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1	@chapter Protocol Options
2	@c man begin PROTOCOL OPTIONS
3
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.
7
8	Options may be set by specifying -@var{option} @var{value} in the
9	FFmpeg tools, or by setting the value explicitly in the
10	@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
11	for programmatic use.
12
13	The list of supported options follows:
14
15	@table @option
16	@item protocol_whitelist @var{list} (@emph{input})
17	Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
18	prefixed by "-" are disabled.
19	All protocols are allowed by default but protocols used by an another
20	protocol (nested protocols) are restricted to a per protocol subset.
21	@end table
22
23	@c man end PROTOCOL OPTIONS
24
25	@chapter Protocols
26	@c man begin PROTOCOLS
27
28	Protocols are configured elements in FFmpeg that enable access to
29	resources that require specific protocols.
30
31	When you configure your FFmpeg build, all the supported protocols are
32	enabled by default. You can list all available ones using the
33	configure option "--list-protocols".
34
35	You can disable all the protocols using the configure option
36	"--disable-protocols", and selectively enable a protocol using the
37	option "--enable-protocol=@var{PROTOCOL}", or you can disable a
38	particular protocol using the option
39	"--disable-protocol=@var{PROTOCOL}".
40
41	The option "-protocols" of the ff* tools will display the list of
42	supported protocols.
43
44	All protocols accept the following options:
45
46	@table @option
47	@item rw_timeout
48	Maximum time to wait for (network) read/write operations to complete,
49	in microseconds.
50	@end table
51
52	A description of the currently available protocols follows.
53
54	@section async
55
56	Asynchronous data filling wrapper for input stream.
57
58	Fill data in a background thread, to decouple I/O operation from demux thread.
59
60	@example
61	async:@var{URL}
62	async:http://host/resource
63	async:cache:http://host/resource
64	@end example
65
66	@section bluray
67
68	Read BluRay playlist.
69
70	The accepted options are:
71	@table @option
72
73	@item angle
74	BluRay angle
75
76	@item chapter
77	Start chapter (1...N)
78
79	@item playlist
80	Playlist to read (BDMV/PLAYLIST/?????.mpls)
81
82	@end table
83
84	Examples:
85
86	Read longest playlist from BluRay mounted to /mnt/bluray:
87	@example
88	bluray:/mnt/bluray
89	@end example
90
91	Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
92	@example
93	-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
94	@end example
95
96	@section cache
97
98	Caching wrapper for input stream.
99
100	Cache the input stream to temporary file. It brings seeking capability to live streams.
101
102	@example
103	cache:@var{URL}
104	@end example
105
106	@section concat
107
108	Physical concatenation protocol.
109
110	Read and seek from many resources in sequence as if they were
111	a unique resource.
112
113	A URL accepted by this protocol has the syntax:
114	@example
115	concat:@var{URL1}|@var{URL2}|...|@var{URLN}
116	@end example
117
118	where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
119	resource to be concatenated, each one possibly specifying a distinct
120	protocol.
121
122	For example to read a sequence of files @file{split1.mpeg},
123	@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
124	command:
125	@example
126	ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
127	@end example
128
129	Note that you may need to escape the character "|" which is special for
130	many shells.
131
132	@section crypto
133
134	AES-encrypted stream reading protocol.
135
136	The accepted options are:
137	@table @option
138	@item key
139	Set the AES decryption key binary block from given hexadecimal representation.
140
141	@item iv
142	Set the AES decryption initialization vector binary block from given hexadecimal representation.
143	@end table
144
145	Accepted URL formats:
146	@example
147	crypto:@var{URL}
148	crypto+@var{URL}
149	@end example
150
151	@section data
152
153	Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
154
155	For example, to convert a GIF file given inline with @command{ffmpeg}:
156	@example
157	ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
158	@end example
159
160	@section file
161
162	File access protocol.
163
164	Read from or write to a file.
165
166	A file URL can have the form:
167	@example
168	file:@var{filename}
169	@end example
170
171	where @var{filename} is the path of the file to read.
172
173	An URL that does not have a protocol prefix will be assumed to be a
174	file URL. Depending on the build, an URL that looks like a Windows
175	path with the drive letter at the beginning will also be assumed to be
176	a file URL (usually not the case in builds for unix-like systems).
177
178	For example to read from a file @file{input.mpeg} with @command{ffmpeg}
179	use the command:
180	@example
181	ffmpeg -i file:input.mpeg output.mpeg
182	@end example
183
184	This protocol accepts the following options:
185
186	@table @option
187	@item truncate
188	Truncate existing files on write, if set to 1. A value of 0 prevents
189	truncating. Default value is 1.
190
191	@item blocksize
192	Set I/O operation maximum block size, in bytes. Default value is
193	@code{INT_MAX}, which results in not limiting the requested block size.
194	Setting this value reasonably low improves user termination request reaction
195	time, which is valuable for files on slow medium.
196	@end table
197
198	@section ftp
199
200	FTP (File Transfer Protocol).
201
202	Read from or write to remote resources using FTP protocol.
203
204	Following syntax is required.
205	@example
206	ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
207	@end example
208
209	This protocol accepts the following options.
210
211	@table @option
212	@item timeout
213	Set timeout in microseconds of socket I/O operations used by the underlying low level
214	operation. By default it is set to -1, which means that the timeout is
215	not specified.
216
217	@item ftp-anonymous-password
218	Password used when login as anonymous user. Typically an e-mail address
219	should be used.
220
221	@item ftp-write-seekable
222	Control seekability of connection during encoding. If set to 1 the
223	resource is supposed to be seekable, if set to 0 it is assumed not
224	to be seekable. Default value is 0.
225	@end table
226
227	NOTE: Protocol can be used as output, but it is recommended to not do
228	it, unless special care is taken (tests, customized server configuration
229	etc.). Different FTP servers behave in different way during seek
230	operation. ff* tools may produce incomplete content due to server limitations.
231
232	This protocol accepts the following options:
233
234	@table @option
235	@item follow
236	If set to 1, the protocol will retry reading at the end of the file, allowing
237	reading files that still are being written. In order for this to terminate,
238	you either need to use the rw_timeout option, or use the interrupt callback
239	(for API users).
240
241	@end table
242
243	@section gopher
244
245	Gopher protocol.
246
247	@section hls
248
249	Read Apple HTTP Live Streaming compliant segmented stream as
250	a uniform one. The M3U8 playlists describing the segments can be
251	remote HTTP resources or local files, accessed using the standard
252	file protocol.
253	The nested protocol is declared by specifying
254	"+@var{proto}" after the hls URI scheme name, where @var{proto}
255	is either "file" or "http".
256
257	@example
258	hls+http://host/path/to/remote/resource.m3u8
259	hls+file://path/to/local/resource.m3u8
260	@end example
261
262	Using this protocol is discouraged - the hls demuxer should work
263	just as well (if not, please report the issues) and is more complete.
264	To use the hls demuxer instead, simply use the direct URLs to the
265	m3u8 files.
266
267	@section http
268
269	HTTP (Hyper Text Transfer Protocol).
270
271	This protocol accepts the following options:
272
273	@table @option
274	@item seekable
275	Control seekability of connection. If set to 1 the resource is
276	supposed to be seekable, if set to 0 it is assumed not to be seekable,
277	if set to -1 it will try to autodetect if it is seekable. Default
278	value is -1.
279
280	@item chunked_post
281	If set to 1 use chunked Transfer-Encoding for posts, default is 1.
282
283	@item content_type
284	Set a specific content type for the POST messages or for listen mode.
285
286	@item http_proxy
287	set HTTP proxy to tunnel through e.g. http://example.com:1234
288
289	@item headers
290	Set custom HTTP headers, can override built in default headers. The
291	value must be a string encoding the headers.
292
293	@item multiple_requests
294	Use persistent connections if set to 1, default is 0.
295
296	@item post_data
297	Set custom HTTP post data.
298
299	@item user_agent
300	Override the User-Agent header. If not specified the protocol will use a
301	string describing the libavformat build. ("Lavf/<version>")
302
303	@item user-agent
304	This is a deprecated option, you can use user_agent instead it.
305
306	@item timeout
307	Set timeout in microseconds of socket I/O operations used by the underlying low level
308	operation. By default it is set to -1, which means that the timeout is
309	not specified.
310
311	@item reconnect_at_eof
312	If set then eof is treated like an error and causes reconnection, this is useful
313	for live / endless streams.
314
315	@item reconnect_streamed
316	If set then even streamed/non seekable streams will be reconnected on errors.
317
318	@item reconnect_delay_max
319	Sets the maximum delay in seconds after which to give up reconnecting
320
321	@item mime_type
322	Export the MIME type.
323
324	@item icy
325	If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
326	supports this, the metadata has to be retrieved by the application by reading
327	the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
328	The default is 1.
329
330	@item icy_metadata_headers
331	If the server supports ICY metadata, this contains the ICY-specific HTTP reply
332	headers, separated by newline characters.
333
334	@item icy_metadata_packet
335	If the server supports ICY metadata, and @option{icy} was set to 1, this
336	contains the last non-empty metadata packet sent by the server. It should be
337	polled in regular intervals by applications interested in mid-stream metadata
338	updates.
339
340	@item cookies
341	Set the cookies to be sent in future requests. The format of each cookie is the
342	same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
343	delimited by a newline character.
344
345	@item offset
346	Set initial byte offset.
347
348	@item end_offset
349	Try to limit the request to bytes preceding this offset.
350
351	@item method
352	When used as a client option it sets the HTTP method for the request.
353
354	When used as a server option it sets the HTTP method that is going to be
355	expected from the client(s).
356	If the expected and the received HTTP method do not match the client will
357	be given a Bad Request response.
358	When unset the HTTP method is not checked for now. This will be replaced by
359	autodetection in the future.
360
361	@item listen
362	If set to 1 enables experimental HTTP server. This can be used to send data when
363	used as an output option, or read data from a client with HTTP POST when used as
364	an input option.
365	If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
366	in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
367	@example
368	# Server side (sending):
369	ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
370
371	# Client side (receiving):
372	ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
373
374	# Client can also be done with wget:
375	wget http://@var{server}:@var{port} -O somefile.ogg
376
377	# Server side (receiving):
378	ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
379
380	# Client side (sending):
381	ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
382
383	# Client can also be done with wget:
384	wget --post-file=somefile.ogg http://@var{server}:@var{port}
385	@end example
386
387	@end table
388
389	@subsection HTTP Cookies
390
391	Some HTTP requests will be denied unless cookie values are passed in with the
392	request. The @option{cookies} option allows these cookies to be specified. At
393	the very least, each cookie must specify a value along with a path and domain.
394	HTTP requests that match both the domain and path will automatically include the
395	cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
396	by a newline.
397
398	The required syntax to play a stream specifying a cookie is:
399	@example
400	ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
401	@end example
402
403	@section Icecast
404
405	Icecast protocol (stream to Icecast servers)
406
407	This protocol accepts the following options:
408
409	@table @option
410	@item ice_genre
411	Set the stream genre.
412
413	@item ice_name
414	Set the stream name.
415
416	@item ice_description
417	Set the stream description.
418
419	@item ice_url
420	Set the stream website URL.
421
422	@item ice_public
423	Set if the stream should be public.
424	The default is 0 (not public).
425
426	@item user_agent
427	Override the User-Agent header. If not specified a string of the form
428	"Lavf/<version>" will be used.
429
430	@item password
431	Set the Icecast mountpoint password.
432
433	@item content_type
434	Set the stream content type. This must be set if it is different from
435	audio/mpeg.
436
437	@item legacy_icecast
438	This enables support for Icecast versions < 2.4.0, that do not support the
439	HTTP PUT method but the SOURCE method.
440
441	@end table
442
443	@example
444	icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
445	@end example
446
447	@section mmst
448
449	MMS (Microsoft Media Server) protocol over TCP.
450
451	@section mmsh
452
453	MMS (Microsoft Media Server) protocol over HTTP.
454
455	The required syntax is:
456	@example
457	mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
458	@end example
459
460	@section md5
461
462	MD5 output protocol.
463
464	Computes the MD5 hash of the data to be written, and on close writes
465	this to the designated output or stdout if none is specified. It can
466	be used to test muxers without writing an actual file.
467
468	Some examples follow.
469	@example
470	# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
471	ffmpeg -i input.flv -f avi -y md5:output.avi.md5
472
473	# Write the MD5 hash of the encoded AVI file to stdout.
474	ffmpeg -i input.flv -f avi -y md5:
475	@end example
476
477	Note that some formats (typically MOV) require the output protocol to
478	be seekable, so they will fail with the MD5 output protocol.
479
480	@section pipe
481
482	UNIX pipe access protocol.
483
484	Read and write from UNIX pipes.
485
486	The accepted syntax is:
487	@example
488	pipe:[@var{number}]
489	@end example
490
491	@var{number} is the number corresponding to the file descriptor of the
492	pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
493	is not specified, by default the stdout file descriptor will be used
494	for writing, stdin for reading.
495
496	For example to read from stdin with @command{ffmpeg}:
497	@example
498	cat test.wav | ffmpeg -i pipe:0
499	# ...this is the same as...
500	cat test.wav | ffmpeg -i pipe:
501	@end example
502
503	For writing to stdout with @command{ffmpeg}:
504	@example
505	ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
506	# ...this is the same as...
507	ffmpeg -i test.wav -f avi pipe: | cat > test.avi
508	@end example
509
510	This protocol accepts the following options:
511
512	@table @option
513	@item blocksize
514	Set I/O operation maximum block size, in bytes. Default value is
515	@code{INT_MAX}, which results in not limiting the requested block size.
516	Setting this value reasonably low improves user termination request reaction
517	time, which is valuable if data transmission is slow.
518	@end table
519
520	Note that some formats (typically MOV), require the output protocol to
521	be seekable, so they will fail with the pipe output protocol.
522
523	@section prompeg
524
525	Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
526
527	The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
528	for MPEG-2 Transport Streams sent over RTP.
529
530	This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
531	the @code{rtp} protocol.
532
533	The required syntax is:
534	@example
535	-f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
536	@end example
537
538	The destination UDP ports are @code{port + 2} for the column FEC stream
539	and @code{port + 4} for the row FEC stream.
540
541	This protocol accepts the following options:
542	@table @option
543
544	@item l=@var{n}
545	The number of columns (4-20, LxD <= 100)
546
547	@item d=@var{n}
548	The number of rows (4-20, LxD <= 100)
549
550	@end table
551
552	Example usage:
553
554	@example
555	-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
556	@end example
557
558	@section rtmp
559
560	Real-Time Messaging Protocol.
561
562	The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
563	content across a TCP/IP network.
564
565	The required syntax is:
566	@example
567	rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
568	@end example
569
570	The accepted parameters are:
571	@table @option
572
573	@item username
574	An optional username (mostly for publishing).
575
576	@item password
577	An optional password (mostly for publishing).
578
579	@item server
580	The address of the RTMP server.
581
582	@item port
583	The number of the TCP port to use (by default is 1935).
584
585	@item app
586	It is the name of the application to access. It usually corresponds to
587	the path where the application is installed on the RTMP server
588	(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
589	the value parsed from the URI through the @code{rtmp_app} option, too.
590
591	@item playpath
592	It is the path or name of the resource to play with reference to the
593	application specified in @var{app}, may be prefixed by "mp4:". You
594	can override the value parsed from the URI through the @code{rtmp_playpath}
595	option, too.
596
597	@item listen
598	Act as a server, listening for an incoming connection.
599
600	@item timeout
601	Maximum time to wait for the incoming connection. Implies listen.
602	@end table
603
604	Additionally, the following parameters can be set via command line options
605	(or in code via @code{AVOption}s):
606	@table @option
607
608	@item rtmp_app
609	Name of application to connect on the RTMP server. This option
610	overrides the parameter specified in the URI.
611
612	@item rtmp_buffer
613	Set the client buffer time in milliseconds. The default is 3000.
614
615	@item rtmp_conn
616	Extra arbitrary AMF connection parameters, parsed from a string,
617	e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
618	Each value is prefixed by a single character denoting the type,
619	B for Boolean, N for number, S for string, O for object, or Z for null,
620	followed by a colon. For Booleans the data must be either 0 or 1 for
621	FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
622	1 to end or begin an object, respectively. Data items in subobjects may
623	be named, by prefixing the type with 'N' and specifying the name before
624	the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
625	times to construct arbitrary AMF sequences.
626
627	@item rtmp_flashver
628	Version of the Flash plugin used to run the SWF player. The default
629	is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
630	<libavformat version>).)
631
632	@item rtmp_flush_interval
633	Number of packets flushed in the same request (RTMPT only). The default
634	is 10.
635
636	@item rtmp_live
637	Specify that the media is a live stream. No resuming or seeking in
638	live streams is possible. The default value is @code{any}, which means the
639	subscriber first tries to play the live stream specified in the
640	playpath. If a live stream of that name is not found, it plays the
641	recorded stream. The other possible values are @code{live} and
642	@code{recorded}.
643
644	@item rtmp_pageurl
645	URL of the web page in which the media was embedded. By default no
646	value will be sent.
647
648	@item rtmp_playpath
649	Stream identifier to play or to publish. This option overrides the
650	parameter specified in the URI.
651
652	@item rtmp_subscribe
653	Name of live stream to subscribe to. By default no value will be sent.
654	It is only sent if the option is specified or if rtmp_live
655	is set to live.
656
657	@item rtmp_swfhash
658	SHA256 hash of the decompressed SWF file (32 bytes).
659
660	@item rtmp_swfsize
661	Size of the decompressed SWF file, required for SWFVerification.
662
663	@item rtmp_swfurl
664	URL of the SWF player for the media. By default no value will be sent.
665
666	@item rtmp_swfverify
667	URL to player swf file, compute hash/size automatically.
668
669	@item rtmp_tcurl
670	URL of the target stream. Defaults to proto://host[:port]/app.
671
672	@end table
673
674	For example to read with @command{ffplay} a multimedia resource named
675	"sample" from the application "vod" from an RTMP server "myserver":
676	@example
677	ffplay rtmp://myserver/vod/sample
678	@end example
679
680	To publish to a password protected server, passing the playpath and
681	app names separately:
682	@example
683	ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
684	@end example
685
686	@section rtmpe
687
688	Encrypted Real-Time Messaging Protocol.
689
690	The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
691	streaming multimedia content within standard cryptographic primitives,
692	consisting of Diffie-Hellman key exchange and HMACSHA256, generating
693	a pair of RC4 keys.
694
695	@section rtmps
696
697	Real-Time Messaging Protocol over a secure SSL connection.
698
699	The Real-Time Messaging Protocol (RTMPS) is used for streaming
700	multimedia content across an encrypted connection.
701
702	@section rtmpt
703
704	Real-Time Messaging Protocol tunneled through HTTP.
705
706	The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
707	for streaming multimedia content within HTTP requests to traverse
708	firewalls.
709
710	@section rtmpte
711
712	Encrypted Real-Time Messaging Protocol tunneled through HTTP.
713
714	The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
715	is used for streaming multimedia content within HTTP requests to traverse
716	firewalls.
717
718	@section rtmpts
719
720	Real-Time Messaging Protocol tunneled through HTTPS.
721
722	The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
723	for streaming multimedia content within HTTPS requests to traverse
724	firewalls.
725
726	@section libsmbclient
727
728	libsmbclient permits one to manipulate CIFS/SMB network resources.
729
730	Following syntax is required.
731
732	@example
733	smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
734	@end example
735
736	This protocol accepts the following options.
737
738	@table @option
739	@item timeout
740	Set timeout in milliseconds of socket I/O operations used by the underlying
741	low level operation. By default it is set to -1, which means that the timeout
742	is not specified.
743
744	@item truncate
745	Truncate existing files on write, if set to 1. A value of 0 prevents
746	truncating. Default value is 1.
747
748	@item workgroup
749	Set the workgroup used for making connections. By default workgroup is not specified.
750
751	@end table
752
753	For more information see: @url{http://www.samba.org/}.
754
755	@section libssh
756
757	Secure File Transfer Protocol via libssh
758
759	Read from or write to remote resources using SFTP protocol.
760
761	Following syntax is required.
762
763	@example
764	sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
765	@end example
766
767	This protocol accepts the following options.
768
769	@table @option
770	@item timeout
771	Set timeout of socket I/O operations used by the underlying low level
772	operation. By default it is set to -1, which means that the timeout
773	is not specified.
774
775	@item truncate
776	Truncate existing files on write, if set to 1. A value of 0 prevents
777	truncating. Default value is 1.
778
779	@item private_key
780	Specify the path of the file containing private key to use during authorization.
781	By default libssh searches for keys in the @file{~/.ssh/} directory.
782
783	@end table
784
785	Example: Play a file stored on remote server.
786
787	@example
788	ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
789	@end example
790
791	@section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
792
793	Real-Time Messaging Protocol and its variants supported through
794	librtmp.
795
796	Requires the presence of the librtmp headers and library during
797	configuration. You need to explicitly configure the build with
798	"--enable-librtmp". If enabled this will replace the native RTMP
799	protocol.
800
801	This protocol provides most client functions and a few server
802	functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
803	encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
804	variants of these encrypted types (RTMPTE, RTMPTS).
805
806	The required syntax is:
807	@example
808	@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
809	@end example
810
811	where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
812	"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
813	@var{server}, @var{port}, @var{app} and @var{playpath} have the same
814	meaning as specified for the RTMP native protocol.
815	@var{options} contains a list of space-separated options of the form
816	@var{key}=@var{val}.
817
818	See the librtmp manual page (man 3 librtmp) for more information.
819
820	For example, to stream a file in real-time to an RTMP server using
821	@command{ffmpeg}:
822	@example
823	ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
824	@end example
825
826	To play the same stream using @command{ffplay}:
827	@example
828	ffplay "rtmp://myserver/live/mystream live=1"
829	@end example
830
831	@section rtp
832
833	Real-time Transport Protocol.
834
835	The required syntax for an RTP URL is:
836	rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
837
838	@var{port} specifies the RTP port to use.
839
840	The following URL options are supported:
841
842	@table @option
843
844	@item ttl=@var{n}
845	Set the TTL (Time-To-Live) value (for multicast only).
846
847	@item rtcpport=@var{n}
848	Set the remote RTCP port to @var{n}.
849
850	@item localrtpport=@var{n}
851	Set the local RTP port to @var{n}.
852
853	@item localrtcpport=@var{n}'
854	Set the local RTCP port to @var{n}.
855
856	@item pkt_size=@var{n}
857	Set max packet size (in bytes) to @var{n}.
858
859	@item connect=0|1
860	Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
861	to 0).
862
863	@item sources=@var{ip}[,@var{ip}]
864	List allowed source IP addresses.
865
866	@item block=@var{ip}[,@var{ip}]
867	List disallowed (blocked) source IP addresses.
868
869	@item write_to_source=0|1
870	Send packets to the source address of the latest received packet (if
871	set to 1) or to a default remote address (if set to 0).
872
873	@item localport=@var{n}
874	Set the local RTP port to @var{n}.
875
876	This is a deprecated option. Instead, @option{localrtpport} should be
877	used.
878
879	@end table
880
881	Important notes:
882
883	@enumerate
884
885	@item
886	If @option{rtcpport} is not set the RTCP port will be set to the RTP
887	port value plus 1.
888
889	@item
890	If @option{localrtpport} (the local RTP port) is not set any available
891	port will be used for the local RTP and RTCP ports.
892
893	@item
894	If @option{localrtcpport} (the local RTCP port) is not set it will be
895	set to the local RTP port value plus 1.
896	@end enumerate
897
898	@section rtsp
899
900	Real-Time Streaming Protocol.
901
902	RTSP is not technically a protocol handler in libavformat, it is a demuxer
903	and muxer. The demuxer supports both normal RTSP (with data transferred
904	over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
905	data transferred over RDT).
906
907	The muxer can be used to send a stream using RTSP ANNOUNCE to a server
908	supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
909	@uref{https://github.com/revmischa/rtsp-server, RTSP server}).
910
911	The required syntax for a RTSP url is:
912	@example
913	rtsp://@var{hostname}[:@var{port}]/@var{path}
914	@end example
915
916	Options can be set on the @command{ffmpeg}/@command{ffplay} command
917	line, or set in code via @code{AVOption}s or in
918	@code{avformat_open_input}.
919
920	The following options are supported.
921
922	@table @option
923	@item initial_pause
924	Do not start playing the stream immediately if set to 1. Default value
925	is 0.
926
927	@item rtsp_transport
928	Set RTSP transport protocols.
929
930	It accepts the following values:
931	@table @samp
932	@item udp
933	Use UDP as lower transport protocol.
934
935	@item tcp
936	Use TCP (interleaving within the RTSP control channel) as lower
937	transport protocol.
938
939	@item udp_multicast
940	Use UDP multicast as lower transport protocol.
941
942	@item http
943	Use HTTP tunneling as lower transport protocol, which is useful for
944	passing proxies.
945	@end table
946
947	Multiple lower transport protocols may be specified, in that case they are
948	tried one at a time (if the setup of one fails, the next one is tried).
949	For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
950
951	@item rtsp_flags
952	Set RTSP flags.
953
954	The following values are accepted:
955	@table @samp
956	@item filter_src
957	Accept packets only from negotiated peer address and port.
958	@item listen
959	Act as a server, listening for an incoming connection.
960	@item prefer_tcp
961	Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
962	@end table
963
964	Default value is @samp{none}.
965
966	@item allowed_media_types
967	Set media types to accept from the server.
968
969	The following flags are accepted:
970	@table @samp
971	@item video
972	@item audio
973	@item data
974	@end table
975
976	By default it accepts all media types.
977
978	@item min_port
979	Set minimum local UDP port. Default value is 5000.
980
981	@item max_port
982	Set maximum local UDP port. Default value is 65000.
983
984	@item timeout
985	Set maximum timeout (in seconds) to wait for incoming connections.
986
987	A value of -1 means infinite (default). This option implies the
988	@option{rtsp_flags} set to @samp{listen}.
989
990	@item reorder_queue_size
991	Set number of packets to buffer for handling of reordered packets.
992
993	@item stimeout
994	Set socket TCP I/O timeout in microseconds.
995
996	@item user-agent
997	Override User-Agent header. If not specified, it defaults to the
998	libavformat identifier string.
999	@end table
1000
1001	When receiving data over UDP, the demuxer tries to reorder received packets
1002	(since they may arrive out of order, or packets may get lost totally). This
1003	can be disabled by setting the maximum demuxing delay to zero (via
1004	the @code{max_delay} field of AVFormatContext).
1005
1006	When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1007	streams to display can be chosen with @code{-vst} @var{n} and
1008	@code{-ast} @var{n} for video and audio respectively, and can be switched
1009	on the fly by pressing @code{v} and @code{a}.
1010
1011	@subsection Examples
1012
1013	The following examples all make use of the @command{ffplay} and
1014	@command{ffmpeg} tools.
1015
1016	@itemize
1017	@item
1018	Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1019	@example
1020	ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1021	@end example
1022
1023	@item
1024	Watch a stream tunneled over HTTP:
1025	@example
1026	ffplay -rtsp_transport http rtsp://server/video.mp4
1027	@end example
1028
1029	@item
1030	Send a stream in realtime to a RTSP server, for others to watch:
1031	@example
1032	ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1033	@end example
1034
1035	@item
1036	Receive a stream in realtime:
1037	@example
1038	ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1039	@end example
1040	@end itemize
1041
1042	@section sap
1043
1044	Session Announcement Protocol (RFC 2974). This is not technically a
1045	protocol handler in libavformat, it is a muxer and demuxer.
1046	It is used for signalling of RTP streams, by announcing the SDP for the
1047	streams regularly on a separate port.
1048
1049	@subsection Muxer
1050
1051	The syntax for a SAP url given to the muxer is:
1052	@example
1053	sap://@var{destination}[:@var{port}][?@var{options}]
1054	@end example
1055
1056	The RTP packets are sent to @var{destination} on port @var{port},
1057	or to port 5004 if no port is specified.
1058	@var{options} is a @code{&}-separated list. The following options
1059	are supported:
1060
1061	@table @option
1062
1063	@item announce_addr=@var{address}
1064	Specify the destination IP address for sending the announcements to.
1065	If omitted, the announcements are sent to the commonly used SAP
1066	announcement multicast address 224.2.127.254 (sap.mcast.net), or
1067	ff0e::2:7ffe if @var{destination} is an IPv6 address.
1068
1069	@item announce_port=@var{port}
1070	Specify the port to send the announcements on, defaults to
1071	9875 if not specified.
1072
1073	@item ttl=@var{ttl}
1074	Specify the time to live value for the announcements and RTP packets,
1075	defaults to 255.
1076
1077	@item same_port=@var{0|1}
1078	If set to 1, send all RTP streams on the same port pair. If zero (the
1079	default), all streams are sent on unique ports, with each stream on a
1080	port 2 numbers higher than the previous.
1081	VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1082	The RTP stack in libavformat for receiving requires all streams to be sent
1083	on unique ports.
1084	@end table
1085
1086	Example command lines follow.
1087
1088	To broadcast a stream on the local subnet, for watching in VLC:
1089
1090	@example
1091	ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1092	@end example
1093
1094	Similarly, for watching in @command{ffplay}:
1095
1096	@example
1097	ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1098	@end example
1099
1100	And for watching in @command{ffplay}, over IPv6:
1101
1102	@example
1103	ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1104	@end example
1105
1106	@subsection Demuxer
1107
1108	The syntax for a SAP url given to the demuxer is:
1109	@example
1110	sap://[@var{address}][:@var{port}]
1111	@end example
1112
1113	@var{address} is the multicast address to listen for announcements on,
1114	if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1115	is the port that is listened on, 9875 if omitted.
1116
1117	The demuxers listens for announcements on the given address and port.
1118	Once an announcement is received, it tries to receive that particular stream.
1119
1120	Example command lines follow.
1121
1122	To play back the first stream announced on the normal SAP multicast address:
1123
1124	@example
1125	ffplay sap://
1126	@end example
1127
1128	To play back the first stream announced on one the default IPv6 SAP multicast address:
1129
1130	@example
1131	ffplay sap://[ff0e::2:7ffe]
1132	@end example
1133
1134	@section sctp
1135
1136	Stream Control Transmission Protocol.
1137
1138	The accepted URL syntax is:
1139	@example
1140	sctp://@var{host}:@var{port}[?@var{options}]
1141	@end example
1142
1143	The protocol accepts the following options:
1144	@table @option
1145	@item listen
1146	If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1147
1148	@item max_streams
1149	Set the maximum number of streams. By default no limit is set.
1150	@end table
1151
1152	@section srtp
1153
1154	Secure Real-time Transport Protocol.
1155
1156	The accepted options are:
1157	@table @option
1158	@item srtp_in_suite
1159	@item srtp_out_suite
1160	Select input and output encoding suites.
1161
1162	Supported values:
1163	@table @samp
1164	@item AES_CM_128_HMAC_SHA1_80
1165	@item SRTP_AES128_CM_HMAC_SHA1_80
1166	@item AES_CM_128_HMAC_SHA1_32
1167	@item SRTP_AES128_CM_HMAC_SHA1_32
1168	@end table
1169
1170	@item srtp_in_params
1171	@item srtp_out_params
1172	Set input and output encoding parameters, which are expressed by a
1173	base64-encoded representation of a binary block. The first 16 bytes of
1174	this binary block are used as master key, the following 14 bytes are
1175	used as master salt.
1176	@end table
1177
1178	@section subfile
1179
1180	Virtually extract a segment of a file or another stream.
1181	The underlying stream must be seekable.
1182
1183	Accepted options:
1184	@table @option
1185	@item start
1186	Start offset of the extracted segment, in bytes.
1187	@item end
1188	End offset of the extracted segment, in bytes.
1189	@end table
1190
1191	Examples:
1192
1193	Extract a chapter from a DVD VOB file (start and end sectors obtained
1194	externally and multiplied by 2048):
1195	@example
1196	subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1197	@end example
1198
1199	Play an AVI file directly from a TAR archive:
1200	@example
1201	subfile,,start,183241728,end,366490624,,:archive.tar
1202	@end example
1203
1204	@section tee
1205
1206	Writes the output to multiple protocols. The individual outputs are separated
1207	by |
1208
1209	@example
1210	tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1211	@end example
1212
1213	@section tcp
1214
1215	Transmission Control Protocol.
1216
1217	The required syntax for a TCP url is:
1218	@example
1219	tcp://@var{hostname}:@var{port}[?@var{options}]
1220	@end example
1221
1222	@var{options} contains a list of &-separated options of the form
1223	@var{key}=@var{val}.
1224
1225	The list of supported options follows.
1226
1227	@table @option
1228	@item listen=@var{1|0}
1229	Listen for an incoming connection. Default value is 0.
1230
1231	@item timeout=@var{microseconds}
1232	Set raise error timeout, expressed in microseconds.
1233
1234	This option is only relevant in read mode: if no data arrived in more
1235	than this time interval, raise error.
1236
1237	@item listen_timeout=@var{milliseconds}
1238	Set listen timeout, expressed in milliseconds.
1239
1240	@item recv_buffer_size=@var{bytes}
1241	Set receive buffer size, expressed bytes.
1242
1243	@item send_buffer_size=@var{bytes}
1244	Set send buffer size, expressed bytes.
1245	@end table
1246
1247	The following example shows how to setup a listening TCP connection
1248	with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1249	@example
1250	ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1251	ffplay tcp://@var{hostname}:@var{port}
1252	@end example
1253
1254	@section tls
1255
1256	Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1257
1258	The required syntax for a TLS/SSL url is:
1259	@example
1260	tls://@var{hostname}:@var{port}[?@var{options}]
1261	@end example
1262
1263	The following parameters can be set via command line options
1264	(or in code via @code{AVOption}s):
1265
1266	@table @option
1267
1268	@item ca_file, cafile=@var{filename}
1269	A file containing certificate authority (CA) root certificates to treat
1270	as trusted. If the linked TLS library contains a default this might not
1271	need to be specified for verification to work, but not all libraries and
1272	setups have defaults built in.
1273	The file must be in OpenSSL PEM format.
1274
1275	@item tls_verify=@var{1|0}
1276	If enabled, try to verify the peer that we are communicating with.
1277	Note, if using OpenSSL, this currently only makes sure that the
1278	peer certificate is signed by one of the root certificates in the CA
1279	database, but it does not validate that the certificate actually
1280	matches the host name we are trying to connect to. (With GnuTLS,
1281	the host name is validated as well.)
1282
1283	This is disabled by default since it requires a CA database to be
1284	provided by the caller in many cases.
1285
1286	@item cert_file, cert=@var{filename}
1287	A file containing a certificate to use in the handshake with the peer.
1288	(When operating as server, in listen mode, this is more often required
1289	by the peer, while client certificates only are mandated in certain
1290	setups.)
1291
1292	@item key_file, key=@var{filename}
1293	A file containing the private key for the certificate.
1294
1295	@item listen=@var{1|0}
1296	If enabled, listen for connections on the provided port, and assume
1297	the server role in the handshake instead of the client role.
1298
1299	@end table
1300
1301	Example command lines:
1302
1303	To create a TLS/SSL server that serves an input stream.
1304
1305	@example
1306	ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1307	@end example
1308
1309	To play back a stream from the TLS/SSL server using @command{ffplay}:
1310
1311	@example
1312	ffplay tls://@var{hostname}:@var{port}
1313	@end example
1314
1315	@section udp
1316
1317	User Datagram Protocol.
1318
1319	The required syntax for an UDP URL is:
1320	@example
1321	udp://@var{hostname}:@var{port}[?@var{options}]
1322	@end example
1323
1324	@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1325
1326	In case threading is enabled on the system, a circular buffer is used
1327	to store the incoming data, which allows one to reduce loss of data due to
1328	UDP socket buffer overruns. The @var{fifo_size} and
1329	@var{overrun_nonfatal} options are related to this buffer.
1330
1331	The list of supported options follows.
1332
1333	@table @option
1334	@item buffer_size=@var{size}
1335	Set the UDP maximum socket buffer size in bytes. This is used to set either
1336	the receive or send buffer size, depending on what the socket is used for.
1337	Default is 64KB. See also @var{fifo_size}.
1338
1339	@item bitrate=@var{bitrate}
1340	If set to nonzero, the output will have the specified constant bitrate if the
1341	input has enough packets to sustain it.
1342
1343	@item burst_bits=@var{bits}
1344	When using @var{bitrate} this specifies the maximum number of bits in
1345	packet bursts.
1346
1347	@item localport=@var{port}
1348	Override the local UDP port to bind with.
1349
1350	@item localaddr=@var{addr}
1351	Choose the local IP address. This is useful e.g. if sending multicast
1352	and the host has multiple interfaces, where the user can choose
1353	which interface to send on by specifying the IP address of that interface.
1354
1355	@item pkt_size=@var{size}
1356	Set the size in bytes of UDP packets.
1357
1358	@item reuse=@var{1|0}
1359	Explicitly allow or disallow reusing UDP sockets.
1360
1361	@item ttl=@var{ttl}
1362	Set the time to live value (for multicast only).
1363
1364	@item connect=@var{1|0}
1365	Initialize the UDP socket with @code{connect()}. In this case, the
1366	destination address can't be changed with ff_udp_set_remote_url later.
1367	If the destination address isn't known at the start, this option can
1368	be specified in ff_udp_set_remote_url, too.
1369	This allows finding out the source address for the packets with getsockname,
1370	and makes writes return with AVERROR(ECONNREFUSED) if "destination
1371	unreachable" is received.
1372	For receiving, this gives the benefit of only receiving packets from
1373	the specified peer address/port.
1374
1375	@item sources=@var{address}[,@var{address}]
1376	Only receive packets sent to the multicast group from one of the
1377	specified sender IP addresses.
1378
1379	@item block=@var{address}[,@var{address}]
1380	Ignore packets sent to the multicast group from the specified
1381	sender IP addresses.
1382
1383	@item fifo_size=@var{units}
1384	Set the UDP receiving circular buffer size, expressed as a number of
1385	packets with size of 188 bytes. If not specified defaults to 7*4096.
1386
1387	@item overrun_nonfatal=@var{1|0}
1388	Survive in case of UDP receiving circular buffer overrun. Default
1389	value is 0.
1390
1391	@item timeout=@var{microseconds}
1392	Set raise error timeout, expressed in microseconds.
1393
1394	This option is only relevant in read mode: if no data arrived in more
1395	than this time interval, raise error.
1396
1397	@item broadcast=@var{1|0}
1398	Explicitly allow or disallow UDP broadcasting.
1399
1400	Note that broadcasting may not work properly on networks having
1401	a broadcast storm protection.
1402	@end table
1403
1404	@subsection Examples
1405
1406	@itemize
1407	@item
1408	Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1409	@example
1410	ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1411	@end example
1412
1413	@item
1414	Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1415	sized UDP packets, using a large input buffer:
1416	@example
1417	ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1418	@end example
1419
1420	@item
1421	Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1422	@example
1423	ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1424	@end example
1425	@end itemize
1426
1427	@section unix
1428
1429	Unix local socket
1430
1431	The required syntax for a Unix socket URL is:
1432
1433	@example
1434	unix://@var{filepath}
1435	@end example
1436
1437	The following parameters can be set via command line options
1438	(or in code via @code{AVOption}s):
1439
1440	@table @option
1441	@item timeout
1442	Timeout in ms.
1443	@item listen
1444	Create the Unix socket in listening mode.
1445	@end table
1446
1447	@c man end PROTOCOLS
1448