@chapter Protocols
 @c man begin PROTOCOLS
 

 Protocols are configured elements in FFmpeg which allow to access

 resources which require the use of a particular protocol.
 

 When you configure your FFmpeg 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 bluray
 
 Read BluRay playlist.
 
 The accepted options are:
 @table @option
 
 @item angle
 BluRay angle
 
 @item chapter
 Start chapter (1...N)
 
 @item playlist
 Playlist to read (BDMV/PLAYLIST/?????.mpls)
 
 @end table
 
 Examples:
 
 Read longest playlist from BluRay mounted to /mnt/bluray:
 @example
 bluray:/mnt/bluray
 @end example
 
 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
 @example
 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
 @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 @command{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 @command{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 hls
 
 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.
 The nested protocol is declared by specifying
 "+@var{proto}" after the hls URI scheme name, where @var{proto}
 is either "file" or "http".
 
 @example
 hls+http://host/path/to/remote/resource.m3u8
 hls+file://path/to/local/resource.m3u8
 @end example
 

 Using this protocol is discouraged - the hls demuxer should work
 just as well (if not, please report the issues) and is more complete.
 To use the hls demuxer instead, simply use the direct URLs to the
 m3u8 files.
 

 @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 @command{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 @command{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 multimedia
 content across a TCP/IP network.

 
 The required syntax is:
 @example

 rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@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.). You can override
 the value parsed from the URI through the @code{rtmp_app} option, too.

 
 @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:". You
 can override the value parsed from the URI through the @code{rtmp_playpath}
 option, too.
 

 @item listen
 Act as a server, listening for an incoming connection.
 
 @item timeout
 Maximum time to wait for the incoming connection. Implies listen.

 @end table
 
 Additionally, the following parameters can be set via command line options
 (or in code via @code{AVOption}s):
 @table @option
 
 @item rtmp_app
 Name of application to connect on the RTMP server. This option
 overrides the parameter specified in the URI.
 

 @item rtmp_buffer
 Set the client buffer time in milliseconds. The default is 3000.
 

 @item rtmp_conn
 Extra arbitrary AMF connection parameters, parsed from a string,
 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
 Each value is prefixed by a single character denoting the type,
 B for Boolean, N for number, S for string, O for object, or Z for null,
 followed by a colon. For Booleans the data must be either 0 or 1 for
 FALSE or TRUE, respectively.  Likewise for Objects the data must be 0 or
 1 to end or begin an object, respectively. Data items in subobjects may
 be named, by prefixing the type with 'N' and specifying the name before
 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
 times to construct arbitrary AMF sequences.
 

 @item rtmp_flashver
 Version of the Flash plugin used to run the SWF player. The default
 is LNX 9,0,124,2.
 

 @item rtmp_flush_interval
 Number of packets flushed in the same request (RTMPT only). The default
 is 10.
 

 @item rtmp_live
 Specify that the media is a live stream. No resuming or seeking in
 live streams is possible. The default value is @code{any}, which means the
 subscriber first tries to play the live stream specified in the
 playpath. If a live stream of that name is not found, it plays the
 recorded stream. The other possible values are @code{live} and
 @code{recorded}.
 

 @item rtmp_pageurl
 URL of the web page in which the media was embedded. By default no
 value will be sent.
 

 @item rtmp_playpath
 Stream identifier to play or to publish. This option overrides the
 parameter specified in the URI.
 

 @item rtmp_subscribe
 Name of live stream to subscribe to. By default no value will be sent.
 It is only sent if the option is specified or if rtmp_live
 is set to live.
 

 @item rtmp_swfhash
 SHA256 hash of the decompressed SWF file (32 bytes).
 
 @item rtmp_swfsize
 Size of the decompressed SWF file, required for SWFVerification.
 

 @item rtmp_swfurl
 URL of the SWF player for the media. By default no value will be sent.
 

 @item rtmp_swfverify
 URL to player swf file, compute hash/size automatically.
 

 @item rtmp_tcurl

 URL of the target stream. Defaults to proto://host[:port]/app.

 
 @end table
 

 For example to read with @command{ffplay} a multimedia resource named

 "sample" from the application "vod" from an RTMP server "myserver":
 @example
 ffplay rtmp://myserver/vod/sample
 @end example
 

 @section rtmpe
 
 Encrypted Real-Time Messaging Protocol.
 
 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
 streaming multimedia content within standard cryptographic primitives,
 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
 a pair of RC4 keys.
 

 @section rtmps
 
 Real-Time Messaging Protocol over a secure SSL connection.
 
 The Real-Time Messaging Protocol (RTMPS) is used for streaming
 multimedia content across an encrypted connection.
 

 @section rtmpt
 
 Real-Time Messaging Protocol tunneled through HTTP.
 
 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
 for streaming multimedia content within HTTP requests to traverse
 firewalls.
 

 @section rtmpte
 
 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
 
 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
 is used for streaming multimedia content within HTTP requests to traverse
 firewalls.
 

 @section rtmpts
 
 Real-Time Messaging Protocol tunneled through HTTPS.
 
 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
 for streaming multimedia content within HTTPS requests to traverse
 firewalls.
 

 @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 explicitly 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

 @command{ffmpeg}:

 @example
 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
 @end example
 

 To play the same stream using @command{ffplay}:

 @example
 ffplay "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 @command{ffmpeg}/@command{ffplay} 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.

 @item listen
 Act as a server, listening for an incoming connection.

 @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). This
 can be disabled by setting the maximum demuxing delay to zero (via
 the @code{max_delay} field of AVFormatContext).

 When watching multi-bitrate Real-RTSP streams with @command{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_transport udp rtsp://server/video.mp4

 @end example
 
 To watch a stream tunneled over HTTP:
 
 @example

 ffplay -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
 

 To receive a stream in realtime:
 
 @example

 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}

 @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 @command{ffplay}:

 
 @example
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
 @end example
 

 And for watching in @command{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
 

 @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
 

 @item timeout=@var{microseconds}
 In read mode: if no data arrived in more than this time interval, raise error.
 In write mode: if socket cannot be written in more than this time interval, raise error.
 This also sets timeout on TCP connection establishing.
 

 @example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
 @end example
 
 @end table
 

 @section tls
 
 Transport Layer Security/Secure Sockets Layer
 
 The required syntax for a TLS/SSL url is:
 @example
 tls://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
 @table @option
 
 @item listen
 Act as a server, listening for an incoming connection.
 
 @item cafile=@var{filename}
 Certificate authority file. The file must be in OpenSSL PEM format.
 
 @item cert=@var{filename}
 Certificate file. The file must be in OpenSSL PEM format.
 
 @item key=@var{filename}
 Private key file.
 
 @item verify=@var{0|1}
 Verify the peer's certificate.
 
 @end table
 
 Example command lines:
 
 To create a TLS/SSL server that serves an input stream.
 
 @example
 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
 @end example
 
 To play back a stream from the TLS/SSL server using @command{ffplay}:
 
 @example
 ffplay tls://@var{hostname}:@var{port}
 @end example
 

 @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 &-separated options of the form @var{key}=@var{val}.
 

 In case threading is enabled on the system, a circular buffer is used
 to store the incoming data, which allows to reduce loss of data due to
 UDP socket buffer overruns. The @var{fifo_size} and
 @var{overrun_nonfatal} options are related to this buffer.
 

 The list of supported options follows.

 
 @table @option
 
 @item buffer_size=@var{size}

 Set the UDP socket buffer size in bytes. This is used both for the
 receiving and the sending buffer size.

 
 @item localport=@var{port}

 Override the local UDP port to bind with.

 @item localaddr=@var{addr}
 Choose the local IP address. This is useful e.g. if sending multicast
 and the host has multiple interfaces, where the user can choose
 which interface to send on by specifying the IP address of that interface.
 

 @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.

 
 @item sources=@var{address}[,@var{address}]
 Only receive packets sent to the multicast group from one of the
 specified sender IP addresses.
 
 @item block=@var{address}[,@var{address}]
 Ignore packets sent to the multicast group from the specified
 sender IP addresses.

 
 @item fifo_size=@var{units}
 Set the UDP receiving circular buffer size, expressed as a number of
 packets with size of 188 bytes. If not specified defaults to 7*4096.
 
 @item overrun_nonfatal=@var{1|0}
 Survive in case of UDP receiving circular buffer overrun. Default
 value is 0.

 
 @item timeout=@var{microseconds}
 In read mode: if no data arrived in more than this time interval, raise error.

 @end table
 

 Some usage examples of the UDP protocol with @command{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