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 tunneled through HTTP.
284 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
285 for streaming multimedia content within HTTP requests to traverse
288 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
290 Real-Time Messaging Protocol and its variants supported through
293 Requires the presence of the librtmp headers and library during
294 configuration. You need to explicitly configure the build with
295 "--enable-librtmp". If enabled this will replace the native RTMP
298 This protocol provides most client functions and a few server
299 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
300 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
301 variants of these encrypted types (RTMPTE, RTMPTS).
303 The required syntax is:
305 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
308 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
309 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
310 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
311 meaning as specified for the RTMP native protocol.
312 @var{options} contains a list of space-separated options of the form
315 See the librtmp manual page (man 3 librtmp) for more information.
317 For example, to stream a file in real-time to an RTMP server using
320 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
323 To play the same stream using @command{ffplay}:
325 ffplay "rtmp://myserver/live/mystream live=1"
334 RTSP is not technically a protocol handler in libavformat, it is a demuxer
335 and muxer. The demuxer supports both normal RTSP (with data transferred
336 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
337 data transferred over RDT).
339 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
340 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
341 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
343 The required syntax for a RTSP url is:
345 rtsp://@var{hostname}[:@var{port}]/@var{path}
348 The following options (set on the @command{ffmpeg}/@command{ffplay} command
349 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
352 Flags for @code{rtsp_transport}:
357 Use UDP as lower transport protocol.
360 Use TCP (interleaving within the RTSP control channel) as lower
364 Use UDP multicast as lower transport protocol.
367 Use HTTP tunneling as lower transport protocol, which is useful for
371 Multiple lower transport protocols may be specified, in that case they are
372 tried one at a time (if the setup of one fails, the next one is tried).
373 For the muxer, only the @code{tcp} and @code{udp} options are supported.
375 Flags for @code{rtsp_flags}:
379 Accept packets only from negotiated peer address and port.
382 When receiving data over UDP, the demuxer tries to reorder received packets
383 (since they may arrive out of order, or packets may get lost totally). This
384 can be disabled by setting the maximum demuxing delay to zero (via
385 the @code{max_delay} field of AVFormatContext).
387 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
388 streams to display can be chosen with @code{-vst} @var{n} and
389 @code{-ast} @var{n} for video and audio respectively, and can be switched
390 on the fly by pressing @code{v} and @code{a}.
392 Example command lines:
394 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
397 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
400 To watch a stream tunneled over HTTP:
403 ffplay -rtsp_transport http rtsp://server/video.mp4
406 To send a stream in realtime to a RTSP server, for others to watch:
409 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
414 Session Announcement Protocol (RFC 2974). This is not technically a
415 protocol handler in libavformat, it is a muxer and demuxer.
416 It is used for signalling of RTP streams, by announcing the SDP for the
417 streams regularly on a separate port.
421 The syntax for a SAP url given to the muxer is:
423 sap://@var{destination}[:@var{port}][?@var{options}]
426 The RTP packets are sent to @var{destination} on port @var{port},
427 or to port 5004 if no port is specified.
428 @var{options} is a @code{&}-separated list. The following options
433 @item announce_addr=@var{address}
434 Specify the destination IP address for sending the announcements to.
435 If omitted, the announcements are sent to the commonly used SAP
436 announcement multicast address 224.2.127.254 (sap.mcast.net), or
437 ff0e::2:7ffe if @var{destination} is an IPv6 address.
439 @item announce_port=@var{port}
440 Specify the port to send the announcements on, defaults to
441 9875 if not specified.
444 Specify the time to live value for the announcements and RTP packets,
447 @item same_port=@var{0|1}
448 If set to 1, send all RTP streams on the same port pair. If zero (the
449 default), all streams are sent on unique ports, with each stream on a
450 port 2 numbers higher than the previous.
451 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
452 The RTP stack in libavformat for receiving requires all streams to be sent
456 Example command lines follow.
458 To broadcast a stream on the local subnet, for watching in VLC:
461 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
464 Similarly, for watching in @command{ffplay}:
467 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
470 And for watching in @command{ffplay}, over IPv6:
473 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
478 The syntax for a SAP url given to the demuxer is:
480 sap://[@var{address}][:@var{port}]
483 @var{address} is the multicast address to listen for announcements on,
484 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
485 is the port that is listened on, 9875 if omitted.
487 The demuxers listens for announcements on the given address and port.
488 Once an announcement is received, it tries to receive that particular stream.
490 Example command lines follow.
492 To play back the first stream announced on the normal SAP multicast address:
498 To play back the first stream announced on one the default IPv6 SAP multicast address:
501 ffplay sap://[ff0e::2:7ffe]
506 Trasmission Control Protocol.
508 The required syntax for a TCP url is:
510 tcp://@var{hostname}:@var{port}[?@var{options}]
516 Listen for an incoming connection
519 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
520 ffplay tcp://@var{hostname}:@var{port}
527 User Datagram Protocol.
529 The required syntax for a UDP url is:
531 udp://@var{hostname}:@var{port}[?@var{options}]
534 @var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
535 Follow the list of supported options.
539 @item buffer_size=@var{size}
540 set the UDP buffer size in bytes
542 @item localport=@var{port}
543 override the local UDP port to bind with
545 @item localaddr=@var{addr}
546 Choose the local IP address. This is useful e.g. if sending multicast
547 and the host has multiple interfaces, where the user can choose
548 which interface to send on by specifying the IP address of that interface.
550 @item pkt_size=@var{size}
551 set the size in bytes of UDP packets
553 @item reuse=@var{1|0}
554 explicitly allow or disallow reusing UDP sockets
557 set the time to live value (for multicast only)
559 @item connect=@var{1|0}
560 Initialize the UDP socket with @code{connect()}. In this case, the
561 destination address can't be changed with ff_udp_set_remote_url later.
562 If the destination address isn't known at the start, this option can
563 be specified in ff_udp_set_remote_url, too.
564 This allows finding out the source address for the packets with getsockname,
565 and makes writes return with AVERROR(ECONNREFUSED) if "destination
566 unreachable" is received.
567 For receiving, this gives the benefit of only receiving packets from
568 the specified peer address/port.
571 Some usage examples of the udp protocol with @command{ffmpeg} follow.
573 To stream over UDP to a remote endpoint:
575 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
578 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
580 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
583 To receive over UDP from a remote endpoint:
585 ffmpeg -i udp://[@var{multicast-address}]:@var{port}