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
264 Export the MIME type.
267 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
268 supports this, the metadata has to be retrieved by the application by reading
269 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
272 @item icy_metadata_headers
273 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
274 headers, separated by newline characters.
276 @item icy_metadata_packet
277 If the server supports ICY metadata, and @option{icy} was set to 1, this
278 contains the last non-empty metadata packet sent by the server. It should be
279 polled in regular intervals by applications interested in mid-stream metadata
283 Set the cookies to be sent in future requests. The format of each cookie is the
284 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
285 delimited by a newline character.
288 Set initial byte offset.
291 Try to limit the request to bytes preceding this offset.
294 When used as a client option it sets the HTTP method for the request.
296 When used as a server option it sets the HTTP method that is going to be
297 expected from the client(s).
298 If the expected and the received HTTP method do not match the client will
299 be given a Bad Request response.
300 When unset the HTTP method is not checked for now. This will be replaced by
301 autodetection in the future.
304 If set to 1 enables experimental HTTP server. This can be used to send data when
305 used as an output option, or read data from a client with HTTP POST when used as
307 If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
308 in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
310 # Server side (sending):
311 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
313 # Client side (receiving):
314 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
316 # Client can also be done with wget:
317 wget http://@var{server}:@var{port} -O somefile.ogg
319 # Server side (receiving):
320 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
322 # Client side (sending):
323 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
325 # Client can also be done with wget:
326 wget --post-file=somefile.ogg http://@var{server}:@var{port}
331 @subsection HTTP Cookies
333 Some HTTP requests will be denied unless cookie values are passed in with the
334 request. The @option{cookies} option allows these cookies to be specified. At
335 the very least, each cookie must specify a value along with a path and domain.
336 HTTP requests that match both the domain and path will automatically include the
337 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
340 The required syntax to play a stream specifying a cookie is:
342 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
347 Icecast protocol (stream to Icecast servers)
349 This protocol accepts the following options:
353 Set the stream genre.
358 @item ice_description
359 Set the stream description.
362 Set the stream website URL.
365 Set if the stream should be public.
366 The default is 0 (not public).
369 Override the User-Agent header. If not specified a string of the form
370 "Lavf/<version>" will be used.
373 Set the Icecast mountpoint password.
376 Set the stream content type. This must be set if it is different from
380 This enables support for Icecast versions < 2.4.0, that do not support the
381 HTTP PUT method but the SOURCE method.
386 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
391 MMS (Microsoft Media Server) protocol over TCP.
395 MMS (Microsoft Media Server) protocol over HTTP.
397 The required syntax is:
399 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
406 Computes the MD5 hash of the data to be written, and on close writes
407 this to the designated output or stdout if none is specified. It can
408 be used to test muxers without writing an actual file.
410 Some examples follow.
412 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
413 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
415 # Write the MD5 hash of the encoded AVI file to stdout.
416 ffmpeg -i input.flv -f avi -y md5:
419 Note that some formats (typically MOV) require the output protocol to
420 be seekable, so they will fail with the MD5 output protocol.
424 UNIX pipe access protocol.
426 Read and write from UNIX pipes.
428 The accepted syntax is:
433 @var{number} is the number corresponding to the file descriptor of the
434 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
435 is not specified, by default the stdout file descriptor will be used
436 for writing, stdin for reading.
438 For example to read from stdin with @command{ffmpeg}:
440 cat test.wav | ffmpeg -i pipe:0
441 # ...this is the same as...
442 cat test.wav | ffmpeg -i pipe:
445 For writing to stdout with @command{ffmpeg}:
447 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
448 # ...this is the same as...
449 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
452 This protocol accepts the following options:
456 Set I/O operation maximum block size, in bytes. Default value is
457 @code{INT_MAX}, which results in not limiting the requested block size.
458 Setting this value reasonably low improves user termination request reaction
459 time, which is valuable if data transmission is slow.
462 Note that some formats (typically MOV), require the output protocol to
463 be seekable, so they will fail with the pipe output protocol.
467 Real-Time Messaging Protocol.
469 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
470 content across a TCP/IP network.
472 The required syntax is:
474 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
477 The accepted parameters are:
481 An optional username (mostly for publishing).
484 An optional password (mostly for publishing).
487 The address of the RTMP server.
490 The number of the TCP port to use (by default is 1935).
493 It is the name of the application to access. It usually corresponds to
494 the path where the application is installed on the RTMP server
495 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
496 the value parsed from the URI through the @code{rtmp_app} option, too.
499 It is the path or name of the resource to play with reference to the
500 application specified in @var{app}, may be prefixed by "mp4:". You
501 can override the value parsed from the URI through the @code{rtmp_playpath}
505 Act as a server, listening for an incoming connection.
508 Maximum time to wait for the incoming connection. Implies listen.
511 Additionally, the following parameters can be set via command line options
512 (or in code via @code{AVOption}s):
516 Name of application to connect on the RTMP server. This option
517 overrides the parameter specified in the URI.
520 Set the client buffer time in milliseconds. The default is 3000.
523 Extra arbitrary AMF connection parameters, parsed from a string,
524 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
525 Each value is prefixed by a single character denoting the type,
526 B for Boolean, N for number, S for string, O for object, or Z for null,
527 followed by a colon. For Booleans the data must be either 0 or 1 for
528 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
529 1 to end or begin an object, respectively. Data items in subobjects may
530 be named, by prefixing the type with 'N' and specifying the name before
531 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
532 times to construct arbitrary AMF sequences.
535 Version of the Flash plugin used to run the SWF player. The default
536 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
537 <libavformat version>).)
539 @item rtmp_flush_interval
540 Number of packets flushed in the same request (RTMPT only). The default
544 Specify that the media is a live stream. No resuming or seeking in
545 live streams is possible. The default value is @code{any}, which means the
546 subscriber first tries to play the live stream specified in the
547 playpath. If a live stream of that name is not found, it plays the
548 recorded stream. The other possible values are @code{live} and
552 URL of the web page in which the media was embedded. By default no
556 Stream identifier to play or to publish. This option overrides the
557 parameter specified in the URI.
560 Name of live stream to subscribe to. By default no value will be sent.
561 It is only sent if the option is specified or if rtmp_live
565 SHA256 hash of the decompressed SWF file (32 bytes).
568 Size of the decompressed SWF file, required for SWFVerification.
571 URL of the SWF player for the media. By default no value will be sent.
574 URL to player swf file, compute hash/size automatically.
577 URL of the target stream. Defaults to proto://host[:port]/app.
581 For example to read with @command{ffplay} a multimedia resource named
582 "sample" from the application "vod" from an RTMP server "myserver":
584 ffplay rtmp://myserver/vod/sample
587 To publish to a password protected server, passing the playpath and
588 app names separately:
590 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
595 Encrypted Real-Time Messaging Protocol.
597 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
598 streaming multimedia content within standard cryptographic primitives,
599 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
604 Real-Time Messaging Protocol over a secure SSL connection.
606 The Real-Time Messaging Protocol (RTMPS) is used for streaming
607 multimedia content across an encrypted connection.
611 Real-Time Messaging Protocol tunneled through HTTP.
613 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
614 for streaming multimedia content within HTTP requests to traverse
619 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
621 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
622 is used for streaming multimedia content within HTTP requests to traverse
627 Real-Time Messaging Protocol tunneled through HTTPS.
629 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
630 for streaming multimedia content within HTTPS requests to traverse
633 @section libsmbclient
635 libsmbclient permits one to manipulate CIFS/SMB network resources.
637 Following syntax is required.
640 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
643 This protocol accepts the following options.
647 Set timeout in miliseconds of socket I/O operations used by the underlying
648 low level operation. By default it is set to -1, which means that the timeout
652 Truncate existing files on write, if set to 1. A value of 0 prevents
653 truncating. Default value is 1.
656 Set the workgroup used for making connections. By default workgroup is not specified.
660 For more information see: @url{http://www.samba.org/}.
664 Secure File Transfer Protocol via libssh
666 Read from or write to remote resources using SFTP protocol.
668 Following syntax is required.
671 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
674 This protocol accepts the following options.
678 Set timeout of socket I/O operations used by the underlying low level
679 operation. By default it is set to -1, which means that the timeout
683 Truncate existing files on write, if set to 1. A value of 0 prevents
684 truncating. Default value is 1.
687 Specify the path of the file containing private key to use during authorization.
688 By default libssh searches for keys in the @file{~/.ssh/} directory.
692 Example: Play a file stored on remote server.
695 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
698 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
700 Real-Time Messaging Protocol and its variants supported through
703 Requires the presence of the librtmp headers and library during
704 configuration. You need to explicitly configure the build with
705 "--enable-librtmp". If enabled this will replace the native RTMP
708 This protocol provides most client functions and a few server
709 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
710 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
711 variants of these encrypted types (RTMPTE, RTMPTS).
713 The required syntax is:
715 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
718 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
719 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
720 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
721 meaning as specified for the RTMP native protocol.
722 @var{options} contains a list of space-separated options of the form
725 See the librtmp manual page (man 3 librtmp) for more information.
727 For example, to stream a file in real-time to an RTMP server using
730 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
733 To play the same stream using @command{ffplay}:
735 ffplay "rtmp://myserver/live/mystream live=1"
740 Real-time Transport Protocol.
742 The required syntax for an RTP URL is:
743 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
745 @var{port} specifies the RTP port to use.
747 The following URL options are supported:
752 Set the TTL (Time-To-Live) value (for multicast only).
754 @item rtcpport=@var{n}
755 Set the remote RTCP port to @var{n}.
757 @item localrtpport=@var{n}
758 Set the local RTP port to @var{n}.
760 @item localrtcpport=@var{n}'
761 Set the local RTCP port to @var{n}.
763 @item pkt_size=@var{n}
764 Set max packet size (in bytes) to @var{n}.
767 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
770 @item sources=@var{ip}[,@var{ip}]
771 List allowed source IP addresses.
773 @item block=@var{ip}[,@var{ip}]
774 List disallowed (blocked) source IP addresses.
776 @item write_to_source=0|1
777 Send packets to the source address of the latest received packet (if
778 set to 1) or to a default remote address (if set to 0).
780 @item localport=@var{n}
781 Set the local RTP port to @var{n}.
783 This is a deprecated option. Instead, @option{localrtpport} should be
793 If @option{rtcpport} is not set the RTCP port will be set to the RTP
797 If @option{localrtpport} (the local RTP port) is not set any available
798 port will be used for the local RTP and RTCP ports.
801 If @option{localrtcpport} (the local RTCP port) is not set it will be
802 set to the local RTP port value plus 1.
807 Real-Time Streaming Protocol.
809 RTSP is not technically a protocol handler in libavformat, it is a demuxer
810 and muxer. The demuxer supports both normal RTSP (with data transferred
811 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
812 data transferred over RDT).
814 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
815 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
816 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
818 The required syntax for a RTSP url is:
820 rtsp://@var{hostname}[:@var{port}]/@var{path}
823 Options can be set on the @command{ffmpeg}/@command{ffplay} command
824 line, or set in code via @code{AVOption}s or in
825 @code{avformat_open_input}.
827 The following options are supported.
831 Do not start playing the stream immediately if set to 1. Default value
835 Set RTSP transport protocols.
837 It accepts the following values:
840 Use UDP as lower transport protocol.
843 Use TCP (interleaving within the RTSP control channel) as lower
847 Use UDP multicast as lower transport protocol.
850 Use HTTP tunneling as lower transport protocol, which is useful for
854 Multiple lower transport protocols may be specified, in that case they are
855 tried one at a time (if the setup of one fails, the next one is tried).
856 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
861 The following values are accepted:
864 Accept packets only from negotiated peer address and port.
866 Act as a server, listening for an incoming connection.
868 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
871 Default value is @samp{none}.
873 @item allowed_media_types
874 Set media types to accept from the server.
876 The following flags are accepted:
883 By default it accepts all media types.
886 Set minimum local UDP port. Default value is 5000.
889 Set maximum local UDP port. Default value is 65000.
892 Set maximum timeout (in seconds) to wait for incoming connections.
894 A value of -1 means infinite (default). This option implies the
895 @option{rtsp_flags} set to @samp{listen}.
897 @item reorder_queue_size
898 Set number of packets to buffer for handling of reordered packets.
901 Set socket TCP I/O timeout in microseconds.
904 Override User-Agent header. If not specified, it defaults to the
905 libavformat identifier string.
908 When receiving data over UDP, the demuxer tries to reorder received packets
909 (since they may arrive out of order, or packets may get lost totally). This
910 can be disabled by setting the maximum demuxing delay to zero (via
911 the @code{max_delay} field of AVFormatContext).
913 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
914 streams to display can be chosen with @code{-vst} @var{n} and
915 @code{-ast} @var{n} for video and audio respectively, and can be switched
916 on the fly by pressing @code{v} and @code{a}.
920 The following examples all make use of the @command{ffplay} and
921 @command{ffmpeg} tools.
925 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
927 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
931 Watch a stream tunneled over HTTP:
933 ffplay -rtsp_transport http rtsp://server/video.mp4
937 Send a stream in realtime to a RTSP server, for others to watch:
939 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
943 Receive a stream in realtime:
945 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
951 Session Announcement Protocol (RFC 2974). This is not technically a
952 protocol handler in libavformat, it is a muxer and demuxer.
953 It is used for signalling of RTP streams, by announcing the SDP for the
954 streams regularly on a separate port.
958 The syntax for a SAP url given to the muxer is:
960 sap://@var{destination}[:@var{port}][?@var{options}]
963 The RTP packets are sent to @var{destination} on port @var{port},
964 or to port 5004 if no port is specified.
965 @var{options} is a @code{&}-separated list. The following options
970 @item announce_addr=@var{address}
971 Specify the destination IP address for sending the announcements to.
972 If omitted, the announcements are sent to the commonly used SAP
973 announcement multicast address 224.2.127.254 (sap.mcast.net), or
974 ff0e::2:7ffe if @var{destination} is an IPv6 address.
976 @item announce_port=@var{port}
977 Specify the port to send the announcements on, defaults to
978 9875 if not specified.
981 Specify the time to live value for the announcements and RTP packets,
984 @item same_port=@var{0|1}
985 If set to 1, send all RTP streams on the same port pair. If zero (the
986 default), all streams are sent on unique ports, with each stream on a
987 port 2 numbers higher than the previous.
988 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
989 The RTP stack in libavformat for receiving requires all streams to be sent
993 Example command lines follow.
995 To broadcast a stream on the local subnet, for watching in VLC:
998 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1001 Similarly, for watching in @command{ffplay}:
1004 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1007 And for watching in @command{ffplay}, over IPv6:
1010 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1015 The syntax for a SAP url given to the demuxer is:
1017 sap://[@var{address}][:@var{port}]
1020 @var{address} is the multicast address to listen for announcements on,
1021 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1022 is the port that is listened on, 9875 if omitted.
1024 The demuxers listens for announcements on the given address and port.
1025 Once an announcement is received, it tries to receive that particular stream.
1027 Example command lines follow.
1029 To play back the first stream announced on the normal SAP multicast address:
1035 To play back the first stream announced on one the default IPv6 SAP multicast address:
1038 ffplay sap://[ff0e::2:7ffe]
1043 Stream Control Transmission Protocol.
1045 The accepted URL syntax is:
1047 sctp://@var{host}:@var{port}[?@var{options}]
1050 The protocol accepts the following options:
1053 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1056 Set the maximum number of streams. By default no limit is set.
1061 Secure Real-time Transport Protocol.
1063 The accepted options are:
1066 @item srtp_out_suite
1067 Select input and output encoding suites.
1071 @item AES_CM_128_HMAC_SHA1_80
1072 @item SRTP_AES128_CM_HMAC_SHA1_80
1073 @item AES_CM_128_HMAC_SHA1_32
1074 @item SRTP_AES128_CM_HMAC_SHA1_32
1077 @item srtp_in_params
1078 @item srtp_out_params
1079 Set input and output encoding parameters, which are expressed by a
1080 base64-encoded representation of a binary block. The first 16 bytes of
1081 this binary block are used as master key, the following 14 bytes are
1082 used as master salt.
1087 Virtually extract a segment of a file or another stream.
1088 The underlying stream must be seekable.
1093 Start offset of the extracted segment, in bytes.
1095 End offset of the extracted segment, in bytes.
1100 Extract a chapter from a DVD VOB file (start and end sectors obtained
1101 externally and multiplied by 2048):
1103 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1106 Play an AVI file directly from a TAR archive:
1108 subfile,,start,183241728,end,366490624,,:archive.tar
1113 Transmission Control Protocol.
1115 The required syntax for a TCP url is:
1117 tcp://@var{hostname}:@var{port}[?@var{options}]
1120 @var{options} contains a list of &-separated options of the form
1121 @var{key}=@var{val}.
1123 The list of supported options follows.
1126 @item listen=@var{1|0}
1127 Listen for an incoming connection. Default value is 0.
1129 @item timeout=@var{microseconds}
1130 Set raise error timeout, expressed in microseconds.
1132 This option is only relevant in read mode: if no data arrived in more
1133 than this time interval, raise error.
1135 @item listen_timeout=@var{milliseconds}
1136 Set listen timeout, expressed in milliseconds.
1139 The following example shows how to setup a listening TCP connection
1140 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1142 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1143 ffplay tcp://@var{hostname}:@var{port}
1148 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1150 The required syntax for a TLS/SSL url is:
1152 tls://@var{hostname}:@var{port}[?@var{options}]
1155 The following parameters can be set via command line options
1156 (or in code via @code{AVOption}s):
1160 @item ca_file, cafile=@var{filename}
1161 A file containing certificate authority (CA) root certificates to treat
1162 as trusted. If the linked TLS library contains a default this might not
1163 need to be specified for verification to work, but not all libraries and
1164 setups have defaults built in.
1165 The file must be in OpenSSL PEM format.
1167 @item tls_verify=@var{1|0}
1168 If enabled, try to verify the peer that we are communicating with.
1169 Note, if using OpenSSL, this currently only makes sure that the
1170 peer certificate is signed by one of the root certificates in the CA
1171 database, but it does not validate that the certificate actually
1172 matches the host name we are trying to connect to. (With GnuTLS,
1173 the host name is validated as well.)
1175 This is disabled by default since it requires a CA database to be
1176 provided by the caller in many cases.
1178 @item cert_file, cert=@var{filename}
1179 A file containing a certificate to use in the handshake with the peer.
1180 (When operating as server, in listen mode, this is more often required
1181 by the peer, while client certificates only are mandated in certain
1184 @item key_file, key=@var{filename}
1185 A file containing the private key for the certificate.
1187 @item listen=@var{1|0}
1188 If enabled, listen for connections on the provided port, and assume
1189 the server role in the handshake instead of the client role.
1193 Example command lines:
1195 To create a TLS/SSL server that serves an input stream.
1198 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1201 To play back a stream from the TLS/SSL server using @command{ffplay}:
1204 ffplay tls://@var{hostname}:@var{port}
1209 User Datagram Protocol.
1211 The required syntax for an UDP URL is:
1213 udp://@var{hostname}:@var{port}[?@var{options}]
1216 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1218 In case threading is enabled on the system, a circular buffer is used
1219 to store the incoming data, which allows one to reduce loss of data due to
1220 UDP socket buffer overruns. The @var{fifo_size} and
1221 @var{overrun_nonfatal} options are related to this buffer.
1223 The list of supported options follows.
1226 @item buffer_size=@var{size}
1227 Set the UDP maximum socket buffer size in bytes. This is used to set either
1228 the receive or send buffer size, depending on what the socket is used for.
1229 Default is 64KB. See also @var{fifo_size}.
1231 @item localport=@var{port}
1232 Override the local UDP port to bind with.
1234 @item localaddr=@var{addr}
1235 Choose the local IP address. This is useful e.g. if sending multicast
1236 and the host has multiple interfaces, where the user can choose
1237 which interface to send on by specifying the IP address of that interface.
1239 @item pkt_size=@var{size}
1240 Set the size in bytes of UDP packets.
1242 @item reuse=@var{1|0}
1243 Explicitly allow or disallow reusing UDP sockets.
1246 Set the time to live value (for multicast only).
1248 @item connect=@var{1|0}
1249 Initialize the UDP socket with @code{connect()}. In this case, the
1250 destination address can't be changed with ff_udp_set_remote_url later.
1251 If the destination address isn't known at the start, this option can
1252 be specified in ff_udp_set_remote_url, too.
1253 This allows finding out the source address for the packets with getsockname,
1254 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1255 unreachable" is received.
1256 For receiving, this gives the benefit of only receiving packets from
1257 the specified peer address/port.
1259 @item sources=@var{address}[,@var{address}]
1260 Only receive packets sent to the multicast group from one of the
1261 specified sender IP addresses.
1263 @item block=@var{address}[,@var{address}]
1264 Ignore packets sent to the multicast group from the specified
1265 sender IP addresses.
1267 @item fifo_size=@var{units}
1268 Set the UDP receiving circular buffer size, expressed as a number of
1269 packets with size of 188 bytes. If not specified defaults to 7*4096.
1271 @item overrun_nonfatal=@var{1|0}
1272 Survive in case of UDP receiving circular buffer overrun. Default
1275 @item timeout=@var{microseconds}
1276 Set raise error timeout, expressed in microseconds.
1278 This option is only relevant in read mode: if no data arrived in more
1279 than this time interval, raise error.
1281 @item broadcast=@var{1|0}
1282 Explicitly allow or disallow UDP broadcasting.
1284 Note that broadcasting may not work properly on networks having
1285 a broadcast storm protection.
1288 @subsection Examples
1292 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1294 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1298 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1299 sized UDP packets, using a large input buffer:
1301 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1305 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1307 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1315 The required syntax for a Unix socket URL is:
1318 unix://@var{filepath}
1321 The following parameters can be set via command line options
1322 (or in code via @code{AVOption}s):
1328 Create the Unix socket in listening mode.
1331 @c man end PROTOCOLS