doc/protocols.texi
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 @chapter Protocols
 @c man begin PROTOCOLS
 
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 Protocols are configured elements in FFmpeg which allow to access
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 resources which require the use of a particular protocol.
 
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 When you configure your FFmpeg build, all the supported protocols are
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 enabled by default. You can list all available ones using the
 configure option "--list-protocols".
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 You can disable all the protocols using the configure option
 "--disable-protocols", and selectively enable a protocol using the
 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
 particular protocol using the option
 "--disable-protocol=@var{PROTOCOL}".
 
 The option "-protocols" of the ff* tools will display the list of
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 supported protocols.
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 A description of the currently available protocols follows.
 
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 @section applehttp
 
 Read Apple HTTP Live Streaming compliant segmented stream as
 a uniform one. The M3U8 playlists describing the segments can be
 remote HTTP resources or local files, accessed using the standard
 file protocol.
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 HTTP is default, specific protocol can be declared by specifying
 "+@var{proto}" after the applehttp URI scheme name, where @var{proto}
 is either "file" or "http".
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 @example
 applehttp://host/path/to/remote/resource.m3u8
 applehttp+http://host/path/to/remote/resource.m3u8
 applehttp+file://path/to/local/resource.m3u8
 @end example
 
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 @section concat
 
 Physical concatenation protocol.
 
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 Allow to read and seek from many resource in sequence as if they were
 a unique resource.
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 A URL accepted by this protocol has the syntax:
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 @example
 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
 @end example
 
 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
 resource to be concatenated, each one possibly specifying a distinct
 protocol.
 
 For example to read a sequence of files @file{split1.mpeg},
 @file{split2.mpeg}, @file{split3.mpeg} with @file{ffplay} use the
 command:
 @example
 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 @end example
 
 Note that you may need to escape the character "|" which is special for
 many shells.
 
 @section file
 
 File access protocol.
 
 Allow to read from or read to a file.
 
 For example to read from a file @file{input.mpeg} with @file{ffmpeg}
 use the command:
 @example
 ffmpeg -i file:input.mpeg output.mpeg
 @end example
 
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 The ff* tools default to the file protocol, that is a resource
 specified with the name "FILE.mpeg" is interpreted as the URL
 "file:FILE.mpeg".
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 @section gopher
 
 Gopher protocol.
 
 @section http
 
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 HTTP (Hyper Text Transfer Protocol).
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 @section mmst
 
 MMS (Microsoft Media Server) protocol over TCP.
 
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 @section mmsh
 
 MMS (Microsoft Media Server) protocol over HTTP.
 
 The required syntax is:
 @example
 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
 @end example
 
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 @section md5
 
 MD5 output protocol.
 
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 Computes the MD5 hash of the data to be written, and on close writes
 this to the designated output or stdout if none is specified. It can
 be used to test muxers without writing an actual file.
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 Some examples follow.
 @example
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 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
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 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
 
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 # Write the MD5 hash of the encoded AVI file to stdout.
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 ffmpeg -i input.flv -f avi -y md5:
 @end example
 
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 Note that some formats (typically MOV) require the output protocol to
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 be seekable, so they will fail with the MD5 output protocol.
 
 @section pipe
 
 UNIX pipe access protocol.
 
 Allow to read and write from UNIX pipes.
 
 The accepted syntax is:
 @example
 pipe:[@var{number}]
 @end example
 
 @var{number} is the number corresponding to the file descriptor of the
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 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).  If @var{number}
 is not specified, by default the stdout file descriptor will be used
 for writing, stdin for reading.
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 For example to read from stdin with @file{ffmpeg}:
 @example
 cat test.wav | ffmpeg -i pipe:0
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 # ...this is the same as...
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 cat test.wav | ffmpeg -i pipe:
 @end example
 
 For writing to stdout with @file{ffmpeg}:
 @example
 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
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 # ...this is the same as...
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 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
 @end example
 
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 Note that some formats (typically MOV), require the output protocol to
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 be seekable, so they will fail with the pipe output protocol.
 
 @section rtmp
 
 Real-Time Messaging Protocol.
 
 The Real-Time Messaging Protocol (RTMP) is used for streaming multimeā€
 dia content across a TCP/IP network.
 
 The required syntax is:
 @example
 rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
 @end example
 
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 The accepted parameters are:
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 @table @option
 
 @item server
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 The address of the RTMP server.
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 @item port
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 The number of the TCP port to use (by default is 1935).
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 @item app
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 It is the name of the application to access. It usually corresponds to
 the path where the application is installed on the RTMP server
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 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
 
 @item playpath
 It is the path or name of the resource to play with reference to the
 application specified in @var{app}, may be prefixed by "mp4:".
 
 @end table
 
 For example to read with @file{ffplay} a multimedia resource named
 "sample" from the application "vod" from an RTMP server "myserver":
 @example
 ffplay rtmp://myserver/vod/sample
 @end example
 
 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
 
 Real-Time Messaging Protocol and its variants supported through
 librtmp.
 
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 Requires the presence of the librtmp headers and library during
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 configuration. You need to explicitely configure the build with
 "--enable-librtmp". If enabled this will replace the native RTMP
 protocol.
 
 This protocol provides most client functions and a few server
 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
 variants of these encrypted types (RTMPTE, RTMPTS).
 
 The required syntax is:
 @example
 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
 @end example
 
 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
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 meaning as specified for the RTMP native protocol.
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 @var{options} contains a list of space-separated options of the form
 @var{key}=@var{val}.
 
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 See the librtmp manual page (man 3 librtmp) for more information.
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 For example, to stream a file in real-time to an RTMP server using
 @file{ffmpeg}:
 @example
 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
 @end example
 
 To play the same stream using @file{ffplay}:
 @example
 ffplay "rtmp://myserver/live/mystream live=1"
 @end example
 
 @section rtp
 
 Real-Time Protocol.
 
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 @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.
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 @item filter_src
 Accept packets only from negotiated peer address and port.
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 @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
 
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 @section sap
 
 Session Announcement Protocol (RFC 2974). This is not technically a
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 protocol handler in libavformat, it is a muxer and demuxer.
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 It is used for signalling of RTP streams, by announcing the SDP for the
 streams regularly on a separate port.
 
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 @subsection Muxer
 
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 The syntax for a SAP url given to the muxer is:
 @example
 sap://@var{destination}[:@var{port}][?@var{options}]
 @end example
 
 The RTP packets are sent to @var{destination} on port @var{port},
 or to port 5004 if no port is specified.
 @var{options} is a @code{&}-separated list. The following options
 are supported:
 
 @table @option
 
 @item announce_addr=@var{address}
 Specify the destination IP address for sending the announcements to.
 If omitted, the announcements are sent to the commonly used SAP
 announcement multicast address 224.2.127.254 (sap.mcast.net), or
 ff0e::2:7ffe if @var{destination} is an IPv6 address.
 
 @item announce_port=@var{port}
 Specify the port to send the announcements on, defaults to
 9875 if not specified.
 
 @item ttl=@var{ttl}
 Specify the time to live value for the announcements and RTP packets,
 defaults to 255.
 
 @item same_port=@var{0|1}
 If set to 1, send all RTP streams on the same port pair. If zero (the
 default), all streams are sent on unique ports, with each stream on a
 port 2 numbers higher than the previous.
 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
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 The RTP stack in libavformat for receiving requires all streams to be sent
 on unique ports.
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 @end table
 
 Example command lines follow.
 
 To broadcast a stream on the local subnet, for watching in VLC:
 
 @example
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
 @end example
 
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 Similarly, for watching in ffplay:
 
 @example
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
 @end example
 
 And for watching in ffplay, over IPv6:
 
 @example
 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
 @end example
 
 @subsection Demuxer
 
 The syntax for a SAP url given to the demuxer is:
 @example
 sap://[@var{address}][:@var{port}]
 @end example
 
 @var{address} is the multicast address to listen for announcements on,
 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
 is the port that is listened on, 9875 if omitted.
 
 The demuxers listens for announcements on the given address and port.
 Once an announcement is received, it tries to receive that particular stream.
 
 Example command lines follow.
 
 To play back the first stream announced on the normal SAP multicast address:
 
 @example
 ffplay sap://
 @end example
 
 To play back the first stream announced on one the default IPv6 SAP multicast address:
 
 @example
 ffplay sap://[ff0e::2:7ffe]
 @end example
 
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 @section tcp
 
 Trasmission Control Protocol.
 
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 The required syntax for a TCP url is:
 @example
 tcp://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
 @table @option
 
 @item listen
 Listen for an incoming connection
 
 @example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
 @end example
 
 @end table
 
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 @section udp
 
 User Datagram Protocol.
 
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 The required syntax for a UDP url is:
 @example
 udp://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
 @var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
 Follow the list of supported options.
 
 @table @option
 
 @item buffer_size=@var{size}
 set the UDP buffer size in bytes
 
 @item localport=@var{port}
 override the local UDP port to bind with
 
 @item pkt_size=@var{size}
 set the size in bytes of UDP packets
 
 @item reuse=@var{1|0}
 explicitly allow or disallow reusing UDP sockets
 
 @item ttl=@var{ttl}
 set the time to live value (for multicast only)
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 @item connect=@var{1|0}
 Initialize the UDP socket with @code{connect()}. In this case, the
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 destination address can't be changed with ff_udp_set_remote_url later.
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 If the destination address isn't known at the start, this option can
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 be specified in ff_udp_set_remote_url, too.
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 This allows finding out the source address for the packets with getsockname,
 and makes writes return with AVERROR(ECONNREFUSED) if "destination
 unreachable" is received.
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 For receiving, this gives the benefit of only receiving packets from
 the specified peer address/port.
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 @end table
 
 Some usage examples of the udp protocol with @file{ffmpeg} follow.
 
 To stream over UDP to a remote endpoint:
 @example
 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
 @end example
 
 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
 @example
 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
 @end example
 
 To receive over UDP from a remote endpoint:
 @example
 ffmpeg -i udp://[@var{multicast-address}]:@var{port}
 @end example
 
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 @c man end PROTOCOLS