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:
99 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
100 supports this, the metadata has to be retrieved by the application by reading
101 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
104 @item icy_metadata_headers
105 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
106 headers, separated by newline characters.
108 @item icy_metadata_packet
109 If the server supports ICY metadata, and @option{icy} was set to 1, this
110 contains the last non-empty metadata packet sent by the server. It should be
111 polled in regular intervals by applications interested in mid-stream metadata
117 MMS (Microsoft Media Server) protocol over TCP.
121 MMS (Microsoft Media Server) protocol over HTTP.
123 The required syntax is:
125 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
132 Computes the MD5 hash of the data to be written, and on close writes
133 this to the designated output or stdout if none is specified. It can
134 be used to test muxers without writing an actual file.
136 Some examples follow.
138 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
139 avconv -i input.flv -f avi -y md5:output.avi.md5
141 # Write the MD5 hash of the encoded AVI file to stdout.
142 avconv -i input.flv -f avi -y md5:
145 Note that some formats (typically MOV) require the output protocol to
146 be seekable, so they will fail with the MD5 output protocol.
150 UNIX pipe access protocol.
152 Allow to read and write from UNIX pipes.
154 The accepted syntax is:
159 @var{number} is the number corresponding to the file descriptor of the
160 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
161 is not specified, by default the stdout file descriptor will be used
162 for writing, stdin for reading.
164 For example to read from stdin with @command{avconv}:
166 cat test.wav | avconv -i pipe:0
167 # ...this is the same as...
168 cat test.wav | avconv -i pipe:
171 For writing to stdout with @command{avconv}:
173 avconv -i test.wav -f avi pipe:1 | cat > test.avi
174 # ...this is the same as...
175 avconv -i test.wav -f avi pipe: | cat > test.avi
178 Note that some formats (typically MOV), require the output protocol to
179 be seekable, so they will fail with the pipe output protocol.
183 Real-Time Messaging Protocol.
185 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
186 content across a TCP/IP network.
188 The required syntax is:
190 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
193 The accepted parameters are:
197 An optional username (mostly for publishing).
200 An optional password (mostly for publishing).
203 The address of the RTMP server.
206 The number of the TCP port to use (by default is 1935).
209 It is the name of the application to access. It usually corresponds to
210 the path where the application is installed on the RTMP server
211 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
212 the value parsed from the URI through the @code{rtmp_app} option, too.
215 It is the path or name of the resource to play with reference to the
216 application specified in @var{app}, may be prefixed by "mp4:". You
217 can override the value parsed from the URI through the @code{rtmp_playpath}
221 Act as a server, listening for an incoming connection.
224 Maximum time to wait for the incoming connection. Implies listen.
227 Additionally, the following parameters can be set via command line options
228 (or in code via @code{AVOption}s):
232 Name of application to connect on the RTMP server. This option
233 overrides the parameter specified in the URI.
236 Set the client buffer time in milliseconds. The default is 3000.
239 Extra arbitrary AMF connection parameters, parsed from a string,
240 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
241 Each value is prefixed by a single character denoting the type,
242 B for Boolean, N for number, S for string, O for object, or Z for null,
243 followed by a colon. For Booleans the data must be either 0 or 1 for
244 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
245 1 to end or begin an object, respectively. Data items in subobjects may
246 be named, by prefixing the type with 'N' and specifying the name before
247 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
248 times to construct arbitrary AMF sequences.
251 Version of the Flash plugin used to run the SWF player. The default
252 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
253 <libavformat version>).)
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{avplay} a multimedia resource named
298 "sample" from the application "vod" from an RTMP server "myserver":
300 avplay rtmp://myserver/vod/sample
303 To publish to a password protected server, passing the playpath and
304 app names separately:
306 avconv -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
311 Encrypted Real-Time Messaging Protocol.
313 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
314 streaming multimedia content within standard cryptographic primitives,
315 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
320 Real-Time Messaging Protocol over a secure SSL connection.
322 The Real-Time Messaging Protocol (RTMPS) is used for streaming
323 multimedia content across an encrypted connection.
327 Real-Time Messaging Protocol tunneled through HTTP.
329 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
330 for streaming multimedia content within HTTP requests to traverse
335 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
337 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
338 is used for streaming multimedia content within HTTP requests to traverse
343 Real-Time Messaging Protocol tunneled through HTTPS.
345 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
346 for streaming multimedia content within HTTPS requests to traverse
349 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
351 Real-Time Messaging Protocol and its variants supported through
354 Requires the presence of the librtmp headers and library during
355 configuration. You need to explicitly configure the build with
356 "--enable-librtmp". If enabled this will replace the native RTMP
359 This protocol provides most client functions and a few server
360 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
361 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
362 variants of these encrypted types (RTMPTE, RTMPTS).
364 The required syntax is:
366 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
369 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
370 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
371 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
372 meaning as specified for the RTMP native protocol.
373 @var{options} contains a list of space-separated options of the form
376 See the librtmp manual page (man 3 librtmp) for more information.
378 For example, to stream a file in real-time to an RTMP server using
381 avconv -re -i myfile -f flv rtmp://myserver/live/mystream
384 To play the same stream using @command{avplay}:
386 avplay "rtmp://myserver/live/mystream live=1"
395 RTSP is not technically a protocol handler in libavformat, it is a demuxer
396 and muxer. The demuxer supports both normal RTSP (with data transferred
397 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
398 data transferred over RDT).
400 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
401 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
402 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
404 The required syntax for a RTSP url is:
406 rtsp://@var{hostname}[:@var{port}]/@var{path}
409 The following options (set on the @command{avconv}/@command{avplay} command
410 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
413 Flags for @code{rtsp_transport}:
418 Use UDP as lower transport protocol.
421 Use TCP (interleaving within the RTSP control channel) as lower
425 Use UDP multicast as lower transport protocol.
428 Use HTTP tunneling as lower transport protocol, which is useful for
432 Multiple lower transport protocols may be specified, in that case they are
433 tried one at a time (if the setup of one fails, the next one is tried).
434 For the muxer, only the @code{tcp} and @code{udp} options are supported.
436 Flags for @code{rtsp_flags}:
440 Accept packets only from negotiated peer address and port.
442 Act as a server, listening for an incoming connection.
445 When receiving data over UDP, the demuxer tries to reorder received packets
446 (since they may arrive out of order, or packets may get lost totally). This
447 can be disabled by setting the maximum demuxing delay to zero (via
448 the @code{max_delay} field of AVFormatContext).
450 When watching multi-bitrate Real-RTSP streams with @command{avplay}, the
451 streams to display can be chosen with @code{-vst} @var{n} and
452 @code{-ast} @var{n} for video and audio respectively, and can be switched
453 on the fly by pressing @code{v} and @code{a}.
455 Example command lines:
457 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
460 avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
463 To watch a stream tunneled over HTTP:
466 avplay -rtsp_transport http rtsp://server/video.mp4
469 To send a stream in realtime to a RTSP server, for others to watch:
472 avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
475 To receive a stream in realtime:
478 avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
483 Session Announcement Protocol (RFC 2974). This is not technically a
484 protocol handler in libavformat, it is a muxer and demuxer.
485 It is used for signalling of RTP streams, by announcing the SDP for the
486 streams regularly on a separate port.
490 The syntax for a SAP url given to the muxer is:
492 sap://@var{destination}[:@var{port}][?@var{options}]
495 The RTP packets are sent to @var{destination} on port @var{port},
496 or to port 5004 if no port is specified.
497 @var{options} is a @code{&}-separated list. The following options
502 @item announce_addr=@var{address}
503 Specify the destination IP address for sending the announcements to.
504 If omitted, the announcements are sent to the commonly used SAP
505 announcement multicast address 224.2.127.254 (sap.mcast.net), or
506 ff0e::2:7ffe if @var{destination} is an IPv6 address.
508 @item announce_port=@var{port}
509 Specify the port to send the announcements on, defaults to
510 9875 if not specified.
513 Specify the time to live value for the announcements and RTP packets,
516 @item same_port=@var{0|1}
517 If set to 1, send all RTP streams on the same port pair. If zero (the
518 default), all streams are sent on unique ports, with each stream on a
519 port 2 numbers higher than the previous.
520 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
521 The RTP stack in libavformat for receiving requires all streams to be sent
525 Example command lines follow.
527 To broadcast a stream on the local subnet, for watching in VLC:
530 avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
533 Similarly, for watching in avplay:
536 avconv -re -i @var{input} -f sap sap://224.0.0.255
539 And for watching in avplay, over IPv6:
542 avconv -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
547 The syntax for a SAP url given to the demuxer is:
549 sap://[@var{address}][:@var{port}]
552 @var{address} is the multicast address to listen for announcements on,
553 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
554 is the port that is listened on, 9875 if omitted.
556 The demuxers listens for announcements on the given address and port.
557 Once an announcement is received, it tries to receive that particular stream.
559 Example command lines follow.
561 To play back the first stream announced on the normal SAP multicast address:
567 To play back the first stream announced on one the default IPv6 SAP multicast address:
570 avplay sap://[ff0e::2:7ffe]
575 Trasmission Control Protocol.
577 The required syntax for a TCP url is:
579 tcp://@var{hostname}:@var{port}[?@var{options}]
585 Listen for an incoming connection
588 avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
589 avplay tcp://@var{hostname}:@var{port}
596 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
598 The required syntax for a TLS url is:
600 tls://@var{hostname}:@var{port}
603 The following parameters can be set via command line options
604 (or in code via @code{AVOption}s):
609 A file containing certificate authority (CA) root certificates to treat
610 as trusted. If the linked TLS library contains a default this might not
611 need to be specified for verification to work, but not all libraries and
612 setups have defaults built in.
614 @item tls_verify=@var{1|0}
615 If enabled, try to verify the peer that we are communicating with.
616 Note, if using OpenSSL, this currently only makes sure that the
617 peer certificate is signed by one of the root certificates in the CA
618 database, but it does not validate that the certificate actually
619 matches the host name we are trying to connect to. (With GnuTLS,
620 the host name is validated as well.)
622 This is disabled by default since it requires a CA database to be
623 provided by the caller in many cases.
626 A file containing a certificate to use in the handshake with the peer.
627 (When operating as server, in listen mode, this is more often required
628 by the peer, while client certificates only are mandated in certain
632 A file containing the private key for the certificate.
634 @item listen=@var{1|0}
635 If enabled, listen for connections on the provided port, and assume
636 the server role in the handshake instead of the client role.
642 User Datagram Protocol.
644 The required syntax for a UDP url is:
646 udp://@var{hostname}:@var{port}[?@var{options}]
649 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
650 Follow the list of supported options.
654 @item buffer_size=@var{size}
655 set the UDP buffer size in bytes
657 @item localport=@var{port}
658 override the local UDP port to bind with
660 @item localaddr=@var{addr}
661 Choose the local IP address. This is useful e.g. if sending multicast
662 and the host has multiple interfaces, where the user can choose
663 which interface to send on by specifying the IP address of that interface.
665 @item pkt_size=@var{size}
666 set the size in bytes of UDP packets
668 @item reuse=@var{1|0}
669 explicitly allow or disallow reusing UDP sockets
672 set the time to live value (for multicast only)
674 @item connect=@var{1|0}
675 Initialize the UDP socket with @code{connect()}. In this case, the
676 destination address can't be changed with ff_udp_set_remote_url later.
677 If the destination address isn't known at the start, this option can
678 be specified in ff_udp_set_remote_url, too.
679 This allows finding out the source address for the packets with getsockname,
680 and makes writes return with AVERROR(ECONNREFUSED) if "destination
681 unreachable" is received.
682 For receiving, this gives the benefit of only receiving packets from
683 the specified peer address/port.
685 @item sources=@var{address}[,@var{address}]
686 Only receive packets sent to the multicast group from one of the
687 specified sender IP addresses.
689 @item block=@var{address}[,@var{address}]
690 Ignore packets sent to the multicast group from the specified
694 Some usage examples of the udp protocol with @command{avconv} follow.
696 To stream over UDP to a remote endpoint:
698 avconv -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
701 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
703 avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
706 To receive over UDP from a remote endpoint:
708 avconv -i udp://[@var{multicast-address}]:@var{port}
715 The required syntax for a Unix socket URL is:
718 unix://@var{filepath}
721 The following parameters can be set via command line options
722 (or in code via @code{AVOption}s):
728 Create the Unix socket in listening mode.