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 in milliseconds for retransmission of data.
716 Default value is 0 which means the librist default (1 sec). Maximum value is 30
720 Set maximum packet size for sending data. 1316 by default.
723 Set loglevel for RIST logging messages. You only need to set this if you
724 explicitly want to enable debug level messages or packet loss simulation,
725 otherwise the regular loglevel is respected.
728 Set override of encryption secret, by default is unset.
731 Set encryption type, by default is disabled.
732 Acceptable values are 128 and 256.
737 Real-Time Messaging Protocol.
739 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
740 content across a TCP/IP network.
742 The required syntax is:
744 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
747 The accepted parameters are:
751 An optional username (mostly for publishing).
754 An optional password (mostly for publishing).
757 The address of the RTMP server.
760 The number of the TCP port to use (by default is 1935).
763 It is the name of the application to access. It usually corresponds to
764 the path where the application is installed on the RTMP server
765 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
766 the value parsed from the URI through the @code{rtmp_app} option, too.
769 It is the path or name of the resource to play with reference to the
770 application specified in @var{app}, may be prefixed by "mp4:". You
771 can override the value parsed from the URI through the @code{rtmp_playpath}
775 Act as a server, listening for an incoming connection.
778 Maximum time to wait for the incoming connection. Implies listen.
781 Additionally, the following parameters can be set via command line options
782 (or in code via @code{AVOption}s):
786 Name of application to connect on the RTMP server. This option
787 overrides the parameter specified in the URI.
790 Set the client buffer time in milliseconds. The default is 3000.
793 Extra arbitrary AMF connection parameters, parsed from a string,
794 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
795 Each value is prefixed by a single character denoting the type,
796 B for Boolean, N for number, S for string, O for object, or Z for null,
797 followed by a colon. For Booleans the data must be either 0 or 1 for
798 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
799 1 to end or begin an object, respectively. Data items in subobjects may
800 be named, by prefixing the type with 'N' and specifying the name before
801 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
802 times to construct arbitrary AMF sequences.
805 Version of the Flash plugin used to run the SWF player. The default
806 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
807 <libavformat version>).)
809 @item rtmp_flush_interval
810 Number of packets flushed in the same request (RTMPT only). The default
814 Specify that the media is a live stream. No resuming or seeking in
815 live streams is possible. The default value is @code{any}, which means the
816 subscriber first tries to play the live stream specified in the
817 playpath. If a live stream of that name is not found, it plays the
818 recorded stream. The other possible values are @code{live} and
822 URL of the web page in which the media was embedded. By default no
826 Stream identifier to play or to publish. This option overrides the
827 parameter specified in the URI.
830 Name of live stream to subscribe to. By default no value will be sent.
831 It is only sent if the option is specified or if rtmp_live
835 SHA256 hash of the decompressed SWF file (32 bytes).
838 Size of the decompressed SWF file, required for SWFVerification.
841 URL of the SWF player for the media. By default no value will be sent.
844 URL to player swf file, compute hash/size automatically.
847 URL of the target stream. Defaults to proto://host[:port]/app.
851 For example to read with @command{ffplay} a multimedia resource named
852 "sample" from the application "vod" from an RTMP server "myserver":
854 ffplay rtmp://myserver/vod/sample
857 To publish to a password protected server, passing the playpath and
858 app names separately:
860 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
865 Encrypted Real-Time Messaging Protocol.
867 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
868 streaming multimedia content within standard cryptographic primitives,
869 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
874 Real-Time Messaging Protocol over a secure SSL connection.
876 The Real-Time Messaging Protocol (RTMPS) is used for streaming
877 multimedia content across an encrypted connection.
881 Real-Time Messaging Protocol tunneled through HTTP.
883 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
884 for streaming multimedia content within HTTP requests to traverse
889 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
891 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
892 is used for streaming multimedia content within HTTP requests to traverse
897 Real-Time Messaging Protocol tunneled through HTTPS.
899 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
900 for streaming multimedia content within HTTPS requests to traverse
903 @section libsmbclient
905 libsmbclient permits one to manipulate CIFS/SMB network resources.
907 Following syntax is required.
910 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
913 This protocol accepts the following options.
917 Set timeout in milliseconds of socket I/O operations used by the underlying
918 low level operation. By default it is set to -1, which means that the timeout
922 Truncate existing files on write, if set to 1. A value of 0 prevents
923 truncating. Default value is 1.
926 Set the workgroup used for making connections. By default workgroup is not specified.
930 For more information see: @url{http://www.samba.org/}.
934 Secure File Transfer Protocol via libssh
936 Read from or write to remote resources using SFTP protocol.
938 Following syntax is required.
941 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
944 This protocol accepts the following options.
948 Set timeout of socket I/O operations used by the underlying low level
949 operation. By default it is set to -1, which means that the timeout
953 Truncate existing files on write, if set to 1. A value of 0 prevents
954 truncating. Default value is 1.
957 Specify the path of the file containing private key to use during authorization.
958 By default libssh searches for keys in the @file{~/.ssh/} directory.
962 Example: Play a file stored on remote server.
965 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
968 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
970 Real-Time Messaging Protocol and its variants supported through
973 Requires the presence of the librtmp headers and library during
974 configuration. You need to explicitly configure the build with
975 "--enable-librtmp". If enabled this will replace the native RTMP
978 This protocol provides most client functions and a few server
979 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
980 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
981 variants of these encrypted types (RTMPTE, RTMPTS).
983 The required syntax is:
985 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
988 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
989 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
990 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
991 meaning as specified for the RTMP native protocol.
992 @var{options} contains a list of space-separated options of the form
995 See the librtmp manual page (man 3 librtmp) for more information.
997 For example, to stream a file in real-time to an RTMP server using
1000 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
1003 To play the same stream using @command{ffplay}:
1005 ffplay "rtmp://myserver/live/mystream live=1"
1010 Real-time Transport Protocol.
1012 The required syntax for an RTP URL is:
1013 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
1015 @var{port} specifies the RTP port to use.
1017 The following URL options are supported:
1022 Set the TTL (Time-To-Live) value (for multicast only).
1024 @item rtcpport=@var{n}
1025 Set the remote RTCP port to @var{n}.
1027 @item localrtpport=@var{n}
1028 Set the local RTP port to @var{n}.
1030 @item localrtcpport=@var{n}'
1031 Set the local RTCP port to @var{n}.
1033 @item pkt_size=@var{n}
1034 Set max packet size (in bytes) to @var{n}.
1036 @item buffer_size=@var{size}
1037 Set the maximum UDP socket buffer size in bytes.
1040 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
1043 @item sources=@var{ip}[,@var{ip}]
1044 List allowed source IP addresses.
1046 @item block=@var{ip}[,@var{ip}]
1047 List disallowed (blocked) source IP addresses.
1049 @item write_to_source=0|1
1050 Send packets to the source address of the latest received packet (if
1051 set to 1) or to a default remote address (if set to 0).
1053 @item localport=@var{n}
1054 Set the local RTP port to @var{n}.
1056 @item timeout=@var{n}
1057 Set timeout (in microseconds) of socket I/O operations to @var{n}.
1059 This is a deprecated option. Instead, @option{localrtpport} should be
1069 If @option{rtcpport} is not set the RTCP port will be set to the RTP
1073 If @option{localrtpport} (the local RTP port) is not set any available
1074 port will be used for the local RTP and RTCP ports.
1077 If @option{localrtcpport} (the local RTCP port) is not set it will be
1078 set to the local RTP port value plus 1.
1083 Real-Time Streaming Protocol.
1085 RTSP is not technically a protocol handler in libavformat, it is a demuxer
1086 and muxer. The demuxer supports both normal RTSP (with data transferred
1087 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
1088 data transferred over RDT).
1090 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1091 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
1092 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
1094 The required syntax for a RTSP url is:
1096 rtsp://@var{hostname}[:@var{port}]/@var{path}
1099 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1100 line, or set in code via @code{AVOption}s or in
1101 @code{avformat_open_input}.
1103 The following options are supported.
1107 Do not start playing the stream immediately if set to 1. Default value
1110 @item rtsp_transport
1111 Set RTSP transport protocols.
1113 It accepts the following values:
1116 Use UDP as lower transport protocol.
1119 Use TCP (interleaving within the RTSP control channel) as lower
1123 Use UDP multicast as lower transport protocol.
1126 Use HTTP tunneling as lower transport protocol, which is useful for
1130 Multiple lower transport protocols may be specified, in that case they are
1131 tried one at a time (if the setup of one fails, the next one is tried).
1132 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
1137 The following values are accepted:
1140 Accept packets only from negotiated peer address and port.
1142 Act as a server, listening for an incoming connection.
1144 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
1147 Default value is @samp{none}.
1149 @item allowed_media_types
1150 Set media types to accept from the server.
1152 The following flags are accepted:
1159 By default it accepts all media types.
1162 Set minimum local UDP port. Default value is 5000.
1165 Set maximum local UDP port. Default value is 65000.
1168 This option is deprecated. Use @option{listen_timeout} instead. Set maximum timeout (in seconds) to wait for incoming connections.
1170 A value of -1 means infinite (default). This option implies the
1171 @option{rtsp_flags} set to @samp{listen}.
1173 @item listen_timeout
1174 Set maximum timeout (in seconds) to establish an initial connection. Setting
1175 @option{listen_timeout} > 0 sets @option{rtsp_flags} to @samp{listen}. Default is -1
1176 which means an infinite timeout when @samp{listen} mode is set.
1178 @item reorder_queue_size
1179 Set number of packets to buffer for handling of reordered packets.
1182 Set socket TCP I/O timeout in microseconds.
1185 This option is deprecated. Use @option{user_agent} instead. Override User-Agent header. If not specified, it defaults to the
1186 libavformat identifier string.
1189 Override User-Agent header. If not specified, it defaults to the
1190 libavformat identifier string.
1193 When receiving data over UDP, the demuxer tries to reorder received packets
1194 (since they may arrive out of order, or packets may get lost totally). This
1195 can be disabled by setting the maximum demuxing delay to zero (via
1196 the @code{max_delay} field of AVFormatContext).
1198 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1199 streams to display can be chosen with @code{-vst} @var{n} and
1200 @code{-ast} @var{n} for video and audio respectively, and can be switched
1201 on the fly by pressing @code{v} and @code{a}.
1203 @subsection Examples
1205 The following examples all make use of the @command{ffplay} and
1206 @command{ffmpeg} tools.
1210 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1212 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1216 Watch a stream tunneled over HTTP:
1218 ffplay -rtsp_transport http rtsp://server/video.mp4
1222 Send a stream in realtime to a RTSP server, for others to watch:
1224 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1228 Receive a stream in realtime:
1230 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1236 Session Announcement Protocol (RFC 2974). This is not technically a
1237 protocol handler in libavformat, it is a muxer and demuxer.
1238 It is used for signalling of RTP streams, by announcing the SDP for the
1239 streams regularly on a separate port.
1243 The syntax for a SAP url given to the muxer is:
1245 sap://@var{destination}[:@var{port}][?@var{options}]
1248 The RTP packets are sent to @var{destination} on port @var{port},
1249 or to port 5004 if no port is specified.
1250 @var{options} is a @code{&}-separated list. The following options
1255 @item announce_addr=@var{address}
1256 Specify the destination IP address for sending the announcements to.
1257 If omitted, the announcements are sent to the commonly used SAP
1258 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1259 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1261 @item announce_port=@var{port}
1262 Specify the port to send the announcements on, defaults to
1263 9875 if not specified.
1266 Specify the time to live value for the announcements and RTP packets,
1269 @item same_port=@var{0|1}
1270 If set to 1, send all RTP streams on the same port pair. If zero (the
1271 default), all streams are sent on unique ports, with each stream on a
1272 port 2 numbers higher than the previous.
1273 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1274 The RTP stack in libavformat for receiving requires all streams to be sent
1278 Example command lines follow.
1280 To broadcast a stream on the local subnet, for watching in VLC:
1283 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1286 Similarly, for watching in @command{ffplay}:
1289 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1292 And for watching in @command{ffplay}, over IPv6:
1295 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1300 The syntax for a SAP url given to the demuxer is:
1302 sap://[@var{address}][:@var{port}]
1305 @var{address} is the multicast address to listen for announcements on,
1306 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1307 is the port that is listened on, 9875 if omitted.
1309 The demuxers listens for announcements on the given address and port.
1310 Once an announcement is received, it tries to receive that particular stream.
1312 Example command lines follow.
1314 To play back the first stream announced on the normal SAP multicast address:
1320 To play back the first stream announced on one the default IPv6 SAP multicast address:
1323 ffplay sap://[ff0e::2:7ffe]
1328 Stream Control Transmission Protocol.
1330 The accepted URL syntax is:
1332 sctp://@var{host}:@var{port}[?@var{options}]
1335 The protocol accepts the following options:
1338 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1341 Set the maximum number of streams. By default no limit is set.
1346 Haivision Secure Reliable Transport Protocol via libsrt.
1348 The supported syntax for a SRT URL is:
1350 srt://@var{hostname}:@var{port}[?@var{options}]
1353 @var{options} contains a list of &-separated options of the form
1354 @var{key}=@var{val}.
1359 @var{options} srt://@var{hostname}:@var{port}
1362 @var{options} contains a list of '-@var{key} @var{val}'
1365 This protocol accepts the following options.
1368 @item connect_timeout=@var{milliseconds}
1369 Connection timeout; SRT cannot connect for RTT > 1500 msec
1370 (2 handshake exchanges) with the default connect timeout of
1371 3 seconds. This option applies to the caller and rendezvous
1372 connection modes. The connect timeout is 10 times the value
1373 set for the rendezvous mode (which can be used as a
1374 workaround for this connection problem with earlier versions).
1376 @item ffs=@var{bytes}
1377 Flight Flag Size (Window Size), in bytes. FFS is actually an
1378 internal parameter and you should set it to not less than
1379 @option{recv_buffer_size} and @option{mss}. The default value
1380 is relatively large, therefore unless you set a very large receiver buffer,
1381 you do not need to change this option. Default value is 25600.
1383 @item inputbw=@var{bytes/seconds}
1384 Sender nominal input rate, in bytes per seconds. Used along with
1385 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1386 calculate maximum sending rate when recovery packets are sent
1387 along with the main media stream:
1388 @option{inputbw} * (100 + @option{oheadbw}) / 100
1389 if @option{inputbw} is not set while @option{maxbw} is set to
1390 relative (0), the actual input rate is evaluated inside
1391 the library. Default value is 0.
1393 @item iptos=@var{tos}
1394 IP Type of Service. Applies to sender only. Default value is 0xB8.
1396 @item ipttl=@var{ttl}
1397 IP Time To Live. Applies to sender only. Default value is 64.
1399 @item latency=@var{microseconds}
1400 Timestamp-based Packet Delivery Delay.
1401 Used to absorb bursts of missed packet retransmissions.
1402 This flag sets both @option{rcvlatency} and @option{peerlatency}
1403 to the same value. Note that prior to version 1.3.0
1404 this is the only flag to set the latency, however
1405 this is effectively equivalent to setting @option{peerlatency},
1406 when side is sender and @option{rcvlatency}
1407 when side is receiver, and the bidirectional stream
1408 sending is not supported.
1410 @item listen_timeout=@var{microseconds}
1411 Set socket listen timeout.
1413 @item maxbw=@var{bytes/seconds}
1414 Maximum sending bandwidth, in bytes per seconds.
1415 -1 infinite (CSRTCC limit is 30mbps)
1416 0 relative to input rate (see @option{inputbw})
1417 >0 absolute limit value
1418 Default value is 0 (relative)
1420 @item mode=@var{caller|listener|rendezvous}
1422 @option{caller} opens client connection.
1423 @option{listener} starts server to listen for incoming connections.
1424 @option{rendezvous} use Rendez-Vous connection mode.
1425 Default value is caller.
1427 @item mss=@var{bytes}
1428 Maximum Segment Size, in bytes. Used for buffer allocation
1429 and rate calculation using a packet counter assuming fully
1430 filled packets. The smallest MSS between the peers is
1431 used. This is 1500 by default in the overall internet.
1432 This is the maximum size of the UDP packet and can be
1433 only decreased, unless you have some unusual dedicated
1434 network settings. Default value is 1500.
1436 @item nakreport=@var{1|0}
1437 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1438 periodically until a lost packet is retransmitted or
1439 intentionally dropped. Default value is 1.
1441 @item oheadbw=@var{percents}
1442 Recovery bandwidth overhead above input rate, in percents.
1443 See @option{inputbw}. Default value is 25%.
1445 @item passphrase=@var{string}
1446 HaiCrypt Encryption/Decryption Passphrase string, length
1447 from 10 to 79 characters. The passphrase is the shared
1448 secret between the sender and the receiver. It is used
1449 to generate the Key Encrypting Key using PBKDF2
1450 (Password-Based Key Derivation Function). It is used
1451 only if @option{pbkeylen} is non-zero. It is used on
1452 the receiver only if the received data is encrypted.
1453 The configured passphrase cannot be recovered (write-only).
1455 @item enforced_encryption=@var{1|0}
1456 If true, both connection parties must have the same password
1457 set (including empty, that is, with no encryption). If the
1458 password doesn't match or only one side is unencrypted,
1459 the connection is rejected. Default is true.
1461 @item kmrefreshrate=@var{packets}
1462 The number of packets to be transmitted after which the
1463 encryption key is switched to a new key. Default is -1.
1464 -1 means auto (0x1000000 in srt library). The range for
1465 this option is integers in the 0 - @code{INT_MAX}.
1467 @item kmpreannounce=@var{packets}
1468 The interval between when a new encryption key is sent and
1469 when switchover occurs. This value also applies to the
1470 subsequent interval between when switchover occurs and
1471 when the old encryption key is decommissioned. Default is -1.
1472 -1 means auto (0x1000 in srt library). The range for
1473 this option is integers in the 0 - @code{INT_MAX}.
1475 @item payload_size=@var{bytes}
1476 Sets the maximum declared size of a packet transferred
1477 during the single call to the sending function in Live
1478 mode. Use 0 if this value isn't used (which is default in
1480 Default is -1 (automatic), which typically means MPEG-TS;
1481 if you are going to use SRT
1482 to send any different kind of payload, such as, for example,
1483 wrapping a live stream in very small frames, then you can
1484 use a bigger maximum frame size, though not greater than
1487 @item pkt_size=@var{bytes}
1488 Alias for @samp{payload_size}.
1490 @item peerlatency=@var{microseconds}
1491 The latency value (as described in @option{rcvlatency}) that is
1492 set by the sender side as a minimum value for the receiver.
1494 @item pbkeylen=@var{bytes}
1495 Sender encryption key length, in bytes.
1496 Only can be set to 0, 16, 24 and 32.
1497 Enable sender encryption if not 0.
1498 Not required on receiver (set to 0),
1499 key size obtained from sender in HaiCrypt handshake.
1502 @item rcvlatency=@var{microseconds}
1503 The time that should elapse since the moment when the
1504 packet was sent and the moment when it's delivered to
1505 the receiver application in the receiving function.
1506 This time should be a buffer time large enough to cover
1507 the time spent for sending, unexpectedly extended RTT
1508 time, and the time needed to retransmit the lost UDP
1509 packet. The effective latency value will be the maximum
1510 of this options' value and the value of @option{peerlatency}
1511 set by the peer side. Before version 1.3.0 this option
1512 is only available as @option{latency}.
1514 @item recv_buffer_size=@var{bytes}
1515 Set UDP receive buffer size, expressed in bytes.
1517 @item send_buffer_size=@var{bytes}
1518 Set UDP send buffer size, expressed in bytes.
1520 @item timeout=@var{microseconds}
1521 Set raise error timeouts for read, write and connect operations. Note that the
1522 SRT library has internal timeouts which can be controlled separately, the
1523 value set here is only a cap on those.
1525 @item tlpktdrop=@var{1|0}
1526 Too-late Packet Drop. When enabled on receiver, it skips
1527 missing packets that have not been delivered in time and
1528 delivers the following packets to the application when
1529 their time-to-play has come. It also sends a fake ACK to
1530 the sender. When enabled on sender and enabled on the
1531 receiving peer, the sender drops the older packets that
1532 have no chance of being delivered in time. It was
1533 automatically enabled in the sender if the receiver
1536 @item sndbuf=@var{bytes}
1537 Set send buffer size, expressed in bytes.
1539 @item rcvbuf=@var{bytes}
1540 Set receive buffer size, expressed in bytes.
1542 Receive buffer must not be greater than @option{ffs}.
1544 @item lossmaxttl=@var{packets}
1545 The value up to which the Reorder Tolerance may grow. When
1546 Reorder Tolerance is > 0, then packet loss report is delayed
1547 until that number of packets come in. Reorder Tolerance
1548 increases every time a "belated" packet has come, but it
1549 wasn't due to retransmission (that is, when UDP packets tend
1550 to come out of order), with the difference between the latest
1551 sequence and this packet's sequence, and not more than the
1552 value of this option. By default it's 0, which means that this
1553 mechanism is turned off, and the loss report is always sent
1554 immediately upon experiencing a "gap" in sequences.
1557 The minimum SRT version that is required from the peer. A connection
1558 to a peer that does not satisfy the minimum version requirement
1561 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1564 @item streamid=@var{string}
1565 A string limited to 512 characters that can be set on the socket prior
1566 to connecting. This stream ID will be able to be retrieved by the
1567 listener side from the socket that is returned from srt_accept and
1568 was connected by a socket with that set stream ID. SRT does not enforce
1569 any special interpretation of the contents of this string.
1570 This option doesn’t make sense in Rendezvous connection; the result
1571 might be that simply one side will override the value from the other
1572 side and it’s the matter of luck which one would win
1574 @item smoother=@var{live|file}
1575 The type of Smoother used for the transmission for that socket, which
1576 is responsible for the transmission and congestion control. The Smoother
1577 type must be exactly the same on both connecting parties, otherwise
1578 the connection is rejected.
1580 @item messageapi=@var{1|0}
1581 When set, this socket uses the Message API, otherwise it uses Buffer
1582 API. Note that in live mode (see @option{transtype}) there’s only
1583 message API available. In File mode you can chose to use one of two modes:
1585 Stream API (default, when this option is false). In this mode you may
1586 send as many data as you wish with one sending instruction, or even use
1587 dedicated functions that read directly from a file. The internal facility
1588 will take care of any speed and congestion control. When receiving, you
1589 can also receive as many data as desired, the data not extracted will be
1590 waiting for the next call. There is no boundary between data portions in
1593 Message API. In this mode your single sending instruction passes exactly
1594 one piece of data that has boundaries (a message). Contrary to Live mode,
1595 this message may span across multiple UDP packets and the only size
1596 limitation is that it shall fit as a whole in the sending buffer. The
1597 receiver shall use as large buffer as necessary to receive the message,
1598 otherwise the message will not be given up. When the message is not
1599 complete (not all packets received or there was a packet loss) it will
1602 @item transtype=@var{live|file}
1603 Sets the transmission type for the socket, in particular, setting this
1604 option sets multiple other parameters to their default values as required
1605 for a particular transmission type.
1607 live: Set options as for live transmission. In this mode, you should
1608 send by one sending instruction only so many data that fit in one UDP packet,
1609 and limited to the value defined first in @option{payload_size} (1316 is
1610 default in this mode). There is no speed control in this mode, only the
1611 bandwidth control, if configured, in order to not exceed the bandwidth with
1612 the overhead transmission (retransmitted and control packets).
1614 file: Set options as for non-live transmission. See @option{messageapi}
1615 for further explanations
1617 @item linger=@var{seconds}
1618 The number of seconds that the socket waits for unsent data when closing.
1619 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
1620 seconds in file mode). The range for this option is integers in the
1625 For more information see: @url{https://github.com/Haivision/srt}.
1629 Secure Real-time Transport Protocol.
1631 The accepted options are:
1634 @item srtp_out_suite
1635 Select input and output encoding suites.
1639 @item AES_CM_128_HMAC_SHA1_80
1640 @item SRTP_AES128_CM_HMAC_SHA1_80
1641 @item AES_CM_128_HMAC_SHA1_32
1642 @item SRTP_AES128_CM_HMAC_SHA1_32
1645 @item srtp_in_params
1646 @item srtp_out_params
1647 Set input and output encoding parameters, which are expressed by a
1648 base64-encoded representation of a binary block. The first 16 bytes of
1649 this binary block are used as master key, the following 14 bytes are
1650 used as master salt.
1655 Virtually extract a segment of a file or another stream.
1656 The underlying stream must be seekable.
1661 Start offset of the extracted segment, in bytes.
1663 End offset of the extracted segment, in bytes.
1664 If set to 0, extract till end of file.
1669 Extract a chapter from a DVD VOB file (start and end sectors obtained
1670 externally and multiplied by 2048):
1672 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1675 Play an AVI file directly from a TAR archive:
1677 subfile,,start,183241728,end,366490624,,:archive.tar
1680 Play a MPEG-TS file from start offset till end:
1682 subfile,,start,32815239,end,0,,:video.ts
1687 Writes the output to multiple protocols. The individual outputs are separated
1691 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1696 Transmission Control Protocol.
1698 The required syntax for a TCP url is:
1700 tcp://@var{hostname}:@var{port}[?@var{options}]
1703 @var{options} contains a list of &-separated options of the form
1704 @var{key}=@var{val}.
1706 The list of supported options follows.
1709 @item listen=@var{2|1|0}
1710 Listen for an incoming connection. 0 disables listen, 1 enables listen in
1711 single client mode, 2 enables listen in multi-client mode. Default value is 0.
1713 @item timeout=@var{microseconds}
1714 Set raise error timeout, expressed in microseconds.
1716 This option is only relevant in read mode: if no data arrived in more
1717 than this time interval, raise error.
1719 @item listen_timeout=@var{milliseconds}
1720 Set listen timeout, expressed in milliseconds.
1722 @item recv_buffer_size=@var{bytes}
1723 Set receive buffer size, expressed bytes.
1725 @item send_buffer_size=@var{bytes}
1726 Set send buffer size, expressed bytes.
1728 @item tcp_nodelay=@var{1|0}
1729 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1731 @item tcp_mss=@var{bytes}
1732 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1735 The following example shows how to setup a listening TCP connection
1736 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1738 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1739 ffplay tcp://@var{hostname}:@var{port}
1744 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1746 The required syntax for a TLS/SSL url is:
1748 tls://@var{hostname}:@var{port}[?@var{options}]
1751 The following parameters can be set via command line options
1752 (or in code via @code{AVOption}s):
1756 @item ca_file, cafile=@var{filename}
1757 A file containing certificate authority (CA) root certificates to treat
1758 as trusted. If the linked TLS library contains a default this might not
1759 need to be specified for verification to work, but not all libraries and
1760 setups have defaults built in.
1761 The file must be in OpenSSL PEM format.
1763 @item tls_verify=@var{1|0}
1764 If enabled, try to verify the peer that we are communicating with.
1765 Note, if using OpenSSL, this currently only makes sure that the
1766 peer certificate is signed by one of the root certificates in the CA
1767 database, but it does not validate that the certificate actually
1768 matches the host name we are trying to connect to. (With other backends,
1769 the host name is validated as well.)
1771 This is disabled by default since it requires a CA database to be
1772 provided by the caller in many cases.
1774 @item cert_file, cert=@var{filename}
1775 A file containing a certificate to use in the handshake with the peer.
1776 (When operating as server, in listen mode, this is more often required
1777 by the peer, while client certificates only are mandated in certain
1780 @item key_file, key=@var{filename}
1781 A file containing the private key for the certificate.
1783 @item listen=@var{1|0}
1784 If enabled, listen for connections on the provided port, and assume
1785 the server role in the handshake instead of the client role.
1788 The HTTP proxy to tunnel through, e.g. @code{http://example.com:1234}.
1789 The proxy must support the CONNECT method.
1793 Example command lines:
1795 To create a TLS/SSL server that serves an input stream.
1798 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1801 To play back a stream from the TLS/SSL server using @command{ffplay}:
1804 ffplay tls://@var{hostname}:@var{port}
1809 User Datagram Protocol.
1811 The required syntax for an UDP URL is:
1813 udp://@var{hostname}:@var{port}[?@var{options}]
1816 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1818 In case threading is enabled on the system, a circular buffer is used
1819 to store the incoming data, which allows one to reduce loss of data due to
1820 UDP socket buffer overruns. The @var{fifo_size} and
1821 @var{overrun_nonfatal} options are related to this buffer.
1823 The list of supported options follows.
1826 @item buffer_size=@var{size}
1827 Set the UDP maximum socket buffer size in bytes. This is used to set either
1828 the receive or send buffer size, depending on what the socket is used for.
1829 Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
1831 @item bitrate=@var{bitrate}
1832 If set to nonzero, the output will have the specified constant bitrate if the
1833 input has enough packets to sustain it.
1835 @item burst_bits=@var{bits}
1836 When using @var{bitrate} this specifies the maximum number of bits in
1839 @item localport=@var{port}
1840 Override the local UDP port to bind with.
1842 @item localaddr=@var{addr}
1843 Local IP address of a network interface used for sending packets or joining
1846 @item pkt_size=@var{size}
1847 Set the size in bytes of UDP packets.
1849 @item reuse=@var{1|0}
1850 Explicitly allow or disallow reusing UDP sockets.
1853 Set the time to live value (for multicast only).
1855 @item connect=@var{1|0}
1856 Initialize the UDP socket with @code{connect()}. In this case, the
1857 destination address can't be changed with ff_udp_set_remote_url later.
1858 If the destination address isn't known at the start, this option can
1859 be specified in ff_udp_set_remote_url, too.
1860 This allows finding out the source address for the packets with getsockname,
1861 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1862 unreachable" is received.
1863 For receiving, this gives the benefit of only receiving packets from
1864 the specified peer address/port.
1866 @item sources=@var{address}[,@var{address}]
1867 Only receive packets sent from the specified addresses. In case of multicast,
1868 also subscribe to multicast traffic coming from these addresses only.
1870 @item block=@var{address}[,@var{address}]
1871 Ignore packets sent from the specified addresses. In case of multicast, also
1872 exclude the source addresses in the multicast subscription.
1874 @item fifo_size=@var{units}
1875 Set the UDP receiving circular buffer size, expressed as a number of
1876 packets with size of 188 bytes. If not specified defaults to 7*4096.
1878 @item overrun_nonfatal=@var{1|0}
1879 Survive in case of UDP receiving circular buffer overrun. Default
1882 @item timeout=@var{microseconds}
1883 Set raise error timeout, expressed in microseconds.
1885 This option is only relevant in read mode: if no data arrived in more
1886 than this time interval, raise error.
1888 @item broadcast=@var{1|0}
1889 Explicitly allow or disallow UDP broadcasting.
1891 Note that broadcasting may not work properly on networks having
1892 a broadcast storm protection.
1895 @subsection Examples
1899 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1901 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1905 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1906 sized UDP packets, using a large input buffer:
1908 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1912 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1914 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1922 The required syntax for a Unix socket URL is:
1925 unix://@var{filepath}
1928 The following parameters can be set via command line options
1929 (or in code via @code{AVOption}s):
1935 Create the Unix socket in listening mode.
1940 ZeroMQ asynchronous messaging using the libzmq library.
1942 This library supports unicast streaming to multiple clients without relying on
1945 The required syntax for streaming or connecting to a stream is:
1947 zmq:tcp://ip-address:port
1951 Create a localhost stream on port 5555:
1953 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1956 Multiple clients may connect to the stream using:
1958 ffplay zmq:tcp://127.0.0.1:5555
1961 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
1962 The server side binds to a port and publishes data. Clients connect to the
1963 server (via IP address/port) and subscribe to the stream. The order in which
1964 the server and client start generally does not matter.
1966 ffmpeg must be compiled with the --enable-libzmq option to support
1969 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1970 line. The following options are supported:
1975 Forces the maximum packet size for sending/receiving data. The default value is
1976 131,072 bytes. On the server side, this sets the maximum size of sent packets
1977 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
1978 packets. Note that pkt_size on the clients should be equal to or greater than
1979 pkt_size on the server. Otherwise the received message may be truncated causing
1985 @c man end PROTOCOLS