Real Time Streaming Protocol
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The Real Time Streaming Protocol (RTSP), developed by the IETF and published 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.
Some RTSP servers use RTP as the transport protocol for the actual audio/video data. Many RTSP servers use RealNetworks's proprietary RDT as the transport protocol.
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[edit] Clients
[edit] RTSP commands
RTSP requests are based on HTTP 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 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] See also
[edit] External links
- RTSP.org, a central information repository about RTSP. (Site DOA 2006-04-25)
- RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
- RFC 2326, Real Time Streaming Protocol (RTSP).
- Fenice, Server from the Open Source Software Suite (LS)³ - Libre Streaming, Libre Software, Libre Standards ([1])
- Helix Community RDT implementation, C++ source code repository.
- RTSP Streaming Sessions, an overview of the process flow for RTSP sessions within the Helix/RealNetworks DNA Client
- protocol.helixcommunity.org Protocol specifications of Helix/RealNetworks implementation.
- myIPTV.org Overview of RTSP with code examples and simple RTSP testing client.