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 MMS (Microsoft Media Server) protocol over TCP.
147 MMS (Microsoft Media Server) protocol over HTTP.
149 The required syntax is:
151 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
158 Computes the MD5 hash of the data to be written, and on close writes
159 this to the designated output or stdout if none is specified. It can
160 be used to test muxers without writing an actual file.
162 Some examples follow.
164 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
165 avconv -i input.flv -f avi -y md5:output.avi.md5
167 # Write the MD5 hash of the encoded AVI file to stdout.
168 avconv -i input.flv -f avi -y md5:
171 Note that some formats (typically MOV) require the output protocol to
172 be seekable, so they will fail with the MD5 output protocol.
176 UNIX pipe access protocol.
178 Allow to read and write from UNIX pipes.
180 The accepted syntax is:
185 @var{number} is the number corresponding to the file descriptor of the
186 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
187 is not specified, by default the stdout file descriptor will be used
188 for writing, stdin for reading.
190 For example to read from stdin with @command{avconv}:
192 cat test.wav | avconv -i pipe:0
193 # ...this is the same as...
194 cat test.wav | avconv -i pipe:
197 For writing to stdout with @command{avconv}:
199 avconv -i test.wav -f avi pipe:1 | cat > test.avi
200 # ...this is the same as...
201 avconv -i test.wav -f avi pipe: | cat > test.avi
204 Note that some formats (typically MOV), require the output protocol to
205 be seekable, so they will fail with the pipe output protocol.
209 Real-Time Messaging Protocol.
211 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
212 content across a TCP/IP network.
214 The required syntax is:
216 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
219 The accepted parameters are:
223 An optional username (mostly for publishing).
226 An optional password (mostly for publishing).
229 The address of the RTMP server.
232 The number of the TCP port to use (by default is 1935).
235 It is the name of the application to access. It usually corresponds to
236 the path where the application is installed on the RTMP server
237 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
238 the value parsed from the URI through the @code{rtmp_app} option, too.
241 It is the path or name of the resource to play with reference to the
242 application specified in @var{app}, may be prefixed by "mp4:". You
243 can override the value parsed from the URI through the @code{rtmp_playpath}
247 Act as a server, listening for an incoming connection.
250 Maximum time to wait for the incoming connection. Implies listen.
253 Additionally, the following parameters can be set via command line options
254 (or in code via @code{AVOption}s):
258 Name of application to connect on the RTMP server. This option
259 overrides the parameter specified in the URI.
262 Set the client buffer time in milliseconds. The default is 3000.
265 Extra arbitrary AMF connection parameters, parsed from a string,
266 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
267 Each value is prefixed by a single character denoting the type,
268 B for Boolean, N for number, S for string, O for object, or Z for null,
269 followed by a colon. For Booleans the data must be either 0 or 1 for
270 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
271 1 to end or begin an object, respectively. Data items in subobjects may
272 be named, by prefixing the type with 'N' and specifying the name before
273 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
274 times to construct arbitrary AMF sequences.
277 Version of the Flash plugin used to run the SWF player. The default
278 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
279 <libavformat version>).)
281 @item rtmp_flush_interval
282 Number of packets flushed in the same request (RTMPT only). The default
286 Specify that the media is a live stream. No resuming or seeking in
287 live streams is possible. The default value is @code{any}, which means the
288 subscriber first tries to play the live stream specified in the
289 playpath. If a live stream of that name is not found, it plays the
290 recorded stream. The other possible values are @code{live} and
294 URL of the web page in which the media was embedded. By default no
298 Stream identifier to play or to publish. This option overrides the
299 parameter specified in the URI.
302 Name of live stream to subscribe to. By default no value will be sent.
303 It is only sent if the option is specified or if rtmp_live
307 SHA256 hash of the decompressed SWF file (32 bytes).
310 Size of the decompressed SWF file, required for SWFVerification.
313 URL of the SWF player for the media. By default no value will be sent.
316 URL to player swf file, compute hash/size automatically.
319 URL of the target stream. Defaults to proto://host[:port]/app.
323 For example to read with @command{avplay} a multimedia resource named
324 "sample" from the application "vod" from an RTMP server "myserver":
326 avplay rtmp://myserver/vod/sample
329 To publish to a password protected server, passing the playpath and
330 app names separately:
332 avconv -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
337 Encrypted Real-Time Messaging Protocol.
339 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
340 streaming multimedia content within standard cryptographic primitives,
341 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
346 Real-Time Messaging Protocol over a secure SSL connection.
348 The Real-Time Messaging Protocol (RTMPS) is used for streaming
349 multimedia content across an encrypted connection.
353 Real-Time Messaging Protocol tunneled through HTTP.
355 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
356 for streaming multimedia content within HTTP requests to traverse
361 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
363 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
364 is used for streaming multimedia content within HTTP requests to traverse
369 Real-Time Messaging Protocol tunneled through HTTPS.
371 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
372 for streaming multimedia content within HTTPS requests to traverse
375 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
377 Real-Time Messaging Protocol and its variants supported through
380 Requires the presence of the librtmp headers and library during
381 configuration. You need to explicitly configure the build with
382 "--enable-librtmp". If enabled this will replace the native RTMP
385 This protocol provides most client functions and a few server
386 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
387 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
388 variants of these encrypted types (RTMPTE, RTMPTS).
390 The required syntax is:
392 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
395 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
396 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
397 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
398 meaning as specified for the RTMP native protocol.
399 @var{options} contains a list of space-separated options of the form
402 See the librtmp manual page (man 3 librtmp) for more information.
404 For example, to stream a file in real-time to an RTMP server using
407 avconv -re -i myfile -f flv rtmp://myserver/live/mystream
410 To play the same stream using @command{avplay}:
412 avplay "rtmp://myserver/live/mystream live=1"
421 RTSP is not technically a protocol handler in libavformat, it is a demuxer
422 and muxer. The demuxer supports both normal RTSP (with data transferred
423 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
424 data transferred over RDT).
426 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
427 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
428 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
430 The required syntax for a RTSP url is:
432 rtsp://@var{hostname}[:@var{port}]/@var{path}
435 The following options (set on the @command{avconv}/@command{avplay} command
436 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
439 Flags for @code{rtsp_transport}:
444 Use UDP as lower transport protocol.
447 Use TCP (interleaving within the RTSP control channel) as lower
451 Use UDP multicast as lower transport protocol.
454 Use HTTP tunneling as lower transport protocol, which is useful for
458 Multiple lower transport protocols may be specified, in that case they are
459 tried one at a time (if the setup of one fails, the next one is tried).
460 For the muxer, only the @code{tcp} and @code{udp} options are supported.
462 Flags for @code{rtsp_flags}:
466 Accept packets only from negotiated peer address and port.
468 Act as a server, listening for an incoming connection.
471 When receiving data over UDP, the demuxer tries to reorder received packets
472 (since they may arrive out of order, or packets may get lost totally). This
473 can be disabled by setting the maximum demuxing delay to zero (via
474 the @code{max_delay} field of AVFormatContext).
476 When watching multi-bitrate Real-RTSP streams with @command{avplay}, the
477 streams to display can be chosen with @code{-vst} @var{n} and
478 @code{-ast} @var{n} for video and audio respectively, and can be switched
479 on the fly by pressing @code{v} and @code{a}.
481 Example command lines:
483 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
486 avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
489 To watch a stream tunneled over HTTP:
492 avplay -rtsp_transport http rtsp://server/video.mp4
495 To send a stream in realtime to a RTSP server, for others to watch:
498 avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
501 To receive a stream in realtime:
504 avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
509 Session Announcement Protocol (RFC 2974). This is not technically a
510 protocol handler in libavformat, it is a muxer and demuxer.
511 It is used for signalling of RTP streams, by announcing the SDP for the
512 streams regularly on a separate port.
516 The syntax for a SAP url given to the muxer is:
518 sap://@var{destination}[:@var{port}][?@var{options}]
521 The RTP packets are sent to @var{destination} on port @var{port},
522 or to port 5004 if no port is specified.
523 @var{options} is a @code{&}-separated list. The following options
528 @item announce_addr=@var{address}
529 Specify the destination IP address for sending the announcements to.
530 If omitted, the announcements are sent to the commonly used SAP
531 announcement multicast address 224.2.127.254 (sap.mcast.net), or
532 ff0e::2:7ffe if @var{destination} is an IPv6 address.
534 @item announce_port=@var{port}
535 Specify the port to send the announcements on, defaults to
536 9875 if not specified.
539 Specify the time to live value for the announcements and RTP packets,
542 @item same_port=@var{0|1}
543 If set to 1, send all RTP streams on the same port pair. If zero (the
544 default), all streams are sent on unique ports, with each stream on a
545 port 2 numbers higher than the previous.
546 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
547 The RTP stack in libavformat for receiving requires all streams to be sent
551 Example command lines follow.
553 To broadcast a stream on the local subnet, for watching in VLC:
556 avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
559 Similarly, for watching in avplay:
562 avconv -re -i @var{input} -f sap sap://224.0.0.255
565 And for watching in avplay, over IPv6:
568 avconv -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
573 The syntax for a SAP url given to the demuxer is:
575 sap://[@var{address}][:@var{port}]
578 @var{address} is the multicast address to listen for announcements on,
579 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
580 is the port that is listened on, 9875 if omitted.
582 The demuxers listens for announcements on the given address and port.
583 Once an announcement is received, it tries to receive that particular stream.
585 Example command lines follow.
587 To play back the first stream announced on the normal SAP multicast address:
593 To play back the first stream announced on one the default IPv6 SAP multicast address:
596 avplay sap://[ff0e::2:7ffe]
601 Trasmission Control Protocol.
603 The required syntax for a TCP url is:
605 tcp://@var{hostname}:@var{port}[?@var{options}]
611 Listen for an incoming connection
614 avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
615 avplay tcp://@var{hostname}:@var{port}
622 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
624 The required syntax for a TLS url is:
626 tls://@var{hostname}:@var{port}
629 The following parameters can be set via command line options
630 (or in code via @code{AVOption}s):
635 A file containing certificate authority (CA) root certificates to treat
636 as trusted. If the linked TLS library contains a default this might not
637 need to be specified for verification to work, but not all libraries and
638 setups have defaults built in.
640 @item tls_verify=@var{1|0}
641 If enabled, try to verify the peer that we are communicating with.
642 Note, if using OpenSSL, this currently only makes sure that the
643 peer certificate is signed by one of the root certificates in the CA
644 database, but it does not validate that the certificate actually
645 matches the host name we are trying to connect to. (With GnuTLS,
646 the host name is validated as well.)
648 This is disabled by default since it requires a CA database to be
649 provided by the caller in many cases.
652 A file containing a certificate to use in the handshake with the peer.
653 (When operating as server, in listen mode, this is more often required
654 by the peer, while client certificates only are mandated in certain
658 A file containing the private key for the certificate.
660 @item listen=@var{1|0}
661 If enabled, listen for connections on the provided port, and assume
662 the server role in the handshake instead of the client role.
668 User Datagram Protocol.
670 The required syntax for a UDP url is:
672 udp://@var{hostname}:@var{port}[?@var{options}]
675 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
676 Follow the list of supported options.
680 @item buffer_size=@var{size}
681 set the UDP buffer size in bytes
683 @item localport=@var{port}
684 override the local UDP port to bind with
686 @item localaddr=@var{addr}
687 Choose the local IP address. This is useful e.g. if sending multicast
688 and the host has multiple interfaces, where the user can choose
689 which interface to send on by specifying the IP address of that interface.
691 @item pkt_size=@var{size}
692 set the size in bytes of UDP packets
694 @item reuse=@var{1|0}
695 explicitly allow or disallow reusing UDP sockets
698 set the time to live value (for multicast only)
700 @item connect=@var{1|0}
701 Initialize the UDP socket with @code{connect()}. In this case, the
702 destination address can't be changed with ff_udp_set_remote_url later.
703 If the destination address isn't known at the start, this option can
704 be specified in ff_udp_set_remote_url, too.
705 This allows finding out the source address for the packets with getsockname,
706 and makes writes return with AVERROR(ECONNREFUSED) if "destination
707 unreachable" is received.
708 For receiving, this gives the benefit of only receiving packets from
709 the specified peer address/port.
711 @item sources=@var{address}[,@var{address}]
712 Only receive packets sent to the multicast group from one of the
713 specified sender IP addresses.
715 @item block=@var{address}[,@var{address}]
716 Ignore packets sent to the multicast group from the specified
720 Some usage examples of the udp protocol with @command{avconv} follow.
722 To stream over UDP to a remote endpoint:
724 avconv -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
727 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
729 avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
732 To receive over UDP from a remote endpoint:
734 avconv -i udp://[@var{multicast-address}]:@var{port}
741 The required syntax for a Unix socket URL is:
744 unix://@var{filepath}
747 The following parameters can be set via command line options
748 (or in code via @code{AVOption}s):
754 Create the Unix socket in listening mode.