4 Protocols are configured elements in FFmpeg which allow to access
5 resources which require the use of a particular protocol.
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 Physical concatenation protocol.
56 Allow to read and seek from many resource in sequence as if they were
59 A URL accepted by this protocol has the syntax:
61 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
64 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
65 resource to be concatenated, each one possibly specifying a distinct
68 For example to read a sequence of files @file{split1.mpeg},
69 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
72 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
75 Note that you may need to escape the character "|" which is special for
80 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
82 For example, to convert a GIF file given inline with @command{ffmpeg}:
84 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
91 Allow to read from or read to a file.
93 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
96 ffmpeg -i file:input.mpeg output.mpeg
99 The ff* tools default to the file protocol, that is a resource
100 specified with the name "FILE.mpeg" is interpreted as the URL
109 Read Apple HTTP Live Streaming compliant segmented stream as
110 a uniform one. The M3U8 playlists describing the segments can be
111 remote HTTP resources or local files, accessed using the standard
113 The nested protocol is declared by specifying
114 "+@var{proto}" after the hls URI scheme name, where @var{proto}
115 is either "file" or "http".
118 hls+http://host/path/to/remote/resource.m3u8
119 hls+file://path/to/local/resource.m3u8
122 Using this protocol is discouraged - the hls demuxer should work
123 just as well (if not, please report the issues) and is more complete.
124 To use the hls demuxer instead, simply use the direct URLs to the
129 HTTP (Hyper Text Transfer Protocol).
131 This protocol accepts the following options.
135 Control seekability of connection. If set to 1 the resource is
136 supposed to be seekable, if set to 0 it is assumed not to be seekable,
137 if set to -1 it will try to autodetect if it is seekable. Default
141 If set to 1 use chunked transfer-encoding for posts, default is 1.
144 Set custom HTTP headers, can override built in default headers. The
145 value must be a string encoding the headers.
148 Force a content type.
151 Override User-Agent header. If not specified the protocol will use a
152 string describing the libavformat build.
154 @item multiple_requests
155 Use persistent connections if set to 1. By default it is 0.
158 Set custom HTTP post data.
161 Set timeout of socket I/O operations used by the underlying low level
162 operation. By default it is set to -1, which means that the timeout is
169 Set the cookies to be sent in future requests. The format of each cookie is the
170 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
171 delimited by a newline character.
174 @subsection HTTP Cookies
176 Some HTTP requests will be denied unless cookie values are passed in with the
177 request. The @option{cookies} option allows these cookies to be specified. At
178 the very least, each cookie must specify a value along with a path and domain.
179 HTTP requests that match both the domain and path will automatically include the
180 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
183 The required syntax to play a stream specifying a cookie is:
185 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
190 MMS (Microsoft Media Server) protocol over TCP.
194 MMS (Microsoft Media Server) protocol over HTTP.
196 The required syntax is:
198 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
205 Computes the MD5 hash of the data to be written, and on close writes
206 this to the designated output or stdout if none is specified. It can
207 be used to test muxers without writing an actual file.
209 Some examples follow.
211 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
212 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
214 # Write the MD5 hash of the encoded AVI file to stdout.
215 ffmpeg -i input.flv -f avi -y md5:
218 Note that some formats (typically MOV) require the output protocol to
219 be seekable, so they will fail with the MD5 output protocol.
223 UNIX pipe access protocol.
225 Allow to read and write from UNIX pipes.
227 The accepted syntax is:
232 @var{number} is the number corresponding to the file descriptor of the
233 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
234 is not specified, by default the stdout file descriptor will be used
235 for writing, stdin for reading.
237 For example to read from stdin with @command{ffmpeg}:
239 cat test.wav | ffmpeg -i pipe:0
240 # ...this is the same as...
241 cat test.wav | ffmpeg -i pipe:
244 For writing to stdout with @command{ffmpeg}:
246 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
247 # ...this is the same as...
248 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
251 Note that some formats (typically MOV), require the output protocol to
252 be seekable, so they will fail with the pipe output protocol.
256 Real-Time Messaging Protocol.
258 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
259 content across a TCP/IP network.
261 The required syntax is:
263 rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
266 The accepted parameters are:
270 The address of the RTMP server.
273 The number of the TCP port to use (by default is 1935).
276 It is the name of the application to access. It usually corresponds to
277 the path where the application is installed on the RTMP server
278 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
279 the value parsed from the URI through the @code{rtmp_app} option, too.
282 It is the path or name of the resource to play with reference to the
283 application specified in @var{app}, may be prefixed by "mp4:". You
284 can override the value parsed from the URI through the @code{rtmp_playpath}
288 Act as a server, listening for an incoming connection.
291 Maximum time to wait for the incoming connection. Implies listen.
294 Additionally, the following parameters can be set via command line options
295 (or in code via @code{AVOption}s):
299 Name of application to connect on the RTMP server. This option
300 overrides the parameter specified in the URI.
303 Set the client buffer time in milliseconds. The default is 3000.
306 Extra arbitrary AMF connection parameters, parsed from a string,
307 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
308 Each value is prefixed by a single character denoting the type,
309 B for Boolean, N for number, S for string, O for object, or Z for null,
310 followed by a colon. For Booleans the data must be either 0 or 1 for
311 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
312 1 to end or begin an object, respectively. Data items in subobjects may
313 be named, by prefixing the type with 'N' and specifying the name before
314 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
315 times to construct arbitrary AMF sequences.
318 Version of the Flash plugin used to run the SWF player. The default
321 @item rtmp_flush_interval
322 Number of packets flushed in the same request (RTMPT only). The default
326 Specify that the media is a live stream. No resuming or seeking in
327 live streams is possible. The default value is @code{any}, which means the
328 subscriber first tries to play the live stream specified in the
329 playpath. If a live stream of that name is not found, it plays the
330 recorded stream. The other possible values are @code{live} and
334 URL of the web page in which the media was embedded. By default no
338 Stream identifier to play or to publish. This option overrides the
339 parameter specified in the URI.
342 Name of live stream to subscribe to. By default no value will be sent.
343 It is only sent if the option is specified or if rtmp_live
347 SHA256 hash of the decompressed SWF file (32 bytes).
350 Size of the decompressed SWF file, required for SWFVerification.
353 URL of the SWF player for the media. By default no value will be sent.
356 URL to player swf file, compute hash/size automatically.
359 URL of the target stream. Defaults to proto://host[:port]/app.
363 For example to read with @command{ffplay} a multimedia resource named
364 "sample" from the application "vod" from an RTMP server "myserver":
366 ffplay rtmp://myserver/vod/sample
371 Encrypted Real-Time Messaging Protocol.
373 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
374 streaming multimedia content within standard cryptographic primitives,
375 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
380 Real-Time Messaging Protocol over a secure SSL connection.
382 The Real-Time Messaging Protocol (RTMPS) is used for streaming
383 multimedia content across an encrypted connection.
387 Real-Time Messaging Protocol tunneled through HTTP.
389 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
390 for streaming multimedia content within HTTP requests to traverse
395 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
397 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
398 is used for streaming multimedia content within HTTP requests to traverse
403 Real-Time Messaging Protocol tunneled through HTTPS.
405 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
406 for streaming multimedia content within HTTPS requests to traverse
409 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
411 Real-Time Messaging Protocol and its variants supported through
414 Requires the presence of the librtmp headers and library during
415 configuration. You need to explicitly configure the build with
416 "--enable-librtmp". If enabled this will replace the native RTMP
419 This protocol provides most client functions and a few server
420 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
421 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
422 variants of these encrypted types (RTMPTE, RTMPTS).
424 The required syntax is:
426 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
429 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
430 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
431 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
432 meaning as specified for the RTMP native protocol.
433 @var{options} contains a list of space-separated options of the form
436 See the librtmp manual page (man 3 librtmp) for more information.
438 For example, to stream a file in real-time to an RTMP server using
441 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
444 To play the same stream using @command{ffplay}:
446 ffplay "rtmp://myserver/live/mystream live=1"
455 RTSP is not technically a protocol handler in libavformat, it is a demuxer
456 and muxer. The demuxer supports both normal RTSP (with data transferred
457 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
458 data transferred over RDT).
460 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
461 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
462 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
464 The required syntax for a RTSP url is:
466 rtsp://@var{hostname}[:@var{port}]/@var{path}
469 The following options (set on the @command{ffmpeg}/@command{ffplay} command
470 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
473 Flags for @code{rtsp_transport}:
478 Use UDP as lower transport protocol.
481 Use TCP (interleaving within the RTSP control channel) as lower
485 Use UDP multicast as lower transport protocol.
488 Use HTTP tunneling as lower transport protocol, which is useful for
492 Multiple lower transport protocols may be specified, in that case they are
493 tried one at a time (if the setup of one fails, the next one is tried).
494 For the muxer, only the @code{tcp} and @code{udp} options are supported.
496 Flags for @code{rtsp_flags}:
500 Accept packets only from negotiated peer address and port.
502 Act as a server, listening for an incoming connection.
505 When receiving data over UDP, the demuxer tries to reorder received packets
506 (since they may arrive out of order, or packets may get lost totally). This
507 can be disabled by setting the maximum demuxing delay to zero (via
508 the @code{max_delay} field of AVFormatContext).
510 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
511 streams to display can be chosen with @code{-vst} @var{n} and
512 @code{-ast} @var{n} for video and audio respectively, and can be switched
513 on the fly by pressing @code{v} and @code{a}.
515 Example command lines:
517 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
520 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
523 To watch a stream tunneled over HTTP:
526 ffplay -rtsp_transport http rtsp://server/video.mp4
529 To send a stream in realtime to a RTSP server, for others to watch:
532 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
535 To receive a stream in realtime:
538 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
543 Socket IO timeout in micro seconds.
548 Session Announcement Protocol (RFC 2974). This is not technically a
549 protocol handler in libavformat, it is a muxer and demuxer.
550 It is used for signalling of RTP streams, by announcing the SDP for the
551 streams regularly on a separate port.
555 The syntax for a SAP url given to the muxer is:
557 sap://@var{destination}[:@var{port}][?@var{options}]
560 The RTP packets are sent to @var{destination} on port @var{port},
561 or to port 5004 if no port is specified.
562 @var{options} is a @code{&}-separated list. The following options
567 @item announce_addr=@var{address}
568 Specify the destination IP address for sending the announcements to.
569 If omitted, the announcements are sent to the commonly used SAP
570 announcement multicast address 224.2.127.254 (sap.mcast.net), or
571 ff0e::2:7ffe if @var{destination} is an IPv6 address.
573 @item announce_port=@var{port}
574 Specify the port to send the announcements on, defaults to
575 9875 if not specified.
578 Specify the time to live value for the announcements and RTP packets,
581 @item same_port=@var{0|1}
582 If set to 1, send all RTP streams on the same port pair. If zero (the
583 default), all streams are sent on unique ports, with each stream on a
584 port 2 numbers higher than the previous.
585 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
586 The RTP stack in libavformat for receiving requires all streams to be sent
590 Example command lines follow.
592 To broadcast a stream on the local subnet, for watching in VLC:
595 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
598 Similarly, for watching in @command{ffplay}:
601 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
604 And for watching in @command{ffplay}, over IPv6:
607 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
612 The syntax for a SAP url given to the demuxer is:
614 sap://[@var{address}][:@var{port}]
617 @var{address} is the multicast address to listen for announcements on,
618 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
619 is the port that is listened on, 9875 if omitted.
621 The demuxers listens for announcements on the given address and port.
622 Once an announcement is received, it tries to receive that particular stream.
624 Example command lines follow.
626 To play back the first stream announced on the normal SAP multicast address:
632 To play back the first stream announced on one the default IPv6 SAP multicast address:
635 ffplay sap://[ff0e::2:7ffe]
640 Trasmission Control Protocol.
642 The required syntax for a TCP url is:
644 tcp://@var{hostname}:@var{port}[?@var{options}]
650 Listen for an incoming connection
652 @item timeout=@var{microseconds}
653 In read mode: if no data arrived in more than this time interval, raise error.
654 In write mode: if socket cannot be written in more than this time interval, raise error.
655 This also sets timeout on TCP connection establishing.
658 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
659 ffplay tcp://@var{hostname}:@var{port}
666 Transport Layer Security/Secure Sockets Layer
668 The required syntax for a TLS/SSL url is:
670 tls://@var{hostname}:@var{port}[?@var{options}]
676 Act as a server, listening for an incoming connection.
678 @item cafile=@var{filename}
679 Certificate authority file. The file must be in OpenSSL PEM format.
681 @item cert=@var{filename}
682 Certificate file. The file must be in OpenSSL PEM format.
684 @item key=@var{filename}
687 @item verify=@var{0|1}
688 Verify the peer's certificate.
692 Example command lines:
694 To create a TLS/SSL server that serves an input stream.
697 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
700 To play back a stream from the TLS/SSL server using @command{ffplay}:
703 ffplay tls://@var{hostname}:@var{port}
708 User Datagram Protocol.
710 The required syntax for a UDP url is:
712 udp://@var{hostname}:@var{port}[?@var{options}]
715 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
717 In case threading is enabled on the system, a circular buffer is used
718 to store the incoming data, which allows to reduce loss of data due to
719 UDP socket buffer overruns. The @var{fifo_size} and
720 @var{overrun_nonfatal} options are related to this buffer.
722 The list of supported options follows.
726 @item buffer_size=@var{size}
727 Set the UDP socket buffer size in bytes. This is used both for the
728 receiving and the sending buffer size.
730 @item localport=@var{port}
731 Override the local UDP port to bind with.
733 @item localaddr=@var{addr}
734 Choose the local IP address. This is useful e.g. if sending multicast
735 and the host has multiple interfaces, where the user can choose
736 which interface to send on by specifying the IP address of that interface.
738 @item pkt_size=@var{size}
739 Set the size in bytes of UDP packets.
741 @item reuse=@var{1|0}
742 Explicitly allow or disallow reusing UDP sockets.
745 Set the time to live value (for multicast only).
747 @item connect=@var{1|0}
748 Initialize the UDP socket with @code{connect()}. In this case, the
749 destination address can't be changed with ff_udp_set_remote_url later.
750 If the destination address isn't known at the start, this option can
751 be specified in ff_udp_set_remote_url, too.
752 This allows finding out the source address for the packets with getsockname,
753 and makes writes return with AVERROR(ECONNREFUSED) if "destination
754 unreachable" is received.
755 For receiving, this gives the benefit of only receiving packets from
756 the specified peer address/port.
758 @item sources=@var{address}[,@var{address}]
759 Only receive packets sent to the multicast group from one of the
760 specified sender IP addresses.
762 @item block=@var{address}[,@var{address}]
763 Ignore packets sent to the multicast group from the specified
766 @item fifo_size=@var{units}
767 Set the UDP receiving circular buffer size, expressed as a number of
768 packets with size of 188 bytes. If not specified defaults to 7*4096.
770 @item overrun_nonfatal=@var{1|0}
771 Survive in case of UDP receiving circular buffer overrun. Default
774 @item timeout=@var{microseconds}
775 In read mode: if no data arrived in more than this time interval, raise error.
778 Some usage examples of the UDP protocol with @command{ffmpeg} follow.
780 To stream over UDP to a remote endpoint:
782 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
785 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
787 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
790 To receive over UDP from a remote endpoint:
792 ffmpeg -i udp://[@var{multicast-address}]:@var{port}