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 Asynchronous data filling wrapper for input stream.
58 Fill data in a background thread, to decouple I/O operation from demux thread.
62 async:http://host/resource
63 async:cache:http://host/resource
70 The accepted options are:
80 Playlist to read (BDMV/PLAYLIST/?????.mpls)
86 Read longest playlist from BluRay mounted to /mnt/bluray:
91 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
93 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
98 Caching wrapper for input stream.
100 Cache the input stream to temporary file. It brings seeking capability to live streams.
108 Physical concatenation protocol.
110 Read and seek from many resources in sequence as if they were
113 A URL accepted by this protocol has the syntax:
115 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
118 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
119 resource to be concatenated, each one possibly specifying a distinct
122 For example to read a sequence of files @file{split1.mpeg},
123 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
126 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
129 Note that you may need to escape the character "|" which is special for
134 AES-encrypted stream reading protocol.
136 The accepted options are:
139 Set the AES decryption key binary block from given hexadecimal representation.
142 Set the AES decryption initialization vector binary block from given hexadecimal representation.
145 Accepted URL formats:
153 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
155 For example, to convert a GIF file given inline with @command{ffmpeg}:
157 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
162 File access protocol.
164 Read from or write to a file.
166 A file URL can have the form:
171 where @var{filename} is the path of the file to read.
173 An URL that does not have a protocol prefix will be assumed to be a
174 file URL. Depending on the build, an URL that looks like a Windows
175 path with the drive letter at the beginning will also be assumed to be
176 a file URL (usually not the case in builds for unix-like systems).
178 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
181 ffmpeg -i file:input.mpeg output.mpeg
184 This protocol accepts the following options:
188 Truncate existing files on write, if set to 1. A value of 0 prevents
189 truncating. Default value is 1.
192 Set I/O operation maximum block size, in bytes. Default value is
193 @code{INT_MAX}, which results in not limiting the requested block size.
194 Setting this value reasonably low improves user termination request reaction
195 time, which is valuable for files on slow medium.
198 If set to 1, the protocol will retry reading at the end of the file, allowing
199 reading files that still are being written. In order for this to terminate,
200 you either need to use the rw_timeout option, or use the interrupt callback
206 FTP (File Transfer Protocol).
208 Read from or write to remote resources using FTP protocol.
210 Following syntax is required.
212 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
215 This protocol accepts the following options.
219 Set timeout in microseconds of socket I/O operations used by the underlying low level
220 operation. By default it is set to -1, which means that the timeout is
223 @item ftp-anonymous-password
224 Password used when login as anonymous user. Typically an e-mail address
227 @item ftp-write-seekable
228 Control seekability of connection during encoding. If set to 1 the
229 resource is supposed to be seekable, if set to 0 it is assumed not
230 to be seekable. Default value is 0.
233 NOTE: Protocol can be used as output, but it is recommended to not do
234 it, unless special care is taken (tests, customized server configuration
235 etc.). Different FTP servers behave in different way during seek
236 operation. ff* tools may produce incomplete content due to server limitations.
244 Read Apple HTTP Live Streaming compliant segmented stream as
245 a uniform one. The M3U8 playlists describing the segments can be
246 remote HTTP resources or local files, accessed using the standard
248 The nested protocol is declared by specifying
249 "+@var{proto}" after the hls URI scheme name, where @var{proto}
250 is either "file" or "http".
253 hls+http://host/path/to/remote/resource.m3u8
254 hls+file://path/to/local/resource.m3u8
257 Using this protocol is discouraged - the hls demuxer should work
258 just as well (if not, please report the issues) and is more complete.
259 To use the hls demuxer instead, simply use the direct URLs to the
264 HTTP (Hyper Text Transfer Protocol).
266 This protocol accepts the following options:
270 Control seekability of connection. If set to 1 the resource is
271 supposed to be seekable, if set to 0 it is assumed not to be seekable,
272 if set to -1 it will try to autodetect if it is seekable. Default
276 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
279 Set a specific content type for the POST messages or for listen mode.
282 set HTTP proxy to tunnel through e.g. http://example.com:1234
285 Set custom HTTP headers, can override built in default headers. The
286 value must be a string encoding the headers.
288 @item multiple_requests
289 Use persistent connections if set to 1, default is 0.
292 Set custom HTTP post data.
295 Set the Referer header. Include 'Referer: URL' header in HTTP request.
298 Override the User-Agent header. If not specified the protocol will use a
299 string describing the libavformat build. ("Lavf/<version>")
302 This is a deprecated option, you can use user_agent instead it.
305 Set timeout in microseconds of socket I/O operations used by the underlying low level
306 operation. By default it is set to -1, which means that the timeout is
309 @item reconnect_at_eof
310 If set then eof is treated like an error and causes reconnection, this is useful
311 for live / endless streams.
313 @item reconnect_streamed
314 If set then even streamed/non seekable streams will be reconnected on errors.
316 @item reconnect_delay_max
317 Sets the maximum delay in seconds after which to give up reconnecting
320 Export the MIME type.
323 Exports the HTTP response version number. Usually "1.0" or "1.1".
326 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
327 supports this, the metadata has to be retrieved by the application by reading
328 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
331 @item icy_metadata_headers
332 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
333 headers, separated by newline characters.
335 @item icy_metadata_packet
336 If the server supports ICY metadata, and @option{icy} was set to 1, this
337 contains the last non-empty metadata packet sent by the server. It should be
338 polled in regular intervals by applications interested in mid-stream metadata
342 Set the cookies to be sent in future requests. The format of each cookie is the
343 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
344 delimited by a newline character.
347 Set initial byte offset.
350 Try to limit the request to bytes preceding this offset.
353 When used as a client option it sets the HTTP method for the request.
355 When used as a server option it sets the HTTP method that is going to be
356 expected from the client(s).
357 If the expected and the received HTTP method do not match the client will
358 be given a Bad Request response.
359 When unset the HTTP method is not checked for now. This will be replaced by
360 autodetection in the future.
363 If set to 1 enables experimental HTTP server. This can be used to send data when
364 used as an output option, or read data from a client with HTTP POST when used as
366 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
367 in ffmpeg.c and thus must not be used as a command line option.
369 # Server side (sending):
370 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
372 # Client side (receiving):
373 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
375 # Client can also be done with wget:
376 wget http://@var{server}:@var{port} -O somefile.ogg
378 # Server side (receiving):
379 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
381 # Client side (sending):
382 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
384 # Client can also be done with wget:
385 wget --post-file=somefile.ogg http://@var{server}:@var{port}
390 @subsection HTTP Cookies
392 Some HTTP requests will be denied unless cookie values are passed in with the
393 request. The @option{cookies} option allows these cookies to be specified. At
394 the very least, each cookie must specify a value along with a path and domain.
395 HTTP requests that match both the domain and path will automatically include the
396 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
399 The required syntax to play a stream specifying a cookie is:
401 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
406 Icecast protocol (stream to Icecast servers)
408 This protocol accepts the following options:
412 Set the stream genre.
417 @item ice_description
418 Set the stream description.
421 Set the stream website URL.
424 Set if the stream should be public.
425 The default is 0 (not public).
428 Override the User-Agent header. If not specified a string of the form
429 "Lavf/<version>" will be used.
432 Set the Icecast mountpoint password.
435 Set the stream content type. This must be set if it is different from
439 This enables support for Icecast versions < 2.4.0, that do not support the
440 HTTP PUT method but the SOURCE method.
445 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
450 MMS (Microsoft Media Server) protocol over TCP.
454 MMS (Microsoft Media Server) protocol over HTTP.
456 The required syntax is:
458 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
465 Computes the MD5 hash of the data to be written, and on close writes
466 this to the designated output or stdout if none is specified. It can
467 be used to test muxers without writing an actual file.
469 Some examples follow.
471 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
472 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
474 # Write the MD5 hash of the encoded AVI file to stdout.
475 ffmpeg -i input.flv -f avi -y md5:
478 Note that some formats (typically MOV) require the output protocol to
479 be seekable, so they will fail with the MD5 output protocol.
483 UNIX pipe access protocol.
485 Read and write from UNIX pipes.
487 The accepted syntax is:
492 @var{number} is the number corresponding to the file descriptor of the
493 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
494 is not specified, by default the stdout file descriptor will be used
495 for writing, stdin for reading.
497 For example to read from stdin with @command{ffmpeg}:
499 cat test.wav | ffmpeg -i pipe:0
500 # ...this is the same as...
501 cat test.wav | ffmpeg -i pipe:
504 For writing to stdout with @command{ffmpeg}:
506 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
507 # ...this is the same as...
508 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
511 This protocol accepts the following options:
515 Set I/O operation maximum block size, in bytes. Default value is
516 @code{INT_MAX}, which results in not limiting the requested block size.
517 Setting this value reasonably low improves user termination request reaction
518 time, which is valuable if data transmission is slow.
521 Note that some formats (typically MOV), require the output protocol to
522 be seekable, so they will fail with the pipe output protocol.
526 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
528 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
529 for MPEG-2 Transport Streams sent over RTP.
531 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
532 the @code{rtp} protocol.
534 The required syntax is:
536 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
539 The destination UDP ports are @code{port + 2} for the column FEC stream
540 and @code{port + 4} for the row FEC stream.
542 This protocol accepts the following options:
546 The number of columns (4-20, LxD <= 100)
549 The number of rows (4-20, LxD <= 100)
556 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
561 Real-Time Messaging Protocol.
563 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
564 content across a TCP/IP network.
566 The required syntax is:
568 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
571 The accepted parameters are:
575 An optional username (mostly for publishing).
578 An optional password (mostly for publishing).
581 The address of the RTMP server.
584 The number of the TCP port to use (by default is 1935).
587 It is the name of the application to access. It usually corresponds to
588 the path where the application is installed on the RTMP server
589 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
590 the value parsed from the URI through the @code{rtmp_app} option, too.
593 It is the path or name of the resource to play with reference to the
594 application specified in @var{app}, may be prefixed by "mp4:". You
595 can override the value parsed from the URI through the @code{rtmp_playpath}
599 Act as a server, listening for an incoming connection.
602 Maximum time to wait for the incoming connection. Implies listen.
605 Additionally, the following parameters can be set via command line options
606 (or in code via @code{AVOption}s):
610 Name of application to connect on the RTMP server. This option
611 overrides the parameter specified in the URI.
614 Set the client buffer time in milliseconds. The default is 3000.
617 Extra arbitrary AMF connection parameters, parsed from a string,
618 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
619 Each value is prefixed by a single character denoting the type,
620 B for Boolean, N for number, S for string, O for object, or Z for null,
621 followed by a colon. For Booleans the data must be either 0 or 1 for
622 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
623 1 to end or begin an object, respectively. Data items in subobjects may
624 be named, by prefixing the type with 'N' and specifying the name before
625 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
626 times to construct arbitrary AMF sequences.
629 Version of the Flash plugin used to run the SWF player. The default
630 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
631 <libavformat version>).)
633 @item rtmp_flush_interval
634 Number of packets flushed in the same request (RTMPT only). The default
638 Specify that the media is a live stream. No resuming or seeking in
639 live streams is possible. The default value is @code{any}, which means the
640 subscriber first tries to play the live stream specified in the
641 playpath. If a live stream of that name is not found, it plays the
642 recorded stream. The other possible values are @code{live} and
646 URL of the web page in which the media was embedded. By default no
650 Stream identifier to play or to publish. This option overrides the
651 parameter specified in the URI.
654 Name of live stream to subscribe to. By default no value will be sent.
655 It is only sent if the option is specified or if rtmp_live
659 SHA256 hash of the decompressed SWF file (32 bytes).
662 Size of the decompressed SWF file, required for SWFVerification.
665 URL of the SWF player for the media. By default no value will be sent.
668 URL to player swf file, compute hash/size automatically.
671 URL of the target stream. Defaults to proto://host[:port]/app.
675 For example to read with @command{ffplay} a multimedia resource named
676 "sample" from the application "vod" from an RTMP server "myserver":
678 ffplay rtmp://myserver/vod/sample
681 To publish to a password protected server, passing the playpath and
682 app names separately:
684 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
689 Encrypted Real-Time Messaging Protocol.
691 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
692 streaming multimedia content within standard cryptographic primitives,
693 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
698 Real-Time Messaging Protocol over a secure SSL connection.
700 The Real-Time Messaging Protocol (RTMPS) is used for streaming
701 multimedia content across an encrypted connection.
705 Real-Time Messaging Protocol tunneled through HTTP.
707 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
708 for streaming multimedia content within HTTP requests to traverse
713 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
715 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
716 is used for streaming multimedia content within HTTP requests to traverse
721 Real-Time Messaging Protocol tunneled through HTTPS.
723 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
724 for streaming multimedia content within HTTPS requests to traverse
727 @section libsmbclient
729 libsmbclient permits one to manipulate CIFS/SMB network resources.
731 Following syntax is required.
734 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
737 This protocol accepts the following options.
741 Set timeout in milliseconds of socket I/O operations used by the underlying
742 low level operation. By default it is set to -1, which means that the timeout
746 Truncate existing files on write, if set to 1. A value of 0 prevents
747 truncating. Default value is 1.
750 Set the workgroup used for making connections. By default workgroup is not specified.
754 For more information see: @url{http://www.samba.org/}.
758 Secure File Transfer Protocol via libssh
760 Read from or write to remote resources using SFTP protocol.
762 Following syntax is required.
765 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
768 This protocol accepts the following options.
772 Set timeout of socket I/O operations used by the underlying low level
773 operation. By default it is set to -1, which means that the timeout
777 Truncate existing files on write, if set to 1. A value of 0 prevents
778 truncating. Default value is 1.
781 Specify the path of the file containing private key to use during authorization.
782 By default libssh searches for keys in the @file{~/.ssh/} directory.
786 Example: Play a file stored on remote server.
789 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
792 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
794 Real-Time Messaging Protocol and its variants supported through
797 Requires the presence of the librtmp headers and library during
798 configuration. You need to explicitly configure the build with
799 "--enable-librtmp". If enabled this will replace the native RTMP
802 This protocol provides most client functions and a few server
803 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
804 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
805 variants of these encrypted types (RTMPTE, RTMPTS).
807 The required syntax is:
809 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
812 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
813 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
814 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
815 meaning as specified for the RTMP native protocol.
816 @var{options} contains a list of space-separated options of the form
819 See the librtmp manual page (man 3 librtmp) for more information.
821 For example, to stream a file in real-time to an RTMP server using
824 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
827 To play the same stream using @command{ffplay}:
829 ffplay "rtmp://myserver/live/mystream live=1"
834 Real-time Transport Protocol.
836 The required syntax for an RTP URL is:
837 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
839 @var{port} specifies the RTP port to use.
841 The following URL options are supported:
846 Set the TTL (Time-To-Live) value (for multicast only).
848 @item rtcpport=@var{n}
849 Set the remote RTCP port to @var{n}.
851 @item localrtpport=@var{n}
852 Set the local RTP port to @var{n}.
854 @item localrtcpport=@var{n}'
855 Set the local RTCP port to @var{n}.
857 @item pkt_size=@var{n}
858 Set max packet size (in bytes) to @var{n}.
861 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
864 @item sources=@var{ip}[,@var{ip}]
865 List allowed source IP addresses.
867 @item block=@var{ip}[,@var{ip}]
868 List disallowed (blocked) source IP addresses.
870 @item write_to_source=0|1
871 Send packets to the source address of the latest received packet (if
872 set to 1) or to a default remote address (if set to 0).
874 @item localport=@var{n}
875 Set the local RTP port to @var{n}.
877 This is a deprecated option. Instead, @option{localrtpport} should be
887 If @option{rtcpport} is not set the RTCP port will be set to the RTP
891 If @option{localrtpport} (the local RTP port) is not set any available
892 port will be used for the local RTP and RTCP ports.
895 If @option{localrtcpport} (the local RTCP port) is not set it will be
896 set to the local RTP port value plus 1.
901 Real-Time Streaming Protocol.
903 RTSP is not technically a protocol handler in libavformat, it is a demuxer
904 and muxer. The demuxer supports both normal RTSP (with data transferred
905 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
906 data transferred over RDT).
908 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
909 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
910 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
912 The required syntax for a RTSP url is:
914 rtsp://@var{hostname}[:@var{port}]/@var{path}
917 Options can be set on the @command{ffmpeg}/@command{ffplay} command
918 line, or set in code via @code{AVOption}s or in
919 @code{avformat_open_input}.
921 The following options are supported.
925 Do not start playing the stream immediately if set to 1. Default value
929 Set RTSP transport protocols.
931 It accepts the following values:
934 Use UDP as lower transport protocol.
937 Use TCP (interleaving within the RTSP control channel) as lower
941 Use UDP multicast as lower transport protocol.
944 Use HTTP tunneling as lower transport protocol, which is useful for
948 Multiple lower transport protocols may be specified, in that case they are
949 tried one at a time (if the setup of one fails, the next one is tried).
950 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
955 The following values are accepted:
958 Accept packets only from negotiated peer address and port.
960 Act as a server, listening for an incoming connection.
962 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
965 Default value is @samp{none}.
967 @item allowed_media_types
968 Set media types to accept from the server.
970 The following flags are accepted:
977 By default it accepts all media types.
980 Set minimum local UDP port. Default value is 5000.
983 Set maximum local UDP port. Default value is 65000.
986 Set maximum timeout (in seconds) to wait for incoming connections.
988 A value of -1 means infinite (default). This option implies the
989 @option{rtsp_flags} set to @samp{listen}.
991 @item reorder_queue_size
992 Set number of packets to buffer for handling of reordered packets.
995 Set socket TCP I/O timeout in microseconds.
998 Override User-Agent header. If not specified, it defaults to the
999 libavformat identifier string.
1002 When receiving data over UDP, the demuxer tries to reorder received packets
1003 (since they may arrive out of order, or packets may get lost totally). This
1004 can be disabled by setting the maximum demuxing delay to zero (via
1005 the @code{max_delay} field of AVFormatContext).
1007 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1008 streams to display can be chosen with @code{-vst} @var{n} and
1009 @code{-ast} @var{n} for video and audio respectively, and can be switched
1010 on the fly by pressing @code{v} and @code{a}.
1012 @subsection Examples
1014 The following examples all make use of the @command{ffplay} and
1015 @command{ffmpeg} tools.
1019 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1021 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1025 Watch a stream tunneled over HTTP:
1027 ffplay -rtsp_transport http rtsp://server/video.mp4
1031 Send a stream in realtime to a RTSP server, for others to watch:
1033 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1037 Receive a stream in realtime:
1039 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1045 Session Announcement Protocol (RFC 2974). This is not technically a
1046 protocol handler in libavformat, it is a muxer and demuxer.
1047 It is used for signalling of RTP streams, by announcing the SDP for the
1048 streams regularly on a separate port.
1052 The syntax for a SAP url given to the muxer is:
1054 sap://@var{destination}[:@var{port}][?@var{options}]
1057 The RTP packets are sent to @var{destination} on port @var{port},
1058 or to port 5004 if no port is specified.
1059 @var{options} is a @code{&}-separated list. The following options
1064 @item announce_addr=@var{address}
1065 Specify the destination IP address for sending the announcements to.
1066 If omitted, the announcements are sent to the commonly used SAP
1067 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1068 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1070 @item announce_port=@var{port}
1071 Specify the port to send the announcements on, defaults to
1072 9875 if not specified.
1075 Specify the time to live value for the announcements and RTP packets,
1078 @item same_port=@var{0|1}
1079 If set to 1, send all RTP streams on the same port pair. If zero (the
1080 default), all streams are sent on unique ports, with each stream on a
1081 port 2 numbers higher than the previous.
1082 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1083 The RTP stack in libavformat for receiving requires all streams to be sent
1087 Example command lines follow.
1089 To broadcast a stream on the local subnet, for watching in VLC:
1092 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1095 Similarly, for watching in @command{ffplay}:
1098 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1101 And for watching in @command{ffplay}, over IPv6:
1104 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1109 The syntax for a SAP url given to the demuxer is:
1111 sap://[@var{address}][:@var{port}]
1114 @var{address} is the multicast address to listen for announcements on,
1115 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1116 is the port that is listened on, 9875 if omitted.
1118 The demuxers listens for announcements on the given address and port.
1119 Once an announcement is received, it tries to receive that particular stream.
1121 Example command lines follow.
1123 To play back the first stream announced on the normal SAP multicast address:
1129 To play back the first stream announced on one the default IPv6 SAP multicast address:
1132 ffplay sap://[ff0e::2:7ffe]
1137 Stream Control Transmission Protocol.
1139 The accepted URL syntax is:
1141 sctp://@var{host}:@var{port}[?@var{options}]
1144 The protocol accepts the following options:
1147 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1150 Set the maximum number of streams. By default no limit is set.
1155 Haivision Secure Reliable Transport Protocol via libsrt.
1157 The supported syntax for a SRT URL is:
1159 srt://@var{hostname}:@var{port}[?@var{options}]
1162 @var{options} contains a list of &-separated options of the form
1163 @var{key}=@var{val}.
1168 @var{options} srt://@var{hostname}:@var{port}
1171 @var{options} contains a list of '-@var{key} @var{val}'
1174 This protocol accepts the following options.
1177 @item connect_timeout
1178 Connection timeout; SRT cannot connect for RTT > 1500 msec
1179 (2 handshake exchanges) with the default connect timeout of
1180 3 seconds. This option applies to the caller and rendezvous
1181 connection modes. The connect timeout is 10 times the value
1182 set for the rendezvous mode (which can be used as a
1183 workaround for this connection problem with earlier versions).
1185 @item ffs=@var{bytes}
1186 Flight Flag Size (Window Size), in bytes. FFS is actually an
1187 internal parameter and you should set it to not less than
1188 @option{recv_buffer_size} and @option{mss}. The default value
1189 is relatively large, therefore unless you set a very large receiver buffer,
1190 you do not need to change this option. Default value is 25600.
1192 @item inputbw=@var{bytes/seconds}
1193 Sender nominal input rate, in bytes per seconds. Used along with
1194 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1195 calculate maximum sending rate when recovery packets are sent
1196 along with the main media stream:
1197 @option{inputbw} * (100 + @option{oheadbw}) / 100
1198 if @option{inputbw} is not set while @option{maxbw} is set to
1199 relative (0), the actual input rate is evaluated inside
1200 the library. Default value is 0.
1202 @item iptos=@var{tos}
1203 IP Type of Service. Applies to sender only. Default value is 0xB8.
1205 @item ipttl=@var{ttl}
1206 IP Time To Live. Applies to sender only. Default value is 64.
1209 Timestamp-based Packet Delivery Delay.
1210 Used to absorb bursts of missed packet retransmissions.
1211 This flag sets both @option{rcvlatency} and @option{peerlatency}
1212 to the same value. Note that prior to version 1.3.0
1213 this is the only flag to set the latency, however
1214 this is effectively equivalent to setting @option{peerlatency},
1215 when side is sender and @option{rcvlatency}
1216 when side is receiver, and the bidirectional stream
1217 sending is not supported.
1219 @item listen_timeout
1220 Set socket listen timeout.
1222 @item maxbw=@var{bytes/seconds}
1223 Maximum sending bandwidth, in bytes per seconds.
1224 -1 infinite (CSRTCC limit is 30mbps)
1225 0 relative to input rate (see @option{inputbw})
1226 >0 absolute limit value
1227 Default value is 0 (relative)
1229 @item mode=@var{caller|listener|rendezvous}
1231 @option{caller} opens client connection.
1232 @option{listener} starts server to listen for incoming connections.
1233 @option{rendezvous} use Rendez-Vous connection mode.
1234 Default value is caller.
1236 @item mss=@var{bytes}
1237 Maximum Segment Size, in bytes. Used for buffer allocation
1238 and rate calculation using a packet counter assuming fully
1239 filled packets. The smallest MSS between the peers is
1240 used. This is 1500 by default in the overall internet.
1241 This is the maximum size of the UDP packet and can be
1242 only decreased, unless you have some unusual dedicated
1243 network settings. Default value is 1500.
1245 @item nakreport=@var{1|0}
1246 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1247 periodically until a lost packet is retransmitted or
1248 intentionally dropped. Default value is 1.
1250 @item oheadbw=@var{percents}
1251 Recovery bandwidth overhead above input rate, in percents.
1252 See @option{inputbw}. Default value is 25%.
1254 @item passphrase=@var{string}
1255 HaiCrypt Encryption/Decryption Passphrase string, length
1256 from 10 to 79 characters. The passphrase is the shared
1257 secret between the sender and the receiver. It is used
1258 to generate the Key Encrypting Key using PBKDF2
1259 (Password-Based Key Derivation Function). It is used
1260 only if @option{pbkeylen} is non-zero. It is used on
1261 the receiver only if the received data is encrypted.
1262 The configured passphrase cannot be recovered (write-only).
1264 @item payload_size=@var{bytes}
1265 Sets the maximum declared size of a packet transferred
1266 during the single call to the sending function in Live
1267 mode. Use 0 if this value isn't used (which is default in
1269 Default is -1 (automatic), which typically means MPEG-TS;
1270 if you are going to use SRT
1271 to send any different kind of payload, such as, for example,
1272 wrapping a live stream in very small frames, then you can
1273 use a bigger maximum frame size, though not greater than
1276 @item pkt_size=@var{bytes}
1277 Alias for @samp{payload_size}.
1280 The latency value (as described in @option{rcvlatency}) that is
1281 set by the sender side as a minimum value for the receiver.
1283 @item pbkeylen=@var{bytes}
1284 Sender encryption key length, in bytes.
1285 Only can be set to 0, 16, 24 and 32.
1286 Enable sender encryption if not 0.
1287 Not required on receiver (set to 0),
1288 key size obtained from sender in HaiCrypt handshake.
1292 The time that should elapse since the moment when the
1293 packet was sent and the moment when it's delivered to
1294 the receiver application in the receiving function.
1295 This time should be a buffer time large enough to cover
1296 the time spent for sending, unexpectedly extended RTT
1297 time, and the time needed to retransmit the lost UDP
1298 packet. The effective latency value will be the maximum
1299 of this options' value and the value of @option{peerlatency}
1300 set by the peer side. Before version 1.3.0 this option
1301 is only available as @option{latency}.
1303 @item recv_buffer_size=@var{bytes}
1304 Set UDP receive buffer size, expressed in bytes.
1306 @item send_buffer_size=@var{bytes}
1307 Set UDP send buffer size, expressed in bytes.
1310 Set raise error timeout for read/write optations.
1312 This option is only relevant in read mode:
1313 if no data arrived in more than this time
1314 interval, raise error.
1316 @item tlpktdrop=@var{1|0}
1317 Too-late Packet Drop. When enabled on receiver, it skips
1318 missing packets that have not been delivered in time and
1319 delivers the following packets to the application when
1320 their time-to-play has come. It also sends a fake ACK to
1321 the sender. When enabled on sender and enabled on the
1322 receiving peer, the sender drops the older packets that
1323 have no chance of being delivered in time. It was
1324 automatically enabled in the sender if the receiver
1327 @item sndbuf=@var{bytes}
1328 Set send buffer size, expressed in bytes.
1330 @item rcvbuf=@var{bytes}
1331 Set receive buffer size, expressed in bytes.
1333 Receive buffer must not be greater than @option{ffs}.
1335 @item lossmaxttl=@var{packets}
1336 The value up to which the Reorder Tolerance may grow. When
1337 Reorder Tolerance is > 0, then packet loss report is delayed
1338 until that number of packets come in. Reorder Tolerance
1339 increases every time a "belated" packet has come, but it
1340 wasn't due to retransmission (that is, when UDP packets tend
1341 to come out of order), with the difference between the latest
1342 sequence and this packet's sequence, and not more than the
1343 value of this option. By default it's 0, which means that this
1344 mechanism is turned off, and the loss report is always sent
1345 immediately upon experiencing a "gap" in sequences.
1348 The minimum SRT version that is required from the peer. A connection
1349 to a peer that does not satisfy the minimum version requirement
1352 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1355 @item streamid=@var{string}
1356 A string limited to 512 characters that can be set on the socket prior
1357 to connecting. This stream ID will be able to be retrieved by the
1358 listener side from the socket that is returned from srt_accept and
1359 was connected by a socket with that set stream ID. SRT does not enforce
1360 any special interpretation of the contents of this string.
1361 This option doesn’t make sense in Rendezvous connection; the result
1362 might be that simply one side will override the value from the other
1363 side and it’s the matter of luck which one would win
1365 @item smoother=@var{live|file}
1366 The type of Smoother used for the transmission for that socket, which
1367 is responsible for the transmission and congestion control. The Smoother
1368 type must be exactly the same on both connecting parties, otherwise
1369 the connection is rejected.
1371 @item messageapi=@var{1|0}
1372 When set, this socket uses the Message API, otherwise it uses Buffer
1373 API. Note that in live mode (see @option{transtype}) there’s only
1374 message API available. In File mode you can chose to use one of two modes:
1376 Stream API (default, when this option is false). In this mode you may
1377 send as many data as you wish with one sending instruction, or even use
1378 dedicated functions that read directly from a file. The internal facility
1379 will take care of any speed and congestion control. When receiving, you
1380 can also receive as many data as desired, the data not extracted will be
1381 waiting for the next call. There is no boundary between data portions in
1384 Message API. In this mode your single sending instruction passes exactly
1385 one piece of data that has boundaries (a message). Contrary to Live mode,
1386 this message may span across multiple UDP packets and the only size
1387 limitation is that it shall fit as a whole in the sending buffer. The
1388 receiver shall use as large buffer as necessary to receive the message,
1389 otherwise the message will not be given up. When the message is not
1390 complete (not all packets received or there was a packet loss) it will
1393 @item transtype=@var{live|file}
1394 Sets the transmission type for the socket, in particular, setting this
1395 option sets multiple other parameters to their default values as required
1396 for a particular transmission type.
1398 live: Set options as for live transmission. In this mode, you should
1399 send by one sending instruction only so many data that fit in one UDP packet,
1400 and limited to the value defined first in @option{payload_size} (1316 is
1401 default in this mode). There is no speed control in this mode, only the
1402 bandwidth control, if configured, in order to not exceed the bandwidth with
1403 the overhead transmission (retransmitted and control packets).
1405 file: Set options as for non-live transmission. See @option{messageapi}
1406 for further explanations
1410 For more information see: @url{https://github.com/Haivision/srt}.
1414 Secure Real-time Transport Protocol.
1416 The accepted options are:
1419 @item srtp_out_suite
1420 Select input and output encoding suites.
1424 @item AES_CM_128_HMAC_SHA1_80
1425 @item SRTP_AES128_CM_HMAC_SHA1_80
1426 @item AES_CM_128_HMAC_SHA1_32
1427 @item SRTP_AES128_CM_HMAC_SHA1_32
1430 @item srtp_in_params
1431 @item srtp_out_params
1432 Set input and output encoding parameters, which are expressed by a
1433 base64-encoded representation of a binary block. The first 16 bytes of
1434 this binary block are used as master key, the following 14 bytes are
1435 used as master salt.
1440 Virtually extract a segment of a file or another stream.
1441 The underlying stream must be seekable.
1446 Start offset of the extracted segment, in bytes.
1448 End offset of the extracted segment, in bytes.
1449 If set to 0, extract till end of file.
1454 Extract a chapter from a DVD VOB file (start and end sectors obtained
1455 externally and multiplied by 2048):
1457 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1460 Play an AVI file directly from a TAR archive:
1462 subfile,,start,183241728,end,366490624,,:archive.tar
1465 Play a MPEG-TS file from start offset till end:
1467 subfile,,start,32815239,end,0,,:video.ts
1472 Writes the output to multiple protocols. The individual outputs are separated
1476 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1481 Transmission Control Protocol.
1483 The required syntax for a TCP url is:
1485 tcp://@var{hostname}:@var{port}[?@var{options}]
1488 @var{options} contains a list of &-separated options of the form
1489 @var{key}=@var{val}.
1491 The list of supported options follows.
1494 @item listen=@var{1|0}
1495 Listen for an incoming connection. Default value is 0.
1497 @item timeout=@var{microseconds}
1498 Set raise error timeout, expressed in microseconds.
1500 This option is only relevant in read mode: if no data arrived in more
1501 than this time interval, raise error.
1503 @item listen_timeout=@var{milliseconds}
1504 Set listen timeout, expressed in milliseconds.
1506 @item recv_buffer_size=@var{bytes}
1507 Set receive buffer size, expressed bytes.
1509 @item send_buffer_size=@var{bytes}
1510 Set send buffer size, expressed bytes.
1512 @item tcp_nodelay=@var{1|0}
1513 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1515 @item tcp_mss=@var{bytes}
1516 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1519 The following example shows how to setup a listening TCP connection
1520 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1522 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1523 ffplay tcp://@var{hostname}:@var{port}
1528 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1530 The required syntax for a TLS/SSL url is:
1532 tls://@var{hostname}:@var{port}[?@var{options}]
1535 The following parameters can be set via command line options
1536 (or in code via @code{AVOption}s):
1540 @item ca_file, cafile=@var{filename}
1541 A file containing certificate authority (CA) root certificates to treat
1542 as trusted. If the linked TLS library contains a default this might not
1543 need to be specified for verification to work, but not all libraries and
1544 setups have defaults built in.
1545 The file must be in OpenSSL PEM format.
1547 @item tls_verify=@var{1|0}
1548 If enabled, try to verify the peer that we are communicating with.
1549 Note, if using OpenSSL, this currently only makes sure that the
1550 peer certificate is signed by one of the root certificates in the CA
1551 database, but it does not validate that the certificate actually
1552 matches the host name we are trying to connect to. (With other backends,
1553 the host name is validated as well.)
1555 This is disabled by default since it requires a CA database to be
1556 provided by the caller in many cases.
1558 @item cert_file, cert=@var{filename}
1559 A file containing a certificate to use in the handshake with the peer.
1560 (When operating as server, in listen mode, this is more often required
1561 by the peer, while client certificates only are mandated in certain
1564 @item key_file, key=@var{filename}
1565 A file containing the private key for the certificate.
1567 @item listen=@var{1|0}
1568 If enabled, listen for connections on the provided port, and assume
1569 the server role in the handshake instead of the client role.
1573 Example command lines:
1575 To create a TLS/SSL server that serves an input stream.
1578 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1581 To play back a stream from the TLS/SSL server using @command{ffplay}:
1584 ffplay tls://@var{hostname}:@var{port}
1589 User Datagram Protocol.
1591 The required syntax for an UDP URL is:
1593 udp://@var{hostname}:@var{port}[?@var{options}]
1596 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1598 In case threading is enabled on the system, a circular buffer is used
1599 to store the incoming data, which allows one to reduce loss of data due to
1600 UDP socket buffer overruns. The @var{fifo_size} and
1601 @var{overrun_nonfatal} options are related to this buffer.
1603 The list of supported options follows.
1606 @item buffer_size=@var{size}
1607 Set the UDP maximum socket buffer size in bytes. This is used to set either
1608 the receive or send buffer size, depending on what the socket is used for.
1609 Default is 64KB. See also @var{fifo_size}.
1611 @item bitrate=@var{bitrate}
1612 If set to nonzero, the output will have the specified constant bitrate if the
1613 input has enough packets to sustain it.
1615 @item burst_bits=@var{bits}
1616 When using @var{bitrate} this specifies the maximum number of bits in
1619 @item localport=@var{port}
1620 Override the local UDP port to bind with.
1622 @item localaddr=@var{addr}
1623 Local IP address of a network interface used for sending packets or joining
1626 @item pkt_size=@var{size}
1627 Set the size in bytes of UDP packets.
1629 @item reuse=@var{1|0}
1630 Explicitly allow or disallow reusing UDP sockets.
1633 Set the time to live value (for multicast only).
1635 @item connect=@var{1|0}
1636 Initialize the UDP socket with @code{connect()}. In this case, the
1637 destination address can't be changed with ff_udp_set_remote_url later.
1638 If the destination address isn't known at the start, this option can
1639 be specified in ff_udp_set_remote_url, too.
1640 This allows finding out the source address for the packets with getsockname,
1641 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1642 unreachable" is received.
1643 For receiving, this gives the benefit of only receiving packets from
1644 the specified peer address/port.
1646 @item sources=@var{address}[,@var{address}]
1647 Only receive packets sent from the specified addresses. In case of multicast,
1648 also subscribe to multicast traffic coming from these addresses only.
1650 @item block=@var{address}[,@var{address}]
1651 Ignore packets sent from the specified addresses. In case of multicast, also
1652 exclude the source addresses in the multicast subscription.
1654 @item fifo_size=@var{units}
1655 Set the UDP receiving circular buffer size, expressed as a number of
1656 packets with size of 188 bytes. If not specified defaults to 7*4096.
1658 @item overrun_nonfatal=@var{1|0}
1659 Survive in case of UDP receiving circular buffer overrun. Default
1662 @item timeout=@var{microseconds}
1663 Set raise error timeout, expressed in microseconds.
1665 This option is only relevant in read mode: if no data arrived in more
1666 than this time interval, raise error.
1668 @item broadcast=@var{1|0}
1669 Explicitly allow or disallow UDP broadcasting.
1671 Note that broadcasting may not work properly on networks having
1672 a broadcast storm protection.
1675 @subsection Examples
1679 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1681 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1685 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1686 sized UDP packets, using a large input buffer:
1688 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1692 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1694 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1702 The required syntax for a Unix socket URL is:
1705 unix://@var{filepath}
1708 The following parameters can be set via command line options
1709 (or in code via @code{AVOption}s):
1715 Create the Unix socket in listening mode.
1718 @c man end PROTOCOLS