Real Time Streaming Protocol - Protocol Directives

Protocol Directives

While similar in some ways to HTTP, RTSP defines control sequences useful in controlling multimedia playback. While HTTP is stateless, RTSP has state; an identifier is used when needed to track concurrent sessions. Like HTTP, RTSP uses TCP to maintain an end-to-end connection and, while most RTSP control messages are sent by the client to the server, some commands travel in the other direction (i.e. from server to client).

Presented here are the basic RTSP requests. Some typical HTTP requests, like the OPTIONS request, are also available. The default transport layer port number is 554.

OPTIONS
An OPTIONS request returns the request types the server will accept.
C->S: OPTIONS rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 1 Require: implicit-play Proxy-Require: gzipped-messages S->C: RTSP/1.0 200 OK CSeq: 1 Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
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 (deprecated and very rarely used) and TCP transports. This reply includes the presentation description, typically in Session Description Protocol (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 each for audio and video.
C->S: DESCRIBE rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 2 S->C: RTSP/1.0 200 OK CSeq: 2 Content-Base: rtsp://example.com/media.mp4 Content-Type: application/sdp Content-Length: 460 m=video 0 RTP/AVP 96 a=control:streamid=0 a=range:npt=0-7.741000 a=length:npt=7.741000 a=rtpmap:96 MP4V-ES/5544 a=mimetype:string;"video/MP4V-ES" a=AvgBitRate:integer;304018 a=StreamName:string;"hinted video track" m=audio 0 RTP/AVP 97 a=control:streamid=1 a=range:npt=0-7.712000 a=length:npt=7.712000 a=rtpmap:97 mpeg4-generic/32000/2 a=mimetype:string;"audio/mpeg4-generic" a=AvgBitRate:integer;65790 a=StreamName:string;"hinted audio track"
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.
C->S: SETUP rtsp://example.com/media.mp4/streamid=0 RTSP/1.0 CSeq: 3 Transport: RTP/AVP;unicast;client_port=8000-8001 S->C: RTSP/1.0 200 OK CSeq: 3 Transport: RTP/AVP;unicast;client_port=8000-8001;server_port=9000-9001 Session: 12345678
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.
C->S: PLAY rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 4 Range: npt=5-20 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 4 Session: 12345678 RTP-Info: url=rtsp://example.com/media.mp4/streamid=0;seq=9810092;rtptime=3450012
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. A range parameter on a PAUSE request specifies when to pause. When the range parameter is omitted, the pause occurs immediately and indefinitely.
C->S: PAUSE rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 5 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 5 Session: 12345678
RECORD
This method initiates recording a range of media data according to the presentation description. The time stamp reflects start and end time(UTC). If no time range is given, use the start or end time provided in the presentation description. If the session has already started, commence recording immediately. The server decides whether to store the recorded data under the request URI or another URI. If the server does not use the request URI, the response should be 201 and contain an entity which describes the states of the request and refers to the new resource, and a Location header.
C->S: RECORD rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 6 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 6 Session: 12345678
ANNOUNCE
The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a presentation or media object identified by the request URL to a server. When sent from server to client, ANNOUNCE updates the session description in real-time. If a new media stream is added to a presentation (e.g., during a live presentation), the whole presentation description should be sent again, rather than just the additional components, so that components can be deleted.
C->S: ANNOUNCE rtsp:rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 7 Date: 23 Jan 1997 15:35:06 GMT Session: 12345678 Content-Type: application/sdp Content-Length: 332 v=0 o=mhandley 2890844526 2890845468 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 S->C: RTSP/1.0 200 OK CSeq: 7
TEARDOWN
A TEARDOWN request is used to terminate the session. It stops all media streams and frees all session related data on the server.
C->S: TEARDOWN rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 8 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 8
GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter of a presentation or stream specified in the URI. 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").
S->C: GET_PARAMETER rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 9 Content-Type: text/parameters Session: 12345678 Content-Length: 15 packets_received jitter C->S: RTSP/1.0 200 OK CSeq: 9 Content-Length: 46 Content-Type: text/parameters packets_received: 10 jitter: 0.3838
SET_PARAMETER
This method requests to set the value of a parameter for a presentation or stream specified by the URI.
C->S: SET_PARAMETER rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 10 Content-length: 20 Content-type: text/parameters barparam: barstuff S->C: RTSP/1.0 451 Invalid Parameter CSeq: 10 Content-length: 10 Content-type: text/parameters barparam
REDIRECT

: A redirect request informs the client that it must connect to another server location. It contains the mandatory header Location, which indicates that the client should issue requests for that URL. It may contain the parameter Range, which indicates when the redirection takes effect. If the client wants to continue to send or receive media for this URI, the client MUST issue a TEARDOWN request for the current session and a SETUP for the new session at the designated host.

S->C: REDIRECT rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 11 Location: rtsp://bigserver.com:8001 Range: clock=19960213T143205Z-
Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server to interleave RTSP methods and stream data. This interleaving should generally be avoided unless necessary since it complicates client and server operation and imposes additional overhead. 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 hexadecimal), 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.
C->S: SETUP rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 3 Transport: RTP/AVP/TCP;interleaved=0-1 S->C: RTSP/1.0 200 OK CSeq: 3 Date: 05 Jun 1997 18:57:18 GMT Transport: RTP/AVP/TCP;interleaved=0-1 Session: 12345678 C->S: PLAY rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 4 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 4 Session: 12345678 Date: 05 Jun 1997 18:59:15 GMT RTP-Info: url=rtsp://example.com/media.mp4; seq=232433;rtptime=972948234 S->C: $\000{2 byte length}{"length" bytes data, w/RTP header} S->C: $\000{2 byte length}{"length" bytes data, w/RTP header} S->C: $\001{2 byte length}{"length" bytes RTCP packet}

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