4 Protocols are configured elements in FFmpeg that enable access to
5 resources that require specific protocols.
7 When you configure your FFmpeg build, all the supported protocols are
8 enabled by default. You can list all available ones using the
9 configure option "--list-protocols".
11 You can disable all the protocols using the configure option
12 "--disable-protocols", and selectively enable a protocol using the
13 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
14 particular protocol using the option
15 "--disable-protocol=@var{PROTOCOL}".
17 The option "-protocols" of the ff* tools will display the list of
20 A description of the currently available protocols follows.
24 Asynchronous data filling wrapper for input stream.
26 Fill data in a background thread, to decouple I/O operation from demux thread.
30 async:http://host/resource
31 async:cache:http://host/resource
38 The accepted options are:
48 Playlist to read (BDMV/PLAYLIST/?????.mpls)
54 Read longest playlist from BluRay mounted to /mnt/bluray:
59 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
61 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
66 Caching wrapper for input stream.
68 Cache the input stream to temporary file. It brings seeking capability to live streams.
76 Physical concatenation protocol.
78 Read and seek from many resources in sequence as if they were
81 A URL accepted by this protocol has the syntax:
83 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
86 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
87 resource to be concatenated, each one possibly specifying a distinct
90 For example to read a sequence of files @file{split1.mpeg},
91 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
94 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
97 Note that you may need to escape the character "|" which is special for
102 AES-encrypted stream reading protocol.
104 The accepted options are:
107 Set the AES decryption key binary block from given hexadecimal representation.
110 Set the AES decryption initialization vector binary block from given hexadecimal representation.
113 Accepted URL formats:
121 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
123 For example, to convert a GIF file given inline with @command{ffmpeg}:
125 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
130 File access protocol.
132 Read from or write to a file.
134 A file URL can have the form:
139 where @var{filename} is the path of the file to read.
141 An URL that does not have a protocol prefix will be assumed to be a
142 file URL. Depending on the build, an URL that looks like a Windows
143 path with the drive letter at the beginning will also be assumed to be
144 a file URL (usually not the case in builds for unix-like systems).
146 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
149 ffmpeg -i file:input.mpeg output.mpeg
152 This protocol accepts the following options:
156 Truncate existing files on write, if set to 1. A value of 0 prevents
157 truncating. Default value is 1.
160 Set I/O operation maximum block size, in bytes. Default value is
161 @code{INT_MAX}, which results in not limiting the requested block size.
162 Setting this value reasonably low improves user termination request reaction
163 time, which is valuable for files on slow medium.
168 FTP (File Transfer Protocol).
170 Read from or write to remote resources using FTP protocol.
172 Following syntax is required.
174 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
177 This protocol accepts the following options.
181 Set timeout in microseconds of socket I/O operations used by the underlying low level
182 operation. By default it is set to -1, which means that the timeout is
185 @item ftp-anonymous-password
186 Password used when login as anonymous user. Typically an e-mail address
189 @item ftp-write-seekable
190 Control seekability of connection during encoding. If set to 1 the
191 resource is supposed to be seekable, if set to 0 it is assumed not
192 to be seekable. Default value is 0.
195 NOTE: Protocol can be used as output, but it is recommended to not do
196 it, unless special care is taken (tests, customized server configuration
197 etc.). Different FTP servers behave in different way during seek
198 operation. ff* tools may produce incomplete content due to server limitations.
206 Read Apple HTTP Live Streaming compliant segmented stream as
207 a uniform one. The M3U8 playlists describing the segments can be
208 remote HTTP resources or local files, accessed using the standard
210 The nested protocol is declared by specifying
211 "+@var{proto}" after the hls URI scheme name, where @var{proto}
212 is either "file" or "http".
215 hls+http://host/path/to/remote/resource.m3u8
216 hls+file://path/to/local/resource.m3u8
219 Using this protocol is discouraged - the hls demuxer should work
220 just as well (if not, please report the issues) and is more complete.
221 To use the hls demuxer instead, simply use the direct URLs to the
226 HTTP (Hyper Text Transfer Protocol).
228 This protocol accepts the following options:
232 Control seekability of connection. If set to 1 the resource is
233 supposed to be seekable, if set to 0 it is assumed not to be seekable,
234 if set to -1 it will try to autodetect if it is seekable. Default
238 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
241 Set a specific content type for the POST messages.
244 set HTTP proxy to tunnel through e.g. http://example.com:1234
247 Set custom HTTP headers, can override built in default headers. The
248 value must be a string encoding the headers.
250 @item multiple_requests
251 Use persistent connections if set to 1, default is 0.
254 Set custom HTTP post data.
258 Override the User-Agent header. If not specified the protocol will use a
259 string describing the libavformat build. ("Lavf/<version>")
262 Set timeout in microseconds of socket I/O operations used by the underlying low level
263 operation. By default it is set to -1, which means that the timeout is
266 @item reconnect_at_eof
267 If set then eof is treated like an error and causes reconnection, this is useful
268 for live / endless streams.
270 @item reconnect_streamed
271 If set then even streamed/non seekable streams will be reconnected on errors.
273 @item reconnect_delay_max
274 Sets the maximum delay in seconds after which to give up reconnecting
277 Export the MIME type.
280 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
281 supports this, the metadata has to be retrieved by the application by reading
282 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
285 @item icy_metadata_headers
286 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
287 headers, separated by newline characters.
289 @item icy_metadata_packet
290 If the server supports ICY metadata, and @option{icy} was set to 1, this
291 contains the last non-empty metadata packet sent by the server. It should be
292 polled in regular intervals by applications interested in mid-stream metadata
296 Set the cookies to be sent in future requests. The format of each cookie is the
297 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
298 delimited by a newline character.
301 Set initial byte offset.
304 Try to limit the request to bytes preceding this offset.
307 When used as a client option it sets the HTTP method for the request.
309 When used as a server option it sets the HTTP method that is going to be
310 expected from the client(s).
311 If the expected and the received HTTP method do not match the client will
312 be given a Bad Request response.
313 When unset the HTTP method is not checked for now. This will be replaced by
314 autodetection in the future.
317 If set to 1 enables experimental HTTP server. This can be used to send data when
318 used as an output option, or read data from a client with HTTP POST when used as
320 If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
321 in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
323 # Server side (sending):
324 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
326 # Client side (receiving):
327 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
329 # Client can also be done with wget:
330 wget http://@var{server}:@var{port} -O somefile.ogg
332 # Server side (receiving):
333 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
335 # Client side (sending):
336 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
338 # Client can also be done with wget:
339 wget --post-file=somefile.ogg http://@var{server}:@var{port}
344 @subsection HTTP Cookies
346 Some HTTP requests will be denied unless cookie values are passed in with the
347 request. The @option{cookies} option allows these cookies to be specified. At
348 the very least, each cookie must specify a value along with a path and domain.
349 HTTP requests that match both the domain and path will automatically include the
350 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
353 The required syntax to play a stream specifying a cookie is:
355 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
360 Icecast protocol (stream to Icecast servers)
362 This protocol accepts the following options:
366 Set the stream genre.
371 @item ice_description
372 Set the stream description.
375 Set the stream website URL.
378 Set if the stream should be public.
379 The default is 0 (not public).
382 Override the User-Agent header. If not specified a string of the form
383 "Lavf/<version>" will be used.
386 Set the Icecast mountpoint password.
389 Set the stream content type. This must be set if it is different from
393 This enables support for Icecast versions < 2.4.0, that do not support the
394 HTTP PUT method but the SOURCE method.
399 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
404 MMS (Microsoft Media Server) protocol over TCP.
408 MMS (Microsoft Media Server) protocol over HTTP.
410 The required syntax is:
412 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
419 Computes the MD5 hash of the data to be written, and on close writes
420 this to the designated output or stdout if none is specified. It can
421 be used to test muxers without writing an actual file.
423 Some examples follow.
425 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
426 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
428 # Write the MD5 hash of the encoded AVI file to stdout.
429 ffmpeg -i input.flv -f avi -y md5:
432 Note that some formats (typically MOV) require the output protocol to
433 be seekable, so they will fail with the MD5 output protocol.
437 UNIX pipe access protocol.
439 Read and write from UNIX pipes.
441 The accepted syntax is:
446 @var{number} is the number corresponding to the file descriptor of the
447 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
448 is not specified, by default the stdout file descriptor will be used
449 for writing, stdin for reading.
451 For example to read from stdin with @command{ffmpeg}:
453 cat test.wav | ffmpeg -i pipe:0
454 # ...this is the same as...
455 cat test.wav | ffmpeg -i pipe:
458 For writing to stdout with @command{ffmpeg}:
460 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
461 # ...this is the same as...
462 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
465 This protocol accepts the following options:
469 Set I/O operation maximum block size, in bytes. Default value is
470 @code{INT_MAX}, which results in not limiting the requested block size.
471 Setting this value reasonably low improves user termination request reaction
472 time, which is valuable if data transmission is slow.
475 Note that some formats (typically MOV), require the output protocol to
476 be seekable, so they will fail with the pipe output protocol.
480 Real-Time Messaging Protocol.
482 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
483 content across a TCP/IP network.
485 The required syntax is:
487 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
490 The accepted parameters are:
494 An optional username (mostly for publishing).
497 An optional password (mostly for publishing).
500 The address of the RTMP server.
503 The number of the TCP port to use (by default is 1935).
506 It is the name of the application to access. It usually corresponds to
507 the path where the application is installed on the RTMP server
508 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
509 the value parsed from the URI through the @code{rtmp_app} option, too.
512 It is the path or name of the resource to play with reference to the
513 application specified in @var{app}, may be prefixed by "mp4:". You
514 can override the value parsed from the URI through the @code{rtmp_playpath}
518 Act as a server, listening for an incoming connection.
521 Maximum time to wait for the incoming connection. Implies listen.
524 Additionally, the following parameters can be set via command line options
525 (or in code via @code{AVOption}s):
529 Name of application to connect on the RTMP server. This option
530 overrides the parameter specified in the URI.
533 Set the client buffer time in milliseconds. The default is 3000.
536 Extra arbitrary AMF connection parameters, parsed from a string,
537 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
538 Each value is prefixed by a single character denoting the type,
539 B for Boolean, N for number, S for string, O for object, or Z for null,
540 followed by a colon. For Booleans the data must be either 0 or 1 for
541 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
542 1 to end or begin an object, respectively. Data items in subobjects may
543 be named, by prefixing the type with 'N' and specifying the name before
544 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
545 times to construct arbitrary AMF sequences.
548 Version of the Flash plugin used to run the SWF player. The default
549 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
550 <libavformat version>).)
552 @item rtmp_flush_interval
553 Number of packets flushed in the same request (RTMPT only). The default
557 Specify that the media is a live stream. No resuming or seeking in
558 live streams is possible. The default value is @code{any}, which means the
559 subscriber first tries to play the live stream specified in the
560 playpath. If a live stream of that name is not found, it plays the
561 recorded stream. The other possible values are @code{live} and
565 URL of the web page in which the media was embedded. By default no
569 Stream identifier to play or to publish. This option overrides the
570 parameter specified in the URI.
573 Name of live stream to subscribe to. By default no value will be sent.
574 It is only sent if the option is specified or if rtmp_live
578 SHA256 hash of the decompressed SWF file (32 bytes).
581 Size of the decompressed SWF file, required for SWFVerification.
584 URL of the SWF player for the media. By default no value will be sent.
587 URL to player swf file, compute hash/size automatically.
590 URL of the target stream. Defaults to proto://host[:port]/app.
594 For example to read with @command{ffplay} a multimedia resource named
595 "sample" from the application "vod" from an RTMP server "myserver":
597 ffplay rtmp://myserver/vod/sample
600 To publish to a password protected server, passing the playpath and
601 app names separately:
603 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
608 Encrypted Real-Time Messaging Protocol.
610 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
611 streaming multimedia content within standard cryptographic primitives,
612 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
617 Real-Time Messaging Protocol over a secure SSL connection.
619 The Real-Time Messaging Protocol (RTMPS) is used for streaming
620 multimedia content across an encrypted connection.
624 Real-Time Messaging Protocol tunneled through HTTP.
626 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
627 for streaming multimedia content within HTTP requests to traverse
632 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
634 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
635 is used for streaming multimedia content within HTTP requests to traverse
640 Real-Time Messaging Protocol tunneled through HTTPS.
642 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
643 for streaming multimedia content within HTTPS requests to traverse
646 @section libsmbclient
648 libsmbclient permits one to manipulate CIFS/SMB network resources.
650 Following syntax is required.
653 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
656 This protocol accepts the following options.
660 Set timeout in miliseconds of socket I/O operations used by the underlying
661 low level operation. By default it is set to -1, which means that the timeout
665 Truncate existing files on write, if set to 1. A value of 0 prevents
666 truncating. Default value is 1.
669 Set the workgroup used for making connections. By default workgroup is not specified.
673 For more information see: @url{http://www.samba.org/}.
677 Secure File Transfer Protocol via libssh
679 Read from or write to remote resources using SFTP protocol.
681 Following syntax is required.
684 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
687 This protocol accepts the following options.
691 Set timeout of socket I/O operations used by the underlying low level
692 operation. By default it is set to -1, which means that the timeout
696 Truncate existing files on write, if set to 1. A value of 0 prevents
697 truncating. Default value is 1.
700 Specify the path of the file containing private key to use during authorization.
701 By default libssh searches for keys in the @file{~/.ssh/} directory.
705 Example: Play a file stored on remote server.
708 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
711 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
713 Real-Time Messaging Protocol and its variants supported through
716 Requires the presence of the librtmp headers and library during
717 configuration. You need to explicitly configure the build with
718 "--enable-librtmp". If enabled this will replace the native RTMP
721 This protocol provides most client functions and a few server
722 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
723 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
724 variants of these encrypted types (RTMPTE, RTMPTS).
726 The required syntax is:
728 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
731 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
732 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
733 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
734 meaning as specified for the RTMP native protocol.
735 @var{options} contains a list of space-separated options of the form
738 See the librtmp manual page (man 3 librtmp) for more information.
740 For example, to stream a file in real-time to an RTMP server using
743 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
746 To play the same stream using @command{ffplay}:
748 ffplay "rtmp://myserver/live/mystream live=1"
753 Real-time Transport Protocol.
755 The required syntax for an RTP URL is:
756 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
758 @var{port} specifies the RTP port to use.
760 The following URL options are supported:
765 Set the TTL (Time-To-Live) value (for multicast only).
767 @item rtcpport=@var{n}
768 Set the remote RTCP port to @var{n}.
770 @item localrtpport=@var{n}
771 Set the local RTP port to @var{n}.
773 @item localrtcpport=@var{n}'
774 Set the local RTCP port to @var{n}.
776 @item pkt_size=@var{n}
777 Set max packet size (in bytes) to @var{n}.
780 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
783 @item sources=@var{ip}[,@var{ip}]
784 List allowed source IP addresses.
786 @item block=@var{ip}[,@var{ip}]
787 List disallowed (blocked) source IP addresses.
789 @item write_to_source=0|1
790 Send packets to the source address of the latest received packet (if
791 set to 1) or to a default remote address (if set to 0).
793 @item localport=@var{n}
794 Set the local RTP port to @var{n}.
796 This is a deprecated option. Instead, @option{localrtpport} should be
806 If @option{rtcpport} is not set the RTCP port will be set to the RTP
810 If @option{localrtpport} (the local RTP port) is not set any available
811 port will be used for the local RTP and RTCP ports.
814 If @option{localrtcpport} (the local RTCP port) is not set it will be
815 set to the local RTP port value plus 1.
820 Real-Time Streaming Protocol.
822 RTSP is not technically a protocol handler in libavformat, it is a demuxer
823 and muxer. The demuxer supports both normal RTSP (with data transferred
824 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
825 data transferred over RDT).
827 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
828 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
829 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
831 The required syntax for a RTSP url is:
833 rtsp://@var{hostname}[:@var{port}]/@var{path}
836 Options can be set on the @command{ffmpeg}/@command{ffplay} command
837 line, or set in code via @code{AVOption}s or in
838 @code{avformat_open_input}.
840 The following options are supported.
844 Do not start playing the stream immediately if set to 1. Default value
848 Set RTSP transport protocols.
850 It accepts the following values:
853 Use UDP as lower transport protocol.
856 Use TCP (interleaving within the RTSP control channel) as lower
860 Use UDP multicast as lower transport protocol.
863 Use HTTP tunneling as lower transport protocol, which is useful for
867 Multiple lower transport protocols may be specified, in that case they are
868 tried one at a time (if the setup of one fails, the next one is tried).
869 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
874 The following values are accepted:
877 Accept packets only from negotiated peer address and port.
879 Act as a server, listening for an incoming connection.
881 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
884 Default value is @samp{none}.
886 @item allowed_media_types
887 Set media types to accept from the server.
889 The following flags are accepted:
896 By default it accepts all media types.
899 Set minimum local UDP port. Default value is 5000.
902 Set maximum local UDP port. Default value is 65000.
905 Set maximum timeout (in seconds) to wait for incoming connections.
907 A value of -1 means infinite (default). This option implies the
908 @option{rtsp_flags} set to @samp{listen}.
910 @item reorder_queue_size
911 Set number of packets to buffer for handling of reordered packets.
914 Set socket TCP I/O timeout in microseconds.
917 Override User-Agent header. If not specified, it defaults to the
918 libavformat identifier string.
921 When receiving data over UDP, the demuxer tries to reorder received packets
922 (since they may arrive out of order, or packets may get lost totally). This
923 can be disabled by setting the maximum demuxing delay to zero (via
924 the @code{max_delay} field of AVFormatContext).
926 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
927 streams to display can be chosen with @code{-vst} @var{n} and
928 @code{-ast} @var{n} for video and audio respectively, and can be switched
929 on the fly by pressing @code{v} and @code{a}.
933 The following examples all make use of the @command{ffplay} and
934 @command{ffmpeg} tools.
938 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
940 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
944 Watch a stream tunneled over HTTP:
946 ffplay -rtsp_transport http rtsp://server/video.mp4
950 Send a stream in realtime to a RTSP server, for others to watch:
952 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
956 Receive a stream in realtime:
958 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
964 Session Announcement Protocol (RFC 2974). This is not technically a
965 protocol handler in libavformat, it is a muxer and demuxer.
966 It is used for signalling of RTP streams, by announcing the SDP for the
967 streams regularly on a separate port.
971 The syntax for a SAP url given to the muxer is:
973 sap://@var{destination}[:@var{port}][?@var{options}]
976 The RTP packets are sent to @var{destination} on port @var{port},
977 or to port 5004 if no port is specified.
978 @var{options} is a @code{&}-separated list. The following options
983 @item announce_addr=@var{address}
984 Specify the destination IP address for sending the announcements to.
985 If omitted, the announcements are sent to the commonly used SAP
986 announcement multicast address 224.2.127.254 (sap.mcast.net), or
987 ff0e::2:7ffe if @var{destination} is an IPv6 address.
989 @item announce_port=@var{port}
990 Specify the port to send the announcements on, defaults to
991 9875 if not specified.
994 Specify the time to live value for the announcements and RTP packets,
997 @item same_port=@var{0|1}
998 If set to 1, send all RTP streams on the same port pair. If zero (the
999 default), all streams are sent on unique ports, with each stream on a
1000 port 2 numbers higher than the previous.
1001 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1002 The RTP stack in libavformat for receiving requires all streams to be sent
1006 Example command lines follow.
1008 To broadcast a stream on the local subnet, for watching in VLC:
1011 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1014 Similarly, for watching in @command{ffplay}:
1017 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1020 And for watching in @command{ffplay}, over IPv6:
1023 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1028 The syntax for a SAP url given to the demuxer is:
1030 sap://[@var{address}][:@var{port}]
1033 @var{address} is the multicast address to listen for announcements on,
1034 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1035 is the port that is listened on, 9875 if omitted.
1037 The demuxers listens for announcements on the given address and port.
1038 Once an announcement is received, it tries to receive that particular stream.
1040 Example command lines follow.
1042 To play back the first stream announced on the normal SAP multicast address:
1048 To play back the first stream announced on one the default IPv6 SAP multicast address:
1051 ffplay sap://[ff0e::2:7ffe]
1056 Stream Control Transmission Protocol.
1058 The accepted URL syntax is:
1060 sctp://@var{host}:@var{port}[?@var{options}]
1063 The protocol accepts the following options:
1066 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1069 Set the maximum number of streams. By default no limit is set.
1074 Secure Real-time Transport Protocol.
1076 The accepted options are:
1079 @item srtp_out_suite
1080 Select input and output encoding suites.
1084 @item AES_CM_128_HMAC_SHA1_80
1085 @item SRTP_AES128_CM_HMAC_SHA1_80
1086 @item AES_CM_128_HMAC_SHA1_32
1087 @item SRTP_AES128_CM_HMAC_SHA1_32
1090 @item srtp_in_params
1091 @item srtp_out_params
1092 Set input and output encoding parameters, which are expressed by a
1093 base64-encoded representation of a binary block. The first 16 bytes of
1094 this binary block are used as master key, the following 14 bytes are
1095 used as master salt.
1100 Virtually extract a segment of a file or another stream.
1101 The underlying stream must be seekable.
1106 Start offset of the extracted segment, in bytes.
1108 End offset of the extracted segment, in bytes.
1113 Extract a chapter from a DVD VOB file (start and end sectors obtained
1114 externally and multiplied by 2048):
1116 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1119 Play an AVI file directly from a TAR archive:
1121 subfile,,start,183241728,end,366490624,,:archive.tar
1126 Transmission Control Protocol.
1128 The required syntax for a TCP url is:
1130 tcp://@var{hostname}:@var{port}[?@var{options}]
1133 @var{options} contains a list of &-separated options of the form
1134 @var{key}=@var{val}.
1136 The list of supported options follows.
1139 @item listen=@var{1|0}
1140 Listen for an incoming connection. Default value is 0.
1142 @item timeout=@var{microseconds}
1143 Set raise error timeout, expressed in microseconds.
1145 This option is only relevant in read mode: if no data arrived in more
1146 than this time interval, raise error.
1148 @item listen_timeout=@var{milliseconds}
1149 Set listen timeout, expressed in milliseconds.
1151 @item recv_buffer_size=@var{bytes}
1152 Set receive buffer size, expressed bytes.
1154 @item send_buffer_size=@var{bytes}
1155 Set send buffer size, expressed bytes.
1158 The following example shows how to setup a listening TCP connection
1159 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1161 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1162 ffplay tcp://@var{hostname}:@var{port}
1167 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1169 The required syntax for a TLS/SSL url is:
1171 tls://@var{hostname}:@var{port}[?@var{options}]
1174 The following parameters can be set via command line options
1175 (or in code via @code{AVOption}s):
1179 @item ca_file, cafile=@var{filename}
1180 A file containing certificate authority (CA) root certificates to treat
1181 as trusted. If the linked TLS library contains a default this might not
1182 need to be specified for verification to work, but not all libraries and
1183 setups have defaults built in.
1184 The file must be in OpenSSL PEM format.
1186 @item tls_verify=@var{1|0}
1187 If enabled, try to verify the peer that we are communicating with.
1188 Note, if using OpenSSL, this currently only makes sure that the
1189 peer certificate is signed by one of the root certificates in the CA
1190 database, but it does not validate that the certificate actually
1191 matches the host name we are trying to connect to. (With GnuTLS,
1192 the host name is validated as well.)
1194 This is disabled by default since it requires a CA database to be
1195 provided by the caller in many cases.
1197 @item cert_file, cert=@var{filename}
1198 A file containing a certificate to use in the handshake with the peer.
1199 (When operating as server, in listen mode, this is more often required
1200 by the peer, while client certificates only are mandated in certain
1203 @item key_file, key=@var{filename}
1204 A file containing the private key for the certificate.
1206 @item listen=@var{1|0}
1207 If enabled, listen for connections on the provided port, and assume
1208 the server role in the handshake instead of the client role.
1212 Example command lines:
1214 To create a TLS/SSL server that serves an input stream.
1217 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1220 To play back a stream from the TLS/SSL server using @command{ffplay}:
1223 ffplay tls://@var{hostname}:@var{port}
1228 User Datagram Protocol.
1230 The required syntax for an UDP URL is:
1232 udp://@var{hostname}:@var{port}[?@var{options}]
1235 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1237 In case threading is enabled on the system, a circular buffer is used
1238 to store the incoming data, which allows one to reduce loss of data due to
1239 UDP socket buffer overruns. The @var{fifo_size} and
1240 @var{overrun_nonfatal} options are related to this buffer.
1242 The list of supported options follows.
1245 @item buffer_size=@var{size}
1246 Set the UDP maximum socket buffer size in bytes. This is used to set either
1247 the receive or send buffer size, depending on what the socket is used for.
1248 Default is 64KB. See also @var{fifo_size}.
1250 @item localport=@var{port}
1251 Override the local UDP port to bind with.
1253 @item localaddr=@var{addr}
1254 Choose the local IP address. This is useful e.g. if sending multicast
1255 and the host has multiple interfaces, where the user can choose
1256 which interface to send on by specifying the IP address of that interface.
1258 @item pkt_size=@var{size}
1259 Set the size in bytes of UDP packets.
1261 @item reuse=@var{1|0}
1262 Explicitly allow or disallow reusing UDP sockets.
1265 Set the time to live value (for multicast only).
1267 @item connect=@var{1|0}
1268 Initialize the UDP socket with @code{connect()}. In this case, the
1269 destination address can't be changed with ff_udp_set_remote_url later.
1270 If the destination address isn't known at the start, this option can
1271 be specified in ff_udp_set_remote_url, too.
1272 This allows finding out the source address for the packets with getsockname,
1273 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1274 unreachable" is received.
1275 For receiving, this gives the benefit of only receiving packets from
1276 the specified peer address/port.
1278 @item sources=@var{address}[,@var{address}]
1279 Only receive packets sent to the multicast group from one of the
1280 specified sender IP addresses.
1282 @item block=@var{address}[,@var{address}]
1283 Ignore packets sent to the multicast group from the specified
1284 sender IP addresses.
1286 @item fifo_size=@var{units}
1287 Set the UDP receiving circular buffer size, expressed as a number of
1288 packets with size of 188 bytes. If not specified defaults to 7*4096.
1290 @item overrun_nonfatal=@var{1|0}
1291 Survive in case of UDP receiving circular buffer overrun. Default
1294 @item timeout=@var{microseconds}
1295 Set raise error timeout, expressed in microseconds.
1297 This option is only relevant in read mode: if no data arrived in more
1298 than this time interval, raise error.
1300 @item broadcast=@var{1|0}
1301 Explicitly allow or disallow UDP broadcasting.
1303 Note that broadcasting may not work properly on networks having
1304 a broadcast storm protection.
1307 @subsection Examples
1311 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1313 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1317 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1318 sized UDP packets, using a large input buffer:
1320 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1324 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1326 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1334 The required syntax for a Unix socket URL is:
1337 unix://@var{filepath}
1340 The following parameters can be set via command line options
1341 (or in code via @code{AVOption}s):
1347 Create the Unix socket in listening mode.
1350 @c man end PROTOCOLS