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.
348 The Gopher protocol with TLS encapsulation.
352 Read Apple HTTP Live Streaming compliant segmented stream as
353 a uniform one. The M3U8 playlists describing the segments can be
354 remote HTTP resources or local files, accessed using the standard
356 The nested protocol is declared by specifying
357 "+@var{proto}" after the hls URI scheme name, where @var{proto}
358 is either "file" or "http".
361 hls+http://host/path/to/remote/resource.m3u8
362 hls+file://path/to/local/resource.m3u8
365 Using this protocol is discouraged - the hls demuxer should work
366 just as well (if not, please report the issues) and is more complete.
367 To use the hls demuxer instead, simply use the direct URLs to the
372 HTTP (Hyper Text Transfer Protocol).
374 This protocol accepts the following options:
378 Control seekability of connection. If set to 1 the resource is
379 supposed to be seekable, if set to 0 it is assumed not to be seekable,
380 if set to -1 it will try to autodetect if it is seekable. Default
384 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
387 Set a specific content type for the POST messages or for listen mode.
390 set HTTP proxy to tunnel through e.g. http://example.com:1234
393 Set custom HTTP headers, can override built in default headers. The
394 value must be a string encoding the headers.
396 @item multiple_requests
397 Use persistent connections if set to 1, default is 0.
400 Set custom HTTP post data.
403 Set the Referer header. Include 'Referer: URL' header in HTTP request.
406 Override the User-Agent header. If not specified the protocol will use a
407 string describing the libavformat build. ("Lavf/<version>")
410 This is a deprecated option, you can use user_agent instead it.
412 @item reconnect_at_eof
413 If set then eof is treated like an error and causes reconnection, this is useful
414 for live / endless streams.
416 @item reconnect_streamed
417 If set then even streamed/non seekable streams will be reconnected on errors.
419 @item reconnect_on_network_error
420 Reconnect automatically in case of TCP/TLS errors during connect.
422 @item reconnect_on_http_error
423 A comma separated list of HTTP status codes to reconnect on. The list can
424 include specific status codes (e.g. '503') or the strings '4xx' / '5xx'.
426 @item reconnect_delay_max
427 Sets the maximum delay in seconds after which to give up reconnecting
430 Export the MIME type.
433 Exports the HTTP response version number. Usually "1.0" or "1.1".
436 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
437 supports this, the metadata has to be retrieved by the application by reading
438 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
441 @item icy_metadata_headers
442 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
443 headers, separated by newline characters.
445 @item icy_metadata_packet
446 If the server supports ICY metadata, and @option{icy} was set to 1, this
447 contains the last non-empty metadata packet sent by the server. It should be
448 polled in regular intervals by applications interested in mid-stream metadata
452 Set the cookies to be sent in future requests. The format of each cookie is the
453 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
454 delimited by a newline character.
457 Set initial byte offset.
460 Try to limit the request to bytes preceding this offset.
463 When used as a client option it sets the HTTP method for the request.
465 When used as a server option it sets the HTTP method that is going to be
466 expected from the client(s).
467 If the expected and the received HTTP method do not match the client will
468 be given a Bad Request response.
469 When unset the HTTP method is not checked for now. This will be replaced by
470 autodetection in the future.
473 If set to 1 enables experimental HTTP server. This can be used to send data when
474 used as an output option, or read data from a client with HTTP POST when used as
476 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
477 in ffmpeg.c and thus must not be used as a command line option.
479 # Server side (sending):
480 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
482 # Client side (receiving):
483 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
485 # Client can also be done with wget:
486 wget http://@var{server}:@var{port} -O somefile.ogg
488 # Server side (receiving):
489 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
491 # Client side (sending):
492 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
494 # Client can also be done with wget:
495 wget --post-file=somefile.ogg http://@var{server}:@var{port}
498 @item send_expect_100
499 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
500 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
505 Set HTTP authentication type. No option for Digest, since this method requires
506 getting nonce parameters from the server first and can't be used straight away like
511 Choose the HTTP authentication type automatically. This is the default.
514 Choose the HTTP basic authentication.
516 Basic authentication sends a Base64-encoded string that contains a user name and password
517 for the client. Base64 is not a form of encryption and should be considered the same as
518 sending the user name and password in clear text (Base64 is a reversible encoding).
519 If a resource needs to be protected, strongly consider using an authentication scheme
520 other than basic authentication. HTTPS/TLS should be used with basic authentication.
521 Without these additional security enhancements, basic authentication should not be used
522 to protect sensitive or valuable information.
527 @subsection HTTP Cookies
529 Some HTTP requests will be denied unless cookie values are passed in with the
530 request. The @option{cookies} option allows these cookies to be specified. At
531 the very least, each cookie must specify a value along with a path and domain.
532 HTTP requests that match both the domain and path will automatically include the
533 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
536 The required syntax to play a stream specifying a cookie is:
538 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
543 Icecast protocol (stream to Icecast servers)
545 This protocol accepts the following options:
549 Set the stream genre.
554 @item ice_description
555 Set the stream description.
558 Set the stream website URL.
561 Set if the stream should be public.
562 The default is 0 (not public).
565 Override the User-Agent header. If not specified a string of the form
566 "Lavf/<version>" will be used.
569 Set the Icecast mountpoint password.
572 Set the stream content type. This must be set if it is different from
576 This enables support for Icecast versions < 2.4.0, that do not support the
577 HTTP PUT method but the SOURCE method.
580 Establish a TLS (HTTPS) connection to Icecast.
585 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
590 MMS (Microsoft Media Server) protocol over TCP.
594 MMS (Microsoft Media Server) protocol over HTTP.
596 The required syntax is:
598 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
605 Computes the MD5 hash of the data to be written, and on close writes
606 this to the designated output or stdout if none is specified. It can
607 be used to test muxers without writing an actual file.
609 Some examples follow.
611 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
612 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
614 # Write the MD5 hash of the encoded AVI file to stdout.
615 ffmpeg -i input.flv -f avi -y md5:
618 Note that some formats (typically MOV) require the output protocol to
619 be seekable, so they will fail with the MD5 output protocol.
623 UNIX pipe access protocol.
625 Read and write from UNIX pipes.
627 The accepted syntax is:
632 @var{number} is the number corresponding to the file descriptor of the
633 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
634 is not specified, by default the stdout file descriptor will be used
635 for writing, stdin for reading.
637 For example to read from stdin with @command{ffmpeg}:
639 cat test.wav | ffmpeg -i pipe:0
640 # ...this is the same as...
641 cat test.wav | ffmpeg -i pipe:
644 For writing to stdout with @command{ffmpeg}:
646 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
647 # ...this is the same as...
648 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
651 This protocol accepts the following options:
655 Set I/O operation maximum block size, in bytes. Default value is
656 @code{INT_MAX}, which results in not limiting the requested block size.
657 Setting this value reasonably low improves user termination request reaction
658 time, which is valuable if data transmission is slow.
661 Note that some formats (typically MOV), require the output protocol to
662 be seekable, so they will fail with the pipe output protocol.
666 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
668 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
669 for MPEG-2 Transport Streams sent over RTP.
671 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
672 the @code{rtp} protocol.
674 The required syntax is:
676 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
679 The destination UDP ports are @code{port + 2} for the column FEC stream
680 and @code{port + 4} for the row FEC stream.
682 This protocol accepts the following options:
686 The number of columns (4-20, LxD <= 100)
689 The number of rows (4-20, LxD <= 100)
696 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
701 Reliable Internet Streaming Transport protocol
703 The accepted options are:
715 Set internal RIST buffer size for retransmission of data.
718 Set maximum packet size for sending data. 1316 by default.
721 Set loglevel for RIST logging messages.
724 Set override of encryption secret, by default is unset.
727 Set encryption type, by default is disabled.
728 Acceptable values are 128 and 256.
733 Real-Time Messaging Protocol.
735 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
736 content across a TCP/IP network.
738 The required syntax is:
740 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
743 The accepted parameters are:
747 An optional username (mostly for publishing).
750 An optional password (mostly for publishing).
753 The address of the RTMP server.
756 The number of the TCP port to use (by default is 1935).
759 It is the name of the application to access. It usually corresponds to
760 the path where the application is installed on the RTMP server
761 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
762 the value parsed from the URI through the @code{rtmp_app} option, too.
765 It is the path or name of the resource to play with reference to the
766 application specified in @var{app}, may be prefixed by "mp4:". You
767 can override the value parsed from the URI through the @code{rtmp_playpath}
771 Act as a server, listening for an incoming connection.
774 Maximum time to wait for the incoming connection. Implies listen.
777 Additionally, the following parameters can be set via command line options
778 (or in code via @code{AVOption}s):
782 Name of application to connect on the RTMP server. This option
783 overrides the parameter specified in the URI.
786 Set the client buffer time in milliseconds. The default is 3000.
789 Extra arbitrary AMF connection parameters, parsed from a string,
790 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
791 Each value is prefixed by a single character denoting the type,
792 B for Boolean, N for number, S for string, O for object, or Z for null,
793 followed by a colon. For Booleans the data must be either 0 or 1 for
794 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
795 1 to end or begin an object, respectively. Data items in subobjects may
796 be named, by prefixing the type with 'N' and specifying the name before
797 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
798 times to construct arbitrary AMF sequences.
801 Version of the Flash plugin used to run the SWF player. The default
802 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
803 <libavformat version>).)
805 @item rtmp_flush_interval
806 Number of packets flushed in the same request (RTMPT only). The default
810 Specify that the media is a live stream. No resuming or seeking in
811 live streams is possible. The default value is @code{any}, which means the
812 subscriber first tries to play the live stream specified in the
813 playpath. If a live stream of that name is not found, it plays the
814 recorded stream. The other possible values are @code{live} and
818 URL of the web page in which the media was embedded. By default no
822 Stream identifier to play or to publish. This option overrides the
823 parameter specified in the URI.
826 Name of live stream to subscribe to. By default no value will be sent.
827 It is only sent if the option is specified or if rtmp_live
831 SHA256 hash of the decompressed SWF file (32 bytes).
834 Size of the decompressed SWF file, required for SWFVerification.
837 URL of the SWF player for the media. By default no value will be sent.
840 URL to player swf file, compute hash/size automatically.
843 URL of the target stream. Defaults to proto://host[:port]/app.
847 For example to read with @command{ffplay} a multimedia resource named
848 "sample" from the application "vod" from an RTMP server "myserver":
850 ffplay rtmp://myserver/vod/sample
853 To publish to a password protected server, passing the playpath and
854 app names separately:
856 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
861 Encrypted Real-Time Messaging Protocol.
863 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
864 streaming multimedia content within standard cryptographic primitives,
865 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
870 Real-Time Messaging Protocol over a secure SSL connection.
872 The Real-Time Messaging Protocol (RTMPS) is used for streaming
873 multimedia content across an encrypted connection.
877 Real-Time Messaging Protocol tunneled through HTTP.
879 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
880 for streaming multimedia content within HTTP requests to traverse
885 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
887 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
888 is used for streaming multimedia content within HTTP requests to traverse
893 Real-Time Messaging Protocol tunneled through HTTPS.
895 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
896 for streaming multimedia content within HTTPS requests to traverse
899 @section libsmbclient
901 libsmbclient permits one to manipulate CIFS/SMB network resources.
903 Following syntax is required.
906 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
909 This protocol accepts the following options.
913 Set timeout in milliseconds of socket I/O operations used by the underlying
914 low level operation. By default it is set to -1, which means that the timeout
918 Truncate existing files on write, if set to 1. A value of 0 prevents
919 truncating. Default value is 1.
922 Set the workgroup used for making connections. By default workgroup is not specified.
926 For more information see: @url{http://www.samba.org/}.
930 Secure File Transfer Protocol via libssh
932 Read from or write to remote resources using SFTP protocol.
934 Following syntax is required.
937 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
940 This protocol accepts the following options.
944 Set timeout of socket I/O operations used by the underlying low level
945 operation. By default it is set to -1, which means that the timeout
949 Truncate existing files on write, if set to 1. A value of 0 prevents
950 truncating. Default value is 1.
953 Specify the path of the file containing private key to use during authorization.
954 By default libssh searches for keys in the @file{~/.ssh/} directory.
958 Example: Play a file stored on remote server.
961 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
964 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
966 Real-Time Messaging Protocol and its variants supported through
969 Requires the presence of the librtmp headers and library during
970 configuration. You need to explicitly configure the build with
971 "--enable-librtmp". If enabled this will replace the native RTMP
974 This protocol provides most client functions and a few server
975 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
976 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
977 variants of these encrypted types (RTMPTE, RTMPTS).
979 The required syntax is:
981 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
984 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
985 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
986 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
987 meaning as specified for the RTMP native protocol.
988 @var{options} contains a list of space-separated options of the form
991 See the librtmp manual page (man 3 librtmp) for more information.
993 For example, to stream a file in real-time to an RTMP server using
996 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
999 To play the same stream using @command{ffplay}:
1001 ffplay "rtmp://myserver/live/mystream live=1"
1006 Real-time Transport Protocol.
1008 The required syntax for an RTP URL is:
1009 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
1011 @var{port} specifies the RTP port to use.
1013 The following URL options are supported:
1018 Set the TTL (Time-To-Live) value (for multicast only).
1020 @item rtcpport=@var{n}
1021 Set the remote RTCP port to @var{n}.
1023 @item localrtpport=@var{n}
1024 Set the local RTP port to @var{n}.
1026 @item localrtcpport=@var{n}'
1027 Set the local RTCP port to @var{n}.
1029 @item pkt_size=@var{n}
1030 Set max packet size (in bytes) to @var{n}.
1033 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
1036 @item sources=@var{ip}[,@var{ip}]
1037 List allowed source IP addresses.
1039 @item block=@var{ip}[,@var{ip}]
1040 List disallowed (blocked) source IP addresses.
1042 @item write_to_source=0|1
1043 Send packets to the source address of the latest received packet (if
1044 set to 1) or to a default remote address (if set to 0).
1046 @item localport=@var{n}
1047 Set the local RTP port to @var{n}.
1049 @item timeout=@var{n}
1050 Set timeout (in microseconds) of socket I/O operations to @var{n}.
1052 This is a deprecated option. Instead, @option{localrtpport} should be
1062 If @option{rtcpport} is not set the RTCP port will be set to the RTP
1066 If @option{localrtpport} (the local RTP port) is not set any available
1067 port will be used for the local RTP and RTCP ports.
1070 If @option{localrtcpport} (the local RTCP port) is not set it will be
1071 set to the local RTP port value plus 1.
1076 Real-Time Streaming Protocol.
1078 RTSP is not technically a protocol handler in libavformat, it is a demuxer
1079 and muxer. The demuxer supports both normal RTSP (with data transferred
1080 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
1081 data transferred over RDT).
1083 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1084 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
1085 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
1087 The required syntax for a RTSP url is:
1089 rtsp://@var{hostname}[:@var{port}]/@var{path}
1092 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1093 line, or set in code via @code{AVOption}s or in
1094 @code{avformat_open_input}.
1096 The following options are supported.
1100 Do not start playing the stream immediately if set to 1. Default value
1103 @item rtsp_transport
1104 Set RTSP transport protocols.
1106 It accepts the following values:
1109 Use UDP as lower transport protocol.
1112 Use TCP (interleaving within the RTSP control channel) as lower
1116 Use UDP multicast as lower transport protocol.
1119 Use HTTP tunneling as lower transport protocol, which is useful for
1123 Multiple lower transport protocols may be specified, in that case they are
1124 tried one at a time (if the setup of one fails, the next one is tried).
1125 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
1130 The following values are accepted:
1133 Accept packets only from negotiated peer address and port.
1135 Act as a server, listening for an incoming connection.
1137 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
1140 Default value is @samp{none}.
1142 @item allowed_media_types
1143 Set media types to accept from the server.
1145 The following flags are accepted:
1152 By default it accepts all media types.
1155 Set minimum local UDP port. Default value is 5000.
1158 Set maximum local UDP port. Default value is 65000.
1161 Set maximum timeout (in seconds) to wait for incoming connections.
1163 A value of -1 means infinite (default). This option implies the
1164 @option{rtsp_flags} set to @samp{listen}.
1166 @item reorder_queue_size
1167 Set number of packets to buffer for handling of reordered packets.
1170 Set socket TCP I/O timeout in microseconds.
1173 Override User-Agent header. If not specified, it defaults to the
1174 libavformat identifier string.
1177 When receiving data over UDP, the demuxer tries to reorder received packets
1178 (since they may arrive out of order, or packets may get lost totally). This
1179 can be disabled by setting the maximum demuxing delay to zero (via
1180 the @code{max_delay} field of AVFormatContext).
1182 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1183 streams to display can be chosen with @code{-vst} @var{n} and
1184 @code{-ast} @var{n} for video and audio respectively, and can be switched
1185 on the fly by pressing @code{v} and @code{a}.
1187 @subsection Examples
1189 The following examples all make use of the @command{ffplay} and
1190 @command{ffmpeg} tools.
1194 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1196 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1200 Watch a stream tunneled over HTTP:
1202 ffplay -rtsp_transport http rtsp://server/video.mp4
1206 Send a stream in realtime to a RTSP server, for others to watch:
1208 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1212 Receive a stream in realtime:
1214 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1220 Session Announcement Protocol (RFC 2974). This is not technically a
1221 protocol handler in libavformat, it is a muxer and demuxer.
1222 It is used for signalling of RTP streams, by announcing the SDP for the
1223 streams regularly on a separate port.
1227 The syntax for a SAP url given to the muxer is:
1229 sap://@var{destination}[:@var{port}][?@var{options}]
1232 The RTP packets are sent to @var{destination} on port @var{port},
1233 or to port 5004 if no port is specified.
1234 @var{options} is a @code{&}-separated list. The following options
1239 @item announce_addr=@var{address}
1240 Specify the destination IP address for sending the announcements to.
1241 If omitted, the announcements are sent to the commonly used SAP
1242 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1243 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1245 @item announce_port=@var{port}
1246 Specify the port to send the announcements on, defaults to
1247 9875 if not specified.
1250 Specify the time to live value for the announcements and RTP packets,
1253 @item same_port=@var{0|1}
1254 If set to 1, send all RTP streams on the same port pair. If zero (the
1255 default), all streams are sent on unique ports, with each stream on a
1256 port 2 numbers higher than the previous.
1257 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1258 The RTP stack in libavformat for receiving requires all streams to be sent
1262 Example command lines follow.
1264 To broadcast a stream on the local subnet, for watching in VLC:
1267 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1270 Similarly, for watching in @command{ffplay}:
1273 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1276 And for watching in @command{ffplay}, over IPv6:
1279 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1284 The syntax for a SAP url given to the demuxer is:
1286 sap://[@var{address}][:@var{port}]
1289 @var{address} is the multicast address to listen for announcements on,
1290 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1291 is the port that is listened on, 9875 if omitted.
1293 The demuxers listens for announcements on the given address and port.
1294 Once an announcement is received, it tries to receive that particular stream.
1296 Example command lines follow.
1298 To play back the first stream announced on the normal SAP multicast address:
1304 To play back the first stream announced on one the default IPv6 SAP multicast address:
1307 ffplay sap://[ff0e::2:7ffe]
1312 Stream Control Transmission Protocol.
1314 The accepted URL syntax is:
1316 sctp://@var{host}:@var{port}[?@var{options}]
1319 The protocol accepts the following options:
1322 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1325 Set the maximum number of streams. By default no limit is set.
1330 Haivision Secure Reliable Transport Protocol via libsrt.
1332 The supported syntax for a SRT URL is:
1334 srt://@var{hostname}:@var{port}[?@var{options}]
1337 @var{options} contains a list of &-separated options of the form
1338 @var{key}=@var{val}.
1343 @var{options} srt://@var{hostname}:@var{port}
1346 @var{options} contains a list of '-@var{key} @var{val}'
1349 This protocol accepts the following options.
1352 @item connect_timeout=@var{milliseconds}
1353 Connection timeout; SRT cannot connect for RTT > 1500 msec
1354 (2 handshake exchanges) with the default connect timeout of
1355 3 seconds. This option applies to the caller and rendezvous
1356 connection modes. The connect timeout is 10 times the value
1357 set for the rendezvous mode (which can be used as a
1358 workaround for this connection problem with earlier versions).
1360 @item ffs=@var{bytes}
1361 Flight Flag Size (Window Size), in bytes. FFS is actually an
1362 internal parameter and you should set it to not less than
1363 @option{recv_buffer_size} and @option{mss}. The default value
1364 is relatively large, therefore unless you set a very large receiver buffer,
1365 you do not need to change this option. Default value is 25600.
1367 @item inputbw=@var{bytes/seconds}
1368 Sender nominal input rate, in bytes per seconds. Used along with
1369 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1370 calculate maximum sending rate when recovery packets are sent
1371 along with the main media stream:
1372 @option{inputbw} * (100 + @option{oheadbw}) / 100
1373 if @option{inputbw} is not set while @option{maxbw} is set to
1374 relative (0), the actual input rate is evaluated inside
1375 the library. Default value is 0.
1377 @item iptos=@var{tos}
1378 IP Type of Service. Applies to sender only. Default value is 0xB8.
1380 @item ipttl=@var{ttl}
1381 IP Time To Live. Applies to sender only. Default value is 64.
1383 @item latency=@var{microseconds}
1384 Timestamp-based Packet Delivery Delay.
1385 Used to absorb bursts of missed packet retransmissions.
1386 This flag sets both @option{rcvlatency} and @option{peerlatency}
1387 to the same value. Note that prior to version 1.3.0
1388 this is the only flag to set the latency, however
1389 this is effectively equivalent to setting @option{peerlatency},
1390 when side is sender and @option{rcvlatency}
1391 when side is receiver, and the bidirectional stream
1392 sending is not supported.
1394 @item listen_timeout=@var{microseconds}
1395 Set socket listen timeout.
1397 @item maxbw=@var{bytes/seconds}
1398 Maximum sending bandwidth, in bytes per seconds.
1399 -1 infinite (CSRTCC limit is 30mbps)
1400 0 relative to input rate (see @option{inputbw})
1401 >0 absolute limit value
1402 Default value is 0 (relative)
1404 @item mode=@var{caller|listener|rendezvous}
1406 @option{caller} opens client connection.
1407 @option{listener} starts server to listen for incoming connections.
1408 @option{rendezvous} use Rendez-Vous connection mode.
1409 Default value is caller.
1411 @item mss=@var{bytes}
1412 Maximum Segment Size, in bytes. Used for buffer allocation
1413 and rate calculation using a packet counter assuming fully
1414 filled packets. The smallest MSS between the peers is
1415 used. This is 1500 by default in the overall internet.
1416 This is the maximum size of the UDP packet and can be
1417 only decreased, unless you have some unusual dedicated
1418 network settings. Default value is 1500.
1420 @item nakreport=@var{1|0}
1421 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1422 periodically until a lost packet is retransmitted or
1423 intentionally dropped. Default value is 1.
1425 @item oheadbw=@var{percents}
1426 Recovery bandwidth overhead above input rate, in percents.
1427 See @option{inputbw}. Default value is 25%.
1429 @item passphrase=@var{string}
1430 HaiCrypt Encryption/Decryption Passphrase string, length
1431 from 10 to 79 characters. The passphrase is the shared
1432 secret between the sender and the receiver. It is used
1433 to generate the Key Encrypting Key using PBKDF2
1434 (Password-Based Key Derivation Function). It is used
1435 only if @option{pbkeylen} is non-zero. It is used on
1436 the receiver only if the received data is encrypted.
1437 The configured passphrase cannot be recovered (write-only).
1439 @item enforced_encryption=@var{1|0}
1440 If true, both connection parties must have the same password
1441 set (including empty, that is, with no encryption). If the
1442 password doesn't match or only one side is unencrypted,
1443 the connection is rejected. Default is true.
1445 @item kmrefreshrate=@var{packets}
1446 The number of packets to be transmitted after which the
1447 encryption key is switched to a new key. Default is -1.
1448 -1 means auto (0x1000000 in srt library). The range for
1449 this option is integers in the 0 - @code{INT_MAX}.
1451 @item kmpreannounce=@var{packets}
1452 The interval between when a new encryption key is sent and
1453 when switchover occurs. This value also applies to the
1454 subsequent interval between when switchover occurs and
1455 when the old encryption key is decommissioned. Default is -1.
1456 -1 means auto (0x1000 in srt library). The range for
1457 this option is integers in the 0 - @code{INT_MAX}.
1459 @item payload_size=@var{bytes}
1460 Sets the maximum declared size of a packet transferred
1461 during the single call to the sending function in Live
1462 mode. Use 0 if this value isn't used (which is default in
1464 Default is -1 (automatic), which typically means MPEG-TS;
1465 if you are going to use SRT
1466 to send any different kind of payload, such as, for example,
1467 wrapping a live stream in very small frames, then you can
1468 use a bigger maximum frame size, though not greater than
1471 @item pkt_size=@var{bytes}
1472 Alias for @samp{payload_size}.
1474 @item peerlatency=@var{microseconds}
1475 The latency value (as described in @option{rcvlatency}) that is
1476 set by the sender side as a minimum value for the receiver.
1478 @item pbkeylen=@var{bytes}
1479 Sender encryption key length, in bytes.
1480 Only can be set to 0, 16, 24 and 32.
1481 Enable sender encryption if not 0.
1482 Not required on receiver (set to 0),
1483 key size obtained from sender in HaiCrypt handshake.
1486 @item rcvlatency=@var{microseconds}
1487 The time that should elapse since the moment when the
1488 packet was sent and the moment when it's delivered to
1489 the receiver application in the receiving function.
1490 This time should be a buffer time large enough to cover
1491 the time spent for sending, unexpectedly extended RTT
1492 time, and the time needed to retransmit the lost UDP
1493 packet. The effective latency value will be the maximum
1494 of this options' value and the value of @option{peerlatency}
1495 set by the peer side. Before version 1.3.0 this option
1496 is only available as @option{latency}.
1498 @item recv_buffer_size=@var{bytes}
1499 Set UDP receive buffer size, expressed in bytes.
1501 @item send_buffer_size=@var{bytes}
1502 Set UDP send buffer size, expressed in bytes.
1504 @item timeout=@var{microseconds}
1505 Set raise error timeouts for read, write and connect operations. Note that the
1506 SRT library has internal timeouts which can be controlled separately, the
1507 value set here is only a cap on those.
1509 @item tlpktdrop=@var{1|0}
1510 Too-late Packet Drop. When enabled on receiver, it skips
1511 missing packets that have not been delivered in time and
1512 delivers the following packets to the application when
1513 their time-to-play has come. It also sends a fake ACK to
1514 the sender. When enabled on sender and enabled on the
1515 receiving peer, the sender drops the older packets that
1516 have no chance of being delivered in time. It was
1517 automatically enabled in the sender if the receiver
1520 @item sndbuf=@var{bytes}
1521 Set send buffer size, expressed in bytes.
1523 @item rcvbuf=@var{bytes}
1524 Set receive buffer size, expressed in bytes.
1526 Receive buffer must not be greater than @option{ffs}.
1528 @item lossmaxttl=@var{packets}
1529 The value up to which the Reorder Tolerance may grow. When
1530 Reorder Tolerance is > 0, then packet loss report is delayed
1531 until that number of packets come in. Reorder Tolerance
1532 increases every time a "belated" packet has come, but it
1533 wasn't due to retransmission (that is, when UDP packets tend
1534 to come out of order), with the difference between the latest
1535 sequence and this packet's sequence, and not more than the
1536 value of this option. By default it's 0, which means that this
1537 mechanism is turned off, and the loss report is always sent
1538 immediately upon experiencing a "gap" in sequences.
1541 The minimum SRT version that is required from the peer. A connection
1542 to a peer that does not satisfy the minimum version requirement
1545 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1548 @item streamid=@var{string}
1549 A string limited to 512 characters that can be set on the socket prior
1550 to connecting. This stream ID will be able to be retrieved by the
1551 listener side from the socket that is returned from srt_accept and
1552 was connected by a socket with that set stream ID. SRT does not enforce
1553 any special interpretation of the contents of this string.
1554 This option doesn’t make sense in Rendezvous connection; the result
1555 might be that simply one side will override the value from the other
1556 side and it’s the matter of luck which one would win
1558 @item smoother=@var{live|file}
1559 The type of Smoother used for the transmission for that socket, which
1560 is responsible for the transmission and congestion control. The Smoother
1561 type must be exactly the same on both connecting parties, otherwise
1562 the connection is rejected.
1564 @item messageapi=@var{1|0}
1565 When set, this socket uses the Message API, otherwise it uses Buffer
1566 API. Note that in live mode (see @option{transtype}) there’s only
1567 message API available. In File mode you can chose to use one of two modes:
1569 Stream API (default, when this option is false). In this mode you may
1570 send as many data as you wish with one sending instruction, or even use
1571 dedicated functions that read directly from a file. The internal facility
1572 will take care of any speed and congestion control. When receiving, you
1573 can also receive as many data as desired, the data not extracted will be
1574 waiting for the next call. There is no boundary between data portions in
1577 Message API. In this mode your single sending instruction passes exactly
1578 one piece of data that has boundaries (a message). Contrary to Live mode,
1579 this message may span across multiple UDP packets and the only size
1580 limitation is that it shall fit as a whole in the sending buffer. The
1581 receiver shall use as large buffer as necessary to receive the message,
1582 otherwise the message will not be given up. When the message is not
1583 complete (not all packets received or there was a packet loss) it will
1586 @item transtype=@var{live|file}
1587 Sets the transmission type for the socket, in particular, setting this
1588 option sets multiple other parameters to their default values as required
1589 for a particular transmission type.
1591 live: Set options as for live transmission. In this mode, you should
1592 send by one sending instruction only so many data that fit in one UDP packet,
1593 and limited to the value defined first in @option{payload_size} (1316 is
1594 default in this mode). There is no speed control in this mode, only the
1595 bandwidth control, if configured, in order to not exceed the bandwidth with
1596 the overhead transmission (retransmitted and control packets).
1598 file: Set options as for non-live transmission. See @option{messageapi}
1599 for further explanations
1601 @item linger=@var{seconds}
1602 The number of seconds that the socket waits for unsent data when closing.
1603 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
1604 seconds in file mode). The range for this option is integers in the
1609 For more information see: @url{https://github.com/Haivision/srt}.
1613 Secure Real-time Transport Protocol.
1615 The accepted options are:
1618 @item srtp_out_suite
1619 Select input and output encoding suites.
1623 @item AES_CM_128_HMAC_SHA1_80
1624 @item SRTP_AES128_CM_HMAC_SHA1_80
1625 @item AES_CM_128_HMAC_SHA1_32
1626 @item SRTP_AES128_CM_HMAC_SHA1_32
1629 @item srtp_in_params
1630 @item srtp_out_params
1631 Set input and output encoding parameters, which are expressed by a
1632 base64-encoded representation of a binary block. The first 16 bytes of
1633 this binary block are used as master key, the following 14 bytes are
1634 used as master salt.
1639 Virtually extract a segment of a file or another stream.
1640 The underlying stream must be seekable.
1645 Start offset of the extracted segment, in bytes.
1647 End offset of the extracted segment, in bytes.
1648 If set to 0, extract till end of file.
1653 Extract a chapter from a DVD VOB file (start and end sectors obtained
1654 externally and multiplied by 2048):
1656 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1659 Play an AVI file directly from a TAR archive:
1661 subfile,,start,183241728,end,366490624,,:archive.tar
1664 Play a MPEG-TS file from start offset till end:
1666 subfile,,start,32815239,end,0,,:video.ts
1671 Writes the output to multiple protocols. The individual outputs are separated
1675 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1680 Transmission Control Protocol.
1682 The required syntax for a TCP url is:
1684 tcp://@var{hostname}:@var{port}[?@var{options}]
1687 @var{options} contains a list of &-separated options of the form
1688 @var{key}=@var{val}.
1690 The list of supported options follows.
1693 @item listen=@var{2|1|0}
1694 Listen for an incoming connection. 0 disables listen, 1 enables listen in
1695 single client mode, 2 enables listen in multi-client mode. Default value is 0.
1697 @item timeout=@var{microseconds}
1698 Set raise error timeout, expressed in microseconds.
1700 This option is only relevant in read mode: if no data arrived in more
1701 than this time interval, raise error.
1703 @item listen_timeout=@var{milliseconds}
1704 Set listen timeout, expressed in milliseconds.
1706 @item recv_buffer_size=@var{bytes}
1707 Set receive buffer size, expressed bytes.
1709 @item send_buffer_size=@var{bytes}
1710 Set send buffer size, expressed bytes.
1712 @item tcp_nodelay=@var{1|0}
1713 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1715 @item tcp_mss=@var{bytes}
1716 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1719 The following example shows how to setup a listening TCP connection
1720 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1722 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1723 ffplay tcp://@var{hostname}:@var{port}
1728 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1730 The required syntax for a TLS/SSL url is:
1732 tls://@var{hostname}:@var{port}[?@var{options}]
1735 The following parameters can be set via command line options
1736 (or in code via @code{AVOption}s):
1740 @item ca_file, cafile=@var{filename}
1741 A file containing certificate authority (CA) root certificates to treat
1742 as trusted. If the linked TLS library contains a default this might not
1743 need to be specified for verification to work, but not all libraries and
1744 setups have defaults built in.
1745 The file must be in OpenSSL PEM format.
1747 @item tls_verify=@var{1|0}
1748 If enabled, try to verify the peer that we are communicating with.
1749 Note, if using OpenSSL, this currently only makes sure that the
1750 peer certificate is signed by one of the root certificates in the CA
1751 database, but it does not validate that the certificate actually
1752 matches the host name we are trying to connect to. (With other backends,
1753 the host name is validated as well.)
1755 This is disabled by default since it requires a CA database to be
1756 provided by the caller in many cases.
1758 @item cert_file, cert=@var{filename}
1759 A file containing a certificate to use in the handshake with the peer.
1760 (When operating as server, in listen mode, this is more often required
1761 by the peer, while client certificates only are mandated in certain
1764 @item key_file, key=@var{filename}
1765 A file containing the private key for the certificate.
1767 @item listen=@var{1|0}
1768 If enabled, listen for connections on the provided port, and assume
1769 the server role in the handshake instead of the client role.
1773 Example command lines:
1775 To create a TLS/SSL server that serves an input stream.
1778 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1781 To play back a stream from the TLS/SSL server using @command{ffplay}:
1784 ffplay tls://@var{hostname}:@var{port}
1789 User Datagram Protocol.
1791 The required syntax for an UDP URL is:
1793 udp://@var{hostname}:@var{port}[?@var{options}]
1796 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1798 In case threading is enabled on the system, a circular buffer is used
1799 to store the incoming data, which allows one to reduce loss of data due to
1800 UDP socket buffer overruns. The @var{fifo_size} and
1801 @var{overrun_nonfatal} options are related to this buffer.
1803 The list of supported options follows.
1806 @item buffer_size=@var{size}
1807 Set the UDP maximum socket buffer size in bytes. This is used to set either
1808 the receive or send buffer size, depending on what the socket is used for.
1809 Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
1811 @item bitrate=@var{bitrate}
1812 If set to nonzero, the output will have the specified constant bitrate if the
1813 input has enough packets to sustain it.
1815 @item burst_bits=@var{bits}
1816 When using @var{bitrate} this specifies the maximum number of bits in
1819 @item localport=@var{port}
1820 Override the local UDP port to bind with.
1822 @item localaddr=@var{addr}
1823 Local IP address of a network interface used for sending packets or joining
1826 @item pkt_size=@var{size}
1827 Set the size in bytes of UDP packets.
1829 @item reuse=@var{1|0}
1830 Explicitly allow or disallow reusing UDP sockets.
1833 Set the time to live value (for multicast only).
1835 @item connect=@var{1|0}
1836 Initialize the UDP socket with @code{connect()}. In this case, the
1837 destination address can't be changed with ff_udp_set_remote_url later.
1838 If the destination address isn't known at the start, this option can
1839 be specified in ff_udp_set_remote_url, too.
1840 This allows finding out the source address for the packets with getsockname,
1841 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1842 unreachable" is received.
1843 For receiving, this gives the benefit of only receiving packets from
1844 the specified peer address/port.
1846 @item sources=@var{address}[,@var{address}]
1847 Only receive packets sent from the specified addresses. In case of multicast,
1848 also subscribe to multicast traffic coming from these addresses only.
1850 @item block=@var{address}[,@var{address}]
1851 Ignore packets sent from the specified addresses. In case of multicast, also
1852 exclude the source addresses in the multicast subscription.
1854 @item fifo_size=@var{units}
1855 Set the UDP receiving circular buffer size, expressed as a number of
1856 packets with size of 188 bytes. If not specified defaults to 7*4096.
1858 @item overrun_nonfatal=@var{1|0}
1859 Survive in case of UDP receiving circular buffer overrun. Default
1862 @item timeout=@var{microseconds}
1863 Set raise error timeout, expressed in microseconds.
1865 This option is only relevant in read mode: if no data arrived in more
1866 than this time interval, raise error.
1868 @item broadcast=@var{1|0}
1869 Explicitly allow or disallow UDP broadcasting.
1871 Note that broadcasting may not work properly on networks having
1872 a broadcast storm protection.
1875 @subsection Examples
1879 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1881 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1885 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1886 sized UDP packets, using a large input buffer:
1888 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1892 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1894 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1902 The required syntax for a Unix socket URL is:
1905 unix://@var{filepath}
1908 The following parameters can be set via command line options
1909 (or in code via @code{AVOption}s):
1915 Create the Unix socket in listening mode.
1920 ZeroMQ asynchronous messaging using the libzmq library.
1922 This library supports unicast streaming to multiple clients without relying on
1925 The required syntax for streaming or connecting to a stream is:
1927 zmq:tcp://ip-address:port
1931 Create a localhost stream on port 5555:
1933 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1936 Multiple clients may connect to the stream using:
1938 ffplay zmq:tcp://127.0.0.1:5555
1941 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
1942 The server side binds to a port and publishes data. Clients connect to the
1943 server (via IP address/port) and subscribe to the stream. The order in which
1944 the server and client start generally does not matter.
1946 ffmpeg must be compiled with the --enable-libzmq option to support
1949 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1950 line. The following options are supported:
1955 Forces the maximum packet size for sending/receiving data. The default value is
1956 131,072 bytes. On the server side, this sets the maximum size of sent packets
1957 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
1958 packets. Note that pkt_size on the clients should be equal to or greater than
1959 pkt_size on the server. Otherwise the received message may be truncated causing
1965 @c man end PROTOCOLS