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 If set to 1 enables experimental HTTP server. This can be used to send data when
283 used as an output option, or read data from a client with HTTP POST when used as
286 # Server side (sending):
287 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
289 # Client side (receiving):
290 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
292 # Client can also be done with wget:
293 wget http://@var{server}:@var{port} -O somefile.ogg
295 # Server side (receiving):
296 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
298 # Client side (sending):
299 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
301 # Client can also be done with wget:
302 wget --post-file=somefile.ogg http://@var{server}:@var{port}
307 @subsection HTTP Cookies
309 Some HTTP requests will be denied unless cookie values are passed in with the
310 request. The @option{cookies} option allows these cookies to be specified. At
311 the very least, each cookie must specify a value along with a path and domain.
312 HTTP requests that match both the domain and path will automatically include the
313 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
316 The required syntax to play a stream specifying a cookie is:
318 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
323 Icecast protocol (stream to Icecast servers)
325 This protocol accepts the following options:
329 Set the stream genre.
334 @item ice_description
335 Set the stream description.
338 Set the stream website URL.
341 Set if the stream should be public.
342 The default is 0 (not public).
345 Override the User-Agent header. If not specified a string of the form
346 "Lavf/<version>" will be used.
349 Set the Icecast mountpoint password.
352 Set the stream content type. This must be set if it is different from
356 This enables support for Icecast versions < 2.4.0, that do not support the
357 HTTP PUT method but the SOURCE method.
362 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
367 MMS (Microsoft Media Server) protocol over TCP.
371 MMS (Microsoft Media Server) protocol over HTTP.
373 The required syntax is:
375 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
382 Computes the MD5 hash of the data to be written, and on close writes
383 this to the designated output or stdout if none is specified. It can
384 be used to test muxers without writing an actual file.
386 Some examples follow.
388 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
389 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
391 # Write the MD5 hash of the encoded AVI file to stdout.
392 ffmpeg -i input.flv -f avi -y md5:
395 Note that some formats (typically MOV) require the output protocol to
396 be seekable, so they will fail with the MD5 output protocol.
400 UNIX pipe access protocol.
402 Read and write from UNIX pipes.
404 The accepted syntax is:
409 @var{number} is the number corresponding to the file descriptor of the
410 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
411 is not specified, by default the stdout file descriptor will be used
412 for writing, stdin for reading.
414 For example to read from stdin with @command{ffmpeg}:
416 cat test.wav | ffmpeg -i pipe:0
417 # ...this is the same as...
418 cat test.wav | ffmpeg -i pipe:
421 For writing to stdout with @command{ffmpeg}:
423 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
424 # ...this is the same as...
425 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
428 This protocol accepts the following options:
432 Set I/O operation maximum block size, in bytes. Default value is
433 @code{INT_MAX}, which results in not limiting the requested block size.
434 Setting this value reasonably low improves user termination request reaction
435 time, which is valuable if data transmission is slow.
438 Note that some formats (typically MOV), require the output protocol to
439 be seekable, so they will fail with the pipe output protocol.
443 Real-Time Messaging Protocol.
445 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
446 content across a TCP/IP network.
448 The required syntax is:
450 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
453 The accepted parameters are:
457 An optional username (mostly for publishing).
460 An optional password (mostly for publishing).
463 The address of the RTMP server.
466 The number of the TCP port to use (by default is 1935).
469 It is the name of the application to access. It usually corresponds to
470 the path where the application is installed on the RTMP server
471 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
472 the value parsed from the URI through the @code{rtmp_app} option, too.
475 It is the path or name of the resource to play with reference to the
476 application specified in @var{app}, may be prefixed by "mp4:". You
477 can override the value parsed from the URI through the @code{rtmp_playpath}
481 Act as a server, listening for an incoming connection.
484 Maximum time to wait for the incoming connection. Implies listen.
487 Additionally, the following parameters can be set via command line options
488 (or in code via @code{AVOption}s):
492 Name of application to connect on the RTMP server. This option
493 overrides the parameter specified in the URI.
496 Set the client buffer time in milliseconds. The default is 3000.
499 Extra arbitrary AMF connection parameters, parsed from a string,
500 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
501 Each value is prefixed by a single character denoting the type,
502 B for Boolean, N for number, S for string, O for object, or Z for null,
503 followed by a colon. For Booleans the data must be either 0 or 1 for
504 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
505 1 to end or begin an object, respectively. Data items in subobjects may
506 be named, by prefixing the type with 'N' and specifying the name before
507 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
508 times to construct arbitrary AMF sequences.
511 Version of the Flash plugin used to run the SWF player. The default
512 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
513 <libavformat version>).)
515 @item rtmp_flush_interval
516 Number of packets flushed in the same request (RTMPT only). The default
520 Specify that the media is a live stream. No resuming or seeking in
521 live streams is possible. The default value is @code{any}, which means the
522 subscriber first tries to play the live stream specified in the
523 playpath. If a live stream of that name is not found, it plays the
524 recorded stream. The other possible values are @code{live} and
528 URL of the web page in which the media was embedded. By default no
532 Stream identifier to play or to publish. This option overrides the
533 parameter specified in the URI.
536 Name of live stream to subscribe to. By default no value will be sent.
537 It is only sent if the option is specified or if rtmp_live
541 SHA256 hash of the decompressed SWF file (32 bytes).
544 Size of the decompressed SWF file, required for SWFVerification.
547 URL of the SWF player for the media. By default no value will be sent.
550 URL to player swf file, compute hash/size automatically.
553 URL of the target stream. Defaults to proto://host[:port]/app.
557 For example to read with @command{ffplay} a multimedia resource named
558 "sample" from the application "vod" from an RTMP server "myserver":
560 ffplay rtmp://myserver/vod/sample
563 To publish to a password protected server, passing the playpath and
564 app names separately:
566 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
571 Encrypted Real-Time Messaging Protocol.
573 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
574 streaming multimedia content within standard cryptographic primitives,
575 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
580 Real-Time Messaging Protocol over a secure SSL connection.
582 The Real-Time Messaging Protocol (RTMPS) is used for streaming
583 multimedia content across an encrypted connection.
587 Real-Time Messaging Protocol tunneled through HTTP.
589 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
590 for streaming multimedia content within HTTP requests to traverse
595 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
597 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
598 is used for streaming multimedia content within HTTP requests to traverse
603 Real-Time Messaging Protocol tunneled through HTTPS.
605 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
606 for streaming multimedia content within HTTPS requests to traverse
609 @section libsmbclient
611 libsmbclient permits one to manipulate CIFS/SMB network resources.
613 Following syntax is required.
616 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
619 This protocol accepts the following options.
623 Set timeout in miliseconds of socket I/O operations used by the underlying
624 low level operation. By default it is set to -1, which means that the timeout
628 Truncate existing files on write, if set to 1. A value of 0 prevents
629 truncating. Default value is 1.
632 Set the workgroup used for making connections. By default workgroup is not specified.
636 For more information see: @url{http://www.samba.org/}.
640 Secure File Transfer Protocol via libssh
642 Read from or write to remote resources using SFTP protocol.
644 Following syntax is required.
647 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
650 This protocol accepts the following options.
654 Set timeout of socket I/O operations used by the underlying low level
655 operation. By default it is set to -1, which means that the timeout
659 Truncate existing files on write, if set to 1. A value of 0 prevents
660 truncating. Default value is 1.
663 Specify the path of the file containing private key to use during authorization.
664 By default libssh searches for keys in the @file{~/.ssh/} directory.
668 Example: Play a file stored on remote server.
671 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
674 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
676 Real-Time Messaging Protocol and its variants supported through
679 Requires the presence of the librtmp headers and library during
680 configuration. You need to explicitly configure the build with
681 "--enable-librtmp". If enabled this will replace the native RTMP
684 This protocol provides most client functions and a few server
685 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
686 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
687 variants of these encrypted types (RTMPTE, RTMPTS).
689 The required syntax is:
691 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
694 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
695 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
696 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
697 meaning as specified for the RTMP native protocol.
698 @var{options} contains a list of space-separated options of the form
701 See the librtmp manual page (man 3 librtmp) for more information.
703 For example, to stream a file in real-time to an RTMP server using
706 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
709 To play the same stream using @command{ffplay}:
711 ffplay "rtmp://myserver/live/mystream live=1"
716 Real-time Transport Protocol.
718 The required syntax for an RTP URL is:
719 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
721 @var{port} specifies the RTP port to use.
723 The following URL options are supported:
728 Set the TTL (Time-To-Live) value (for multicast only).
730 @item rtcpport=@var{n}
731 Set the remote RTCP port to @var{n}.
733 @item localrtpport=@var{n}
734 Set the local RTP port to @var{n}.
736 @item localrtcpport=@var{n}'
737 Set the local RTCP port to @var{n}.
739 @item pkt_size=@var{n}
740 Set max packet size (in bytes) to @var{n}.
743 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
746 @item sources=@var{ip}[,@var{ip}]
747 List allowed source IP addresses.
749 @item block=@var{ip}[,@var{ip}]
750 List disallowed (blocked) source IP addresses.
752 @item write_to_source=0|1
753 Send packets to the source address of the latest received packet (if
754 set to 1) or to a default remote address (if set to 0).
756 @item localport=@var{n}
757 Set the local RTP port to @var{n}.
759 This is a deprecated option. Instead, @option{localrtpport} should be
769 If @option{rtcpport} is not set the RTCP port will be set to the RTP
773 If @option{localrtpport} (the local RTP port) is not set any available
774 port will be used for the local RTP and RTCP ports.
777 If @option{localrtcpport} (the local RTCP port) is not set it will be
778 set to the local RTP port value plus 1.
783 Real-Time Streaming Protocol.
785 RTSP is not technically a protocol handler in libavformat, it is a demuxer
786 and muxer. The demuxer supports both normal RTSP (with data transferred
787 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
788 data transferred over RDT).
790 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
791 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
792 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
794 The required syntax for a RTSP url is:
796 rtsp://@var{hostname}[:@var{port}]/@var{path}
799 Options can be set on the @command{ffmpeg}/@command{ffplay} command
800 line, or set in code via @code{AVOption}s or in
801 @code{avformat_open_input}.
803 The following options are supported.
807 Do not start playing the stream immediately if set to 1. Default value
811 Set RTSP transport protocols.
813 It accepts the following values:
816 Use UDP as lower transport protocol.
819 Use TCP (interleaving within the RTSP control channel) as lower
823 Use UDP multicast as lower transport protocol.
826 Use HTTP tunneling as lower transport protocol, which is useful for
830 Multiple lower transport protocols may be specified, in that case they are
831 tried one at a time (if the setup of one fails, the next one is tried).
832 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
837 The following values are accepted:
840 Accept packets only from negotiated peer address and port.
842 Act as a server, listening for an incoming connection.
844 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
847 Default value is @samp{none}.
849 @item allowed_media_types
850 Set media types to accept from the server.
852 The following flags are accepted:
859 By default it accepts all media types.
862 Set minimum local UDP port. Default value is 5000.
865 Set maximum local UDP port. Default value is 65000.
868 Set maximum timeout (in seconds) to wait for incoming connections.
870 A value of -1 means infinite (default). This option implies the
871 @option{rtsp_flags} set to @samp{listen}.
873 @item reorder_queue_size
874 Set number of packets to buffer for handling of reordered packets.
877 Set socket TCP I/O timeout in microseconds.
880 Override User-Agent header. If not specified, it defaults to the
881 libavformat identifier string.
884 When receiving data over UDP, the demuxer tries to reorder received packets
885 (since they may arrive out of order, or packets may get lost totally). This
886 can be disabled by setting the maximum demuxing delay to zero (via
887 the @code{max_delay} field of AVFormatContext).
889 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
890 streams to display can be chosen with @code{-vst} @var{n} and
891 @code{-ast} @var{n} for video and audio respectively, and can be switched
892 on the fly by pressing @code{v} and @code{a}.
896 The following examples all make use of the @command{ffplay} and
897 @command{ffmpeg} tools.
901 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
903 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
907 Watch a stream tunneled over HTTP:
909 ffplay -rtsp_transport http rtsp://server/video.mp4
913 Send a stream in realtime to a RTSP server, for others to watch:
915 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
919 Receive a stream in realtime:
921 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
927 Session Announcement Protocol (RFC 2974). This is not technically a
928 protocol handler in libavformat, it is a muxer and demuxer.
929 It is used for signalling of RTP streams, by announcing the SDP for the
930 streams regularly on a separate port.
934 The syntax for a SAP url given to the muxer is:
936 sap://@var{destination}[:@var{port}][?@var{options}]
939 The RTP packets are sent to @var{destination} on port @var{port},
940 or to port 5004 if no port is specified.
941 @var{options} is a @code{&}-separated list. The following options
946 @item announce_addr=@var{address}
947 Specify the destination IP address for sending the announcements to.
948 If omitted, the announcements are sent to the commonly used SAP
949 announcement multicast address 224.2.127.254 (sap.mcast.net), or
950 ff0e::2:7ffe if @var{destination} is an IPv6 address.
952 @item announce_port=@var{port}
953 Specify the port to send the announcements on, defaults to
954 9875 if not specified.
957 Specify the time to live value for the announcements and RTP packets,
960 @item same_port=@var{0|1}
961 If set to 1, send all RTP streams on the same port pair. If zero (the
962 default), all streams are sent on unique ports, with each stream on a
963 port 2 numbers higher than the previous.
964 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
965 The RTP stack in libavformat for receiving requires all streams to be sent
969 Example command lines follow.
971 To broadcast a stream on the local subnet, for watching in VLC:
974 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
977 Similarly, for watching in @command{ffplay}:
980 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
983 And for watching in @command{ffplay}, over IPv6:
986 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
991 The syntax for a SAP url given to the demuxer is:
993 sap://[@var{address}][:@var{port}]
996 @var{address} is the multicast address to listen for announcements on,
997 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
998 is the port that is listened on, 9875 if omitted.
1000 The demuxers listens for announcements on the given address and port.
1001 Once an announcement is received, it tries to receive that particular stream.
1003 Example command lines follow.
1005 To play back the first stream announced on the normal SAP multicast address:
1011 To play back the first stream announced on one the default IPv6 SAP multicast address:
1014 ffplay sap://[ff0e::2:7ffe]
1019 Stream Control Transmission Protocol.
1021 The accepted URL syntax is:
1023 sctp://@var{host}:@var{port}[?@var{options}]
1026 The protocol accepts the following options:
1029 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1032 Set the maximum number of streams. By default no limit is set.
1037 Secure Real-time Transport Protocol.
1039 The accepted options are:
1042 @item srtp_out_suite
1043 Select input and output encoding suites.
1047 @item AES_CM_128_HMAC_SHA1_80
1048 @item SRTP_AES128_CM_HMAC_SHA1_80
1049 @item AES_CM_128_HMAC_SHA1_32
1050 @item SRTP_AES128_CM_HMAC_SHA1_32
1053 @item srtp_in_params
1054 @item srtp_out_params
1055 Set input and output encoding parameters, which are expressed by a
1056 base64-encoded representation of a binary block. The first 16 bytes of
1057 this binary block are used as master key, the following 14 bytes are
1058 used as master salt.
1063 Virtually extract a segment of a file or another stream.
1064 The underlying stream must be seekable.
1069 Start offset of the extracted segment, in bytes.
1071 End offset of the extracted segment, in bytes.
1076 Extract a chapter from a DVD VOB file (start and end sectors obtained
1077 externally and multiplied by 2048):
1079 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1082 Play an AVI file directly from a TAR archive:
1084 subfile,,start,183241728,end,366490624,,:archive.tar
1089 Transmission Control Protocol.
1091 The required syntax for a TCP url is:
1093 tcp://@var{hostname}:@var{port}[?@var{options}]
1096 @var{options} contains a list of &-separated options of the form
1097 @var{key}=@var{val}.
1099 The list of supported options follows.
1102 @item listen=@var{1|0}
1103 Listen for an incoming connection. Default value is 0.
1105 @item timeout=@var{microseconds}
1106 Set raise error timeout, expressed in microseconds.
1108 This option is only relevant in read mode: if no data arrived in more
1109 than this time interval, raise error.
1111 @item listen_timeout=@var{milliseconds}
1112 Set listen timeout, expressed in milliseconds.
1115 The following example shows how to setup a listening TCP connection
1116 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1118 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1119 ffplay tcp://@var{hostname}:@var{port}
1124 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1126 The required syntax for a TLS/SSL url is:
1128 tls://@var{hostname}:@var{port}[?@var{options}]
1131 The following parameters can be set via command line options
1132 (or in code via @code{AVOption}s):
1136 @item ca_file, cafile=@var{filename}
1137 A file containing certificate authority (CA) root certificates to treat
1138 as trusted. If the linked TLS library contains a default this might not
1139 need to be specified for verification to work, but not all libraries and
1140 setups have defaults built in.
1141 The file must be in OpenSSL PEM format.
1143 @item tls_verify=@var{1|0}
1144 If enabled, try to verify the peer that we are communicating with.
1145 Note, if using OpenSSL, this currently only makes sure that the
1146 peer certificate is signed by one of the root certificates in the CA
1147 database, but it does not validate that the certificate actually
1148 matches the host name we are trying to connect to. (With GnuTLS,
1149 the host name is validated as well.)
1151 This is disabled by default since it requires a CA database to be
1152 provided by the caller in many cases.
1154 @item cert_file, cert=@var{filename}
1155 A file containing a certificate to use in the handshake with the peer.
1156 (When operating as server, in listen mode, this is more often required
1157 by the peer, while client certificates only are mandated in certain
1160 @item key_file, key=@var{filename}
1161 A file containing the private key for the certificate.
1163 @item listen=@var{1|0}
1164 If enabled, listen for connections on the provided port, and assume
1165 the server role in the handshake instead of the client role.
1169 Example command lines:
1171 To create a TLS/SSL server that serves an input stream.
1174 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1177 To play back a stream from the TLS/SSL server using @command{ffplay}:
1180 ffplay tls://@var{hostname}:@var{port}
1185 User Datagram Protocol.
1187 The required syntax for an UDP URL is:
1189 udp://@var{hostname}:@var{port}[?@var{options}]
1192 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1194 In case threading is enabled on the system, a circular buffer is used
1195 to store the incoming data, which allows one to reduce loss of data due to
1196 UDP socket buffer overruns. The @var{fifo_size} and
1197 @var{overrun_nonfatal} options are related to this buffer.
1199 The list of supported options follows.
1202 @item buffer_size=@var{size}
1203 Set the UDP maximum socket buffer size in bytes. This is used to set either
1204 the receive or send buffer size, depending on what the socket is used for.
1205 Default is 64KB. See also @var{fifo_size}.
1207 @item localport=@var{port}
1208 Override the local UDP port to bind with.
1210 @item localaddr=@var{addr}
1211 Choose the local IP address. This is useful e.g. if sending multicast
1212 and the host has multiple interfaces, where the user can choose
1213 which interface to send on by specifying the IP address of that interface.
1215 @item pkt_size=@var{size}
1216 Set the size in bytes of UDP packets.
1218 @item reuse=@var{1|0}
1219 Explicitly allow or disallow reusing UDP sockets.
1222 Set the time to live value (for multicast only).
1224 @item connect=@var{1|0}
1225 Initialize the UDP socket with @code{connect()}. In this case, the
1226 destination address can't be changed with ff_udp_set_remote_url later.
1227 If the destination address isn't known at the start, this option can
1228 be specified in ff_udp_set_remote_url, too.
1229 This allows finding out the source address for the packets with getsockname,
1230 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1231 unreachable" is received.
1232 For receiving, this gives the benefit of only receiving packets from
1233 the specified peer address/port.
1235 @item sources=@var{address}[,@var{address}]
1236 Only receive packets sent to the multicast group from one of the
1237 specified sender IP addresses.
1239 @item block=@var{address}[,@var{address}]
1240 Ignore packets sent to the multicast group from the specified
1241 sender IP addresses.
1243 @item fifo_size=@var{units}
1244 Set the UDP receiving circular buffer size, expressed as a number of
1245 packets with size of 188 bytes. If not specified defaults to 7*4096.
1247 @item overrun_nonfatal=@var{1|0}
1248 Survive in case of UDP receiving circular buffer overrun. Default
1251 @item timeout=@var{microseconds}
1252 Set raise error timeout, expressed in microseconds.
1254 This option is only relevant in read mode: if no data arrived in more
1255 than this time interval, raise error.
1257 @item broadcast=@var{1|0}
1258 Explicitly allow or disallow UDP broadcasting.
1260 Note that broadcasting may not work properly on networks having
1261 a broadcast storm protection.
1264 @subsection Examples
1268 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1270 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1274 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1275 sized UDP packets, using a large input buffer:
1277 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1281 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1283 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1291 The required syntax for a Unix socket URL is:
1294 unix://@var{filepath}
1297 The following parameters can be set via command line options
1298 (or in code via @code{AVOption}s):
1304 Create the Unix socket in listening mode.
1307 @c man end PROTOCOLS