/fizzle/videotrack; | seq=57654;rtptime=2792482193 | Session: 12345678 | C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 | CSeq: 835 | Session: 12345678 | Range: npt=14- | S->C: RTSP/1.0 200 OK | CSeq: 835 | Date: 23 Jan 1997 15:35:09 GMT | Server: PhonyServer 1.0 | Range: npt=14-15, npt=13-20 | Session: 12345678 | If a client issues a PAUSE request and the server acknowledges and | enters the READY state, the proper server response, if the player | issues another PAUSE, is still 200 OK. The 200 OK response MUST | include the Range header with the current pause point, even if the | PAUSE request is asking for some other pause point. See examples | below: Examples: C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 834 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 834 Session: 12345678 Date: 23 Jan 1997 15:35:06 GMT Range: npt=45.76- C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 835 Session: 12345678 Range: 86- H. Schulzrinne et. al. [Page 45] Internet Draft RTSP October 27, 2003 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 client to server request stops the stream delivery for | the given URI, freeing the resources associated with it. TEARDOWN | MAY be done using either an aggregated or a media control URI. | However some restrictions apply depending on the current state. The | TEARDOWN request SHALL contain a Session header indicating what | session the request applies to. | A TEARDOWN using the aggregated control URI or the media URI in a | session under non-aggregated control MAY be done in any state (Ready, | and Play). A successful request SHALL result in that media delivery | is immediately halted and the session state is destroyed. This SHALL | be indicated through the lack of a Session header in the response. | A TEARDOWN using a media URI in an aggregated session MAY only be | done in Ready state. Such a request only removes the indicated media | stream and associated resources from the session. This may result in | that a session returns to non-aggregated control. In the response to | TEARDOWN request resulting in that the session still exist SHALL | contain a Session header to indicate this. | Note, the indication with the session header if sessions state remain | may not be done correctly by a RFC 2326 client, but will be for any | server signalling the "play.basic" tag. Example: C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 892 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 892 Server: PhonyServer 1.0 H. Schulzrinne et. al. [Page 46] Internet Draft RTSP October 27, 2003 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 be defined after further experimentation. 11.8 SET_PARAMETER This method requests to set the value of a parameter for a presentation 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 H. Schulzrinne et. al. [Page 47] Internet Draft RTSP October 27, 2003 MAY disallow changing parameter values. If the receiver of the request does not understand or can locate a parameter error 451 (Parameter Not Understood) SHALL be used. In the case a parameter is not allowed to change the error code 458 (Parameter Is Read-Only). The response body SHOULD contain only the parameters that has errors. Otherwise no body SHALL be returned. Note: transport parameters for the media stream MUST only be set with the SETUP command. Restricting setting transport parameters to SETUP is for the benefit of firewalls. The parameters are split in a fine-grained fashion so that there can be more meaningful error indications. However, it may make sense to allow the setting of several parameters if an atomic setting is desirable. Imagine device control where the client does not want the camera to pan unless it can also tilt to the right angle at the same time. Example: C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 421 Content-length: 20 Content-type: text/parameters barparam: barstuff S->C: RTSP/1.0 451 Parameter Not Understood CSeq: 421 Content-length: 10 Content-type: text/parameters barparam The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation. 11.9 REDIRECT H. Schulzrinne et. al. [Page 48] Internet Draft RTSP October 27, 2003 A redirect request informs the client that it MUST connect to another | server location. The REDIRECT request MAY contain the header | Location, which indicates that the client should issue requests for | that URL. The lack of a Location header in the response SHALL be | interpreted as that the server can't any longer fulfill the current | request, but has no alternative at the present where the client | continue. | If a REDIRECT request contains a Session header, it is end-to-end and | applies only to the given session. If there are proxies in the | request chain, they SHOULD NOT disconnect the control channel unless | there are no remaining sessions. If the Location header is included | it SHALL contain a full absolute URI pointing out the resource to | reconnect too, i.e. the Location SHALL NOT contain only host and | port. | If a REDIRECT request does not contain a Session header, it is next- | hop and applies also to the control connection. If the Location | header is included it SHOULD only contain an absolute URI containing | the host address and OPTIONAL the port number. If there are proxies | in the request chain, they SHOULD do all of the following: (1) | respond to the REDIRECT request, (2) disconnect the control channel | from the requestor, (3) reconnect to the given host address, and (4) | pass the request to each applicable client (typically those clients | with an active session or unanswered request from the requestor). | Note that the proxy is responsible for accepting the REDIRECT | response from its clients and these responses MUST NOT be passed on | to either the requesting or the destination server. The redirect request MAY contain the header Range, which indicates when the redirection takes effect. If the Range contains a "time=" value that is the wall clock time that the redirection MUST at the latest take place. When the "time=" parameter is present the range value MUST be ignored. However the range entered MUST be syntactical correct and SHALL point at the beginning of any on-demand content. If no time parameter is part of the Range header then redirection SHALL take place when the media playout from the server reaches the given time. The range value MUST be a single value in the open ended form, e.g. npt=59-. A server upon receiving a successful (2xx) response for a REDIRECT | request without any Range header SHALL consider the session as | removed and can free any session state. For this type of requests the | rest of this paragraph applies. The server MAY close the signalling | connection upon receiving the response for REDIRECT requests without | a Session header. The client SHOULD close the signaling connection | after having given the 2xx response to a REDIRECT response, unless it | has several sessions on the server. If the client has multiple | H. Schulzrinne et. al. [Page 49] Internet Draft RTSP October 27, 2003 session on the server it SHOULD close the connection when it has | received and responded to REDIRECT requests for all sessions. | A client receiving a REDIRECT request with a Range header SHALL issue | a TEARDOWN request when the in indicated redirect point is reached. | The client SHOULD for REDIRECT requests with Range header close the | signalling connection after a 2xx response on its TEARDOWN request. | The normal connection considerations apply for the server. This | differentiation from REDIRECT requests without range headers is to | allow clear an explicit state handling. As the state in the server | needs to be kept until the point of redirection, the handling becomes | more clear if the client is required to tear down the session at that | point. | If the client wants to continue to send or receive media for this | resource, the client will have to establish a new session with the | designated host. A client SHOULD issue a new DESCRIBE request with | the URL given in the Location header, unless the URL only contains a | host address. In the cases the Location only contains a host address | the client MAY assume that the media on the server it is redirected | to is identical. Identical media means that all media configuration | information from the old session still is valid except for the host | address. In the case of absolute URLs in the location header the | media redirected to can be either identical, slightly different or | totally different. This is the reason why a new DESCRIBE request | SHOULD be issued. This example request redirects traffic for this session to the new server at the given absolute time: S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 732 Location: rtsp://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 liveness checking. It has no side effects. The issuer of the request MUST include a session header with the session ID of the session that is being checked for liveness. Prior to using this method, an OPTIONS method is RECOMMENDED to be H. Schulzrinne et. al. [Page 50] Internet Draft RTSP October 27, 2003 issued in the direction which the PING method would be used. This method MUST NOT be used if support is not indicated by the Public header. Note: That an 501 (Not Implemented) response means that the keep-alive timer has not been updated. When a proxy is in use, PING with a * indicates a single-hop liveness check, whereas PING with a URL including an host address indicates an end-to-end liveness check. Example: C->S: PING * RTSP/1.0 CSeq: 123 Session:12345678 S->C: RTSP/1.0 200 OK CSeq: 123 Session:12345678 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, followed by the length of the encapsulated binary data as a binary, two-byte integer in network byte order. The stream data follows immediately afterwards, without a CRLF, but including the upper-layer protocol headers. Each $ block 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 : H. Schulzrinne et. al. [Page 51] Internet Draft RTSP October 27, 2003 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 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 directions. RTCP is needed for synchronization when two or more streams are interleaved in such a fashion. Also, this provides a convenient way to tunnel RTP/RTCP packets through the TCP control connection when required by the network configuration and transfer them onto UDP when possible. C->S: SETUP rtsp://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 H. Schulzrinne et. al. [Page 52] Internet Draft RTSP October 27, 2003 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]. 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 indicate what time period it may still be able to record. Since other processes on the server may be consuming storage space simultaneously, 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 comments are given for some to how they apply to RTSP. Within RTSP, redirection may be used for load balancing or redirecting stream requests to a server topologically closer to the client. Mechanisms to determine topological proximity are beyond the scope of this specification. If the the Location header is used in a response it SHALL contain an | absolute URI pointing out the media resource the client is redirected | to, the URI SHALL NOT only contain the host name. H. Schulzrinne et. al. [Page 53] Internet Draft RTSP October 27, 2003 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 automatically 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 redirects, e.g. load balancing. It is RECOMMENDED that one set the reason phrase to something more meaningful than "Found" in these cases. The user client SHOULD redirect automatically to the given 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. 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: o Date o ETag and/or Content-Location, if the header would have been sent in a 200 response to the same request. o Expires, Cache-Control, and/or Vary, if the field-value might differ from that sent in any previous response for the same variant. This response is independent for the DESCRIBE and SETUP requests. That is, a 304 response to DESCRIBE does NOT imply that the resource content is unchanged and a 304 response to SETUP does NOT imply that the resource description is unchanged. The ETag and If-Match headers may be used to link the DESCRIBE and SETUP in this manner. H. Schulzrinne et. al. [Page 54] Internet Draft RTSP October 27, 2003 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 modifications [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. 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 failure. 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 H. Schulzrinne et. al. [Page 55] Internet Draft RTSP October 27, 2003 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 b