1 Introduction

1 Introduction

1.1 Purpose

The Real-Time Streaming Protocol (RTSP) establishes and controls either a single or several time-synchronized streams of continuous media such as audio and video. It does not typically deliver the continuous streams itself, although interleaving of the continuous media stream with the control stream is possible (see Section 10.12). In other words, RTSP acts as a "network remote control" for multimedia servers.

The set of streams to be controlled is defined by a presentation description. This memorandum does not define a format for a presentation description.

There is no notion of an RTSP connection; instead, a server maintains a session labeled by an identifier. An RTSP session is in no way tied to a transport-level connection such as a TCP connection. During an RTSP session, an RTSP client may open and close many reliable transport connections to the server to issue RTSP requests. Alternatively, it may use a connectionless transport protocol such as UDP.

The streams controlled by RTSP may use RTP [1], 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 [2] so that extension mechanisms to HTTP can in most cases also be added to RTSP. However, RTSP differs in a number of important aspects from HTTP:

• RTSP introduces a number of new methods and has a different protocol identifier. * An RTSP server needs to maintain state by default in almost all cases, as opposed to the stateless nature of HTTP. * Both an RTSP server and client can issue requests. * Data is carried out-of-band by a different protocol. (There is an exception to this.) * RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, consistent with current HTML internationalization efforts [3]. * The Request-URI always contains the absolute URI. Because of backward compatibility with a historical blunder, HTTP/1.1 [2] 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 be "invited" to join an existing conference, either to play back media into the presentation or to record all or a subset of the media in a 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 [2].

1.2 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].

1.3 Terminology

Some of the terminology has been adopted from HTTP/1.1 [2]. Terms not listed here are defined as in HTTP/1.1.

Aggregate control: The control of the multiple streams using a single timeline by the server. For audio/video feeds, this means that the client may issue a single play or pause message to control both the audio and video feeds.

Conference: a multiparty, multimedia presentation, where "multi" implies greater than or equal to one.

Client: The client requests continuous media data from the media server.

Connection: A transport layer virtual circuit established between two programs for the purpose of communication.

Container file: A file which may contain multiple media streams which often comprise a presentation when played together. RTSP servers may offer aggregate control on these files, though the concept of a container file is not embedded in the protocol.

Continuous media: Data where there is a timing relationship between source and sink; that is, the sink must reproduce the timing relationship that existed at the source. The most common examples of continuous media are audio and motion video. Continuous media can be real-time (interactive), where there is a "tight" timing relationship between source and sink, or streaming (playback), where the relationship is less strict.

Entity: The information transferred as the payload of a request or response. An entity consists of metainformation in the form of entity-header fields and content in the form of an entity- body, as described in Section 8.

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 or recording 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 15 and transmitted via a connection or a connectionless protocol.

Participant: Member of a conference. A participant may be a machine, e.g., a media record or 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 [6]) 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: An RTSP response. If an HTTP response is meant, that is indicated explicitly.

Request: An RTSP request. If an HTTP request is meant, that is indicated explicitly.

RTSP session: A complete RTSP "transaction", e.g., the viewing of a movie. A session typically consists of a client setting up a transport mechanism for the continuous media stream (SETUP), starting the stream with PLAY or RECORD, and closing the

stream with TEARDOWN.

Transport initialization: The negotiation of transport information (e.g., port numbers, transport protocols) between the client and the server.

1.4 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. All HTTP authentication mechanisms such as basic (RFC 2068 [2, Section 11.1]) and digest authentication (RFC 2069 [8]) are directly applicable. One may also reuse transport or network layer security mechanisms.

Transport-independent: RTSP may use either an unreliable datagram protocol (UDP) (RFC 768 [9]), a reliable datagram protocol (RDP, RFC 1151, not widely used [10]) or a reliable stream protocol such as TCP (RFC 793 [11]) as it implements application-level reliability.

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.

Control of recording devices: The protocol can control both recording and playback devices, as well as devices that can alternate between the two modes ("VCR").

Separation of stream control and conference initiation: Stream control is divorced from inviting a media server to a conference. The only requirement is that the conference initiation protocol either provides or can be used to create a unique conference identifier. In particular, SIP [12] or H.323 [13] 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 [14]) 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 [15,16]) 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.

1.5 Extending RTSP

Since not all media servers have the same functionality, media servers by necessity will support different sets of requests. For example:

• A server may only be capable of playback thus has no need to support the RECORD request. * A server may not be capable of seeking (absolute positioning) if it is to support live events only. * Some servers may not support setting stream parameters and thus not support GET_PARAMETER and SET_PARAMETER.

A server SHOULD implement all header fields described in Section 12.

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 [2], where the methods described in [H19.6] 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:

• 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 12.32). * New methods can be added. If the recipient of the message does not understand the request, it responds with error code 501 (Not implemented) and the sender should not attempt to use this method again. A client may also use the OPTIONS method to inquire about methods supported by the server. The server SHOULD list the methods it supports using the Public response header. * A new version of the protocol can be defined, allowing almost all aspects (except the position of the protocol version number) to change.

1.6 Overall Operation

Each presentation and media stream may be identified by an RTSP URL. The overall presentation and the properties of the media the presentation is made up of are defined by a presentation description file, the format of which is outside the scope of this specification. The presentation description file may be obtained by the client using

HTTP or other means such as email and may not necessarily be stored on the media server.

For the purposes of this specification, a presentation description is assumed to describe one or more presentations, each of which maintains a common time axis. For simplicity of exposition and without loss of generality, it is assumed that the presentation description contains exactly one such presentation. A presentation may contain several media streams.

The presentation description file contains a description of the media streams making up the presentation, including their encodings, language, and other parameters that enable the client to choose the most appropriate combination of media. In this presentation description, each media stream that is individually controllable by RTSP is identified by an 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.

1.7 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 Section A.

Many methods in RTSP do not contribute to state. However, the following play a central role in defining the allocation and usage of stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and TEARDOWN.

SETUP: Causes the server to allocate resources for a stream and start an RTSP session.

PLAY and RECORD: Starts data transmission on a stream allocated via SETUP.

PAUSE: Temporarily halts a stream without freeing server resources.

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 12.37) to identify the RTSP session whose state is being manipulated. The server generates session identifiers in response to SETUP requests (Section 10.4).

1.8 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 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 not stateless; 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.

While most real-time media will use RTP as a transport protocol, RTSP is not tied to RTP.

RTSP assumes the existence of a presentation description format that can express both static and temporal properties of a presentation containing several media streams.