4 Protocols are configured elements in FFmpeg that enable access to
5 resources that require specific protocols.
7 When you configure your FFmpeg build, all the supported protocols are
8 enabled by default. You can list all available ones using the
9 configure option "--list-protocols".
11 You can disable all the protocols using the configure option
12 "--disable-protocols", and selectively enable a protocol using the
13 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
14 particular protocol using the option
15 "--disable-protocol=@var{PROTOCOL}".
17 The option "-protocols" of the ff* tools will display the list of
20 A description of the currently available protocols follows.
24 Asynchronous data filling wrapper for input stream.
26 Fill data in a background thread, to decouple I/O operation from demux thread.
30 async:http://host/resource
31 async:cache:http://host/resource
38 The accepted options are:
48 Playlist to read (BDMV/PLAYLIST/?????.mpls)
54 Read longest playlist from BluRay mounted to /mnt/bluray:
59 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
61 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
66 Caching wrapper for input stream.
68 Cache the input stream to temporary file. It brings seeking capability to live streams.
76 Physical concatenation protocol.
78 Read and seek from many resources in sequence as if they were
81 A URL accepted by this protocol has the syntax:
83 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
86 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
87 resource to be concatenated, each one possibly specifying a distinct
90 For example to read a sequence of files @file{split1.mpeg},
91 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
94 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
97 Note that you may need to escape the character "|" which is special for
102 AES-encrypted stream reading protocol.
104 The accepted options are:
107 Set the AES decryption key binary block from given hexadecimal representation.
110 Set the AES decryption initialization vector binary block from given hexadecimal representation.
113 Accepted URL formats:
121 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
123 For example, to convert a GIF file given inline with @command{ffmpeg}:
125 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
130 File access protocol.
132 Read from or write to a file.
134 A file URL can have the form:
139 where @var{filename} is the path of the file to read.
141 An URL that does not have a protocol prefix will be assumed to be a
142 file URL. Depending on the build, an URL that looks like a Windows
143 path with the drive letter at the beginning will also be assumed to be
144 a file URL (usually not the case in builds for unix-like systems).
146 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
149 ffmpeg -i file:input.mpeg output.mpeg
152 This protocol accepts the following options:
156 Truncate existing files on write, if set to 1. A value of 0 prevents
157 truncating. Default value is 1.
160 Set I/O operation maximum block size, in bytes. Default value is
161 @code{INT_MAX}, which results in not limiting the requested block size.
162 Setting this value reasonably low improves user termination request reaction
163 time, which is valuable for files on slow medium.
168 FTP (File Transfer Protocol).
170 Read from or write to remote resources using FTP protocol.
172 Following syntax is required.
174 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
177 This protocol accepts the following options.
181 Set timeout in microseconds of socket I/O operations used by the underlying low level
182 operation. By default it is set to -1, which means that the timeout is
185 @item ftp-anonymous-password
186 Password used when login as anonymous user. Typically an e-mail address
189 @item ftp-write-seekable
190 Control seekability of connection during encoding. If set to 1 the
191 resource is supposed to be seekable, if set to 0 it is assumed not
192 to be seekable. Default value is 0.
195 NOTE: Protocol can be used as output, but it is recommended to not do
196 it, unless special care is taken (tests, customized server configuration
197 etc.). Different FTP servers behave in different way during seek
198 operation. ff* tools may produce incomplete content due to server limitations.
206 Read Apple HTTP Live Streaming compliant segmented stream as
207 a uniform one. The M3U8 playlists describing the segments can be
208 remote HTTP resources or local files, accessed using the standard
210 The nested protocol is declared by specifying
211 "+@var{proto}" after the hls URI scheme name, where @var{proto}
212 is either "file" or "http".
215 hls+http://host/path/to/remote/resource.m3u8
216 hls+file://path/to/local/resource.m3u8
219 Using this protocol is discouraged - the hls demuxer should work
220 just as well (if not, please report the issues) and is more complete.
221 To use the hls demuxer instead, simply use the direct URLs to the
226 HTTP (Hyper Text Transfer Protocol).
228 This protocol accepts the following options:
232 Control seekability of connection. If set to 1 the resource is
233 supposed to be seekable, if set to 0 it is assumed not to be seekable,
234 if set to -1 it will try to autodetect if it is seekable. Default
238 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
241 Set a specific content type for the POST messages.
244 Set custom HTTP headers, can override built in default headers. The
245 value must be a string encoding the headers.
247 @item multiple_requests
248 Use persistent connections if set to 1, default is 0.
251 Set custom HTTP post data.
255 Override the User-Agent header. If not specified the protocol will use a
256 string describing the libavformat build. ("Lavf/<version>")
259 Set timeout in microseconds of socket I/O operations used by the underlying low level
260 operation. By default it is set to -1, which means that the timeout is
263 @item reconnect_at_eof
264 If set then eof is treated like an error and causes reconnection, this is usefull
265 for live / endless streams.
267 @item reconnect_streamed
268 If set then even streamed/non seekable streams will be reconnected on errors.
271 Export the MIME type.
274 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
275 supports this, the metadata has to be retrieved by the application by reading
276 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
279 @item icy_metadata_headers
280 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
281 headers, separated by newline characters.
283 @item icy_metadata_packet
284 If the server supports ICY metadata, and @option{icy} was set to 1, this
285 contains the last non-empty metadata packet sent by the server. It should be
286 polled in regular intervals by applications interested in mid-stream metadata
290 Set the cookies to be sent in future requests. The format of each cookie is the
291 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
292 delimited by a newline character.
295 Set initial byte offset.
298 Try to limit the request to bytes preceding this offset.
301 When used as a client option it sets the HTTP method for the request.
303 When used as a server option it sets the HTTP method that is going to be
304 expected from the client(s).
305 If the expected and the received HTTP method do not match the client will
306 be given a Bad Request response.
307 When unset the HTTP method is not checked for now. This will be replaced by
308 autodetection in the future.
311 If set to 1 enables experimental HTTP server. This can be used to send data when
312 used as an output option, or read data from a client with HTTP POST when used as
314 If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
315 in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
317 # Server side (sending):
318 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
320 # Client side (receiving):
321 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
323 # Client can also be done with wget:
324 wget http://@var{server}:@var{port} -O somefile.ogg
326 # Server side (receiving):
327 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
329 # Client side (sending):
330 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
332 # Client can also be done with wget:
333 wget --post-file=somefile.ogg http://@var{server}:@var{port}
338 @subsection HTTP Cookies
340 Some HTTP requests will be denied unless cookie values are passed in with the
341 request. The @option{cookies} option allows these cookies to be specified. At
342 the very least, each cookie must specify a value along with a path and domain.
343 HTTP requests that match both the domain and path will automatically include the
344 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
347 The required syntax to play a stream specifying a cookie is:
349 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
354 Icecast protocol (stream to Icecast servers)
356 This protocol accepts the following options:
360 Set the stream genre.
365 @item ice_description
366 Set the stream description.
369 Set the stream website URL.
372 Set if the stream should be public.
373 The default is 0 (not public).
376 Override the User-Agent header. If not specified a string of the form
377 "Lavf/<version>" will be used.
380 Set the Icecast mountpoint password.
383 Set the stream content type. This must be set if it is different from
387 This enables support for Icecast versions < 2.4.0, that do not support the
388 HTTP PUT method but the SOURCE method.
393 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
398 MMS (Microsoft Media Server) protocol over TCP.
402 MMS (Microsoft Media Server) protocol over HTTP.
404 The required syntax is:
406 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
413 Computes the MD5 hash of the data to be written, and on close writes
414 this to the designated output or stdout if none is specified. It can
415 be used to test muxers without writing an actual file.
417 Some examples follow.
419 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
420 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
422 # Write the MD5 hash of the encoded AVI file to stdout.
423 ffmpeg -i input.flv -f avi -y md5:
426 Note that some formats (typically MOV) require the output protocol to
427 be seekable, so they will fail with the MD5 output protocol.
431 UNIX pipe access protocol.
433 Read and write from UNIX pipes.
435 The accepted syntax is:
440 @var{number} is the number corresponding to the file descriptor of the
441 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
442 is not specified, by default the stdout file descriptor will be used
443 for writing, stdin for reading.
445 For example to read from stdin with @command{ffmpeg}:
447 cat test.wav | ffmpeg -i pipe:0
448 # ...this is the same as...
449 cat test.wav | ffmpeg -i pipe:
452 For writing to stdout with @command{ffmpeg}:
454 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
455 # ...this is the same as...
456 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
459 This protocol accepts the following options:
463 Set I/O operation maximum block size, in bytes. Default value is
464 @code{INT_MAX}, which results in not limiting the requested block size.
465 Setting this value reasonably low improves user termination request reaction
466 time, which is valuable if data transmission is slow.
469 Note that some formats (typically MOV), require the output protocol to
470 be seekable, so they will fail with the pipe output protocol.
474 Real-Time Messaging Protocol.
476 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
477 content across a TCP/IP network.
479 The required syntax is:
481 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
484 The accepted parameters are:
488 An optional username (mostly for publishing).
491 An optional password (mostly for publishing).
494 The address of the RTMP server.
497 The number of the TCP port to use (by default is 1935).
500 It is the name of the application to access. It usually corresponds to
501 the path where the application is installed on the RTMP server
502 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
503 the value parsed from the URI through the @code{rtmp_app} option, too.
506 It is the path or name of the resource to play with reference to the
507 application specified in @var{app}, may be prefixed by "mp4:". You
508 can override the value parsed from the URI through the @code{rtmp_playpath}
512 Act as a server, listening for an incoming connection.
515 Maximum time to wait for the incoming connection. Implies listen.
518 Additionally, the following parameters can be set via command line options
519 (or in code via @code{AVOption}s):
523 Name of application to connect on the RTMP server. This option
524 overrides the parameter specified in the URI.
527 Set the client buffer time in milliseconds. The default is 3000.
530 Extra arbitrary AMF connection parameters, parsed from a string,
531 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
532 Each value is prefixed by a single character denoting the type,
533 B for Boolean, N for number, S for string, O for object, or Z for null,
534 followed by a colon. For Booleans the data must be either 0 or 1 for
535 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
536 1 to end or begin an object, respectively. Data items in subobjects may
537 be named, by prefixing the type with 'N' and specifying the name before
538 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
539 times to construct arbitrary AMF sequences.
542 Version of the Flash plugin used to run the SWF player. The default
543 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
544 <libavformat version>).)
546 @item rtmp_flush_interval
547 Number of packets flushed in the same request (RTMPT only). The default
551 Specify that the media is a live stream. No resuming or seeking in
552 live streams is possible. The default value is @code{any}, which means the
553 subscriber first tries to play the live stream specified in the
554 playpath. If a live stream of that name is not found, it plays the
555 recorded stream. The other possible values are @code{live} and
559 URL of the web page in which the media was embedded. By default no
563 Stream identifier to play or to publish. This option overrides the
564 parameter specified in the URI.
567 Name of live stream to subscribe to. By default no value will be sent.
568 It is only sent if the option is specified or if rtmp_live
572 SHA256 hash of the decompressed SWF file (32 bytes).
575 Size of the decompressed SWF file, required for SWFVerification.
578 URL of the SWF player for the media. By default no value will be sent.
581 URL to player swf file, compute hash/size automatically.
584 URL of the target stream. Defaults to proto://host[:port]/app.
588 For example to read with @command{ffplay} a multimedia resource named
589 "sample" from the application "vod" from an RTMP server "myserver":
591 ffplay rtmp://myserver/vod/sample
594 To publish to a password protected server, passing the playpath and
595 app names separately:
597 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
602 Encrypted Real-Time Messaging Protocol.
604 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
605 streaming multimedia content within standard cryptographic primitives,
606 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
611 Real-Time Messaging Protocol over a secure SSL connection.
613 The Real-Time Messaging Protocol (RTMPS) is used for streaming
614 multimedia content across an encrypted connection.
618 Real-Time Messaging Protocol tunneled through HTTP.
620 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
621 for streaming multimedia content within HTTP requests to traverse
626 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
628 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
629 is used for streaming multimedia content within HTTP requests to traverse
634 Real-Time Messaging Protocol tunneled through HTTPS.
636 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
637 for streaming multimedia content within HTTPS requests to traverse
640 @section libsmbclient
642 libsmbclient permits one to manipulate CIFS/SMB network resources.
644 Following syntax is required.
647 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
650 This protocol accepts the following options.
654 Set timeout in miliseconds of socket I/O operations used by the underlying
655 low level operation. By default it is set to -1, which means that the timeout
659 Truncate existing files on write, if set to 1. A value of 0 prevents
660 truncating. Default value is 1.
663 Set the workgroup used for making connections. By default workgroup is not specified.
667 For more information see: @url{http://www.samba.org/}.
671 Secure File Transfer Protocol via libssh
673 Read from or write to remote resources using SFTP protocol.
675 Following syntax is required.
678 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
681 This protocol accepts the following options.
685 Set timeout of socket I/O operations used by the underlying low level
686 operation. By default it is set to -1, which means that the timeout
690 Truncate existing files on write, if set to 1. A value of 0 prevents
691 truncating. Default value is 1.
694 Specify the path of the file containing private key to use during authorization.
695 By default libssh searches for keys in the @file{~/.ssh/} directory.
699 Example: Play a file stored on remote server.
702 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
705 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
707 Real-Time Messaging Protocol and its variants supported through
710 Requires the presence of the librtmp headers and library during
711 configuration. You need to explicitly configure the build with
712 "--enable-librtmp". If enabled this will replace the native RTMP
715 This protocol provides most client functions and a few server
716 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
717 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
718 variants of these encrypted types (RTMPTE, RTMPTS).
720 The required syntax is:
722 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
725 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
726 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
727 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
728 meaning as specified for the RTMP native protocol.
729 @var{options} contains a list of space-separated options of the form
732 See the librtmp manual page (man 3 librtmp) for more information.
734 For example, to stream a file in real-time to an RTMP server using
737 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
740 To play the same stream using @command{ffplay}:
742 ffplay "rtmp://myserver/live/mystream live=1"
747 Real-time Transport Protocol.
749 The required syntax for an RTP URL is:
750 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
752 @var{port} specifies the RTP port to use.
754 The following URL options are supported:
759 Set the TTL (Time-To-Live) value (for multicast only).
761 @item rtcpport=@var{n}
762 Set the remote RTCP port to @var{n}.
764 @item localrtpport=@var{n}
765 Set the local RTP port to @var{n}.
767 @item localrtcpport=@var{n}'
768 Set the local RTCP port to @var{n}.
770 @item pkt_size=@var{n}
771 Set max packet size (in bytes) to @var{n}.
774 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
777 @item sources=@var{ip}[,@var{ip}]
778 List allowed source IP addresses.
780 @item block=@var{ip}[,@var{ip}]
781 List disallowed (blocked) source IP addresses.
783 @item write_to_source=0|1
784 Send packets to the source address of the latest received packet (if
785 set to 1) or to a default remote address (if set to 0).
787 @item localport=@var{n}
788 Set the local RTP port to @var{n}.
790 This is a deprecated option. Instead, @option{localrtpport} should be
800 If @option{rtcpport} is not set the RTCP port will be set to the RTP
804 If @option{localrtpport} (the local RTP port) is not set any available
805 port will be used for the local RTP and RTCP ports.
808 If @option{localrtcpport} (the local RTCP port) is not set it will be
809 set to the local RTP port value plus 1.
814 Real-Time Streaming Protocol.
816 RTSP is not technically a protocol handler in libavformat, it is a demuxer
817 and muxer. The demuxer supports both normal RTSP (with data transferred
818 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
819 data transferred over RDT).
821 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
822 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
823 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
825 The required syntax for a RTSP url is:
827 rtsp://@var{hostname}[:@var{port}]/@var{path}
830 Options can be set on the @command{ffmpeg}/@command{ffplay} command
831 line, or set in code via @code{AVOption}s or in
832 @code{avformat_open_input}.
834 The following options are supported.
838 Do not start playing the stream immediately if set to 1. Default value
842 Set RTSP transport protocols.
844 It accepts the following values:
847 Use UDP as lower transport protocol.
850 Use TCP (interleaving within the RTSP control channel) as lower
854 Use UDP multicast as lower transport protocol.
857 Use HTTP tunneling as lower transport protocol, which is useful for
861 Multiple lower transport protocols may be specified, in that case they are
862 tried one at a time (if the setup of one fails, the next one is tried).
863 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
868 The following values are accepted:
871 Accept packets only from negotiated peer address and port.
873 Act as a server, listening for an incoming connection.
875 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
878 Default value is @samp{none}.
880 @item allowed_media_types
881 Set media types to accept from the server.
883 The following flags are accepted:
890 By default it accepts all media types.
893 Set minimum local UDP port. Default value is 5000.
896 Set maximum local UDP port. Default value is 65000.
899 Set maximum timeout (in seconds) to wait for incoming connections.
901 A value of -1 means infinite (default). This option implies the
902 @option{rtsp_flags} set to @samp{listen}.
904 @item reorder_queue_size
905 Set number of packets to buffer for handling of reordered packets.
908 Set socket TCP I/O timeout in microseconds.
911 Override User-Agent header. If not specified, it defaults to the
912 libavformat identifier string.
915 When receiving data over UDP, the demuxer tries to reorder received packets
916 (since they may arrive out of order, or packets may get lost totally). This
917 can be disabled by setting the maximum demuxing delay to zero (via
918 the @code{max_delay} field of AVFormatContext).
920 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
921 streams to display can be chosen with @code{-vst} @var{n} and
922 @code{-ast} @var{n} for video and audio respectively, and can be switched
923 on the fly by pressing @code{v} and @code{a}.
927 The following examples all make use of the @command{ffplay} and
928 @command{ffmpeg} tools.
932 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
934 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
938 Watch a stream tunneled over HTTP:
940 ffplay -rtsp_transport http rtsp://server/video.mp4
944 Send a stream in realtime to a RTSP server, for others to watch:
946 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
950 Receive a stream in realtime:
952 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
958 Session Announcement Protocol (RFC 2974). This is not technically a
959 protocol handler in libavformat, it is a muxer and demuxer.
960 It is used for signalling of RTP streams, by announcing the SDP for the
961 streams regularly on a separate port.
965 The syntax for a SAP url given to the muxer is:
967 sap://@var{destination}[:@var{port}][?@var{options}]
970 The RTP packets are sent to @var{destination} on port @var{port},
971 or to port 5004 if no port is specified.
972 @var{options} is a @code{&}-separated list. The following options
977 @item announce_addr=@var{address}
978 Specify the destination IP address for sending the announcements to.
979 If omitted, the announcements are sent to the commonly used SAP
980 announcement multicast address 224.2.127.254 (sap.mcast.net), or
981 ff0e::2:7ffe if @var{destination} is an IPv6 address.
983 @item announce_port=@var{port}
984 Specify the port to send the announcements on, defaults to
985 9875 if not specified.
988 Specify the time to live value for the announcements and RTP packets,
991 @item same_port=@var{0|1}
992 If set to 1, send all RTP streams on the same port pair. If zero (the
993 default), all streams are sent on unique ports, with each stream on a
994 port 2 numbers higher than the previous.
995 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
996 The RTP stack in libavformat for receiving requires all streams to be sent
1000 Example command lines follow.
1002 To broadcast a stream on the local subnet, for watching in VLC:
1005 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1008 Similarly, for watching in @command{ffplay}:
1011 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1014 And for watching in @command{ffplay}, over IPv6:
1017 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1022 The syntax for a SAP url given to the demuxer is:
1024 sap://[@var{address}][:@var{port}]
1027 @var{address} is the multicast address to listen for announcements on,
1028 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1029 is the port that is listened on, 9875 if omitted.
1031 The demuxers listens for announcements on the given address and port.
1032 Once an announcement is received, it tries to receive that particular stream.
1034 Example command lines follow.
1036 To play back the first stream announced on the normal SAP multicast address:
1042 To play back the first stream announced on one the default IPv6 SAP multicast address:
1045 ffplay sap://[ff0e::2:7ffe]
1050 Stream Control Transmission Protocol.
1052 The accepted URL syntax is:
1054 sctp://@var{host}:@var{port}[?@var{options}]
1057 The protocol accepts the following options:
1060 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1063 Set the maximum number of streams. By default no limit is set.
1068 Secure Real-time Transport Protocol.
1070 The accepted options are:
1073 @item srtp_out_suite
1074 Select input and output encoding suites.
1078 @item AES_CM_128_HMAC_SHA1_80
1079 @item SRTP_AES128_CM_HMAC_SHA1_80
1080 @item AES_CM_128_HMAC_SHA1_32
1081 @item SRTP_AES128_CM_HMAC_SHA1_32
1084 @item srtp_in_params
1085 @item srtp_out_params
1086 Set input and output encoding parameters, which are expressed by a
1087 base64-encoded representation of a binary block. The first 16 bytes of
1088 this binary block are used as master key, the following 14 bytes are
1089 used as master salt.
1094 Virtually extract a segment of a file or another stream.
1095 The underlying stream must be seekable.
1100 Start offset of the extracted segment, in bytes.
1102 End offset of the extracted segment, in bytes.
1107 Extract a chapter from a DVD VOB file (start and end sectors obtained
1108 externally and multiplied by 2048):
1110 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1113 Play an AVI file directly from a TAR archive:
1115 subfile,,start,183241728,end,366490624,,:archive.tar
1120 Transmission Control Protocol.
1122 The required syntax for a TCP url is:
1124 tcp://@var{hostname}:@var{port}[?@var{options}]
1127 @var{options} contains a list of &-separated options of the form
1128 @var{key}=@var{val}.
1130 The list of supported options follows.
1133 @item listen=@var{1|0}
1134 Listen for an incoming connection. Default value is 0.
1136 @item timeout=@var{microseconds}
1137 Set raise error timeout, expressed in microseconds.
1139 This option is only relevant in read mode: if no data arrived in more
1140 than this time interval, raise error.
1142 @item listen_timeout=@var{milliseconds}
1143 Set listen timeout, expressed in milliseconds.
1146 The following example shows how to setup a listening TCP connection
1147 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1149 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1150 ffplay tcp://@var{hostname}:@var{port}
1155 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1157 The required syntax for a TLS/SSL url is:
1159 tls://@var{hostname}:@var{port}[?@var{options}]
1162 The following parameters can be set via command line options
1163 (or in code via @code{AVOption}s):
1167 @item ca_file, cafile=@var{filename}
1168 A file containing certificate authority (CA) root certificates to treat
1169 as trusted. If the linked TLS library contains a default this might not
1170 need to be specified for verification to work, but not all libraries and
1171 setups have defaults built in.
1172 The file must be in OpenSSL PEM format.
1174 @item tls_verify=@var{1|0}
1175 If enabled, try to verify the peer that we are communicating with.
1176 Note, if using OpenSSL, this currently only makes sure that the
1177 peer certificate is signed by one of the root certificates in the CA
1178 database, but it does not validate that the certificate actually
1179 matches the host name we are trying to connect to. (With GnuTLS,
1180 the host name is validated as well.)
1182 This is disabled by default since it requires a CA database to be
1183 provided by the caller in many cases.
1185 @item cert_file, cert=@var{filename}
1186 A file containing a certificate to use in the handshake with the peer.
1187 (When operating as server, in listen mode, this is more often required
1188 by the peer, while client certificates only are mandated in certain
1191 @item key_file, key=@var{filename}
1192 A file containing the private key for the certificate.
1194 @item listen=@var{1|0}
1195 If enabled, listen for connections on the provided port, and assume
1196 the server role in the handshake instead of the client role.
1200 Example command lines:
1202 To create a TLS/SSL server that serves an input stream.
1205 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1208 To play back a stream from the TLS/SSL server using @command{ffplay}:
1211 ffplay tls://@var{hostname}:@var{port}
1216 User Datagram Protocol.
1218 The required syntax for an UDP URL is:
1220 udp://@var{hostname}:@var{port}[?@var{options}]
1223 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1225 In case threading is enabled on the system, a circular buffer is used
1226 to store the incoming data, which allows one to reduce loss of data due to
1227 UDP socket buffer overruns. The @var{fifo_size} and
1228 @var{overrun_nonfatal} options are related to this buffer.
1230 The list of supported options follows.
1233 @item buffer_size=@var{size}
1234 Set the UDP maximum socket buffer size in bytes. This is used to set either
1235 the receive or send buffer size, depending on what the socket is used for.
1236 Default is 64KB. See also @var{fifo_size}.
1238 @item localport=@var{port}
1239 Override the local UDP port to bind with.
1241 @item localaddr=@var{addr}
1242 Choose the local IP address. This is useful e.g. if sending multicast
1243 and the host has multiple interfaces, where the user can choose
1244 which interface to send on by specifying the IP address of that interface.
1246 @item pkt_size=@var{size}
1247 Set the size in bytes of UDP packets.
1249 @item reuse=@var{1|0}
1250 Explicitly allow or disallow reusing UDP sockets.
1253 Set the time to live value (for multicast only).
1255 @item connect=@var{1|0}
1256 Initialize the UDP socket with @code{connect()}. In this case, the
1257 destination address can't be changed with ff_udp_set_remote_url later.
1258 If the destination address isn't known at the start, this option can
1259 be specified in ff_udp_set_remote_url, too.
1260 This allows finding out the source address for the packets with getsockname,
1261 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1262 unreachable" is received.
1263 For receiving, this gives the benefit of only receiving packets from
1264 the specified peer address/port.
1266 @item sources=@var{address}[,@var{address}]
1267 Only receive packets sent to the multicast group from one of the
1268 specified sender IP addresses.
1270 @item block=@var{address}[,@var{address}]
1271 Ignore packets sent to the multicast group from the specified
1272 sender IP addresses.
1274 @item fifo_size=@var{units}
1275 Set the UDP receiving circular buffer size, expressed as a number of
1276 packets with size of 188 bytes. If not specified defaults to 7*4096.
1278 @item overrun_nonfatal=@var{1|0}
1279 Survive in case of UDP receiving circular buffer overrun. Default
1282 @item timeout=@var{microseconds}
1283 Set raise error timeout, expressed in microseconds.
1285 This option is only relevant in read mode: if no data arrived in more
1286 than this time interval, raise error.
1288 @item broadcast=@var{1|0}
1289 Explicitly allow or disallow UDP broadcasting.
1291 Note that broadcasting may not work properly on networks having
1292 a broadcast storm protection.
1295 @subsection Examples
1299 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1301 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1305 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1306 sized UDP packets, using a large input buffer:
1308 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1312 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1314 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1322 The required syntax for a Unix socket URL is:
1325 unix://@var{filepath}
1328 The following parameters can be set via command line options
1329 (or in code via @code{AVOption}s):
1335 Create the Unix socket in listening mode.
1338 @c man end PROTOCOLS