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 useful
265 for live / endless streams.
267 @item reconnect_streamed
268 If set then even streamed/non seekable streams will be reconnected on errors.
270 @item reconnect_delay_max
271 Sets the maximum delay in seconds after which to give up reconnecting
274 Export the MIME type.
277 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
278 supports this, the metadata has to be retrieved by the application by reading
279 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
282 @item icy_metadata_headers
283 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
284 headers, separated by newline characters.
286 @item icy_metadata_packet
287 If the server supports ICY metadata, and @option{icy} was set to 1, this
288 contains the last non-empty metadata packet sent by the server. It should be
289 polled in regular intervals by applications interested in mid-stream metadata
293 Set the cookies to be sent in future requests. The format of each cookie is the
294 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
295 delimited by a newline character.
298 Set initial byte offset.
301 Try to limit the request to bytes preceding this offset.
304 When used as a client option it sets the HTTP method for the request.
306 When used as a server option it sets the HTTP method that is going to be
307 expected from the client(s).
308 If the expected and the received HTTP method do not match the client will
309 be given a Bad Request response.
310 When unset the HTTP method is not checked for now. This will be replaced by
311 autodetection in the future.
314 If set to 1 enables experimental HTTP server. This can be used to send data when
315 used as an output option, or read data from a client with HTTP POST when used as
317 If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
318 in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
320 # Server side (sending):
321 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
323 # Client side (receiving):
324 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
326 # Client can also be done with wget:
327 wget http://@var{server}:@var{port} -O somefile.ogg
329 # Server side (receiving):
330 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
332 # Client side (sending):
333 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
335 # Client can also be done with wget:
336 wget --post-file=somefile.ogg http://@var{server}:@var{port}
341 @subsection HTTP Cookies
343 Some HTTP requests will be denied unless cookie values are passed in with the
344 request. The @option{cookies} option allows these cookies to be specified. At
345 the very least, each cookie must specify a value along with a path and domain.
346 HTTP requests that match both the domain and path will automatically include the
347 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
350 The required syntax to play a stream specifying a cookie is:
352 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
357 Icecast protocol (stream to Icecast servers)
359 This protocol accepts the following options:
363 Set the stream genre.
368 @item ice_description
369 Set the stream description.
372 Set the stream website URL.
375 Set if the stream should be public.
376 The default is 0 (not public).
379 Override the User-Agent header. If not specified a string of the form
380 "Lavf/<version>" will be used.
383 Set the Icecast mountpoint password.
386 Set the stream content type. This must be set if it is different from
390 This enables support for Icecast versions < 2.4.0, that do not support the
391 HTTP PUT method but the SOURCE method.
396 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
401 MMS (Microsoft Media Server) protocol over TCP.
405 MMS (Microsoft Media Server) protocol over HTTP.
407 The required syntax is:
409 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
416 Computes the MD5 hash of the data to be written, and on close writes
417 this to the designated output or stdout if none is specified. It can
418 be used to test muxers without writing an actual file.
420 Some examples follow.
422 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
423 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
425 # Write the MD5 hash of the encoded AVI file to stdout.
426 ffmpeg -i input.flv -f avi -y md5:
429 Note that some formats (typically MOV) require the output protocol to
430 be seekable, so they will fail with the MD5 output protocol.
434 UNIX pipe access protocol.
436 Read and write from UNIX pipes.
438 The accepted syntax is:
443 @var{number} is the number corresponding to the file descriptor of the
444 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
445 is not specified, by default the stdout file descriptor will be used
446 for writing, stdin for reading.
448 For example to read from stdin with @command{ffmpeg}:
450 cat test.wav | ffmpeg -i pipe:0
451 # ...this is the same as...
452 cat test.wav | ffmpeg -i pipe:
455 For writing to stdout with @command{ffmpeg}:
457 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
458 # ...this is the same as...
459 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
462 This protocol accepts the following options:
466 Set I/O operation maximum block size, in bytes. Default value is
467 @code{INT_MAX}, which results in not limiting the requested block size.
468 Setting this value reasonably low improves user termination request reaction
469 time, which is valuable if data transmission is slow.
472 Note that some formats (typically MOV), require the output protocol to
473 be seekable, so they will fail with the pipe output protocol.
477 Real-Time Messaging Protocol.
479 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
480 content across a TCP/IP network.
482 The required syntax is:
484 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
487 The accepted parameters are:
491 An optional username (mostly for publishing).
494 An optional password (mostly for publishing).
497 The address of the RTMP server.
500 The number of the TCP port to use (by default is 1935).
503 It is the name of the application to access. It usually corresponds to
504 the path where the application is installed on the RTMP server
505 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
506 the value parsed from the URI through the @code{rtmp_app} option, too.
509 It is the path or name of the resource to play with reference to the
510 application specified in @var{app}, may be prefixed by "mp4:". You
511 can override the value parsed from the URI through the @code{rtmp_playpath}
515 Act as a server, listening for an incoming connection.
518 Maximum time to wait for the incoming connection. Implies listen.
521 Additionally, the following parameters can be set via command line options
522 (or in code via @code{AVOption}s):
526 Name of application to connect on the RTMP server. This option
527 overrides the parameter specified in the URI.
530 Set the client buffer time in milliseconds. The default is 3000.
533 Extra arbitrary AMF connection parameters, parsed from a string,
534 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
535 Each value is prefixed by a single character denoting the type,
536 B for Boolean, N for number, S for string, O for object, or Z for null,
537 followed by a colon. For Booleans the data must be either 0 or 1 for
538 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
539 1 to end or begin an object, respectively. Data items in subobjects may
540 be named, by prefixing the type with 'N' and specifying the name before
541 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
542 times to construct arbitrary AMF sequences.
545 Version of the Flash plugin used to run the SWF player. The default
546 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
547 <libavformat version>).)
549 @item rtmp_flush_interval
550 Number of packets flushed in the same request (RTMPT only). The default
554 Specify that the media is a live stream. No resuming or seeking in
555 live streams is possible. The default value is @code{any}, which means the
556 subscriber first tries to play the live stream specified in the
557 playpath. If a live stream of that name is not found, it plays the
558 recorded stream. The other possible values are @code{live} and
562 URL of the web page in which the media was embedded. By default no
566 Stream identifier to play or to publish. This option overrides the
567 parameter specified in the URI.
570 Name of live stream to subscribe to. By default no value will be sent.
571 It is only sent if the option is specified or if rtmp_live
575 SHA256 hash of the decompressed SWF file (32 bytes).
578 Size of the decompressed SWF file, required for SWFVerification.
581 URL of the SWF player for the media. By default no value will be sent.
584 URL to player swf file, compute hash/size automatically.
587 URL of the target stream. Defaults to proto://host[:port]/app.
591 For example to read with @command{ffplay} a multimedia resource named
592 "sample" from the application "vod" from an RTMP server "myserver":
594 ffplay rtmp://myserver/vod/sample
597 To publish to a password protected server, passing the playpath and
598 app names separately:
600 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
605 Encrypted Real-Time Messaging Protocol.
607 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
608 streaming multimedia content within standard cryptographic primitives,
609 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
614 Real-Time Messaging Protocol over a secure SSL connection.
616 The Real-Time Messaging Protocol (RTMPS) is used for streaming
617 multimedia content across an encrypted connection.
621 Real-Time Messaging Protocol tunneled through HTTP.
623 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
624 for streaming multimedia content within HTTP requests to traverse
629 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
631 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
632 is used for streaming multimedia content within HTTP requests to traverse
637 Real-Time Messaging Protocol tunneled through HTTPS.
639 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
640 for streaming multimedia content within HTTPS requests to traverse
643 @section libsmbclient
645 libsmbclient permits one to manipulate CIFS/SMB network resources.
647 Following syntax is required.
650 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
653 This protocol accepts the following options.
657 Set timeout in miliseconds of socket I/O operations used by the underlying
658 low level operation. By default it is set to -1, which means that the timeout
662 Truncate existing files on write, if set to 1. A value of 0 prevents
663 truncating. Default value is 1.
666 Set the workgroup used for making connections. By default workgroup is not specified.
670 For more information see: @url{http://www.samba.org/}.
674 Secure File Transfer Protocol via libssh
676 Read from or write to remote resources using SFTP protocol.
678 Following syntax is required.
681 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
684 This protocol accepts the following options.
688 Set timeout of socket I/O operations used by the underlying low level
689 operation. By default it is set to -1, which means that the timeout
693 Truncate existing files on write, if set to 1. A value of 0 prevents
694 truncating. Default value is 1.
697 Specify the path of the file containing private key to use during authorization.
698 By default libssh searches for keys in the @file{~/.ssh/} directory.
702 Example: Play a file stored on remote server.
705 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
708 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
710 Real-Time Messaging Protocol and its variants supported through
713 Requires the presence of the librtmp headers and library during
714 configuration. You need to explicitly configure the build with
715 "--enable-librtmp". If enabled this will replace the native RTMP
718 This protocol provides most client functions and a few server
719 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
720 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
721 variants of these encrypted types (RTMPTE, RTMPTS).
723 The required syntax is:
725 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
728 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
729 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
730 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
731 meaning as specified for the RTMP native protocol.
732 @var{options} contains a list of space-separated options of the form
735 See the librtmp manual page (man 3 librtmp) for more information.
737 For example, to stream a file in real-time to an RTMP server using
740 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
743 To play the same stream using @command{ffplay}:
745 ffplay "rtmp://myserver/live/mystream live=1"
750 Real-time Transport Protocol.
752 The required syntax for an RTP URL is:
753 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
755 @var{port} specifies the RTP port to use.
757 The following URL options are supported:
762 Set the TTL (Time-To-Live) value (for multicast only).
764 @item rtcpport=@var{n}
765 Set the remote RTCP port to @var{n}.
767 @item localrtpport=@var{n}
768 Set the local RTP port to @var{n}.
770 @item localrtcpport=@var{n}'
771 Set the local RTCP port to @var{n}.
773 @item pkt_size=@var{n}
774 Set max packet size (in bytes) to @var{n}.
777 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
780 @item sources=@var{ip}[,@var{ip}]
781 List allowed source IP addresses.
783 @item block=@var{ip}[,@var{ip}]
784 List disallowed (blocked) source IP addresses.
786 @item write_to_source=0|1
787 Send packets to the source address of the latest received packet (if
788 set to 1) or to a default remote address (if set to 0).
790 @item localport=@var{n}
791 Set the local RTP port to @var{n}.
793 This is a deprecated option. Instead, @option{localrtpport} should be
803 If @option{rtcpport} is not set the RTCP port will be set to the RTP
807 If @option{localrtpport} (the local RTP port) is not set any available
808 port will be used for the local RTP and RTCP ports.
811 If @option{localrtcpport} (the local RTCP port) is not set it will be
812 set to the local RTP port value plus 1.
817 Real-Time Streaming Protocol.
819 RTSP is not technically a protocol handler in libavformat, it is a demuxer
820 and muxer. The demuxer supports both normal RTSP (with data transferred
821 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
822 data transferred over RDT).
824 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
825 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
826 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
828 The required syntax for a RTSP url is:
830 rtsp://@var{hostname}[:@var{port}]/@var{path}
833 Options can be set on the @command{ffmpeg}/@command{ffplay} command
834 line, or set in code via @code{AVOption}s or in
835 @code{avformat_open_input}.
837 The following options are supported.
841 Do not start playing the stream immediately if set to 1. Default value
845 Set RTSP transport protocols.
847 It accepts the following values:
850 Use UDP as lower transport protocol.
853 Use TCP (interleaving within the RTSP control channel) as lower
857 Use UDP multicast as lower transport protocol.
860 Use HTTP tunneling as lower transport protocol, which is useful for
864 Multiple lower transport protocols may be specified, in that case they are
865 tried one at a time (if the setup of one fails, the next one is tried).
866 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
871 The following values are accepted:
874 Accept packets only from negotiated peer address and port.
876 Act as a server, listening for an incoming connection.
878 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
881 Default value is @samp{none}.
883 @item allowed_media_types
884 Set media types to accept from the server.
886 The following flags are accepted:
893 By default it accepts all media types.
896 Set minimum local UDP port. Default value is 5000.
899 Set maximum local UDP port. Default value is 65000.
902 Set maximum timeout (in seconds) to wait for incoming connections.
904 A value of -1 means infinite (default). This option implies the
905 @option{rtsp_flags} set to @samp{listen}.
907 @item reorder_queue_size
908 Set number of packets to buffer for handling of reordered packets.
911 Set socket TCP I/O timeout in microseconds.
914 Override User-Agent header. If not specified, it defaults to the
915 libavformat identifier string.
918 When receiving data over UDP, the demuxer tries to reorder received packets
919 (since they may arrive out of order, or packets may get lost totally). This
920 can be disabled by setting the maximum demuxing delay to zero (via
921 the @code{max_delay} field of AVFormatContext).
923 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
924 streams to display can be chosen with @code{-vst} @var{n} and
925 @code{-ast} @var{n} for video and audio respectively, and can be switched
926 on the fly by pressing @code{v} and @code{a}.
930 The following examples all make use of the @command{ffplay} and
931 @command{ffmpeg} tools.
935 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
937 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
941 Watch a stream tunneled over HTTP:
943 ffplay -rtsp_transport http rtsp://server/video.mp4
947 Send a stream in realtime to a RTSP server, for others to watch:
949 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
953 Receive a stream in realtime:
955 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
961 Session Announcement Protocol (RFC 2974). This is not technically a
962 protocol handler in libavformat, it is a muxer and demuxer.
963 It is used for signalling of RTP streams, by announcing the SDP for the
964 streams regularly on a separate port.
968 The syntax for a SAP url given to the muxer is:
970 sap://@var{destination}[:@var{port}][?@var{options}]
973 The RTP packets are sent to @var{destination} on port @var{port},
974 or to port 5004 if no port is specified.
975 @var{options} is a @code{&}-separated list. The following options
980 @item announce_addr=@var{address}
981 Specify the destination IP address for sending the announcements to.
982 If omitted, the announcements are sent to the commonly used SAP
983 announcement multicast address 224.2.127.254 (sap.mcast.net), or
984 ff0e::2:7ffe if @var{destination} is an IPv6 address.
986 @item announce_port=@var{port}
987 Specify the port to send the announcements on, defaults to
988 9875 if not specified.
991 Specify the time to live value for the announcements and RTP packets,
994 @item same_port=@var{0|1}
995 If set to 1, send all RTP streams on the same port pair. If zero (the
996 default), all streams are sent on unique ports, with each stream on a
997 port 2 numbers higher than the previous.
998 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
999 The RTP stack in libavformat for receiving requires all streams to be sent
1003 Example command lines follow.
1005 To broadcast a stream on the local subnet, for watching in VLC:
1008 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1011 Similarly, for watching in @command{ffplay}:
1014 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1017 And for watching in @command{ffplay}, over IPv6:
1020 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1025 The syntax for a SAP url given to the demuxer is:
1027 sap://[@var{address}][:@var{port}]
1030 @var{address} is the multicast address to listen for announcements on,
1031 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1032 is the port that is listened on, 9875 if omitted.
1034 The demuxers listens for announcements on the given address and port.
1035 Once an announcement is received, it tries to receive that particular stream.
1037 Example command lines follow.
1039 To play back the first stream announced on the normal SAP multicast address:
1045 To play back the first stream announced on one the default IPv6 SAP multicast address:
1048 ffplay sap://[ff0e::2:7ffe]
1053 Stream Control Transmission Protocol.
1055 The accepted URL syntax is:
1057 sctp://@var{host}:@var{port}[?@var{options}]
1060 The protocol accepts the following options:
1063 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1066 Set the maximum number of streams. By default no limit is set.
1071 Secure Real-time Transport Protocol.
1073 The accepted options are:
1076 @item srtp_out_suite
1077 Select input and output encoding suites.
1081 @item AES_CM_128_HMAC_SHA1_80
1082 @item SRTP_AES128_CM_HMAC_SHA1_80
1083 @item AES_CM_128_HMAC_SHA1_32
1084 @item SRTP_AES128_CM_HMAC_SHA1_32
1087 @item srtp_in_params
1088 @item srtp_out_params
1089 Set input and output encoding parameters, which are expressed by a
1090 base64-encoded representation of a binary block. The first 16 bytes of
1091 this binary block are used as master key, the following 14 bytes are
1092 used as master salt.
1097 Virtually extract a segment of a file or another stream.
1098 The underlying stream must be seekable.
1103 Start offset of the extracted segment, in bytes.
1105 End offset of the extracted segment, in bytes.
1110 Extract a chapter from a DVD VOB file (start and end sectors obtained
1111 externally and multiplied by 2048):
1113 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1116 Play an AVI file directly from a TAR archive:
1118 subfile,,start,183241728,end,366490624,,:archive.tar
1123 Transmission Control Protocol.
1125 The required syntax for a TCP url is:
1127 tcp://@var{hostname}:@var{port}[?@var{options}]
1130 @var{options} contains a list of &-separated options of the form
1131 @var{key}=@var{val}.
1133 The list of supported options follows.
1136 @item listen=@var{1|0}
1137 Listen for an incoming connection. Default value is 0.
1139 @item timeout=@var{microseconds}
1140 Set raise error timeout, expressed in microseconds.
1142 This option is only relevant in read mode: if no data arrived in more
1143 than this time interval, raise error.
1145 @item listen_timeout=@var{milliseconds}
1146 Set listen timeout, expressed in milliseconds.
1149 The following example shows how to setup a listening TCP connection
1150 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1152 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1153 ffplay tcp://@var{hostname}:@var{port}
1158 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1160 The required syntax for a TLS/SSL url is:
1162 tls://@var{hostname}:@var{port}[?@var{options}]
1165 The following parameters can be set via command line options
1166 (or in code via @code{AVOption}s):
1170 @item ca_file, cafile=@var{filename}
1171 A file containing certificate authority (CA) root certificates to treat
1172 as trusted. If the linked TLS library contains a default this might not
1173 need to be specified for verification to work, but not all libraries and
1174 setups have defaults built in.
1175 The file must be in OpenSSL PEM format.
1177 @item tls_verify=@var{1|0}
1178 If enabled, try to verify the peer that we are communicating with.
1179 Note, if using OpenSSL, this currently only makes sure that the
1180 peer certificate is signed by one of the root certificates in the CA
1181 database, but it does not validate that the certificate actually
1182 matches the host name we are trying to connect to. (With GnuTLS,
1183 the host name is validated as well.)
1185 This is disabled by default since it requires a CA database to be
1186 provided by the caller in many cases.
1188 @item cert_file, cert=@var{filename}
1189 A file containing a certificate to use in the handshake with the peer.
1190 (When operating as server, in listen mode, this is more often required
1191 by the peer, while client certificates only are mandated in certain
1194 @item key_file, key=@var{filename}
1195 A file containing the private key for the certificate.
1197 @item listen=@var{1|0}
1198 If enabled, listen for connections on the provided port, and assume
1199 the server role in the handshake instead of the client role.
1203 Example command lines:
1205 To create a TLS/SSL server that serves an input stream.
1208 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1211 To play back a stream from the TLS/SSL server using @command{ffplay}:
1214 ffplay tls://@var{hostname}:@var{port}
1219 User Datagram Protocol.
1221 The required syntax for an UDP URL is:
1223 udp://@var{hostname}:@var{port}[?@var{options}]
1226 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1228 In case threading is enabled on the system, a circular buffer is used
1229 to store the incoming data, which allows one to reduce loss of data due to
1230 UDP socket buffer overruns. The @var{fifo_size} and
1231 @var{overrun_nonfatal} options are related to this buffer.
1233 The list of supported options follows.
1236 @item buffer_size=@var{size}
1237 Set the UDP maximum socket buffer size in bytes. This is used to set either
1238 the receive or send buffer size, depending on what the socket is used for.
1239 Default is 64KB. See also @var{fifo_size}.
1241 @item localport=@var{port}
1242 Override the local UDP port to bind with.
1244 @item localaddr=@var{addr}
1245 Choose the local IP address. This is useful e.g. if sending multicast
1246 and the host has multiple interfaces, where the user can choose
1247 which interface to send on by specifying the IP address of that interface.
1249 @item pkt_size=@var{size}
1250 Set the size in bytes of UDP packets.
1252 @item reuse=@var{1|0}
1253 Explicitly allow or disallow reusing UDP sockets.
1256 Set the time to live value (for multicast only).
1258 @item connect=@var{1|0}
1259 Initialize the UDP socket with @code{connect()}. In this case, the
1260 destination address can't be changed with ff_udp_set_remote_url later.
1261 If the destination address isn't known at the start, this option can
1262 be specified in ff_udp_set_remote_url, too.
1263 This allows finding out the source address for the packets with getsockname,
1264 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1265 unreachable" is received.
1266 For receiving, this gives the benefit of only receiving packets from
1267 the specified peer address/port.
1269 @item sources=@var{address}[,@var{address}]
1270 Only receive packets sent to the multicast group from one of the
1271 specified sender IP addresses.
1273 @item block=@var{address}[,@var{address}]
1274 Ignore packets sent to the multicast group from the specified
1275 sender IP addresses.
1277 @item fifo_size=@var{units}
1278 Set the UDP receiving circular buffer size, expressed as a number of
1279 packets with size of 188 bytes. If not specified defaults to 7*4096.
1281 @item overrun_nonfatal=@var{1|0}
1282 Survive in case of UDP receiving circular buffer overrun. Default
1285 @item timeout=@var{microseconds}
1286 Set raise error timeout, expressed in microseconds.
1288 This option is only relevant in read mode: if no data arrived in more
1289 than this time interval, raise error.
1291 @item broadcast=@var{1|0}
1292 Explicitly allow or disallow UDP broadcasting.
1294 Note that broadcasting may not work properly on networks having
1295 a broadcast storm protection.
1298 @subsection Examples
1302 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1304 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1308 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1309 sized UDP packets, using a large input buffer:
1311 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1315 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1317 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1325 The required syntax for a Unix socket URL is:
1328 unix://@var{filepath}
1331 The following parameters can be set via command line options
1332 (or in code via @code{AVOption}s):
1338 Create the Unix socket in listening mode.
1341 @c man end PROTOCOLS