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
82 Allow to read from or read to a file.
84 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
87 ffmpeg -i file:input.mpeg output.mpeg
90 The ff* tools default to the file protocol, that is a resource
91 specified with the name "FILE.mpeg" is interpreted as the URL
100 Read Apple HTTP Live Streaming compliant segmented stream as
101 a uniform one. The M3U8 playlists describing the segments can be
102 remote HTTP resources or local files, accessed using the standard
104 The nested protocol is declared by specifying
105 "+@var{proto}" after the hls URI scheme name, where @var{proto}
106 is either "file" or "http".
109 hls+http://host/path/to/remote/resource.m3u8
110 hls+file://path/to/local/resource.m3u8
113 Using this protocol is discouraged - the hls demuxer should work
114 just as well (if not, please report the issues) and is more complete.
115 To use the hls demuxer instead, simply use the direct URLs to the
120 HTTP (Hyper Text Transfer Protocol).
124 MMS (Microsoft Media Server) protocol over TCP.
128 MMS (Microsoft Media Server) protocol over HTTP.
130 The required syntax is:
132 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
139 Computes the MD5 hash of the data to be written, and on close writes
140 this to the designated output or stdout if none is specified. It can
141 be used to test muxers without writing an actual file.
143 Some examples follow.
145 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
146 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
148 # Write the MD5 hash of the encoded AVI file to stdout.
149 ffmpeg -i input.flv -f avi -y md5:
152 Note that some formats (typically MOV) require the output protocol to
153 be seekable, so they will fail with the MD5 output protocol.
157 UNIX pipe access protocol.
159 Allow to read and write from UNIX pipes.
161 The accepted syntax is:
166 @var{number} is the number corresponding to the file descriptor of the
167 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
168 is not specified, by default the stdout file descriptor will be used
169 for writing, stdin for reading.
171 For example to read from stdin with @command{ffmpeg}:
173 cat test.wav | ffmpeg -i pipe:0
174 # ...this is the same as...
175 cat test.wav | ffmpeg -i pipe:
178 For writing to stdout with @command{ffmpeg}:
180 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
181 # ...this is the same as...
182 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
185 Note that some formats (typically MOV), require the output protocol to
186 be seekable, so they will fail with the pipe output protocol.
190 Real-Time Messaging Protocol.
192 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
193 content across a TCP/IP network.
195 The required syntax is:
197 rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
200 The accepted parameters are:
204 The address of the RTMP server.
207 The number of the TCP port to use (by default is 1935).
210 It is the name of the application to access. It usually corresponds to
211 the path where the application is installed on the RTMP server
212 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
213 the value parsed from the URI through the @code{rtmp_app} option, too.
216 It is the path or name of the resource to play with reference to the
217 application specified in @var{app}, may be prefixed by "mp4:". You
218 can override the value parsed from the URI through the @code{rtmp_playpath}
223 Additionally, the following parameters can be set via command line options
224 (or in code via @code{AVOption}s):
228 Name of application to connect on the RTMP server. This option
229 overrides the parameter specified in the URI.
232 Set the client buffer time in milliseconds. The default is 3000.
235 Extra arbitrary AMF connection parameters, parsed from a string,
236 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
237 Each value is prefixed by a single character denoting the type,
238 B for Boolean, N for number, S for string, O for object, or Z for null,
239 followed by a colon. For Booleans the data must be either 0 or 1 for
240 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
241 1 to end or begin an object, respectively. Data items in subobjects may
242 be named, by prefixing the type with 'N' and specifying the name before
243 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
244 times to construct arbitrary AMF sequences.
247 Version of the Flash plugin used to run the SWF player. The default
250 @item rtmp_flush_interval
251 Number of packets flushed in the same request (RTMPT only). The default
255 Specify that the media is a live stream. No resuming or seeking in
256 live streams is possible. The default value is @code{any}, which means the
257 subscriber first tries to play the live stream specified in the
258 playpath. If a live stream of that name is not found, it plays the
259 recorded stream. The other possible values are @code{live} and
263 URL of the web page in which the media was embedded. By default no
267 Stream identifier to play or to publish. This option overrides the
268 parameter specified in the URI.
271 Name of live stream to subscribe to. By default no value will be sent.
272 It is only sent if the option is specified or if rtmp_live
276 URL of the SWF player for the media. By default no value will be sent.
279 URL of the target stream. Defaults to proto://host[:port]/app.
283 For example to read with @command{ffplay} a multimedia resource named
284 "sample" from the application "vod" from an RTMP server "myserver":
286 ffplay rtmp://myserver/vod/sample
291 Encrypted Real-Time Messaging Protocol.
293 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
294 streaming multimedia content within standard cryptographic primitives,
295 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
300 Real-Time Messaging Protocol over a secure SSL connection.
302 The Real-Time Messaging Protocol (RTMPS) is used for streaming
303 multimedia content across an encrypted connection.
307 Real-Time Messaging Protocol tunneled through HTTP.
309 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
310 for streaming multimedia content within HTTP requests to traverse
315 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
317 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
318 is used for streaming multimedia content within HTTP requests to traverse
323 Real-Time Messaging Protocol tunneled through HTTPS.
325 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
326 for streaming multimedia content within HTTPS requests to traverse
329 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
331 Real-Time Messaging Protocol and its variants supported through
334 Requires the presence of the librtmp headers and library during
335 configuration. You need to explicitly configure the build with
336 "--enable-librtmp". If enabled this will replace the native RTMP
339 This protocol provides most client functions and a few server
340 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
341 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
342 variants of these encrypted types (RTMPTE, RTMPTS).
344 The required syntax is:
346 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
349 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
350 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
351 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
352 meaning as specified for the RTMP native protocol.
353 @var{options} contains a list of space-separated options of the form
356 See the librtmp manual page (man 3 librtmp) for more information.
358 For example, to stream a file in real-time to an RTMP server using
361 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
364 To play the same stream using @command{ffplay}:
366 ffplay "rtmp://myserver/live/mystream live=1"
375 RTSP is not technically a protocol handler in libavformat, it is a demuxer
376 and muxer. The demuxer supports both normal RTSP (with data transferred
377 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
378 data transferred over RDT).
380 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
381 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
382 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
384 The required syntax for a RTSP url is:
386 rtsp://@var{hostname}[:@var{port}]/@var{path}
389 The following options (set on the @command{ffmpeg}/@command{ffplay} command
390 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
393 Flags for @code{rtsp_transport}:
398 Use UDP as lower transport protocol.
401 Use TCP (interleaving within the RTSP control channel) as lower
405 Use UDP multicast as lower transport protocol.
408 Use HTTP tunneling as lower transport protocol, which is useful for
412 Multiple lower transport protocols may be specified, in that case they are
413 tried one at a time (if the setup of one fails, the next one is tried).
414 For the muxer, only the @code{tcp} and @code{udp} options are supported.
416 Flags for @code{rtsp_flags}:
420 Accept packets only from negotiated peer address and port.
422 Act as a server, listening for an incoming connection.
425 When receiving data over UDP, the demuxer tries to reorder received packets
426 (since they may arrive out of order, or packets may get lost totally). This
427 can be disabled by setting the maximum demuxing delay to zero (via
428 the @code{max_delay} field of AVFormatContext).
430 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
431 streams to display can be chosen with @code{-vst} @var{n} and
432 @code{-ast} @var{n} for video and audio respectively, and can be switched
433 on the fly by pressing @code{v} and @code{a}.
435 Example command lines:
437 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
440 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
443 To watch a stream tunneled over HTTP:
446 ffplay -rtsp_transport http rtsp://server/video.mp4
449 To send a stream in realtime to a RTSP server, for others to watch:
452 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
455 To receive a stream in realtime:
458 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
463 Session Announcement Protocol (RFC 2974). This is not technically a
464 protocol handler in libavformat, it is a muxer and demuxer.
465 It is used for signalling of RTP streams, by announcing the SDP for the
466 streams regularly on a separate port.
470 The syntax for a SAP url given to the muxer is:
472 sap://@var{destination}[:@var{port}][?@var{options}]
475 The RTP packets are sent to @var{destination} on port @var{port},
476 or to port 5004 if no port is specified.
477 @var{options} is a @code{&}-separated list. The following options
482 @item announce_addr=@var{address}
483 Specify the destination IP address for sending the announcements to.
484 If omitted, the announcements are sent to the commonly used SAP
485 announcement multicast address 224.2.127.254 (sap.mcast.net), or
486 ff0e::2:7ffe if @var{destination} is an IPv6 address.
488 @item announce_port=@var{port}
489 Specify the port to send the announcements on, defaults to
490 9875 if not specified.
493 Specify the time to live value for the announcements and RTP packets,
496 @item same_port=@var{0|1}
497 If set to 1, send all RTP streams on the same port pair. If zero (the
498 default), all streams are sent on unique ports, with each stream on a
499 port 2 numbers higher than the previous.
500 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
501 The RTP stack in libavformat for receiving requires all streams to be sent
505 Example command lines follow.
507 To broadcast a stream on the local subnet, for watching in VLC:
510 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
513 Similarly, for watching in @command{ffplay}:
516 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
519 And for watching in @command{ffplay}, over IPv6:
522 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
527 The syntax for a SAP url given to the demuxer is:
529 sap://[@var{address}][:@var{port}]
532 @var{address} is the multicast address to listen for announcements on,
533 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
534 is the port that is listened on, 9875 if omitted.
536 The demuxers listens for announcements on the given address and port.
537 Once an announcement is received, it tries to receive that particular stream.
539 Example command lines follow.
541 To play back the first stream announced on the normal SAP multicast address:
547 To play back the first stream announced on one the default IPv6 SAP multicast address:
550 ffplay sap://[ff0e::2:7ffe]
555 Trasmission Control Protocol.
557 The required syntax for a TCP url is:
559 tcp://@var{hostname}:@var{port}[?@var{options}]
565 Listen for an incoming connection
568 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
569 ffplay tcp://@var{hostname}:@var{port}
576 Transport Layer Security/Secure Sockets Layer
578 The required syntax for a TLS/SSL url is:
580 tls://@var{hostname}:@var{port}[?@var{options}]
586 Act as a server, listening for an incoming connection.
588 @item cafile=@var{filename}
589 Certificate authority file. The file must be in OpenSSL PEM format.
591 @item cert=@var{filename}
592 Certificate file. The file must be in OpenSSL PEM format.
594 @item key=@var{filename}
597 @item verify=@var{0|1}
598 Verify the peer's certificate.
602 Example command lines:
604 To create a TLS/SSL server that serves an input stream.
607 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
610 To play back a stream from the TLS/SSL server using @command{ffplay}:
613 ffplay tls://@var{hostname}:@var{port}
618 User Datagram Protocol.
620 The required syntax for a UDP url is:
622 udp://@var{hostname}:@var{port}[?@var{options}]
625 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
627 In case threading is enabled on the system, a circular buffer is used
628 to store the incoming data, which allows to reduce loss of data due to
629 UDP socket buffer overruns. The @var{fifo_size} and
630 @var{overrun_nonfatal} options are related to this buffer.
632 The list of supported options follows.
636 @item buffer_size=@var{size}
637 Set the UDP socket buffer size in bytes. This is used both for the
638 receiving and the sending buffer size.
640 @item localport=@var{port}
641 Override the local UDP port to bind with.
643 @item localaddr=@var{addr}
644 Choose the local IP address. This is useful e.g. if sending multicast
645 and the host has multiple interfaces, where the user can choose
646 which interface to send on by specifying the IP address of that interface.
648 @item pkt_size=@var{size}
649 Set the size in bytes of UDP packets.
651 @item reuse=@var{1|0}
652 Explicitly allow or disallow reusing UDP sockets.
655 Set the time to live value (for multicast only).
657 @item connect=@var{1|0}
658 Initialize the UDP socket with @code{connect()}. In this case, the
659 destination address can't be changed with ff_udp_set_remote_url later.
660 If the destination address isn't known at the start, this option can
661 be specified in ff_udp_set_remote_url, too.
662 This allows finding out the source address for the packets with getsockname,
663 and makes writes return with AVERROR(ECONNREFUSED) if "destination
664 unreachable" is received.
665 For receiving, this gives the benefit of only receiving packets from
666 the specified peer address/port.
668 @item sources=@var{address}[,@var{address}]
669 Only receive packets sent to the multicast group from one of the
670 specified sender IP addresses.
672 @item block=@var{address}[,@var{address}]
673 Ignore packets sent to the multicast group from the specified
676 @item fifo_size=@var{units}
677 Set the UDP receiving circular buffer size, expressed as a number of
678 packets with size of 188 bytes. If not specified defaults to 7*4096.
680 @item overrun_nonfatal=@var{1|0}
681 Survive in case of UDP receiving circular buffer overrun. Default
685 Some usage examples of the UDP protocol with @command{ffmpeg} follow.
687 To stream over UDP to a remote endpoint:
689 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
692 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
694 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
697 To receive over UDP from a remote endpoint:
699 ffmpeg -i udp://[@var{multicast-address}]:@var{port}