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 Stream identifier to play or to publish. This option overrides the
264 parameter specified in the URI.
267 URL of the SWF player for the media. By default no value will be sent.
270 URL of the target stream.
274 For example to read with @command{ffplay} a multimedia resource named
275 "sample" from the application "vod" from an RTMP server "myserver":
277 ffplay rtmp://myserver/vod/sample
282 Real-Time Messaging Protocol over a secure SSL connection.
284 The Real-Time Messaging Protocol (RTMPS) is used for streaming
285 multimedia content across an encrypted connection.
289 Real-Time Messaging Protocol tunneled through HTTP.
291 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
292 for streaming multimedia content within HTTP requests to traverse
297 Real-Time Messaging Protocol tunneled through HTTPS.
299 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
300 for streaming multimedia content within HTTPS requests to traverse
303 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
305 Real-Time Messaging Protocol and its variants supported through
308 Requires the presence of the librtmp headers and library during
309 configuration. You need to explicitly configure the build with
310 "--enable-librtmp". If enabled this will replace the native RTMP
313 This protocol provides most client functions and a few server
314 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
315 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
316 variants of these encrypted types (RTMPTE, RTMPTS).
318 The required syntax is:
320 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
323 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
324 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
325 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
326 meaning as specified for the RTMP native protocol.
327 @var{options} contains a list of space-separated options of the form
330 See the librtmp manual page (man 3 librtmp) for more information.
332 For example, to stream a file in real-time to an RTMP server using
335 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
338 To play the same stream using @command{ffplay}:
340 ffplay "rtmp://myserver/live/mystream live=1"
349 RTSP is not technically a protocol handler in libavformat, it is a demuxer
350 and muxer. The demuxer supports both normal RTSP (with data transferred
351 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
352 data transferred over RDT).
354 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
355 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
356 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
358 The required syntax for a RTSP url is:
360 rtsp://@var{hostname}[:@var{port}]/@var{path}
363 The following options (set on the @command{ffmpeg}/@command{ffplay} command
364 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
367 Flags for @code{rtsp_transport}:
372 Use UDP as lower transport protocol.
375 Use TCP (interleaving within the RTSP control channel) as lower
379 Use UDP multicast as lower transport protocol.
382 Use HTTP tunneling as lower transport protocol, which is useful for
386 Multiple lower transport protocols may be specified, in that case they are
387 tried one at a time (if the setup of one fails, the next one is tried).
388 For the muxer, only the @code{tcp} and @code{udp} options are supported.
390 Flags for @code{rtsp_flags}:
394 Accept packets only from negotiated peer address and port.
396 Act as a server, listening for an incoming connection.
399 When receiving data over UDP, the demuxer tries to reorder received packets
400 (since they may arrive out of order, or packets may get lost totally). This
401 can be disabled by setting the maximum demuxing delay to zero (via
402 the @code{max_delay} field of AVFormatContext).
404 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
405 streams to display can be chosen with @code{-vst} @var{n} and
406 @code{-ast} @var{n} for video and audio respectively, and can be switched
407 on the fly by pressing @code{v} and @code{a}.
409 Example command lines:
411 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
414 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
417 To watch a stream tunneled over HTTP:
420 ffplay -rtsp_transport http rtsp://server/video.mp4
423 To send a stream in realtime to a RTSP server, for others to watch:
426 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
429 To receive a stream in realtime:
432 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
437 Session Announcement Protocol (RFC 2974). This is not technically a
438 protocol handler in libavformat, it is a muxer and demuxer.
439 It is used for signalling of RTP streams, by announcing the SDP for the
440 streams regularly on a separate port.
444 The syntax for a SAP url given to the muxer is:
446 sap://@var{destination}[:@var{port}][?@var{options}]
449 The RTP packets are sent to @var{destination} on port @var{port},
450 or to port 5004 if no port is specified.
451 @var{options} is a @code{&}-separated list. The following options
456 @item announce_addr=@var{address}
457 Specify the destination IP address for sending the announcements to.
458 If omitted, the announcements are sent to the commonly used SAP
459 announcement multicast address 224.2.127.254 (sap.mcast.net), or
460 ff0e::2:7ffe if @var{destination} is an IPv6 address.
462 @item announce_port=@var{port}
463 Specify the port to send the announcements on, defaults to
464 9875 if not specified.
467 Specify the time to live value for the announcements and RTP packets,
470 @item same_port=@var{0|1}
471 If set to 1, send all RTP streams on the same port pair. If zero (the
472 default), all streams are sent on unique ports, with each stream on a
473 port 2 numbers higher than the previous.
474 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
475 The RTP stack in libavformat for receiving requires all streams to be sent
479 Example command lines follow.
481 To broadcast a stream on the local subnet, for watching in VLC:
484 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
487 Similarly, for watching in @command{ffplay}:
490 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
493 And for watching in @command{ffplay}, over IPv6:
496 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
501 The syntax for a SAP url given to the demuxer is:
503 sap://[@var{address}][:@var{port}]
506 @var{address} is the multicast address to listen for announcements on,
507 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
508 is the port that is listened on, 9875 if omitted.
510 The demuxers listens for announcements on the given address and port.
511 Once an announcement is received, it tries to receive that particular stream.
513 Example command lines follow.
515 To play back the first stream announced on the normal SAP multicast address:
521 To play back the first stream announced on one the default IPv6 SAP multicast address:
524 ffplay sap://[ff0e::2:7ffe]
529 Trasmission Control Protocol.
531 The required syntax for a TCP url is:
533 tcp://@var{hostname}:@var{port}[?@var{options}]
539 Listen for an incoming connection
542 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
543 ffplay tcp://@var{hostname}:@var{port}
550 User Datagram Protocol.
552 The required syntax for a UDP url is:
554 udp://@var{hostname}:@var{port}[?@var{options}]
557 @var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
558 Follow the list of supported options.
562 @item buffer_size=@var{size}
563 set the UDP buffer size in bytes
565 @item localport=@var{port}
566 override the local UDP port to bind with
568 @item localaddr=@var{addr}
569 Choose the local IP address. This is useful e.g. if sending multicast
570 and the host has multiple interfaces, where the user can choose
571 which interface to send on by specifying the IP address of that interface.
573 @item pkt_size=@var{size}
574 set the size in bytes of UDP packets
576 @item reuse=@var{1|0}
577 explicitly allow or disallow reusing UDP sockets
580 set the time to live value (for multicast only)
582 @item connect=@var{1|0}
583 Initialize the UDP socket with @code{connect()}. In this case, the
584 destination address can't be changed with ff_udp_set_remote_url later.
585 If the destination address isn't known at the start, this option can
586 be specified in ff_udp_set_remote_url, too.
587 This allows finding out the source address for the packets with getsockname,
588 and makes writes return with AVERROR(ECONNREFUSED) if "destination
589 unreachable" is received.
590 For receiving, this gives the benefit of only receiving packets from
591 the specified peer address/port.
593 @item sources=@var{address}[,@var{address}]
594 Only receive packets sent to the multicast group from one of the
595 specified sender IP addresses.
597 @item block=@var{address}[,@var{address}]
598 Ignore packets sent to the multicast group from the specified
602 Some usage examples of the udp protocol with @command{ffmpeg} follow.
604 To stream over UDP to a remote endpoint:
606 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
609 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
611 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
614 To receive over UDP from a remote endpoint:
616 ffmpeg -i udp://[@var{multicast-address}]:@var{port}