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.
116 Asynchronous data filling wrapper for input stream.
118 Fill data in a background thread, to decouple I/O operation from demux thread.
122 async:http://host/resource
123 async:cache:http://host/resource
128 Read BluRay playlist.
130 The accepted options are:
137 Start chapter (1...N)
140 Playlist to read (BDMV/PLAYLIST/?????.mpls)
146 Read longest playlist from BluRay mounted to /mnt/bluray:
151 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
153 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
158 Caching wrapper for input stream.
160 Cache the input stream to temporary file. It brings seeking capability to live streams.
168 Physical concatenation protocol.
170 Read and seek from many resources in sequence as if they were
173 A URL accepted by this protocol has the syntax:
175 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
178 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
179 resource to be concatenated, each one possibly specifying a distinct
182 For example to read a sequence of files @file{split1.mpeg},
183 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
186 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
189 Note that you may need to escape the character "|" which is special for
194 AES-encrypted stream reading protocol.
196 The accepted options are:
199 Set the AES decryption key binary block from given hexadecimal representation.
202 Set the AES decryption initialization vector binary block from given hexadecimal representation.
205 Accepted URL formats:
213 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
215 For example, to convert a GIF file given inline with @command{ffmpeg}:
217 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
222 File access protocol.
224 Read from or write to a file.
226 A file URL can have the form:
231 where @var{filename} is the path of the file to read.
233 An URL that does not have a protocol prefix will be assumed to be a
234 file URL. Depending on the build, an URL that looks like a Windows
235 path with the drive letter at the beginning will also be assumed to be
236 a file URL (usually not the case in builds for unix-like systems).
238 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
241 ffmpeg -i file:input.mpeg output.mpeg
244 This protocol accepts the following options:
248 Truncate existing files on write, if set to 1. A value of 0 prevents
249 truncating. Default value is 1.
252 Set I/O operation maximum block size, in bytes. Default value is
253 @code{INT_MAX}, which results in not limiting the requested block size.
254 Setting this value reasonably low improves user termination request reaction
255 time, which is valuable for files on slow medium.
258 If set to 1, the protocol will retry reading at the end of the file, allowing
259 reading files that still are being written. In order for this to terminate,
260 you either need to use the rw_timeout option, or use the interrupt callback
264 Controls if seekability is advertised on the file. 0 means non-seekable, -1
265 means auto (seekable for normal files, non-seekable for named pipes).
267 Many demuxers handle seekable and non-seekable resources differently,
268 overriding this might speed up opening certain files at the cost of losing some
269 features (e.g. accurate seeking).
274 FTP (File Transfer Protocol).
276 Read from or write to remote resources using FTP protocol.
278 Following syntax is required.
280 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
283 This protocol accepts the following options.
287 Set timeout in microseconds of socket I/O operations used by the underlying low level
288 operation. By default it is set to -1, which means that the timeout is
292 Set a user to be used for authenticating to the FTP server. This is overridden by the
296 Set a password to be used for authenticating to the FTP server. This is overridden by
297 the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
299 @item ftp-anonymous-password
300 Password used when login as anonymous user. Typically an e-mail address
303 @item ftp-write-seekable
304 Control seekability of connection during encoding. If set to 1 the
305 resource is supposed to be seekable, if set to 0 it is assumed not
306 to be seekable. Default value is 0.
309 NOTE: Protocol can be used as output, but it is recommended to not do
310 it, unless special care is taken (tests, customized server configuration
311 etc.). Different FTP servers behave in different way during seek
312 operation. ff* tools may produce incomplete content due to server limitations.
320 Read Apple HTTP Live Streaming compliant segmented stream as
321 a uniform one. The M3U8 playlists describing the segments can be
322 remote HTTP resources or local files, accessed using the standard
324 The nested protocol is declared by specifying
325 "+@var{proto}" after the hls URI scheme name, where @var{proto}
326 is either "file" or "http".
329 hls+http://host/path/to/remote/resource.m3u8
330 hls+file://path/to/local/resource.m3u8
333 Using this protocol is discouraged - the hls demuxer should work
334 just as well (if not, please report the issues) and is more complete.
335 To use the hls demuxer instead, simply use the direct URLs to the
340 HTTP (Hyper Text Transfer Protocol).
342 This protocol accepts the following options:
346 Control seekability of connection. If set to 1 the resource is
347 supposed to be seekable, if set to 0 it is assumed not to be seekable,
348 if set to -1 it will try to autodetect if it is seekable. Default
352 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
355 Set a specific content type for the POST messages or for listen mode.
358 set HTTP proxy to tunnel through e.g. http://example.com:1234
361 Set custom HTTP headers, can override built in default headers. The
362 value must be a string encoding the headers.
364 @item multiple_requests
365 Use persistent connections if set to 1, default is 0.
368 Set custom HTTP post data.
371 Set the Referer header. Include 'Referer: URL' header in HTTP request.
374 Override the User-Agent header. If not specified the protocol will use a
375 string describing the libavformat build. ("Lavf/<version>")
378 This is a deprecated option, you can use user_agent instead it.
381 Set timeout in microseconds of socket I/O operations used by the underlying low level
382 operation. By default it is set to -1, which means that the timeout is
385 @item reconnect_at_eof
386 If set then eof is treated like an error and causes reconnection, this is useful
387 for live / endless streams.
389 @item reconnect_streamed
390 If set then even streamed/non seekable streams will be reconnected on errors.
392 @item reconnect_delay_max
393 Sets the maximum delay in seconds after which to give up reconnecting
396 Export the MIME type.
399 Exports the HTTP response version number. Usually "1.0" or "1.1".
402 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
403 supports this, the metadata has to be retrieved by the application by reading
404 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
407 @item icy_metadata_headers
408 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
409 headers, separated by newline characters.
411 @item icy_metadata_packet
412 If the server supports ICY metadata, and @option{icy} was set to 1, this
413 contains the last non-empty metadata packet sent by the server. It should be
414 polled in regular intervals by applications interested in mid-stream metadata
418 Set the cookies to be sent in future requests. The format of each cookie is the
419 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
420 delimited by a newline character.
423 Set initial byte offset.
426 Try to limit the request to bytes preceding this offset.
429 When used as a client option it sets the HTTP method for the request.
431 When used as a server option it sets the HTTP method that is going to be
432 expected from the client(s).
433 If the expected and the received HTTP method do not match the client will
434 be given a Bad Request response.
435 When unset the HTTP method is not checked for now. This will be replaced by
436 autodetection in the future.
439 If set to 1 enables experimental HTTP server. This can be used to send data when
440 used as an output option, or read data from a client with HTTP POST when used as
442 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
443 in ffmpeg.c and thus must not be used as a command line option.
445 # Server side (sending):
446 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
448 # Client side (receiving):
449 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
451 # Client can also be done with wget:
452 wget http://@var{server}:@var{port} -O somefile.ogg
454 # Server side (receiving):
455 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
457 # Client side (sending):
458 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
460 # Client can also be done with wget:
461 wget --post-file=somefile.ogg http://@var{server}:@var{port}
464 @item send_expect_100
465 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
466 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
471 @subsection HTTP Cookies
473 Some HTTP requests will be denied unless cookie values are passed in with the
474 request. The @option{cookies} option allows these cookies to be specified. At
475 the very least, each cookie must specify a value along with a path and domain.
476 HTTP requests that match both the domain and path will automatically include the
477 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
480 The required syntax to play a stream specifying a cookie is:
482 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
487 Icecast protocol (stream to Icecast servers)
489 This protocol accepts the following options:
493 Set the stream genre.
498 @item ice_description
499 Set the stream description.
502 Set the stream website URL.
505 Set if the stream should be public.
506 The default is 0 (not public).
509 Override the User-Agent header. If not specified a string of the form
510 "Lavf/<version>" will be used.
513 Set the Icecast mountpoint password.
516 Set the stream content type. This must be set if it is different from
520 This enables support for Icecast versions < 2.4.0, that do not support the
521 HTTP PUT method but the SOURCE method.
526 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
531 MMS (Microsoft Media Server) protocol over TCP.
535 MMS (Microsoft Media Server) protocol over HTTP.
537 The required syntax is:
539 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
546 Computes the MD5 hash of the data to be written, and on close writes
547 this to the designated output or stdout if none is specified. It can
548 be used to test muxers without writing an actual file.
550 Some examples follow.
552 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
553 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
555 # Write the MD5 hash of the encoded AVI file to stdout.
556 ffmpeg -i input.flv -f avi -y md5:
559 Note that some formats (typically MOV) require the output protocol to
560 be seekable, so they will fail with the MD5 output protocol.
564 UNIX pipe access protocol.
566 Read and write from UNIX pipes.
568 The accepted syntax is:
573 @var{number} is the number corresponding to the file descriptor of the
574 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
575 is not specified, by default the stdout file descriptor will be used
576 for writing, stdin for reading.
578 For example to read from stdin with @command{ffmpeg}:
580 cat test.wav | ffmpeg -i pipe:0
581 # ...this is the same as...
582 cat test.wav | ffmpeg -i pipe:
585 For writing to stdout with @command{ffmpeg}:
587 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
588 # ...this is the same as...
589 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
592 This protocol accepts the following options:
596 Set I/O operation maximum block size, in bytes. Default value is
597 @code{INT_MAX}, which results in not limiting the requested block size.
598 Setting this value reasonably low improves user termination request reaction
599 time, which is valuable if data transmission is slow.
602 Note that some formats (typically MOV), require the output protocol to
603 be seekable, so they will fail with the pipe output protocol.
607 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
609 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
610 for MPEG-2 Transport Streams sent over RTP.
612 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
613 the @code{rtp} protocol.
615 The required syntax is:
617 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
620 The destination UDP ports are @code{port + 2} for the column FEC stream
621 and @code{port + 4} for the row FEC stream.
623 This protocol accepts the following options:
627 The number of columns (4-20, LxD <= 100)
630 The number of rows (4-20, LxD <= 100)
637 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
642 Real-Time Messaging Protocol.
644 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
645 content across a TCP/IP network.
647 The required syntax is:
649 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
652 The accepted parameters are:
656 An optional username (mostly for publishing).
659 An optional password (mostly for publishing).
662 The address of the RTMP server.
665 The number of the TCP port to use (by default is 1935).
668 It is the name of the application to access. It usually corresponds to
669 the path where the application is installed on the RTMP server
670 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
671 the value parsed from the URI through the @code{rtmp_app} option, too.
674 It is the path or name of the resource to play with reference to the
675 application specified in @var{app}, may be prefixed by "mp4:". You
676 can override the value parsed from the URI through the @code{rtmp_playpath}
680 Act as a server, listening for an incoming connection.
683 Maximum time to wait for the incoming connection. Implies listen.
686 Additionally, the following parameters can be set via command line options
687 (or in code via @code{AVOption}s):
691 Name of application to connect on the RTMP server. This option
692 overrides the parameter specified in the URI.
695 Set the client buffer time in milliseconds. The default is 3000.
698 Extra arbitrary AMF connection parameters, parsed from a string,
699 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
700 Each value is prefixed by a single character denoting the type,
701 B for Boolean, N for number, S for string, O for object, or Z for null,
702 followed by a colon. For Booleans the data must be either 0 or 1 for
703 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
704 1 to end or begin an object, respectively. Data items in subobjects may
705 be named, by prefixing the type with 'N' and specifying the name before
706 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
707 times to construct arbitrary AMF sequences.
710 Version of the Flash plugin used to run the SWF player. The default
711 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
712 <libavformat version>).)
714 @item rtmp_flush_interval
715 Number of packets flushed in the same request (RTMPT only). The default
719 Specify that the media is a live stream. No resuming or seeking in
720 live streams is possible. The default value is @code{any}, which means the
721 subscriber first tries to play the live stream specified in the
722 playpath. If a live stream of that name is not found, it plays the
723 recorded stream. The other possible values are @code{live} and
727 URL of the web page in which the media was embedded. By default no
731 Stream identifier to play or to publish. This option overrides the
732 parameter specified in the URI.
735 Name of live stream to subscribe to. By default no value will be sent.
736 It is only sent if the option is specified or if rtmp_live
740 SHA256 hash of the decompressed SWF file (32 bytes).
743 Size of the decompressed SWF file, required for SWFVerification.
746 URL of the SWF player for the media. By default no value will be sent.
749 URL to player swf file, compute hash/size automatically.
752 URL of the target stream. Defaults to proto://host[:port]/app.
756 For example to read with @command{ffplay} a multimedia resource named
757 "sample" from the application "vod" from an RTMP server "myserver":
759 ffplay rtmp://myserver/vod/sample
762 To publish to a password protected server, passing the playpath and
763 app names separately:
765 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
770 Encrypted Real-Time Messaging Protocol.
772 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
773 streaming multimedia content within standard cryptographic primitives,
774 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
779 Real-Time Messaging Protocol over a secure SSL connection.
781 The Real-Time Messaging Protocol (RTMPS) is used for streaming
782 multimedia content across an encrypted connection.
786 Real-Time Messaging Protocol tunneled through HTTP.
788 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
789 for streaming multimedia content within HTTP requests to traverse
794 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
796 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
797 is used for streaming multimedia content within HTTP requests to traverse
802 Real-Time Messaging Protocol tunneled through HTTPS.
804 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
805 for streaming multimedia content within HTTPS requests to traverse
808 @section libsmbclient
810 libsmbclient permits one to manipulate CIFS/SMB network resources.
812 Following syntax is required.
815 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
818 This protocol accepts the following options.
822 Set timeout in milliseconds of socket I/O operations used by the underlying
823 low level operation. By default it is set to -1, which means that the timeout
827 Truncate existing files on write, if set to 1. A value of 0 prevents
828 truncating. Default value is 1.
831 Set the workgroup used for making connections. By default workgroup is not specified.
835 For more information see: @url{http://www.samba.org/}.
839 Secure File Transfer Protocol via libssh
841 Read from or write to remote resources using SFTP protocol.
843 Following syntax is required.
846 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
849 This protocol accepts the following options.
853 Set timeout of socket I/O operations used by the underlying low level
854 operation. By default it is set to -1, which means that the timeout
858 Truncate existing files on write, if set to 1. A value of 0 prevents
859 truncating. Default value is 1.
862 Specify the path of the file containing private key to use during authorization.
863 By default libssh searches for keys in the @file{~/.ssh/} directory.
867 Example: Play a file stored on remote server.
870 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
873 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
875 Real-Time Messaging Protocol and its variants supported through
878 Requires the presence of the librtmp headers and library during
879 configuration. You need to explicitly configure the build with
880 "--enable-librtmp". If enabled this will replace the native RTMP
883 This protocol provides most client functions and a few server
884 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
885 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
886 variants of these encrypted types (RTMPTE, RTMPTS).
888 The required syntax is:
890 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
893 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
894 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
895 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
896 meaning as specified for the RTMP native protocol.
897 @var{options} contains a list of space-separated options of the form
900 See the librtmp manual page (man 3 librtmp) for more information.
902 For example, to stream a file in real-time to an RTMP server using
905 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
908 To play the same stream using @command{ffplay}:
910 ffplay "rtmp://myserver/live/mystream live=1"
915 Real-time Transport Protocol.
917 The required syntax for an RTP URL is:
918 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
920 @var{port} specifies the RTP port to use.
922 The following URL options are supported:
927 Set the TTL (Time-To-Live) value (for multicast only).
929 @item rtcpport=@var{n}
930 Set the remote RTCP port to @var{n}.
932 @item localrtpport=@var{n}
933 Set the local RTP port to @var{n}.
935 @item localrtcpport=@var{n}'
936 Set the local RTCP port to @var{n}.
938 @item pkt_size=@var{n}
939 Set max packet size (in bytes) to @var{n}.
942 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
945 @item sources=@var{ip}[,@var{ip}]
946 List allowed source IP addresses.
948 @item block=@var{ip}[,@var{ip}]
949 List disallowed (blocked) source IP addresses.
951 @item write_to_source=0|1
952 Send packets to the source address of the latest received packet (if
953 set to 1) or to a default remote address (if set to 0).
955 @item localport=@var{n}
956 Set the local RTP port to @var{n}.
958 This is a deprecated option. Instead, @option{localrtpport} should be
968 If @option{rtcpport} is not set the RTCP port will be set to the RTP
972 If @option{localrtpport} (the local RTP port) is not set any available
973 port will be used for the local RTP and RTCP ports.
976 If @option{localrtcpport} (the local RTCP port) is not set it will be
977 set to the local RTP port value plus 1.
982 Real-Time Streaming Protocol.
984 RTSP is not technically a protocol handler in libavformat, it is a demuxer
985 and muxer. The demuxer supports both normal RTSP (with data transferred
986 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
987 data transferred over RDT).
989 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
990 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
991 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
993 The required syntax for a RTSP url is:
995 rtsp://@var{hostname}[:@var{port}]/@var{path}
998 Options can be set on the @command{ffmpeg}/@command{ffplay} command
999 line, or set in code via @code{AVOption}s or in
1000 @code{avformat_open_input}.
1002 The following options are supported.
1006 Do not start playing the stream immediately if set to 1. Default value
1009 @item rtsp_transport
1010 Set RTSP transport protocols.
1012 It accepts the following values:
1015 Use UDP as lower transport protocol.
1018 Use TCP (interleaving within the RTSP control channel) as lower
1022 Use UDP multicast as lower transport protocol.
1025 Use HTTP tunneling as lower transport protocol, which is useful for
1029 Multiple lower transport protocols may be specified, in that case they are
1030 tried one at a time (if the setup of one fails, the next one is tried).
1031 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
1036 The following values are accepted:
1039 Accept packets only from negotiated peer address and port.
1041 Act as a server, listening for an incoming connection.
1043 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
1046 Default value is @samp{none}.
1048 @item allowed_media_types
1049 Set media types to accept from the server.
1051 The following flags are accepted:
1058 By default it accepts all media types.
1061 Set minimum local UDP port. Default value is 5000.
1064 Set maximum local UDP port. Default value is 65000.
1067 Set maximum timeout (in seconds) to wait for incoming connections.
1069 A value of -1 means infinite (default). This option implies the
1070 @option{rtsp_flags} set to @samp{listen}.
1072 @item reorder_queue_size
1073 Set number of packets to buffer for handling of reordered packets.
1076 Set socket TCP I/O timeout in microseconds.
1079 Override User-Agent header. If not specified, it defaults to the
1080 libavformat identifier string.
1083 When receiving data over UDP, the demuxer tries to reorder received packets
1084 (since they may arrive out of order, or packets may get lost totally). This
1085 can be disabled by setting the maximum demuxing delay to zero (via
1086 the @code{max_delay} field of AVFormatContext).
1088 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1089 streams to display can be chosen with @code{-vst} @var{n} and
1090 @code{-ast} @var{n} for video and audio respectively, and can be switched
1091 on the fly by pressing @code{v} and @code{a}.
1093 @subsection Examples
1095 The following examples all make use of the @command{ffplay} and
1096 @command{ffmpeg} tools.
1100 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1102 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1106 Watch a stream tunneled over HTTP:
1108 ffplay -rtsp_transport http rtsp://server/video.mp4
1112 Send a stream in realtime to a RTSP server, for others to watch:
1114 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1118 Receive a stream in realtime:
1120 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1126 Session Announcement Protocol (RFC 2974). This is not technically a
1127 protocol handler in libavformat, it is a muxer and demuxer.
1128 It is used for signalling of RTP streams, by announcing the SDP for the
1129 streams regularly on a separate port.
1133 The syntax for a SAP url given to the muxer is:
1135 sap://@var{destination}[:@var{port}][?@var{options}]
1138 The RTP packets are sent to @var{destination} on port @var{port},
1139 or to port 5004 if no port is specified.
1140 @var{options} is a @code{&}-separated list. The following options
1145 @item announce_addr=@var{address}
1146 Specify the destination IP address for sending the announcements to.
1147 If omitted, the announcements are sent to the commonly used SAP
1148 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1149 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1151 @item announce_port=@var{port}
1152 Specify the port to send the announcements on, defaults to
1153 9875 if not specified.
1156 Specify the time to live value for the announcements and RTP packets,
1159 @item same_port=@var{0|1}
1160 If set to 1, send all RTP streams on the same port pair. If zero (the
1161 default), all streams are sent on unique ports, with each stream on a
1162 port 2 numbers higher than the previous.
1163 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1164 The RTP stack in libavformat for receiving requires all streams to be sent
1168 Example command lines follow.
1170 To broadcast a stream on the local subnet, for watching in VLC:
1173 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1176 Similarly, for watching in @command{ffplay}:
1179 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1182 And for watching in @command{ffplay}, over IPv6:
1185 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1190 The syntax for a SAP url given to the demuxer is:
1192 sap://[@var{address}][:@var{port}]
1195 @var{address} is the multicast address to listen for announcements on,
1196 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1197 is the port that is listened on, 9875 if omitted.
1199 The demuxers listens for announcements on the given address and port.
1200 Once an announcement is received, it tries to receive that particular stream.
1202 Example command lines follow.
1204 To play back the first stream announced on the normal SAP multicast address:
1210 To play back the first stream announced on one the default IPv6 SAP multicast address:
1213 ffplay sap://[ff0e::2:7ffe]
1218 Stream Control Transmission Protocol.
1220 The accepted URL syntax is:
1222 sctp://@var{host}:@var{port}[?@var{options}]
1225 The protocol accepts the following options:
1228 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1231 Set the maximum number of streams. By default no limit is set.
1236 Haivision Secure Reliable Transport Protocol via libsrt.
1238 The supported syntax for a SRT URL is:
1240 srt://@var{hostname}:@var{port}[?@var{options}]
1243 @var{options} contains a list of &-separated options of the form
1244 @var{key}=@var{val}.
1249 @var{options} srt://@var{hostname}:@var{port}
1252 @var{options} contains a list of '-@var{key} @var{val}'
1255 This protocol accepts the following options.
1258 @item connect_timeout=@var{milliseconds}
1259 Connection timeout; SRT cannot connect for RTT > 1500 msec
1260 (2 handshake exchanges) with the default connect timeout of
1261 3 seconds. This option applies to the caller and rendezvous
1262 connection modes. The connect timeout is 10 times the value
1263 set for the rendezvous mode (which can be used as a
1264 workaround for this connection problem with earlier versions).
1266 @item ffs=@var{bytes}
1267 Flight Flag Size (Window Size), in bytes. FFS is actually an
1268 internal parameter and you should set it to not less than
1269 @option{recv_buffer_size} and @option{mss}. The default value
1270 is relatively large, therefore unless you set a very large receiver buffer,
1271 you do not need to change this option. Default value is 25600.
1273 @item inputbw=@var{bytes/seconds}
1274 Sender nominal input rate, in bytes per seconds. Used along with
1275 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1276 calculate maximum sending rate when recovery packets are sent
1277 along with the main media stream:
1278 @option{inputbw} * (100 + @option{oheadbw}) / 100
1279 if @option{inputbw} is not set while @option{maxbw} is set to
1280 relative (0), the actual input rate is evaluated inside
1281 the library. Default value is 0.
1283 @item iptos=@var{tos}
1284 IP Type of Service. Applies to sender only. Default value is 0xB8.
1286 @item ipttl=@var{ttl}
1287 IP Time To Live. Applies to sender only. Default value is 64.
1289 @item latency=@var{microseconds}
1290 Timestamp-based Packet Delivery Delay.
1291 Used to absorb bursts of missed packet retransmissions.
1292 This flag sets both @option{rcvlatency} and @option{peerlatency}
1293 to the same value. Note that prior to version 1.3.0
1294 this is the only flag to set the latency, however
1295 this is effectively equivalent to setting @option{peerlatency},
1296 when side is sender and @option{rcvlatency}
1297 when side is receiver, and the bidirectional stream
1298 sending is not supported.
1300 @item listen_timeout=@var{microseconds}
1301 Set socket listen timeout.
1303 @item maxbw=@var{bytes/seconds}
1304 Maximum sending bandwidth, in bytes per seconds.
1305 -1 infinite (CSRTCC limit is 30mbps)
1306 0 relative to input rate (see @option{inputbw})
1307 >0 absolute limit value
1308 Default value is 0 (relative)
1310 @item mode=@var{caller|listener|rendezvous}
1312 @option{caller} opens client connection.
1313 @option{listener} starts server to listen for incoming connections.
1314 @option{rendezvous} use Rendez-Vous connection mode.
1315 Default value is caller.
1317 @item mss=@var{bytes}
1318 Maximum Segment Size, in bytes. Used for buffer allocation
1319 and rate calculation using a packet counter assuming fully
1320 filled packets. The smallest MSS between the peers is
1321 used. This is 1500 by default in the overall internet.
1322 This is the maximum size of the UDP packet and can be
1323 only decreased, unless you have some unusual dedicated
1324 network settings. Default value is 1500.
1326 @item nakreport=@var{1|0}
1327 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1328 periodically until a lost packet is retransmitted or
1329 intentionally dropped. Default value is 1.
1331 @item oheadbw=@var{percents}
1332 Recovery bandwidth overhead above input rate, in percents.
1333 See @option{inputbw}. Default value is 25%.
1335 @item passphrase=@var{string}
1336 HaiCrypt Encryption/Decryption Passphrase string, length
1337 from 10 to 79 characters. The passphrase is the shared
1338 secret between the sender and the receiver. It is used
1339 to generate the Key Encrypting Key using PBKDF2
1340 (Password-Based Key Derivation Function). It is used
1341 only if @option{pbkeylen} is non-zero. It is used on
1342 the receiver only if the received data is encrypted.
1343 The configured passphrase cannot be recovered (write-only).
1345 @item enforced_encryption=@var{1|0}
1346 If true, both connection parties must have the same password
1347 set (including empty, that is, with no encryption). If the
1348 password doesn't match or only one side is unencrypted,
1349 the connection is rejected. Default is true.
1351 @item kmrefreshrate=@var{packets}
1352 The number of packets to be transmitted after which the
1353 encryption key is switched to a new key. Default is -1.
1354 -1 means auto (0x1000000 in srt library). The range for
1355 this option is integers in the 0 - @code{INT_MAX}.
1357 @item kmpreannounce=@var{packets}
1358 The interval between when a new encryption key is sent and
1359 when switchover occurs. This value also applies to the
1360 subsequent interval between when switchover occurs and
1361 when the old encryption key is decommissioned. Default is -1.
1362 -1 means auto (0x1000 in srt library). The range for
1363 this option is integers in the 0 - @code{INT_MAX}.
1365 @item payload_size=@var{bytes}
1366 Sets the maximum declared size of a packet transferred
1367 during the single call to the sending function in Live
1368 mode. Use 0 if this value isn't used (which is default in
1370 Default is -1 (automatic), which typically means MPEG-TS;
1371 if you are going to use SRT
1372 to send any different kind of payload, such as, for example,
1373 wrapping a live stream in very small frames, then you can
1374 use a bigger maximum frame size, though not greater than
1377 @item pkt_size=@var{bytes}
1378 Alias for @samp{payload_size}.
1380 @item peerlatency=@var{microseconds}
1381 The latency value (as described in @option{rcvlatency}) that is
1382 set by the sender side as a minimum value for the receiver.
1384 @item pbkeylen=@var{bytes}
1385 Sender encryption key length, in bytes.
1386 Only can be set to 0, 16, 24 and 32.
1387 Enable sender encryption if not 0.
1388 Not required on receiver (set to 0),
1389 key size obtained from sender in HaiCrypt handshake.
1392 @item rcvlatency=@var{microseconds}
1393 The time that should elapse since the moment when the
1394 packet was sent and the moment when it's delivered to
1395 the receiver application in the receiving function.
1396 This time should be a buffer time large enough to cover
1397 the time spent for sending, unexpectedly extended RTT
1398 time, and the time needed to retransmit the lost UDP
1399 packet. The effective latency value will be the maximum
1400 of this options' value and the value of @option{peerlatency}
1401 set by the peer side. Before version 1.3.0 this option
1402 is only available as @option{latency}.
1404 @item recv_buffer_size=@var{bytes}
1405 Set UDP receive buffer size, expressed in bytes.
1407 @item send_buffer_size=@var{bytes}
1408 Set UDP send buffer size, expressed in bytes.
1410 @item timeout=@var{microseconds}
1411 Set raise error timeouts for read, write and connect operations. Note that the
1412 SRT library has internal timeouts which can be controlled separately, the
1413 value set here is only a cap on those.
1415 @item tlpktdrop=@var{1|0}
1416 Too-late Packet Drop. When enabled on receiver, it skips
1417 missing packets that have not been delivered in time and
1418 delivers the following packets to the application when
1419 their time-to-play has come. It also sends a fake ACK to
1420 the sender. When enabled on sender and enabled on the
1421 receiving peer, the sender drops the older packets that
1422 have no chance of being delivered in time. It was
1423 automatically enabled in the sender if the receiver
1426 @item sndbuf=@var{bytes}
1427 Set send buffer size, expressed in bytes.
1429 @item rcvbuf=@var{bytes}
1430 Set receive buffer size, expressed in bytes.
1432 Receive buffer must not be greater than @option{ffs}.
1434 @item lossmaxttl=@var{packets}
1435 The value up to which the Reorder Tolerance may grow. When
1436 Reorder Tolerance is > 0, then packet loss report is delayed
1437 until that number of packets come in. Reorder Tolerance
1438 increases every time a "belated" packet has come, but it
1439 wasn't due to retransmission (that is, when UDP packets tend
1440 to come out of order), with the difference between the latest
1441 sequence and this packet's sequence, and not more than the
1442 value of this option. By default it's 0, which means that this
1443 mechanism is turned off, and the loss report is always sent
1444 immediately upon experiencing a "gap" in sequences.
1447 The minimum SRT version that is required from the peer. A connection
1448 to a peer that does not satisfy the minimum version requirement
1451 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1454 @item streamid=@var{string}
1455 A string limited to 512 characters that can be set on the socket prior
1456 to connecting. This stream ID will be able to be retrieved by the
1457 listener side from the socket that is returned from srt_accept and
1458 was connected by a socket with that set stream ID. SRT does not enforce
1459 any special interpretation of the contents of this string.
1460 This option doesn’t make sense in Rendezvous connection; the result
1461 might be that simply one side will override the value from the other
1462 side and it’s the matter of luck which one would win
1464 @item smoother=@var{live|file}
1465 The type of Smoother used for the transmission for that socket, which
1466 is responsible for the transmission and congestion control. The Smoother
1467 type must be exactly the same on both connecting parties, otherwise
1468 the connection is rejected.
1470 @item messageapi=@var{1|0}
1471 When set, this socket uses the Message API, otherwise it uses Buffer
1472 API. Note that in live mode (see @option{transtype}) there’s only
1473 message API available. In File mode you can chose to use one of two modes:
1475 Stream API (default, when this option is false). In this mode you may
1476 send as many data as you wish with one sending instruction, or even use
1477 dedicated functions that read directly from a file. The internal facility
1478 will take care of any speed and congestion control. When receiving, you
1479 can also receive as many data as desired, the data not extracted will be
1480 waiting for the next call. There is no boundary between data portions in
1483 Message API. In this mode your single sending instruction passes exactly
1484 one piece of data that has boundaries (a message). Contrary to Live mode,
1485 this message may span across multiple UDP packets and the only size
1486 limitation is that it shall fit as a whole in the sending buffer. The
1487 receiver shall use as large buffer as necessary to receive the message,
1488 otherwise the message will not be given up. When the message is not
1489 complete (not all packets received or there was a packet loss) it will
1492 @item transtype=@var{live|file}
1493 Sets the transmission type for the socket, in particular, setting this
1494 option sets multiple other parameters to their default values as required
1495 for a particular transmission type.
1497 live: Set options as for live transmission. In this mode, you should
1498 send by one sending instruction only so many data that fit in one UDP packet,
1499 and limited to the value defined first in @option{payload_size} (1316 is
1500 default in this mode). There is no speed control in this mode, only the
1501 bandwidth control, if configured, in order to not exceed the bandwidth with
1502 the overhead transmission (retransmitted and control packets).
1504 file: Set options as for non-live transmission. See @option{messageapi}
1505 for further explanations
1507 @item linger=@var{seconds}
1508 The number of seconds that the socket waits for unsent data when closing.
1509 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
1510 seconds in file mode). The range for this option is integers in the
1515 For more information see: @url{https://github.com/Haivision/srt}.
1519 Secure Real-time Transport Protocol.
1521 The accepted options are:
1524 @item srtp_out_suite
1525 Select input and output encoding suites.
1529 @item AES_CM_128_HMAC_SHA1_80
1530 @item SRTP_AES128_CM_HMAC_SHA1_80
1531 @item AES_CM_128_HMAC_SHA1_32
1532 @item SRTP_AES128_CM_HMAC_SHA1_32
1535 @item srtp_in_params
1536 @item srtp_out_params
1537 Set input and output encoding parameters, which are expressed by a
1538 base64-encoded representation of a binary block. The first 16 bytes of
1539 this binary block are used as master key, the following 14 bytes are
1540 used as master salt.
1545 Virtually extract a segment of a file or another stream.
1546 The underlying stream must be seekable.
1551 Start offset of the extracted segment, in bytes.
1553 End offset of the extracted segment, in bytes.
1554 If set to 0, extract till end of file.
1559 Extract a chapter from a DVD VOB file (start and end sectors obtained
1560 externally and multiplied by 2048):
1562 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1565 Play an AVI file directly from a TAR archive:
1567 subfile,,start,183241728,end,366490624,,:archive.tar
1570 Play a MPEG-TS file from start offset till end:
1572 subfile,,start,32815239,end,0,,:video.ts
1577 Writes the output to multiple protocols. The individual outputs are separated
1581 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1586 Transmission Control Protocol.
1588 The required syntax for a TCP url is:
1590 tcp://@var{hostname}:@var{port}[?@var{options}]
1593 @var{options} contains a list of &-separated options of the form
1594 @var{key}=@var{val}.
1596 The list of supported options follows.
1599 @item listen=@var{1|0}
1600 Listen for an incoming connection. Default value is 0.
1602 @item timeout=@var{microseconds}
1603 Set raise error timeout, expressed in microseconds.
1605 This option is only relevant in read mode: if no data arrived in more
1606 than this time interval, raise error.
1608 @item listen_timeout=@var{milliseconds}
1609 Set listen timeout, expressed in milliseconds.
1611 @item recv_buffer_size=@var{bytes}
1612 Set receive buffer size, expressed bytes.
1614 @item send_buffer_size=@var{bytes}
1615 Set send buffer size, expressed bytes.
1617 @item tcp_nodelay=@var{1|0}
1618 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1620 @item tcp_mss=@var{bytes}
1621 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1624 The following example shows how to setup a listening TCP connection
1625 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1627 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1628 ffplay tcp://@var{hostname}:@var{port}
1633 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1635 The required syntax for a TLS/SSL url is:
1637 tls://@var{hostname}:@var{port}[?@var{options}]
1640 The following parameters can be set via command line options
1641 (or in code via @code{AVOption}s):
1645 @item ca_file, cafile=@var{filename}
1646 A file containing certificate authority (CA) root certificates to treat
1647 as trusted. If the linked TLS library contains a default this might not
1648 need to be specified for verification to work, but not all libraries and
1649 setups have defaults built in.
1650 The file must be in OpenSSL PEM format.
1652 @item tls_verify=@var{1|0}
1653 If enabled, try to verify the peer that we are communicating with.
1654 Note, if using OpenSSL, this currently only makes sure that the
1655 peer certificate is signed by one of the root certificates in the CA
1656 database, but it does not validate that the certificate actually
1657 matches the host name we are trying to connect to. (With other backends,
1658 the host name is validated as well.)
1660 This is disabled by default since it requires a CA database to be
1661 provided by the caller in many cases.
1663 @item cert_file, cert=@var{filename}
1664 A file containing a certificate to use in the handshake with the peer.
1665 (When operating as server, in listen mode, this is more often required
1666 by the peer, while client certificates only are mandated in certain
1669 @item key_file, key=@var{filename}
1670 A file containing the private key for the certificate.
1672 @item listen=@var{1|0}
1673 If enabled, listen for connections on the provided port, and assume
1674 the server role in the handshake instead of the client role.
1678 Example command lines:
1680 To create a TLS/SSL server that serves an input stream.
1683 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1686 To play back a stream from the TLS/SSL server using @command{ffplay}:
1689 ffplay tls://@var{hostname}:@var{port}
1694 User Datagram Protocol.
1696 The required syntax for an UDP URL is:
1698 udp://@var{hostname}:@var{port}[?@var{options}]
1701 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1703 In case threading is enabled on the system, a circular buffer is used
1704 to store the incoming data, which allows one to reduce loss of data due to
1705 UDP socket buffer overruns. The @var{fifo_size} and
1706 @var{overrun_nonfatal} options are related to this buffer.
1708 The list of supported options follows.
1711 @item buffer_size=@var{size}
1712 Set the UDP maximum socket buffer size in bytes. This is used to set either
1713 the receive or send buffer size, depending on what the socket is used for.
1714 Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
1716 @item bitrate=@var{bitrate}
1717 If set to nonzero, the output will have the specified constant bitrate if the
1718 input has enough packets to sustain it.
1720 @item burst_bits=@var{bits}
1721 When using @var{bitrate} this specifies the maximum number of bits in
1724 @item localport=@var{port}
1725 Override the local UDP port to bind with.
1727 @item localaddr=@var{addr}
1728 Local IP address of a network interface used for sending packets or joining
1731 @item pkt_size=@var{size}
1732 Set the size in bytes of UDP packets.
1734 @item reuse=@var{1|0}
1735 Explicitly allow or disallow reusing UDP sockets.
1738 Set the time to live value (for multicast only).
1740 @item connect=@var{1|0}
1741 Initialize the UDP socket with @code{connect()}. In this case, the
1742 destination address can't be changed with ff_udp_set_remote_url later.
1743 If the destination address isn't known at the start, this option can
1744 be specified in ff_udp_set_remote_url, too.
1745 This allows finding out the source address for the packets with getsockname,
1746 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1747 unreachable" is received.
1748 For receiving, this gives the benefit of only receiving packets from
1749 the specified peer address/port.
1751 @item sources=@var{address}[,@var{address}]
1752 Only receive packets sent from the specified addresses. In case of multicast,
1753 also subscribe to multicast traffic coming from these addresses only.
1755 @item block=@var{address}[,@var{address}]
1756 Ignore packets sent from the specified addresses. In case of multicast, also
1757 exclude the source addresses in the multicast subscription.
1759 @item fifo_size=@var{units}
1760 Set the UDP receiving circular buffer size, expressed as a number of
1761 packets with size of 188 bytes. If not specified defaults to 7*4096.
1763 @item overrun_nonfatal=@var{1|0}
1764 Survive in case of UDP receiving circular buffer overrun. Default
1767 @item timeout=@var{microseconds}
1768 Set raise error timeout, expressed in microseconds.
1770 This option is only relevant in read mode: if no data arrived in more
1771 than this time interval, raise error.
1773 @item broadcast=@var{1|0}
1774 Explicitly allow or disallow UDP broadcasting.
1776 Note that broadcasting may not work properly on networks having
1777 a broadcast storm protection.
1780 @subsection Examples
1784 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1786 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1790 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1791 sized UDP packets, using a large input buffer:
1793 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1797 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1799 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1807 The required syntax for a Unix socket URL is:
1810 unix://@var{filepath}
1813 The following parameters can be set via command line options
1814 (or in code via @code{AVOption}s):
1820 Create the Unix socket in listening mode.
1825 ZeroMQ asynchronous messaging using the libzmq library.
1827 This library supports unicast streaming to multiple clients without relying on
1830 The required syntax for streaming or connecting to a stream is:
1832 zmq:tcp://ip-address:port
1836 Create a localhost stream on port 5555:
1838 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1841 Multiple clients may connect to the stream using:
1843 ffplay zmq:tcp://127.0.0.1:5555
1846 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
1847 The server side binds to a port and publishes data. Clients connect to the
1848 server (via IP address/port) and subscribe to the stream. The order in which
1849 the server and client start generally does not matter.
1851 ffmpeg must be compiled with the --enable-libzmq option to support
1854 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1855 line. The following options are supported:
1860 Forces the maximum packet size for sending/receiving data. The default value is
1861 131,072 bytes. On the server side, this sets the maximum size of sent packets
1862 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
1863 packets. Note that pkt_size on the clients should be equal to or greater than
1864 pkt_size on the server. Otherwise the received message may be truncated causing
1870 @c man end PROTOCOLS