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}
222 Act as a server, listening for an incoming connection.
225 Maximum time to wait for the incoming connection. Implies listen.
228 Additionally, the following parameters can be set via command line options
229 (or in code via @code{AVOption}s):
233 Name of application to connect on the RTMP server. This option
234 overrides the parameter specified in the URI.
237 Set the client buffer time in milliseconds. The default is 3000.
240 Extra arbitrary AMF connection parameters, parsed from a string,
241 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
242 Each value is prefixed by a single character denoting the type,
243 B for Boolean, N for number, S for string, O for object, or Z for null,
244 followed by a colon. For Booleans the data must be either 0 or 1 for
245 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
246 1 to end or begin an object, respectively. Data items in subobjects may
247 be named, by prefixing the type with 'N' and specifying the name before
248 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
249 times to construct arbitrary AMF sequences.
252 Version of the Flash plugin used to run the SWF player. The default
255 @item rtmp_flush_interval
256 Number of packets flushed in the same request (RTMPT only). The default
260 Specify that the media is a live stream. No resuming or seeking in
261 live streams is possible. The default value is @code{any}, which means the
262 subscriber first tries to play the live stream specified in the
263 playpath. If a live stream of that name is not found, it plays the
264 recorded stream. The other possible values are @code{live} and
268 URL of the web page in which the media was embedded. By default no
272 Stream identifier to play or to publish. This option overrides the
273 parameter specified in the URI.
276 Name of live stream to subscribe to. By default no value will be sent.
277 It is only sent if the option is specified or if rtmp_live
281 SHA256 hash of the decompressed SWF file (32 bytes).
284 Size of the decompressed SWF file, required for SWFVerification.
287 URL of the SWF player for the media. By default no value will be sent.
290 URL to player swf file, compute hash/size automatically.
293 URL of the target stream. Defaults to proto://host[:port]/app.
297 For example to read with @command{ffplay} a multimedia resource named
298 "sample" from the application "vod" from an RTMP server "myserver":
300 ffplay rtmp://myserver/vod/sample
305 Encrypted Real-Time Messaging Protocol.
307 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
308 streaming multimedia content within standard cryptographic primitives,
309 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
314 Real-Time Messaging Protocol over a secure SSL connection.
316 The Real-Time Messaging Protocol (RTMPS) is used for streaming
317 multimedia content across an encrypted connection.
321 Real-Time Messaging Protocol tunneled through HTTP.
323 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
324 for streaming multimedia content within HTTP requests to traverse
329 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
331 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
332 is used for streaming multimedia content within HTTP requests to traverse
337 Real-Time Messaging Protocol tunneled through HTTPS.
339 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
340 for streaming multimedia content within HTTPS requests to traverse
343 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
345 Real-Time Messaging Protocol and its variants supported through
348 Requires the presence of the librtmp headers and library during
349 configuration. You need to explicitly configure the build with
350 "--enable-librtmp". If enabled this will replace the native RTMP
353 This protocol provides most client functions and a few server
354 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
355 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
356 variants of these encrypted types (RTMPTE, RTMPTS).
358 The required syntax is:
360 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
363 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
364 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
365 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
366 meaning as specified for the RTMP native protocol.
367 @var{options} contains a list of space-separated options of the form
370 See the librtmp manual page (man 3 librtmp) for more information.
372 For example, to stream a file in real-time to an RTMP server using
375 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
378 To play the same stream using @command{ffplay}:
380 ffplay "rtmp://myserver/live/mystream live=1"
389 RTSP is not technically a protocol handler in libavformat, it is a demuxer
390 and muxer. The demuxer supports both normal RTSP (with data transferred
391 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
392 data transferred over RDT).
394 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
395 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
396 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
398 The required syntax for a RTSP url is:
400 rtsp://@var{hostname}[:@var{port}]/@var{path}
403 The following options (set on the @command{ffmpeg}/@command{ffplay} command
404 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
407 Flags for @code{rtsp_transport}:
412 Use UDP as lower transport protocol.
415 Use TCP (interleaving within the RTSP control channel) as lower
419 Use UDP multicast as lower transport protocol.
422 Use HTTP tunneling as lower transport protocol, which is useful for
426 Multiple lower transport protocols may be specified, in that case they are
427 tried one at a time (if the setup of one fails, the next one is tried).
428 For the muxer, only the @code{tcp} and @code{udp} options are supported.
430 Flags for @code{rtsp_flags}:
434 Accept packets only from negotiated peer address and port.
436 Act as a server, listening for an incoming connection.
439 When receiving data over UDP, the demuxer tries to reorder received packets
440 (since they may arrive out of order, or packets may get lost totally). This
441 can be disabled by setting the maximum demuxing delay to zero (via
442 the @code{max_delay} field of AVFormatContext).
444 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
445 streams to display can be chosen with @code{-vst} @var{n} and
446 @code{-ast} @var{n} for video and audio respectively, and can be switched
447 on the fly by pressing @code{v} and @code{a}.
449 Example command lines:
451 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
454 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
457 To watch a stream tunneled over HTTP:
460 ffplay -rtsp_transport http rtsp://server/video.mp4
463 To send a stream in realtime to a RTSP server, for others to watch:
466 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
469 To receive a stream in realtime:
472 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
477 Session Announcement Protocol (RFC 2974). This is not technically a
478 protocol handler in libavformat, it is a muxer and demuxer.
479 It is used for signalling of RTP streams, by announcing the SDP for the
480 streams regularly on a separate port.
484 The syntax for a SAP url given to the muxer is:
486 sap://@var{destination}[:@var{port}][?@var{options}]
489 The RTP packets are sent to @var{destination} on port @var{port},
490 or to port 5004 if no port is specified.
491 @var{options} is a @code{&}-separated list. The following options
496 @item announce_addr=@var{address}
497 Specify the destination IP address for sending the announcements to.
498 If omitted, the announcements are sent to the commonly used SAP
499 announcement multicast address 224.2.127.254 (sap.mcast.net), or
500 ff0e::2:7ffe if @var{destination} is an IPv6 address.
502 @item announce_port=@var{port}
503 Specify the port to send the announcements on, defaults to
504 9875 if not specified.
507 Specify the time to live value for the announcements and RTP packets,
510 @item same_port=@var{0|1}
511 If set to 1, send all RTP streams on the same port pair. If zero (the
512 default), all streams are sent on unique ports, with each stream on a
513 port 2 numbers higher than the previous.
514 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
515 The RTP stack in libavformat for receiving requires all streams to be sent
519 Example command lines follow.
521 To broadcast a stream on the local subnet, for watching in VLC:
524 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
527 Similarly, for watching in @command{ffplay}:
530 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
533 And for watching in @command{ffplay}, over IPv6:
536 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
541 The syntax for a SAP url given to the demuxer is:
543 sap://[@var{address}][:@var{port}]
546 @var{address} is the multicast address to listen for announcements on,
547 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
548 is the port that is listened on, 9875 if omitted.
550 The demuxers listens for announcements on the given address and port.
551 Once an announcement is received, it tries to receive that particular stream.
553 Example command lines follow.
555 To play back the first stream announced on the normal SAP multicast address:
561 To play back the first stream announced on one the default IPv6 SAP multicast address:
564 ffplay sap://[ff0e::2:7ffe]
569 Trasmission Control Protocol.
571 The required syntax for a TCP url is:
573 tcp://@var{hostname}:@var{port}[?@var{options}]
579 Listen for an incoming connection
581 @item timeout=@var{microseconds}
582 In read mode: if no data arrived in more than this time interval, raise error.
583 In write mode: if socket cannot be written in more than this time interval, raise error.
584 This also sets timeout on TCP connection establishing.
587 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
588 ffplay tcp://@var{hostname}:@var{port}
595 Transport Layer Security/Secure Sockets Layer
597 The required syntax for a TLS/SSL url is:
599 tls://@var{hostname}:@var{port}[?@var{options}]
605 Act as a server, listening for an incoming connection.
607 @item cafile=@var{filename}
608 Certificate authority file. The file must be in OpenSSL PEM format.
610 @item cert=@var{filename}
611 Certificate file. The file must be in OpenSSL PEM format.
613 @item key=@var{filename}
616 @item verify=@var{0|1}
617 Verify the peer's certificate.
621 Example command lines:
623 To create a TLS/SSL server that serves an input stream.
626 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
629 To play back a stream from the TLS/SSL server using @command{ffplay}:
632 ffplay tls://@var{hostname}:@var{port}
637 User Datagram Protocol.
639 The required syntax for a UDP url is:
641 udp://@var{hostname}:@var{port}[?@var{options}]
644 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
646 In case threading is enabled on the system, a circular buffer is used
647 to store the incoming data, which allows to reduce loss of data due to
648 UDP socket buffer overruns. The @var{fifo_size} and
649 @var{overrun_nonfatal} options are related to this buffer.
651 The list of supported options follows.
655 @item buffer_size=@var{size}
656 Set the UDP socket buffer size in bytes. This is used both for the
657 receiving and the sending buffer size.
659 @item localport=@var{port}
660 Override the local UDP port to bind with.
662 @item localaddr=@var{addr}
663 Choose the local IP address. This is useful e.g. if sending multicast
664 and the host has multiple interfaces, where the user can choose
665 which interface to send on by specifying the IP address of that interface.
667 @item pkt_size=@var{size}
668 Set the size in bytes of UDP packets.
670 @item reuse=@var{1|0}
671 Explicitly allow or disallow reusing UDP sockets.
674 Set the time to live value (for multicast only).
676 @item connect=@var{1|0}
677 Initialize the UDP socket with @code{connect()}. In this case, the
678 destination address can't be changed with ff_udp_set_remote_url later.
679 If the destination address isn't known at the start, this option can
680 be specified in ff_udp_set_remote_url, too.
681 This allows finding out the source address for the packets with getsockname,
682 and makes writes return with AVERROR(ECONNREFUSED) if "destination
683 unreachable" is received.
684 For receiving, this gives the benefit of only receiving packets from
685 the specified peer address/port.
687 @item sources=@var{address}[,@var{address}]
688 Only receive packets sent to the multicast group from one of the
689 specified sender IP addresses.
691 @item block=@var{address}[,@var{address}]
692 Ignore packets sent to the multicast group from the specified
695 @item fifo_size=@var{units}
696 Set the UDP receiving circular buffer size, expressed as a number of
697 packets with size of 188 bytes. If not specified defaults to 7*4096.
699 @item overrun_nonfatal=@var{1|0}
700 Survive in case of UDP receiving circular buffer overrun. Default
703 @item timeout=@var{microseconds}
704 In read mode: if no data arrived in more than this time interval, raise error.
707 Some usage examples of the UDP protocol with @command{ffmpeg} follow.
709 To stream over UDP to a remote endpoint:
711 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
714 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
716 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
719 To receive over UDP from a remote endpoint:
721 ffmpeg -i udp://[@var{multicast-address}]:@var{port}