Real Time Streaming Protocol

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The Real Time Streaming Protocol (RTSP), developed by the IETF and created in 1998 as RFC 2326, is a protocol for use in streaming media systems which allows a client to remotely control a streaming media server, issuing VCR-like commands such as "play" and "pause", and allowing time-based access to files on a server.

Most RTSP servers use the standards-based RTP as the transport protocol for the actual audio/video data, acting somewhat as a metadata channel. The RTSP server from RealNetworks also features Real's proprietary RDT as the transport protocol.

The set of standards that include RTSP and RTP are unfortunately not sufficiently complete or specific to ensure interoperablity and each client/server implementation tends to be a little different.[citation needed]

Contents

[edit] Clients

[edit] RTSP commands

The protocol is similar in syntax and operation to HTTP but RTSP adds new requests. While HTTP is stateless, RTSP is a stateful protocol. A session ID is used to keep track of sessions when needed. This way, no permanent TCP connection is needed. RTSP messages are sent from client to server, although some exceptions exist where the server will send to the client. Below are the basic RTSP requests. A number of typical HTTP requests, like an OPTION request, are also frequently used.

[edit] DESCRIBE

A DESCRIBE request includes an RTSP URL (rtsp://...), and the type of reply data that can be handled. The default port for the RTSP protocol is 554 for both UDP and TCP transports.

The reply includes the presentation description, typically in SDP format. Among other things, the presentation description lists the media streams controlled with the aggregate URL. In the typical case, there is one media stream for audio and one for video.

[edit] SETUP

A SETUP request specifies how a single media stream must be transported. This must be done before a PLAY request is sent.

The request contains the media stream URL and a transport specifier. This specifier typically includes a local port for receiving RTP data (audio or video), and another for RTCP data (meta information).

The server reply usually confirms the chosen parameters, and fills in the missing parts, such as the server's chosen ports. Each media stream must be configured using SETUP before an aggregate play request may be sent.

[edit] PLAY

A PLAY request will cause one or all media streams to be played. Play requests can be stacked by sending multiple PLAY requests.

The URL may be the aggregate URL (to play all media streams), or a single media stream URL (to play only that stream). A range can be specified. If no range is specified, the stream is played from the beginning and plays to the end, or, if the stream is paused, it is resumed at the point it was paused..

[edit] PAUSE

A PAUSE request temporarily halts one or all media streams, so it can later be resumed with a PLAY request.

The request contains an aggregate or media stream URL. When to pause can be specified with a range parameter. The range parameter can be left out to pause immediately.

[edit] RECORD

The RECORD request can be used to send a stream to the server for storage.

[edit] TEARDOWN

A TEARDOWN request is used to terminate the session. It stops all media streams and frees all session related data on the server.

[edit] Disadvantages

Because RTSP is a real-time protocol, it is not as practical as http or ftp when trying to download files onto a hard drive for later use.

[edit] Server implementations

[edit] See also

[edit] External links

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