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.
26 The accepted options are:
36 Playlist to read (BDMV/PLAYLIST/?????.mpls)
42 Read longest playlist from BluRay mounted to /mnt/bluray:
47 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
49 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
54 Caching wrapper for input stream.
56 Cache the input stream to temporary file. It brings seeking capability to live streams.
64 Physical concatenation protocol.
66 Read and seek from many resources in sequence as if they were
69 A URL accepted by this protocol has the syntax:
71 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
74 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
75 resource to be concatenated, each one possibly specifying a distinct
78 For example to read a sequence of files @file{split1.mpeg},
79 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
82 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
85 Note that you may need to escape the character "|" which is special for
90 AES-encrypted stream reading protocol.
92 The accepted options are:
95 Set the AES decryption key binary block from given hexadecimal representation.
98 Set the AES decryption initialization vector binary block from given hexadecimal representation.
101 Accepted URL formats:
109 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
111 For example, to convert a GIF file given inline with @command{ffmpeg}:
113 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
118 File access protocol.
120 Read from or write to a file.
122 A file URL can have the form:
127 where @var{filename} is the path of the file to read.
129 An URL that does not have a protocol prefix will be assumed to be a
130 file URL. Depending on the build, an URL that looks like a Windows
131 path with the drive letter at the beginning will also be assumed to be
132 a file URL (usually not the case in builds for unix-like systems).
134 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
137 ffmpeg -i file:input.mpeg output.mpeg
140 This protocol accepts the following options:
144 Truncate existing files on write, if set to 1. A value of 0 prevents
145 truncating. Default value is 1.
148 Set I/O operation maximum block size, in bytes. Default value is
149 @code{INT_MAX}, which results in not limiting the requested block size.
150 Setting this value reasonably low improves user termination request reaction
151 time, which is valuable for files on slow medium.
156 FTP (File Transfer Protocol).
158 Read from or write to remote resources using FTP protocol.
160 Following syntax is required.
162 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
165 This protocol accepts the following options.
169 Set timeout in microseconds of socket I/O operations used by the underlying low level
170 operation. By default it is set to -1, which means that the timeout is
173 @item ftp-anonymous-password
174 Password used when login as anonymous user. Typically an e-mail address
177 @item ftp-write-seekable
178 Control seekability of connection during encoding. If set to 1 the
179 resource is supposed to be seekable, if set to 0 it is assumed not
180 to be seekable. Default value is 0.
183 NOTE: Protocol can be used as output, but it is recommended to not do
184 it, unless special care is taken (tests, customized server configuration
185 etc.). Different FTP servers behave in different way during seek
186 operation. ff* tools may produce incomplete content due to server limitations.
194 Read Apple HTTP Live Streaming compliant segmented stream as
195 a uniform one. The M3U8 playlists describing the segments can be
196 remote HTTP resources or local files, accessed using the standard
198 The nested protocol is declared by specifying
199 "+@var{proto}" after the hls URI scheme name, where @var{proto}
200 is either "file" or "http".
203 hls+http://host/path/to/remote/resource.m3u8
204 hls+file://path/to/local/resource.m3u8
207 Using this protocol is discouraged - the hls demuxer should work
208 just as well (if not, please report the issues) and is more complete.
209 To use the hls demuxer instead, simply use the direct URLs to the
214 HTTP (Hyper Text Transfer Protocol).
216 This protocol accepts the following options:
220 Control seekability of connection. If set to 1 the resource is
221 supposed to be seekable, if set to 0 it is assumed not to be seekable,
222 if set to -1 it will try to autodetect if it is seekable. Default
226 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
229 Set a specific content type for the POST messages.
232 Set custom HTTP headers, can override built in default headers. The
233 value must be a string encoding the headers.
235 @item multiple_requests
236 Use persistent connections if set to 1, default is 0.
239 Set custom HTTP post data.
243 Override the User-Agent header. If not specified the protocol will use a
244 string describing the libavformat build. ("Lavf/<version>")
247 Set timeout in microseconds of socket I/O operations used by the underlying low level
248 operation. By default it is set to -1, which means that the timeout is
252 Export the MIME type.
255 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
256 supports this, the metadata has to be retrieved by the application by reading
257 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
260 @item icy_metadata_headers
261 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
262 headers, separated by newline characters.
264 @item icy_metadata_packet
265 If the server supports ICY metadata, and @option{icy} was set to 1, this
266 contains the last non-empty metadata packet sent by the server. It should be
267 polled in regular intervals by applications interested in mid-stream metadata
271 Set the cookies to be sent in future requests. The format of each cookie is the
272 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
273 delimited by a newline character.
276 Set initial byte offset.
279 Try to limit the request to bytes preceding this offset.
282 When used as a client option it sets the HTTP method for the request.
284 When used as a server option it sets the HTTP method that is going to be
285 expected from the client(s).
286 If the expected and the received HTTP method do not match the client will
287 be given a Bad Request response.
288 When unset the HTTP method is not checked for now. This will be replaced by
289 autodetection in the future.
292 If set to 1 enables experimental HTTP server. This can be used to send data when
293 used as an output option, or read data from a client with HTTP POST when used as
296 # Server side (sending):
297 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
299 # Client side (receiving):
300 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
302 # Client can also be done with wget:
303 wget http://@var{server}:@var{port} -O somefile.ogg
305 # Server side (receiving):
306 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
308 # Client side (sending):
309 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
311 # Client can also be done with wget:
312 wget --post-file=somefile.ogg http://@var{server}:@var{port}
317 @subsection HTTP Cookies
319 Some HTTP requests will be denied unless cookie values are passed in with the
320 request. The @option{cookies} option allows these cookies to be specified. At
321 the very least, each cookie must specify a value along with a path and domain.
322 HTTP requests that match both the domain and path will automatically include the
323 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
326 The required syntax to play a stream specifying a cookie is:
328 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
333 Icecast protocol (stream to Icecast servers)
335 This protocol accepts the following options:
339 Set the stream genre.
344 @item ice_description
345 Set the stream description.
348 Set the stream website URL.
351 Set if the stream should be public.
352 The default is 0 (not public).
355 Override the User-Agent header. If not specified a string of the form
356 "Lavf/<version>" will be used.
359 Set the Icecast mountpoint password.
362 Set the stream content type. This must be set if it is different from
366 This enables support for Icecast versions < 2.4.0, that do not support the
367 HTTP PUT method but the SOURCE method.
372 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
377 MMS (Microsoft Media Server) protocol over TCP.
381 MMS (Microsoft Media Server) protocol over HTTP.
383 The required syntax is:
385 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
392 Computes the MD5 hash of the data to be written, and on close writes
393 this to the designated output or stdout if none is specified. It can
394 be used to test muxers without writing an actual file.
396 Some examples follow.
398 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
399 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
401 # Write the MD5 hash of the encoded AVI file to stdout.
402 ffmpeg -i input.flv -f avi -y md5:
405 Note that some formats (typically MOV) require the output protocol to
406 be seekable, so they will fail with the MD5 output protocol.
410 UNIX pipe access protocol.
412 Read and write from UNIX pipes.
414 The accepted syntax is:
419 @var{number} is the number corresponding to the file descriptor of the
420 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
421 is not specified, by default the stdout file descriptor will be used
422 for writing, stdin for reading.
424 For example to read from stdin with @command{ffmpeg}:
426 cat test.wav | ffmpeg -i pipe:0
427 # ...this is the same as...
428 cat test.wav | ffmpeg -i pipe:
431 For writing to stdout with @command{ffmpeg}:
433 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
434 # ...this is the same as...
435 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
438 This protocol accepts the following options:
442 Set I/O operation maximum block size, in bytes. Default value is
443 @code{INT_MAX}, which results in not limiting the requested block size.
444 Setting this value reasonably low improves user termination request reaction
445 time, which is valuable if data transmission is slow.
448 Note that some formats (typically MOV), require the output protocol to
449 be seekable, so they will fail with the pipe output protocol.
453 Real-Time Messaging Protocol.
455 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
456 content across a TCP/IP network.
458 The required syntax is:
460 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
463 The accepted parameters are:
467 An optional username (mostly for publishing).
470 An optional password (mostly for publishing).
473 The address of the RTMP server.
476 The number of the TCP port to use (by default is 1935).
479 It is the name of the application to access. It usually corresponds to
480 the path where the application is installed on the RTMP server
481 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
482 the value parsed from the URI through the @code{rtmp_app} option, too.
485 It is the path or name of the resource to play with reference to the
486 application specified in @var{app}, may be prefixed by "mp4:". You
487 can override the value parsed from the URI through the @code{rtmp_playpath}
491 Act as a server, listening for an incoming connection.
494 Maximum time to wait for the incoming connection. Implies listen.
497 Additionally, the following parameters can be set via command line options
498 (or in code via @code{AVOption}s):
502 Name of application to connect on the RTMP server. This option
503 overrides the parameter specified in the URI.
506 Set the client buffer time in milliseconds. The default is 3000.
509 Extra arbitrary AMF connection parameters, parsed from a string,
510 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
511 Each value is prefixed by a single character denoting the type,
512 B for Boolean, N for number, S for string, O for object, or Z for null,
513 followed by a colon. For Booleans the data must be either 0 or 1 for
514 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
515 1 to end or begin an object, respectively. Data items in subobjects may
516 be named, by prefixing the type with 'N' and specifying the name before
517 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
518 times to construct arbitrary AMF sequences.
521 Version of the Flash plugin used to run the SWF player. The default
522 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
523 <libavformat version>).)
525 @item rtmp_flush_interval
526 Number of packets flushed in the same request (RTMPT only). The default
530 Specify that the media is a live stream. No resuming or seeking in
531 live streams is possible. The default value is @code{any}, which means the
532 subscriber first tries to play the live stream specified in the
533 playpath. If a live stream of that name is not found, it plays the
534 recorded stream. The other possible values are @code{live} and
538 URL of the web page in which the media was embedded. By default no
542 Stream identifier to play or to publish. This option overrides the
543 parameter specified in the URI.
546 Name of live stream to subscribe to. By default no value will be sent.
547 It is only sent if the option is specified or if rtmp_live
551 SHA256 hash of the decompressed SWF file (32 bytes).
554 Size of the decompressed SWF file, required for SWFVerification.
557 URL of the SWF player for the media. By default no value will be sent.
560 URL to player swf file, compute hash/size automatically.
563 URL of the target stream. Defaults to proto://host[:port]/app.
567 For example to read with @command{ffplay} a multimedia resource named
568 "sample" from the application "vod" from an RTMP server "myserver":
570 ffplay rtmp://myserver/vod/sample
573 To publish to a password protected server, passing the playpath and
574 app names separately:
576 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
581 Encrypted Real-Time Messaging Protocol.
583 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
584 streaming multimedia content within standard cryptographic primitives,
585 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
590 Real-Time Messaging Protocol over a secure SSL connection.
592 The Real-Time Messaging Protocol (RTMPS) is used for streaming
593 multimedia content across an encrypted connection.
597 Real-Time Messaging Protocol tunneled through HTTP.
599 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
600 for streaming multimedia content within HTTP requests to traverse
605 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
607 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
608 is used for streaming multimedia content within HTTP requests to traverse
613 Real-Time Messaging Protocol tunneled through HTTPS.
615 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
616 for streaming multimedia content within HTTPS requests to traverse
619 @section libsmbclient
621 libsmbclient permits one to manipulate CIFS/SMB network resources.
623 Following syntax is required.
626 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
629 This protocol accepts the following options.
633 Set timeout in miliseconds of socket I/O operations used by the underlying
634 low level operation. By default it is set to -1, which means that the timeout
638 Truncate existing files on write, if set to 1. A value of 0 prevents
639 truncating. Default value is 1.
642 Set the workgroup used for making connections. By default workgroup is not specified.
646 For more information see: @url{http://www.samba.org/}.
650 Secure File Transfer Protocol via libssh
652 Read from or write to remote resources using SFTP protocol.
654 Following syntax is required.
657 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
660 This protocol accepts the following options.
664 Set timeout of socket I/O operations used by the underlying low level
665 operation. By default it is set to -1, which means that the timeout
669 Truncate existing files on write, if set to 1. A value of 0 prevents
670 truncating. Default value is 1.
673 Specify the path of the file containing private key to use during authorization.
674 By default libssh searches for keys in the @file{~/.ssh/} directory.
678 Example: Play a file stored on remote server.
681 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
684 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
686 Real-Time Messaging Protocol and its variants supported through
689 Requires the presence of the librtmp headers and library during
690 configuration. You need to explicitly configure the build with
691 "--enable-librtmp". If enabled this will replace the native RTMP
694 This protocol provides most client functions and a few server
695 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
696 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
697 variants of these encrypted types (RTMPTE, RTMPTS).
699 The required syntax is:
701 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
704 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
705 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
706 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
707 meaning as specified for the RTMP native protocol.
708 @var{options} contains a list of space-separated options of the form
711 See the librtmp manual page (man 3 librtmp) for more information.
713 For example, to stream a file in real-time to an RTMP server using
716 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
719 To play the same stream using @command{ffplay}:
721 ffplay "rtmp://myserver/live/mystream live=1"
726 Real-time Transport Protocol.
728 The required syntax for an RTP URL is:
729 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
731 @var{port} specifies the RTP port to use.
733 The following URL options are supported:
738 Set the TTL (Time-To-Live) value (for multicast only).
740 @item rtcpport=@var{n}
741 Set the remote RTCP port to @var{n}.
743 @item localrtpport=@var{n}
744 Set the local RTP port to @var{n}.
746 @item localrtcpport=@var{n}'
747 Set the local RTCP port to @var{n}.
749 @item pkt_size=@var{n}
750 Set max packet size (in bytes) to @var{n}.
753 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
756 @item sources=@var{ip}[,@var{ip}]
757 List allowed source IP addresses.
759 @item block=@var{ip}[,@var{ip}]
760 List disallowed (blocked) source IP addresses.
762 @item write_to_source=0|1
763 Send packets to the source address of the latest received packet (if
764 set to 1) or to a default remote address (if set to 0).
766 @item localport=@var{n}
767 Set the local RTP port to @var{n}.
769 This is a deprecated option. Instead, @option{localrtpport} should be
779 If @option{rtcpport} is not set the RTCP port will be set to the RTP
783 If @option{localrtpport} (the local RTP port) is not set any available
784 port will be used for the local RTP and RTCP ports.
787 If @option{localrtcpport} (the local RTCP port) is not set it will be
788 set to the local RTP port value plus 1.
793 Real-Time Streaming Protocol.
795 RTSP is not technically a protocol handler in libavformat, it is a demuxer
796 and muxer. The demuxer supports both normal RTSP (with data transferred
797 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
798 data transferred over RDT).
800 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
801 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
802 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
804 The required syntax for a RTSP url is:
806 rtsp://@var{hostname}[:@var{port}]/@var{path}
809 Options can be set on the @command{ffmpeg}/@command{ffplay} command
810 line, or set in code via @code{AVOption}s or in
811 @code{avformat_open_input}.
813 The following options are supported.
817 Do not start playing the stream immediately if set to 1. Default value
821 Set RTSP transport protocols.
823 It accepts the following values:
826 Use UDP as lower transport protocol.
829 Use TCP (interleaving within the RTSP control channel) as lower
833 Use UDP multicast as lower transport protocol.
836 Use HTTP tunneling as lower transport protocol, which is useful for
840 Multiple lower transport protocols may be specified, in that case they are
841 tried one at a time (if the setup of one fails, the next one is tried).
842 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
847 The following values are accepted:
850 Accept packets only from negotiated peer address and port.
852 Act as a server, listening for an incoming connection.
854 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
857 Default value is @samp{none}.
859 @item allowed_media_types
860 Set media types to accept from the server.
862 The following flags are accepted:
869 By default it accepts all media types.
872 Set minimum local UDP port. Default value is 5000.
875 Set maximum local UDP port. Default value is 65000.
878 Set maximum timeout (in seconds) to wait for incoming connections.
880 A value of -1 means infinite (default). This option implies the
881 @option{rtsp_flags} set to @samp{listen}.
883 @item reorder_queue_size
884 Set number of packets to buffer for handling of reordered packets.
887 Set socket TCP I/O timeout in microseconds.
890 Override User-Agent header. If not specified, it defaults to the
891 libavformat identifier string.
894 When receiving data over UDP, the demuxer tries to reorder received packets
895 (since they may arrive out of order, or packets may get lost totally). This
896 can be disabled by setting the maximum demuxing delay to zero (via
897 the @code{max_delay} field of AVFormatContext).
899 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
900 streams to display can be chosen with @code{-vst} @var{n} and
901 @code{-ast} @var{n} for video and audio respectively, and can be switched
902 on the fly by pressing @code{v} and @code{a}.
906 The following examples all make use of the @command{ffplay} and
907 @command{ffmpeg} tools.
911 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
913 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
917 Watch a stream tunneled over HTTP:
919 ffplay -rtsp_transport http rtsp://server/video.mp4
923 Send a stream in realtime to a RTSP server, for others to watch:
925 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
929 Receive a stream in realtime:
931 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
937 Session Announcement Protocol (RFC 2974). This is not technically a
938 protocol handler in libavformat, it is a muxer and demuxer.
939 It is used for signalling of RTP streams, by announcing the SDP for the
940 streams regularly on a separate port.
944 The syntax for a SAP url given to the muxer is:
946 sap://@var{destination}[:@var{port}][?@var{options}]
949 The RTP packets are sent to @var{destination} on port @var{port},
950 or to port 5004 if no port is specified.
951 @var{options} is a @code{&}-separated list. The following options
956 @item announce_addr=@var{address}
957 Specify the destination IP address for sending the announcements to.
958 If omitted, the announcements are sent to the commonly used SAP
959 announcement multicast address 224.2.127.254 (sap.mcast.net), or
960 ff0e::2:7ffe if @var{destination} is an IPv6 address.
962 @item announce_port=@var{port}
963 Specify the port to send the announcements on, defaults to
964 9875 if not specified.
967 Specify the time to live value for the announcements and RTP packets,
970 @item same_port=@var{0|1}
971 If set to 1, send all RTP streams on the same port pair. If zero (the
972 default), all streams are sent on unique ports, with each stream on a
973 port 2 numbers higher than the previous.
974 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
975 The RTP stack in libavformat for receiving requires all streams to be sent
979 Example command lines follow.
981 To broadcast a stream on the local subnet, for watching in VLC:
984 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
987 Similarly, for watching in @command{ffplay}:
990 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
993 And for watching in @command{ffplay}, over IPv6:
996 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1001 The syntax for a SAP url given to the demuxer is:
1003 sap://[@var{address}][:@var{port}]
1006 @var{address} is the multicast address to listen for announcements on,
1007 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1008 is the port that is listened on, 9875 if omitted.
1010 The demuxers listens for announcements on the given address and port.
1011 Once an announcement is received, it tries to receive that particular stream.
1013 Example command lines follow.
1015 To play back the first stream announced on the normal SAP multicast address:
1021 To play back the first stream announced on one the default IPv6 SAP multicast address:
1024 ffplay sap://[ff0e::2:7ffe]
1029 Stream Control Transmission Protocol.
1031 The accepted URL syntax is:
1033 sctp://@var{host}:@var{port}[?@var{options}]
1036 The protocol accepts the following options:
1039 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1042 Set the maximum number of streams. By default no limit is set.
1047 Secure Real-time Transport Protocol.
1049 The accepted options are:
1052 @item srtp_out_suite
1053 Select input and output encoding suites.
1057 @item AES_CM_128_HMAC_SHA1_80
1058 @item SRTP_AES128_CM_HMAC_SHA1_80
1059 @item AES_CM_128_HMAC_SHA1_32
1060 @item SRTP_AES128_CM_HMAC_SHA1_32
1063 @item srtp_in_params
1064 @item srtp_out_params
1065 Set input and output encoding parameters, which are expressed by a
1066 base64-encoded representation of a binary block. The first 16 bytes of
1067 this binary block are used as master key, the following 14 bytes are
1068 used as master salt.
1073 Virtually extract a segment of a file or another stream.
1074 The underlying stream must be seekable.
1079 Start offset of the extracted segment, in bytes.
1081 End offset of the extracted segment, in bytes.
1086 Extract a chapter from a DVD VOB file (start and end sectors obtained
1087 externally and multiplied by 2048):
1089 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1092 Play an AVI file directly from a TAR archive:
1094 subfile,,start,183241728,end,366490624,,:archive.tar
1099 Transmission Control Protocol.
1101 The required syntax for a TCP url is:
1103 tcp://@var{hostname}:@var{port}[?@var{options}]
1106 @var{options} contains a list of &-separated options of the form
1107 @var{key}=@var{val}.
1109 The list of supported options follows.
1112 @item listen=@var{1|0}
1113 Listen for an incoming connection. Default value is 0.
1115 @item timeout=@var{microseconds}
1116 Set raise error timeout, expressed in microseconds.
1118 This option is only relevant in read mode: if no data arrived in more
1119 than this time interval, raise error.
1121 @item listen_timeout=@var{milliseconds}
1122 Set listen timeout, expressed in milliseconds.
1125 The following example shows how to setup a listening TCP connection
1126 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1128 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1129 ffplay tcp://@var{hostname}:@var{port}
1134 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1136 The required syntax for a TLS/SSL url is:
1138 tls://@var{hostname}:@var{port}[?@var{options}]
1141 The following parameters can be set via command line options
1142 (or in code via @code{AVOption}s):
1146 @item ca_file, cafile=@var{filename}
1147 A file containing certificate authority (CA) root certificates to treat
1148 as trusted. If the linked TLS library contains a default this might not
1149 need to be specified for verification to work, but not all libraries and
1150 setups have defaults built in.
1151 The file must be in OpenSSL PEM format.
1153 @item tls_verify=@var{1|0}
1154 If enabled, try to verify the peer that we are communicating with.
1155 Note, if using OpenSSL, this currently only makes sure that the
1156 peer certificate is signed by one of the root certificates in the CA
1157 database, but it does not validate that the certificate actually
1158 matches the host name we are trying to connect to. (With GnuTLS,
1159 the host name is validated as well.)
1161 This is disabled by default since it requires a CA database to be
1162 provided by the caller in many cases.
1164 @item cert_file, cert=@var{filename}
1165 A file containing a certificate to use in the handshake with the peer.
1166 (When operating as server, in listen mode, this is more often required
1167 by the peer, while client certificates only are mandated in certain
1170 @item key_file, key=@var{filename}
1171 A file containing the private key for the certificate.
1173 @item listen=@var{1|0}
1174 If enabled, listen for connections on the provided port, and assume
1175 the server role in the handshake instead of the client role.
1179 Example command lines:
1181 To create a TLS/SSL server that serves an input stream.
1184 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1187 To play back a stream from the TLS/SSL server using @command{ffplay}:
1190 ffplay tls://@var{hostname}:@var{port}
1195 User Datagram Protocol.
1197 The required syntax for an UDP URL is:
1199 udp://@var{hostname}:@var{port}[?@var{options}]
1202 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1204 In case threading is enabled on the system, a circular buffer is used
1205 to store the incoming data, which allows one to reduce loss of data due to
1206 UDP socket buffer overruns. The @var{fifo_size} and
1207 @var{overrun_nonfatal} options are related to this buffer.
1209 The list of supported options follows.
1212 @item buffer_size=@var{size}
1213 Set the UDP maximum socket buffer size in bytes. This is used to set either
1214 the receive or send buffer size, depending on what the socket is used for.
1215 Default is 64KB. See also @var{fifo_size}.
1217 @item localport=@var{port}
1218 Override the local UDP port to bind with.
1220 @item localaddr=@var{addr}
1221 Choose the local IP address. This is useful e.g. if sending multicast
1222 and the host has multiple interfaces, where the user can choose
1223 which interface to send on by specifying the IP address of that interface.
1225 @item pkt_size=@var{size}
1226 Set the size in bytes of UDP packets.
1228 @item reuse=@var{1|0}
1229 Explicitly allow or disallow reusing UDP sockets.
1232 Set the time to live value (for multicast only).
1234 @item connect=@var{1|0}
1235 Initialize the UDP socket with @code{connect()}. In this case, the
1236 destination address can't be changed with ff_udp_set_remote_url later.
1237 If the destination address isn't known at the start, this option can
1238 be specified in ff_udp_set_remote_url, too.
1239 This allows finding out the source address for the packets with getsockname,
1240 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1241 unreachable" is received.
1242 For receiving, this gives the benefit of only receiving packets from
1243 the specified peer address/port.
1245 @item sources=@var{address}[,@var{address}]
1246 Only receive packets sent to the multicast group from one of the
1247 specified sender IP addresses.
1249 @item block=@var{address}[,@var{address}]
1250 Ignore packets sent to the multicast group from the specified
1251 sender IP addresses.
1253 @item fifo_size=@var{units}
1254 Set the UDP receiving circular buffer size, expressed as a number of
1255 packets with size of 188 bytes. If not specified defaults to 7*4096.
1257 @item overrun_nonfatal=@var{1|0}
1258 Survive in case of UDP receiving circular buffer overrun. Default
1261 @item timeout=@var{microseconds}
1262 Set raise error timeout, expressed in microseconds.
1264 This option is only relevant in read mode: if no data arrived in more
1265 than this time interval, raise error.
1267 @item broadcast=@var{1|0}
1268 Explicitly allow or disallow UDP broadcasting.
1270 Note that broadcasting may not work properly on networks having
1271 a broadcast storm protection.
1274 @subsection Examples
1278 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1280 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1284 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1285 sized UDP packets, using a large input buffer:
1287 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1291 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1293 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1301 The required syntax for a Unix socket URL is:
1304 unix://@var{filepath}
1307 The following parameters can be set via command line options
1308 (or in code via @code{AVOption}s):
1314 Create the Unix socket in listening mode.
1317 @c man end PROTOCOLS