The Real Time Streaming Protocol (RTSP) is an application-level network protocol designed for multiplexing and packetizing multimedia transport streams (such as interactive media, video and audio) over a suitable transport protocol. RTSP is used in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between endpoints. Clients of media servers issue commands such as play, record and pause, to facilitate real-time control of the media streaming from the server to a client (video on demand) or from a client to the server (voice recording).
RTSP was developed by RealNetworks, Netscape[1] and Columbia University.[2] The first draft was submitted to IETF in October 1996 by Netscape and Progressive Networks, after which Henning Schulzrinne from Columbia University submitted "RTSP՚" ("RTSP prime") in December 1996.[3][4] The two drafts were merged for standardization by the Multiparty Multimedia Session Control Working Group (MMUSIC WG) of the Internet Engineering Task Force (IETF) and further drafts were published by the working group.[5][6] The Proposed Standard for RTSP was published as RFC 2326 in 1998.[7] RTSP 2.0 published as RFC 7826 in 2016 as a replacement of RTSP 1.0. RTSP 2.0 is based on RTSP 1.0 but is not backwards compatible other than in the basic version negotiation mechanism, and remains a "Proposed Standard".[8]
Internet protocol suite |
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Application layer |
Transport layer |
Internet layer |
Link layer |
Main article: Real-time Transport Protocol |
The transmission of streaming data itself is not a task of RTSP. Most RTSP servers use the Real-time Transport Protocol (RTP) in conjunction with Real-time Control Protocol (RTCP) for media stream delivery. However, some vendors implement proprietary transport protocols. The RTSP server software from RealNetworks, for example, also used RealNetworks' proprietary Real Data Transport (RDT).
While similar in some ways to HTTP, RTSP defines control sequences useful in controlling multimedia playback. While HTTP is stateless, RTSP has a 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[7] for both TCP and UDP, the latter being rarely used for the control requests.
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
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"
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;ssrc=1234ABCD Session: 12345678 C->S: SETUP rtsp://example.com/media.mp4/streamid=1 RTSP/1.0 CSeq: 3 Transport: RTP/AVP;unicast;client_port=8002-8003 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 3 Transport: RTP/AVP;unicast;client_port=8002-8003;server_port=9002-9003;ssrc=1234ABCD Session: 12345678
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
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
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
The ANNOUNCE method serves two purposes:
C->S: ANNOUNCE 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
C->S: TEARDOWN rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 8 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 8
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
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
S->C: REDIRECT rtsp://example.com/media.mp4 RTSP/1.0 CSeq: 11 Location: rtsp://bigserver.com:8001 Range: clock=19960213T143205Z-
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}
RTSP using RTP and RTCP allows for the implementation of rate adaptation.[9]
Many CCTV / Security cameras, often called IP cameras, support RTSP streaming too, especially those with ONVIF profiles G, S, T.