4 Protocols are configured elements in Libav which allow to access
5 resources which require the use of a particular protocol.
7 When you configure your Libav 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 av* tools will display the list of
20 A description of the currently available protocols follows.
24 Physical concatenation protocol.
26 Allow to read and seek from many resource in sequence as if they were
29 A URL accepted by this protocol has the syntax:
31 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
34 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
35 resource to be concatenated, each one possibly specifying a distinct
38 For example to read a sequence of files @file{split1.mpeg},
39 @file{split2.mpeg}, @file{split3.mpeg} with @command{avplay} use the
42 avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
45 Note that you may need to escape the character "|" which is special for
52 Allow to read from or read to a file.
54 For example to read from a file @file{input.mpeg} with @command{avconv}
57 avconv -i file:input.mpeg output.mpeg
60 The av* tools default to the file protocol, that is a resource
61 specified with the name "FILE.mpeg" is interpreted as the URL
70 Read Apple HTTP Live Streaming compliant segmented stream as
71 a uniform one. The M3U8 playlists describing the segments can be
72 remote HTTP resources or local files, accessed using the standard
74 The nested protocol is declared by specifying
75 "+@var{proto}" after the hls URI scheme name, where @var{proto}
76 is either "file" or "http".
79 hls+http://host/path/to/remote/resource.m3u8
80 hls+file://path/to/local/resource.m3u8
83 Using this protocol is discouraged - the hls demuxer should work
84 just as well (if not, please report the issues) and is more complete.
85 To use the hls demuxer instead, simply use the direct URLs to the
90 HTTP (Hyper Text Transfer Protocol).
92 This protocol accepts the following options:
96 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
99 Set a specific content type for the POST messages.
102 Set custom HTTP headers, can override built in default headers. The
103 value must be a string encoding the headers.
105 @item multiple_requests
106 Use persistent connections if set to 1, default is 0.
109 Set custom HTTP post data.
112 Override the User-Agent header. If not specified a string of the form
113 "Lavf/<version>" will be used.
116 Export the MIME type.
119 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
120 supports this, the metadata has to be retrieved by the application by reading
121 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
124 @item icy_metadata_headers
125 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
126 headers, separated by newline characters.
128 @item icy_metadata_packet
129 If the server supports ICY metadata, and @option{icy} was set to 1, this
130 contains the last non-empty metadata packet sent by the server. It should be
131 polled in regular intervals by applications interested in mid-stream metadata
135 Set initial byte offset.
138 Try to limit the request to bytes preceding this offset.
143 Icecast (stream to Icecast servers)
145 This protocol accepts the following options:
149 Set the stream genre.
154 @item ice_description
155 Set the stream description.
158 Set the stream website URL.
161 Set if the stream should be public or not.
162 The default is 0 (not public).
165 Override the User-Agent header. If not specified a string of the form
166 "Lavf/<version>" will be used.
169 Set the Icecast mountpoint password.
172 Set the stream content type. This must be set if it is different from
176 This enables support for Icecast versions < 2.4.0, that do not support the
177 HTTP PUT method but the SOURCE method.
183 MMS (Microsoft Media Server) protocol over TCP.
187 MMS (Microsoft Media Server) protocol over HTTP.
189 The required syntax is:
191 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
198 Computes the MD5 hash of the data to be written, and on close writes
199 this to the designated output or stdout if none is specified. It can
200 be used to test muxers without writing an actual file.
202 Some examples follow.
204 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
205 avconv -i input.flv -f avi -y md5:output.avi.md5
207 # Write the MD5 hash of the encoded AVI file to stdout.
208 avconv -i input.flv -f avi -y md5:
211 Note that some formats (typically MOV) require the output protocol to
212 be seekable, so they will fail with the MD5 output protocol.
216 UNIX pipe access protocol.
218 Allow to read and write from UNIX pipes.
220 The accepted syntax is:
225 @var{number} is the number corresponding to the file descriptor of the
226 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
227 is not specified, by default the stdout file descriptor will be used
228 for writing, stdin for reading.
230 For example to read from stdin with @command{avconv}:
232 cat test.wav | avconv -i pipe:0
233 # ...this is the same as...
234 cat test.wav | avconv -i pipe:
237 For writing to stdout with @command{avconv}:
239 avconv -i test.wav -f avi pipe:1 | cat > test.avi
240 # ...this is the same as...
241 avconv -i test.wav -f avi pipe: | cat > test.avi
244 Note that some formats (typically MOV), require the output protocol to
245 be seekable, so they will fail with the pipe output protocol.
249 Real-Time Messaging Protocol.
251 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
252 content across a TCP/IP network.
254 The required syntax is:
256 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
259 The accepted parameters are:
263 An optional username (mostly for publishing).
266 An optional password (mostly for publishing).
269 The address of the RTMP server.
272 The number of the TCP port to use (by default is 1935).
275 It is the name of the application to access. It usually corresponds to
276 the path where the application is installed on the RTMP server
277 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
278 the value parsed from the URI through the @code{rtmp_app} option, too.
281 It is the path or name of the resource to play with reference to the
282 application specified in @var{app}, may be prefixed by "mp4:". You
283 can override the value parsed from the URI through the @code{rtmp_playpath}
287 Act as a server, listening for an incoming connection.
290 Maximum time to wait for the incoming connection. Implies listen.
293 Additionally, the following parameters can be set via command line options
294 (or in code via @code{AVOption}s):
298 Name of application to connect on the RTMP server. This option
299 overrides the parameter specified in the URI.
302 Set the client buffer time in milliseconds. The default is 3000.
305 Extra arbitrary AMF connection parameters, parsed from a string,
306 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
307 Each value is prefixed by a single character denoting the type,
308 B for Boolean, N for number, S for string, O for object, or Z for null,
309 followed by a colon. For Booleans the data must be either 0 or 1 for
310 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
311 1 to end or begin an object, respectively. Data items in subobjects may
312 be named, by prefixing the type with 'N' and specifying the name before
313 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
314 times to construct arbitrary AMF sequences.
317 Version of the Flash plugin used to run the SWF player. The default
318 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
319 <libavformat version>).)
321 @item rtmp_flush_interval
322 Number of packets flushed in the same request (RTMPT only). The default
326 Specify that the media is a live stream. No resuming or seeking in
327 live streams is possible. The default value is @code{any}, which means the
328 subscriber first tries to play the live stream specified in the
329 playpath. If a live stream of that name is not found, it plays the
330 recorded stream. The other possible values are @code{live} and
334 URL of the web page in which the media was embedded. By default no
338 Stream identifier to play or to publish. This option overrides the
339 parameter specified in the URI.
342 Name of live stream to subscribe to. By default no value will be sent.
343 It is only sent if the option is specified or if rtmp_live
347 SHA256 hash of the decompressed SWF file (32 bytes).
350 Size of the decompressed SWF file, required for SWFVerification.
353 URL of the SWF player for the media. By default no value will be sent.
356 URL to player swf file, compute hash/size automatically.
359 URL of the target stream. Defaults to proto://host[:port]/app.
363 For example to read with @command{avplay} a multimedia resource named
364 "sample" from the application "vod" from an RTMP server "myserver":
366 avplay rtmp://myserver/vod/sample
369 To publish to a password protected server, passing the playpath and
370 app names separately:
372 avconv -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
377 Encrypted Real-Time Messaging Protocol.
379 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
380 streaming multimedia content within standard cryptographic primitives,
381 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
386 Real-Time Messaging Protocol over a secure SSL connection.
388 The Real-Time Messaging Protocol (RTMPS) is used for streaming
389 multimedia content across an encrypted connection.
393 Real-Time Messaging Protocol tunneled through HTTP.
395 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
396 for streaming multimedia content within HTTP requests to traverse
401 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
403 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
404 is used for streaming multimedia content within HTTP requests to traverse
409 Real-Time Messaging Protocol tunneled through HTTPS.
411 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
412 for streaming multimedia content within HTTPS requests to traverse
415 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
417 Real-Time Messaging Protocol and its variants supported through
420 Requires the presence of the librtmp headers and library during
421 configuration. You need to explicitly configure the build with
422 "--enable-librtmp". If enabled this will replace the native RTMP
425 This protocol provides most client functions and a few server
426 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
427 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
428 variants of these encrypted types (RTMPTE, RTMPTS).
430 The required syntax is:
432 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
435 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
436 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
437 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
438 meaning as specified for the RTMP native protocol.
439 @var{options} contains a list of space-separated options of the form
442 See the librtmp manual page (man 3 librtmp) for more information.
444 For example, to stream a file in real-time to an RTMP server using
447 avconv -re -i myfile -f flv rtmp://myserver/live/mystream
450 To play the same stream using @command{avplay}:
452 avplay "rtmp://myserver/live/mystream live=1"
461 RTSP is not technically a protocol handler in libavformat, it is a demuxer
462 and muxer. The demuxer supports both normal RTSP (with data transferred
463 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
464 data transferred over RDT).
466 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
467 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
468 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
470 The required syntax for a RTSP url is:
472 rtsp://@var{hostname}[:@var{port}]/@var{path}
475 The following options (set on the @command{avconv}/@command{avplay} command
476 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
479 Flags for @code{rtsp_transport}:
484 Use UDP as lower transport protocol.
487 Use TCP (interleaving within the RTSP control channel) as lower
491 Use UDP multicast as lower transport protocol.
494 Use HTTP tunneling as lower transport protocol, which is useful for
498 Multiple lower transport protocols may be specified, in that case they are
499 tried one at a time (if the setup of one fails, the next one is tried).
500 For the muxer, only the @code{tcp} and @code{udp} options are supported.
502 Flags for @code{rtsp_flags}:
506 Accept packets only from negotiated peer address and port.
508 Act as a server, listening for an incoming connection.
511 When receiving data over UDP, the demuxer tries to reorder received packets
512 (since they may arrive out of order, or packets may get lost totally). This
513 can be disabled by setting the maximum demuxing delay to zero (via
514 the @code{max_delay} field of AVFormatContext).
516 When watching multi-bitrate Real-RTSP streams with @command{avplay}, the
517 streams to display can be chosen with @code{-vst} @var{n} and
518 @code{-ast} @var{n} for video and audio respectively, and can be switched
519 on the fly by pressing @code{v} and @code{a}.
521 Example command lines:
523 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
526 avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
529 To watch a stream tunneled over HTTP:
532 avplay -rtsp_transport http rtsp://server/video.mp4
535 To send a stream in realtime to a RTSP server, for others to watch:
538 avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
541 To receive a stream in realtime:
544 avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
549 Session Announcement Protocol (RFC 2974). This is not technically a
550 protocol handler in libavformat, it is a muxer and demuxer.
551 It is used for signalling of RTP streams, by announcing the SDP for the
552 streams regularly on a separate port.
556 The syntax for a SAP url given to the muxer is:
558 sap://@var{destination}[:@var{port}][?@var{options}]
561 The RTP packets are sent to @var{destination} on port @var{port},
562 or to port 5004 if no port is specified.
563 @var{options} is a @code{&}-separated list. The following options
568 @item announce_addr=@var{address}
569 Specify the destination IP address for sending the announcements to.
570 If omitted, the announcements are sent to the commonly used SAP
571 announcement multicast address 224.2.127.254 (sap.mcast.net), or
572 ff0e::2:7ffe if @var{destination} is an IPv6 address.
574 @item announce_port=@var{port}
575 Specify the port to send the announcements on, defaults to
576 9875 if not specified.
579 Specify the time to live value for the announcements and RTP packets,
582 @item same_port=@var{0|1}
583 If set to 1, send all RTP streams on the same port pair. If zero (the
584 default), all streams are sent on unique ports, with each stream on a
585 port 2 numbers higher than the previous.
586 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
587 The RTP stack in libavformat for receiving requires all streams to be sent
591 Example command lines follow.
593 To broadcast a stream on the local subnet, for watching in VLC:
596 avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
599 Similarly, for watching in avplay:
602 avconv -re -i @var{input} -f sap sap://224.0.0.255
605 And for watching in avplay, over IPv6:
608 avconv -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
613 The syntax for a SAP url given to the demuxer is:
615 sap://[@var{address}][:@var{port}]
618 @var{address} is the multicast address to listen for announcements on,
619 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
620 is the port that is listened on, 9875 if omitted.
622 The demuxers listens for announcements on the given address and port.
623 Once an announcement is received, it tries to receive that particular stream.
625 Example command lines follow.
627 To play back the first stream announced on the normal SAP multicast address:
633 To play back the first stream announced on one the default IPv6 SAP multicast address:
636 avplay sap://[ff0e::2:7ffe]
641 Trasmission Control Protocol.
643 The required syntax for a TCP url is:
645 tcp://@var{hostname}:@var{port}[?@var{options}]
651 Listen for an incoming connection
654 avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
655 avplay tcp://@var{hostname}:@var{port}
662 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
664 The required syntax for a TLS url is:
666 tls://@var{hostname}:@var{port}
669 The following parameters can be set via command line options
670 (or in code via @code{AVOption}s):
675 A file containing certificate authority (CA) root certificates to treat
676 as trusted. If the linked TLS library contains a default this might not
677 need to be specified for verification to work, but not all libraries and
678 setups have defaults built in.
680 @item tls_verify=@var{1|0}
681 If enabled, try to verify the peer that we are communicating with.
682 Note, if using OpenSSL, this currently only makes sure that the
683 peer certificate is signed by one of the root certificates in the CA
684 database, but it does not validate that the certificate actually
685 matches the host name we are trying to connect to. (With GnuTLS,
686 the host name is validated as well.)
688 This is disabled by default since it requires a CA database to be
689 provided by the caller in many cases.
692 A file containing a certificate to use in the handshake with the peer.
693 (When operating as server, in listen mode, this is more often required
694 by the peer, while client certificates only are mandated in certain
698 A file containing the private key for the certificate.
700 @item listen=@var{1|0}
701 If enabled, listen for connections on the provided port, and assume
702 the server role in the handshake instead of the client role.
708 User Datagram Protocol.
710 The required syntax for a UDP url is:
712 udp://@var{hostname}:@var{port}[?@var{options}]
715 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
716 Follow the list of supported options.
720 @item buffer_size=@var{size}
721 set the UDP buffer size in bytes
723 @item localport=@var{port}
724 override the local UDP port to bind with
726 @item localaddr=@var{addr}
727 Choose the local IP address. This is useful e.g. if sending multicast
728 and the host has multiple interfaces, where the user can choose
729 which interface to send on by specifying the IP address of that interface.
731 @item pkt_size=@var{size}
732 set the size in bytes of UDP packets
734 @item reuse=@var{1|0}
735 explicitly allow or disallow reusing UDP sockets
738 set the time to live value (for multicast only)
740 @item connect=@var{1|0}
741 Initialize the UDP socket with @code{connect()}. In this case, the
742 destination address can't be changed with ff_udp_set_remote_url later.
743 If the destination address isn't known at the start, this option can
744 be specified in ff_udp_set_remote_url, too.
745 This allows finding out the source address for the packets with getsockname,
746 and makes writes return with AVERROR(ECONNREFUSED) if "destination
747 unreachable" is received.
748 For receiving, this gives the benefit of only receiving packets from
749 the specified peer address/port.
751 @item sources=@var{address}[,@var{address}]
752 Only receive packets sent to the multicast group from one of the
753 specified sender IP addresses.
755 @item block=@var{address}[,@var{address}]
756 Ignore packets sent to the multicast group from the specified
760 Some usage examples of the udp protocol with @command{avconv} follow.
762 To stream over UDP to a remote endpoint:
764 avconv -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
767 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
769 avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
772 To receive over UDP from a remote endpoint:
774 avconv -i udp://[@var{multicast-address}]:@var{port}
781 The required syntax for a Unix socket URL is:
784 unix://@var{filepath}
787 The following parameters can be set via command line options
788 (or in code via @code{AVOption}s):
794 Create the Unix socket in listening mode.