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.
395 @item reconnect_at_eof
396 If set then eof is treated like an error and causes reconnection, this is useful
397 for live / endless streams.
399 @item reconnect_streamed
400 If set then even streamed/non seekable streams will be reconnected on errors.
402 @item reconnect_delay_max
403 Sets the maximum delay in seconds after which to give up reconnecting
406 Export the MIME type.
409 Exports the HTTP response version number. Usually "1.0" or "1.1".
412 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
413 supports this, the metadata has to be retrieved by the application by reading
414 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
417 @item icy_metadata_headers
418 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
419 headers, separated by newline characters.
421 @item icy_metadata_packet
422 If the server supports ICY metadata, and @option{icy} was set to 1, this
423 contains the last non-empty metadata packet sent by the server. It should be
424 polled in regular intervals by applications interested in mid-stream metadata
428 Set the cookies to be sent in future requests. The format of each cookie is the
429 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
430 delimited by a newline character.
433 Set initial byte offset.
436 Try to limit the request to bytes preceding this offset.
439 When used as a client option it sets the HTTP method for the request.
441 When used as a server option it sets the HTTP method that is going to be
442 expected from the client(s).
443 If the expected and the received HTTP method do not match the client will
444 be given a Bad Request response.
445 When unset the HTTP method is not checked for now. This will be replaced by
446 autodetection in the future.
449 If set to 1 enables experimental HTTP server. This can be used to send data when
450 used as an output option, or read data from a client with HTTP POST when used as
452 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
453 in ffmpeg.c and thus must not be used as a command line option.
455 # Server side (sending):
456 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
458 # Client side (receiving):
459 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
461 # Client can also be done with wget:
462 wget http://@var{server}:@var{port} -O somefile.ogg
464 # Server side (receiving):
465 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
467 # Client side (sending):
468 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
470 # Client can also be done with wget:
471 wget --post-file=somefile.ogg http://@var{server}:@var{port}
474 @item send_expect_100
475 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
476 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
481 Set HTTP authentication type. No option for Digest, since this method requires
482 getting nonce parameters from the server first and can't be used straight away like
487 Choose the HTTP authentication type automatically. This is the default.
490 Choose the HTTP basic authentication.
492 Basic authentication sends a Base64-encoded string that contains a user name and password
493 for the client. Base64 is not a form of encryption and should be considered the same as
494 sending the user name and password in clear text (Base64 is a reversible encoding).
495 If a resource needs to be protected, strongly consider using an authentication scheme
496 other than basic authentication. HTTPS/TLS should be used with basic authentication.
497 Without these additional security enhancements, basic authentication should not be used
498 to protect sensitive or valuable information.
503 @subsection HTTP Cookies
505 Some HTTP requests will be denied unless cookie values are passed in with the
506 request. The @option{cookies} option allows these cookies to be specified. At
507 the very least, each cookie must specify a value along with a path and domain.
508 HTTP requests that match both the domain and path will automatically include the
509 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
512 The required syntax to play a stream specifying a cookie is:
514 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
519 Icecast protocol (stream to Icecast servers)
521 This protocol accepts the following options:
525 Set the stream genre.
530 @item ice_description
531 Set the stream description.
534 Set the stream website URL.
537 Set if the stream should be public.
538 The default is 0 (not public).
541 Override the User-Agent header. If not specified a string of the form
542 "Lavf/<version>" will be used.
545 Set the Icecast mountpoint password.
548 Set the stream content type. This must be set if it is different from
552 This enables support for Icecast versions < 2.4.0, that do not support the
553 HTTP PUT method but the SOURCE method.
556 Establish a TLS (HTTPS) connection to Icecast.
561 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
566 MMS (Microsoft Media Server) protocol over TCP.
570 MMS (Microsoft Media Server) protocol over HTTP.
572 The required syntax is:
574 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
581 Computes the MD5 hash of the data to be written, and on close writes
582 this to the designated output or stdout if none is specified. It can
583 be used to test muxers without writing an actual file.
585 Some examples follow.
587 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
588 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
590 # Write the MD5 hash of the encoded AVI file to stdout.
591 ffmpeg -i input.flv -f avi -y md5:
594 Note that some formats (typically MOV) require the output protocol to
595 be seekable, so they will fail with the MD5 output protocol.
599 UNIX pipe access protocol.
601 Read and write from UNIX pipes.
603 The accepted syntax is:
608 @var{number} is the number corresponding to the file descriptor of the
609 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
610 is not specified, by default the stdout file descriptor will be used
611 for writing, stdin for reading.
613 For example to read from stdin with @command{ffmpeg}:
615 cat test.wav | ffmpeg -i pipe:0
616 # ...this is the same as...
617 cat test.wav | ffmpeg -i pipe:
620 For writing to stdout with @command{ffmpeg}:
622 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
623 # ...this is the same as...
624 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
627 This protocol accepts the following options:
631 Set I/O operation maximum block size, in bytes. Default value is
632 @code{INT_MAX}, which results in not limiting the requested block size.
633 Setting this value reasonably low improves user termination request reaction
634 time, which is valuable if data transmission is slow.
637 Note that some formats (typically MOV), require the output protocol to
638 be seekable, so they will fail with the pipe output protocol.
642 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
644 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
645 for MPEG-2 Transport Streams sent over RTP.
647 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
648 the @code{rtp} protocol.
650 The required syntax is:
652 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
655 The destination UDP ports are @code{port + 2} for the column FEC stream
656 and @code{port + 4} for the row FEC stream.
658 This protocol accepts the following options:
662 The number of columns (4-20, LxD <= 100)
665 The number of rows (4-20, LxD <= 100)
672 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
677 Real-Time Messaging Protocol.
679 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
680 content across a TCP/IP network.
682 The required syntax is:
684 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
687 The accepted parameters are:
691 An optional username (mostly for publishing).
694 An optional password (mostly for publishing).
697 The address of the RTMP server.
700 The number of the TCP port to use (by default is 1935).
703 It is the name of the application to access. It usually corresponds to
704 the path where the application is installed on the RTMP server
705 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
706 the value parsed from the URI through the @code{rtmp_app} option, too.
709 It is the path or name of the resource to play with reference to the
710 application specified in @var{app}, may be prefixed by "mp4:". You
711 can override the value parsed from the URI through the @code{rtmp_playpath}
715 Act as a server, listening for an incoming connection.
718 Maximum time to wait for the incoming connection. Implies listen.
721 Additionally, the following parameters can be set via command line options
722 (or in code via @code{AVOption}s):
726 Name of application to connect on the RTMP server. This option
727 overrides the parameter specified in the URI.
730 Set the client buffer time in milliseconds. The default is 3000.
733 Extra arbitrary AMF connection parameters, parsed from a string,
734 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
735 Each value is prefixed by a single character denoting the type,
736 B for Boolean, N for number, S for string, O for object, or Z for null,
737 followed by a colon. For Booleans the data must be either 0 or 1 for
738 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
739 1 to end or begin an object, respectively. Data items in subobjects may
740 be named, by prefixing the type with 'N' and specifying the name before
741 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
742 times to construct arbitrary AMF sequences.
745 Version of the Flash plugin used to run the SWF player. The default
746 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
747 <libavformat version>).)
749 @item rtmp_flush_interval
750 Number of packets flushed in the same request (RTMPT only). The default
754 Specify that the media is a live stream. No resuming or seeking in
755 live streams is possible. The default value is @code{any}, which means the
756 subscriber first tries to play the live stream specified in the
757 playpath. If a live stream of that name is not found, it plays the
758 recorded stream. The other possible values are @code{live} and
762 URL of the web page in which the media was embedded. By default no
766 Stream identifier to play or to publish. This option overrides the
767 parameter specified in the URI.
770 Name of live stream to subscribe to. By default no value will be sent.
771 It is only sent if the option is specified or if rtmp_live
775 SHA256 hash of the decompressed SWF file (32 bytes).
778 Size of the decompressed SWF file, required for SWFVerification.
781 URL of the SWF player for the media. By default no value will be sent.
784 URL to player swf file, compute hash/size automatically.
787 URL of the target stream. Defaults to proto://host[:port]/app.
791 For example to read with @command{ffplay} a multimedia resource named
792 "sample" from the application "vod" from an RTMP server "myserver":
794 ffplay rtmp://myserver/vod/sample
797 To publish to a password protected server, passing the playpath and
798 app names separately:
800 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
805 Encrypted Real-Time Messaging Protocol.
807 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
808 streaming multimedia content within standard cryptographic primitives,
809 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
814 Real-Time Messaging Protocol over a secure SSL connection.
816 The Real-Time Messaging Protocol (RTMPS) is used for streaming
817 multimedia content across an encrypted connection.
821 Real-Time Messaging Protocol tunneled through HTTP.
823 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
824 for streaming multimedia content within HTTP requests to traverse
829 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
831 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
832 is used for streaming multimedia content within HTTP requests to traverse
837 Real-Time Messaging Protocol tunneled through HTTPS.
839 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
840 for streaming multimedia content within HTTPS requests to traverse
843 @section libsmbclient
845 libsmbclient permits one to manipulate CIFS/SMB network resources.
847 Following syntax is required.
850 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
853 This protocol accepts the following options.
857 Set timeout in milliseconds of socket I/O operations used by the underlying
858 low level operation. By default it is set to -1, which means that the timeout
862 Truncate existing files on write, if set to 1. A value of 0 prevents
863 truncating. Default value is 1.
866 Set the workgroup used for making connections. By default workgroup is not specified.
870 For more information see: @url{http://www.samba.org/}.
874 Secure File Transfer Protocol via libssh
876 Read from or write to remote resources using SFTP protocol.
878 Following syntax is required.
881 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
884 This protocol accepts the following options.
888 Set timeout of socket I/O operations used by the underlying low level
889 operation. By default it is set to -1, which means that the timeout
893 Truncate existing files on write, if set to 1. A value of 0 prevents
894 truncating. Default value is 1.
897 Specify the path of the file containing private key to use during authorization.
898 By default libssh searches for keys in the @file{~/.ssh/} directory.
902 Example: Play a file stored on remote server.
905 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
908 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
910 Real-Time Messaging Protocol and its variants supported through
913 Requires the presence of the librtmp headers and library during
914 configuration. You need to explicitly configure the build with
915 "--enable-librtmp". If enabled this will replace the native RTMP
918 This protocol provides most client functions and a few server
919 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
920 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
921 variants of these encrypted types (RTMPTE, RTMPTS).
923 The required syntax is:
925 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
928 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
929 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
930 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
931 meaning as specified for the RTMP native protocol.
932 @var{options} contains a list of space-separated options of the form
935 See the librtmp manual page (man 3 librtmp) for more information.
937 For example, to stream a file in real-time to an RTMP server using
940 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
943 To play the same stream using @command{ffplay}:
945 ffplay "rtmp://myserver/live/mystream live=1"
950 Real-time Transport Protocol.
952 The required syntax for an RTP URL is:
953 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
955 @var{port} specifies the RTP port to use.
957 The following URL options are supported:
962 Set the TTL (Time-To-Live) value (for multicast only).
964 @item rtcpport=@var{n}
965 Set the remote RTCP port to @var{n}.
967 @item localrtpport=@var{n}
968 Set the local RTP port to @var{n}.
970 @item localrtcpport=@var{n}'
971 Set the local RTCP port to @var{n}.
973 @item pkt_size=@var{n}
974 Set max packet size (in bytes) to @var{n}.
977 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
980 @item sources=@var{ip}[,@var{ip}]
981 List allowed source IP addresses.
983 @item block=@var{ip}[,@var{ip}]
984 List disallowed (blocked) source IP addresses.
986 @item write_to_source=0|1
987 Send packets to the source address of the latest received packet (if
988 set to 1) or to a default remote address (if set to 0).
990 @item localport=@var{n}
991 Set the local RTP port to @var{n}.
993 This is a deprecated option. Instead, @option{localrtpport} should be
1003 If @option{rtcpport} is not set the RTCP port will be set to the RTP
1007 If @option{localrtpport} (the local RTP port) is not set any available
1008 port will be used for the local RTP and RTCP ports.
1011 If @option{localrtcpport} (the local RTCP port) is not set it will be
1012 set to the local RTP port value plus 1.
1017 Real-Time Streaming Protocol.
1019 RTSP is not technically a protocol handler in libavformat, it is a demuxer
1020 and muxer. The demuxer supports both normal RTSP (with data transferred
1021 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
1022 data transferred over RDT).
1024 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1025 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
1026 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
1028 The required syntax for a RTSP url is:
1030 rtsp://@var{hostname}[:@var{port}]/@var{path}
1033 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1034 line, or set in code via @code{AVOption}s or in
1035 @code{avformat_open_input}.
1037 The following options are supported.
1041 Do not start playing the stream immediately if set to 1. Default value
1044 @item rtsp_transport
1045 Set RTSP transport protocols.
1047 It accepts the following values:
1050 Use UDP as lower transport protocol.
1053 Use TCP (interleaving within the RTSP control channel) as lower
1057 Use UDP multicast as lower transport protocol.
1060 Use HTTP tunneling as lower transport protocol, which is useful for
1064 Multiple lower transport protocols may be specified, in that case they are
1065 tried one at a time (if the setup of one fails, the next one is tried).
1066 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
1071 The following values are accepted:
1074 Accept packets only from negotiated peer address and port.
1076 Act as a server, listening for an incoming connection.
1078 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
1081 Default value is @samp{none}.
1083 @item allowed_media_types
1084 Set media types to accept from the server.
1086 The following flags are accepted:
1093 By default it accepts all media types.
1096 Set minimum local UDP port. Default value is 5000.
1099 Set maximum local UDP port. Default value is 65000.
1102 Set maximum timeout (in seconds) to wait for incoming connections.
1104 A value of -1 means infinite (default). This option implies the
1105 @option{rtsp_flags} set to @samp{listen}.
1107 @item reorder_queue_size
1108 Set number of packets to buffer for handling of reordered packets.
1111 Set socket TCP I/O timeout in microseconds.
1114 Override User-Agent header. If not specified, it defaults to the
1115 libavformat identifier string.
1118 When receiving data over UDP, the demuxer tries to reorder received packets
1119 (since they may arrive out of order, or packets may get lost totally). This
1120 can be disabled by setting the maximum demuxing delay to zero (via
1121 the @code{max_delay} field of AVFormatContext).
1123 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1124 streams to display can be chosen with @code{-vst} @var{n} and
1125 @code{-ast} @var{n} for video and audio respectively, and can be switched
1126 on the fly by pressing @code{v} and @code{a}.
1128 @subsection Examples
1130 The following examples all make use of the @command{ffplay} and
1131 @command{ffmpeg} tools.
1135 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1137 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1141 Watch a stream tunneled over HTTP:
1143 ffplay -rtsp_transport http rtsp://server/video.mp4
1147 Send a stream in realtime to a RTSP server, for others to watch:
1149 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1153 Receive a stream in realtime:
1155 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1161 Session Announcement Protocol (RFC 2974). This is not technically a
1162 protocol handler in libavformat, it is a muxer and demuxer.
1163 It is used for signalling of RTP streams, by announcing the SDP for the
1164 streams regularly on a separate port.
1168 The syntax for a SAP url given to the muxer is:
1170 sap://@var{destination}[:@var{port}][?@var{options}]
1173 The RTP packets are sent to @var{destination} on port @var{port},
1174 or to port 5004 if no port is specified.
1175 @var{options} is a @code{&}-separated list. The following options
1180 @item announce_addr=@var{address}
1181 Specify the destination IP address for sending the announcements to.
1182 If omitted, the announcements are sent to the commonly used SAP
1183 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1184 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1186 @item announce_port=@var{port}
1187 Specify the port to send the announcements on, defaults to
1188 9875 if not specified.
1191 Specify the time to live value for the announcements and RTP packets,
1194 @item same_port=@var{0|1}
1195 If set to 1, send all RTP streams on the same port pair. If zero (the
1196 default), all streams are sent on unique ports, with each stream on a
1197 port 2 numbers higher than the previous.
1198 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1199 The RTP stack in libavformat for receiving requires all streams to be sent
1203 Example command lines follow.
1205 To broadcast a stream on the local subnet, for watching in VLC:
1208 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1211 Similarly, for watching in @command{ffplay}:
1214 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1217 And for watching in @command{ffplay}, over IPv6:
1220 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1225 The syntax for a SAP url given to the demuxer is:
1227 sap://[@var{address}][:@var{port}]
1230 @var{address} is the multicast address to listen for announcements on,
1231 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1232 is the port that is listened on, 9875 if omitted.
1234 The demuxers listens for announcements on the given address and port.
1235 Once an announcement is received, it tries to receive that particular stream.
1237 Example command lines follow.
1239 To play back the first stream announced on the normal SAP multicast address:
1245 To play back the first stream announced on one the default IPv6 SAP multicast address:
1248 ffplay sap://[ff0e::2:7ffe]
1253 Stream Control Transmission Protocol.
1255 The accepted URL syntax is:
1257 sctp://@var{host}:@var{port}[?@var{options}]
1260 The protocol accepts the following options:
1263 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1266 Set the maximum number of streams. By default no limit is set.
1271 Haivision Secure Reliable Transport Protocol via libsrt.
1273 The supported syntax for a SRT URL is:
1275 srt://@var{hostname}:@var{port}[?@var{options}]
1278 @var{options} contains a list of &-separated options of the form
1279 @var{key}=@var{val}.
1284 @var{options} srt://@var{hostname}:@var{port}
1287 @var{options} contains a list of '-@var{key} @var{val}'
1290 This protocol accepts the following options.
1293 @item connect_timeout=@var{milliseconds}
1294 Connection timeout; SRT cannot connect for RTT > 1500 msec
1295 (2 handshake exchanges) with the default connect timeout of
1296 3 seconds. This option applies to the caller and rendezvous
1297 connection modes. The connect timeout is 10 times the value
1298 set for the rendezvous mode (which can be used as a
1299 workaround for this connection problem with earlier versions).
1301 @item ffs=@var{bytes}
1302 Flight Flag Size (Window Size), in bytes. FFS is actually an
1303 internal parameter and you should set it to not less than
1304 @option{recv_buffer_size} and @option{mss}. The default value
1305 is relatively large, therefore unless you set a very large receiver buffer,
1306 you do not need to change this option. Default value is 25600.
1308 @item inputbw=@var{bytes/seconds}
1309 Sender nominal input rate, in bytes per seconds. Used along with
1310 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1311 calculate maximum sending rate when recovery packets are sent
1312 along with the main media stream:
1313 @option{inputbw} * (100 + @option{oheadbw}) / 100
1314 if @option{inputbw} is not set while @option{maxbw} is set to
1315 relative (0), the actual input rate is evaluated inside
1316 the library. Default value is 0.
1318 @item iptos=@var{tos}
1319 IP Type of Service. Applies to sender only. Default value is 0xB8.
1321 @item ipttl=@var{ttl}
1322 IP Time To Live. Applies to sender only. Default value is 64.
1324 @item latency=@var{microseconds}
1325 Timestamp-based Packet Delivery Delay.
1326 Used to absorb bursts of missed packet retransmissions.
1327 This flag sets both @option{rcvlatency} and @option{peerlatency}
1328 to the same value. Note that prior to version 1.3.0
1329 this is the only flag to set the latency, however
1330 this is effectively equivalent to setting @option{peerlatency},
1331 when side is sender and @option{rcvlatency}
1332 when side is receiver, and the bidirectional stream
1333 sending is not supported.
1335 @item listen_timeout=@var{microseconds}
1336 Set socket listen timeout.
1338 @item maxbw=@var{bytes/seconds}
1339 Maximum sending bandwidth, in bytes per seconds.
1340 -1 infinite (CSRTCC limit is 30mbps)
1341 0 relative to input rate (see @option{inputbw})
1342 >0 absolute limit value
1343 Default value is 0 (relative)
1345 @item mode=@var{caller|listener|rendezvous}
1347 @option{caller} opens client connection.
1348 @option{listener} starts server to listen for incoming connections.
1349 @option{rendezvous} use Rendez-Vous connection mode.
1350 Default value is caller.
1352 @item mss=@var{bytes}
1353 Maximum Segment Size, in bytes. Used for buffer allocation
1354 and rate calculation using a packet counter assuming fully
1355 filled packets. The smallest MSS between the peers is
1356 used. This is 1500 by default in the overall internet.
1357 This is the maximum size of the UDP packet and can be
1358 only decreased, unless you have some unusual dedicated
1359 network settings. Default value is 1500.
1361 @item nakreport=@var{1|0}
1362 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1363 periodically until a lost packet is retransmitted or
1364 intentionally dropped. Default value is 1.
1366 @item oheadbw=@var{percents}
1367 Recovery bandwidth overhead above input rate, in percents.
1368 See @option{inputbw}. Default value is 25%.
1370 @item passphrase=@var{string}
1371 HaiCrypt Encryption/Decryption Passphrase string, length
1372 from 10 to 79 characters. The passphrase is the shared
1373 secret between the sender and the receiver. It is used
1374 to generate the Key Encrypting Key using PBKDF2
1375 (Password-Based Key Derivation Function). It is used
1376 only if @option{pbkeylen} is non-zero. It is used on
1377 the receiver only if the received data is encrypted.
1378 The configured passphrase cannot be recovered (write-only).
1380 @item enforced_encryption=@var{1|0}
1381 If true, both connection parties must have the same password
1382 set (including empty, that is, with no encryption). If the
1383 password doesn't match or only one side is unencrypted,
1384 the connection is rejected. Default is true.
1386 @item kmrefreshrate=@var{packets}
1387 The number of packets to be transmitted after which the
1388 encryption key is switched to a new key. Default is -1.
1389 -1 means auto (0x1000000 in srt library). The range for
1390 this option is integers in the 0 - @code{INT_MAX}.
1392 @item kmpreannounce=@var{packets}
1393 The interval between when a new encryption key is sent and
1394 when switchover occurs. This value also applies to the
1395 subsequent interval between when switchover occurs and
1396 when the old encryption key is decommissioned. Default is -1.
1397 -1 means auto (0x1000 in srt library). The range for
1398 this option is integers in the 0 - @code{INT_MAX}.
1400 @item payload_size=@var{bytes}
1401 Sets the maximum declared size of a packet transferred
1402 during the single call to the sending function in Live
1403 mode. Use 0 if this value isn't used (which is default in
1405 Default is -1 (automatic), which typically means MPEG-TS;
1406 if you are going to use SRT
1407 to send any different kind of payload, such as, for example,
1408 wrapping a live stream in very small frames, then you can
1409 use a bigger maximum frame size, though not greater than
1412 @item pkt_size=@var{bytes}
1413 Alias for @samp{payload_size}.
1415 @item peerlatency=@var{microseconds}
1416 The latency value (as described in @option{rcvlatency}) that is
1417 set by the sender side as a minimum value for the receiver.
1419 @item pbkeylen=@var{bytes}
1420 Sender encryption key length, in bytes.
1421 Only can be set to 0, 16, 24 and 32.
1422 Enable sender encryption if not 0.
1423 Not required on receiver (set to 0),
1424 key size obtained from sender in HaiCrypt handshake.
1427 @item rcvlatency=@var{microseconds}
1428 The time that should elapse since the moment when the
1429 packet was sent and the moment when it's delivered to
1430 the receiver application in the receiving function.
1431 This time should be a buffer time large enough to cover
1432 the time spent for sending, unexpectedly extended RTT
1433 time, and the time needed to retransmit the lost UDP
1434 packet. The effective latency value will be the maximum
1435 of this options' value and the value of @option{peerlatency}
1436 set by the peer side. Before version 1.3.0 this option
1437 is only available as @option{latency}.
1439 @item recv_buffer_size=@var{bytes}
1440 Set UDP receive buffer size, expressed in bytes.
1442 @item send_buffer_size=@var{bytes}
1443 Set UDP send buffer size, expressed in bytes.
1445 @item timeout=@var{microseconds}
1446 Set raise error timeouts for read, write and connect operations. Note that the
1447 SRT library has internal timeouts which can be controlled separately, the
1448 value set here is only a cap on those.
1450 @item tlpktdrop=@var{1|0}
1451 Too-late Packet Drop. When enabled on receiver, it skips
1452 missing packets that have not been delivered in time and
1453 delivers the following packets to the application when
1454 their time-to-play has come. It also sends a fake ACK to
1455 the sender. When enabled on sender and enabled on the
1456 receiving peer, the sender drops the older packets that
1457 have no chance of being delivered in time. It was
1458 automatically enabled in the sender if the receiver
1461 @item sndbuf=@var{bytes}
1462 Set send buffer size, expressed in bytes.
1464 @item rcvbuf=@var{bytes}
1465 Set receive buffer size, expressed in bytes.
1467 Receive buffer must not be greater than @option{ffs}.
1469 @item lossmaxttl=@var{packets}
1470 The value up to which the Reorder Tolerance may grow. When
1471 Reorder Tolerance is > 0, then packet loss report is delayed
1472 until that number of packets come in. Reorder Tolerance
1473 increases every time a "belated" packet has come, but it
1474 wasn't due to retransmission (that is, when UDP packets tend
1475 to come out of order), with the difference between the latest
1476 sequence and this packet's sequence, and not more than the
1477 value of this option. By default it's 0, which means that this
1478 mechanism is turned off, and the loss report is always sent
1479 immediately upon experiencing a "gap" in sequences.
1482 The minimum SRT version that is required from the peer. A connection
1483 to a peer that does not satisfy the minimum version requirement
1486 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1489 @item streamid=@var{string}
1490 A string limited to 512 characters that can be set on the socket prior
1491 to connecting. This stream ID will be able to be retrieved by the
1492 listener side from the socket that is returned from srt_accept and
1493 was connected by a socket with that set stream ID. SRT does not enforce
1494 any special interpretation of the contents of this string.
1495 This option doesn’t make sense in Rendezvous connection; the result
1496 might be that simply one side will override the value from the other
1497 side and it’s the matter of luck which one would win
1499 @item smoother=@var{live|file}
1500 The type of Smoother used for the transmission for that socket, which
1501 is responsible for the transmission and congestion control. The Smoother
1502 type must be exactly the same on both connecting parties, otherwise
1503 the connection is rejected.
1505 @item messageapi=@var{1|0}
1506 When set, this socket uses the Message API, otherwise it uses Buffer
1507 API. Note that in live mode (see @option{transtype}) there’s only
1508 message API available. In File mode you can chose to use one of two modes:
1510 Stream API (default, when this option is false). In this mode you may
1511 send as many data as you wish with one sending instruction, or even use
1512 dedicated functions that read directly from a file. The internal facility
1513 will take care of any speed and congestion control. When receiving, you
1514 can also receive as many data as desired, the data not extracted will be
1515 waiting for the next call. There is no boundary between data portions in
1518 Message API. In this mode your single sending instruction passes exactly
1519 one piece of data that has boundaries (a message). Contrary to Live mode,
1520 this message may span across multiple UDP packets and the only size
1521 limitation is that it shall fit as a whole in the sending buffer. The
1522 receiver shall use as large buffer as necessary to receive the message,
1523 otherwise the message will not be given up. When the message is not
1524 complete (not all packets received or there was a packet loss) it will
1527 @item transtype=@var{live|file}
1528 Sets the transmission type for the socket, in particular, setting this
1529 option sets multiple other parameters to their default values as required
1530 for a particular transmission type.
1532 live: Set options as for live transmission. In this mode, you should
1533 send by one sending instruction only so many data that fit in one UDP packet,
1534 and limited to the value defined first in @option{payload_size} (1316 is
1535 default in this mode). There is no speed control in this mode, only the
1536 bandwidth control, if configured, in order to not exceed the bandwidth with
1537 the overhead transmission (retransmitted and control packets).
1539 file: Set options as for non-live transmission. See @option{messageapi}
1540 for further explanations
1542 @item linger=@var{seconds}
1543 The number of seconds that the socket waits for unsent data when closing.
1544 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
1545 seconds in file mode). The range for this option is integers in the
1550 For more information see: @url{https://github.com/Haivision/srt}.
1554 Secure Real-time Transport Protocol.
1556 The accepted options are:
1559 @item srtp_out_suite
1560 Select input and output encoding suites.
1564 @item AES_CM_128_HMAC_SHA1_80
1565 @item SRTP_AES128_CM_HMAC_SHA1_80
1566 @item AES_CM_128_HMAC_SHA1_32
1567 @item SRTP_AES128_CM_HMAC_SHA1_32
1570 @item srtp_in_params
1571 @item srtp_out_params
1572 Set input and output encoding parameters, which are expressed by a
1573 base64-encoded representation of a binary block. The first 16 bytes of
1574 this binary block are used as master key, the following 14 bytes are
1575 used as master salt.
1580 Virtually extract a segment of a file or another stream.
1581 The underlying stream must be seekable.
1586 Start offset of the extracted segment, in bytes.
1588 End offset of the extracted segment, in bytes.
1589 If set to 0, extract till end of file.
1594 Extract a chapter from a DVD VOB file (start and end sectors obtained
1595 externally and multiplied by 2048):
1597 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1600 Play an AVI file directly from a TAR archive:
1602 subfile,,start,183241728,end,366490624,,:archive.tar
1605 Play a MPEG-TS file from start offset till end:
1607 subfile,,start,32815239,end,0,,:video.ts
1612 Writes the output to multiple protocols. The individual outputs are separated
1616 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1621 Transmission Control Protocol.
1623 The required syntax for a TCP url is:
1625 tcp://@var{hostname}:@var{port}[?@var{options}]
1628 @var{options} contains a list of &-separated options of the form
1629 @var{key}=@var{val}.
1631 The list of supported options follows.
1634 @item listen=@var{1|0}
1635 Listen for an incoming connection. Default value is 0.
1637 @item timeout=@var{microseconds}
1638 Set raise error timeout, expressed in microseconds.
1640 This option is only relevant in read mode: if no data arrived in more
1641 than this time interval, raise error.
1643 @item listen_timeout=@var{milliseconds}
1644 Set listen timeout, expressed in milliseconds.
1646 @item recv_buffer_size=@var{bytes}
1647 Set receive buffer size, expressed bytes.
1649 @item send_buffer_size=@var{bytes}
1650 Set send buffer size, expressed bytes.
1652 @item tcp_nodelay=@var{1|0}
1653 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1655 @item tcp_mss=@var{bytes}
1656 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1659 The following example shows how to setup a listening TCP connection
1660 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1662 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1663 ffplay tcp://@var{hostname}:@var{port}
1668 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1670 The required syntax for a TLS/SSL url is:
1672 tls://@var{hostname}:@var{port}[?@var{options}]
1675 The following parameters can be set via command line options
1676 (or in code via @code{AVOption}s):
1680 @item ca_file, cafile=@var{filename}
1681 A file containing certificate authority (CA) root certificates to treat
1682 as trusted. If the linked TLS library contains a default this might not
1683 need to be specified for verification to work, but not all libraries and
1684 setups have defaults built in.
1685 The file must be in OpenSSL PEM format.
1687 @item tls_verify=@var{1|0}
1688 If enabled, try to verify the peer that we are communicating with.
1689 Note, if using OpenSSL, this currently only makes sure that the
1690 peer certificate is signed by one of the root certificates in the CA
1691 database, but it does not validate that the certificate actually
1692 matches the host name we are trying to connect to. (With other backends,
1693 the host name is validated as well.)
1695 This is disabled by default since it requires a CA database to be
1696 provided by the caller in many cases.
1698 @item cert_file, cert=@var{filename}
1699 A file containing a certificate to use in the handshake with the peer.
1700 (When operating as server, in listen mode, this is more often required
1701 by the peer, while client certificates only are mandated in certain
1704 @item key_file, key=@var{filename}
1705 A file containing the private key for the certificate.
1707 @item listen=@var{1|0}
1708 If enabled, listen for connections on the provided port, and assume
1709 the server role in the handshake instead of the client role.
1713 Example command lines:
1715 To create a TLS/SSL server that serves an input stream.
1718 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1721 To play back a stream from the TLS/SSL server using @command{ffplay}:
1724 ffplay tls://@var{hostname}:@var{port}
1729 User Datagram Protocol.
1731 The required syntax for an UDP URL is:
1733 udp://@var{hostname}:@var{port}[?@var{options}]
1736 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1738 In case threading is enabled on the system, a circular buffer is used
1739 to store the incoming data, which allows one to reduce loss of data due to
1740 UDP socket buffer overruns. The @var{fifo_size} and
1741 @var{overrun_nonfatal} options are related to this buffer.
1743 The list of supported options follows.
1746 @item buffer_size=@var{size}
1747 Set the UDP maximum socket buffer size in bytes. This is used to set either
1748 the receive or send buffer size, depending on what the socket is used for.
1749 Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
1751 @item bitrate=@var{bitrate}
1752 If set to nonzero, the output will have the specified constant bitrate if the
1753 input has enough packets to sustain it.
1755 @item burst_bits=@var{bits}
1756 When using @var{bitrate} this specifies the maximum number of bits in
1759 @item localport=@var{port}
1760 Override the local UDP port to bind with.
1762 @item localaddr=@var{addr}
1763 Local IP address of a network interface used for sending packets or joining
1766 @item pkt_size=@var{size}
1767 Set the size in bytes of UDP packets.
1769 @item reuse=@var{1|0}
1770 Explicitly allow or disallow reusing UDP sockets.
1773 Set the time to live value (for multicast only).
1775 @item connect=@var{1|0}
1776 Initialize the UDP socket with @code{connect()}. In this case, the
1777 destination address can't be changed with ff_udp_set_remote_url later.
1778 If the destination address isn't known at the start, this option can
1779 be specified in ff_udp_set_remote_url, too.
1780 This allows finding out the source address for the packets with getsockname,
1781 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1782 unreachable" is received.
1783 For receiving, this gives the benefit of only receiving packets from
1784 the specified peer address/port.
1786 @item sources=@var{address}[,@var{address}]
1787 Only receive packets sent from the specified addresses. In case of multicast,
1788 also subscribe to multicast traffic coming from these addresses only.
1790 @item block=@var{address}[,@var{address}]
1791 Ignore packets sent from the specified addresses. In case of multicast, also
1792 exclude the source addresses in the multicast subscription.
1794 @item fifo_size=@var{units}
1795 Set the UDP receiving circular buffer size, expressed as a number of
1796 packets with size of 188 bytes. If not specified defaults to 7*4096.
1798 @item overrun_nonfatal=@var{1|0}
1799 Survive in case of UDP receiving circular buffer overrun. Default
1802 @item timeout=@var{microseconds}
1803 Set raise error timeout, expressed in microseconds.
1805 This option is only relevant in read mode: if no data arrived in more
1806 than this time interval, raise error.
1808 @item broadcast=@var{1|0}
1809 Explicitly allow or disallow UDP broadcasting.
1811 Note that broadcasting may not work properly on networks having
1812 a broadcast storm protection.
1815 @subsection Examples
1819 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1821 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1825 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1826 sized UDP packets, using a large input buffer:
1828 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1832 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1834 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1842 The required syntax for a Unix socket URL is:
1845 unix://@var{filepath}
1848 The following parameters can be set via command line options
1849 (or in code via @code{AVOption}s):
1855 Create the Unix socket in listening mode.
1860 ZeroMQ asynchronous messaging using the libzmq library.
1862 This library supports unicast streaming to multiple clients without relying on
1865 The required syntax for streaming or connecting to a stream is:
1867 zmq:tcp://ip-address:port
1871 Create a localhost stream on port 5555:
1873 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1876 Multiple clients may connect to the stream using:
1878 ffplay zmq:tcp://127.0.0.1:5555
1881 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
1882 The server side binds to a port and publishes data. Clients connect to the
1883 server (via IP address/port) and subscribe to the stream. The order in which
1884 the server and client start generally does not matter.
1886 ffmpeg must be compiled with the --enable-libzmq option to support
1889 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1890 line. The following options are supported:
1895 Forces the maximum packet size for sending/receiving data. The default value is
1896 131,072 bytes. On the server side, this sets the maximum size of sent packets
1897 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
1898 packets. Note that pkt_size on the clients should be equal to or greater than
1899 pkt_size on the server. Otherwise the received message may be truncated causing
1905 @c man end PROTOCOLS