1 @chapter Protocol Options
2 @c man begin PROTOCOL OPTIONS
4 The libavformat library provides some generic global options, which
5 can be set on all the protocols. In addition each protocol may support
6 so-called private options, which are specific for that component.
8 The list of supported options follows:
11 @item protocol_whitelist @var{list} (@emph{input})
12 Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
13 prefixed by "-" are disabled.
14 All protocols are allowed by default but protocols used by an another
15 protocol (nested protocols) are restricted to a per protocol subset.
18 @c man end PROTOCOL OPTIONS
21 @c man begin PROTOCOLS
23 Protocols are configured elements in FFmpeg that enable access to
24 resources that require specific protocols.
26 When you configure your FFmpeg build, all the supported protocols are
27 enabled by default. You can list all available ones using the
28 configure option "--list-protocols".
30 You can disable all the protocols using the configure option
31 "--disable-protocols", and selectively enable a protocol using the
32 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
33 particular protocol using the option
34 "--disable-protocol=@var{PROTOCOL}".
36 The option "-protocols" of the ff* tools will display the list of
39 A description of the currently available protocols follows.
43 Asynchronous data filling wrapper for input stream.
45 Fill data in a background thread, to decouple I/O operation from demux thread.
49 async:http://host/resource
50 async:cache:http://host/resource
57 The accepted options are:
67 Playlist to read (BDMV/PLAYLIST/?????.mpls)
73 Read longest playlist from BluRay mounted to /mnt/bluray:
78 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
80 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
85 Caching wrapper for input stream.
87 Cache the input stream to temporary file. It brings seeking capability to live streams.
95 Physical concatenation protocol.
97 Read and seek from many resources in sequence as if they were
100 A URL accepted by this protocol has the syntax:
102 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
105 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
106 resource to be concatenated, each one possibly specifying a distinct
109 For example to read a sequence of files @file{split1.mpeg},
110 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
113 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
116 Note that you may need to escape the character "|" which is special for
121 AES-encrypted stream reading protocol.
123 The accepted options are:
126 Set the AES decryption key binary block from given hexadecimal representation.
129 Set the AES decryption initialization vector binary block from given hexadecimal representation.
132 Accepted URL formats:
140 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
142 For example, to convert a GIF file given inline with @command{ffmpeg}:
144 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
149 File access protocol.
151 Read from or write to a file.
153 A file URL can have the form:
158 where @var{filename} is the path of the file to read.
160 An URL that does not have a protocol prefix will be assumed to be a
161 file URL. Depending on the build, an URL that looks like a Windows
162 path with the drive letter at the beginning will also be assumed to be
163 a file URL (usually not the case in builds for unix-like systems).
165 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
168 ffmpeg -i file:input.mpeg output.mpeg
171 This protocol accepts the following options:
175 Truncate existing files on write, if set to 1. A value of 0 prevents
176 truncating. Default value is 1.
179 Set I/O operation maximum block size, in bytes. Default value is
180 @code{INT_MAX}, which results in not limiting the requested block size.
181 Setting this value reasonably low improves user termination request reaction
182 time, which is valuable for files on slow medium.
187 FTP (File Transfer Protocol).
189 Read from or write to remote resources using FTP protocol.
191 Following syntax is required.
193 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
196 This protocol accepts the following options.
200 Set timeout in microseconds of socket I/O operations used by the underlying low level
201 operation. By default it is set to -1, which means that the timeout is
204 @item ftp-anonymous-password
205 Password used when login as anonymous user. Typically an e-mail address
208 @item ftp-write-seekable
209 Control seekability of connection during encoding. If set to 1 the
210 resource is supposed to be seekable, if set to 0 it is assumed not
211 to be seekable. Default value is 0.
214 NOTE: Protocol can be used as output, but it is recommended to not do
215 it, unless special care is taken (tests, customized server configuration
216 etc.). Different FTP servers behave in different way during seek
217 operation. ff* tools may produce incomplete content due to server limitations.
225 Read Apple HTTP Live Streaming compliant segmented stream as
226 a uniform one. The M3U8 playlists describing the segments can be
227 remote HTTP resources or local files, accessed using the standard
229 The nested protocol is declared by specifying
230 "+@var{proto}" after the hls URI scheme name, where @var{proto}
231 is either "file" or "http".
234 hls+http://host/path/to/remote/resource.m3u8
235 hls+file://path/to/local/resource.m3u8
238 Using this protocol is discouraged - the hls demuxer should work
239 just as well (if not, please report the issues) and is more complete.
240 To use the hls demuxer instead, simply use the direct URLs to the
245 HTTP (Hyper Text Transfer Protocol).
247 This protocol accepts the following options:
251 Control seekability of connection. If set to 1 the resource is
252 supposed to be seekable, if set to 0 it is assumed not to be seekable,
253 if set to -1 it will try to autodetect if it is seekable. Default
257 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
260 Set a specific content type for the POST messages.
263 set HTTP proxy to tunnel through e.g. http://example.com:1234
266 Set custom HTTP headers, can override built in default headers. The
267 value must be a string encoding the headers.
269 @item multiple_requests
270 Use persistent connections if set to 1, default is 0.
273 Set custom HTTP post data.
277 Override the User-Agent header. If not specified the protocol will use a
278 string describing the libavformat build. ("Lavf/<version>")
281 Set timeout in microseconds of socket I/O operations used by the underlying low level
282 operation. By default it is set to -1, which means that the timeout is
285 @item reconnect_at_eof
286 If set then eof is treated like an error and causes reconnection, this is useful
287 for live / endless streams.
289 @item reconnect_streamed
290 If set then even streamed/non seekable streams will be reconnected on errors.
292 @item reconnect_delay_max
293 Sets the maximum delay in seconds after which to give up reconnecting
296 Export the MIME type.
299 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
300 supports this, the metadata has to be retrieved by the application by reading
301 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
304 @item icy_metadata_headers
305 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
306 headers, separated by newline characters.
308 @item icy_metadata_packet
309 If the server supports ICY metadata, and @option{icy} was set to 1, this
310 contains the last non-empty metadata packet sent by the server. It should be
311 polled in regular intervals by applications interested in mid-stream metadata
315 Set the cookies to be sent in future requests. The format of each cookie is the
316 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
317 delimited by a newline character.
320 Set initial byte offset.
323 Try to limit the request to bytes preceding this offset.
326 When used as a client option it sets the HTTP method for the request.
328 When used as a server option it sets the HTTP method that is going to be
329 expected from the client(s).
330 If the expected and the received HTTP method do not match the client will
331 be given a Bad Request response.
332 When unset the HTTP method is not checked for now. This will be replaced by
333 autodetection in the future.
336 If set to 1 enables experimental HTTP server. This can be used to send data when
337 used as an output option, or read data from a client with HTTP POST when used as
339 If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
340 in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
342 # Server side (sending):
343 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
345 # Client side (receiving):
346 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
348 # Client can also be done with wget:
349 wget http://@var{server}:@var{port} -O somefile.ogg
351 # Server side (receiving):
352 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
354 # Client side (sending):
355 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
357 # Client can also be done with wget:
358 wget --post-file=somefile.ogg http://@var{server}:@var{port}
363 @subsection HTTP Cookies
365 Some HTTP requests will be denied unless cookie values are passed in with the
366 request. The @option{cookies} option allows these cookies to be specified. At
367 the very least, each cookie must specify a value along with a path and domain.
368 HTTP requests that match both the domain and path will automatically include the
369 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
372 The required syntax to play a stream specifying a cookie is:
374 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
379 Icecast protocol (stream to Icecast servers)
381 This protocol accepts the following options:
385 Set the stream genre.
390 @item ice_description
391 Set the stream description.
394 Set the stream website URL.
397 Set if the stream should be public.
398 The default is 0 (not public).
401 Override the User-Agent header. If not specified a string of the form
402 "Lavf/<version>" will be used.
405 Set the Icecast mountpoint password.
408 Set the stream content type. This must be set if it is different from
412 This enables support for Icecast versions < 2.4.0, that do not support the
413 HTTP PUT method but the SOURCE method.
418 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
423 MMS (Microsoft Media Server) protocol over TCP.
427 MMS (Microsoft Media Server) protocol over HTTP.
429 The required syntax is:
431 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
438 Computes the MD5 hash of the data to be written, and on close writes
439 this to the designated output or stdout if none is specified. It can
440 be used to test muxers without writing an actual file.
442 Some examples follow.
444 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
445 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
447 # Write the MD5 hash of the encoded AVI file to stdout.
448 ffmpeg -i input.flv -f avi -y md5:
451 Note that some formats (typically MOV) require the output protocol to
452 be seekable, so they will fail with the MD5 output protocol.
456 UNIX pipe access protocol.
458 Read and write from UNIX pipes.
460 The accepted syntax is:
465 @var{number} is the number corresponding to the file descriptor of the
466 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
467 is not specified, by default the stdout file descriptor will be used
468 for writing, stdin for reading.
470 For example to read from stdin with @command{ffmpeg}:
472 cat test.wav | ffmpeg -i pipe:0
473 # ...this is the same as...
474 cat test.wav | ffmpeg -i pipe:
477 For writing to stdout with @command{ffmpeg}:
479 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
480 # ...this is the same as...
481 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
484 This protocol accepts the following options:
488 Set I/O operation maximum block size, in bytes. Default value is
489 @code{INT_MAX}, which results in not limiting the requested block size.
490 Setting this value reasonably low improves user termination request reaction
491 time, which is valuable if data transmission is slow.
494 Note that some formats (typically MOV), require the output protocol to
495 be seekable, so they will fail with the pipe output protocol.
499 Real-Time Messaging Protocol.
501 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
502 content across a TCP/IP network.
504 The required syntax is:
506 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
509 The accepted parameters are:
513 An optional username (mostly for publishing).
516 An optional password (mostly for publishing).
519 The address of the RTMP server.
522 The number of the TCP port to use (by default is 1935).
525 It is the name of the application to access. It usually corresponds to
526 the path where the application is installed on the RTMP server
527 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
528 the value parsed from the URI through the @code{rtmp_app} option, too.
531 It is the path or name of the resource to play with reference to the
532 application specified in @var{app}, may be prefixed by "mp4:". You
533 can override the value parsed from the URI through the @code{rtmp_playpath}
537 Act as a server, listening for an incoming connection.
540 Maximum time to wait for the incoming connection. Implies listen.
543 Additionally, the following parameters can be set via command line options
544 (or in code via @code{AVOption}s):
548 Name of application to connect on the RTMP server. This option
549 overrides the parameter specified in the URI.
552 Set the client buffer time in milliseconds. The default is 3000.
555 Extra arbitrary AMF connection parameters, parsed from a string,
556 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
557 Each value is prefixed by a single character denoting the type,
558 B for Boolean, N for number, S for string, O for object, or Z for null,
559 followed by a colon. For Booleans the data must be either 0 or 1 for
560 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
561 1 to end or begin an object, respectively. Data items in subobjects may
562 be named, by prefixing the type with 'N' and specifying the name before
563 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
564 times to construct arbitrary AMF sequences.
567 Version of the Flash plugin used to run the SWF player. The default
568 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
569 <libavformat version>).)
571 @item rtmp_flush_interval
572 Number of packets flushed in the same request (RTMPT only). The default
576 Specify that the media is a live stream. No resuming or seeking in
577 live streams is possible. The default value is @code{any}, which means the
578 subscriber first tries to play the live stream specified in the
579 playpath. If a live stream of that name is not found, it plays the
580 recorded stream. The other possible values are @code{live} and
584 URL of the web page in which the media was embedded. By default no
588 Stream identifier to play or to publish. This option overrides the
589 parameter specified in the URI.
592 Name of live stream to subscribe to. By default no value will be sent.
593 It is only sent if the option is specified or if rtmp_live
597 SHA256 hash of the decompressed SWF file (32 bytes).
600 Size of the decompressed SWF file, required for SWFVerification.
603 URL of the SWF player for the media. By default no value will be sent.
606 URL to player swf file, compute hash/size automatically.
609 URL of the target stream. Defaults to proto://host[:port]/app.
613 For example to read with @command{ffplay} a multimedia resource named
614 "sample" from the application "vod" from an RTMP server "myserver":
616 ffplay rtmp://myserver/vod/sample
619 To publish to a password protected server, passing the playpath and
620 app names separately:
622 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
627 Encrypted Real-Time Messaging Protocol.
629 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
630 streaming multimedia content within standard cryptographic primitives,
631 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
636 Real-Time Messaging Protocol over a secure SSL connection.
638 The Real-Time Messaging Protocol (RTMPS) is used for streaming
639 multimedia content across an encrypted connection.
643 Real-Time Messaging Protocol tunneled through HTTP.
645 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
646 for streaming multimedia content within HTTP requests to traverse
651 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
653 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
654 is used for streaming multimedia content within HTTP requests to traverse
659 Real-Time Messaging Protocol tunneled through HTTPS.
661 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
662 for streaming multimedia content within HTTPS requests to traverse
665 @section libsmbclient
667 libsmbclient permits one to manipulate CIFS/SMB network resources.
669 Following syntax is required.
672 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
675 This protocol accepts the following options.
679 Set timeout in miliseconds of socket I/O operations used by the underlying
680 low level operation. By default it is set to -1, which means that the timeout
684 Truncate existing files on write, if set to 1. A value of 0 prevents
685 truncating. Default value is 1.
688 Set the workgroup used for making connections. By default workgroup is not specified.
692 For more information see: @url{http://www.samba.org/}.
696 Secure File Transfer Protocol via libssh
698 Read from or write to remote resources using SFTP protocol.
700 Following syntax is required.
703 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
706 This protocol accepts the following options.
710 Set timeout of socket I/O operations used by the underlying low level
711 operation. By default it is set to -1, which means that the timeout
715 Truncate existing files on write, if set to 1. A value of 0 prevents
716 truncating. Default value is 1.
719 Specify the path of the file containing private key to use during authorization.
720 By default libssh searches for keys in the @file{~/.ssh/} directory.
724 Example: Play a file stored on remote server.
727 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
730 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
732 Real-Time Messaging Protocol and its variants supported through
735 Requires the presence of the librtmp headers and library during
736 configuration. You need to explicitly configure the build with
737 "--enable-librtmp". If enabled this will replace the native RTMP
740 This protocol provides most client functions and a few server
741 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
742 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
743 variants of these encrypted types (RTMPTE, RTMPTS).
745 The required syntax is:
747 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
750 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
751 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
752 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
753 meaning as specified for the RTMP native protocol.
754 @var{options} contains a list of space-separated options of the form
757 See the librtmp manual page (man 3 librtmp) for more information.
759 For example, to stream a file in real-time to an RTMP server using
762 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
765 To play the same stream using @command{ffplay}:
767 ffplay "rtmp://myserver/live/mystream live=1"
772 Real-time Transport Protocol.
774 The required syntax for an RTP URL is:
775 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
777 @var{port} specifies the RTP port to use.
779 The following URL options are supported:
784 Set the TTL (Time-To-Live) value (for multicast only).
786 @item rtcpport=@var{n}
787 Set the remote RTCP port to @var{n}.
789 @item localrtpport=@var{n}
790 Set the local RTP port to @var{n}.
792 @item localrtcpport=@var{n}'
793 Set the local RTCP port to @var{n}.
795 @item pkt_size=@var{n}
796 Set max packet size (in bytes) to @var{n}.
799 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
802 @item sources=@var{ip}[,@var{ip}]
803 List allowed source IP addresses.
805 @item block=@var{ip}[,@var{ip}]
806 List disallowed (blocked) source IP addresses.
808 @item write_to_source=0|1
809 Send packets to the source address of the latest received packet (if
810 set to 1) or to a default remote address (if set to 0).
812 @item localport=@var{n}
813 Set the local RTP port to @var{n}.
815 This is a deprecated option. Instead, @option{localrtpport} should be
825 If @option{rtcpport} is not set the RTCP port will be set to the RTP
829 If @option{localrtpport} (the local RTP port) is not set any available
830 port will be used for the local RTP and RTCP ports.
833 If @option{localrtcpport} (the local RTCP port) is not set it will be
834 set to the local RTP port value plus 1.
839 Real-Time Streaming Protocol.
841 RTSP is not technically a protocol handler in libavformat, it is a demuxer
842 and muxer. The demuxer supports both normal RTSP (with data transferred
843 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
844 data transferred over RDT).
846 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
847 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
848 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
850 The required syntax for a RTSP url is:
852 rtsp://@var{hostname}[:@var{port}]/@var{path}
855 Options can be set on the @command{ffmpeg}/@command{ffplay} command
856 line, or set in code via @code{AVOption}s or in
857 @code{avformat_open_input}.
859 The following options are supported.
863 Do not start playing the stream immediately if set to 1. Default value
867 Set RTSP transport protocols.
869 It accepts the following values:
872 Use UDP as lower transport protocol.
875 Use TCP (interleaving within the RTSP control channel) as lower
879 Use UDP multicast as lower transport protocol.
882 Use HTTP tunneling as lower transport protocol, which is useful for
886 Multiple lower transport protocols may be specified, in that case they are
887 tried one at a time (if the setup of one fails, the next one is tried).
888 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
893 The following values are accepted:
896 Accept packets only from negotiated peer address and port.
898 Act as a server, listening for an incoming connection.
900 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
903 Default value is @samp{none}.
905 @item allowed_media_types
906 Set media types to accept from the server.
908 The following flags are accepted:
915 By default it accepts all media types.
918 Set minimum local UDP port. Default value is 5000.
921 Set maximum local UDP port. Default value is 65000.
924 Set maximum timeout (in seconds) to wait for incoming connections.
926 A value of -1 means infinite (default). This option implies the
927 @option{rtsp_flags} set to @samp{listen}.
929 @item reorder_queue_size
930 Set number of packets to buffer for handling of reordered packets.
933 Set socket TCP I/O timeout in microseconds.
936 Override User-Agent header. If not specified, it defaults to the
937 libavformat identifier string.
940 When receiving data over UDP, the demuxer tries to reorder received packets
941 (since they may arrive out of order, or packets may get lost totally). This
942 can be disabled by setting the maximum demuxing delay to zero (via
943 the @code{max_delay} field of AVFormatContext).
945 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
946 streams to display can be chosen with @code{-vst} @var{n} and
947 @code{-ast} @var{n} for video and audio respectively, and can be switched
948 on the fly by pressing @code{v} and @code{a}.
952 The following examples all make use of the @command{ffplay} and
953 @command{ffmpeg} tools.
957 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
959 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
963 Watch a stream tunneled over HTTP:
965 ffplay -rtsp_transport http rtsp://server/video.mp4
969 Send a stream in realtime to a RTSP server, for others to watch:
971 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
975 Receive a stream in realtime:
977 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
983 Session Announcement Protocol (RFC 2974). This is not technically a
984 protocol handler in libavformat, it is a muxer and demuxer.
985 It is used for signalling of RTP streams, by announcing the SDP for the
986 streams regularly on a separate port.
990 The syntax for a SAP url given to the muxer is:
992 sap://@var{destination}[:@var{port}][?@var{options}]
995 The RTP packets are sent to @var{destination} on port @var{port},
996 or to port 5004 if no port is specified.
997 @var{options} is a @code{&}-separated list. The following options
1002 @item announce_addr=@var{address}
1003 Specify the destination IP address for sending the announcements to.
1004 If omitted, the announcements are sent to the commonly used SAP
1005 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1006 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1008 @item announce_port=@var{port}
1009 Specify the port to send the announcements on, defaults to
1010 9875 if not specified.
1013 Specify the time to live value for the announcements and RTP packets,
1016 @item same_port=@var{0|1}
1017 If set to 1, send all RTP streams on the same port pair. If zero (the
1018 default), all streams are sent on unique ports, with each stream on a
1019 port 2 numbers higher than the previous.
1020 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1021 The RTP stack in libavformat for receiving requires all streams to be sent
1025 Example command lines follow.
1027 To broadcast a stream on the local subnet, for watching in VLC:
1030 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1033 Similarly, for watching in @command{ffplay}:
1036 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1039 And for watching in @command{ffplay}, over IPv6:
1042 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1047 The syntax for a SAP url given to the demuxer is:
1049 sap://[@var{address}][:@var{port}]
1052 @var{address} is the multicast address to listen for announcements on,
1053 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1054 is the port that is listened on, 9875 if omitted.
1056 The demuxers listens for announcements on the given address and port.
1057 Once an announcement is received, it tries to receive that particular stream.
1059 Example command lines follow.
1061 To play back the first stream announced on the normal SAP multicast address:
1067 To play back the first stream announced on one the default IPv6 SAP multicast address:
1070 ffplay sap://[ff0e::2:7ffe]
1075 Stream Control Transmission Protocol.
1077 The accepted URL syntax is:
1079 sctp://@var{host}:@var{port}[?@var{options}]
1082 The protocol accepts the following options:
1085 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1088 Set the maximum number of streams. By default no limit is set.
1093 Secure Real-time Transport Protocol.
1095 The accepted options are:
1098 @item srtp_out_suite
1099 Select input and output encoding suites.
1103 @item AES_CM_128_HMAC_SHA1_80
1104 @item SRTP_AES128_CM_HMAC_SHA1_80
1105 @item AES_CM_128_HMAC_SHA1_32
1106 @item SRTP_AES128_CM_HMAC_SHA1_32
1109 @item srtp_in_params
1110 @item srtp_out_params
1111 Set input and output encoding parameters, which are expressed by a
1112 base64-encoded representation of a binary block. The first 16 bytes of
1113 this binary block are used as master key, the following 14 bytes are
1114 used as master salt.
1119 Virtually extract a segment of a file or another stream.
1120 The underlying stream must be seekable.
1125 Start offset of the extracted segment, in bytes.
1127 End offset of the extracted segment, in bytes.
1132 Extract a chapter from a DVD VOB file (start and end sectors obtained
1133 externally and multiplied by 2048):
1135 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1138 Play an AVI file directly from a TAR archive:
1140 subfile,,start,183241728,end,366490624,,:archive.tar
1145 Transmission Control Protocol.
1147 The required syntax for a TCP url is:
1149 tcp://@var{hostname}:@var{port}[?@var{options}]
1152 @var{options} contains a list of &-separated options of the form
1153 @var{key}=@var{val}.
1155 The list of supported options follows.
1158 @item listen=@var{1|0}
1159 Listen for an incoming connection. Default value is 0.
1161 @item timeout=@var{microseconds}
1162 Set raise error timeout, expressed in microseconds.
1164 This option is only relevant in read mode: if no data arrived in more
1165 than this time interval, raise error.
1167 @item listen_timeout=@var{milliseconds}
1168 Set listen timeout, expressed in milliseconds.
1170 @item recv_buffer_size=@var{bytes}
1171 Set receive buffer size, expressed bytes.
1173 @item send_buffer_size=@var{bytes}
1174 Set send buffer size, expressed bytes.
1177 The following example shows how to setup a listening TCP connection
1178 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1180 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1181 ffplay tcp://@var{hostname}:@var{port}
1186 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1188 The required syntax for a TLS/SSL url is:
1190 tls://@var{hostname}:@var{port}[?@var{options}]
1193 The following parameters can be set via command line options
1194 (or in code via @code{AVOption}s):
1198 @item ca_file, cafile=@var{filename}
1199 A file containing certificate authority (CA) root certificates to treat
1200 as trusted. If the linked TLS library contains a default this might not
1201 need to be specified for verification to work, but not all libraries and
1202 setups have defaults built in.
1203 The file must be in OpenSSL PEM format.
1205 @item tls_verify=@var{1|0}
1206 If enabled, try to verify the peer that we are communicating with.
1207 Note, if using OpenSSL, this currently only makes sure that the
1208 peer certificate is signed by one of the root certificates in the CA
1209 database, but it does not validate that the certificate actually
1210 matches the host name we are trying to connect to. (With GnuTLS,
1211 the host name is validated as well.)
1213 This is disabled by default since it requires a CA database to be
1214 provided by the caller in many cases.
1216 @item cert_file, cert=@var{filename}
1217 A file containing a certificate to use in the handshake with the peer.
1218 (When operating as server, in listen mode, this is more often required
1219 by the peer, while client certificates only are mandated in certain
1222 @item key_file, key=@var{filename}
1223 A file containing the private key for the certificate.
1225 @item listen=@var{1|0}
1226 If enabled, listen for connections on the provided port, and assume
1227 the server role in the handshake instead of the client role.
1231 Example command lines:
1233 To create a TLS/SSL server that serves an input stream.
1236 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1239 To play back a stream from the TLS/SSL server using @command{ffplay}:
1242 ffplay tls://@var{hostname}:@var{port}
1247 User Datagram Protocol.
1249 The required syntax for an UDP URL is:
1251 udp://@var{hostname}:@var{port}[?@var{options}]
1254 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1256 In case threading is enabled on the system, a circular buffer is used
1257 to store the incoming data, which allows one to reduce loss of data due to
1258 UDP socket buffer overruns. The @var{fifo_size} and
1259 @var{overrun_nonfatal} options are related to this buffer.
1261 The list of supported options follows.
1264 @item buffer_size=@var{size}
1265 Set the UDP maximum socket buffer size in bytes. This is used to set either
1266 the receive or send buffer size, depending on what the socket is used for.
1267 Default is 64KB. See also @var{fifo_size}.
1269 @item localport=@var{port}
1270 Override the local UDP port to bind with.
1272 @item localaddr=@var{addr}
1273 Choose the local IP address. This is useful e.g. if sending multicast
1274 and the host has multiple interfaces, where the user can choose
1275 which interface to send on by specifying the IP address of that interface.
1277 @item pkt_size=@var{size}
1278 Set the size in bytes of UDP packets.
1280 @item reuse=@var{1|0}
1281 Explicitly allow or disallow reusing UDP sockets.
1284 Set the time to live value (for multicast only).
1286 @item connect=@var{1|0}
1287 Initialize the UDP socket with @code{connect()}. In this case, the
1288 destination address can't be changed with ff_udp_set_remote_url later.
1289 If the destination address isn't known at the start, this option can
1290 be specified in ff_udp_set_remote_url, too.
1291 This allows finding out the source address for the packets with getsockname,
1292 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1293 unreachable" is received.
1294 For receiving, this gives the benefit of only receiving packets from
1295 the specified peer address/port.
1297 @item sources=@var{address}[,@var{address}]
1298 Only receive packets sent to the multicast group from one of the
1299 specified sender IP addresses.
1301 @item block=@var{address}[,@var{address}]
1302 Ignore packets sent to the multicast group from the specified
1303 sender IP addresses.
1305 @item fifo_size=@var{units}
1306 Set the UDP receiving circular buffer size, expressed as a number of
1307 packets with size of 188 bytes. If not specified defaults to 7*4096.
1309 @item overrun_nonfatal=@var{1|0}
1310 Survive in case of UDP receiving circular buffer overrun. Default
1313 @item timeout=@var{microseconds}
1314 Set raise error timeout, expressed in microseconds.
1316 This option is only relevant in read mode: if no data arrived in more
1317 than this time interval, raise error.
1319 @item broadcast=@var{1|0}
1320 Explicitly allow or disallow UDP broadcasting.
1322 Note that broadcasting may not work properly on networks having
1323 a broadcast storm protection.
1326 @subsection Examples
1330 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1332 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1336 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1337 sized UDP packets, using a large input buffer:
1339 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1343 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1345 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1353 The required syntax for a Unix socket URL is:
1356 unix://@var{filepath}
1359 The following parameters can be set via command line options
1360 (or in code via @code{AVOption}s):
1366 Create the Unix socket in listening mode.
1369 @c man end PROTOCOLS