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][/vhost]
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
71 fields is "guest". Name of virtual host on broker can be set with vhost. The
74 Muliple subscribers may stream from the broker using the command:
76 ffplay amqp://[[user]:[password]@@]hostname[:port][/vhost]
79 In RabbitMQ all data published to the broker flows through a specific exchange,
80 and each subscribing client has an assigned queue/buffer. When a packet arrives
81 at an exchange, it may be copied to a client's queue depending on the exchange
82 and routing_key fields.
84 The following options are supported:
89 Sets the exchange to use on the broker. RabbitMQ has several predefined
90 exchanges: "amq.direct" is the default exchange, where the publisher and
91 subscriber must have a matching routing_key; "amq.fanout" is the same as a
92 broadcast operation (i.e. the data is forwarded to all queues on the fanout
93 exchange independent of the routing_key); and "amq.topic" is similar to
94 "amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ
98 Sets the routing key. The default value is "amqp". The routing key is used on
99 the "amq.direct" and "amq.topic" exchanges to decide whether packets are written
100 to the queue of a subscriber.
103 Maximum size of each packet sent/received to the broker. Default is 131072.
104 Minimum is 4096 and max is any large value (representable by an int). When
105 receiving packets, this sets an internal buffer size in FFmpeg. It should be
106 equal to or greater than the size of the published packets to the broker. Otherwise
107 the received message may be truncated causing decoding errors.
109 @item connection_timeout
110 The timeout in seconds during the initial connection to the broker. The
111 default value is rw_timeout, or 5 seconds if rw_timeout is not set.
113 @item delivery_mode @var{mode}
114 Sets the delivery mode of each message sent to broker.
115 The following values are accepted:
118 Delivery mode set to "persistent" (2). This is the default value.
119 Messages may be written to the broker's disk depending on its setup.
122 Delivery mode set to "non-persistent" (1).
123 Messages will stay in broker's memory unless the broker is under memory
132 Asynchronous data filling wrapper for input stream.
134 Fill data in a background thread, to decouple I/O operation from demux thread.
138 async:http://host/resource
139 async:cache:http://host/resource
144 Read BluRay playlist.
146 The accepted options are:
153 Start chapter (1...N)
156 Playlist to read (BDMV/PLAYLIST/?????.mpls)
162 Read longest playlist from BluRay mounted to /mnt/bluray:
167 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
169 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
174 Caching wrapper for input stream.
176 Cache the input stream to temporary file. It brings seeking capability to live streams.
184 Physical concatenation protocol.
186 Read and seek from many resources in sequence as if they were
189 A URL accepted by this protocol has the syntax:
191 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
194 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
195 resource to be concatenated, each one possibly specifying a distinct
198 For example to read a sequence of files @file{split1.mpeg},
199 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
202 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
205 Note that you may need to escape the character "|" which is special for
210 AES-encrypted stream reading protocol.
212 The accepted options are:
215 Set the AES decryption key binary block from given hexadecimal representation.
218 Set the AES decryption initialization vector binary block from given hexadecimal representation.
221 Accepted URL formats:
229 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
231 For example, to convert a GIF file given inline with @command{ffmpeg}:
233 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
238 File access protocol.
240 Read from or write to a file.
242 A file URL can have the form:
247 where @var{filename} is the path of the file to read.
249 An URL that does not have a protocol prefix will be assumed to be a
250 file URL. Depending on the build, an URL that looks like a Windows
251 path with the drive letter at the beginning will also be assumed to be
252 a file URL (usually not the case in builds for unix-like systems).
254 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
257 ffmpeg -i file:input.mpeg output.mpeg
260 This protocol accepts the following options:
264 Truncate existing files on write, if set to 1. A value of 0 prevents
265 truncating. Default value is 1.
268 Set I/O operation maximum block size, in bytes. Default value is
269 @code{INT_MAX}, which results in not limiting the requested block size.
270 Setting this value reasonably low improves user termination request reaction
271 time, which is valuable for files on slow medium.
274 If set to 1, the protocol will retry reading at the end of the file, allowing
275 reading files that still are being written. In order for this to terminate,
276 you either need to use the rw_timeout option, or use the interrupt callback
280 Controls if seekability is advertised on the file. 0 means non-seekable, -1
281 means auto (seekable for normal files, non-seekable for named pipes).
283 Many demuxers handle seekable and non-seekable resources differently,
284 overriding this might speed up opening certain files at the cost of losing some
285 features (e.g. accurate seeking).
290 FTP (File Transfer Protocol).
292 Read from or write to remote resources using FTP protocol.
294 Following syntax is required.
296 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
299 This protocol accepts the following options.
303 Set timeout in microseconds of socket I/O operations used by the underlying low level
304 operation. By default it is set to -1, which means that the timeout is
308 Set a user to be used for authenticating to the FTP server. This is overridden by the
312 Set a password to be used for authenticating to the FTP server. This is overridden by
313 the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
315 @item ftp-anonymous-password
316 Password used when login as anonymous user. Typically an e-mail address
319 @item ftp-write-seekable
320 Control seekability of connection during encoding. If set to 1 the
321 resource is supposed to be seekable, if set to 0 it is assumed not
322 to be seekable. Default value is 0.
325 NOTE: Protocol can be used as output, but it is recommended to not do
326 it, unless special care is taken (tests, customized server configuration
327 etc.). Different FTP servers behave in different way during seek
328 operation. ff* tools may produce incomplete content due to server limitations.
336 Read Apple HTTP Live Streaming compliant segmented stream as
337 a uniform one. The M3U8 playlists describing the segments can be
338 remote HTTP resources or local files, accessed using the standard
340 The nested protocol is declared by specifying
341 "+@var{proto}" after the hls URI scheme name, where @var{proto}
342 is either "file" or "http".
345 hls+http://host/path/to/remote/resource.m3u8
346 hls+file://path/to/local/resource.m3u8
349 Using this protocol is discouraged - the hls demuxer should work
350 just as well (if not, please report the issues) and is more complete.
351 To use the hls demuxer instead, simply use the direct URLs to the
356 HTTP (Hyper Text Transfer Protocol).
358 This protocol accepts the following options:
362 Control seekability of connection. If set to 1 the resource is
363 supposed to be seekable, if set to 0 it is assumed not to be seekable,
364 if set to -1 it will try to autodetect if it is seekable. Default
368 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
371 Set a specific content type for the POST messages or for listen mode.
374 set HTTP proxy to tunnel through e.g. http://example.com:1234
377 Set custom HTTP headers, can override built in default headers. The
378 value must be a string encoding the headers.
380 @item multiple_requests
381 Use persistent connections if set to 1, default is 0.
384 Set custom HTTP post data.
387 Set the Referer header. Include 'Referer: URL' header in HTTP request.
390 Override the User-Agent header. If not specified the protocol will use a
391 string describing the libavformat build. ("Lavf/<version>")
394 This is a deprecated option, you can use user_agent instead it.
396 @item reconnect_at_eof
397 If set then eof is treated like an error and causes reconnection, this is useful
398 for live / endless streams.
400 @item reconnect_streamed
401 If set then even streamed/non seekable streams will be reconnected on errors.
403 @item reconnect_delay_max
404 Sets the maximum delay in seconds after which to give up reconnecting
407 Export the MIME type.
410 Exports the HTTP response version number. Usually "1.0" or "1.1".
413 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
414 supports this, the metadata has to be retrieved by the application by reading
415 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
418 @item icy_metadata_headers
419 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
420 headers, separated by newline characters.
422 @item icy_metadata_packet
423 If the server supports ICY metadata, and @option{icy} was set to 1, this
424 contains the last non-empty metadata packet sent by the server. It should be
425 polled in regular intervals by applications interested in mid-stream metadata
429 Set the cookies to be sent in future requests. The format of each cookie is the
430 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
431 delimited by a newline character.
434 Set initial byte offset.
437 Try to limit the request to bytes preceding this offset.
440 When used as a client option it sets the HTTP method for the request.
442 When used as a server option it sets the HTTP method that is going to be
443 expected from the client(s).
444 If the expected and the received HTTP method do not match the client will
445 be given a Bad Request response.
446 When unset the HTTP method is not checked for now. This will be replaced by
447 autodetection in the future.
450 If set to 1 enables experimental HTTP server. This can be used to send data when
451 used as an output option, or read data from a client with HTTP POST when used as
453 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
454 in ffmpeg.c and thus must not be used as a command line option.
456 # Server side (sending):
457 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
459 # Client side (receiving):
460 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
462 # Client can also be done with wget:
463 wget http://@var{server}:@var{port} -O somefile.ogg
465 # Server side (receiving):
466 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
468 # Client side (sending):
469 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
471 # Client can also be done with wget:
472 wget --post-file=somefile.ogg http://@var{server}:@var{port}
475 @item send_expect_100
476 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
477 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
482 Set HTTP authentication type. No option for Digest, since this method requires
483 getting nonce parameters from the server first and can't be used straight away like
488 Choose the HTTP authentication type automatically. This is the default.
491 Choose the HTTP basic authentication.
493 Basic authentication sends a Base64-encoded string that contains a user name and password
494 for the client. Base64 is not a form of encryption and should be considered the same as
495 sending the user name and password in clear text (Base64 is a reversible encoding).
496 If a resource needs to be protected, strongly consider using an authentication scheme
497 other than basic authentication. HTTPS/TLS should be used with basic authentication.
498 Without these additional security enhancements, basic authentication should not be used
499 to protect sensitive or valuable information.
504 @subsection HTTP Cookies
506 Some HTTP requests will be denied unless cookie values are passed in with the
507 request. The @option{cookies} option allows these cookies to be specified. At
508 the very least, each cookie must specify a value along with a path and domain.
509 HTTP requests that match both the domain and path will automatically include the
510 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
513 The required syntax to play a stream specifying a cookie is:
515 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
520 Icecast protocol (stream to Icecast servers)
522 This protocol accepts the following options:
526 Set the stream genre.
531 @item ice_description
532 Set the stream description.
535 Set the stream website URL.
538 Set if the stream should be public.
539 The default is 0 (not public).
542 Override the User-Agent header. If not specified a string of the form
543 "Lavf/<version>" will be used.
546 Set the Icecast mountpoint password.
549 Set the stream content type. This must be set if it is different from
553 This enables support for Icecast versions < 2.4.0, that do not support the
554 HTTP PUT method but the SOURCE method.
557 Establish a TLS (HTTPS) connection to Icecast.
562 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
567 MMS (Microsoft Media Server) protocol over TCP.
571 MMS (Microsoft Media Server) protocol over HTTP.
573 The required syntax is:
575 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
582 Computes the MD5 hash of the data to be written, and on close writes
583 this to the designated output or stdout if none is specified. It can
584 be used to test muxers without writing an actual file.
586 Some examples follow.
588 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
589 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
591 # Write the MD5 hash of the encoded AVI file to stdout.
592 ffmpeg -i input.flv -f avi -y md5:
595 Note that some formats (typically MOV) require the output protocol to
596 be seekable, so they will fail with the MD5 output protocol.
600 UNIX pipe access protocol.
602 Read and write from UNIX pipes.
604 The accepted syntax is:
609 @var{number} is the number corresponding to the file descriptor of the
610 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
611 is not specified, by default the stdout file descriptor will be used
612 for writing, stdin for reading.
614 For example to read from stdin with @command{ffmpeg}:
616 cat test.wav | ffmpeg -i pipe:0
617 # ...this is the same as...
618 cat test.wav | ffmpeg -i pipe:
621 For writing to stdout with @command{ffmpeg}:
623 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
624 # ...this is the same as...
625 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
628 This protocol accepts the following options:
632 Set I/O operation maximum block size, in bytes. Default value is
633 @code{INT_MAX}, which results in not limiting the requested block size.
634 Setting this value reasonably low improves user termination request reaction
635 time, which is valuable if data transmission is slow.
638 Note that some formats (typically MOV), require the output protocol to
639 be seekable, so they will fail with the pipe output protocol.
643 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
645 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
646 for MPEG-2 Transport Streams sent over RTP.
648 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
649 the @code{rtp} protocol.
651 The required syntax is:
653 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
656 The destination UDP ports are @code{port + 2} for the column FEC stream
657 and @code{port + 4} for the row FEC stream.
659 This protocol accepts the following options:
663 The number of columns (4-20, LxD <= 100)
666 The number of rows (4-20, LxD <= 100)
673 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
678 Real-Time Messaging Protocol.
680 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
681 content across a TCP/IP network.
683 The required syntax is:
685 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
688 The accepted parameters are:
692 An optional username (mostly for publishing).
695 An optional password (mostly for publishing).
698 The address of the RTMP server.
701 The number of the TCP port to use (by default is 1935).
704 It is the name of the application to access. It usually corresponds to
705 the path where the application is installed on the RTMP server
706 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
707 the value parsed from the URI through the @code{rtmp_app} option, too.
710 It is the path or name of the resource to play with reference to the
711 application specified in @var{app}, may be prefixed by "mp4:". You
712 can override the value parsed from the URI through the @code{rtmp_playpath}
716 Act as a server, listening for an incoming connection.
719 Maximum time to wait for the incoming connection. Implies listen.
722 Additionally, the following parameters can be set via command line options
723 (or in code via @code{AVOption}s):
727 Name of application to connect on the RTMP server. This option
728 overrides the parameter specified in the URI.
731 Set the client buffer time in milliseconds. The default is 3000.
734 Extra arbitrary AMF connection parameters, parsed from a string,
735 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
736 Each value is prefixed by a single character denoting the type,
737 B for Boolean, N for number, S for string, O for object, or Z for null,
738 followed by a colon. For Booleans the data must be either 0 or 1 for
739 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
740 1 to end or begin an object, respectively. Data items in subobjects may
741 be named, by prefixing the type with 'N' and specifying the name before
742 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
743 times to construct arbitrary AMF sequences.
746 Version of the Flash plugin used to run the SWF player. The default
747 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
748 <libavformat version>).)
750 @item rtmp_flush_interval
751 Number of packets flushed in the same request (RTMPT only). The default
755 Specify that the media is a live stream. No resuming or seeking in
756 live streams is possible. The default value is @code{any}, which means the
757 subscriber first tries to play the live stream specified in the
758 playpath. If a live stream of that name is not found, it plays the
759 recorded stream. The other possible values are @code{live} and
763 URL of the web page in which the media was embedded. By default no
767 Stream identifier to play or to publish. This option overrides the
768 parameter specified in the URI.
771 Name of live stream to subscribe to. By default no value will be sent.
772 It is only sent if the option is specified or if rtmp_live
776 SHA256 hash of the decompressed SWF file (32 bytes).
779 Size of the decompressed SWF file, required for SWFVerification.
782 URL of the SWF player for the media. By default no value will be sent.
785 URL to player swf file, compute hash/size automatically.
788 URL of the target stream. Defaults to proto://host[:port]/app.
792 For example to read with @command{ffplay} a multimedia resource named
793 "sample" from the application "vod" from an RTMP server "myserver":
795 ffplay rtmp://myserver/vod/sample
798 To publish to a password protected server, passing the playpath and
799 app names separately:
801 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
806 Encrypted Real-Time Messaging Protocol.
808 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
809 streaming multimedia content within standard cryptographic primitives,
810 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
815 Real-Time Messaging Protocol over a secure SSL connection.
817 The Real-Time Messaging Protocol (RTMPS) is used for streaming
818 multimedia content across an encrypted connection.
822 Real-Time Messaging Protocol tunneled through HTTP.
824 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
825 for streaming multimedia content within HTTP requests to traverse
830 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
832 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
833 is used for streaming multimedia content within HTTP requests to traverse
838 Real-Time Messaging Protocol tunneled through HTTPS.
840 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
841 for streaming multimedia content within HTTPS requests to traverse
844 @section libsmbclient
846 libsmbclient permits one to manipulate CIFS/SMB network resources.
848 Following syntax is required.
851 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
854 This protocol accepts the following options.
858 Set timeout in milliseconds of socket I/O operations used by the underlying
859 low level operation. By default it is set to -1, which means that the timeout
863 Truncate existing files on write, if set to 1. A value of 0 prevents
864 truncating. Default value is 1.
867 Set the workgroup used for making connections. By default workgroup is not specified.
871 For more information see: @url{http://www.samba.org/}.
875 Secure File Transfer Protocol via libssh
877 Read from or write to remote resources using SFTP protocol.
879 Following syntax is required.
882 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
885 This protocol accepts the following options.
889 Set timeout of socket I/O operations used by the underlying low level
890 operation. By default it is set to -1, which means that the timeout
894 Truncate existing files on write, if set to 1. A value of 0 prevents
895 truncating. Default value is 1.
898 Specify the path of the file containing private key to use during authorization.
899 By default libssh searches for keys in the @file{~/.ssh/} directory.
903 Example: Play a file stored on remote server.
906 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
909 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
911 Real-Time Messaging Protocol and its variants supported through
914 Requires the presence of the librtmp headers and library during
915 configuration. You need to explicitly configure the build with
916 "--enable-librtmp". If enabled this will replace the native RTMP
919 This protocol provides most client functions and a few server
920 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
921 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
922 variants of these encrypted types (RTMPTE, RTMPTS).
924 The required syntax is:
926 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
929 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
930 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
931 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
932 meaning as specified for the RTMP native protocol.
933 @var{options} contains a list of space-separated options of the form
936 See the librtmp manual page (man 3 librtmp) for more information.
938 For example, to stream a file in real-time to an RTMP server using
941 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
944 To play the same stream using @command{ffplay}:
946 ffplay "rtmp://myserver/live/mystream live=1"
951 Real-time Transport Protocol.
953 The required syntax for an RTP URL is:
954 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
956 @var{port} specifies the RTP port to use.
958 The following URL options are supported:
963 Set the TTL (Time-To-Live) value (for multicast only).
965 @item rtcpport=@var{n}
966 Set the remote RTCP port to @var{n}.
968 @item localrtpport=@var{n}
969 Set the local RTP port to @var{n}.
971 @item localrtcpport=@var{n}'
972 Set the local RTCP port to @var{n}.
974 @item pkt_size=@var{n}
975 Set max packet size (in bytes) to @var{n}.
978 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
981 @item sources=@var{ip}[,@var{ip}]
982 List allowed source IP addresses.
984 @item block=@var{ip}[,@var{ip}]
985 List disallowed (blocked) source IP addresses.
987 @item write_to_source=0|1
988 Send packets to the source address of the latest received packet (if
989 set to 1) or to a default remote address (if set to 0).
991 @item localport=@var{n}
992 Set the local RTP port to @var{n}.
994 @item timeout=@var{n}
995 Set timeout (in microseconds) of socket I/O operations to @var{n}.
997 This is a deprecated option. Instead, @option{localrtpport} should be
1007 If @option{rtcpport} is not set the RTCP port will be set to the RTP
1011 If @option{localrtpport} (the local RTP port) is not set any available
1012 port will be used for the local RTP and RTCP ports.
1015 If @option{localrtcpport} (the local RTCP port) is not set it will be
1016 set to the local RTP port value plus 1.
1021 Real-Time Streaming Protocol.
1023 RTSP is not technically a protocol handler in libavformat, it is a demuxer
1024 and muxer. The demuxer supports both normal RTSP (with data transferred
1025 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
1026 data transferred over RDT).
1028 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1029 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
1030 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
1032 The required syntax for a RTSP url is:
1034 rtsp://@var{hostname}[:@var{port}]/@var{path}
1037 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1038 line, or set in code via @code{AVOption}s or in
1039 @code{avformat_open_input}.
1041 The following options are supported.
1045 Do not start playing the stream immediately if set to 1. Default value
1048 @item rtsp_transport
1049 Set RTSP transport protocols.
1051 It accepts the following values:
1054 Use UDP as lower transport protocol.
1057 Use TCP (interleaving within the RTSP control channel) as lower
1061 Use UDP multicast as lower transport protocol.
1064 Use HTTP tunneling as lower transport protocol, which is useful for
1068 Multiple lower transport protocols may be specified, in that case they are
1069 tried one at a time (if the setup of one fails, the next one is tried).
1070 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
1075 The following values are accepted:
1078 Accept packets only from negotiated peer address and port.
1080 Act as a server, listening for an incoming connection.
1082 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
1085 Default value is @samp{none}.
1087 @item allowed_media_types
1088 Set media types to accept from the server.
1090 The following flags are accepted:
1097 By default it accepts all media types.
1100 Set minimum local UDP port. Default value is 5000.
1103 Set maximum local UDP port. Default value is 65000.
1106 Set maximum timeout (in seconds) to wait for incoming connections.
1108 A value of -1 means infinite (default). This option implies the
1109 @option{rtsp_flags} set to @samp{listen}.
1111 @item reorder_queue_size
1112 Set number of packets to buffer for handling of reordered packets.
1115 Set socket TCP I/O timeout in microseconds.
1118 Override User-Agent header. If not specified, it defaults to the
1119 libavformat identifier string.
1122 When receiving data over UDP, the demuxer tries to reorder received packets
1123 (since they may arrive out of order, or packets may get lost totally). This
1124 can be disabled by setting the maximum demuxing delay to zero (via
1125 the @code{max_delay} field of AVFormatContext).
1127 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1128 streams to display can be chosen with @code{-vst} @var{n} and
1129 @code{-ast} @var{n} for video and audio respectively, and can be switched
1130 on the fly by pressing @code{v} and @code{a}.
1132 @subsection Examples
1134 The following examples all make use of the @command{ffplay} and
1135 @command{ffmpeg} tools.
1139 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1141 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1145 Watch a stream tunneled over HTTP:
1147 ffplay -rtsp_transport http rtsp://server/video.mp4
1151 Send a stream in realtime to a RTSP server, for others to watch:
1153 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1157 Receive a stream in realtime:
1159 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1165 Session Announcement Protocol (RFC 2974). This is not technically a
1166 protocol handler in libavformat, it is a muxer and demuxer.
1167 It is used for signalling of RTP streams, by announcing the SDP for the
1168 streams regularly on a separate port.
1172 The syntax for a SAP url given to the muxer is:
1174 sap://@var{destination}[:@var{port}][?@var{options}]
1177 The RTP packets are sent to @var{destination} on port @var{port},
1178 or to port 5004 if no port is specified.
1179 @var{options} is a @code{&}-separated list. The following options
1184 @item announce_addr=@var{address}
1185 Specify the destination IP address for sending the announcements to.
1186 If omitted, the announcements are sent to the commonly used SAP
1187 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1188 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1190 @item announce_port=@var{port}
1191 Specify the port to send the announcements on, defaults to
1192 9875 if not specified.
1195 Specify the time to live value for the announcements and RTP packets,
1198 @item same_port=@var{0|1}
1199 If set to 1, send all RTP streams on the same port pair. If zero (the
1200 default), all streams are sent on unique ports, with each stream on a
1201 port 2 numbers higher than the previous.
1202 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1203 The RTP stack in libavformat for receiving requires all streams to be sent
1207 Example command lines follow.
1209 To broadcast a stream on the local subnet, for watching in VLC:
1212 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1215 Similarly, for watching in @command{ffplay}:
1218 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1221 And for watching in @command{ffplay}, over IPv6:
1224 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1229 The syntax for a SAP url given to the demuxer is:
1231 sap://[@var{address}][:@var{port}]
1234 @var{address} is the multicast address to listen for announcements on,
1235 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1236 is the port that is listened on, 9875 if omitted.
1238 The demuxers listens for announcements on the given address and port.
1239 Once an announcement is received, it tries to receive that particular stream.
1241 Example command lines follow.
1243 To play back the first stream announced on the normal SAP multicast address:
1249 To play back the first stream announced on one the default IPv6 SAP multicast address:
1252 ffplay sap://[ff0e::2:7ffe]
1257 Stream Control Transmission Protocol.
1259 The accepted URL syntax is:
1261 sctp://@var{host}:@var{port}[?@var{options}]
1264 The protocol accepts the following options:
1267 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1270 Set the maximum number of streams. By default no limit is set.
1275 Haivision Secure Reliable Transport Protocol via libsrt.
1277 The supported syntax for a SRT URL is:
1279 srt://@var{hostname}:@var{port}[?@var{options}]
1282 @var{options} contains a list of &-separated options of the form
1283 @var{key}=@var{val}.
1288 @var{options} srt://@var{hostname}:@var{port}
1291 @var{options} contains a list of '-@var{key} @var{val}'
1294 This protocol accepts the following options.
1297 @item connect_timeout=@var{milliseconds}
1298 Connection timeout; SRT cannot connect for RTT > 1500 msec
1299 (2 handshake exchanges) with the default connect timeout of
1300 3 seconds. This option applies to the caller and rendezvous
1301 connection modes. The connect timeout is 10 times the value
1302 set for the rendezvous mode (which can be used as a
1303 workaround for this connection problem with earlier versions).
1305 @item ffs=@var{bytes}
1306 Flight Flag Size (Window Size), in bytes. FFS is actually an
1307 internal parameter and you should set it to not less than
1308 @option{recv_buffer_size} and @option{mss}. The default value
1309 is relatively large, therefore unless you set a very large receiver buffer,
1310 you do not need to change this option. Default value is 25600.
1312 @item inputbw=@var{bytes/seconds}
1313 Sender nominal input rate, in bytes per seconds. Used along with
1314 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1315 calculate maximum sending rate when recovery packets are sent
1316 along with the main media stream:
1317 @option{inputbw} * (100 + @option{oheadbw}) / 100
1318 if @option{inputbw} is not set while @option{maxbw} is set to
1319 relative (0), the actual input rate is evaluated inside
1320 the library. Default value is 0.
1322 @item iptos=@var{tos}
1323 IP Type of Service. Applies to sender only. Default value is 0xB8.
1325 @item ipttl=@var{ttl}
1326 IP Time To Live. Applies to sender only. Default value is 64.
1328 @item latency=@var{microseconds}
1329 Timestamp-based Packet Delivery Delay.
1330 Used to absorb bursts of missed packet retransmissions.
1331 This flag sets both @option{rcvlatency} and @option{peerlatency}
1332 to the same value. Note that prior to version 1.3.0
1333 this is the only flag to set the latency, however
1334 this is effectively equivalent to setting @option{peerlatency},
1335 when side is sender and @option{rcvlatency}
1336 when side is receiver, and the bidirectional stream
1337 sending is not supported.
1339 @item listen_timeout=@var{microseconds}
1340 Set socket listen timeout.
1342 @item maxbw=@var{bytes/seconds}
1343 Maximum sending bandwidth, in bytes per seconds.
1344 -1 infinite (CSRTCC limit is 30mbps)
1345 0 relative to input rate (see @option{inputbw})
1346 >0 absolute limit value
1347 Default value is 0 (relative)
1349 @item mode=@var{caller|listener|rendezvous}
1351 @option{caller} opens client connection.
1352 @option{listener} starts server to listen for incoming connections.
1353 @option{rendezvous} use Rendez-Vous connection mode.
1354 Default value is caller.
1356 @item mss=@var{bytes}
1357 Maximum Segment Size, in bytes. Used for buffer allocation
1358 and rate calculation using a packet counter assuming fully
1359 filled packets. The smallest MSS between the peers is
1360 used. This is 1500 by default in the overall internet.
1361 This is the maximum size of the UDP packet and can be
1362 only decreased, unless you have some unusual dedicated
1363 network settings. Default value is 1500.
1365 @item nakreport=@var{1|0}
1366 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1367 periodically until a lost packet is retransmitted or
1368 intentionally dropped. Default value is 1.
1370 @item oheadbw=@var{percents}
1371 Recovery bandwidth overhead above input rate, in percents.
1372 See @option{inputbw}. Default value is 25%.
1374 @item passphrase=@var{string}
1375 HaiCrypt Encryption/Decryption Passphrase string, length
1376 from 10 to 79 characters. The passphrase is the shared
1377 secret between the sender and the receiver. It is used
1378 to generate the Key Encrypting Key using PBKDF2
1379 (Password-Based Key Derivation Function). It is used
1380 only if @option{pbkeylen} is non-zero. It is used on
1381 the receiver only if the received data is encrypted.
1382 The configured passphrase cannot be recovered (write-only).
1384 @item enforced_encryption=@var{1|0}
1385 If true, both connection parties must have the same password
1386 set (including empty, that is, with no encryption). If the
1387 password doesn't match or only one side is unencrypted,
1388 the connection is rejected. Default is true.
1390 @item kmrefreshrate=@var{packets}
1391 The number of packets to be transmitted after which the
1392 encryption key is switched to a new key. Default is -1.
1393 -1 means auto (0x1000000 in srt library). The range for
1394 this option is integers in the 0 - @code{INT_MAX}.
1396 @item kmpreannounce=@var{packets}
1397 The interval between when a new encryption key is sent and
1398 when switchover occurs. This value also applies to the
1399 subsequent interval between when switchover occurs and
1400 when the old encryption key is decommissioned. Default is -1.
1401 -1 means auto (0x1000 in srt library). The range for
1402 this option is integers in the 0 - @code{INT_MAX}.
1404 @item payload_size=@var{bytes}
1405 Sets the maximum declared size of a packet transferred
1406 during the single call to the sending function in Live
1407 mode. Use 0 if this value isn't used (which is default in
1409 Default is -1 (automatic), which typically means MPEG-TS;
1410 if you are going to use SRT
1411 to send any different kind of payload, such as, for example,
1412 wrapping a live stream in very small frames, then you can
1413 use a bigger maximum frame size, though not greater than
1416 @item pkt_size=@var{bytes}
1417 Alias for @samp{payload_size}.
1419 @item peerlatency=@var{microseconds}
1420 The latency value (as described in @option{rcvlatency}) that is
1421 set by the sender side as a minimum value for the receiver.
1423 @item pbkeylen=@var{bytes}
1424 Sender encryption key length, in bytes.
1425 Only can be set to 0, 16, 24 and 32.
1426 Enable sender encryption if not 0.
1427 Not required on receiver (set to 0),
1428 key size obtained from sender in HaiCrypt handshake.
1431 @item rcvlatency=@var{microseconds}
1432 The time that should elapse since the moment when the
1433 packet was sent and the moment when it's delivered to
1434 the receiver application in the receiving function.
1435 This time should be a buffer time large enough to cover
1436 the time spent for sending, unexpectedly extended RTT
1437 time, and the time needed to retransmit the lost UDP
1438 packet. The effective latency value will be the maximum
1439 of this options' value and the value of @option{peerlatency}
1440 set by the peer side. Before version 1.3.0 this option
1441 is only available as @option{latency}.
1443 @item recv_buffer_size=@var{bytes}
1444 Set UDP receive buffer size, expressed in bytes.
1446 @item send_buffer_size=@var{bytes}
1447 Set UDP send buffer size, expressed in bytes.
1449 @item timeout=@var{microseconds}
1450 Set raise error timeouts for read, write and connect operations. Note that the
1451 SRT library has internal timeouts which can be controlled separately, the
1452 value set here is only a cap on those.
1454 @item tlpktdrop=@var{1|0}
1455 Too-late Packet Drop. When enabled on receiver, it skips
1456 missing packets that have not been delivered in time and
1457 delivers the following packets to the application when
1458 their time-to-play has come. It also sends a fake ACK to
1459 the sender. When enabled on sender and enabled on the
1460 receiving peer, the sender drops the older packets that
1461 have no chance of being delivered in time. It was
1462 automatically enabled in the sender if the receiver
1465 @item sndbuf=@var{bytes}
1466 Set send buffer size, expressed in bytes.
1468 @item rcvbuf=@var{bytes}
1469 Set receive buffer size, expressed in bytes.
1471 Receive buffer must not be greater than @option{ffs}.
1473 @item lossmaxttl=@var{packets}
1474 The value up to which the Reorder Tolerance may grow. When
1475 Reorder Tolerance is > 0, then packet loss report is delayed
1476 until that number of packets come in. Reorder Tolerance
1477 increases every time a "belated" packet has come, but it
1478 wasn't due to retransmission (that is, when UDP packets tend
1479 to come out of order), with the difference between the latest
1480 sequence and this packet's sequence, and not more than the
1481 value of this option. By default it's 0, which means that this
1482 mechanism is turned off, and the loss report is always sent
1483 immediately upon experiencing a "gap" in sequences.
1486 The minimum SRT version that is required from the peer. A connection
1487 to a peer that does not satisfy the minimum version requirement
1490 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1493 @item streamid=@var{string}
1494 A string limited to 512 characters that can be set on the socket prior
1495 to connecting. This stream ID will be able to be retrieved by the
1496 listener side from the socket that is returned from srt_accept and
1497 was connected by a socket with that set stream ID. SRT does not enforce
1498 any special interpretation of the contents of this string.
1499 This option doesn’t make sense in Rendezvous connection; the result
1500 might be that simply one side will override the value from the other
1501 side and it’s the matter of luck which one would win
1503 @item smoother=@var{live|file}
1504 The type of Smoother used for the transmission for that socket, which
1505 is responsible for the transmission and congestion control. The Smoother
1506 type must be exactly the same on both connecting parties, otherwise
1507 the connection is rejected.
1509 @item messageapi=@var{1|0}
1510 When set, this socket uses the Message API, otherwise it uses Buffer
1511 API. Note that in live mode (see @option{transtype}) there’s only
1512 message API available. In File mode you can chose to use one of two modes:
1514 Stream API (default, when this option is false). In this mode you may
1515 send as many data as you wish with one sending instruction, or even use
1516 dedicated functions that read directly from a file. The internal facility
1517 will take care of any speed and congestion control. When receiving, you
1518 can also receive as many data as desired, the data not extracted will be
1519 waiting for the next call. There is no boundary between data portions in
1522 Message API. In this mode your single sending instruction passes exactly
1523 one piece of data that has boundaries (a message). Contrary to Live mode,
1524 this message may span across multiple UDP packets and the only size
1525 limitation is that it shall fit as a whole in the sending buffer. The
1526 receiver shall use as large buffer as necessary to receive the message,
1527 otherwise the message will not be given up. When the message is not
1528 complete (not all packets received or there was a packet loss) it will
1531 @item transtype=@var{live|file}
1532 Sets the transmission type for the socket, in particular, setting this
1533 option sets multiple other parameters to their default values as required
1534 for a particular transmission type.
1536 live: Set options as for live transmission. In this mode, you should
1537 send by one sending instruction only so many data that fit in one UDP packet,
1538 and limited to the value defined first in @option{payload_size} (1316 is
1539 default in this mode). There is no speed control in this mode, only the
1540 bandwidth control, if configured, in order to not exceed the bandwidth with
1541 the overhead transmission (retransmitted and control packets).
1543 file: Set options as for non-live transmission. See @option{messageapi}
1544 for further explanations
1546 @item linger=@var{seconds}
1547 The number of seconds that the socket waits for unsent data when closing.
1548 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
1549 seconds in file mode). The range for this option is integers in the
1554 For more information see: @url{https://github.com/Haivision/srt}.
1558 Secure Real-time Transport Protocol.
1560 The accepted options are:
1563 @item srtp_out_suite
1564 Select input and output encoding suites.
1568 @item AES_CM_128_HMAC_SHA1_80
1569 @item SRTP_AES128_CM_HMAC_SHA1_80
1570 @item AES_CM_128_HMAC_SHA1_32
1571 @item SRTP_AES128_CM_HMAC_SHA1_32
1574 @item srtp_in_params
1575 @item srtp_out_params
1576 Set input and output encoding parameters, which are expressed by a
1577 base64-encoded representation of a binary block. The first 16 bytes of
1578 this binary block are used as master key, the following 14 bytes are
1579 used as master salt.
1584 Virtually extract a segment of a file or another stream.
1585 The underlying stream must be seekable.
1590 Start offset of the extracted segment, in bytes.
1592 End offset of the extracted segment, in bytes.
1593 If set to 0, extract till end of file.
1598 Extract a chapter from a DVD VOB file (start and end sectors obtained
1599 externally and multiplied by 2048):
1601 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1604 Play an AVI file directly from a TAR archive:
1606 subfile,,start,183241728,end,366490624,,:archive.tar
1609 Play a MPEG-TS file from start offset till end:
1611 subfile,,start,32815239,end,0,,:video.ts
1616 Writes the output to multiple protocols. The individual outputs are separated
1620 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1625 Transmission Control Protocol.
1627 The required syntax for a TCP url is:
1629 tcp://@var{hostname}:@var{port}[?@var{options}]
1632 @var{options} contains a list of &-separated options of the form
1633 @var{key}=@var{val}.
1635 The list of supported options follows.
1638 @item listen=@var{1|0}
1639 Listen for an incoming connection. Default value is 0.
1641 @item timeout=@var{microseconds}
1642 Set raise error timeout, expressed in microseconds.
1644 This option is only relevant in read mode: if no data arrived in more
1645 than this time interval, raise error.
1647 @item listen_timeout=@var{milliseconds}
1648 Set listen timeout, expressed in milliseconds.
1650 @item recv_buffer_size=@var{bytes}
1651 Set receive buffer size, expressed bytes.
1653 @item send_buffer_size=@var{bytes}
1654 Set send buffer size, expressed bytes.
1656 @item tcp_nodelay=@var{1|0}
1657 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1659 @item tcp_mss=@var{bytes}
1660 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1663 The following example shows how to setup a listening TCP connection
1664 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1666 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1667 ffplay tcp://@var{hostname}:@var{port}
1672 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1674 The required syntax for a TLS/SSL url is:
1676 tls://@var{hostname}:@var{port}[?@var{options}]
1679 The following parameters can be set via command line options
1680 (or in code via @code{AVOption}s):
1684 @item ca_file, cafile=@var{filename}
1685 A file containing certificate authority (CA) root certificates to treat
1686 as trusted. If the linked TLS library contains a default this might not
1687 need to be specified for verification to work, but not all libraries and
1688 setups have defaults built in.
1689 The file must be in OpenSSL PEM format.
1691 @item tls_verify=@var{1|0}
1692 If enabled, try to verify the peer that we are communicating with.
1693 Note, if using OpenSSL, this currently only makes sure that the
1694 peer certificate is signed by one of the root certificates in the CA
1695 database, but it does not validate that the certificate actually
1696 matches the host name we are trying to connect to. (With other backends,
1697 the host name is validated as well.)
1699 This is disabled by default since it requires a CA database to be
1700 provided by the caller in many cases.
1702 @item cert_file, cert=@var{filename}
1703 A file containing a certificate to use in the handshake with the peer.
1704 (When operating as server, in listen mode, this is more often required
1705 by the peer, while client certificates only are mandated in certain
1708 @item key_file, key=@var{filename}
1709 A file containing the private key for the certificate.
1711 @item listen=@var{1|0}
1712 If enabled, listen for connections on the provided port, and assume
1713 the server role in the handshake instead of the client role.
1717 Example command lines:
1719 To create a TLS/SSL server that serves an input stream.
1722 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1725 To play back a stream from the TLS/SSL server using @command{ffplay}:
1728 ffplay tls://@var{hostname}:@var{port}
1733 User Datagram Protocol.
1735 The required syntax for an UDP URL is:
1737 udp://@var{hostname}:@var{port}[?@var{options}]
1740 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1742 In case threading is enabled on the system, a circular buffer is used
1743 to store the incoming data, which allows one to reduce loss of data due to
1744 UDP socket buffer overruns. The @var{fifo_size} and
1745 @var{overrun_nonfatal} options are related to this buffer.
1747 The list of supported options follows.
1750 @item buffer_size=@var{size}
1751 Set the UDP maximum socket buffer size in bytes. This is used to set either
1752 the receive or send buffer size, depending on what the socket is used for.
1753 Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
1755 @item bitrate=@var{bitrate}
1756 If set to nonzero, the output will have the specified constant bitrate if the
1757 input has enough packets to sustain it.
1759 @item burst_bits=@var{bits}
1760 When using @var{bitrate} this specifies the maximum number of bits in
1763 @item localport=@var{port}
1764 Override the local UDP port to bind with.
1766 @item localaddr=@var{addr}
1767 Local IP address of a network interface used for sending packets or joining
1770 @item pkt_size=@var{size}
1771 Set the size in bytes of UDP packets.
1773 @item reuse=@var{1|0}
1774 Explicitly allow or disallow reusing UDP sockets.
1777 Set the time to live value (for multicast only).
1779 @item connect=@var{1|0}
1780 Initialize the UDP socket with @code{connect()}. In this case, the
1781 destination address can't be changed with ff_udp_set_remote_url later.
1782 If the destination address isn't known at the start, this option can
1783 be specified in ff_udp_set_remote_url, too.
1784 This allows finding out the source address for the packets with getsockname,
1785 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1786 unreachable" is received.
1787 For receiving, this gives the benefit of only receiving packets from
1788 the specified peer address/port.
1790 @item sources=@var{address}[,@var{address}]
1791 Only receive packets sent from the specified addresses. In case of multicast,
1792 also subscribe to multicast traffic coming from these addresses only.
1794 @item block=@var{address}[,@var{address}]
1795 Ignore packets sent from the specified addresses. In case of multicast, also
1796 exclude the source addresses in the multicast subscription.
1798 @item fifo_size=@var{units}
1799 Set the UDP receiving circular buffer size, expressed as a number of
1800 packets with size of 188 bytes. If not specified defaults to 7*4096.
1802 @item overrun_nonfatal=@var{1|0}
1803 Survive in case of UDP receiving circular buffer overrun. Default
1806 @item timeout=@var{microseconds}
1807 Set raise error timeout, expressed in microseconds.
1809 This option is only relevant in read mode: if no data arrived in more
1810 than this time interval, raise error.
1812 @item broadcast=@var{1|0}
1813 Explicitly allow or disallow UDP broadcasting.
1815 Note that broadcasting may not work properly on networks having
1816 a broadcast storm protection.
1819 @subsection Examples
1823 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1825 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1829 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1830 sized UDP packets, using a large input buffer:
1832 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1836 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1838 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1846 The required syntax for a Unix socket URL is:
1849 unix://@var{filepath}
1852 The following parameters can be set via command line options
1853 (or in code via @code{AVOption}s):
1859 Create the Unix socket in listening mode.
1864 ZeroMQ asynchronous messaging using the libzmq library.
1866 This library supports unicast streaming to multiple clients without relying on
1869 The required syntax for streaming or connecting to a stream is:
1871 zmq:tcp://ip-address:port
1875 Create a localhost stream on port 5555:
1877 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1880 Multiple clients may connect to the stream using:
1882 ffplay zmq:tcp://127.0.0.1:5555
1885 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
1886 The server side binds to a port and publishes data. Clients connect to the
1887 server (via IP address/port) and subscribe to the stream. The order in which
1888 the server and client start generally does not matter.
1890 ffmpeg must be compiled with the --enable-libzmq option to support
1893 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1894 line. The following options are supported:
1899 Forces the maximum packet size for sending/receiving data. The default value is
1900 131,072 bytes. On the server side, this sets the maximum size of sent packets
1901 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
1902 packets. Note that pkt_size on the clients should be equal to or greater than
1903 pkt_size on the server. Otherwise the received message may be truncated causing
1909 @c man end PROTOCOLS