@chapter Protocols @c man begin PROTOCOLS Protocols are configured elements in Libav which allow to access resources which require the use of a particular protocol. When you configure your Libav build, all the supported protocols are enabled by default. You can list all available ones using the configure option "--list-protocols". 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 supported protocols. A description of the currently available protocols follows. @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. 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". @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 @section concat Physical concatenation protocol. Allow to read and seek from many resource in sequence as if they were a unique resource. A URL accepted by this protocol has the syntax: @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{avplay} use the command: @example avplay 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 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". @section gopher Gopher protocol. @section http HTTP (Hyper Text Transfer Protocol). @section mmst MMS (Microsoft Media Server) protocol over TCP. @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 @section md5 MD5 output protocol. 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. Some examples follow. @example # Write the MD5 hash of the encoded AVI file to the file output.avi.md5. ffmpeg -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. ffmpeg -i input.flv -f avi -y md5: @end example Note that some formats (typically MOV) require the output protocol to 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 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. For example to read from stdin with @file{ffmpeg}: @example cat test.wav | ffmpeg -i pipe:0 # ...this is the same as... 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 # ...this is the same as... ffmpeg -i test.wav -f avi pipe: | cat > test.avi @end example Note that some formats (typically MOV), require the output protocol to 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 The accepted parameters are: @table @option @item server The address of the RTMP server. @item port The number of the TCP port to use (by default is 1935). @item app It is the name of the application to access. It usually corresponds to the path where the application is installed on the RTMP server (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{avplay} a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver": @example avplay rtmp://myserver/vod/sample @end example @section rtmp, rtmpe, rtmps, rtmpt, rtmpte Real-Time Messaging Protocol and its variants supported through librtmp. Requires the presence of the librtmp headers and library during 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 meaning as specified for the RTMP native protocol. @var{options} contains a list of space-separated options of the form @var{key}=@var{val}. See the librtmp manual page (man 3 librtmp) for more information. 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{avplay}: @example avplay "rtmp://myserver/live/mystream live=1" @end example @section rtp 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 @uref{http://github.com/revmischa/rtsp-server, RTSP server}). The required syntax for a RTSP url is: @example rtsp://@var{hostname}[:@var{port}]/@var{path} @end example The following options (set on the @file{avconv}/@file{avplay} command line, or set in code via @code{AVOption}s or in @code{avformat_open_input}), are supported: Flags for @code{rtsp_transport}: @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 udp_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. Flags for @code{rtsp_flags}: @table @option @item filter_src Accept packets only from negotiated peer address and port. @end table 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{avplay}, 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 avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 @end example To watch a stream tunneled over HTTP: @example avplay -rtsp_transport http rtsp://server/video.mp4 @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 sap Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in libavformat, it is a muxer and demuxer. It is used for signalling of RTP streams, by announcing the SDP for the streams regularly on a separate port. @subsection Muxer 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. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports. @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 Similarly, for watching in avplay: @example ffmpeg -re -i @var{input} -f sap sap://224.0.0.255 @end example And for watching in avplay, 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 avplay sap:// @end example To play back the first stream announced on one the default IPv6 SAP multicast address: @example avplay sap://[ff0e::2:7ffe] @end example @section tcp Trasmission Control Protocol. 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 avplay tcp://@var{hostname}:@var{port} @end example @end table @section udp User Datagram Protocol. 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) @item connect=@var{1|0} Initialize the UDP socket with @code{connect()}. In this case, the destination address can't be changed with ff_udp_set_remote_url later. If the destination address isn't known at the start, this option can be specified in ff_udp_set_remote_url, too. This allows finding out the source address for the packets with getsockname, and makes writes return with AVERROR(ECONNREFUSED) if "destination unreachable" is received. For receiving, this gives the benefit of only receiving packets from the specified peer address/port. @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 @c man end PROTOCOLS