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.
178 The accepted options are:
181 @item read_ahead_limit
182 Amount in bytes that may be read ahead when seeking isn't supported. Range is -1 to INT_MAX.
183 -1 for unlimited. Default is 65536.
194 Physical concatenation protocol.
196 Read and seek from many resources in sequence as if they were
199 A URL accepted by this protocol has the syntax:
201 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
204 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
205 resource to be concatenated, each one possibly specifying a distinct
208 For example to read a sequence of files @file{split1.mpeg},
209 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
212 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
215 Note that you may need to escape the character "|" which is special for
220 AES-encrypted stream reading protocol.
222 The accepted options are:
225 Set the AES decryption key binary block from given hexadecimal representation.
228 Set the AES decryption initialization vector binary block from given hexadecimal representation.
231 Accepted URL formats:
239 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
241 For example, to convert a GIF file given inline with @command{ffmpeg}:
243 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
248 File access protocol.
250 Read from or write to a file.
252 A file URL can have the form:
257 where @var{filename} is the path of the file to read.
259 An URL that does not have a protocol prefix will be assumed to be a
260 file URL. Depending on the build, an URL that looks like a Windows
261 path with the drive letter at the beginning will also be assumed to be
262 a file URL (usually not the case in builds for unix-like systems).
264 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
267 ffmpeg -i file:input.mpeg output.mpeg
270 This protocol accepts the following options:
274 Truncate existing files on write, if set to 1. A value of 0 prevents
275 truncating. Default value is 1.
278 Set I/O operation maximum block size, in bytes. Default value is
279 @code{INT_MAX}, which results in not limiting the requested block size.
280 Setting this value reasonably low improves user termination request reaction
281 time, which is valuable for files on slow medium.
284 If set to 1, the protocol will retry reading at the end of the file, allowing
285 reading files that still are being written. In order for this to terminate,
286 you either need to use the rw_timeout option, or use the interrupt callback
290 Controls if seekability is advertised on the file. 0 means non-seekable, -1
291 means auto (seekable for normal files, non-seekable for named pipes).
293 Many demuxers handle seekable and non-seekable resources differently,
294 overriding this might speed up opening certain files at the cost of losing some
295 features (e.g. accurate seeking).
300 FTP (File Transfer Protocol).
302 Read from or write to remote resources using FTP protocol.
304 Following syntax is required.
306 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
309 This protocol accepts the following options.
313 Set timeout in microseconds of socket I/O operations used by the underlying low level
314 operation. By default it is set to -1, which means that the timeout is
318 Set a user to be used for authenticating to the FTP server. This is overridden by the
322 Set a password to be used for authenticating to the FTP server. This is overridden by
323 the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
325 @item ftp-anonymous-password
326 Password used when login as anonymous user. Typically an e-mail address
329 @item ftp-write-seekable
330 Control seekability of connection during encoding. If set to 1 the
331 resource is supposed to be seekable, if set to 0 it is assumed not
332 to be seekable. Default value is 0.
335 NOTE: Protocol can be used as output, but it is recommended to not do
336 it, unless special care is taken (tests, customized server configuration
337 etc.). Different FTP servers behave in different way during seek
338 operation. ff* tools may produce incomplete content due to server limitations.
346 Read Apple HTTP Live Streaming compliant segmented stream as
347 a uniform one. The M3U8 playlists describing the segments can be
348 remote HTTP resources or local files, accessed using the standard
350 The nested protocol is declared by specifying
351 "+@var{proto}" after the hls URI scheme name, where @var{proto}
352 is either "file" or "http".
355 hls+http://host/path/to/remote/resource.m3u8
356 hls+file://path/to/local/resource.m3u8
359 Using this protocol is discouraged - the hls demuxer should work
360 just as well (if not, please report the issues) and is more complete.
361 To use the hls demuxer instead, simply use the direct URLs to the
366 HTTP (Hyper Text Transfer Protocol).
368 This protocol accepts the following options:
372 Control seekability of connection. If set to 1 the resource is
373 supposed to be seekable, if set to 0 it is assumed not to be seekable,
374 if set to -1 it will try to autodetect if it is seekable. Default
378 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
381 Set a specific content type for the POST messages or for listen mode.
384 set HTTP proxy to tunnel through e.g. http://example.com:1234
387 Set custom HTTP headers, can override built in default headers. The
388 value must be a string encoding the headers.
390 @item multiple_requests
391 Use persistent connections if set to 1, default is 0.
394 Set custom HTTP post data.
397 Set the Referer header. Include 'Referer: URL' header in HTTP request.
400 Override the User-Agent header. If not specified the protocol will use a
401 string describing the libavformat build. ("Lavf/<version>")
404 This is a deprecated option, you can use user_agent instead it.
406 @item reconnect_at_eof
407 If set then eof is treated like an error and causes reconnection, this is useful
408 for live / endless streams.
410 @item reconnect_streamed
411 If set then even streamed/non seekable streams will be reconnected on errors.
413 @item reconnect_on_network_error
414 Reconnect automatically in case of TCP/TLS errors during connect.
416 @item reconnect_on_http_error
417 A comma separated list of HTTP status codes to reconnect on. The list can
418 include specific status codes (e.g. '503') or the strings '4xx' / '5xx'.
420 @item reconnect_delay_max
421 Sets the maximum delay in seconds after which to give up reconnecting
424 Export the MIME type.
427 Exports the HTTP response version number. Usually "1.0" or "1.1".
430 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
431 supports this, the metadata has to be retrieved by the application by reading
432 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
435 @item icy_metadata_headers
436 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
437 headers, separated by newline characters.
439 @item icy_metadata_packet
440 If the server supports ICY metadata, and @option{icy} was set to 1, this
441 contains the last non-empty metadata packet sent by the server. It should be
442 polled in regular intervals by applications interested in mid-stream metadata
446 Set the cookies to be sent in future requests. The format of each cookie is the
447 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
448 delimited by a newline character.
451 Set initial byte offset.
454 Try to limit the request to bytes preceding this offset.
457 When used as a client option it sets the HTTP method for the request.
459 When used as a server option it sets the HTTP method that is going to be
460 expected from the client(s).
461 If the expected and the received HTTP method do not match the client will
462 be given a Bad Request response.
463 When unset the HTTP method is not checked for now. This will be replaced by
464 autodetection in the future.
467 If set to 1 enables experimental HTTP server. This can be used to send data when
468 used as an output option, or read data from a client with HTTP POST when used as
470 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
471 in ffmpeg.c and thus must not be used as a command line option.
473 # Server side (sending):
474 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
476 # Client side (receiving):
477 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
479 # Client can also be done with wget:
480 wget http://@var{server}:@var{port} -O somefile.ogg
482 # Server side (receiving):
483 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
485 # Client side (sending):
486 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
488 # Client can also be done with wget:
489 wget --post-file=somefile.ogg http://@var{server}:@var{port}
492 @item send_expect_100
493 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
494 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
499 Set HTTP authentication type. No option for Digest, since this method requires
500 getting nonce parameters from the server first and can't be used straight away like
505 Choose the HTTP authentication type automatically. This is the default.
508 Choose the HTTP basic authentication.
510 Basic authentication sends a Base64-encoded string that contains a user name and password
511 for the client. Base64 is not a form of encryption and should be considered the same as
512 sending the user name and password in clear text (Base64 is a reversible encoding).
513 If a resource needs to be protected, strongly consider using an authentication scheme
514 other than basic authentication. HTTPS/TLS should be used with basic authentication.
515 Without these additional security enhancements, basic authentication should not be used
516 to protect sensitive or valuable information.
521 @subsection HTTP Cookies
523 Some HTTP requests will be denied unless cookie values are passed in with the
524 request. The @option{cookies} option allows these cookies to be specified. At
525 the very least, each cookie must specify a value along with a path and domain.
526 HTTP requests that match both the domain and path will automatically include the
527 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
530 The required syntax to play a stream specifying a cookie is:
532 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
537 Icecast protocol (stream to Icecast servers)
539 This protocol accepts the following options:
543 Set the stream genre.
548 @item ice_description
549 Set the stream description.
552 Set the stream website URL.
555 Set if the stream should be public.
556 The default is 0 (not public).
559 Override the User-Agent header. If not specified a string of the form
560 "Lavf/<version>" will be used.
563 Set the Icecast mountpoint password.
566 Set the stream content type. This must be set if it is different from
570 This enables support for Icecast versions < 2.4.0, that do not support the
571 HTTP PUT method but the SOURCE method.
574 Establish a TLS (HTTPS) connection to Icecast.
579 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
584 MMS (Microsoft Media Server) protocol over TCP.
588 MMS (Microsoft Media Server) protocol over HTTP.
590 The required syntax is:
592 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
599 Computes the MD5 hash of the data to be written, and on close writes
600 this to the designated output or stdout if none is specified. It can
601 be used to test muxers without writing an actual file.
603 Some examples follow.
605 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
606 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
608 # Write the MD5 hash of the encoded AVI file to stdout.
609 ffmpeg -i input.flv -f avi -y md5:
612 Note that some formats (typically MOV) require the output protocol to
613 be seekable, so they will fail with the MD5 output protocol.
617 UNIX pipe access protocol.
619 Read and write from UNIX pipes.
621 The accepted syntax is:
626 @var{number} is the number corresponding to the file descriptor of the
627 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
628 is not specified, by default the stdout file descriptor will be used
629 for writing, stdin for reading.
631 For example to read from stdin with @command{ffmpeg}:
633 cat test.wav | ffmpeg -i pipe:0
634 # ...this is the same as...
635 cat test.wav | ffmpeg -i pipe:
638 For writing to stdout with @command{ffmpeg}:
640 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
641 # ...this is the same as...
642 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
645 This protocol accepts the following options:
649 Set I/O operation maximum block size, in bytes. Default value is
650 @code{INT_MAX}, which results in not limiting the requested block size.
651 Setting this value reasonably low improves user termination request reaction
652 time, which is valuable if data transmission is slow.
655 Note that some formats (typically MOV), require the output protocol to
656 be seekable, so they will fail with the pipe output protocol.
660 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
662 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
663 for MPEG-2 Transport Streams sent over RTP.
665 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
666 the @code{rtp} protocol.
668 The required syntax is:
670 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
673 The destination UDP ports are @code{port + 2} for the column FEC stream
674 and @code{port + 4} for the row FEC stream.
676 This protocol accepts the following options:
680 The number of columns (4-20, LxD <= 100)
683 The number of rows (4-20, LxD <= 100)
690 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
695 Real-Time Messaging Protocol.
697 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
698 content across a TCP/IP network.
700 The required syntax is:
702 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
705 The accepted parameters are:
709 An optional username (mostly for publishing).
712 An optional password (mostly for publishing).
715 The address of the RTMP server.
718 The number of the TCP port to use (by default is 1935).
721 It is the name of the application to access. It usually corresponds to
722 the path where the application is installed on the RTMP server
723 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
724 the value parsed from the URI through the @code{rtmp_app} option, too.
727 It is the path or name of the resource to play with reference to the
728 application specified in @var{app}, may be prefixed by "mp4:". You
729 can override the value parsed from the URI through the @code{rtmp_playpath}
733 Act as a server, listening for an incoming connection.
736 Maximum time to wait for the incoming connection. Implies listen.
739 Additionally, the following parameters can be set via command line options
740 (or in code via @code{AVOption}s):
744 Name of application to connect on the RTMP server. This option
745 overrides the parameter specified in the URI.
748 Set the client buffer time in milliseconds. The default is 3000.
751 Extra arbitrary AMF connection parameters, parsed from a string,
752 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
753 Each value is prefixed by a single character denoting the type,
754 B for Boolean, N for number, S for string, O for object, or Z for null,
755 followed by a colon. For Booleans the data must be either 0 or 1 for
756 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
757 1 to end or begin an object, respectively. Data items in subobjects may
758 be named, by prefixing the type with 'N' and specifying the name before
759 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
760 times to construct arbitrary AMF sequences.
763 Version of the Flash plugin used to run the SWF player. The default
764 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
765 <libavformat version>).)
767 @item rtmp_flush_interval
768 Number of packets flushed in the same request (RTMPT only). The default
772 Specify that the media is a live stream. No resuming or seeking in
773 live streams is possible. The default value is @code{any}, which means the
774 subscriber first tries to play the live stream specified in the
775 playpath. If a live stream of that name is not found, it plays the
776 recorded stream. The other possible values are @code{live} and
780 URL of the web page in which the media was embedded. By default no
784 Stream identifier to play or to publish. This option overrides the
785 parameter specified in the URI.
788 Name of live stream to subscribe to. By default no value will be sent.
789 It is only sent if the option is specified or if rtmp_live
793 SHA256 hash of the decompressed SWF file (32 bytes).
796 Size of the decompressed SWF file, required for SWFVerification.
799 URL of the SWF player for the media. By default no value will be sent.
802 URL to player swf file, compute hash/size automatically.
805 URL of the target stream. Defaults to proto://host[:port]/app.
809 For example to read with @command{ffplay} a multimedia resource named
810 "sample" from the application "vod" from an RTMP server "myserver":
812 ffplay rtmp://myserver/vod/sample
815 To publish to a password protected server, passing the playpath and
816 app names separately:
818 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
823 Encrypted Real-Time Messaging Protocol.
825 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
826 streaming multimedia content within standard cryptographic primitives,
827 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
832 Real-Time Messaging Protocol over a secure SSL connection.
834 The Real-Time Messaging Protocol (RTMPS) is used for streaming
835 multimedia content across an encrypted connection.
839 Real-Time Messaging Protocol tunneled through HTTP.
841 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
842 for streaming multimedia content within HTTP requests to traverse
847 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
849 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
850 is used for streaming multimedia content within HTTP requests to traverse
855 Real-Time Messaging Protocol tunneled through HTTPS.
857 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
858 for streaming multimedia content within HTTPS requests to traverse
861 @section libsmbclient
863 libsmbclient permits one to manipulate CIFS/SMB network resources.
865 Following syntax is required.
868 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
871 This protocol accepts the following options.
875 Set timeout in milliseconds of socket I/O operations used by the underlying
876 low level operation. By default it is set to -1, which means that the timeout
880 Truncate existing files on write, if set to 1. A value of 0 prevents
881 truncating. Default value is 1.
884 Set the workgroup used for making connections. By default workgroup is not specified.
888 For more information see: @url{http://www.samba.org/}.
892 Secure File Transfer Protocol via libssh
894 Read from or write to remote resources using SFTP protocol.
896 Following syntax is required.
899 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
902 This protocol accepts the following options.
906 Set timeout of socket I/O operations used by the underlying low level
907 operation. By default it is set to -1, which means that the timeout
911 Truncate existing files on write, if set to 1. A value of 0 prevents
912 truncating. Default value is 1.
915 Specify the path of the file containing private key to use during authorization.
916 By default libssh searches for keys in the @file{~/.ssh/} directory.
920 Example: Play a file stored on remote server.
923 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
926 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
928 Real-Time Messaging Protocol and its variants supported through
931 Requires the presence of the librtmp headers and library during
932 configuration. You need to explicitly configure the build with
933 "--enable-librtmp". If enabled this will replace the native RTMP
936 This protocol provides most client functions and a few server
937 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
938 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
939 variants of these encrypted types (RTMPTE, RTMPTS).
941 The required syntax is:
943 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
946 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
947 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
948 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
949 meaning as specified for the RTMP native protocol.
950 @var{options} contains a list of space-separated options of the form
953 See the librtmp manual page (man 3 librtmp) for more information.
955 For example, to stream a file in real-time to an RTMP server using
958 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
961 To play the same stream using @command{ffplay}:
963 ffplay "rtmp://myserver/live/mystream live=1"
968 Real-time Transport Protocol.
970 The required syntax for an RTP URL is:
971 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
973 @var{port} specifies the RTP port to use.
975 The following URL options are supported:
980 Set the TTL (Time-To-Live) value (for multicast only).
982 @item rtcpport=@var{n}
983 Set the remote RTCP port to @var{n}.
985 @item localrtpport=@var{n}
986 Set the local RTP port to @var{n}.
988 @item localrtcpport=@var{n}'
989 Set the local RTCP port to @var{n}.
991 @item pkt_size=@var{n}
992 Set max packet size (in bytes) to @var{n}.
995 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
998 @item sources=@var{ip}[,@var{ip}]
999 List allowed source IP addresses.
1001 @item block=@var{ip}[,@var{ip}]
1002 List disallowed (blocked) source IP addresses.
1004 @item write_to_source=0|1
1005 Send packets to the source address of the latest received packet (if
1006 set to 1) or to a default remote address (if set to 0).
1008 @item localport=@var{n}
1009 Set the local RTP port to @var{n}.
1011 @item timeout=@var{n}
1012 Set timeout (in microseconds) of socket I/O operations to @var{n}.
1014 This is a deprecated option. Instead, @option{localrtpport} should be
1024 If @option{rtcpport} is not set the RTCP port will be set to the RTP
1028 If @option{localrtpport} (the local RTP port) is not set any available
1029 port will be used for the local RTP and RTCP ports.
1032 If @option{localrtcpport} (the local RTCP port) is not set it will be
1033 set to the local RTP port value plus 1.
1038 Real-Time Streaming Protocol.
1040 RTSP is not technically a protocol handler in libavformat, it is a demuxer
1041 and muxer. The demuxer supports both normal RTSP (with data transferred
1042 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
1043 data transferred over RDT).
1045 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1046 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
1047 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
1049 The required syntax for a RTSP url is:
1051 rtsp://@var{hostname}[:@var{port}]/@var{path}
1054 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1055 line, or set in code via @code{AVOption}s or in
1056 @code{avformat_open_input}.
1058 The following options are supported.
1062 Do not start playing the stream immediately if set to 1. Default value
1065 @item rtsp_transport
1066 Set RTSP transport protocols.
1068 It accepts the following values:
1071 Use UDP as lower transport protocol.
1074 Use TCP (interleaving within the RTSP control channel) as lower
1078 Use UDP multicast as lower transport protocol.
1081 Use HTTP tunneling as lower transport protocol, which is useful for
1085 Multiple lower transport protocols may be specified, in that case they are
1086 tried one at a time (if the setup of one fails, the next one is tried).
1087 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
1092 The following values are accepted:
1095 Accept packets only from negotiated peer address and port.
1097 Act as a server, listening for an incoming connection.
1099 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
1102 Default value is @samp{none}.
1104 @item allowed_media_types
1105 Set media types to accept from the server.
1107 The following flags are accepted:
1114 By default it accepts all media types.
1117 Set minimum local UDP port. Default value is 5000.
1120 Set maximum local UDP port. Default value is 65000.
1123 Set maximum timeout (in seconds) to wait for incoming connections.
1125 A value of -1 means infinite (default). This option implies the
1126 @option{rtsp_flags} set to @samp{listen}.
1128 @item reorder_queue_size
1129 Set number of packets to buffer for handling of reordered packets.
1132 Set socket TCP I/O timeout in microseconds.
1135 Override User-Agent header. If not specified, it defaults to the
1136 libavformat identifier string.
1139 When receiving data over UDP, the demuxer tries to reorder received packets
1140 (since they may arrive out of order, or packets may get lost totally). This
1141 can be disabled by setting the maximum demuxing delay to zero (via
1142 the @code{max_delay} field of AVFormatContext).
1144 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1145 streams to display can be chosen with @code{-vst} @var{n} and
1146 @code{-ast} @var{n} for video and audio respectively, and can be switched
1147 on the fly by pressing @code{v} and @code{a}.
1149 @subsection Examples
1151 The following examples all make use of the @command{ffplay} and
1152 @command{ffmpeg} tools.
1156 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1158 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1162 Watch a stream tunneled over HTTP:
1164 ffplay -rtsp_transport http rtsp://server/video.mp4
1168 Send a stream in realtime to a RTSP server, for others to watch:
1170 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1174 Receive a stream in realtime:
1176 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1182 Session Announcement Protocol (RFC 2974). This is not technically a
1183 protocol handler in libavformat, it is a muxer and demuxer.
1184 It is used for signalling of RTP streams, by announcing the SDP for the
1185 streams regularly on a separate port.
1189 The syntax for a SAP url given to the muxer is:
1191 sap://@var{destination}[:@var{port}][?@var{options}]
1194 The RTP packets are sent to @var{destination} on port @var{port},
1195 or to port 5004 if no port is specified.
1196 @var{options} is a @code{&}-separated list. The following options
1201 @item announce_addr=@var{address}
1202 Specify the destination IP address for sending the announcements to.
1203 If omitted, the announcements are sent to the commonly used SAP
1204 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1205 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1207 @item announce_port=@var{port}
1208 Specify the port to send the announcements on, defaults to
1209 9875 if not specified.
1212 Specify the time to live value for the announcements and RTP packets,
1215 @item same_port=@var{0|1}
1216 If set to 1, send all RTP streams on the same port pair. If zero (the
1217 default), all streams are sent on unique ports, with each stream on a
1218 port 2 numbers higher than the previous.
1219 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1220 The RTP stack in libavformat for receiving requires all streams to be sent
1224 Example command lines follow.
1226 To broadcast a stream on the local subnet, for watching in VLC:
1229 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1232 Similarly, for watching in @command{ffplay}:
1235 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1238 And for watching in @command{ffplay}, over IPv6:
1241 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1246 The syntax for a SAP url given to the demuxer is:
1248 sap://[@var{address}][:@var{port}]
1251 @var{address} is the multicast address to listen for announcements on,
1252 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1253 is the port that is listened on, 9875 if omitted.
1255 The demuxers listens for announcements on the given address and port.
1256 Once an announcement is received, it tries to receive that particular stream.
1258 Example command lines follow.
1260 To play back the first stream announced on the normal SAP multicast address:
1266 To play back the first stream announced on one the default IPv6 SAP multicast address:
1269 ffplay sap://[ff0e::2:7ffe]
1274 Stream Control Transmission Protocol.
1276 The accepted URL syntax is:
1278 sctp://@var{host}:@var{port}[?@var{options}]
1281 The protocol accepts the following options:
1284 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1287 Set the maximum number of streams. By default no limit is set.
1292 Haivision Secure Reliable Transport Protocol via libsrt.
1294 The supported syntax for a SRT URL is:
1296 srt://@var{hostname}:@var{port}[?@var{options}]
1299 @var{options} contains a list of &-separated options of the form
1300 @var{key}=@var{val}.
1305 @var{options} srt://@var{hostname}:@var{port}
1308 @var{options} contains a list of '-@var{key} @var{val}'
1311 This protocol accepts the following options.
1314 @item connect_timeout=@var{milliseconds}
1315 Connection timeout; SRT cannot connect for RTT > 1500 msec
1316 (2 handshake exchanges) with the default connect timeout of
1317 3 seconds. This option applies to the caller and rendezvous
1318 connection modes. The connect timeout is 10 times the value
1319 set for the rendezvous mode (which can be used as a
1320 workaround for this connection problem with earlier versions).
1322 @item ffs=@var{bytes}
1323 Flight Flag Size (Window Size), in bytes. FFS is actually an
1324 internal parameter and you should set it to not less than
1325 @option{recv_buffer_size} and @option{mss}. The default value
1326 is relatively large, therefore unless you set a very large receiver buffer,
1327 you do not need to change this option. Default value is 25600.
1329 @item inputbw=@var{bytes/seconds}
1330 Sender nominal input rate, in bytes per seconds. Used along with
1331 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1332 calculate maximum sending rate when recovery packets are sent
1333 along with the main media stream:
1334 @option{inputbw} * (100 + @option{oheadbw}) / 100
1335 if @option{inputbw} is not set while @option{maxbw} is set to
1336 relative (0), the actual input rate is evaluated inside
1337 the library. Default value is 0.
1339 @item iptos=@var{tos}
1340 IP Type of Service. Applies to sender only. Default value is 0xB8.
1342 @item ipttl=@var{ttl}
1343 IP Time To Live. Applies to sender only. Default value is 64.
1345 @item latency=@var{microseconds}
1346 Timestamp-based Packet Delivery Delay.
1347 Used to absorb bursts of missed packet retransmissions.
1348 This flag sets both @option{rcvlatency} and @option{peerlatency}
1349 to the same value. Note that prior to version 1.3.0
1350 this is the only flag to set the latency, however
1351 this is effectively equivalent to setting @option{peerlatency},
1352 when side is sender and @option{rcvlatency}
1353 when side is receiver, and the bidirectional stream
1354 sending is not supported.
1356 @item listen_timeout=@var{microseconds}
1357 Set socket listen timeout.
1359 @item maxbw=@var{bytes/seconds}
1360 Maximum sending bandwidth, in bytes per seconds.
1361 -1 infinite (CSRTCC limit is 30mbps)
1362 0 relative to input rate (see @option{inputbw})
1363 >0 absolute limit value
1364 Default value is 0 (relative)
1366 @item mode=@var{caller|listener|rendezvous}
1368 @option{caller} opens client connection.
1369 @option{listener} starts server to listen for incoming connections.
1370 @option{rendezvous} use Rendez-Vous connection mode.
1371 Default value is caller.
1373 @item mss=@var{bytes}
1374 Maximum Segment Size, in bytes. Used for buffer allocation
1375 and rate calculation using a packet counter assuming fully
1376 filled packets. The smallest MSS between the peers is
1377 used. This is 1500 by default in the overall internet.
1378 This is the maximum size of the UDP packet and can be
1379 only decreased, unless you have some unusual dedicated
1380 network settings. Default value is 1500.
1382 @item nakreport=@var{1|0}
1383 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1384 periodically until a lost packet is retransmitted or
1385 intentionally dropped. Default value is 1.
1387 @item oheadbw=@var{percents}
1388 Recovery bandwidth overhead above input rate, in percents.
1389 See @option{inputbw}. Default value is 25%.
1391 @item passphrase=@var{string}
1392 HaiCrypt Encryption/Decryption Passphrase string, length
1393 from 10 to 79 characters. The passphrase is the shared
1394 secret between the sender and the receiver. It is used
1395 to generate the Key Encrypting Key using PBKDF2
1396 (Password-Based Key Derivation Function). It is used
1397 only if @option{pbkeylen} is non-zero. It is used on
1398 the receiver only if the received data is encrypted.
1399 The configured passphrase cannot be recovered (write-only).
1401 @item enforced_encryption=@var{1|0}
1402 If true, both connection parties must have the same password
1403 set (including empty, that is, with no encryption). If the
1404 password doesn't match or only one side is unencrypted,
1405 the connection is rejected. Default is true.
1407 @item kmrefreshrate=@var{packets}
1408 The number of packets to be transmitted after which the
1409 encryption key is switched to a new key. Default is -1.
1410 -1 means auto (0x1000000 in srt library). The range for
1411 this option is integers in the 0 - @code{INT_MAX}.
1413 @item kmpreannounce=@var{packets}
1414 The interval between when a new encryption key is sent and
1415 when switchover occurs. This value also applies to the
1416 subsequent interval between when switchover occurs and
1417 when the old encryption key is decommissioned. Default is -1.
1418 -1 means auto (0x1000 in srt library). The range for
1419 this option is integers in the 0 - @code{INT_MAX}.
1421 @item payload_size=@var{bytes}
1422 Sets the maximum declared size of a packet transferred
1423 during the single call to the sending function in Live
1424 mode. Use 0 if this value isn't used (which is default in
1426 Default is -1 (automatic), which typically means MPEG-TS;
1427 if you are going to use SRT
1428 to send any different kind of payload, such as, for example,
1429 wrapping a live stream in very small frames, then you can
1430 use a bigger maximum frame size, though not greater than
1433 @item pkt_size=@var{bytes}
1434 Alias for @samp{payload_size}.
1436 @item peerlatency=@var{microseconds}
1437 The latency value (as described in @option{rcvlatency}) that is
1438 set by the sender side as a minimum value for the receiver.
1440 @item pbkeylen=@var{bytes}
1441 Sender encryption key length, in bytes.
1442 Only can be set to 0, 16, 24 and 32.
1443 Enable sender encryption if not 0.
1444 Not required on receiver (set to 0),
1445 key size obtained from sender in HaiCrypt handshake.
1448 @item rcvlatency=@var{microseconds}
1449 The time that should elapse since the moment when the
1450 packet was sent and the moment when it's delivered to
1451 the receiver application in the receiving function.
1452 This time should be a buffer time large enough to cover
1453 the time spent for sending, unexpectedly extended RTT
1454 time, and the time needed to retransmit the lost UDP
1455 packet. The effective latency value will be the maximum
1456 of this options' value and the value of @option{peerlatency}
1457 set by the peer side. Before version 1.3.0 this option
1458 is only available as @option{latency}.
1460 @item recv_buffer_size=@var{bytes}
1461 Set UDP receive buffer size, expressed in bytes.
1463 @item send_buffer_size=@var{bytes}
1464 Set UDP send buffer size, expressed in bytes.
1466 @item timeout=@var{microseconds}
1467 Set raise error timeouts for read, write and connect operations. Note that the
1468 SRT library has internal timeouts which can be controlled separately, the
1469 value set here is only a cap on those.
1471 @item tlpktdrop=@var{1|0}
1472 Too-late Packet Drop. When enabled on receiver, it skips
1473 missing packets that have not been delivered in time and
1474 delivers the following packets to the application when
1475 their time-to-play has come. It also sends a fake ACK to
1476 the sender. When enabled on sender and enabled on the
1477 receiving peer, the sender drops the older packets that
1478 have no chance of being delivered in time. It was
1479 automatically enabled in the sender if the receiver
1482 @item sndbuf=@var{bytes}
1483 Set send buffer size, expressed in bytes.
1485 @item rcvbuf=@var{bytes}
1486 Set receive buffer size, expressed in bytes.
1488 Receive buffer must not be greater than @option{ffs}.
1490 @item lossmaxttl=@var{packets}
1491 The value up to which the Reorder Tolerance may grow. When
1492 Reorder Tolerance is > 0, then packet loss report is delayed
1493 until that number of packets come in. Reorder Tolerance
1494 increases every time a "belated" packet has come, but it
1495 wasn't due to retransmission (that is, when UDP packets tend
1496 to come out of order), with the difference between the latest
1497 sequence and this packet's sequence, and not more than the
1498 value of this option. By default it's 0, which means that this
1499 mechanism is turned off, and the loss report is always sent
1500 immediately upon experiencing a "gap" in sequences.
1503 The minimum SRT version that is required from the peer. A connection
1504 to a peer that does not satisfy the minimum version requirement
1507 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1510 @item streamid=@var{string}
1511 A string limited to 512 characters that can be set on the socket prior
1512 to connecting. This stream ID will be able to be retrieved by the
1513 listener side from the socket that is returned from srt_accept and
1514 was connected by a socket with that set stream ID. SRT does not enforce
1515 any special interpretation of the contents of this string.
1516 This option doesn’t make sense in Rendezvous connection; the result
1517 might be that simply one side will override the value from the other
1518 side and it’s the matter of luck which one would win
1520 @item smoother=@var{live|file}
1521 The type of Smoother used for the transmission for that socket, which
1522 is responsible for the transmission and congestion control. The Smoother
1523 type must be exactly the same on both connecting parties, otherwise
1524 the connection is rejected.
1526 @item messageapi=@var{1|0}
1527 When set, this socket uses the Message API, otherwise it uses Buffer
1528 API. Note that in live mode (see @option{transtype}) there’s only
1529 message API available. In File mode you can chose to use one of two modes:
1531 Stream API (default, when this option is false). In this mode you may
1532 send as many data as you wish with one sending instruction, or even use
1533 dedicated functions that read directly from a file. The internal facility
1534 will take care of any speed and congestion control. When receiving, you
1535 can also receive as many data as desired, the data not extracted will be
1536 waiting for the next call. There is no boundary between data portions in
1539 Message API. In this mode your single sending instruction passes exactly
1540 one piece of data that has boundaries (a message). Contrary to Live mode,
1541 this message may span across multiple UDP packets and the only size
1542 limitation is that it shall fit as a whole in the sending buffer. The
1543 receiver shall use as large buffer as necessary to receive the message,
1544 otherwise the message will not be given up. When the message is not
1545 complete (not all packets received or there was a packet loss) it will
1548 @item transtype=@var{live|file}
1549 Sets the transmission type for the socket, in particular, setting this
1550 option sets multiple other parameters to their default values as required
1551 for a particular transmission type.
1553 live: Set options as for live transmission. In this mode, you should
1554 send by one sending instruction only so many data that fit in one UDP packet,
1555 and limited to the value defined first in @option{payload_size} (1316 is
1556 default in this mode). There is no speed control in this mode, only the
1557 bandwidth control, if configured, in order to not exceed the bandwidth with
1558 the overhead transmission (retransmitted and control packets).
1560 file: Set options as for non-live transmission. See @option{messageapi}
1561 for further explanations
1563 @item linger=@var{seconds}
1564 The number of seconds that the socket waits for unsent data when closing.
1565 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
1566 seconds in file mode). The range for this option is integers in the
1571 For more information see: @url{https://github.com/Haivision/srt}.
1575 Secure Real-time Transport Protocol.
1577 The accepted options are:
1580 @item srtp_out_suite
1581 Select input and output encoding suites.
1585 @item AES_CM_128_HMAC_SHA1_80
1586 @item SRTP_AES128_CM_HMAC_SHA1_80
1587 @item AES_CM_128_HMAC_SHA1_32
1588 @item SRTP_AES128_CM_HMAC_SHA1_32
1591 @item srtp_in_params
1592 @item srtp_out_params
1593 Set input and output encoding parameters, which are expressed by a
1594 base64-encoded representation of a binary block. The first 16 bytes of
1595 this binary block are used as master key, the following 14 bytes are
1596 used as master salt.
1601 Virtually extract a segment of a file or another stream.
1602 The underlying stream must be seekable.
1607 Start offset of the extracted segment, in bytes.
1609 End offset of the extracted segment, in bytes.
1610 If set to 0, extract till end of file.
1615 Extract a chapter from a DVD VOB file (start and end sectors obtained
1616 externally and multiplied by 2048):
1618 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1621 Play an AVI file directly from a TAR archive:
1623 subfile,,start,183241728,end,366490624,,:archive.tar
1626 Play a MPEG-TS file from start offset till end:
1628 subfile,,start,32815239,end,0,,:video.ts
1633 Writes the output to multiple protocols. The individual outputs are separated
1637 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1642 Transmission Control Protocol.
1644 The required syntax for a TCP url is:
1646 tcp://@var{hostname}:@var{port}[?@var{options}]
1649 @var{options} contains a list of &-separated options of the form
1650 @var{key}=@var{val}.
1652 The list of supported options follows.
1655 @item listen=@var{2|1|0}
1656 Listen for an incoming connection. 0 disables listen, 1 enables listen in
1657 single client mode, 2 enables listen in multi-client mode. Default value is 0.
1659 @item timeout=@var{microseconds}
1660 Set raise error timeout, expressed in microseconds.
1662 This option is only relevant in read mode: if no data arrived in more
1663 than this time interval, raise error.
1665 @item listen_timeout=@var{milliseconds}
1666 Set listen timeout, expressed in milliseconds.
1668 @item recv_buffer_size=@var{bytes}
1669 Set receive buffer size, expressed bytes.
1671 @item send_buffer_size=@var{bytes}
1672 Set send buffer size, expressed bytes.
1674 @item tcp_nodelay=@var{1|0}
1675 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1677 @item tcp_mss=@var{bytes}
1678 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1681 The following example shows how to setup a listening TCP connection
1682 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1684 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1685 ffplay tcp://@var{hostname}:@var{port}
1690 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1692 The required syntax for a TLS/SSL url is:
1694 tls://@var{hostname}:@var{port}[?@var{options}]
1697 The following parameters can be set via command line options
1698 (or in code via @code{AVOption}s):
1702 @item ca_file, cafile=@var{filename}
1703 A file containing certificate authority (CA) root certificates to treat
1704 as trusted. If the linked TLS library contains a default this might not
1705 need to be specified for verification to work, but not all libraries and
1706 setups have defaults built in.
1707 The file must be in OpenSSL PEM format.
1709 @item tls_verify=@var{1|0}
1710 If enabled, try to verify the peer that we are communicating with.
1711 Note, if using OpenSSL, this currently only makes sure that the
1712 peer certificate is signed by one of the root certificates in the CA
1713 database, but it does not validate that the certificate actually
1714 matches the host name we are trying to connect to. (With other backends,
1715 the host name is validated as well.)
1717 This is disabled by default since it requires a CA database to be
1718 provided by the caller in many cases.
1720 @item cert_file, cert=@var{filename}
1721 A file containing a certificate to use in the handshake with the peer.
1722 (When operating as server, in listen mode, this is more often required
1723 by the peer, while client certificates only are mandated in certain
1726 @item key_file, key=@var{filename}
1727 A file containing the private key for the certificate.
1729 @item listen=@var{1|0}
1730 If enabled, listen for connections on the provided port, and assume
1731 the server role in the handshake instead of the client role.
1735 Example command lines:
1737 To create a TLS/SSL server that serves an input stream.
1740 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1743 To play back a stream from the TLS/SSL server using @command{ffplay}:
1746 ffplay tls://@var{hostname}:@var{port}
1751 User Datagram Protocol.
1753 The required syntax for an UDP URL is:
1755 udp://@var{hostname}:@var{port}[?@var{options}]
1758 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1760 In case threading is enabled on the system, a circular buffer is used
1761 to store the incoming data, which allows one to reduce loss of data due to
1762 UDP socket buffer overruns. The @var{fifo_size} and
1763 @var{overrun_nonfatal} options are related to this buffer.
1765 The list of supported options follows.
1768 @item buffer_size=@var{size}
1769 Set the UDP maximum socket buffer size in bytes. This is used to set either
1770 the receive or send buffer size, depending on what the socket is used for.
1771 Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
1773 @item bitrate=@var{bitrate}
1774 If set to nonzero, the output will have the specified constant bitrate if the
1775 input has enough packets to sustain it.
1777 @item burst_bits=@var{bits}
1778 When using @var{bitrate} this specifies the maximum number of bits in
1781 @item localport=@var{port}
1782 Override the local UDP port to bind with.
1784 @item localaddr=@var{addr}
1785 Local IP address of a network interface used for sending packets or joining
1788 @item pkt_size=@var{size}
1789 Set the size in bytes of UDP packets.
1791 @item reuse=@var{1|0}
1792 Explicitly allow or disallow reusing UDP sockets.
1795 Set the time to live value (for multicast only).
1797 @item connect=@var{1|0}
1798 Initialize the UDP socket with @code{connect()}. In this case, the
1799 destination address can't be changed with ff_udp_set_remote_url later.
1800 If the destination address isn't known at the start, this option can
1801 be specified in ff_udp_set_remote_url, too.
1802 This allows finding out the source address for the packets with getsockname,
1803 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1804 unreachable" is received.
1805 For receiving, this gives the benefit of only receiving packets from
1806 the specified peer address/port.
1808 @item sources=@var{address}[,@var{address}]
1809 Only receive packets sent from the specified addresses. In case of multicast,
1810 also subscribe to multicast traffic coming from these addresses only.
1812 @item block=@var{address}[,@var{address}]
1813 Ignore packets sent from the specified addresses. In case of multicast, also
1814 exclude the source addresses in the multicast subscription.
1816 @item fifo_size=@var{units}
1817 Set the UDP receiving circular buffer size, expressed as a number of
1818 packets with size of 188 bytes. If not specified defaults to 7*4096.
1820 @item overrun_nonfatal=@var{1|0}
1821 Survive in case of UDP receiving circular buffer overrun. Default
1824 @item timeout=@var{microseconds}
1825 Set raise error timeout, expressed in microseconds.
1827 This option is only relevant in read mode: if no data arrived in more
1828 than this time interval, raise error.
1830 @item broadcast=@var{1|0}
1831 Explicitly allow or disallow UDP broadcasting.
1833 Note that broadcasting may not work properly on networks having
1834 a broadcast storm protection.
1837 @subsection Examples
1841 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1843 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1847 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1848 sized UDP packets, using a large input buffer:
1850 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1854 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1856 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1864 The required syntax for a Unix socket URL is:
1867 unix://@var{filepath}
1870 The following parameters can be set via command line options
1871 (or in code via @code{AVOption}s):
1877 Create the Unix socket in listening mode.
1882 ZeroMQ asynchronous messaging using the libzmq library.
1884 This library supports unicast streaming to multiple clients without relying on
1887 The required syntax for streaming or connecting to a stream is:
1889 zmq:tcp://ip-address:port
1893 Create a localhost stream on port 5555:
1895 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1898 Multiple clients may connect to the stream using:
1900 ffplay zmq:tcp://127.0.0.1:5555
1903 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
1904 The server side binds to a port and publishes data. Clients connect to the
1905 server (via IP address/port) and subscribe to the stream. The order in which
1906 the server and client start generally does not matter.
1908 ffmpeg must be compiled with the --enable-libzmq option to support
1911 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1912 line. The following options are supported:
1917 Forces the maximum packet size for sending/receiving data. The default value is
1918 131,072 bytes. On the server side, this sets the maximum size of sent packets
1919 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
1920 packets. Note that pkt_size on the clients should be equal to or greater than
1921 pkt_size on the server. Otherwise the received message may be truncated causing
1927 @c man end PROTOCOLS