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authorMartin Storsjö <martin@martin.st>2010-10-04 07:06:58 +0000
committerMartin Storsjö <martin@martin.st>2010-10-04 07:06:58 +0000
commit92c5052db971a2a3dfb7c3fc4317b55f401d7811 (patch)
tree1d4f63d3aacc4730a0e627552a11776c925cc880 /doc/protocols.texi
parent59ac1da26d889dd89ba220eb5d88ed1c518ceb11 (diff)
downloadffmpeg-92c5052db971a2a3dfb7c3fc4317b55f401d7811.tar.gz
Add protocol documentation on RTSP
Originally committed as revision 25330 to svn://svn.ffmpeg.org/ffmpeg/trunk
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diff --git a/doc/protocols.texi b/doc/protocols.texi
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@@ -217,6 +217,76 @@ ffplay "rtmp://myserver/live/mystream live=1"
Real-Time Protocol.
+@section rtsp
+
+RTSP is not technically a protocol handler in libavformat, it is a demuxer
+and muxer. The demuxer supports both normal RTSP (with data transferred
+over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
+data transferred over RDT).
+
+The muxer can be used to send a stream using RTSP ANNOUNCE to a server
+supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
+RTSP server, @url{http://github.com/revmischa/rtsp-server}).
+
+The required syntax for a RTSP url is:
+@example
+rtsp://@var{hostname}[:@var{port}]/@var{path}[?@var{options}]
+@end example
+
+@var{options} is a @code{&}-separated list. The following options
+are supported:
+
+@table @option
+
+@item udp
+Use UDP as lower transport protocol.
+
+@item tcp
+Use TCP (interleaving within the RTSP control channel) as lower
+transport protocol.
+
+@item multicast
+Use UDP multicast as lower transport protocol.
+
+@item http
+Use HTTP tunneling as lower transport protocol, which is useful for
+passing proxies.
+@end table
+
+Multiple lower transport protocols may be specified, in that case they are
+tried one at a time (if the setup of one fails, the next one is tried).
+For the muxer, only the @code{tcp} and @code{udp} options are supported.
+
+When receiving data over UDP, the demuxer tries to reorder received packets
+(since they may arrive out of order, or packets may get lost totally). In
+order for this to be enabled, a maximum delay must be specified in the
+@code{max_delay} field of AVFormatContext.
+
+When watching multi-bitrate Real-RTSP streams with @file{ffplay}, the
+streams to display can be chosen with @code{-vst} @var{n} and
+@code{-ast} @var{n} for video and audio respectively, and can be switched
+on the fly by pressing @code{v} and @code{a}.
+
+Example command lines:
+
+To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
+
+@example
+ffplay -max_delay 500000 rtsp://server/video.mp4?udp
+@end example
+
+To watch a stream tunneled over HTTP:
+
+@example
+ffplay rtsp://server/video.mp4?http
+@end example
+
+To send a stream in realtime to a RTSP server, for others to watch:
+
+@example
+ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
+@end example
+
@section tcp
Trasmission Control Protocol.