4 Protocols are configured elements in FFmpeg that enable access to
5 resources that require specific protocols.
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 Caching wrapper for input stream.
56 Cache the input stream to temporary file. It brings seeking capability to live streams.
64 Physical concatenation protocol.
66 Allow to read and seek from many resource in sequence as if they were
69 A URL accepted by this protocol has the syntax:
71 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
74 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
75 resource to be concatenated, each one possibly specifying a distinct
78 For example to read a sequence of files @file{split1.mpeg},
79 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
82 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
85 Note that you may need to escape the character "|" which is special for
90 AES-encrypted stream reading protocol.
92 The accepted options are:
95 Set the AES decryption key binary block from given hexadecimal representation.
98 Set the AES decryption initialization vector binary block from given hexadecimal representation.
101 Accepted URL formats:
109 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
111 For example, to convert a GIF file given inline with @command{ffmpeg}:
113 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
118 File access protocol.
120 Allow to read from or write to a file.
122 A file URL can have the form:
127 where @var{filename} is the path of the file to read.
129 An URL that does not have a protocol prefix will be assumed to be a
130 file URL. Depending on the build, an URL that looks like a Windows
131 path with the drive letter at the beginning will also be assumed to be
132 a file URL (usually not the case in builds for unix-like systems).
134 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
137 ffmpeg -i file:input.mpeg output.mpeg
140 This protocol accepts the following options:
144 Truncate existing files on write, if set to 1. A value of 0 prevents
145 truncating. Default value is 1.
148 Set I/O operation maximum block size, in bytes. Default value is
149 @code{INT_MAX}, which results in not limiting the requested block size.
150 Setting this value reasonably low improves user termination request reaction
151 time, which is valuable for files on slow medium.
156 FTP (File Transfer Protocol).
158 Allow to read from or write to remote resources using FTP protocol.
160 Following syntax is required.
162 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
165 This protocol accepts the following options.
169 Set timeout of socket I/O operations used by the underlying low level
170 operation. By default it is set to -1, which means that the timeout is
173 @item ftp-anonymous-password
174 Password used when login as anonymous user. Typically an e-mail address
177 @item ftp-write-seekable
178 Control seekability of connection during encoding. If set to 1 the
179 resource is supposed to be seekable, if set to 0 it is assumed not
180 to be seekable. Default value is 0.
183 NOTE: Protocol can be used as output, but it is recommended to not do
184 it, unless special care is taken (tests, customized server configuration
185 etc.). Different FTP servers behave in different way during seek
186 operation. ff* tools may produce incomplete content due to server limitations.
194 Read Apple HTTP Live Streaming compliant segmented stream as
195 a uniform one. The M3U8 playlists describing the segments can be
196 remote HTTP resources or local files, accessed using the standard
198 The nested protocol is declared by specifying
199 "+@var{proto}" after the hls URI scheme name, where @var{proto}
200 is either "file" or "http".
203 hls+http://host/path/to/remote/resource.m3u8
204 hls+file://path/to/local/resource.m3u8
207 Using this protocol is discouraged - the hls demuxer should work
208 just as well (if not, please report the issues) and is more complete.
209 To use the hls demuxer instead, simply use the direct URLs to the
214 HTTP (Hyper Text Transfer Protocol).
216 This protocol accepts the following options.
220 Control seekability of connection. If set to 1 the resource is
221 supposed to be seekable, if set to 0 it is assumed not to be seekable,
222 if set to -1 it will try to autodetect if it is seekable. Default
226 If set to 1 use chunked transfer-encoding for posts, default is 1.
229 Set custom HTTP headers, can override built in default headers. The
230 value must be a string encoding the headers.
233 Force a content type.
236 Override User-Agent header. If not specified the protocol will use a
237 string describing the libavformat build.
239 @item multiple_requests
240 Use persistent connections if set to 1. By default it is 0.
243 Set custom HTTP post data.
246 Set timeout of socket I/O operations used by the underlying low level
247 operation. By default it is set to -1, which means that the timeout is
254 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
255 supports this, the metadata has to be retrieved by the application by reading
256 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
259 @item icy_metadata_headers
260 If the server supports ICY metadata, this contains the ICY specific HTTP reply
261 headers, separated with newline characters.
263 @item icy_metadata_packet
264 If the server supports ICY metadata, and @option{icy} was set to 1, this
265 contains the last non-empty metadata packet sent by the server.
268 Set the cookies to be sent in future requests. The format of each cookie is the
269 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
270 delimited by a newline character.
273 @subsection HTTP Cookies
275 Some HTTP requests will be denied unless cookie values are passed in with the
276 request. The @option{cookies} option allows these cookies to be specified. At
277 the very least, each cookie must specify a value along with a path and domain.
278 HTTP requests that match both the domain and path will automatically include the
279 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
282 The required syntax to play a stream specifying a cookie is:
284 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
289 MMS (Microsoft Media Server) protocol over TCP.
293 MMS (Microsoft Media Server) protocol over HTTP.
295 The required syntax is:
297 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
304 Computes the MD5 hash of the data to be written, and on close writes
305 this to the designated output or stdout if none is specified. It can
306 be used to test muxers without writing an actual file.
308 Some examples follow.
310 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
311 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
313 # Write the MD5 hash of the encoded AVI file to stdout.
314 ffmpeg -i input.flv -f avi -y md5:
317 Note that some formats (typically MOV) require the output protocol to
318 be seekable, so they will fail with the MD5 output protocol.
322 UNIX pipe access protocol.
324 Allow to read and write from UNIX pipes.
326 The accepted syntax is:
331 @var{number} is the number corresponding to the file descriptor of the
332 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
333 is not specified, by default the stdout file descriptor will be used
334 for writing, stdin for reading.
336 For example to read from stdin with @command{ffmpeg}:
338 cat test.wav | ffmpeg -i pipe:0
339 # ...this is the same as...
340 cat test.wav | ffmpeg -i pipe:
343 For writing to stdout with @command{ffmpeg}:
345 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
346 # ...this is the same as...
347 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
350 This protocol accepts the following options:
354 Set I/O operation maximum block size, in bytes. Default value is
355 @code{INT_MAX}, which results in not limiting the requested block size.
356 Setting this value reasonably low improves user termination request reaction
357 time, which is valuable if data transmission is slow.
360 Note that some formats (typically MOV), require the output protocol to
361 be seekable, so they will fail with the pipe output protocol.
365 Real-Time Messaging Protocol.
367 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
368 content across a TCP/IP network.
370 The required syntax is:
372 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
375 The accepted parameters are:
379 An optional username (mostly for publishing).
382 An optional password (mostly for publishing).
385 The address of the RTMP server.
388 The number of the TCP port to use (by default is 1935).
391 It is the name of the application to access. It usually corresponds to
392 the path where the application is installed on the RTMP server
393 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
394 the value parsed from the URI through the @code{rtmp_app} option, too.
397 It is the path or name of the resource to play with reference to the
398 application specified in @var{app}, may be prefixed by "mp4:". You
399 can override the value parsed from the URI through the @code{rtmp_playpath}
403 Act as a server, listening for an incoming connection.
406 Maximum time to wait for the incoming connection. Implies listen.
409 Additionally, the following parameters can be set via command line options
410 (or in code via @code{AVOption}s):
414 Name of application to connect on the RTMP server. This option
415 overrides the parameter specified in the URI.
418 Set the client buffer time in milliseconds. The default is 3000.
421 Extra arbitrary AMF connection parameters, parsed from a string,
422 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
423 Each value is prefixed by a single character denoting the type,
424 B for Boolean, N for number, S for string, O for object, or Z for null,
425 followed by a colon. For Booleans the data must be either 0 or 1 for
426 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
427 1 to end or begin an object, respectively. Data items in subobjects may
428 be named, by prefixing the type with 'N' and specifying the name before
429 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
430 times to construct arbitrary AMF sequences.
433 Version of the Flash plugin used to run the SWF player. The default
434 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
435 <libavformat version>).)
437 @item rtmp_flush_interval
438 Number of packets flushed in the same request (RTMPT only). The default
442 Specify that the media is a live stream. No resuming or seeking in
443 live streams is possible. The default value is @code{any}, which means the
444 subscriber first tries to play the live stream specified in the
445 playpath. If a live stream of that name is not found, it plays the
446 recorded stream. The other possible values are @code{live} and
450 URL of the web page in which the media was embedded. By default no
454 Stream identifier to play or to publish. This option overrides the
455 parameter specified in the URI.
458 Name of live stream to subscribe to. By default no value will be sent.
459 It is only sent if the option is specified or if rtmp_live
463 SHA256 hash of the decompressed SWF file (32 bytes).
466 Size of the decompressed SWF file, required for SWFVerification.
469 URL of the SWF player for the media. By default no value will be sent.
472 URL to player swf file, compute hash/size automatically.
475 URL of the target stream. Defaults to proto://host[:port]/app.
479 For example to read with @command{ffplay} a multimedia resource named
480 "sample" from the application "vod" from an RTMP server "myserver":
482 ffplay rtmp://myserver/vod/sample
485 To publish to a password protected server, passing the playpath and
486 app names separately:
488 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
493 Encrypted Real-Time Messaging Protocol.
495 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
496 streaming multimedia content within standard cryptographic primitives,
497 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
502 Real-Time Messaging Protocol over a secure SSL connection.
504 The Real-Time Messaging Protocol (RTMPS) is used for streaming
505 multimedia content across an encrypted connection.
509 Real-Time Messaging Protocol tunneled through HTTP.
511 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
512 for streaming multimedia content within HTTP requests to traverse
517 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
519 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
520 is used for streaming multimedia content within HTTP requests to traverse
525 Real-Time Messaging Protocol tunneled through HTTPS.
527 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
528 for streaming multimedia content within HTTPS requests to traverse
533 Secure File Transfer Protocol via libssh
535 Allow to read from or write to remote resources using SFTP protocol.
537 Following syntax is required.
540 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
543 This protocol accepts the following options.
547 Set timeout of socket I/O operations used by the underlying low level
548 operation. By default it is set to -1, which means that the timeout
552 Truncate existing files on write, if set to 1. A value of 0 prevents
553 truncating. Default value is 1.
557 Example: Play a file stored on remote server.
560 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
563 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
565 Real-Time Messaging Protocol and its variants supported through
568 Requires the presence of the librtmp headers and library during
569 configuration. You need to explicitly configure the build with
570 "--enable-librtmp". If enabled this will replace the native RTMP
573 This protocol provides most client functions and a few server
574 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
575 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
576 variants of these encrypted types (RTMPTE, RTMPTS).
578 The required syntax is:
580 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
583 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
584 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
585 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
586 meaning as specified for the RTMP native protocol.
587 @var{options} contains a list of space-separated options of the form
590 See the librtmp manual page (man 3 librtmp) for more information.
592 For example, to stream a file in real-time to an RTMP server using
595 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
598 To play the same stream using @command{ffplay}:
600 ffplay "rtmp://myserver/live/mystream live=1"
605 Real-time Transport Protocol.
607 The required syntax for an RTP URL is:
608 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
610 @var{port} specifies the RTP port to use.
612 The following URL options are supported:
617 Set the TTL (Time-To-Live) value (for multicast only).
619 @item rtcpport=@var{n}
620 Set the remote RTCP port to @var{n}.
622 @item localrtpport=@var{n}
623 Set the local RTP port to @var{n}.
625 @item localrtcpport=@var{n}'
626 Set the local RTCP port to @var{n}.
628 @item pkt_size=@var{n}
629 Set max packet size (in bytes) to @var{n}.
632 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
635 @item sources=@var{ip}[,@var{ip}]
636 List allowed source IP addresses.
638 @item block=@var{ip}[,@var{ip}]
639 List disallowed (blocked) source IP addresses.
641 @item write_to_source=0|1
642 Send packets to the source address of the latest received packet (if
643 set to 1) or to a default remote address (if set to 0).
645 @item localport=@var{n}
646 Set the local RTP port to @var{n}.
648 This is a deprecated option. Instead, @option{localrtpport} should be
658 If @option{rtcpport} is not set the RTCP port will be set to the RTP
662 If @option{localrtpport} (the local RTP port) is not set any available
663 port will be used for the local RTP and RTCP ports.
666 If @option{localrtcpport} (the local RTCP port) is not set it will be
667 set to the the local RTP port value plus 1.
672 Real-Time Streaming Protocol.
674 RTSP is not technically a protocol handler in libavformat, it is a demuxer
675 and muxer. The demuxer supports both normal RTSP (with data transferred
676 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
677 data transferred over RDT).
679 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
680 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
681 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
683 The required syntax for a RTSP url is:
685 rtsp://@var{hostname}[:@var{port}]/@var{path}
688 Options can be set on the @command{ffmpeg}/@command{ffplay} command
689 line, or set in code via @code{AVOption}s or in
690 @code{avformat_open_input}.
692 The following options are supported.
696 Do not start playing the stream immediately if set to 1. Default value
700 Set RTSP trasport protocols.
702 It accepts the following values:
705 Use UDP as lower transport protocol.
708 Use TCP (interleaving within the RTSP control channel) as lower
712 Use UDP multicast as lower transport protocol.
715 Use HTTP tunneling as lower transport protocol, which is useful for
719 Multiple lower transport protocols may be specified, in that case they are
720 tried one at a time (if the setup of one fails, the next one is tried).
721 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
726 The following values are accepted:
729 Accept packets only from negotiated peer address and port.
731 Act as a server, listening for an incoming connection.
734 Default value is @samp{none}.
736 @item allowed_media_types
737 Set media types to accept from the server.
739 The following flags are accepted:
746 By default it accepts all media types.
749 Set minimum local UDP port. Default value is 5000.
752 Set maximum local UDP port. Default value is 65000.
755 Set maximum timeout (in seconds) to wait for incoming connections.
757 A value of -1 mean infinite (default). This option implies the
758 @option{rtsp_flags} set to @samp{listen}.
760 @item reorder_queue_size
761 Set number of packets to buffer for handling of reordered packets.
764 Set socket TCP I/O timeout in micro seconds.
767 Override User-Agent header. If not specified, it default to the
768 libavformat identifier string.
771 When receiving data over UDP, the demuxer tries to reorder received packets
772 (since they may arrive out of order, or packets may get lost totally). This
773 can be disabled by setting the maximum demuxing delay to zero (via
774 the @code{max_delay} field of AVFormatContext).
776 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
777 streams to display can be chosen with @code{-vst} @var{n} and
778 @code{-ast} @var{n} for video and audio respectively, and can be switched
779 on the fly by pressing @code{v} and @code{a}.
783 The following examples all make use of the @command{ffplay} and
784 @command{ffmpeg} tools.
788 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
790 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
794 Watch a stream tunneled over HTTP:
796 ffplay -rtsp_transport http rtsp://server/video.mp4
800 Send a stream in realtime to a RTSP server, for others to watch:
802 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
806 Receive a stream in realtime:
808 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
814 Session Announcement Protocol (RFC 2974). This is not technically a
815 protocol handler in libavformat, it is a muxer and demuxer.
816 It is used for signalling of RTP streams, by announcing the SDP for the
817 streams regularly on a separate port.
821 The syntax for a SAP url given to the muxer is:
823 sap://@var{destination}[:@var{port}][?@var{options}]
826 The RTP packets are sent to @var{destination} on port @var{port},
827 or to port 5004 if no port is specified.
828 @var{options} is a @code{&}-separated list. The following options
833 @item announce_addr=@var{address}
834 Specify the destination IP address for sending the announcements to.
835 If omitted, the announcements are sent to the commonly used SAP
836 announcement multicast address 224.2.127.254 (sap.mcast.net), or
837 ff0e::2:7ffe if @var{destination} is an IPv6 address.
839 @item announce_port=@var{port}
840 Specify the port to send the announcements on, defaults to
841 9875 if not specified.
844 Specify the time to live value for the announcements and RTP packets,
847 @item same_port=@var{0|1}
848 If set to 1, send all RTP streams on the same port pair. If zero (the
849 default), all streams are sent on unique ports, with each stream on a
850 port 2 numbers higher than the previous.
851 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
852 The RTP stack in libavformat for receiving requires all streams to be sent
856 Example command lines follow.
858 To broadcast a stream on the local subnet, for watching in VLC:
861 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
864 Similarly, for watching in @command{ffplay}:
867 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
870 And for watching in @command{ffplay}, over IPv6:
873 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
878 The syntax for a SAP url given to the demuxer is:
880 sap://[@var{address}][:@var{port}]
883 @var{address} is the multicast address to listen for announcements on,
884 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
885 is the port that is listened on, 9875 if omitted.
887 The demuxers listens for announcements on the given address and port.
888 Once an announcement is received, it tries to receive that particular stream.
890 Example command lines follow.
892 To play back the first stream announced on the normal SAP multicast address:
898 To play back the first stream announced on one the default IPv6 SAP multicast address:
901 ffplay sap://[ff0e::2:7ffe]
906 Stream Control Transmission Protocol.
908 The accepted URL syntax is:
910 sctp://@var{host}:@var{port}[?@var{options}]
913 The protocol accepts the following options:
916 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
919 Set the maximum number of streams. By default no limit is set.
924 Secure Real-time Transport Protocol.
926 The accepted options are:
930 Select input and output encoding suites.
934 @item AES_CM_128_HMAC_SHA1_80
935 @item SRTP_AES128_CM_HMAC_SHA1_80
936 @item AES_CM_128_HMAC_SHA1_32
937 @item SRTP_AES128_CM_HMAC_SHA1_32
941 @item srtp_out_params
942 Set input and output encoding parameters, which are expressed by a
943 base64-encoded representation of a binary block. The first 16 bytes of
944 this binary block are used as master key, the following 14 bytes are
950 Trasmission Control Protocol.
952 The required syntax for a TCP url is:
954 tcp://@var{hostname}:@var{port}[?@var{options}]
957 @var{options} contains a list of &-separated options of the form
960 The list of supported options follows.
963 @item listen=@var{1|0}
964 Listen for an incoming connection. Default value is 0.
966 @item timeout=@var{microseconds}
967 Set raise error timeout, expressed in microseconds.
969 This option is only relevant in read mode: if no data arrived in more
970 than this time interval, raise error.
972 @item listen_timeout=@var{microseconds}
973 Set listen timeout, expressed in microseconds.
976 The following example shows how to setup a listening TCP connection
977 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
979 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
980 ffplay tcp://@var{hostname}:@var{port}
985 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
987 The required syntax for a TLS/SSL url is:
989 tls://@var{hostname}:@var{port}[?@var{options}]
992 The following parameters can be set via command line options
993 (or in code via @code{AVOption}s):
997 @item ca_file, cafile=@var{filename}
998 A file containing certificate authority (CA) root certificates to treat
999 as trusted. If the linked TLS library contains a default this might not
1000 need to be specified for verification to work, but not all libraries and
1001 setups have defaults built in.
1002 The file must be in OpenSSL PEM format.
1004 @item tls_verify=@var{1|0}
1005 If enabled, try to verify the peer that we are communicating with.
1006 Note, if using OpenSSL, this currently only makes sure that the
1007 peer certificate is signed by one of the root certificates in the CA
1008 database, but it does not validate that the certificate actually
1009 matches the host name we are trying to connect to. (With GnuTLS,
1010 the host name is validated as well.)
1012 This is disabled by default since it requires a CA database to be
1013 provided by the caller in many cases.
1015 @item cert_file, cert=@var{filename}
1016 A file containing a certificate to use in the handshake with the peer.
1017 (When operating as server, in listen mode, this is more often required
1018 by the peer, while client certificates only are mandated in certain
1021 @item key_file, key=@var{filename}
1022 A file containing the private key for the certificate.
1024 @item listen=@var{1|0}
1025 If enabled, listen for connections on the provided port, and assume
1026 the server role in the handshake instead of the client role.
1030 Example command lines:
1032 To create a TLS/SSL server that serves an input stream.
1035 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1038 To play back a stream from the TLS/SSL server using @command{ffplay}:
1041 ffplay tls://@var{hostname}:@var{port}
1046 User Datagram Protocol.
1048 The required syntax for an UDP URL is:
1050 udp://@var{hostname}:@var{port}[?@var{options}]
1053 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1055 In case threading is enabled on the system, a circular buffer is used
1056 to store the incoming data, which allows to reduce loss of data due to
1057 UDP socket buffer overruns. The @var{fifo_size} and
1058 @var{overrun_nonfatal} options are related to this buffer.
1060 The list of supported options follows.
1063 @item buffer_size=@var{size}
1064 Set the UDP socket buffer size in bytes. This is used both for the
1065 receiving and the sending buffer size.
1067 @item localport=@var{port}
1068 Override the local UDP port to bind with.
1070 @item localaddr=@var{addr}
1071 Choose the local IP address. This is useful e.g. if sending multicast
1072 and the host has multiple interfaces, where the user can choose
1073 which interface to send on by specifying the IP address of that interface.
1075 @item pkt_size=@var{size}
1076 Set the size in bytes of UDP packets.
1078 @item reuse=@var{1|0}
1079 Explicitly allow or disallow reusing UDP sockets.
1082 Set the time to live value (for multicast only).
1084 @item connect=@var{1|0}
1085 Initialize the UDP socket with @code{connect()}. In this case, the
1086 destination address can't be changed with ff_udp_set_remote_url later.
1087 If the destination address isn't known at the start, this option can
1088 be specified in ff_udp_set_remote_url, too.
1089 This allows finding out the source address for the packets with getsockname,
1090 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1091 unreachable" is received.
1092 For receiving, this gives the benefit of only receiving packets from
1093 the specified peer address/port.
1095 @item sources=@var{address}[,@var{address}]
1096 Only receive packets sent to the multicast group from one of the
1097 specified sender IP addresses.
1099 @item block=@var{address}[,@var{address}]
1100 Ignore packets sent to the multicast group from the specified
1101 sender IP addresses.
1103 @item fifo_size=@var{units}
1104 Set the UDP receiving circular buffer size, expressed as a number of
1105 packets with size of 188 bytes. If not specified defaults to 7*4096.
1107 @item overrun_nonfatal=@var{1|0}
1108 Survive in case of UDP receiving circular buffer overrun. Default
1111 @item timeout=@var{microseconds}
1112 Set raise error timeout, expressed in microseconds.
1114 This option is only relevant in read mode: if no data arrived in more
1115 than this time interval, raise error.
1118 @subsection Examples
1122 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1124 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1128 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1129 sized UDP packets, using a large input buffer:
1131 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1135 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1137 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1145 The required syntax for a Unix socket URL is:
1148 unix://@var{filepath}
1151 The following parameters can be set via command line options
1152 (or in code via @code{AVOption}s):
1158 Create the Unix socket in listening mode.
1161 @c man end PROTOCOLS