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1@chapter Protocols
2@c man begin PROTOCOLS
3
4Protocols are configured elements in FFmpeg that enable access to
5resources that require specific protocols.
6
7When you configure your FFmpeg build, all the supported protocols are
8enabled by default. You can list all available ones using the
9configure option "--list-protocols".
10
11You can disable all the protocols using the configure option
12"--disable-protocols", and selectively enable a protocol using the
13option "--enable-protocol=@var{PROTOCOL}", or you can disable a
14particular protocol using the option
15"--disable-protocol=@var{PROTOCOL}".
16
17The option "-protocols" of the ff* tools will display the list of
18supported protocols.
19
20A description of the currently available protocols follows.
21
22@section bluray
23
24Read BluRay playlist.
25
26The accepted options are:
27@table @option
28
29@item angle
30BluRay angle
31
32@item chapter
33Start chapter (1...N)
34
35@item playlist
36Playlist to read (BDMV/PLAYLIST/?????.mpls)
37
38@end table
39
40Examples:
41
42Read longest playlist from BluRay mounted to /mnt/bluray:
43@example
44bluray:/mnt/bluray
45@end example
46
47Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
48@example
49-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
50@end example
51
52@section cache
53
54Caching wrapper for input stream.
55
56Cache the input stream to temporary file. It brings seeking capability to live streams.
57
58@example
59cache:@var{URL}
60@end example
61
62@section concat
63
64Physical concatenation protocol.
65
66Allow to read and seek from many resource in sequence as if they were
67a unique resource.
68
69A URL accepted by this protocol has the syntax:
70@example
71concat:@var{URL1}|@var{URL2}|...|@var{URLN}
72@end example
73
74where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
75resource to be concatenated, each one possibly specifying a distinct
76protocol.
77
78For example to read a sequence of files @file{split1.mpeg},
79@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
80command:
81@example
82ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
83@end example
84
85Note that you may need to escape the character "|" which is special for
86many shells.
87
88@section crypto
89
90AES-encrypted stream reading protocol.
91
92The accepted options are:
93@table @option
94@item key
95Set the AES decryption key binary block from given hexadecimal representation.
96
97@item iv
98Set the AES decryption initialization vector binary block from given hexadecimal representation.
99@end table
100
101Accepted URL formats:
102@example
103crypto:@var{URL}
104crypto+@var{URL}
105@end example
106
107@section data
108
109Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
110
111For example, to convert a GIF file given inline with @command{ffmpeg}:
112@example
113ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
114@end example
115
116@section file
117
118File access protocol.
119
120Allow to read from or write to a file.
121
122A file URL can have the form:
123@example
124file:@var{filename}
125@end example
126
127where @var{filename} is the path of the file to read.
128
129An URL that does not have a protocol prefix will be assumed to be a
130file URL. Depending on the build, an URL that looks like a Windows
131path with the drive letter at the beginning will also be assumed to be
132a file URL (usually not the case in builds for unix-like systems).
133
134For example to read from a file @file{input.mpeg} with @command{ffmpeg}
135use the command:
136@example
137ffmpeg -i file:input.mpeg output.mpeg
138@end example
139
140This protocol accepts the following options:
141
142@table @option
143@item truncate
144Truncate existing files on write, if set to 1. A value of 0 prevents
145truncating. Default value is 1.
146
147@item blocksize
148Set I/O operation maximum block size, in bytes. Default value is
149@code{INT_MAX}, which results in not limiting the requested block size.
150Setting this value reasonably low improves user termination request reaction
151time, which is valuable for files on slow medium.
152@end table
153
154@section ftp
155
156FTP (File Transfer Protocol).
157
158Allow to read from or write to remote resources using FTP protocol.
159
160Following syntax is required.
161@example
162ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
163@end example
164
165This protocol accepts the following options.
166
167@table @option
168@item timeout
169Set timeout in microseconds of socket I/O operations used by the underlying low level
170operation. By default it is set to -1, which means that the timeout is
171not specified.
172
173@item ftp-anonymous-password
174Password used when login as anonymous user. Typically an e-mail address
175should be used.
176
177@item ftp-write-seekable
178Control seekability of connection during encoding. If set to 1 the
179resource is supposed to be seekable, if set to 0 it is assumed not
180to be seekable. Default value is 0.
181@end table
182
183NOTE: Protocol can be used as output, but it is recommended to not do
184it, unless special care is taken (tests, customized server configuration
185etc.). Different FTP servers behave in different way during seek
186operation. ff* tools may produce incomplete content due to server limitations.
187
188@section gopher
189
190Gopher protocol.
191
192@section hls
193
194Read Apple HTTP Live Streaming compliant segmented stream as
195a uniform one. The M3U8 playlists describing the segments can be
196remote HTTP resources or local files, accessed using the standard
197file protocol.
198The nested protocol is declared by specifying
199"+@var{proto}" after the hls URI scheme name, where @var{proto}
200is either "file" or "http".
201
202@example
203hls+http://host/path/to/remote/resource.m3u8
204hls+file://path/to/local/resource.m3u8
205@end example
206
207Using this protocol is discouraged - the hls demuxer should work
208just as well (if not, please report the issues) and is more complete.
209To use the hls demuxer instead, simply use the direct URLs to the
210m3u8 files.
211
212@section http
213
214HTTP (Hyper Text Transfer Protocol).
215
216This protocol accepts the following options:
217
218@table @option
219@item seekable
220Control seekability of connection. If set to 1 the resource is
221supposed to be seekable, if set to 0 it is assumed not to be seekable,
222if set to -1 it will try to autodetect if it is seekable. Default
223value is -1.
224
225@item chunked_post
226If set to 1 use chunked Transfer-Encoding for posts, default is 1.
227
228@item content_type
229Set a specific content type for the POST messages.
230
231@item headers
232Set custom HTTP headers, can override built in default headers. The
233value must be a string encoding the headers.
234
235@item multiple_requests
236Use persistent connections if set to 1, default is 0.
237
238@item post_data
239Set custom HTTP post data.
240
241@item user-agent
242@item user_agent
243Override the User-Agent header. If not specified the protocol will use a
244string describing the libavformat build. ("Lavf/<version>")
245
246@item timeout
247Set timeout in microseconds of socket I/O operations used by the underlying low level
248operation. By default it is set to -1, which means that the timeout is
249not specified.
250
251@item mime_type
252Export the MIME type.
253
254@item icy
255If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
256supports this, the metadata has to be retrieved by the application by reading
257the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
258The default is 1.
259
260@item icy_metadata_headers
261If the server supports ICY metadata, this contains the ICY-specific HTTP reply
262headers, separated by newline characters.
263
264@item icy_metadata_packet
265If the server supports ICY metadata, and @option{icy} was set to 1, this
266contains the last non-empty metadata packet sent by the server. It should be
267polled in regular intervals by applications interested in mid-stream metadata
268updates.
269
270@item cookies
271Set the cookies to be sent in future requests. The format of each cookie is the
272same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
273delimited by a newline character.
274
275@item offset
276Set initial byte offset.
277
278@item end_offset
279Try to limit the request to bytes preceding this offset.
280@end table
281
282@subsection HTTP Cookies
283
284Some HTTP requests will be denied unless cookie values are passed in with the
285request. The @option{cookies} option allows these cookies to be specified. At
286the very least, each cookie must specify a value along with a path and domain.
287HTTP requests that match both the domain and path will automatically include the
288cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
289by a newline.
290
291The required syntax to play a stream specifying a cookie is:
292@example
293ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
294@end example
295
296@section Icecast
297
298Icecast protocol (stream to Icecast servers)
299
300This protocol accepts the following options:
301
302@table @option
303@item ice_genre
304Set the stream genre.
305
306@item ice_name
307Set the stream name.
308
309@item ice_description
310Set the stream description.
311
312@item ice_url
313Set the stream website URL.
314
315@item ice_public
316Set if the stream should be public.
317The default is 0 (not public).
318
319@item user_agent
320Override the User-Agent header. If not specified a string of the form
321"Lavf/<version>" will be used.
322
323@item password
324Set the Icecast mountpoint password.
325
326@item content_type
327Set the stream content type. This must be set if it is different from
328audio/mpeg.
329
330@item legacy_icecast
331This enables support for Icecast versions < 2.4.0, that do not support the
332HTTP PUT method but the SOURCE method.
333
334@end table
335
336@example
337icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
338@end example
339
340@section mmst
341
342MMS (Microsoft Media Server) protocol over TCP.
343
344@section mmsh
345
346MMS (Microsoft Media Server) protocol over HTTP.
347
348The required syntax is:
349@example
350mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
351@end example
352
353@section md5
354
355MD5 output protocol.
356
357Computes the MD5 hash of the data to be written, and on close writes
358this to the designated output or stdout if none is specified. It can
359be used to test muxers without writing an actual file.
360
361Some examples follow.
362@example
363# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
364ffmpeg -i input.flv -f avi -y md5:output.avi.md5
365
366# Write the MD5 hash of the encoded AVI file to stdout.
367ffmpeg -i input.flv -f avi -y md5:
368@end example
369
370Note that some formats (typically MOV) require the output protocol to
371be seekable, so they will fail with the MD5 output protocol.
372
373@section pipe
374
375UNIX pipe access protocol.
376
377Allow to read and write from UNIX pipes.
378
379The accepted syntax is:
380@example
381pipe:[@var{number}]
382@end example
383
384@var{number} is the number corresponding to the file descriptor of the
385pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
386is not specified, by default the stdout file descriptor will be used
387for writing, stdin for reading.
388
389For example to read from stdin with @command{ffmpeg}:
390@example
391cat test.wav | ffmpeg -i pipe:0
392# ...this is the same as...
393cat test.wav | ffmpeg -i pipe:
394@end example
395
396For writing to stdout with @command{ffmpeg}:
397@example
398ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
399# ...this is the same as...
400ffmpeg -i test.wav -f avi pipe: | cat > test.avi
401@end example
402
403This protocol accepts the following options:
404
405@table @option
406@item blocksize
407Set I/O operation maximum block size, in bytes. Default value is
408@code{INT_MAX}, which results in not limiting the requested block size.
409Setting this value reasonably low improves user termination request reaction
410time, which is valuable if data transmission is slow.
411@end table
412
413Note that some formats (typically MOV), require the output protocol to
414be seekable, so they will fail with the pipe output protocol.
415
416@section rtmp
417
418Real-Time Messaging Protocol.
419
420The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
421content across a TCP/IP network.
422
423The required syntax is:
424@example
425rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
426@end example
427
428The accepted parameters are:
429@table @option
430
431@item username
432An optional username (mostly for publishing).
433
434@item password
435An optional password (mostly for publishing).
436
437@item server
438The address of the RTMP server.
439
440@item port
441The number of the TCP port to use (by default is 1935).
442
443@item app
444It is the name of the application to access. It usually corresponds to
445the path where the application is installed on the RTMP server
446(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
447the value parsed from the URI through the @code{rtmp_app} option, too.
448
449@item playpath
450It is the path or name of the resource to play with reference to the
451application specified in @var{app}, may be prefixed by "mp4:". You
452can override the value parsed from the URI through the @code{rtmp_playpath}
453option, too.
454
455@item listen
456Act as a server, listening for an incoming connection.
457
458@item timeout
459Maximum time to wait for the incoming connection. Implies listen.
460@end table
461
462Additionally, the following parameters can be set via command line options
463(or in code via @code{AVOption}s):
464@table @option
465
466@item rtmp_app
467Name of application to connect on the RTMP server. This option
468overrides the parameter specified in the URI.
469
470@item rtmp_buffer
471Set the client buffer time in milliseconds. The default is 3000.
472
473@item rtmp_conn
474Extra arbitrary AMF connection parameters, parsed from a string,
475e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
476Each value is prefixed by a single character denoting the type,
477B for Boolean, N for number, S for string, O for object, or Z for null,
478followed by a colon. For Booleans the data must be either 0 or 1 for
479FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
4801 to end or begin an object, respectively. Data items in subobjects may
481be named, by prefixing the type with 'N' and specifying the name before
482the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
483times to construct arbitrary AMF sequences.
484
485@item rtmp_flashver
486Version of the Flash plugin used to run the SWF player. The default
487is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
488<libavformat version>).)
489
490@item rtmp_flush_interval
491Number of packets flushed in the same request (RTMPT only). The default
492is 10.
493
494@item rtmp_live
495Specify that the media is a live stream. No resuming or seeking in
496live streams is possible. The default value is @code{any}, which means the
497subscriber first tries to play the live stream specified in the
498playpath. If a live stream of that name is not found, it plays the
499recorded stream. The other possible values are @code{live} and
500@code{recorded}.
501
502@item rtmp_pageurl
503URL of the web page in which the media was embedded. By default no
504value will be sent.
505
506@item rtmp_playpath
507Stream identifier to play or to publish. This option overrides the
508parameter specified in the URI.
509
510@item rtmp_subscribe
511Name of live stream to subscribe to. By default no value will be sent.
512It is only sent if the option is specified or if rtmp_live
513is set to live.
514
515@item rtmp_swfhash
516SHA256 hash of the decompressed SWF file (32 bytes).
517
518@item rtmp_swfsize
519Size of the decompressed SWF file, required for SWFVerification.
520
521@item rtmp_swfurl
522URL of the SWF player for the media. By default no value will be sent.
523
524@item rtmp_swfverify
525URL to player swf file, compute hash/size automatically.
526
527@item rtmp_tcurl
528URL of the target stream. Defaults to proto://host[:port]/app.
529
530@end table
531
532For example to read with @command{ffplay} a multimedia resource named
533"sample" from the application "vod" from an RTMP server "myserver":
534@example
535ffplay rtmp://myserver/vod/sample
536@end example
537
538To publish to a password protected server, passing the playpath and
539app names separately:
540@example
541ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
542@end example
543
544@section rtmpe
545
546Encrypted Real-Time Messaging Protocol.
547
548The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
549streaming multimedia content within standard cryptographic primitives,
550consisting of Diffie-Hellman key exchange and HMACSHA256, generating
551a pair of RC4 keys.
552
553@section rtmps
554
555Real-Time Messaging Protocol over a secure SSL connection.
556
557The Real-Time Messaging Protocol (RTMPS) is used for streaming
558multimedia content across an encrypted connection.
559
560@section rtmpt
561
562Real-Time Messaging Protocol tunneled through HTTP.
563
564The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
565for streaming multimedia content within HTTP requests to traverse
566firewalls.
567
568@section rtmpte
569
570Encrypted Real-Time Messaging Protocol tunneled through HTTP.
571
572The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
573is used for streaming multimedia content within HTTP requests to traverse
574firewalls.
575
576@section rtmpts
577
578Real-Time Messaging Protocol tunneled through HTTPS.
579
580The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
581for streaming multimedia content within HTTPS requests to traverse
582firewalls.
583
584@section libsmbclient
585
586libsmbclient permits one to manipulate CIFS/SMB network resources.
587
588Following syntax is required.
589
590@example
591smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
592@end example
593
594This protocol accepts the following options.
595
596@table @option
597@item timeout
598Set timeout in miliseconds of socket I/O operations used by the underlying
599low level operation. By default it is set to -1, which means that the timeout
600is not specified.
601
602@item truncate
603Truncate existing files on write, if set to 1. A value of 0 prevents
604truncating. Default value is 1.
605
606@item workgroup
607Set the workgroup used for making connections. By default workgroup is not specified.
608
609@end table
610
611For more information see: @url{http://www.samba.org/}.
612
613@section libssh
614
615Secure File Transfer Protocol via libssh
616
617Allow to read from or write to remote resources using SFTP protocol.
618
619Following syntax is required.
620
621@example
622sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
623@end example
624
625This protocol accepts the following options.
626
627@table @option
628@item timeout
629Set timeout of socket I/O operations used by the underlying low level
630operation. By default it is set to -1, which means that the timeout
631is not specified.
632
633@item truncate
634Truncate existing files on write, if set to 1. A value of 0 prevents
635truncating. Default value is 1.
636
637@item private_key
638Specify the path of the file containing private key to use during authorization.
639By default libssh searches for keys in the @file{~/.ssh/} directory.
640
641@end table
642
643Example: Play a file stored on remote server.
644
645@example
646ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
647@end example
648
649@section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
650
651Real-Time Messaging Protocol and its variants supported through
652librtmp.
653
654Requires the presence of the librtmp headers and library during
655configuration. You need to explicitly configure the build with
656"--enable-librtmp". If enabled this will replace the native RTMP
657protocol.
658
659This protocol provides most client functions and a few server
660functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
661encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
662variants of these encrypted types (RTMPTE, RTMPTS).
663
664The required syntax is:
665@example
666@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
667@end example
668
669where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
670"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
671@var{server}, @var{port}, @var{app} and @var{playpath} have the same
672meaning as specified for the RTMP native protocol.
673@var{options} contains a list of space-separated options of the form
674@var{key}=@var{val}.
675
676See the librtmp manual page (man 3 librtmp) for more information.
677
678For example, to stream a file in real-time to an RTMP server using
679@command{ffmpeg}:
680@example
681ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
682@end example
683
684To play the same stream using @command{ffplay}:
685@example
686ffplay "rtmp://myserver/live/mystream live=1"
687@end example
688
689@section rtp
690
691Real-time Transport Protocol.
692
693The required syntax for an RTP URL is:
694rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
695
696@var{port} specifies the RTP port to use.
697
698The following URL options are supported:
699
700@table @option
701
702@item ttl=@var{n}
703Set the TTL (Time-To-Live) value (for multicast only).
704
705@item rtcpport=@var{n}
706Set the remote RTCP port to @var{n}.
707
708@item localrtpport=@var{n}
709Set the local RTP port to @var{n}.
710
711@item localrtcpport=@var{n}'
712Set the local RTCP port to @var{n}.
713
714@item pkt_size=@var{n}
715Set max packet size (in bytes) to @var{n}.
716
717@item connect=0|1
718Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
719to 0).
720
721@item sources=@var{ip}[,@var{ip}]
722List allowed source IP addresses.
723
724@item block=@var{ip}[,@var{ip}]
725List disallowed (blocked) source IP addresses.
726
727@item write_to_source=0|1
728Send packets to the source address of the latest received packet (if
729set to 1) or to a default remote address (if set to 0).
730
731@item localport=@var{n}
732Set the local RTP port to @var{n}.
733
734This is a deprecated option. Instead, @option{localrtpport} should be
735used.
736
737@end table
738
739Important notes:
740
741@enumerate
742
743@item
744If @option{rtcpport} is not set the RTCP port will be set to the RTP
745port value plus 1.
746
747@item
748If @option{localrtpport} (the local RTP port) is not set any available
749port will be used for the local RTP and RTCP ports.
750
751@item
752If @option{localrtcpport} (the local RTCP port) is not set it will be
f6fa7814 753set to the local RTP port value plus 1.
2ba45a60
DM
754@end enumerate
755
756@section rtsp
757
758Real-Time Streaming Protocol.
759
760RTSP is not technically a protocol handler in libavformat, it is a demuxer
761and muxer. The demuxer supports both normal RTSP (with data transferred
762over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
763data transferred over RDT).
764
765The muxer can be used to send a stream using RTSP ANNOUNCE to a server
766supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
767@uref{https://github.com/revmischa/rtsp-server, RTSP server}).
768
769The required syntax for a RTSP url is:
770@example
771rtsp://@var{hostname}[:@var{port}]/@var{path}
772@end example
773
774Options can be set on the @command{ffmpeg}/@command{ffplay} command
775line, or set in code via @code{AVOption}s or in
776@code{avformat_open_input}.
777
778The following options are supported.
779
780@table @option
781@item initial_pause
782Do not start playing the stream immediately if set to 1. Default value
783is 0.
784
785@item rtsp_transport
786Set RTSP transport protocols.
787
788It accepts the following values:
789@table @samp
790@item udp
791Use UDP as lower transport protocol.
792
793@item tcp
794Use TCP (interleaving within the RTSP control channel) as lower
795transport protocol.
796
797@item udp_multicast
798Use UDP multicast as lower transport protocol.
799
800@item http
801Use HTTP tunneling as lower transport protocol, which is useful for
802passing proxies.
803@end table
804
805Multiple lower transport protocols may be specified, in that case they are
806tried one at a time (if the setup of one fails, the next one is tried).
807For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
808
809@item rtsp_flags
810Set RTSP flags.
811
812The following values are accepted:
813@table @samp
814@item filter_src
815Accept packets only from negotiated peer address and port.
816@item listen
817Act as a server, listening for an incoming connection.
818@item prefer_tcp
819Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
820@end table
821
822Default value is @samp{none}.
823
824@item allowed_media_types
825Set media types to accept from the server.
826
827The following flags are accepted:
828@table @samp
829@item video
830@item audio
831@item data
832@end table
833
834By default it accepts all media types.
835
836@item min_port
837Set minimum local UDP port. Default value is 5000.
838
839@item max_port
840Set maximum local UDP port. Default value is 65000.
841
842@item timeout
843Set maximum timeout (in seconds) to wait for incoming connections.
844
845A value of -1 means infinite (default). This option implies the
846@option{rtsp_flags} set to @samp{listen}.
847
848@item reorder_queue_size
849Set number of packets to buffer for handling of reordered packets.
850
851@item stimeout
852Set socket TCP I/O timeout in microseconds.
853
854@item user-agent
855Override User-Agent header. If not specified, it defaults to the
856libavformat identifier string.
857@end table
858
859When receiving data over UDP, the demuxer tries to reorder received packets
860(since they may arrive out of order, or packets may get lost totally). This
861can be disabled by setting the maximum demuxing delay to zero (via
862the @code{max_delay} field of AVFormatContext).
863
864When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
865streams to display can be chosen with @code{-vst} @var{n} and
866@code{-ast} @var{n} for video and audio respectively, and can be switched
867on the fly by pressing @code{v} and @code{a}.
868
869@subsection Examples
870
871The following examples all make use of the @command{ffplay} and
872@command{ffmpeg} tools.
873
874@itemize
875@item
876Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
877@example
878ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
879@end example
880
881@item
882Watch a stream tunneled over HTTP:
883@example
884ffplay -rtsp_transport http rtsp://server/video.mp4
885@end example
886
887@item
888Send a stream in realtime to a RTSP server, for others to watch:
889@example
890ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
891@end example
892
893@item
894Receive a stream in realtime:
895@example
896ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
897@end example
898@end itemize
899
900@section sap
901
902Session Announcement Protocol (RFC 2974). This is not technically a
903protocol handler in libavformat, it is a muxer and demuxer.
904It is used for signalling of RTP streams, by announcing the SDP for the
905streams regularly on a separate port.
906
907@subsection Muxer
908
909The syntax for a SAP url given to the muxer is:
910@example
911sap://@var{destination}[:@var{port}][?@var{options}]
912@end example
913
914The RTP packets are sent to @var{destination} on port @var{port},
915or to port 5004 if no port is specified.
916@var{options} is a @code{&}-separated list. The following options
917are supported:
918
919@table @option
920
921@item announce_addr=@var{address}
922Specify the destination IP address for sending the announcements to.
923If omitted, the announcements are sent to the commonly used SAP
924announcement multicast address 224.2.127.254 (sap.mcast.net), or
925ff0e::2:7ffe if @var{destination} is an IPv6 address.
926
927@item announce_port=@var{port}
928Specify the port to send the announcements on, defaults to
9299875 if not specified.
930
931@item ttl=@var{ttl}
932Specify the time to live value for the announcements and RTP packets,
933defaults to 255.
934
935@item same_port=@var{0|1}
936If set to 1, send all RTP streams on the same port pair. If zero (the
937default), all streams are sent on unique ports, with each stream on a
938port 2 numbers higher than the previous.
939VLC/Live555 requires this to be set to 1, to be able to receive the stream.
940The RTP stack in libavformat for receiving requires all streams to be sent
941on unique ports.
942@end table
943
944Example command lines follow.
945
946To broadcast a stream on the local subnet, for watching in VLC:
947
948@example
949ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
950@end example
951
952Similarly, for watching in @command{ffplay}:
953
954@example
955ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
956@end example
957
958And for watching in @command{ffplay}, over IPv6:
959
960@example
961ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
962@end example
963
964@subsection Demuxer
965
966The syntax for a SAP url given to the demuxer is:
967@example
968sap://[@var{address}][:@var{port}]
969@end example
970
971@var{address} is the multicast address to listen for announcements on,
972if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
973is the port that is listened on, 9875 if omitted.
974
975The demuxers listens for announcements on the given address and port.
976Once an announcement is received, it tries to receive that particular stream.
977
978Example command lines follow.
979
980To play back the first stream announced on the normal SAP multicast address:
981
982@example
983ffplay sap://
984@end example
985
986To play back the first stream announced on one the default IPv6 SAP multicast address:
987
988@example
989ffplay sap://[ff0e::2:7ffe]
990@end example
991
992@section sctp
993
994Stream Control Transmission Protocol.
995
996The accepted URL syntax is:
997@example
998sctp://@var{host}:@var{port}[?@var{options}]
999@end example
1000
1001The protocol accepts the following options:
1002@table @option
1003@item listen
1004If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1005
1006@item max_streams
1007Set the maximum number of streams. By default no limit is set.
1008@end table
1009
1010@section srtp
1011
1012Secure Real-time Transport Protocol.
1013
1014The accepted options are:
1015@table @option
1016@item srtp_in_suite
1017@item srtp_out_suite
1018Select input and output encoding suites.
1019
1020Supported values:
1021@table @samp
1022@item AES_CM_128_HMAC_SHA1_80
1023@item SRTP_AES128_CM_HMAC_SHA1_80
1024@item AES_CM_128_HMAC_SHA1_32
1025@item SRTP_AES128_CM_HMAC_SHA1_32
1026@end table
1027
1028@item srtp_in_params
1029@item srtp_out_params
1030Set input and output encoding parameters, which are expressed by a
1031base64-encoded representation of a binary block. The first 16 bytes of
1032this binary block are used as master key, the following 14 bytes are
1033used as master salt.
1034@end table
1035
1036@section subfile
1037
1038Virtually extract a segment of a file or another stream.
1039The underlying stream must be seekable.
1040
1041Accepted options:
1042@table @option
1043@item start
1044Start offset of the extracted segment, in bytes.
1045@item end
1046End offset of the extracted segment, in bytes.
1047@end table
1048
1049Examples:
1050
1051Extract a chapter from a DVD VOB file (start and end sectors obtained
1052externally and multiplied by 2048):
1053@example
1054subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1055@end example
1056
1057Play an AVI file directly from a TAR archive:
1058subfile,,start,183241728,end,366490624,,:archive.tar
1059
1060@section tcp
1061
1062Transmission Control Protocol.
1063
1064The required syntax for a TCP url is:
1065@example
1066tcp://@var{hostname}:@var{port}[?@var{options}]
1067@end example
1068
1069@var{options} contains a list of &-separated options of the form
1070@var{key}=@var{val}.
1071
1072The list of supported options follows.
1073
1074@table @option
1075@item listen=@var{1|0}
1076Listen for an incoming connection. Default value is 0.
1077
1078@item timeout=@var{microseconds}
1079Set raise error timeout, expressed in microseconds.
1080
1081This option is only relevant in read mode: if no data arrived in more
1082than this time interval, raise error.
1083
1084@item listen_timeout=@var{microseconds}
1085Set listen timeout, expressed in microseconds.
1086@end table
1087
1088The following example shows how to setup a listening TCP connection
1089with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1090@example
1091ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1092ffplay tcp://@var{hostname}:@var{port}
1093@end example
1094
1095@section tls
1096
1097Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1098
1099The required syntax for a TLS/SSL url is:
1100@example
1101tls://@var{hostname}:@var{port}[?@var{options}]
1102@end example
1103
1104The following parameters can be set via command line options
1105(or in code via @code{AVOption}s):
1106
1107@table @option
1108
1109@item ca_file, cafile=@var{filename}
1110A file containing certificate authority (CA) root certificates to treat
1111as trusted. If the linked TLS library contains a default this might not
1112need to be specified for verification to work, but not all libraries and
1113setups have defaults built in.
1114The file must be in OpenSSL PEM format.
1115
1116@item tls_verify=@var{1|0}
1117If enabled, try to verify the peer that we are communicating with.
1118Note, if using OpenSSL, this currently only makes sure that the
1119peer certificate is signed by one of the root certificates in the CA
1120database, but it does not validate that the certificate actually
1121matches the host name we are trying to connect to. (With GnuTLS,
1122the host name is validated as well.)
1123
1124This is disabled by default since it requires a CA database to be
1125provided by the caller in many cases.
1126
1127@item cert_file, cert=@var{filename}
1128A file containing a certificate to use in the handshake with the peer.
1129(When operating as server, in listen mode, this is more often required
1130by the peer, while client certificates only are mandated in certain
1131setups.)
1132
1133@item key_file, key=@var{filename}
1134A file containing the private key for the certificate.
1135
1136@item listen=@var{1|0}
1137If enabled, listen for connections on the provided port, and assume
1138the server role in the handshake instead of the client role.
1139
1140@end table
1141
1142Example command lines:
1143
1144To create a TLS/SSL server that serves an input stream.
1145
1146@example
1147ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1148@end example
1149
1150To play back a stream from the TLS/SSL server using @command{ffplay}:
1151
1152@example
1153ffplay tls://@var{hostname}:@var{port}
1154@end example
1155
1156@section udp
1157
1158User Datagram Protocol.
1159
1160The required syntax for an UDP URL is:
1161@example
1162udp://@var{hostname}:@var{port}[?@var{options}]
1163@end example
1164
1165@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1166
1167In case threading is enabled on the system, a circular buffer is used
1168to store the incoming data, which allows one to reduce loss of data due to
1169UDP socket buffer overruns. The @var{fifo_size} and
1170@var{overrun_nonfatal} options are related to this buffer.
1171
1172The list of supported options follows.
1173
1174@table @option
1175@item buffer_size=@var{size}
1176Set the UDP maximum socket buffer size in bytes. This is used to set either
1177the receive or send buffer size, depending on what the socket is used for.
1178Default is 64KB. See also @var{fifo_size}.
1179
1180@item localport=@var{port}
1181Override the local UDP port to bind with.
1182
1183@item localaddr=@var{addr}
1184Choose the local IP address. This is useful e.g. if sending multicast
1185and the host has multiple interfaces, where the user can choose
1186which interface to send on by specifying the IP address of that interface.
1187
1188@item pkt_size=@var{size}
1189Set the size in bytes of UDP packets.
1190
1191@item reuse=@var{1|0}
1192Explicitly allow or disallow reusing UDP sockets.
1193
1194@item ttl=@var{ttl}
1195Set the time to live value (for multicast only).
1196
1197@item connect=@var{1|0}
1198Initialize the UDP socket with @code{connect()}. In this case, the
1199destination address can't be changed with ff_udp_set_remote_url later.
1200If the destination address isn't known at the start, this option can
1201be specified in ff_udp_set_remote_url, too.
1202This allows finding out the source address for the packets with getsockname,
1203and makes writes return with AVERROR(ECONNREFUSED) if "destination
1204unreachable" is received.
1205For receiving, this gives the benefit of only receiving packets from
1206the specified peer address/port.
1207
1208@item sources=@var{address}[,@var{address}]
1209Only receive packets sent to the multicast group from one of the
1210specified sender IP addresses.
1211
1212@item block=@var{address}[,@var{address}]
1213Ignore packets sent to the multicast group from the specified
1214sender IP addresses.
1215
1216@item fifo_size=@var{units}
1217Set the UDP receiving circular buffer size, expressed as a number of
1218packets with size of 188 bytes. If not specified defaults to 7*4096.
1219
1220@item overrun_nonfatal=@var{1|0}
1221Survive in case of UDP receiving circular buffer overrun. Default
1222value is 0.
1223
1224@item timeout=@var{microseconds}
1225Set raise error timeout, expressed in microseconds.
1226
1227This option is only relevant in read mode: if no data arrived in more
1228than this time interval, raise error.
1229
1230@item broadcast=@var{1|0}
1231Explicitly allow or disallow UDP broadcasting.
1232
1233Note that broadcasting may not work properly on networks having
1234a broadcast storm protection.
1235@end table
1236
1237@subsection Examples
1238
1239@itemize
1240@item
1241Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1242@example
1243ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1244@end example
1245
1246@item
1247Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1248sized UDP packets, using a large input buffer:
1249@example
1250ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1251@end example
1252
1253@item
1254Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1255@example
1256ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1257@end example
1258@end itemize
1259
1260@section unix
1261
1262Unix local socket
1263
1264The required syntax for a Unix socket URL is:
1265
1266@example
1267unix://@var{filepath}
1268@end example
1269
1270The following parameters can be set via command line options
1271(or in code via @code{AVOption}s):
1272
1273@table @option
1274@item timeout
1275Timeout in ms.
1276@item listen
1277Create the Unix socket in listening mode.
1278@end table
1279
1280@c man end PROTOCOLS