1 @chapter Protocol Options
2 @c man begin PROTOCOL OPTIONS
4 The libavformat library provides some generic global options, which
5 can be set on all the protocols. In addition each protocol may support
6 so-called private options, which are specific for that component.
8 Options may be set by specifying -@var{option} @var{value} in the
9 FFmpeg tools, or by setting the value explicitly in the
10 @code{AVFormatContext} options or using the @file{libavutil/opt.h} API
13 The list of supported options follows:
16 @item protocol_whitelist @var{list} (@emph{input})
17 Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
18 prefixed by "-" are disabled.
19 All protocols are allowed by default but protocols used by an another
20 protocol (nested protocols) are restricted to a per protocol subset.
23 @c man end PROTOCOL OPTIONS
26 @c man begin PROTOCOLS
28 Protocols are configured elements in FFmpeg that enable access to
29 resources that require specific protocols.
31 When you configure your FFmpeg build, all the supported protocols are
32 enabled by default. You can list all available ones using the
33 configure option "--list-protocols".
35 You can disable all the protocols using the configure option
36 "--disable-protocols", and selectively enable a protocol using the
37 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
38 particular protocol using the option
39 "--disable-protocol=@var{PROTOCOL}".
41 The option "-protocols" of the ff* tools will display the list of
44 All protocols accept the following options:
48 Maximum time to wait for (network) read/write operations to complete,
52 A description of the currently available protocols follows.
56 Asynchronous data filling wrapper for input stream.
58 Fill data in a background thread, to decouple I/O operation from demux thread.
62 async:http://host/resource
63 async:cache:http://host/resource
70 The accepted options are:
80 Playlist to read (BDMV/PLAYLIST/?????.mpls)
86 Read longest playlist from BluRay mounted to /mnt/bluray:
91 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
93 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
98 Caching wrapper for input stream.
100 Cache the input stream to temporary file. It brings seeking capability to live streams.
108 Physical concatenation protocol.
110 Read and seek from many resources in sequence as if they were
113 A URL accepted by this protocol has the syntax:
115 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
118 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
119 resource to be concatenated, each one possibly specifying a distinct
122 For example to read a sequence of files @file{split1.mpeg},
123 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
126 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
129 Note that you may need to escape the character "|" which is special for
134 AES-encrypted stream reading protocol.
136 The accepted options are:
139 Set the AES decryption key binary block from given hexadecimal representation.
142 Set the AES decryption initialization vector binary block from given hexadecimal representation.
145 Accepted URL formats:
153 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
155 For example, to convert a GIF file given inline with @command{ffmpeg}:
157 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
162 File access protocol.
164 Read from or write to a file.
166 A file URL can have the form:
171 where @var{filename} is the path of the file to read.
173 An URL that does not have a protocol prefix will be assumed to be a
174 file URL. Depending on the build, an URL that looks like a Windows
175 path with the drive letter at the beginning will also be assumed to be
176 a file URL (usually not the case in builds for unix-like systems).
178 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
181 ffmpeg -i file:input.mpeg output.mpeg
184 This protocol accepts the following options:
188 Truncate existing files on write, if set to 1. A value of 0 prevents
189 truncating. Default value is 1.
192 Set I/O operation maximum block size, in bytes. Default value is
193 @code{INT_MAX}, which results in not limiting the requested block size.
194 Setting this value reasonably low improves user termination request reaction
195 time, which is valuable for files on slow medium.
198 If set to 1, the protocol will retry reading at the end of the file, allowing
199 reading files that still are being written. In order for this to terminate,
200 you either need to use the rw_timeout option, or use the interrupt callback
204 Controls if seekability is advertised on the file. 0 means non-seekable, -1
205 means auto (seekable for normal files, non-seekable for named pipes).
207 Many demuxers handle seekable and non-seekable resources differently,
208 overriding this might speed up opening certain files at the cost of losing some
209 features (e.g. accurate seeking).
214 FTP (File Transfer Protocol).
216 Read from or write to remote resources using FTP protocol.
218 Following syntax is required.
220 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
223 This protocol accepts the following options.
227 Set timeout in microseconds of socket I/O operations used by the underlying low level
228 operation. By default it is set to -1, which means that the timeout is
231 @item ftp-anonymous-password
232 Password used when login as anonymous user. Typically an e-mail address
235 @item ftp-write-seekable
236 Control seekability of connection during encoding. If set to 1 the
237 resource is supposed to be seekable, if set to 0 it is assumed not
238 to be seekable. Default value is 0.
241 NOTE: Protocol can be used as output, but it is recommended to not do
242 it, unless special care is taken (tests, customized server configuration
243 etc.). Different FTP servers behave in different way during seek
244 operation. ff* tools may produce incomplete content due to server limitations.
252 Read Apple HTTP Live Streaming compliant segmented stream as
253 a uniform one. The M3U8 playlists describing the segments can be
254 remote HTTP resources or local files, accessed using the standard
256 The nested protocol is declared by specifying
257 "+@var{proto}" after the hls URI scheme name, where @var{proto}
258 is either "file" or "http".
261 hls+http://host/path/to/remote/resource.m3u8
262 hls+file://path/to/local/resource.m3u8
265 Using this protocol is discouraged - the hls demuxer should work
266 just as well (if not, please report the issues) and is more complete.
267 To use the hls demuxer instead, simply use the direct URLs to the
272 HTTP (Hyper Text Transfer Protocol).
274 This protocol accepts the following options:
278 Control seekability of connection. If set to 1 the resource is
279 supposed to be seekable, if set to 0 it is assumed not to be seekable,
280 if set to -1 it will try to autodetect if it is seekable. Default
284 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
287 Set a specific content type for the POST messages or for listen mode.
290 set HTTP proxy to tunnel through e.g. http://example.com:1234
293 Set custom HTTP headers, can override built in default headers. The
294 value must be a string encoding the headers.
296 @item multiple_requests
297 Use persistent connections if set to 1, default is 0.
300 Set custom HTTP post data.
303 Set the Referer header. Include 'Referer: URL' header in HTTP request.
306 Override the User-Agent header. If not specified the protocol will use a
307 string describing the libavformat build. ("Lavf/<version>")
310 This is a deprecated option, you can use user_agent instead it.
313 Set timeout in microseconds of socket I/O operations used by the underlying low level
314 operation. By default it is set to -1, which means that the timeout is
317 @item reconnect_at_eof
318 If set then eof is treated like an error and causes reconnection, this is useful
319 for live / endless streams.
321 @item reconnect_streamed
322 If set then even streamed/non seekable streams will be reconnected on errors.
324 @item reconnect_delay_max
325 Sets the maximum delay in seconds after which to give up reconnecting
328 Export the MIME type.
331 Exports the HTTP response version number. Usually "1.0" or "1.1".
334 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
335 supports this, the metadata has to be retrieved by the application by reading
336 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
339 @item icy_metadata_headers
340 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
341 headers, separated by newline characters.
343 @item icy_metadata_packet
344 If the server supports ICY metadata, and @option{icy} was set to 1, this
345 contains the last non-empty metadata packet sent by the server. It should be
346 polled in regular intervals by applications interested in mid-stream metadata
350 Set the cookies to be sent in future requests. The format of each cookie is the
351 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
352 delimited by a newline character.
355 Set initial byte offset.
358 Try to limit the request to bytes preceding this offset.
361 When used as a client option it sets the HTTP method for the request.
363 When used as a server option it sets the HTTP method that is going to be
364 expected from the client(s).
365 If the expected and the received HTTP method do not match the client will
366 be given a Bad Request response.
367 When unset the HTTP method is not checked for now. This will be replaced by
368 autodetection in the future.
371 If set to 1 enables experimental HTTP server. This can be used to send data when
372 used as an output option, or read data from a client with HTTP POST when used as
374 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
375 in ffmpeg.c and thus must not be used as a command line option.
377 # Server side (sending):
378 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
380 # Client side (receiving):
381 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
383 # Client can also be done with wget:
384 wget http://@var{server}:@var{port} -O somefile.ogg
386 # Server side (receiving):
387 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
389 # Client side (sending):
390 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
392 # Client can also be done with wget:
393 wget --post-file=somefile.ogg http://@var{server}:@var{port}
396 @item send_expect_100
397 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
398 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
403 @subsection HTTP Cookies
405 Some HTTP requests will be denied unless cookie values are passed in with the
406 request. The @option{cookies} option allows these cookies to be specified. At
407 the very least, each cookie must specify a value along with a path and domain.
408 HTTP requests that match both the domain and path will automatically include the
409 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
412 The required syntax to play a stream specifying a cookie is:
414 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
419 Icecast protocol (stream to Icecast servers)
421 This protocol accepts the following options:
425 Set the stream genre.
430 @item ice_description
431 Set the stream description.
434 Set the stream website URL.
437 Set if the stream should be public.
438 The default is 0 (not public).
441 Override the User-Agent header. If not specified a string of the form
442 "Lavf/<version>" will be used.
445 Set the Icecast mountpoint password.
448 Set the stream content type. This must be set if it is different from
452 This enables support for Icecast versions < 2.4.0, that do not support the
453 HTTP PUT method but the SOURCE method.
458 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
463 MMS (Microsoft Media Server) protocol over TCP.
467 MMS (Microsoft Media Server) protocol over HTTP.
469 The required syntax is:
471 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
478 Computes the MD5 hash of the data to be written, and on close writes
479 this to the designated output or stdout if none is specified. It can
480 be used to test muxers without writing an actual file.
482 Some examples follow.
484 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
485 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
487 # Write the MD5 hash of the encoded AVI file to stdout.
488 ffmpeg -i input.flv -f avi -y md5:
491 Note that some formats (typically MOV) require the output protocol to
492 be seekable, so they will fail with the MD5 output protocol.
496 UNIX pipe access protocol.
498 Read and write from UNIX pipes.
500 The accepted syntax is:
505 @var{number} is the number corresponding to the file descriptor of the
506 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
507 is not specified, by default the stdout file descriptor will be used
508 for writing, stdin for reading.
510 For example to read from stdin with @command{ffmpeg}:
512 cat test.wav | ffmpeg -i pipe:0
513 # ...this is the same as...
514 cat test.wav | ffmpeg -i pipe:
517 For writing to stdout with @command{ffmpeg}:
519 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
520 # ...this is the same as...
521 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
524 This protocol accepts the following options:
528 Set I/O operation maximum block size, in bytes. Default value is
529 @code{INT_MAX}, which results in not limiting the requested block size.
530 Setting this value reasonably low improves user termination request reaction
531 time, which is valuable if data transmission is slow.
534 Note that some formats (typically MOV), require the output protocol to
535 be seekable, so they will fail with the pipe output protocol.
539 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
541 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
542 for MPEG-2 Transport Streams sent over RTP.
544 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
545 the @code{rtp} protocol.
547 The required syntax is:
549 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
552 The destination UDP ports are @code{port + 2} for the column FEC stream
553 and @code{port + 4} for the row FEC stream.
555 This protocol accepts the following options:
559 The number of columns (4-20, LxD <= 100)
562 The number of rows (4-20, LxD <= 100)
569 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
574 Real-Time Messaging Protocol.
576 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
577 content across a TCP/IP network.
579 The required syntax is:
581 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
584 The accepted parameters are:
588 An optional username (mostly for publishing).
591 An optional password (mostly for publishing).
594 The address of the RTMP server.
597 The number of the TCP port to use (by default is 1935).
600 It is the name of the application to access. It usually corresponds to
601 the path where the application is installed on the RTMP server
602 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
603 the value parsed from the URI through the @code{rtmp_app} option, too.
606 It is the path or name of the resource to play with reference to the
607 application specified in @var{app}, may be prefixed by "mp4:". You
608 can override the value parsed from the URI through the @code{rtmp_playpath}
612 Act as a server, listening for an incoming connection.
615 Maximum time to wait for the incoming connection. Implies listen.
618 Additionally, the following parameters can be set via command line options
619 (or in code via @code{AVOption}s):
623 Name of application to connect on the RTMP server. This option
624 overrides the parameter specified in the URI.
627 Set the client buffer time in milliseconds. The default is 3000.
630 Extra arbitrary AMF connection parameters, parsed from a string,
631 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
632 Each value is prefixed by a single character denoting the type,
633 B for Boolean, N for number, S for string, O for object, or Z for null,
634 followed by a colon. For Booleans the data must be either 0 or 1 for
635 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
636 1 to end or begin an object, respectively. Data items in subobjects may
637 be named, by prefixing the type with 'N' and specifying the name before
638 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
639 times to construct arbitrary AMF sequences.
642 Version of the Flash plugin used to run the SWF player. The default
643 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
644 <libavformat version>).)
646 @item rtmp_flush_interval
647 Number of packets flushed in the same request (RTMPT only). The default
651 Specify that the media is a live stream. No resuming or seeking in
652 live streams is possible. The default value is @code{any}, which means the
653 subscriber first tries to play the live stream specified in the
654 playpath. If a live stream of that name is not found, it plays the
655 recorded stream. The other possible values are @code{live} and
659 URL of the web page in which the media was embedded. By default no
663 Stream identifier to play or to publish. This option overrides the
664 parameter specified in the URI.
667 Name of live stream to subscribe to. By default no value will be sent.
668 It is only sent if the option is specified or if rtmp_live
672 SHA256 hash of the decompressed SWF file (32 bytes).
675 Size of the decompressed SWF file, required for SWFVerification.
678 URL of the SWF player for the media. By default no value will be sent.
681 URL to player swf file, compute hash/size automatically.
684 URL of the target stream. Defaults to proto://host[:port]/app.
688 For example to read with @command{ffplay} a multimedia resource named
689 "sample" from the application "vod" from an RTMP server "myserver":
691 ffplay rtmp://myserver/vod/sample
694 To publish to a password protected server, passing the playpath and
695 app names separately:
697 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
702 Encrypted Real-Time Messaging Protocol.
704 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
705 streaming multimedia content within standard cryptographic primitives,
706 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
711 Real-Time Messaging Protocol over a secure SSL connection.
713 The Real-Time Messaging Protocol (RTMPS) is used for streaming
714 multimedia content across an encrypted connection.
718 Real-Time Messaging Protocol tunneled through HTTP.
720 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
721 for streaming multimedia content within HTTP requests to traverse
726 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
728 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
729 is used for streaming multimedia content within HTTP requests to traverse
734 Real-Time Messaging Protocol tunneled through HTTPS.
736 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
737 for streaming multimedia content within HTTPS requests to traverse
740 @section libsmbclient
742 libsmbclient permits one to manipulate CIFS/SMB network resources.
744 Following syntax is required.
747 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
750 This protocol accepts the following options.
754 Set timeout in milliseconds of socket I/O operations used by the underlying
755 low level operation. By default it is set to -1, which means that the timeout
759 Truncate existing files on write, if set to 1. A value of 0 prevents
760 truncating. Default value is 1.
763 Set the workgroup used for making connections. By default workgroup is not specified.
767 For more information see: @url{http://www.samba.org/}.
771 Secure File Transfer Protocol via libssh
773 Read from or write to remote resources using SFTP protocol.
775 Following syntax is required.
778 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
781 This protocol accepts the following options.
785 Set timeout of socket I/O operations used by the underlying low level
786 operation. By default it is set to -1, which means that the timeout
790 Truncate existing files on write, if set to 1. A value of 0 prevents
791 truncating. Default value is 1.
794 Specify the path of the file containing private key to use during authorization.
795 By default libssh searches for keys in the @file{~/.ssh/} directory.
799 Example: Play a file stored on remote server.
802 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
805 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
807 Real-Time Messaging Protocol and its variants supported through
810 Requires the presence of the librtmp headers and library during
811 configuration. You need to explicitly configure the build with
812 "--enable-librtmp". If enabled this will replace the native RTMP
815 This protocol provides most client functions and a few server
816 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
817 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
818 variants of these encrypted types (RTMPTE, RTMPTS).
820 The required syntax is:
822 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
825 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
826 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
827 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
828 meaning as specified for the RTMP native protocol.
829 @var{options} contains a list of space-separated options of the form
832 See the librtmp manual page (man 3 librtmp) for more information.
834 For example, to stream a file in real-time to an RTMP server using
837 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
840 To play the same stream using @command{ffplay}:
842 ffplay "rtmp://myserver/live/mystream live=1"
847 Real-time Transport Protocol.
849 The required syntax for an RTP URL is:
850 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
852 @var{port} specifies the RTP port to use.
854 The following URL options are supported:
859 Set the TTL (Time-To-Live) value (for multicast only).
861 @item rtcpport=@var{n}
862 Set the remote RTCP port to @var{n}.
864 @item localrtpport=@var{n}
865 Set the local RTP port to @var{n}.
867 @item localrtcpport=@var{n}'
868 Set the local RTCP port to @var{n}.
870 @item pkt_size=@var{n}
871 Set max packet size (in bytes) to @var{n}.
874 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
877 @item sources=@var{ip}[,@var{ip}]
878 List allowed source IP addresses.
880 @item block=@var{ip}[,@var{ip}]
881 List disallowed (blocked) source IP addresses.
883 @item write_to_source=0|1
884 Send packets to the source address of the latest received packet (if
885 set to 1) or to a default remote address (if set to 0).
887 @item localport=@var{n}
888 Set the local RTP port to @var{n}.
890 This is a deprecated option. Instead, @option{localrtpport} should be
900 If @option{rtcpport} is not set the RTCP port will be set to the RTP
904 If @option{localrtpport} (the local RTP port) is not set any available
905 port will be used for the local RTP and RTCP ports.
908 If @option{localrtcpport} (the local RTCP port) is not set it will be
909 set to the local RTP port value plus 1.
914 Real-Time Streaming Protocol.
916 RTSP is not technically a protocol handler in libavformat, it is a demuxer
917 and muxer. The demuxer supports both normal RTSP (with data transferred
918 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
919 data transferred over RDT).
921 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
922 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
923 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
925 The required syntax for a RTSP url is:
927 rtsp://@var{hostname}[:@var{port}]/@var{path}
930 Options can be set on the @command{ffmpeg}/@command{ffplay} command
931 line, or set in code via @code{AVOption}s or in
932 @code{avformat_open_input}.
934 The following options are supported.
938 Do not start playing the stream immediately if set to 1. Default value
942 Set RTSP transport protocols.
944 It accepts the following values:
947 Use UDP as lower transport protocol.
950 Use TCP (interleaving within the RTSP control channel) as lower
954 Use UDP multicast as lower transport protocol.
957 Use HTTP tunneling as lower transport protocol, which is useful for
961 Multiple lower transport protocols may be specified, in that case they are
962 tried one at a time (if the setup of one fails, the next one is tried).
963 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
968 The following values are accepted:
971 Accept packets only from negotiated peer address and port.
973 Act as a server, listening for an incoming connection.
975 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
978 Default value is @samp{none}.
980 @item allowed_media_types
981 Set media types to accept from the server.
983 The following flags are accepted:
990 By default it accepts all media types.
993 Set minimum local UDP port. Default value is 5000.
996 Set maximum local UDP port. Default value is 65000.
999 Set maximum timeout (in seconds) to wait for incoming connections.
1001 A value of -1 means infinite (default). This option implies the
1002 @option{rtsp_flags} set to @samp{listen}.
1004 @item reorder_queue_size
1005 Set number of packets to buffer for handling of reordered packets.
1008 Set socket TCP I/O timeout in microseconds.
1011 Override User-Agent header. If not specified, it defaults to the
1012 libavformat identifier string.
1015 When receiving data over UDP, the demuxer tries to reorder received packets
1016 (since they may arrive out of order, or packets may get lost totally). This
1017 can be disabled by setting the maximum demuxing delay to zero (via
1018 the @code{max_delay} field of AVFormatContext).
1020 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1021 streams to display can be chosen with @code{-vst} @var{n} and
1022 @code{-ast} @var{n} for video and audio respectively, and can be switched
1023 on the fly by pressing @code{v} and @code{a}.
1025 @subsection Examples
1027 The following examples all make use of the @command{ffplay} and
1028 @command{ffmpeg} tools.
1032 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1034 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1038 Watch a stream tunneled over HTTP:
1040 ffplay -rtsp_transport http rtsp://server/video.mp4
1044 Send a stream in realtime to a RTSP server, for others to watch:
1046 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1050 Receive a stream in realtime:
1052 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1058 Session Announcement Protocol (RFC 2974). This is not technically a
1059 protocol handler in libavformat, it is a muxer and demuxer.
1060 It is used for signalling of RTP streams, by announcing the SDP for the
1061 streams regularly on a separate port.
1065 The syntax for a SAP url given to the muxer is:
1067 sap://@var{destination}[:@var{port}][?@var{options}]
1070 The RTP packets are sent to @var{destination} on port @var{port},
1071 or to port 5004 if no port is specified.
1072 @var{options} is a @code{&}-separated list. The following options
1077 @item announce_addr=@var{address}
1078 Specify the destination IP address for sending the announcements to.
1079 If omitted, the announcements are sent to the commonly used SAP
1080 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1081 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1083 @item announce_port=@var{port}
1084 Specify the port to send the announcements on, defaults to
1085 9875 if not specified.
1088 Specify the time to live value for the announcements and RTP packets,
1091 @item same_port=@var{0|1}
1092 If set to 1, send all RTP streams on the same port pair. If zero (the
1093 default), all streams are sent on unique ports, with each stream on a
1094 port 2 numbers higher than the previous.
1095 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1096 The RTP stack in libavformat for receiving requires all streams to be sent
1100 Example command lines follow.
1102 To broadcast a stream on the local subnet, for watching in VLC:
1105 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1108 Similarly, for watching in @command{ffplay}:
1111 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1114 And for watching in @command{ffplay}, over IPv6:
1117 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1122 The syntax for a SAP url given to the demuxer is:
1124 sap://[@var{address}][:@var{port}]
1127 @var{address} is the multicast address to listen for announcements on,
1128 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1129 is the port that is listened on, 9875 if omitted.
1131 The demuxers listens for announcements on the given address and port.
1132 Once an announcement is received, it tries to receive that particular stream.
1134 Example command lines follow.
1136 To play back the first stream announced on the normal SAP multicast address:
1142 To play back the first stream announced on one the default IPv6 SAP multicast address:
1145 ffplay sap://[ff0e::2:7ffe]
1150 Stream Control Transmission Protocol.
1152 The accepted URL syntax is:
1154 sctp://@var{host}:@var{port}[?@var{options}]
1157 The protocol accepts the following options:
1160 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1163 Set the maximum number of streams. By default no limit is set.
1168 Haivision Secure Reliable Transport Protocol via libsrt.
1170 The supported syntax for a SRT URL is:
1172 srt://@var{hostname}:@var{port}[?@var{options}]
1175 @var{options} contains a list of &-separated options of the form
1176 @var{key}=@var{val}.
1181 @var{options} srt://@var{hostname}:@var{port}
1184 @var{options} contains a list of '-@var{key} @var{val}'
1187 This protocol accepts the following options.
1190 @item connect_timeout
1191 Connection timeout; SRT cannot connect for RTT > 1500 msec
1192 (2 handshake exchanges) with the default connect timeout of
1193 3 seconds. This option applies to the caller and rendezvous
1194 connection modes. The connect timeout is 10 times the value
1195 set for the rendezvous mode (which can be used as a
1196 workaround for this connection problem with earlier versions).
1198 @item ffs=@var{bytes}
1199 Flight Flag Size (Window Size), in bytes. FFS is actually an
1200 internal parameter and you should set it to not less than
1201 @option{recv_buffer_size} and @option{mss}. The default value
1202 is relatively large, therefore unless you set a very large receiver buffer,
1203 you do not need to change this option. Default value is 25600.
1205 @item inputbw=@var{bytes/seconds}
1206 Sender nominal input rate, in bytes per seconds. Used along with
1207 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1208 calculate maximum sending rate when recovery packets are sent
1209 along with the main media stream:
1210 @option{inputbw} * (100 + @option{oheadbw}) / 100
1211 if @option{inputbw} is not set while @option{maxbw} is set to
1212 relative (0), the actual input rate is evaluated inside
1213 the library. Default value is 0.
1215 @item iptos=@var{tos}
1216 IP Type of Service. Applies to sender only. Default value is 0xB8.
1218 @item ipttl=@var{ttl}
1219 IP Time To Live. Applies to sender only. Default value is 64.
1222 Timestamp-based Packet Delivery Delay.
1223 Used to absorb bursts of missed packet retransmissions.
1224 This flag sets both @option{rcvlatency} and @option{peerlatency}
1225 to the same value. Note that prior to version 1.3.0
1226 this is the only flag to set the latency, however
1227 this is effectively equivalent to setting @option{peerlatency},
1228 when side is sender and @option{rcvlatency}
1229 when side is receiver, and the bidirectional stream
1230 sending is not supported.
1232 @item listen_timeout
1233 Set socket listen timeout.
1235 @item maxbw=@var{bytes/seconds}
1236 Maximum sending bandwidth, in bytes per seconds.
1237 -1 infinite (CSRTCC limit is 30mbps)
1238 0 relative to input rate (see @option{inputbw})
1239 >0 absolute limit value
1240 Default value is 0 (relative)
1242 @item mode=@var{caller|listener|rendezvous}
1244 @option{caller} opens client connection.
1245 @option{listener} starts server to listen for incoming connections.
1246 @option{rendezvous} use Rendez-Vous connection mode.
1247 Default value is caller.
1249 @item mss=@var{bytes}
1250 Maximum Segment Size, in bytes. Used for buffer allocation
1251 and rate calculation using a packet counter assuming fully
1252 filled packets. The smallest MSS between the peers is
1253 used. This is 1500 by default in the overall internet.
1254 This is the maximum size of the UDP packet and can be
1255 only decreased, unless you have some unusual dedicated
1256 network settings. Default value is 1500.
1258 @item nakreport=@var{1|0}
1259 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1260 periodically until a lost packet is retransmitted or
1261 intentionally dropped. Default value is 1.
1263 @item oheadbw=@var{percents}
1264 Recovery bandwidth overhead above input rate, in percents.
1265 See @option{inputbw}. Default value is 25%.
1267 @item passphrase=@var{string}
1268 HaiCrypt Encryption/Decryption Passphrase string, length
1269 from 10 to 79 characters. The passphrase is the shared
1270 secret between the sender and the receiver. It is used
1271 to generate the Key Encrypting Key using PBKDF2
1272 (Password-Based Key Derivation Function). It is used
1273 only if @option{pbkeylen} is non-zero. It is used on
1274 the receiver only if the received data is encrypted.
1275 The configured passphrase cannot be recovered (write-only).
1277 @item payload_size=@var{bytes}
1278 Sets the maximum declared size of a packet transferred
1279 during the single call to the sending function in Live
1280 mode. Use 0 if this value isn't used (which is default in
1282 Default is -1 (automatic), which typically means MPEG-TS;
1283 if you are going to use SRT
1284 to send any different kind of payload, such as, for example,
1285 wrapping a live stream in very small frames, then you can
1286 use a bigger maximum frame size, though not greater than
1289 @item pkt_size=@var{bytes}
1290 Alias for @samp{payload_size}.
1293 The latency value (as described in @option{rcvlatency}) that is
1294 set by the sender side as a minimum value for the receiver.
1296 @item pbkeylen=@var{bytes}
1297 Sender encryption key length, in bytes.
1298 Only can be set to 0, 16, 24 and 32.
1299 Enable sender encryption if not 0.
1300 Not required on receiver (set to 0),
1301 key size obtained from sender in HaiCrypt handshake.
1305 The time that should elapse since the moment when the
1306 packet was sent and the moment when it's delivered to
1307 the receiver application in the receiving function.
1308 This time should be a buffer time large enough to cover
1309 the time spent for sending, unexpectedly extended RTT
1310 time, and the time needed to retransmit the lost UDP
1311 packet. The effective latency value will be the maximum
1312 of this options' value and the value of @option{peerlatency}
1313 set by the peer side. Before version 1.3.0 this option
1314 is only available as @option{latency}.
1316 @item recv_buffer_size=@var{bytes}
1317 Set UDP receive buffer size, expressed in bytes.
1319 @item send_buffer_size=@var{bytes}
1320 Set UDP send buffer size, expressed in bytes.
1323 Set raise error timeout for read/write optations.
1325 This option is only relevant in read mode:
1326 if no data arrived in more than this time
1327 interval, raise error.
1329 @item tlpktdrop=@var{1|0}
1330 Too-late Packet Drop. When enabled on receiver, it skips
1331 missing packets that have not been delivered in time and
1332 delivers the following packets to the application when
1333 their time-to-play has come. It also sends a fake ACK to
1334 the sender. When enabled on sender and enabled on the
1335 receiving peer, the sender drops the older packets that
1336 have no chance of being delivered in time. It was
1337 automatically enabled in the sender if the receiver
1340 @item sndbuf=@var{bytes}
1341 Set send buffer size, expressed in bytes.
1343 @item rcvbuf=@var{bytes}
1344 Set receive buffer size, expressed in bytes.
1346 Receive buffer must not be greater than @option{ffs}.
1348 @item lossmaxttl=@var{packets}
1349 The value up to which the Reorder Tolerance may grow. When
1350 Reorder Tolerance is > 0, then packet loss report is delayed
1351 until that number of packets come in. Reorder Tolerance
1352 increases every time a "belated" packet has come, but it
1353 wasn't due to retransmission (that is, when UDP packets tend
1354 to come out of order), with the difference between the latest
1355 sequence and this packet's sequence, and not more than the
1356 value of this option. By default it's 0, which means that this
1357 mechanism is turned off, and the loss report is always sent
1358 immediately upon experiencing a "gap" in sequences.
1361 The minimum SRT version that is required from the peer. A connection
1362 to a peer that does not satisfy the minimum version requirement
1365 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1368 @item streamid=@var{string}
1369 A string limited to 512 characters that can be set on the socket prior
1370 to connecting. This stream ID will be able to be retrieved by the
1371 listener side from the socket that is returned from srt_accept and
1372 was connected by a socket with that set stream ID. SRT does not enforce
1373 any special interpretation of the contents of this string.
1374 This option doesn’t make sense in Rendezvous connection; the result
1375 might be that simply one side will override the value from the other
1376 side and it’s the matter of luck which one would win
1378 @item smoother=@var{live|file}
1379 The type of Smoother used for the transmission for that socket, which
1380 is responsible for the transmission and congestion control. The Smoother
1381 type must be exactly the same on both connecting parties, otherwise
1382 the connection is rejected.
1384 @item messageapi=@var{1|0}
1385 When set, this socket uses the Message API, otherwise it uses Buffer
1386 API. Note that in live mode (see @option{transtype}) there’s only
1387 message API available. In File mode you can chose to use one of two modes:
1389 Stream API (default, when this option is false). In this mode you may
1390 send as many data as you wish with one sending instruction, or even use
1391 dedicated functions that read directly from a file. The internal facility
1392 will take care of any speed and congestion control. When receiving, you
1393 can also receive as many data as desired, the data not extracted will be
1394 waiting for the next call. There is no boundary between data portions in
1397 Message API. In this mode your single sending instruction passes exactly
1398 one piece of data that has boundaries (a message). Contrary to Live mode,
1399 this message may span across multiple UDP packets and the only size
1400 limitation is that it shall fit as a whole in the sending buffer. The
1401 receiver shall use as large buffer as necessary to receive the message,
1402 otherwise the message will not be given up. When the message is not
1403 complete (not all packets received or there was a packet loss) it will
1406 @item transtype=@var{live|file}
1407 Sets the transmission type for the socket, in particular, setting this
1408 option sets multiple other parameters to their default values as required
1409 for a particular transmission type.
1411 live: Set options as for live transmission. In this mode, you should
1412 send by one sending instruction only so many data that fit in one UDP packet,
1413 and limited to the value defined first in @option{payload_size} (1316 is
1414 default in this mode). There is no speed control in this mode, only the
1415 bandwidth control, if configured, in order to not exceed the bandwidth with
1416 the overhead transmission (retransmitted and control packets).
1418 file: Set options as for non-live transmission. See @option{messageapi}
1419 for further explanations
1423 For more information see: @url{https://github.com/Haivision/srt}.
1427 Secure Real-time Transport Protocol.
1429 The accepted options are:
1432 @item srtp_out_suite
1433 Select input and output encoding suites.
1437 @item AES_CM_128_HMAC_SHA1_80
1438 @item SRTP_AES128_CM_HMAC_SHA1_80
1439 @item AES_CM_128_HMAC_SHA1_32
1440 @item SRTP_AES128_CM_HMAC_SHA1_32
1443 @item srtp_in_params
1444 @item srtp_out_params
1445 Set input and output encoding parameters, which are expressed by a
1446 base64-encoded representation of a binary block. The first 16 bytes of
1447 this binary block are used as master key, the following 14 bytes are
1448 used as master salt.
1453 Virtually extract a segment of a file or another stream.
1454 The underlying stream must be seekable.
1459 Start offset of the extracted segment, in bytes.
1461 End offset of the extracted segment, in bytes.
1462 If set to 0, extract till end of file.
1467 Extract a chapter from a DVD VOB file (start and end sectors obtained
1468 externally and multiplied by 2048):
1470 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1473 Play an AVI file directly from a TAR archive:
1475 subfile,,start,183241728,end,366490624,,:archive.tar
1478 Play a MPEG-TS file from start offset till end:
1480 subfile,,start,32815239,end,0,,:video.ts
1485 Writes the output to multiple protocols. The individual outputs are separated
1489 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1494 Transmission Control Protocol.
1496 The required syntax for a TCP url is:
1498 tcp://@var{hostname}:@var{port}[?@var{options}]
1501 @var{options} contains a list of &-separated options of the form
1502 @var{key}=@var{val}.
1504 The list of supported options follows.
1507 @item listen=@var{1|0}
1508 Listen for an incoming connection. Default value is 0.
1510 @item timeout=@var{microseconds}
1511 Set raise error timeout, expressed in microseconds.
1513 This option is only relevant in read mode: if no data arrived in more
1514 than this time interval, raise error.
1516 @item listen_timeout=@var{milliseconds}
1517 Set listen timeout, expressed in milliseconds.
1519 @item recv_buffer_size=@var{bytes}
1520 Set receive buffer size, expressed bytes.
1522 @item send_buffer_size=@var{bytes}
1523 Set send buffer size, expressed bytes.
1525 @item tcp_nodelay=@var{1|0}
1526 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1528 @item tcp_mss=@var{bytes}
1529 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1532 The following example shows how to setup a listening TCP connection
1533 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1535 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1536 ffplay tcp://@var{hostname}:@var{port}
1541 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1543 The required syntax for a TLS/SSL url is:
1545 tls://@var{hostname}:@var{port}[?@var{options}]
1548 The following parameters can be set via command line options
1549 (or in code via @code{AVOption}s):
1553 @item ca_file, cafile=@var{filename}
1554 A file containing certificate authority (CA) root certificates to treat
1555 as trusted. If the linked TLS library contains a default this might not
1556 need to be specified for verification to work, but not all libraries and
1557 setups have defaults built in.
1558 The file must be in OpenSSL PEM format.
1560 @item tls_verify=@var{1|0}
1561 If enabled, try to verify the peer that we are communicating with.
1562 Note, if using OpenSSL, this currently only makes sure that the
1563 peer certificate is signed by one of the root certificates in the CA
1564 database, but it does not validate that the certificate actually
1565 matches the host name we are trying to connect to. (With other backends,
1566 the host name is validated as well.)
1568 This is disabled by default since it requires a CA database to be
1569 provided by the caller in many cases.
1571 @item cert_file, cert=@var{filename}
1572 A file containing a certificate to use in the handshake with the peer.
1573 (When operating as server, in listen mode, this is more often required
1574 by the peer, while client certificates only are mandated in certain
1577 @item key_file, key=@var{filename}
1578 A file containing the private key for the certificate.
1580 @item listen=@var{1|0}
1581 If enabled, listen for connections on the provided port, and assume
1582 the server role in the handshake instead of the client role.
1586 Example command lines:
1588 To create a TLS/SSL server that serves an input stream.
1591 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1594 To play back a stream from the TLS/SSL server using @command{ffplay}:
1597 ffplay tls://@var{hostname}:@var{port}
1602 User Datagram Protocol.
1604 The required syntax for an UDP URL is:
1606 udp://@var{hostname}:@var{port}[?@var{options}]
1609 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1611 In case threading is enabled on the system, a circular buffer is used
1612 to store the incoming data, which allows one to reduce loss of data due to
1613 UDP socket buffer overruns. The @var{fifo_size} and
1614 @var{overrun_nonfatal} options are related to this buffer.
1616 The list of supported options follows.
1619 @item buffer_size=@var{size}
1620 Set the UDP maximum socket buffer size in bytes. This is used to set either
1621 the receive or send buffer size, depending on what the socket is used for.
1622 Default is 64KB. See also @var{fifo_size}.
1624 @item bitrate=@var{bitrate}
1625 If set to nonzero, the output will have the specified constant bitrate if the
1626 input has enough packets to sustain it.
1628 @item burst_bits=@var{bits}
1629 When using @var{bitrate} this specifies the maximum number of bits in
1632 @item localport=@var{port}
1633 Override the local UDP port to bind with.
1635 @item localaddr=@var{addr}
1636 Local IP address of a network interface used for sending packets or joining
1639 @item pkt_size=@var{size}
1640 Set the size in bytes of UDP packets.
1642 @item reuse=@var{1|0}
1643 Explicitly allow or disallow reusing UDP sockets.
1646 Set the time to live value (for multicast only).
1648 @item connect=@var{1|0}
1649 Initialize the UDP socket with @code{connect()}. In this case, the
1650 destination address can't be changed with ff_udp_set_remote_url later.
1651 If the destination address isn't known at the start, this option can
1652 be specified in ff_udp_set_remote_url, too.
1653 This allows finding out the source address for the packets with getsockname,
1654 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1655 unreachable" is received.
1656 For receiving, this gives the benefit of only receiving packets from
1657 the specified peer address/port.
1659 @item sources=@var{address}[,@var{address}]
1660 Only receive packets sent from the specified addresses. In case of multicast,
1661 also subscribe to multicast traffic coming from these addresses only.
1663 @item block=@var{address}[,@var{address}]
1664 Ignore packets sent from the specified addresses. In case of multicast, also
1665 exclude the source addresses in the multicast subscription.
1667 @item fifo_size=@var{units}
1668 Set the UDP receiving circular buffer size, expressed as a number of
1669 packets with size of 188 bytes. If not specified defaults to 7*4096.
1671 @item overrun_nonfatal=@var{1|0}
1672 Survive in case of UDP receiving circular buffer overrun. Default
1675 @item timeout=@var{microseconds}
1676 Set raise error timeout, expressed in microseconds.
1678 This option is only relevant in read mode: if no data arrived in more
1679 than this time interval, raise error.
1681 @item broadcast=@var{1|0}
1682 Explicitly allow or disallow UDP broadcasting.
1684 Note that broadcasting may not work properly on networks having
1685 a broadcast storm protection.
1688 @subsection Examples
1692 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1694 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1698 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1699 sized UDP packets, using a large input buffer:
1701 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1705 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1707 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1715 The required syntax for a Unix socket URL is:
1718 unix://@var{filepath}
1721 The following parameters can be set via command line options
1722 (or in code via @code{AVOption}s):
1728 Create the Unix socket in listening mode.
1731 @c man end PROTOCOLS