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 The list of supported options follows:
11 @item protocol_whitelist @var{list} (@emph{input})
12 Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
13 prefixed by "-" are disabled.
14 All protocols are allowed by default but protocols used by an another
15 protocol (nested protocols) are restricted to a per protocol subset.
18 @c man end PROTOCOL OPTIONS
21 @c man begin PROTOCOLS
23 Protocols are configured elements in FFmpeg that enable access to
24 resources that require specific protocols.
26 When you configure your FFmpeg build, all the supported protocols are
27 enabled by default. You can list all available ones using the
28 configure option "--list-protocols".
30 You can disable all the protocols using the configure option
31 "--disable-protocols", and selectively enable a protocol using the
32 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
33 particular protocol using the option
34 "--disable-protocol=@var{PROTOCOL}".
36 The option "-protocols" of the ff* tools will display the list of
39 All protocols accept the following options:
43 Maximum time to wait for (network) read/write operations to complete,
47 A description of the currently available protocols follows.
51 Asynchronous data filling wrapper for input stream.
53 Fill data in a background thread, to decouple I/O operation from demux thread.
57 async:http://host/resource
58 async:cache:http://host/resource
65 The accepted options are:
75 Playlist to read (BDMV/PLAYLIST/?????.mpls)
81 Read longest playlist from BluRay mounted to /mnt/bluray:
86 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
88 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
93 Caching wrapper for input stream.
95 Cache the input stream to temporary file. It brings seeking capability to live streams.
103 Physical concatenation protocol.
105 Read and seek from many resources in sequence as if they were
108 A URL accepted by this protocol has the syntax:
110 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
113 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
114 resource to be concatenated, each one possibly specifying a distinct
117 For example to read a sequence of files @file{split1.mpeg},
118 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
121 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
124 Note that you may need to escape the character "|" which is special for
129 AES-encrypted stream reading protocol.
131 The accepted options are:
134 Set the AES decryption key binary block from given hexadecimal representation.
137 Set the AES decryption initialization vector binary block from given hexadecimal representation.
140 Accepted URL formats:
148 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
150 For example, to convert a GIF file given inline with @command{ffmpeg}:
152 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
157 File access protocol.
159 Read from or write to a file.
161 A file URL can have the form:
166 where @var{filename} is the path of the file to read.
168 An URL that does not have a protocol prefix will be assumed to be a
169 file URL. Depending on the build, an URL that looks like a Windows
170 path with the drive letter at the beginning will also be assumed to be
171 a file URL (usually not the case in builds for unix-like systems).
173 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
176 ffmpeg -i file:input.mpeg output.mpeg
179 This protocol accepts the following options:
183 Truncate existing files on write, if set to 1. A value of 0 prevents
184 truncating. Default value is 1.
187 Set I/O operation maximum block size, in bytes. Default value is
188 @code{INT_MAX}, which results in not limiting the requested block size.
189 Setting this value reasonably low improves user termination request reaction
190 time, which is valuable for files on slow medium.
195 FTP (File Transfer Protocol).
197 Read from or write to remote resources using FTP protocol.
199 Following syntax is required.
201 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
204 This protocol accepts the following options.
208 Set timeout in microseconds of socket I/O operations used by the underlying low level
209 operation. By default it is set to -1, which means that the timeout is
212 @item ftp-anonymous-password
213 Password used when login as anonymous user. Typically an e-mail address
216 @item ftp-write-seekable
217 Control seekability of connection during encoding. If set to 1 the
218 resource is supposed to be seekable, if set to 0 it is assumed not
219 to be seekable. Default value is 0.
222 NOTE: Protocol can be used as output, but it is recommended to not do
223 it, unless special care is taken (tests, customized server configuration
224 etc.). Different FTP servers behave in different way during seek
225 operation. ff* tools may produce incomplete content due to server limitations.
227 This protocol accepts the following options:
231 If set to 1, the protocol will retry reading at the end of the file, allowing
232 reading files that still are being written. In order for this to terminate,
233 you either need to use the rw_timeout option, or use the interrupt callback
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 Override the User-Agent header. If not specified the protocol will use a
296 string describing the libavformat build. ("Lavf/<version>")
299 This is a deprecated option, you can use user_agent instead it.
302 Set timeout in microseconds of socket I/O operations used by the underlying low level
303 operation. By default it is set to -1, which means that the timeout is
306 @item reconnect_at_eof
307 If set then eof is treated like an error and causes reconnection, this is useful
308 for live / endless streams.
310 @item reconnect_streamed
311 If set then even streamed/non seekable streams will be reconnected on errors.
313 @item reconnect_delay_max
314 Sets the maximum delay in seconds after which to give up reconnecting
317 Export the MIME type.
320 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
321 supports this, the metadata has to be retrieved by the application by reading
322 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
325 @item icy_metadata_headers
326 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
327 headers, separated by newline characters.
329 @item icy_metadata_packet
330 If the server supports ICY metadata, and @option{icy} was set to 1, this
331 contains the last non-empty metadata packet sent by the server. It should be
332 polled in regular intervals by applications interested in mid-stream metadata
336 Set the cookies to be sent in future requests. The format of each cookie is the
337 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
338 delimited by a newline character.
341 Set initial byte offset.
344 Try to limit the request to bytes preceding this offset.
347 When used as a client option it sets the HTTP method for the request.
349 When used as a server option it sets the HTTP method that is going to be
350 expected from the client(s).
351 If the expected and the received HTTP method do not match the client will
352 be given a Bad Request response.
353 When unset the HTTP method is not checked for now. This will be replaced by
354 autodetection in the future.
357 If set to 1 enables experimental HTTP server. This can be used to send data when
358 used as an output option, or read data from a client with HTTP POST when used as
360 If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
361 in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
363 # Server side (sending):
364 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
366 # Client side (receiving):
367 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
369 # Client can also be done with wget:
370 wget http://@var{server}:@var{port} -O somefile.ogg
372 # Server side (receiving):
373 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
375 # Client side (sending):
376 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
378 # Client can also be done with wget:
379 wget --post-file=somefile.ogg http://@var{server}:@var{port}
384 @subsection HTTP Cookies
386 Some HTTP requests will be denied unless cookie values are passed in with the
387 request. The @option{cookies} option allows these cookies to be specified. At
388 the very least, each cookie must specify a value along with a path and domain.
389 HTTP requests that match both the domain and path will automatically include the
390 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
393 The required syntax to play a stream specifying a cookie is:
395 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
400 Icecast protocol (stream to Icecast servers)
402 This protocol accepts the following options:
406 Set the stream genre.
411 @item ice_description
412 Set the stream description.
415 Set the stream website URL.
418 Set if the stream should be public.
419 The default is 0 (not public).
422 Override the User-Agent header. If not specified a string of the form
423 "Lavf/<version>" will be used.
426 Set the Icecast mountpoint password.
429 Set the stream content type. This must be set if it is different from
433 This enables support for Icecast versions < 2.4.0, that do not support the
434 HTTP PUT method but the SOURCE method.
439 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
444 MMS (Microsoft Media Server) protocol over TCP.
448 MMS (Microsoft Media Server) protocol over HTTP.
450 The required syntax is:
452 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
459 Computes the MD5 hash of the data to be written, and on close writes
460 this to the designated output or stdout if none is specified. It can
461 be used to test muxers without writing an actual file.
463 Some examples follow.
465 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
466 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
468 # Write the MD5 hash of the encoded AVI file to stdout.
469 ffmpeg -i input.flv -f avi -y md5:
472 Note that some formats (typically MOV) require the output protocol to
473 be seekable, so they will fail with the MD5 output protocol.
477 UNIX pipe access protocol.
479 Read and write from UNIX pipes.
481 The accepted syntax is:
486 @var{number} is the number corresponding to the file descriptor of the
487 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
488 is not specified, by default the stdout file descriptor will be used
489 for writing, stdin for reading.
491 For example to read from stdin with @command{ffmpeg}:
493 cat test.wav | ffmpeg -i pipe:0
494 # ...this is the same as...
495 cat test.wav | ffmpeg -i pipe:
498 For writing to stdout with @command{ffmpeg}:
500 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
501 # ...this is the same as...
502 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
505 This protocol accepts the following options:
509 Set I/O operation maximum block size, in bytes. Default value is
510 @code{INT_MAX}, which results in not limiting the requested block size.
511 Setting this value reasonably low improves user termination request reaction
512 time, which is valuable if data transmission is slow.
515 Note that some formats (typically MOV), require the output protocol to
516 be seekable, so they will fail with the pipe output protocol.
520 Real-Time Messaging Protocol.
522 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
523 content across a TCP/IP network.
525 The required syntax is:
527 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
530 The accepted parameters are:
534 An optional username (mostly for publishing).
537 An optional password (mostly for publishing).
540 The address of the RTMP server.
543 The number of the TCP port to use (by default is 1935).
546 It is the name of the application to access. It usually corresponds to
547 the path where the application is installed on the RTMP server
548 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
549 the value parsed from the URI through the @code{rtmp_app} option, too.
552 It is the path or name of the resource to play with reference to the
553 application specified in @var{app}, may be prefixed by "mp4:". You
554 can override the value parsed from the URI through the @code{rtmp_playpath}
558 Act as a server, listening for an incoming connection.
561 Maximum time to wait for the incoming connection. Implies listen.
564 Additionally, the following parameters can be set via command line options
565 (or in code via @code{AVOption}s):
569 Name of application to connect on the RTMP server. This option
570 overrides the parameter specified in the URI.
573 Set the client buffer time in milliseconds. The default is 3000.
576 Extra arbitrary AMF connection parameters, parsed from a string,
577 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
578 Each value is prefixed by a single character denoting the type,
579 B for Boolean, N for number, S for string, O for object, or Z for null,
580 followed by a colon. For Booleans the data must be either 0 or 1 for
581 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
582 1 to end or begin an object, respectively. Data items in subobjects may
583 be named, by prefixing the type with 'N' and specifying the name before
584 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
585 times to construct arbitrary AMF sequences.
588 Version of the Flash plugin used to run the SWF player. The default
589 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
590 <libavformat version>).)
592 @item rtmp_flush_interval
593 Number of packets flushed in the same request (RTMPT only). The default
597 Specify that the media is a live stream. No resuming or seeking in
598 live streams is possible. The default value is @code{any}, which means the
599 subscriber first tries to play the live stream specified in the
600 playpath. If a live stream of that name is not found, it plays the
601 recorded stream. The other possible values are @code{live} and
605 URL of the web page in which the media was embedded. By default no
609 Stream identifier to play or to publish. This option overrides the
610 parameter specified in the URI.
613 Name of live stream to subscribe to. By default no value will be sent.
614 It is only sent if the option is specified or if rtmp_live
618 SHA256 hash of the decompressed SWF file (32 bytes).
621 Size of the decompressed SWF file, required for SWFVerification.
624 URL of the SWF player for the media. By default no value will be sent.
627 URL to player swf file, compute hash/size automatically.
630 URL of the target stream. Defaults to proto://host[:port]/app.
634 For example to read with @command{ffplay} a multimedia resource named
635 "sample" from the application "vod" from an RTMP server "myserver":
637 ffplay rtmp://myserver/vod/sample
640 To publish to a password protected server, passing the playpath and
641 app names separately:
643 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
648 Encrypted Real-Time Messaging Protocol.
650 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
651 streaming multimedia content within standard cryptographic primitives,
652 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
657 Real-Time Messaging Protocol over a secure SSL connection.
659 The Real-Time Messaging Protocol (RTMPS) is used for streaming
660 multimedia content across an encrypted connection.
664 Real-Time Messaging Protocol tunneled through HTTP.
666 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
667 for streaming multimedia content within HTTP requests to traverse
672 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
674 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
675 is used for streaming multimedia content within HTTP requests to traverse
680 Real-Time Messaging Protocol tunneled through HTTPS.
682 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
683 for streaming multimedia content within HTTPS requests to traverse
686 @section libsmbclient
688 libsmbclient permits one to manipulate CIFS/SMB network resources.
690 Following syntax is required.
693 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
696 This protocol accepts the following options.
700 Set timeout in milliseconds of socket I/O operations used by the underlying
701 low level operation. By default it is set to -1, which means that the timeout
705 Truncate existing files on write, if set to 1. A value of 0 prevents
706 truncating. Default value is 1.
709 Set the workgroup used for making connections. By default workgroup is not specified.
713 For more information see: @url{http://www.samba.org/}.
717 Secure File Transfer Protocol via libssh
719 Read from or write to remote resources using SFTP protocol.
721 Following syntax is required.
724 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
727 This protocol accepts the following options.
731 Set timeout of socket I/O operations used by the underlying low level
732 operation. By default it is set to -1, which means that the timeout
736 Truncate existing files on write, if set to 1. A value of 0 prevents
737 truncating. Default value is 1.
740 Specify the path of the file containing private key to use during authorization.
741 By default libssh searches for keys in the @file{~/.ssh/} directory.
745 Example: Play a file stored on remote server.
748 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
751 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
753 Real-Time Messaging Protocol and its variants supported through
756 Requires the presence of the librtmp headers and library during
757 configuration. You need to explicitly configure the build with
758 "--enable-librtmp". If enabled this will replace the native RTMP
761 This protocol provides most client functions and a few server
762 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
763 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
764 variants of these encrypted types (RTMPTE, RTMPTS).
766 The required syntax is:
768 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
771 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
772 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
773 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
774 meaning as specified for the RTMP native protocol.
775 @var{options} contains a list of space-separated options of the form
778 See the librtmp manual page (man 3 librtmp) for more information.
780 For example, to stream a file in real-time to an RTMP server using
783 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
786 To play the same stream using @command{ffplay}:
788 ffplay "rtmp://myserver/live/mystream live=1"
793 Real-time Transport Protocol.
795 The required syntax for an RTP URL is:
796 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
798 @var{port} specifies the RTP port to use.
800 The following URL options are supported:
805 Set the TTL (Time-To-Live) value (for multicast only).
807 @item rtcpport=@var{n}
808 Set the remote RTCP port to @var{n}.
810 @item localrtpport=@var{n}
811 Set the local RTP port to @var{n}.
813 @item localrtcpport=@var{n}'
814 Set the local RTCP port to @var{n}.
816 @item pkt_size=@var{n}
817 Set max packet size (in bytes) to @var{n}.
820 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
823 @item sources=@var{ip}[,@var{ip}]
824 List allowed source IP addresses.
826 @item block=@var{ip}[,@var{ip}]
827 List disallowed (blocked) source IP addresses.
829 @item write_to_source=0|1
830 Send packets to the source address of the latest received packet (if
831 set to 1) or to a default remote address (if set to 0).
833 @item localport=@var{n}
834 Set the local RTP port to @var{n}.
836 This is a deprecated option. Instead, @option{localrtpport} should be
846 If @option{rtcpport} is not set the RTCP port will be set to the RTP
850 If @option{localrtpport} (the local RTP port) is not set any available
851 port will be used for the local RTP and RTCP ports.
854 If @option{localrtcpport} (the local RTCP port) is not set it will be
855 set to the local RTP port value plus 1.
860 Real-Time Streaming Protocol.
862 RTSP is not technically a protocol handler in libavformat, it is a demuxer
863 and muxer. The demuxer supports both normal RTSP (with data transferred
864 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
865 data transferred over RDT).
867 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
868 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
869 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
871 The required syntax for a RTSP url is:
873 rtsp://@var{hostname}[:@var{port}]/@var{path}
876 Options can be set on the @command{ffmpeg}/@command{ffplay} command
877 line, or set in code via @code{AVOption}s or in
878 @code{avformat_open_input}.
880 The following options are supported.
884 Do not start playing the stream immediately if set to 1. Default value
888 Set RTSP transport protocols.
890 It accepts the following values:
893 Use UDP as lower transport protocol.
896 Use TCP (interleaving within the RTSP control channel) as lower
900 Use UDP multicast as lower transport protocol.
903 Use HTTP tunneling as lower transport protocol, which is useful for
907 Multiple lower transport protocols may be specified, in that case they are
908 tried one at a time (if the setup of one fails, the next one is tried).
909 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
914 The following values are accepted:
917 Accept packets only from negotiated peer address and port.
919 Act as a server, listening for an incoming connection.
921 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
924 Default value is @samp{none}.
926 @item allowed_media_types
927 Set media types to accept from the server.
929 The following flags are accepted:
936 By default it accepts all media types.
939 Set minimum local UDP port. Default value is 5000.
942 Set maximum local UDP port. Default value is 65000.
945 Set maximum timeout (in seconds) to wait for incoming connections.
947 A value of -1 means infinite (default). This option implies the
948 @option{rtsp_flags} set to @samp{listen}.
950 @item reorder_queue_size
951 Set number of packets to buffer for handling of reordered packets.
954 Set socket TCP I/O timeout in microseconds.
957 Override User-Agent header. If not specified, it defaults to the
958 libavformat identifier string.
961 When receiving data over UDP, the demuxer tries to reorder received packets
962 (since they may arrive out of order, or packets may get lost totally). This
963 can be disabled by setting the maximum demuxing delay to zero (via
964 the @code{max_delay} field of AVFormatContext).
966 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
967 streams to display can be chosen with @code{-vst} @var{n} and
968 @code{-ast} @var{n} for video and audio respectively, and can be switched
969 on the fly by pressing @code{v} and @code{a}.
973 The following examples all make use of the @command{ffplay} and
974 @command{ffmpeg} tools.
978 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
980 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
984 Watch a stream tunneled over HTTP:
986 ffplay -rtsp_transport http rtsp://server/video.mp4
990 Send a stream in realtime to a RTSP server, for others to watch:
992 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
996 Receive a stream in realtime:
998 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1004 Session Announcement Protocol (RFC 2974). This is not technically a
1005 protocol handler in libavformat, it is a muxer and demuxer.
1006 It is used for signalling of RTP streams, by announcing the SDP for the
1007 streams regularly on a separate port.
1011 The syntax for a SAP url given to the muxer is:
1013 sap://@var{destination}[:@var{port}][?@var{options}]
1016 The RTP packets are sent to @var{destination} on port @var{port},
1017 or to port 5004 if no port is specified.
1018 @var{options} is a @code{&}-separated list. The following options
1023 @item announce_addr=@var{address}
1024 Specify the destination IP address for sending the announcements to.
1025 If omitted, the announcements are sent to the commonly used SAP
1026 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1027 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1029 @item announce_port=@var{port}
1030 Specify the port to send the announcements on, defaults to
1031 9875 if not specified.
1034 Specify the time to live value for the announcements and RTP packets,
1037 @item same_port=@var{0|1}
1038 If set to 1, send all RTP streams on the same port pair. If zero (the
1039 default), all streams are sent on unique ports, with each stream on a
1040 port 2 numbers higher than the previous.
1041 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1042 The RTP stack in libavformat for receiving requires all streams to be sent
1046 Example command lines follow.
1048 To broadcast a stream on the local subnet, for watching in VLC:
1051 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1054 Similarly, for watching in @command{ffplay}:
1057 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1060 And for watching in @command{ffplay}, over IPv6:
1063 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1068 The syntax for a SAP url given to the demuxer is:
1070 sap://[@var{address}][:@var{port}]
1073 @var{address} is the multicast address to listen for announcements on,
1074 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1075 is the port that is listened on, 9875 if omitted.
1077 The demuxers listens for announcements on the given address and port.
1078 Once an announcement is received, it tries to receive that particular stream.
1080 Example command lines follow.
1082 To play back the first stream announced on the normal SAP multicast address:
1088 To play back the first stream announced on one the default IPv6 SAP multicast address:
1091 ffplay sap://[ff0e::2:7ffe]
1096 Stream Control Transmission Protocol.
1098 The accepted URL syntax is:
1100 sctp://@var{host}:@var{port}[?@var{options}]
1103 The protocol accepts the following options:
1106 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1109 Set the maximum number of streams. By default no limit is set.
1114 Secure Real-time Transport Protocol.
1116 The accepted options are:
1119 @item srtp_out_suite
1120 Select input and output encoding suites.
1124 @item AES_CM_128_HMAC_SHA1_80
1125 @item SRTP_AES128_CM_HMAC_SHA1_80
1126 @item AES_CM_128_HMAC_SHA1_32
1127 @item SRTP_AES128_CM_HMAC_SHA1_32
1130 @item srtp_in_params
1131 @item srtp_out_params
1132 Set input and output encoding parameters, which are expressed by a
1133 base64-encoded representation of a binary block. The first 16 bytes of
1134 this binary block are used as master key, the following 14 bytes are
1135 used as master salt.
1140 Virtually extract a segment of a file or another stream.
1141 The underlying stream must be seekable.
1146 Start offset of the extracted segment, in bytes.
1148 End offset of the extracted segment, in bytes.
1153 Extract a chapter from a DVD VOB file (start and end sectors obtained
1154 externally and multiplied by 2048):
1156 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1159 Play an AVI file directly from a TAR archive:
1161 subfile,,start,183241728,end,366490624,,:archive.tar
1166 Writes the output to multiple protocols. The individual outputs are separated
1170 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1175 Transmission Control Protocol.
1177 The required syntax for a TCP url is:
1179 tcp://@var{hostname}:@var{port}[?@var{options}]
1182 @var{options} contains a list of &-separated options of the form
1183 @var{key}=@var{val}.
1185 The list of supported options follows.
1188 @item listen=@var{1|0}
1189 Listen for an incoming connection. Default value is 0.
1191 @item timeout=@var{microseconds}
1192 Set raise error timeout, expressed in microseconds.
1194 This option is only relevant in read mode: if no data arrived in more
1195 than this time interval, raise error.
1197 @item listen_timeout=@var{milliseconds}
1198 Set listen timeout, expressed in milliseconds.
1200 @item recv_buffer_size=@var{bytes}
1201 Set receive buffer size, expressed bytes.
1203 @item send_buffer_size=@var{bytes}
1204 Set send buffer size, expressed bytes.
1207 The following example shows how to setup a listening TCP connection
1208 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1210 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1211 ffplay tcp://@var{hostname}:@var{port}
1216 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1218 The required syntax for a TLS/SSL url is:
1220 tls://@var{hostname}:@var{port}[?@var{options}]
1223 The following parameters can be set via command line options
1224 (or in code via @code{AVOption}s):
1228 @item ca_file, cafile=@var{filename}
1229 A file containing certificate authority (CA) root certificates to treat
1230 as trusted. If the linked TLS library contains a default this might not
1231 need to be specified for verification to work, but not all libraries and
1232 setups have defaults built in.
1233 The file must be in OpenSSL PEM format.
1235 @item tls_verify=@var{1|0}
1236 If enabled, try to verify the peer that we are communicating with.
1237 Note, if using OpenSSL, this currently only makes sure that the
1238 peer certificate is signed by one of the root certificates in the CA
1239 database, but it does not validate that the certificate actually
1240 matches the host name we are trying to connect to. (With GnuTLS,
1241 the host name is validated as well.)
1243 This is disabled by default since it requires a CA database to be
1244 provided by the caller in many cases.
1246 @item cert_file, cert=@var{filename}
1247 A file containing a certificate to use in the handshake with the peer.
1248 (When operating as server, in listen mode, this is more often required
1249 by the peer, while client certificates only are mandated in certain
1252 @item key_file, key=@var{filename}
1253 A file containing the private key for the certificate.
1255 @item listen=@var{1|0}
1256 If enabled, listen for connections on the provided port, and assume
1257 the server role in the handshake instead of the client role.
1261 Example command lines:
1263 To create a TLS/SSL server that serves an input stream.
1266 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1269 To play back a stream from the TLS/SSL server using @command{ffplay}:
1272 ffplay tls://@var{hostname}:@var{port}
1277 User Datagram Protocol.
1279 The required syntax for an UDP URL is:
1281 udp://@var{hostname}:@var{port}[?@var{options}]
1284 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1286 In case threading is enabled on the system, a circular buffer is used
1287 to store the incoming data, which allows one to reduce loss of data due to
1288 UDP socket buffer overruns. The @var{fifo_size} and
1289 @var{overrun_nonfatal} options are related to this buffer.
1291 The list of supported options follows.
1294 @item buffer_size=@var{size}
1295 Set the UDP maximum socket buffer size in bytes. This is used to set either
1296 the receive or send buffer size, depending on what the socket is used for.
1297 Default is 64KB. See also @var{fifo_size}.
1299 @item bitrate=@var{bitrate}
1300 If set to nonzero, the output will have the specified constant bitrate if the
1301 input has enough packets to sustain it.
1303 @item burst_bits=@var{bits}
1304 When using @var{bitrate} this specifies the maximum number of bits in
1307 @item localport=@var{port}
1308 Override the local UDP port to bind with.
1310 @item localaddr=@var{addr}
1311 Choose the local IP address. This is useful e.g. if sending multicast
1312 and the host has multiple interfaces, where the user can choose
1313 which interface to send on by specifying the IP address of that interface.
1315 @item pkt_size=@var{size}
1316 Set the size in bytes of UDP packets.
1318 @item reuse=@var{1|0}
1319 Explicitly allow or disallow reusing UDP sockets.
1322 Set the time to live value (for multicast only).
1324 @item connect=@var{1|0}
1325 Initialize the UDP socket with @code{connect()}. In this case, the
1326 destination address can't be changed with ff_udp_set_remote_url later.
1327 If the destination address isn't known at the start, this option can
1328 be specified in ff_udp_set_remote_url, too.
1329 This allows finding out the source address for the packets with getsockname,
1330 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1331 unreachable" is received.
1332 For receiving, this gives the benefit of only receiving packets from
1333 the specified peer address/port.
1335 @item sources=@var{address}[,@var{address}]
1336 Only receive packets sent to the multicast group from one of the
1337 specified sender IP addresses.
1339 @item block=@var{address}[,@var{address}]
1340 Ignore packets sent to the multicast group from the specified
1341 sender IP addresses.
1343 @item fifo_size=@var{units}
1344 Set the UDP receiving circular buffer size, expressed as a number of
1345 packets with size of 188 bytes. If not specified defaults to 7*4096.
1347 @item overrun_nonfatal=@var{1|0}
1348 Survive in case of UDP receiving circular buffer overrun. Default
1351 @item timeout=@var{microseconds}
1352 Set raise error timeout, expressed in microseconds.
1354 This option is only relevant in read mode: if no data arrived in more
1355 than this time interval, raise error.
1357 @item broadcast=@var{1|0}
1358 Explicitly allow or disallow UDP broadcasting.
1360 Note that broadcasting may not work properly on networks having
1361 a broadcast storm protection.
1364 @subsection Examples
1368 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1370 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1374 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1375 sized UDP packets, using a large input buffer:
1377 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1381 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1383 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1391 The required syntax for a Unix socket URL is:
1394 unix://@var{filepath}
1397 The following parameters can be set via command line options
1398 (or in code via @code{AVOption}s):
1404 Create the Unix socket in listening mode.
1407 @c man end PROTOCOLS