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 Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based
57 publish-subscribe communication protocol.
59 FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate
60 AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ.
62 After starting the broker, an FFmpeg client may stream data to the broker using
66 ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@@]hostname[:port]
69 Where hostname and port (default is 5672) is the address of the broker. The
70 client may also set a user/password for authentication. The default for both
73 Muliple subscribers may stream from the broker using the command:
75 ffplay amqp://[[user]:[password]@@]hostname[:port]
78 In RabbitMQ all data published to the broker flows through a specific exchange,
79 and each subscribing client has an assigned queue/buffer. When a packet arrives
80 at an exchange, it may be copied to a client's queue depending on the exchange
81 and routing_key fields.
83 The following options are supported:
88 Sets the exchange to use on the broker. RabbitMQ has several predefined
89 exchanges: "amq.direct" is the default exchange, where the publisher and
90 subscriber must have a matching routing_key; "amq.fanout" is the same as a
91 broadcast operation (i.e. the data is forwarded to all queues on the fanout
92 exchange independent of the routing_key); and "amq.topic" is similar to
93 "amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ
97 Sets the routing key. The default value is "amqp". The routing key is used on
98 the "amq.direct" and "amq.topic" exchanges to decide whether packets are written
99 to the queue of a subscriber.
102 Maximum size of each packet sent/received to the broker. Default is 131072.
103 Minimum is 4096 and max is any large value (representable by an int). When
104 receiving packets, this sets an internal buffer size in FFmpeg. It should be
105 equal to or greater than the size of the published packets to the broker. Otherwise
106 the received message may be truncated causing decoding errors.
108 @item connection_timeout
109 The timeout in seconds during the initial connection to the broker. The
110 default value is rw_timeout, or 5 seconds if rw_timeout is not set.
112 @item delivery_mode @var{mode}
113 Sets the delivery mode of each message sent to broker.
114 The following values are accepted:
117 Delivery mode set to "persistent" (2). This is the default value.
118 Messages may be written to the broker's disk depending on its setup.
121 Delivery mode set to "non-persistent" (1).
122 Messages will stay in broker's memory unless the broker is under memory
131 Asynchronous data filling wrapper for input stream.
133 Fill data in a background thread, to decouple I/O operation from demux thread.
137 async:http://host/resource
138 async:cache:http://host/resource
143 Read BluRay playlist.
145 The accepted options are:
152 Start chapter (1...N)
155 Playlist to read (BDMV/PLAYLIST/?????.mpls)
161 Read longest playlist from BluRay mounted to /mnt/bluray:
166 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
168 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
173 Caching wrapper for input stream.
175 Cache the input stream to temporary file. It brings seeking capability to live streams.
183 Physical concatenation protocol.
185 Read and seek from many resources in sequence as if they were
188 A URL accepted by this protocol has the syntax:
190 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
193 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
194 resource to be concatenated, each one possibly specifying a distinct
197 For example to read a sequence of files @file{split1.mpeg},
198 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
201 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
204 Note that you may need to escape the character "|" which is special for
209 AES-encrypted stream reading protocol.
211 The accepted options are:
214 Set the AES decryption key binary block from given hexadecimal representation.
217 Set the AES decryption initialization vector binary block from given hexadecimal representation.
220 Accepted URL formats:
228 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
230 For example, to convert a GIF file given inline with @command{ffmpeg}:
232 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
237 File access protocol.
239 Read from or write to a file.
241 A file URL can have the form:
246 where @var{filename} is the path of the file to read.
248 An URL that does not have a protocol prefix will be assumed to be a
249 file URL. Depending on the build, an URL that looks like a Windows
250 path with the drive letter at the beginning will also be assumed to be
251 a file URL (usually not the case in builds for unix-like systems).
253 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
256 ffmpeg -i file:input.mpeg output.mpeg
259 This protocol accepts the following options:
263 Truncate existing files on write, if set to 1. A value of 0 prevents
264 truncating. Default value is 1.
267 Set I/O operation maximum block size, in bytes. Default value is
268 @code{INT_MAX}, which results in not limiting the requested block size.
269 Setting this value reasonably low improves user termination request reaction
270 time, which is valuable for files on slow medium.
273 If set to 1, the protocol will retry reading at the end of the file, allowing
274 reading files that still are being written. In order for this to terminate,
275 you either need to use the rw_timeout option, or use the interrupt callback
279 Controls if seekability is advertised on the file. 0 means non-seekable, -1
280 means auto (seekable for normal files, non-seekable for named pipes).
282 Many demuxers handle seekable and non-seekable resources differently,
283 overriding this might speed up opening certain files at the cost of losing some
284 features (e.g. accurate seeking).
289 FTP (File Transfer Protocol).
291 Read from or write to remote resources using FTP protocol.
293 Following syntax is required.
295 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
298 This protocol accepts the following options.
302 Set timeout in microseconds of socket I/O operations used by the underlying low level
303 operation. By default it is set to -1, which means that the timeout is
307 Set a user to be used for authenticating to the FTP server. This is overridden by the
311 Set a password to be used for authenticating to the FTP server. This is overridden by
312 the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
314 @item ftp-anonymous-password
315 Password used when login as anonymous user. Typically an e-mail address
318 @item ftp-write-seekable
319 Control seekability of connection during encoding. If set to 1 the
320 resource is supposed to be seekable, if set to 0 it is assumed not
321 to be seekable. Default value is 0.
324 NOTE: Protocol can be used as output, but it is recommended to not do
325 it, unless special care is taken (tests, customized server configuration
326 etc.). Different FTP servers behave in different way during seek
327 operation. ff* tools may produce incomplete content due to server limitations.
335 Read Apple HTTP Live Streaming compliant segmented stream as
336 a uniform one. The M3U8 playlists describing the segments can be
337 remote HTTP resources or local files, accessed using the standard
339 The nested protocol is declared by specifying
340 "+@var{proto}" after the hls URI scheme name, where @var{proto}
341 is either "file" or "http".
344 hls+http://host/path/to/remote/resource.m3u8
345 hls+file://path/to/local/resource.m3u8
348 Using this protocol is discouraged - the hls demuxer should work
349 just as well (if not, please report the issues) and is more complete.
350 To use the hls demuxer instead, simply use the direct URLs to the
355 HTTP (Hyper Text Transfer Protocol).
357 This protocol accepts the following options:
361 Control seekability of connection. If set to 1 the resource is
362 supposed to be seekable, if set to 0 it is assumed not to be seekable,
363 if set to -1 it will try to autodetect if it is seekable. Default
367 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
370 Set a specific content type for the POST messages or for listen mode.
373 set HTTP proxy to tunnel through e.g. http://example.com:1234
376 Set custom HTTP headers, can override built in default headers. The
377 value must be a string encoding the headers.
379 @item multiple_requests
380 Use persistent connections if set to 1, default is 0.
383 Set custom HTTP post data.
386 Set the Referer header. Include 'Referer: URL' header in HTTP request.
389 Override the User-Agent header. If not specified the protocol will use a
390 string describing the libavformat build. ("Lavf/<version>")
393 This is a deprecated option, you can use user_agent instead it.
396 Set timeout in microseconds of socket I/O operations used by the underlying low level
397 operation. By default it is set to -1, which means that the timeout is
400 @item reconnect_at_eof
401 If set then eof is treated like an error and causes reconnection, this is useful
402 for live / endless streams.
404 @item reconnect_streamed
405 If set then even streamed/non seekable streams will be reconnected on errors.
407 @item reconnect_delay_max
408 Sets the maximum delay in seconds after which to give up reconnecting
411 Export the MIME type.
414 Exports the HTTP response version number. Usually "1.0" or "1.1".
417 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
418 supports this, the metadata has to be retrieved by the application by reading
419 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
422 @item icy_metadata_headers
423 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
424 headers, separated by newline characters.
426 @item icy_metadata_packet
427 If the server supports ICY metadata, and @option{icy} was set to 1, this
428 contains the last non-empty metadata packet sent by the server. It should be
429 polled in regular intervals by applications interested in mid-stream metadata
433 Set the cookies to be sent in future requests. The format of each cookie is the
434 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
435 delimited by a newline character.
438 Set initial byte offset.
441 Try to limit the request to bytes preceding this offset.
444 When used as a client option it sets the HTTP method for the request.
446 When used as a server option it sets the HTTP method that is going to be
447 expected from the client(s).
448 If the expected and the received HTTP method do not match the client will
449 be given a Bad Request response.
450 When unset the HTTP method is not checked for now. This will be replaced by
451 autodetection in the future.
454 If set to 1 enables experimental HTTP server. This can be used to send data when
455 used as an output option, or read data from a client with HTTP POST when used as
457 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
458 in ffmpeg.c and thus must not be used as a command line option.
460 # Server side (sending):
461 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
463 # Client side (receiving):
464 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
466 # Client can also be done with wget:
467 wget http://@var{server}:@var{port} -O somefile.ogg
469 # Server side (receiving):
470 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
472 # Client side (sending):
473 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
475 # Client can also be done with wget:
476 wget --post-file=somefile.ogg http://@var{server}:@var{port}
479 @item send_expect_100
480 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
481 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
486 @subsection HTTP Cookies
488 Some HTTP requests will be denied unless cookie values are passed in with the
489 request. The @option{cookies} option allows these cookies to be specified. At
490 the very least, each cookie must specify a value along with a path and domain.
491 HTTP requests that match both the domain and path will automatically include the
492 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
495 The required syntax to play a stream specifying a cookie is:
497 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
502 Icecast protocol (stream to Icecast servers)
504 This protocol accepts the following options:
508 Set the stream genre.
513 @item ice_description
514 Set the stream description.
517 Set the stream website URL.
520 Set if the stream should be public.
521 The default is 0 (not public).
524 Override the User-Agent header. If not specified a string of the form
525 "Lavf/<version>" will be used.
528 Set the Icecast mountpoint password.
531 Set the stream content type. This must be set if it is different from
535 This enables support for Icecast versions < 2.4.0, that do not support the
536 HTTP PUT method but the SOURCE method.
539 Establish a TLS (HTTPS) connection to Icecast.
544 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
549 MMS (Microsoft Media Server) protocol over TCP.
553 MMS (Microsoft Media Server) protocol over HTTP.
555 The required syntax is:
557 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
564 Computes the MD5 hash of the data to be written, and on close writes
565 this to the designated output or stdout if none is specified. It can
566 be used to test muxers without writing an actual file.
568 Some examples follow.
570 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
571 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
573 # Write the MD5 hash of the encoded AVI file to stdout.
574 ffmpeg -i input.flv -f avi -y md5:
577 Note that some formats (typically MOV) require the output protocol to
578 be seekable, so they will fail with the MD5 output protocol.
582 UNIX pipe access protocol.
584 Read and write from UNIX pipes.
586 The accepted syntax is:
591 @var{number} is the number corresponding to the file descriptor of the
592 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
593 is not specified, by default the stdout file descriptor will be used
594 for writing, stdin for reading.
596 For example to read from stdin with @command{ffmpeg}:
598 cat test.wav | ffmpeg -i pipe:0
599 # ...this is the same as...
600 cat test.wav | ffmpeg -i pipe:
603 For writing to stdout with @command{ffmpeg}:
605 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
606 # ...this is the same as...
607 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
610 This protocol accepts the following options:
614 Set I/O operation maximum block size, in bytes. Default value is
615 @code{INT_MAX}, which results in not limiting the requested block size.
616 Setting this value reasonably low improves user termination request reaction
617 time, which is valuable if data transmission is slow.
620 Note that some formats (typically MOV), require the output protocol to
621 be seekable, so they will fail with the pipe output protocol.
625 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
627 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
628 for MPEG-2 Transport Streams sent over RTP.
630 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
631 the @code{rtp} protocol.
633 The required syntax is:
635 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
638 The destination UDP ports are @code{port + 2} for the column FEC stream
639 and @code{port + 4} for the row FEC stream.
641 This protocol accepts the following options:
645 The number of columns (4-20, LxD <= 100)
648 The number of rows (4-20, LxD <= 100)
655 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
660 Real-Time Messaging Protocol.
662 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
663 content across a TCP/IP network.
665 The required syntax is:
667 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
670 The accepted parameters are:
674 An optional username (mostly for publishing).
677 An optional password (mostly for publishing).
680 The address of the RTMP server.
683 The number of the TCP port to use (by default is 1935).
686 It is the name of the application to access. It usually corresponds to
687 the path where the application is installed on the RTMP server
688 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
689 the value parsed from the URI through the @code{rtmp_app} option, too.
692 It is the path or name of the resource to play with reference to the
693 application specified in @var{app}, may be prefixed by "mp4:". You
694 can override the value parsed from the URI through the @code{rtmp_playpath}
698 Act as a server, listening for an incoming connection.
701 Maximum time to wait for the incoming connection. Implies listen.
704 Additionally, the following parameters can be set via command line options
705 (or in code via @code{AVOption}s):
709 Name of application to connect on the RTMP server. This option
710 overrides the parameter specified in the URI.
713 Set the client buffer time in milliseconds. The default is 3000.
716 Extra arbitrary AMF connection parameters, parsed from a string,
717 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
718 Each value is prefixed by a single character denoting the type,
719 B for Boolean, N for number, S for string, O for object, or Z for null,
720 followed by a colon. For Booleans the data must be either 0 or 1 for
721 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
722 1 to end or begin an object, respectively. Data items in subobjects may
723 be named, by prefixing the type with 'N' and specifying the name before
724 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
725 times to construct arbitrary AMF sequences.
728 Version of the Flash plugin used to run the SWF player. The default
729 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
730 <libavformat version>).)
732 @item rtmp_flush_interval
733 Number of packets flushed in the same request (RTMPT only). The default
737 Specify that the media is a live stream. No resuming or seeking in
738 live streams is possible. The default value is @code{any}, which means the
739 subscriber first tries to play the live stream specified in the
740 playpath. If a live stream of that name is not found, it plays the
741 recorded stream. The other possible values are @code{live} and
745 URL of the web page in which the media was embedded. By default no
749 Stream identifier to play or to publish. This option overrides the
750 parameter specified in the URI.
753 Name of live stream to subscribe to. By default no value will be sent.
754 It is only sent if the option is specified or if rtmp_live
758 SHA256 hash of the decompressed SWF file (32 bytes).
761 Size of the decompressed SWF file, required for SWFVerification.
764 URL of the SWF player for the media. By default no value will be sent.
767 URL to player swf file, compute hash/size automatically.
770 URL of the target stream. Defaults to proto://host[:port]/app.
774 For example to read with @command{ffplay} a multimedia resource named
775 "sample" from the application "vod" from an RTMP server "myserver":
777 ffplay rtmp://myserver/vod/sample
780 To publish to a password protected server, passing the playpath and
781 app names separately:
783 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
788 Encrypted Real-Time Messaging Protocol.
790 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
791 streaming multimedia content within standard cryptographic primitives,
792 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
797 Real-Time Messaging Protocol over a secure SSL connection.
799 The Real-Time Messaging Protocol (RTMPS) is used for streaming
800 multimedia content across an encrypted connection.
804 Real-Time Messaging Protocol tunneled through HTTP.
806 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
807 for streaming multimedia content within HTTP requests to traverse
812 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
814 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
815 is used for streaming multimedia content within HTTP requests to traverse
820 Real-Time Messaging Protocol tunneled through HTTPS.
822 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
823 for streaming multimedia content within HTTPS requests to traverse
826 @section libsmbclient
828 libsmbclient permits one to manipulate CIFS/SMB network resources.
830 Following syntax is required.
833 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
836 This protocol accepts the following options.
840 Set timeout in milliseconds of socket I/O operations used by the underlying
841 low level operation. By default it is set to -1, which means that the timeout
845 Truncate existing files on write, if set to 1. A value of 0 prevents
846 truncating. Default value is 1.
849 Set the workgroup used for making connections. By default workgroup is not specified.
853 For more information see: @url{http://www.samba.org/}.
857 Secure File Transfer Protocol via libssh
859 Read from or write to remote resources using SFTP protocol.
861 Following syntax is required.
864 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
867 This protocol accepts the following options.
871 Set timeout of socket I/O operations used by the underlying low level
872 operation. By default it is set to -1, which means that the timeout
876 Truncate existing files on write, if set to 1. A value of 0 prevents
877 truncating. Default value is 1.
880 Specify the path of the file containing private key to use during authorization.
881 By default libssh searches for keys in the @file{~/.ssh/} directory.
885 Example: Play a file stored on remote server.
888 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
891 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
893 Real-Time Messaging Protocol and its variants supported through
896 Requires the presence of the librtmp headers and library during
897 configuration. You need to explicitly configure the build with
898 "--enable-librtmp". If enabled this will replace the native RTMP
901 This protocol provides most client functions and a few server
902 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
903 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
904 variants of these encrypted types (RTMPTE, RTMPTS).
906 The required syntax is:
908 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
911 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
912 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
913 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
914 meaning as specified for the RTMP native protocol.
915 @var{options} contains a list of space-separated options of the form
918 See the librtmp manual page (man 3 librtmp) for more information.
920 For example, to stream a file in real-time to an RTMP server using
923 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
926 To play the same stream using @command{ffplay}:
928 ffplay "rtmp://myserver/live/mystream live=1"
933 Real-time Transport Protocol.
935 The required syntax for an RTP URL is:
936 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
938 @var{port} specifies the RTP port to use.
940 The following URL options are supported:
945 Set the TTL (Time-To-Live) value (for multicast only).
947 @item rtcpport=@var{n}
948 Set the remote RTCP port to @var{n}.
950 @item localrtpport=@var{n}
951 Set the local RTP port to @var{n}.
953 @item localrtcpport=@var{n}'
954 Set the local RTCP port to @var{n}.
956 @item pkt_size=@var{n}
957 Set max packet size (in bytes) to @var{n}.
960 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
963 @item sources=@var{ip}[,@var{ip}]
964 List allowed source IP addresses.
966 @item block=@var{ip}[,@var{ip}]
967 List disallowed (blocked) source IP addresses.
969 @item write_to_source=0|1
970 Send packets to the source address of the latest received packet (if
971 set to 1) or to a default remote address (if set to 0).
973 @item localport=@var{n}
974 Set the local RTP port to @var{n}.
976 This is a deprecated option. Instead, @option{localrtpport} should be
986 If @option{rtcpport} is not set the RTCP port will be set to the RTP
990 If @option{localrtpport} (the local RTP port) is not set any available
991 port will be used for the local RTP and RTCP ports.
994 If @option{localrtcpport} (the local RTCP port) is not set it will be
995 set to the local RTP port value plus 1.
1000 Real-Time Streaming Protocol.
1002 RTSP is not technically a protocol handler in libavformat, it is a demuxer
1003 and muxer. The demuxer supports both normal RTSP (with data transferred
1004 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
1005 data transferred over RDT).
1007 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1008 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
1009 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
1011 The required syntax for a RTSP url is:
1013 rtsp://@var{hostname}[:@var{port}]/@var{path}
1016 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1017 line, or set in code via @code{AVOption}s or in
1018 @code{avformat_open_input}.
1020 The following options are supported.
1024 Do not start playing the stream immediately if set to 1. Default value
1027 @item rtsp_transport
1028 Set RTSP transport protocols.
1030 It accepts the following values:
1033 Use UDP as lower transport protocol.
1036 Use TCP (interleaving within the RTSP control channel) as lower
1040 Use UDP multicast as lower transport protocol.
1043 Use HTTP tunneling as lower transport protocol, which is useful for
1047 Multiple lower transport protocols may be specified, in that case they are
1048 tried one at a time (if the setup of one fails, the next one is tried).
1049 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
1054 The following values are accepted:
1057 Accept packets only from negotiated peer address and port.
1059 Act as a server, listening for an incoming connection.
1061 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
1064 Default value is @samp{none}.
1066 @item allowed_media_types
1067 Set media types to accept from the server.
1069 The following flags are accepted:
1076 By default it accepts all media types.
1079 Set minimum local UDP port. Default value is 5000.
1082 Set maximum local UDP port. Default value is 65000.
1085 Set maximum timeout (in seconds) to wait for incoming connections.
1087 A value of -1 means infinite (default). This option implies the
1088 @option{rtsp_flags} set to @samp{listen}.
1090 @item reorder_queue_size
1091 Set number of packets to buffer for handling of reordered packets.
1094 Set socket TCP I/O timeout in microseconds.
1097 Override User-Agent header. If not specified, it defaults to the
1098 libavformat identifier string.
1101 When receiving data over UDP, the demuxer tries to reorder received packets
1102 (since they may arrive out of order, or packets may get lost totally). This
1103 can be disabled by setting the maximum demuxing delay to zero (via
1104 the @code{max_delay} field of AVFormatContext).
1106 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1107 streams to display can be chosen with @code{-vst} @var{n} and
1108 @code{-ast} @var{n} for video and audio respectively, and can be switched
1109 on the fly by pressing @code{v} and @code{a}.
1111 @subsection Examples
1113 The following examples all make use of the @command{ffplay} and
1114 @command{ffmpeg} tools.
1118 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1120 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1124 Watch a stream tunneled over HTTP:
1126 ffplay -rtsp_transport http rtsp://server/video.mp4
1130 Send a stream in realtime to a RTSP server, for others to watch:
1132 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1136 Receive a stream in realtime:
1138 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1144 Session Announcement Protocol (RFC 2974). This is not technically a
1145 protocol handler in libavformat, it is a muxer and demuxer.
1146 It is used for signalling of RTP streams, by announcing the SDP for the
1147 streams regularly on a separate port.
1151 The syntax for a SAP url given to the muxer is:
1153 sap://@var{destination}[:@var{port}][?@var{options}]
1156 The RTP packets are sent to @var{destination} on port @var{port},
1157 or to port 5004 if no port is specified.
1158 @var{options} is a @code{&}-separated list. The following options
1163 @item announce_addr=@var{address}
1164 Specify the destination IP address for sending the announcements to.
1165 If omitted, the announcements are sent to the commonly used SAP
1166 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1167 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1169 @item announce_port=@var{port}
1170 Specify the port to send the announcements on, defaults to
1171 9875 if not specified.
1174 Specify the time to live value for the announcements and RTP packets,
1177 @item same_port=@var{0|1}
1178 If set to 1, send all RTP streams on the same port pair. If zero (the
1179 default), all streams are sent on unique ports, with each stream on a
1180 port 2 numbers higher than the previous.
1181 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1182 The RTP stack in libavformat for receiving requires all streams to be sent
1186 Example command lines follow.
1188 To broadcast a stream on the local subnet, for watching in VLC:
1191 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1194 Similarly, for watching in @command{ffplay}:
1197 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1200 And for watching in @command{ffplay}, over IPv6:
1203 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1208 The syntax for a SAP url given to the demuxer is:
1210 sap://[@var{address}][:@var{port}]
1213 @var{address} is the multicast address to listen for announcements on,
1214 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1215 is the port that is listened on, 9875 if omitted.
1217 The demuxers listens for announcements on the given address and port.
1218 Once an announcement is received, it tries to receive that particular stream.
1220 Example command lines follow.
1222 To play back the first stream announced on the normal SAP multicast address:
1228 To play back the first stream announced on one the default IPv6 SAP multicast address:
1231 ffplay sap://[ff0e::2:7ffe]
1236 Stream Control Transmission Protocol.
1238 The accepted URL syntax is:
1240 sctp://@var{host}:@var{port}[?@var{options}]
1243 The protocol accepts the following options:
1246 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1249 Set the maximum number of streams. By default no limit is set.
1254 Haivision Secure Reliable Transport Protocol via libsrt.
1256 The supported syntax for a SRT URL is:
1258 srt://@var{hostname}:@var{port}[?@var{options}]
1261 @var{options} contains a list of &-separated options of the form
1262 @var{key}=@var{val}.
1267 @var{options} srt://@var{hostname}:@var{port}
1270 @var{options} contains a list of '-@var{key} @var{val}'
1273 This protocol accepts the following options.
1276 @item connect_timeout=@var{milliseconds}
1277 Connection timeout; SRT cannot connect for RTT > 1500 msec
1278 (2 handshake exchanges) with the default connect timeout of
1279 3 seconds. This option applies to the caller and rendezvous
1280 connection modes. The connect timeout is 10 times the value
1281 set for the rendezvous mode (which can be used as a
1282 workaround for this connection problem with earlier versions).
1284 @item ffs=@var{bytes}
1285 Flight Flag Size (Window Size), in bytes. FFS is actually an
1286 internal parameter and you should set it to not less than
1287 @option{recv_buffer_size} and @option{mss}. The default value
1288 is relatively large, therefore unless you set a very large receiver buffer,
1289 you do not need to change this option. Default value is 25600.
1291 @item inputbw=@var{bytes/seconds}
1292 Sender nominal input rate, in bytes per seconds. Used along with
1293 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1294 calculate maximum sending rate when recovery packets are sent
1295 along with the main media stream:
1296 @option{inputbw} * (100 + @option{oheadbw}) / 100
1297 if @option{inputbw} is not set while @option{maxbw} is set to
1298 relative (0), the actual input rate is evaluated inside
1299 the library. Default value is 0.
1301 @item iptos=@var{tos}
1302 IP Type of Service. Applies to sender only. Default value is 0xB8.
1304 @item ipttl=@var{ttl}
1305 IP Time To Live. Applies to sender only. Default value is 64.
1307 @item latency=@var{microseconds}
1308 Timestamp-based Packet Delivery Delay.
1309 Used to absorb bursts of missed packet retransmissions.
1310 This flag sets both @option{rcvlatency} and @option{peerlatency}
1311 to the same value. Note that prior to version 1.3.0
1312 this is the only flag to set the latency, however
1313 this is effectively equivalent to setting @option{peerlatency},
1314 when side is sender and @option{rcvlatency}
1315 when side is receiver, and the bidirectional stream
1316 sending is not supported.
1318 @item listen_timeout=@var{microseconds}
1319 Set socket listen timeout.
1321 @item maxbw=@var{bytes/seconds}
1322 Maximum sending bandwidth, in bytes per seconds.
1323 -1 infinite (CSRTCC limit is 30mbps)
1324 0 relative to input rate (see @option{inputbw})
1325 >0 absolute limit value
1326 Default value is 0 (relative)
1328 @item mode=@var{caller|listener|rendezvous}
1330 @option{caller} opens client connection.
1331 @option{listener} starts server to listen for incoming connections.
1332 @option{rendezvous} use Rendez-Vous connection mode.
1333 Default value is caller.
1335 @item mss=@var{bytes}
1336 Maximum Segment Size, in bytes. Used for buffer allocation
1337 and rate calculation using a packet counter assuming fully
1338 filled packets. The smallest MSS between the peers is
1339 used. This is 1500 by default in the overall internet.
1340 This is the maximum size of the UDP packet and can be
1341 only decreased, unless you have some unusual dedicated
1342 network settings. Default value is 1500.
1344 @item nakreport=@var{1|0}
1345 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1346 periodically until a lost packet is retransmitted or
1347 intentionally dropped. Default value is 1.
1349 @item oheadbw=@var{percents}
1350 Recovery bandwidth overhead above input rate, in percents.
1351 See @option{inputbw}. Default value is 25%.
1353 @item passphrase=@var{string}
1354 HaiCrypt Encryption/Decryption Passphrase string, length
1355 from 10 to 79 characters. The passphrase is the shared
1356 secret between the sender and the receiver. It is used
1357 to generate the Key Encrypting Key using PBKDF2
1358 (Password-Based Key Derivation Function). It is used
1359 only if @option{pbkeylen} is non-zero. It is used on
1360 the receiver only if the received data is encrypted.
1361 The configured passphrase cannot be recovered (write-only).
1363 @item enforced_encryption=@var{1|0}
1364 If true, both connection parties must have the same password
1365 set (including empty, that is, with no encryption). If the
1366 password doesn't match or only one side is unencrypted,
1367 the connection is rejected. Default is true.
1369 @item kmrefreshrate=@var{packets}
1370 The number of packets to be transmitted after which the
1371 encryption key is switched to a new key. Default is -1.
1372 -1 means auto (0x1000000 in srt library). The range for
1373 this option is integers in the 0 - @code{INT_MAX}.
1375 @item kmpreannounce=@var{packets}
1376 The interval between when a new encryption key is sent and
1377 when switchover occurs. This value also applies to the
1378 subsequent interval between when switchover occurs and
1379 when the old encryption key is decommissioned. Default is -1.
1380 -1 means auto (0x1000 in srt library). The range for
1381 this option is integers in the 0 - @code{INT_MAX}.
1383 @item payload_size=@var{bytes}
1384 Sets the maximum declared size of a packet transferred
1385 during the single call to the sending function in Live
1386 mode. Use 0 if this value isn't used (which is default in
1388 Default is -1 (automatic), which typically means MPEG-TS;
1389 if you are going to use SRT
1390 to send any different kind of payload, such as, for example,
1391 wrapping a live stream in very small frames, then you can
1392 use a bigger maximum frame size, though not greater than
1395 @item pkt_size=@var{bytes}
1396 Alias for @samp{payload_size}.
1398 @item peerlatency=@var{microseconds}
1399 The latency value (as described in @option{rcvlatency}) that is
1400 set by the sender side as a minimum value for the receiver.
1402 @item pbkeylen=@var{bytes}
1403 Sender encryption key length, in bytes.
1404 Only can be set to 0, 16, 24 and 32.
1405 Enable sender encryption if not 0.
1406 Not required on receiver (set to 0),
1407 key size obtained from sender in HaiCrypt handshake.
1410 @item rcvlatency=@var{microseconds}
1411 The time that should elapse since the moment when the
1412 packet was sent and the moment when it's delivered to
1413 the receiver application in the receiving function.
1414 This time should be a buffer time large enough to cover
1415 the time spent for sending, unexpectedly extended RTT
1416 time, and the time needed to retransmit the lost UDP
1417 packet. The effective latency value will be the maximum
1418 of this options' value and the value of @option{peerlatency}
1419 set by the peer side. Before version 1.3.0 this option
1420 is only available as @option{latency}.
1422 @item recv_buffer_size=@var{bytes}
1423 Set UDP receive buffer size, expressed in bytes.
1425 @item send_buffer_size=@var{bytes}
1426 Set UDP send buffer size, expressed in bytes.
1428 @item timeout=@var{microseconds}
1429 Set raise error timeouts for read, write and connect operations. Note that the
1430 SRT library has internal timeouts which can be controlled separately, the
1431 value set here is only a cap on those.
1433 @item tlpktdrop=@var{1|0}
1434 Too-late Packet Drop. When enabled on receiver, it skips
1435 missing packets that have not been delivered in time and
1436 delivers the following packets to the application when
1437 their time-to-play has come. It also sends a fake ACK to
1438 the sender. When enabled on sender and enabled on the
1439 receiving peer, the sender drops the older packets that
1440 have no chance of being delivered in time. It was
1441 automatically enabled in the sender if the receiver
1444 @item sndbuf=@var{bytes}
1445 Set send buffer size, expressed in bytes.
1447 @item rcvbuf=@var{bytes}
1448 Set receive buffer size, expressed in bytes.
1450 Receive buffer must not be greater than @option{ffs}.
1452 @item lossmaxttl=@var{packets}
1453 The value up to which the Reorder Tolerance may grow. When
1454 Reorder Tolerance is > 0, then packet loss report is delayed
1455 until that number of packets come in. Reorder Tolerance
1456 increases every time a "belated" packet has come, but it
1457 wasn't due to retransmission (that is, when UDP packets tend
1458 to come out of order), with the difference between the latest
1459 sequence and this packet's sequence, and not more than the
1460 value of this option. By default it's 0, which means that this
1461 mechanism is turned off, and the loss report is always sent
1462 immediately upon experiencing a "gap" in sequences.
1465 The minimum SRT version that is required from the peer. A connection
1466 to a peer that does not satisfy the minimum version requirement
1469 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1472 @item streamid=@var{string}
1473 A string limited to 512 characters that can be set on the socket prior
1474 to connecting. This stream ID will be able to be retrieved by the
1475 listener side from the socket that is returned from srt_accept and
1476 was connected by a socket with that set stream ID. SRT does not enforce
1477 any special interpretation of the contents of this string.
1478 This option doesn’t make sense in Rendezvous connection; the result
1479 might be that simply one side will override the value from the other
1480 side and it’s the matter of luck which one would win
1482 @item smoother=@var{live|file}
1483 The type of Smoother used for the transmission for that socket, which
1484 is responsible for the transmission and congestion control. The Smoother
1485 type must be exactly the same on both connecting parties, otherwise
1486 the connection is rejected.
1488 @item messageapi=@var{1|0}
1489 When set, this socket uses the Message API, otherwise it uses Buffer
1490 API. Note that in live mode (see @option{transtype}) there’s only
1491 message API available. In File mode you can chose to use one of two modes:
1493 Stream API (default, when this option is false). In this mode you may
1494 send as many data as you wish with one sending instruction, or even use
1495 dedicated functions that read directly from a file. The internal facility
1496 will take care of any speed and congestion control. When receiving, you
1497 can also receive as many data as desired, the data not extracted will be
1498 waiting for the next call. There is no boundary between data portions in
1501 Message API. In this mode your single sending instruction passes exactly
1502 one piece of data that has boundaries (a message). Contrary to Live mode,
1503 this message may span across multiple UDP packets and the only size
1504 limitation is that it shall fit as a whole in the sending buffer. The
1505 receiver shall use as large buffer as necessary to receive the message,
1506 otherwise the message will not be given up. When the message is not
1507 complete (not all packets received or there was a packet loss) it will
1510 @item transtype=@var{live|file}
1511 Sets the transmission type for the socket, in particular, setting this
1512 option sets multiple other parameters to their default values as required
1513 for a particular transmission type.
1515 live: Set options as for live transmission. In this mode, you should
1516 send by one sending instruction only so many data that fit in one UDP packet,
1517 and limited to the value defined first in @option{payload_size} (1316 is
1518 default in this mode). There is no speed control in this mode, only the
1519 bandwidth control, if configured, in order to not exceed the bandwidth with
1520 the overhead transmission (retransmitted and control packets).
1522 file: Set options as for non-live transmission. See @option{messageapi}
1523 for further explanations
1525 @item linger=@var{seconds}
1526 The number of seconds that the socket waits for unsent data when closing.
1527 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
1528 seconds in file mode). The range for this option is integers in the
1533 For more information see: @url{https://github.com/Haivision/srt}.
1537 Secure Real-time Transport Protocol.
1539 The accepted options are:
1542 @item srtp_out_suite
1543 Select input and output encoding suites.
1547 @item AES_CM_128_HMAC_SHA1_80
1548 @item SRTP_AES128_CM_HMAC_SHA1_80
1549 @item AES_CM_128_HMAC_SHA1_32
1550 @item SRTP_AES128_CM_HMAC_SHA1_32
1553 @item srtp_in_params
1554 @item srtp_out_params
1555 Set input and output encoding parameters, which are expressed by a
1556 base64-encoded representation of a binary block. The first 16 bytes of
1557 this binary block are used as master key, the following 14 bytes are
1558 used as master salt.
1563 Virtually extract a segment of a file or another stream.
1564 The underlying stream must be seekable.
1569 Start offset of the extracted segment, in bytes.
1571 End offset of the extracted segment, in bytes.
1572 If set to 0, extract till end of file.
1577 Extract a chapter from a DVD VOB file (start and end sectors obtained
1578 externally and multiplied by 2048):
1580 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1583 Play an AVI file directly from a TAR archive:
1585 subfile,,start,183241728,end,366490624,,:archive.tar
1588 Play a MPEG-TS file from start offset till end:
1590 subfile,,start,32815239,end,0,,:video.ts
1595 Writes the output to multiple protocols. The individual outputs are separated
1599 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1604 Transmission Control Protocol.
1606 The required syntax for a TCP url is:
1608 tcp://@var{hostname}:@var{port}[?@var{options}]
1611 @var{options} contains a list of &-separated options of the form
1612 @var{key}=@var{val}.
1614 The list of supported options follows.
1617 @item listen=@var{1|0}
1618 Listen for an incoming connection. Default value is 0.
1620 @item timeout=@var{microseconds}
1621 Set raise error timeout, expressed in microseconds.
1623 This option is only relevant in read mode: if no data arrived in more
1624 than this time interval, raise error.
1626 @item listen_timeout=@var{milliseconds}
1627 Set listen timeout, expressed in milliseconds.
1629 @item recv_buffer_size=@var{bytes}
1630 Set receive buffer size, expressed bytes.
1632 @item send_buffer_size=@var{bytes}
1633 Set send buffer size, expressed bytes.
1635 @item tcp_nodelay=@var{1|0}
1636 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1638 @item tcp_mss=@var{bytes}
1639 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1642 The following example shows how to setup a listening TCP connection
1643 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1645 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1646 ffplay tcp://@var{hostname}:@var{port}
1651 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1653 The required syntax for a TLS/SSL url is:
1655 tls://@var{hostname}:@var{port}[?@var{options}]
1658 The following parameters can be set via command line options
1659 (or in code via @code{AVOption}s):
1663 @item ca_file, cafile=@var{filename}
1664 A file containing certificate authority (CA) root certificates to treat
1665 as trusted. If the linked TLS library contains a default this might not
1666 need to be specified for verification to work, but not all libraries and
1667 setups have defaults built in.
1668 The file must be in OpenSSL PEM format.
1670 @item tls_verify=@var{1|0}
1671 If enabled, try to verify the peer that we are communicating with.
1672 Note, if using OpenSSL, this currently only makes sure that the
1673 peer certificate is signed by one of the root certificates in the CA
1674 database, but it does not validate that the certificate actually
1675 matches the host name we are trying to connect to. (With other backends,
1676 the host name is validated as well.)
1678 This is disabled by default since it requires a CA database to be
1679 provided by the caller in many cases.
1681 @item cert_file, cert=@var{filename}
1682 A file containing a certificate to use in the handshake with the peer.
1683 (When operating as server, in listen mode, this is more often required
1684 by the peer, while client certificates only are mandated in certain
1687 @item key_file, key=@var{filename}
1688 A file containing the private key for the certificate.
1690 @item listen=@var{1|0}
1691 If enabled, listen for connections on the provided port, and assume
1692 the server role in the handshake instead of the client role.
1696 Example command lines:
1698 To create a TLS/SSL server that serves an input stream.
1701 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1704 To play back a stream from the TLS/SSL server using @command{ffplay}:
1707 ffplay tls://@var{hostname}:@var{port}
1712 User Datagram Protocol.
1714 The required syntax for an UDP URL is:
1716 udp://@var{hostname}:@var{port}[?@var{options}]
1719 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1721 In case threading is enabled on the system, a circular buffer is used
1722 to store the incoming data, which allows one to reduce loss of data due to
1723 UDP socket buffer overruns. The @var{fifo_size} and
1724 @var{overrun_nonfatal} options are related to this buffer.
1726 The list of supported options follows.
1729 @item buffer_size=@var{size}
1730 Set the UDP maximum socket buffer size in bytes. This is used to set either
1731 the receive or send buffer size, depending on what the socket is used for.
1732 Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
1734 @item bitrate=@var{bitrate}
1735 If set to nonzero, the output will have the specified constant bitrate if the
1736 input has enough packets to sustain it.
1738 @item burst_bits=@var{bits}
1739 When using @var{bitrate} this specifies the maximum number of bits in
1742 @item localport=@var{port}
1743 Override the local UDP port to bind with.
1745 @item localaddr=@var{addr}
1746 Local IP address of a network interface used for sending packets or joining
1749 @item pkt_size=@var{size}
1750 Set the size in bytes of UDP packets.
1752 @item reuse=@var{1|0}
1753 Explicitly allow or disallow reusing UDP sockets.
1756 Set the time to live value (for multicast only).
1758 @item connect=@var{1|0}
1759 Initialize the UDP socket with @code{connect()}. In this case, the
1760 destination address can't be changed with ff_udp_set_remote_url later.
1761 If the destination address isn't known at the start, this option can
1762 be specified in ff_udp_set_remote_url, too.
1763 This allows finding out the source address for the packets with getsockname,
1764 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1765 unreachable" is received.
1766 For receiving, this gives the benefit of only receiving packets from
1767 the specified peer address/port.
1769 @item sources=@var{address}[,@var{address}]
1770 Only receive packets sent from the specified addresses. In case of multicast,
1771 also subscribe to multicast traffic coming from these addresses only.
1773 @item block=@var{address}[,@var{address}]
1774 Ignore packets sent from the specified addresses. In case of multicast, also
1775 exclude the source addresses in the multicast subscription.
1777 @item fifo_size=@var{units}
1778 Set the UDP receiving circular buffer size, expressed as a number of
1779 packets with size of 188 bytes. If not specified defaults to 7*4096.
1781 @item overrun_nonfatal=@var{1|0}
1782 Survive in case of UDP receiving circular buffer overrun. Default
1785 @item timeout=@var{microseconds}
1786 Set raise error timeout, expressed in microseconds.
1788 This option is only relevant in read mode: if no data arrived in more
1789 than this time interval, raise error.
1791 @item broadcast=@var{1|0}
1792 Explicitly allow or disallow UDP broadcasting.
1794 Note that broadcasting may not work properly on networks having
1795 a broadcast storm protection.
1798 @subsection Examples
1802 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1804 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1808 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1809 sized UDP packets, using a large input buffer:
1811 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1815 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1817 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1825 The required syntax for a Unix socket URL is:
1828 unix://@var{filepath}
1831 The following parameters can be set via command line options
1832 (or in code via @code{AVOption}s):
1838 Create the Unix socket in listening mode.
1843 ZeroMQ asynchronous messaging using the libzmq library.
1845 This library supports unicast streaming to multiple clients without relying on
1848 The required syntax for streaming or connecting to a stream is:
1850 zmq:tcp://ip-address:port
1854 Create a localhost stream on port 5555:
1856 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1859 Multiple clients may connect to the stream using:
1861 ffplay zmq:tcp://127.0.0.1:5555
1864 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
1865 The server side binds to a port and publishes data. Clients connect to the
1866 server (via IP address/port) and subscribe to the stream. The order in which
1867 the server and client start generally does not matter.
1869 ffmpeg must be compiled with the --enable-libzmq option to support
1872 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1873 line. The following options are supported:
1878 Forces the maximum packet size for sending/receiving data. The default value is
1879 131,072 bytes. On the server side, this sets the maximum size of sent packets
1880 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
1881 packets. Note that pkt_size on the clients should be equal to or greater than
1882 pkt_size on the server. Otherwise the received message may be truncated causing
1888 @c man end PROTOCOLS