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
171 MMS (Microsoft Media Server) protocol over TCP.
175 MMS (Microsoft Media Server) protocol over HTTP.
177 The required syntax is:
179 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
186 Computes the MD5 hash of the data to be written, and on close writes
187 this to the designated output or stdout if none is specified. It can
188 be used to test muxers without writing an actual file.
190 Some examples follow.
192 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
193 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
195 # Write the MD5 hash of the encoded AVI file to stdout.
196 ffmpeg -i input.flv -f avi -y md5:
199 Note that some formats (typically MOV) require the output protocol to
200 be seekable, so they will fail with the MD5 output protocol.
204 UNIX pipe access protocol.
206 Allow to read and write from UNIX pipes.
208 The accepted syntax is:
213 @var{number} is the number corresponding to the file descriptor of the
214 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
215 is not specified, by default the stdout file descriptor will be used
216 for writing, stdin for reading.
218 For example to read from stdin with @command{ffmpeg}:
220 cat test.wav | ffmpeg -i pipe:0
221 # ...this is the same as...
222 cat test.wav | ffmpeg -i pipe:
225 For writing to stdout with @command{ffmpeg}:
227 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
228 # ...this is the same as...
229 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
232 Note that some formats (typically MOV), require the output protocol to
233 be seekable, so they will fail with the pipe output protocol.
237 Real-Time Messaging Protocol.
239 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
240 content across a TCP/IP network.
242 The required syntax is:
244 rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
247 The accepted parameters are:
251 The address of the RTMP server.
254 The number of the TCP port to use (by default is 1935).
257 It is the name of the application to access. It usually corresponds to
258 the path where the application is installed on the RTMP server
259 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
260 the value parsed from the URI through the @code{rtmp_app} option, too.
263 It is the path or name of the resource to play with reference to the
264 application specified in @var{app}, may be prefixed by "mp4:". You
265 can override the value parsed from the URI through the @code{rtmp_playpath}
269 Act as a server, listening for an incoming connection.
272 Maximum time to wait for the incoming connection. Implies listen.
275 Additionally, the following parameters can be set via command line options
276 (or in code via @code{AVOption}s):
280 Name of application to connect on the RTMP server. This option
281 overrides the parameter specified in the URI.
284 Set the client buffer time in milliseconds. The default is 3000.
287 Extra arbitrary AMF connection parameters, parsed from a string,
288 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
289 Each value is prefixed by a single character denoting the type,
290 B for Boolean, N for number, S for string, O for object, or Z for null,
291 followed by a colon. For Booleans the data must be either 0 or 1 for
292 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
293 1 to end or begin an object, respectively. Data items in subobjects may
294 be named, by prefixing the type with 'N' and specifying the name before
295 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
296 times to construct arbitrary AMF sequences.
299 Version of the Flash plugin used to run the SWF player. The default
302 @item rtmp_flush_interval
303 Number of packets flushed in the same request (RTMPT only). The default
307 Specify that the media is a live stream. No resuming or seeking in
308 live streams is possible. The default value is @code{any}, which means the
309 subscriber first tries to play the live stream specified in the
310 playpath. If a live stream of that name is not found, it plays the
311 recorded stream. The other possible values are @code{live} and
315 URL of the web page in which the media was embedded. By default no
319 Stream identifier to play or to publish. This option overrides the
320 parameter specified in the URI.
323 Name of live stream to subscribe to. By default no value will be sent.
324 It is only sent if the option is specified or if rtmp_live
328 SHA256 hash of the decompressed SWF file (32 bytes).
331 Size of the decompressed SWF file, required for SWFVerification.
334 URL of the SWF player for the media. By default no value will be sent.
337 URL to player swf file, compute hash/size automatically.
340 URL of the target stream. Defaults to proto://host[:port]/app.
344 For example to read with @command{ffplay} a multimedia resource named
345 "sample" from the application "vod" from an RTMP server "myserver":
347 ffplay rtmp://myserver/vod/sample
352 Encrypted Real-Time Messaging Protocol.
354 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
355 streaming multimedia content within standard cryptographic primitives,
356 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
361 Real-Time Messaging Protocol over a secure SSL connection.
363 The Real-Time Messaging Protocol (RTMPS) is used for streaming
364 multimedia content across an encrypted connection.
368 Real-Time Messaging Protocol tunneled through HTTP.
370 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
371 for streaming multimedia content within HTTP requests to traverse
376 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
378 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
379 is used for streaming multimedia content within HTTP requests to traverse
384 Real-Time Messaging Protocol tunneled through HTTPS.
386 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
387 for streaming multimedia content within HTTPS requests to traverse
390 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
392 Real-Time Messaging Protocol and its variants supported through
395 Requires the presence of the librtmp headers and library during
396 configuration. You need to explicitly configure the build with
397 "--enable-librtmp". If enabled this will replace the native RTMP
400 This protocol provides most client functions and a few server
401 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
402 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
403 variants of these encrypted types (RTMPTE, RTMPTS).
405 The required syntax is:
407 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
410 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
411 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
412 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
413 meaning as specified for the RTMP native protocol.
414 @var{options} contains a list of space-separated options of the form
417 See the librtmp manual page (man 3 librtmp) for more information.
419 For example, to stream a file in real-time to an RTMP server using
422 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
425 To play the same stream using @command{ffplay}:
427 ffplay "rtmp://myserver/live/mystream live=1"
436 RTSP is not technically a protocol handler in libavformat, it is a demuxer
437 and muxer. The demuxer supports both normal RTSP (with data transferred
438 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
439 data transferred over RDT).
441 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
442 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
443 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
445 The required syntax for a RTSP url is:
447 rtsp://@var{hostname}[:@var{port}]/@var{path}
450 The following options (set on the @command{ffmpeg}/@command{ffplay} command
451 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
454 Flags for @code{rtsp_transport}:
459 Use UDP as lower transport protocol.
462 Use TCP (interleaving within the RTSP control channel) as lower
466 Use UDP multicast as lower transport protocol.
469 Use HTTP tunneling as lower transport protocol, which is useful for
473 Multiple lower transport protocols may be specified, in that case they are
474 tried one at a time (if the setup of one fails, the next one is tried).
475 For the muxer, only the @code{tcp} and @code{udp} options are supported.
477 Flags for @code{rtsp_flags}:
481 Accept packets only from negotiated peer address and port.
483 Act as a server, listening for an incoming connection.
486 When receiving data over UDP, the demuxer tries to reorder received packets
487 (since they may arrive out of order, or packets may get lost totally). This
488 can be disabled by setting the maximum demuxing delay to zero (via
489 the @code{max_delay} field of AVFormatContext).
491 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
492 streams to display can be chosen with @code{-vst} @var{n} and
493 @code{-ast} @var{n} for video and audio respectively, and can be switched
494 on the fly by pressing @code{v} and @code{a}.
496 Example command lines:
498 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
501 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
504 To watch a stream tunneled over HTTP:
507 ffplay -rtsp_transport http rtsp://server/video.mp4
510 To send a stream in realtime to a RTSP server, for others to watch:
513 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
516 To receive a stream in realtime:
519 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
524 Session Announcement Protocol (RFC 2974). This is not technically a
525 protocol handler in libavformat, it is a muxer and demuxer.
526 It is used for signalling of RTP streams, by announcing the SDP for the
527 streams regularly on a separate port.
531 The syntax for a SAP url given to the muxer is:
533 sap://@var{destination}[:@var{port}][?@var{options}]
536 The RTP packets are sent to @var{destination} on port @var{port},
537 or to port 5004 if no port is specified.
538 @var{options} is a @code{&}-separated list. The following options
543 @item announce_addr=@var{address}
544 Specify the destination IP address for sending the announcements to.
545 If omitted, the announcements are sent to the commonly used SAP
546 announcement multicast address 224.2.127.254 (sap.mcast.net), or
547 ff0e::2:7ffe if @var{destination} is an IPv6 address.
549 @item announce_port=@var{port}
550 Specify the port to send the announcements on, defaults to
551 9875 if not specified.
554 Specify the time to live value for the announcements and RTP packets,
557 @item same_port=@var{0|1}
558 If set to 1, send all RTP streams on the same port pair. If zero (the
559 default), all streams are sent on unique ports, with each stream on a
560 port 2 numbers higher than the previous.
561 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
562 The RTP stack in libavformat for receiving requires all streams to be sent
566 Example command lines follow.
568 To broadcast a stream on the local subnet, for watching in VLC:
571 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
574 Similarly, for watching in @command{ffplay}:
577 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
580 And for watching in @command{ffplay}, over IPv6:
583 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
588 The syntax for a SAP url given to the demuxer is:
590 sap://[@var{address}][:@var{port}]
593 @var{address} is the multicast address to listen for announcements on,
594 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
595 is the port that is listened on, 9875 if omitted.
597 The demuxers listens for announcements on the given address and port.
598 Once an announcement is received, it tries to receive that particular stream.
600 Example command lines follow.
602 To play back the first stream announced on the normal SAP multicast address:
608 To play back the first stream announced on one the default IPv6 SAP multicast address:
611 ffplay sap://[ff0e::2:7ffe]
616 Trasmission Control Protocol.
618 The required syntax for a TCP url is:
620 tcp://@var{hostname}:@var{port}[?@var{options}]
626 Listen for an incoming connection
628 @item timeout=@var{microseconds}
629 In read mode: if no data arrived in more than this time interval, raise error.
630 In write mode: if socket cannot be written in more than this time interval, raise error.
631 This also sets timeout on TCP connection establishing.
634 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
635 ffplay tcp://@var{hostname}:@var{port}
642 Transport Layer Security/Secure Sockets Layer
644 The required syntax for a TLS/SSL url is:
646 tls://@var{hostname}:@var{port}[?@var{options}]
652 Act as a server, listening for an incoming connection.
654 @item cafile=@var{filename}
655 Certificate authority file. The file must be in OpenSSL PEM format.
657 @item cert=@var{filename}
658 Certificate file. The file must be in OpenSSL PEM format.
660 @item key=@var{filename}
663 @item verify=@var{0|1}
664 Verify the peer's certificate.
668 Example command lines:
670 To create a TLS/SSL server that serves an input stream.
673 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
676 To play back a stream from the TLS/SSL server using @command{ffplay}:
679 ffplay tls://@var{hostname}:@var{port}
684 User Datagram Protocol.
686 The required syntax for a UDP url is:
688 udp://@var{hostname}:@var{port}[?@var{options}]
691 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
693 In case threading is enabled on the system, a circular buffer is used
694 to store the incoming data, which allows to reduce loss of data due to
695 UDP socket buffer overruns. The @var{fifo_size} and
696 @var{overrun_nonfatal} options are related to this buffer.
698 The list of supported options follows.
702 @item buffer_size=@var{size}
703 Set the UDP socket buffer size in bytes. This is used both for the
704 receiving and the sending buffer size.
706 @item localport=@var{port}
707 Override the local UDP port to bind with.
709 @item localaddr=@var{addr}
710 Choose the local IP address. This is useful e.g. if sending multicast
711 and the host has multiple interfaces, where the user can choose
712 which interface to send on by specifying the IP address of that interface.
714 @item pkt_size=@var{size}
715 Set the size in bytes of UDP packets.
717 @item reuse=@var{1|0}
718 Explicitly allow or disallow reusing UDP sockets.
721 Set the time to live value (for multicast only).
723 @item connect=@var{1|0}
724 Initialize the UDP socket with @code{connect()}. In this case, the
725 destination address can't be changed with ff_udp_set_remote_url later.
726 If the destination address isn't known at the start, this option can
727 be specified in ff_udp_set_remote_url, too.
728 This allows finding out the source address for the packets with getsockname,
729 and makes writes return with AVERROR(ECONNREFUSED) if "destination
730 unreachable" is received.
731 For receiving, this gives the benefit of only receiving packets from
732 the specified peer address/port.
734 @item sources=@var{address}[,@var{address}]
735 Only receive packets sent to the multicast group from one of the
736 specified sender IP addresses.
738 @item block=@var{address}[,@var{address}]
739 Ignore packets sent to the multicast group from the specified
742 @item fifo_size=@var{units}
743 Set the UDP receiving circular buffer size, expressed as a number of
744 packets with size of 188 bytes. If not specified defaults to 7*4096.
746 @item overrun_nonfatal=@var{1|0}
747 Survive in case of UDP receiving circular buffer overrun. Default
750 @item timeout=@var{microseconds}
751 In read mode: if no data arrived in more than this time interval, raise error.
754 Some usage examples of the UDP protocol with @command{ffmpeg} follow.
756 To stream over UDP to a remote endpoint:
758 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
761 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
763 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
766 To receive over UDP from a remote endpoint:
768 ffmpeg -i udp://[@var{multicast-address}]:@var{port}