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 read to a file.
122 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
125 ffmpeg -i file:input.mpeg output.mpeg
128 The ff* tools default to the file protocol, that is a resource
129 specified with the name "FILE.mpeg" is interpreted as the URL
132 This protocol accepts the following options:
136 Truncate existing files on write, if set to 1. A value of 0 prevents
137 truncating. Default value is 1.
140 Set I/O operation maximum block size, in bytes. Default value is
141 @code{INT_MAX}, which results in not limiting the requested block size.
142 Setting this value reasonably low improves user termination request reaction
143 time, which is valuable for files on slow medium.
148 FTP (File Transfer Protocol).
150 Allow to read from or write to remote resources using FTP protocol.
152 Following syntax is required.
154 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
157 This protocol accepts the following options.
161 Set timeout of socket I/O operations used by the underlying low level
162 operation. By default it is set to -1, which means that the timeout is
165 @item ftp-anonymous-password
166 Password used when login as anonymous user. Typically an e-mail address
169 @item ftp-write-seekable
170 Control seekability of connection during encoding. If set to 1 the
171 resource is supposed to be seekable, if set to 0 it is assumed not
172 to be seekable. Default value is 0.
175 NOTE: Protocol can be used as output, but it is recommended to not do
176 it, unless special care is taken (tests, customized server configuration
177 etc.). Different FTP servers behave in different way during seek
178 operation. ff* tools may produce incomplete content due to server limitations.
186 Read Apple HTTP Live Streaming compliant segmented stream as
187 a uniform one. The M3U8 playlists describing the segments can be
188 remote HTTP resources or local files, accessed using the standard
190 The nested protocol is declared by specifying
191 "+@var{proto}" after the hls URI scheme name, where @var{proto}
192 is either "file" or "http".
195 hls+http://host/path/to/remote/resource.m3u8
196 hls+file://path/to/local/resource.m3u8
199 Using this protocol is discouraged - the hls demuxer should work
200 just as well (if not, please report the issues) and is more complete.
201 To use the hls demuxer instead, simply use the direct URLs to the
206 HTTP (Hyper Text Transfer Protocol).
208 This protocol accepts the following options.
212 Control seekability of connection. If set to 1 the resource is
213 supposed to be seekable, if set to 0 it is assumed not to be seekable,
214 if set to -1 it will try to autodetect if it is seekable. Default
218 If set to 1 use chunked transfer-encoding for posts, default is 1.
221 Set custom HTTP headers, can override built in default headers. The
222 value must be a string encoding the headers.
225 Force a content type.
228 Override User-Agent header. If not specified the protocol will use a
229 string describing the libavformat build.
231 @item multiple_requests
232 Use persistent connections if set to 1. By default it is 0.
235 Set custom HTTP post data.
238 Set timeout of socket I/O operations used by the underlying low level
239 operation. By default it is set to -1, which means that the timeout is
246 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
247 supports this, the metadata has to be retrieved by the application by reading
248 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
251 @item icy_metadata_headers
252 If the server supports ICY metadata, this contains the ICY specific HTTP reply
253 headers, separated with newline characters.
255 @item icy_metadata_packet
256 If the server supports ICY metadata, and @option{icy} was set to 1, this
257 contains the last non-empty metadata packet sent by the server.
260 Set the cookies to be sent in future requests. The format of each cookie is the
261 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
262 delimited by a newline character.
265 @subsection HTTP Cookies
267 Some HTTP requests will be denied unless cookie values are passed in with the
268 request. The @option{cookies} option allows these cookies to be specified. At
269 the very least, each cookie must specify a value along with a path and domain.
270 HTTP requests that match both the domain and path will automatically include the
271 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
274 The required syntax to play a stream specifying a cookie is:
276 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
281 MMS (Microsoft Media Server) protocol over TCP.
285 MMS (Microsoft Media Server) protocol over HTTP.
287 The required syntax is:
289 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
296 Computes the MD5 hash of the data to be written, and on close writes
297 this to the designated output or stdout if none is specified. It can
298 be used to test muxers without writing an actual file.
300 Some examples follow.
302 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
303 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
305 # Write the MD5 hash of the encoded AVI file to stdout.
306 ffmpeg -i input.flv -f avi -y md5:
309 Note that some formats (typically MOV) require the output protocol to
310 be seekable, so they will fail with the MD5 output protocol.
314 UNIX pipe access protocol.
316 Allow to read and write from UNIX pipes.
318 The accepted syntax is:
323 @var{number} is the number corresponding to the file descriptor of the
324 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
325 is not specified, by default the stdout file descriptor will be used
326 for writing, stdin for reading.
328 For example to read from stdin with @command{ffmpeg}:
330 cat test.wav | ffmpeg -i pipe:0
331 # ...this is the same as...
332 cat test.wav | ffmpeg -i pipe:
335 For writing to stdout with @command{ffmpeg}:
337 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
338 # ...this is the same as...
339 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
342 This protocol accepts the following options:
346 Set I/O operation maximum block size, in bytes. Default value is
347 @code{INT_MAX}, which results in not limiting the requested block size.
348 Setting this value reasonably low improves user termination request reaction
349 time, which is valuable if data transmission is slow.
352 Note that some formats (typically MOV), require the output protocol to
353 be seekable, so they will fail with the pipe output protocol.
357 Real-Time Messaging Protocol.
359 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
360 content across a TCP/IP network.
362 The required syntax is:
364 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
367 The accepted parameters are:
371 An optional username (mostly for publishing).
374 An optional password (mostly for publishing).
377 The address of the RTMP server.
380 The number of the TCP port to use (by default is 1935).
383 It is the name of the application to access. It usually corresponds to
384 the path where the application is installed on the RTMP server
385 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
386 the value parsed from the URI through the @code{rtmp_app} option, too.
389 It is the path or name of the resource to play with reference to the
390 application specified in @var{app}, may be prefixed by "mp4:". You
391 can override the value parsed from the URI through the @code{rtmp_playpath}
395 Act as a server, listening for an incoming connection.
398 Maximum time to wait for the incoming connection. Implies listen.
401 Additionally, the following parameters can be set via command line options
402 (or in code via @code{AVOption}s):
406 Name of application to connect on the RTMP server. This option
407 overrides the parameter specified in the URI.
410 Set the client buffer time in milliseconds. The default is 3000.
413 Extra arbitrary AMF connection parameters, parsed from a string,
414 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
415 Each value is prefixed by a single character denoting the type,
416 B for Boolean, N for number, S for string, O for object, or Z for null,
417 followed by a colon. For Booleans the data must be either 0 or 1 for
418 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
419 1 to end or begin an object, respectively. Data items in subobjects may
420 be named, by prefixing the type with 'N' and specifying the name before
421 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
422 times to construct arbitrary AMF sequences.
425 Version of the Flash plugin used to run the SWF player. The default
426 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
427 <libavformat version>).)
429 @item rtmp_flush_interval
430 Number of packets flushed in the same request (RTMPT only). The default
434 Specify that the media is a live stream. No resuming or seeking in
435 live streams is possible. The default value is @code{any}, which means the
436 subscriber first tries to play the live stream specified in the
437 playpath. If a live stream of that name is not found, it plays the
438 recorded stream. The other possible values are @code{live} and
442 URL of the web page in which the media was embedded. By default no
446 Stream identifier to play or to publish. This option overrides the
447 parameter specified in the URI.
450 Name of live stream to subscribe to. By default no value will be sent.
451 It is only sent if the option is specified or if rtmp_live
455 SHA256 hash of the decompressed SWF file (32 bytes).
458 Size of the decompressed SWF file, required for SWFVerification.
461 URL of the SWF player for the media. By default no value will be sent.
464 URL to player swf file, compute hash/size automatically.
467 URL of the target stream. Defaults to proto://host[:port]/app.
471 For example to read with @command{ffplay} a multimedia resource named
472 "sample" from the application "vod" from an RTMP server "myserver":
474 ffplay rtmp://myserver/vod/sample
477 To publish to a password protected server, passing the playpath and
478 app names separately:
480 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
485 Encrypted Real-Time Messaging Protocol.
487 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
488 streaming multimedia content within standard cryptographic primitives,
489 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
494 Real-Time Messaging Protocol over a secure SSL connection.
496 The Real-Time Messaging Protocol (RTMPS) is used for streaming
497 multimedia content across an encrypted connection.
501 Real-Time Messaging Protocol tunneled through HTTP.
503 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
504 for streaming multimedia content within HTTP requests to traverse
509 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
511 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
512 is used for streaming multimedia content within HTTP requests to traverse
517 Real-Time Messaging Protocol tunneled through HTTPS.
519 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
520 for streaming multimedia content within HTTPS requests to traverse
525 Secure File Transfer Protocol via libssh
527 Allow to read from or write to remote resources using SFTP protocol.
529 Following syntax is required.
532 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
535 This protocol accepts the following options.
539 Set timeout of socket I/O operations used by the underlying low level
540 operation. By default it is set to -1, which means that the timeout
544 Truncate existing files on write, if set to 1. A value of 0 prevents
545 truncating. Default value is 1.
549 Example: Play a file stored on remote server.
552 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
555 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
557 Real-Time Messaging Protocol and its variants supported through
560 Requires the presence of the librtmp headers and library during
561 configuration. You need to explicitly configure the build with
562 "--enable-librtmp". If enabled this will replace the native RTMP
565 This protocol provides most client functions and a few server
566 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
567 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
568 variants of these encrypted types (RTMPTE, RTMPTS).
570 The required syntax is:
572 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
575 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
576 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
577 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
578 meaning as specified for the RTMP native protocol.
579 @var{options} contains a list of space-separated options of the form
582 See the librtmp manual page (man 3 librtmp) for more information.
584 For example, to stream a file in real-time to an RTMP server using
587 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
590 To play the same stream using @command{ffplay}:
592 ffplay "rtmp://myserver/live/mystream live=1"
597 Real-time Transport Protocol.
599 The required syntax for an RTP URL is:
600 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
602 @var{port} specifies the RTP port to use.
604 The following URL options are supported:
609 Set the TTL (Time-To-Live) value (for multicast only).
611 @item rtcpport=@var{n}
612 Set the remote RTCP port to @var{n}.
614 @item localrtpport=@var{n}
615 Set the local RTP port to @var{n}.
617 @item localrtcpport=@var{n}'
618 Set the local RTCP port to @var{n}.
620 @item pkt_size=@var{n}
621 Set max packet size (in bytes) to @var{n}.
624 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
627 @item sources=@var{ip}[,@var{ip}]
628 List allowed source IP addresses.
630 @item block=@var{ip}[,@var{ip}]
631 List disallowed (blocked) source IP addresses.
633 @item write_to_source=0|1
634 Send packets to the source address of the latest received packet (if
635 set to 1) or to a default remote address (if set to 0).
637 @item localport=@var{n}
638 Set the local RTP port to @var{n}.
640 This is a deprecated option. Instead, @option{localrtpport} should be
650 If @option{rtcpport} is not set the RTCP port will be set to the RTP
654 If @option{localrtpport} (the local RTP port) is not set any available
655 port will be used for the local RTP and RTCP ports.
658 If @option{localrtcpport} (the local RTCP port) is not set it will be
659 set to the the local RTP port value plus 1.
664 Real-Time Streaming Protocol.
666 RTSP is not technically a protocol handler in libavformat, it is a demuxer
667 and muxer. The demuxer supports both normal RTSP (with data transferred
668 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
669 data transferred over RDT).
671 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
672 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
673 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
675 The required syntax for a RTSP url is:
677 rtsp://@var{hostname}[:@var{port}]/@var{path}
680 Options can be set on the @command{ffmpeg}/@command{ffplay} command
681 line, or set in code via @code{AVOption}s or in
682 @code{avformat_open_input}.
684 The following options are supported.
688 Do not start playing the stream immediately if set to 1. Default value
692 Set RTSP trasport protocols.
694 It accepts the following values:
697 Use UDP as lower transport protocol.
700 Use TCP (interleaving within the RTSP control channel) as lower
704 Use UDP multicast as lower transport protocol.
707 Use HTTP tunneling as lower transport protocol, which is useful for
711 Multiple lower transport protocols may be specified, in that case they are
712 tried one at a time (if the setup of one fails, the next one is tried).
713 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
718 The following values are accepted:
721 Accept packets only from negotiated peer address and port.
723 Act as a server, listening for an incoming connection.
726 Default value is @samp{none}.
728 @item allowed_media_types
729 Set media types to accept from the server.
731 The following flags are accepted:
738 By default it accepts all media types.
741 Set minimum local UDP port. Default value is 5000.
744 Set maximum local UDP port. Default value is 65000.
747 Set maximum timeout (in seconds) to wait for incoming connections.
749 A value of -1 mean infinite (default). This option implies the
750 @option{rtsp_flags} set to @samp{listen}.
752 @item reorder_queue_size
753 Set number of packets to buffer for handling of reordered packets.
756 Set socket TCP I/O timeout in micro seconds.
759 Override User-Agent header. If not specified, it default to the
760 libavformat identifier string.
763 When receiving data over UDP, the demuxer tries to reorder received packets
764 (since they may arrive out of order, or packets may get lost totally). This
765 can be disabled by setting the maximum demuxing delay to zero (via
766 the @code{max_delay} field of AVFormatContext).
768 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
769 streams to display can be chosen with @code{-vst} @var{n} and
770 @code{-ast} @var{n} for video and audio respectively, and can be switched
771 on the fly by pressing @code{v} and @code{a}.
775 The following examples all make use of the @command{ffplay} and
776 @command{ffmpeg} tools.
780 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
782 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
786 Watch a stream tunneled over HTTP:
788 ffplay -rtsp_transport http rtsp://server/video.mp4
792 Send a stream in realtime to a RTSP server, for others to watch:
794 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
798 Receive a stream in realtime:
800 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
806 Session Announcement Protocol (RFC 2974). This is not technically a
807 protocol handler in libavformat, it is a muxer and demuxer.
808 It is used for signalling of RTP streams, by announcing the SDP for the
809 streams regularly on a separate port.
813 The syntax for a SAP url given to the muxer is:
815 sap://@var{destination}[:@var{port}][?@var{options}]
818 The RTP packets are sent to @var{destination} on port @var{port},
819 or to port 5004 if no port is specified.
820 @var{options} is a @code{&}-separated list. The following options
825 @item announce_addr=@var{address}
826 Specify the destination IP address for sending the announcements to.
827 If omitted, the announcements are sent to the commonly used SAP
828 announcement multicast address 224.2.127.254 (sap.mcast.net), or
829 ff0e::2:7ffe if @var{destination} is an IPv6 address.
831 @item announce_port=@var{port}
832 Specify the port to send the announcements on, defaults to
833 9875 if not specified.
836 Specify the time to live value for the announcements and RTP packets,
839 @item same_port=@var{0|1}
840 If set to 1, send all RTP streams on the same port pair. If zero (the
841 default), all streams are sent on unique ports, with each stream on a
842 port 2 numbers higher than the previous.
843 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
844 The RTP stack in libavformat for receiving requires all streams to be sent
848 Example command lines follow.
850 To broadcast a stream on the local subnet, for watching in VLC:
853 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
856 Similarly, for watching in @command{ffplay}:
859 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
862 And for watching in @command{ffplay}, over IPv6:
865 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
870 The syntax for a SAP url given to the demuxer is:
872 sap://[@var{address}][:@var{port}]
875 @var{address} is the multicast address to listen for announcements on,
876 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
877 is the port that is listened on, 9875 if omitted.
879 The demuxers listens for announcements on the given address and port.
880 Once an announcement is received, it tries to receive that particular stream.
882 Example command lines follow.
884 To play back the first stream announced on the normal SAP multicast address:
890 To play back the first stream announced on one the default IPv6 SAP multicast address:
893 ffplay sap://[ff0e::2:7ffe]
898 Stream Control Transmission Protocol.
900 The accepted URL syntax is:
902 sctp://@var{host}:@var{port}[?@var{options}]
905 The protocol accepts the following options:
908 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
911 Set the maximum number of streams. By default no limit is set.
916 Secure Real-time Transport Protocol.
918 The accepted options are:
922 Select input and output encoding suites.
926 @item AES_CM_128_HMAC_SHA1_80
927 @item SRTP_AES128_CM_HMAC_SHA1_80
928 @item AES_CM_128_HMAC_SHA1_32
929 @item SRTP_AES128_CM_HMAC_SHA1_32
933 @item srtp_out_params
934 Set input and output encoding parameters, which are expressed by a
935 base64-encoded representation of a binary block. The first 16 bytes of
936 this binary block are used as master key, the following 14 bytes are
942 Trasmission Control Protocol.
944 The required syntax for a TCP url is:
946 tcp://@var{hostname}:@var{port}[?@var{options}]
949 @var{options} contains a list of &-separated options of the form
952 The list of supported options follows.
955 @item listen=@var{1|0}
956 Listen for an incoming connection. Default value is 0.
958 @item timeout=@var{microseconds}
959 Set raise error timeout, expressed in microseconds.
961 This option is only relevant in read mode: if no data arrived in more
962 than this time interval, raise error.
964 @item listen_timeout=@var{microseconds}
965 Set listen timeout, expressed in microseconds.
968 The following example shows how to setup a listening TCP connection
969 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
971 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
972 ffplay tcp://@var{hostname}:@var{port}
977 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
979 The required syntax for a TLS/SSL url is:
981 tls://@var{hostname}:@var{port}[?@var{options}]
984 The following parameters can be set via command line options
985 (or in code via @code{AVOption}s):
989 @item ca_file, cafile=@var{filename}
990 A file containing certificate authority (CA) root certificates to treat
991 as trusted. If the linked TLS library contains a default this might not
992 need to be specified for verification to work, but not all libraries and
993 setups have defaults built in.
994 The file must be in OpenSSL PEM format.
996 @item tls_verify=@var{1|0}
997 If enabled, try to verify the peer that we are communicating with.
998 Note, if using OpenSSL, this currently only makes sure that the
999 peer certificate is signed by one of the root certificates in the CA
1000 database, but it does not validate that the certificate actually
1001 matches the host name we are trying to connect to. (With GnuTLS,
1002 the host name is validated as well.)
1004 This is disabled by default since it requires a CA database to be
1005 provided by the caller in many cases.
1007 @item cert_file, cert=@var{filename}
1008 A file containing a certificate to use in the handshake with the peer.
1009 (When operating as server, in listen mode, this is more often required
1010 by the peer, while client certificates only are mandated in certain
1013 @item key_file, key=@var{filename}
1014 A file containing the private key for the certificate.
1016 @item listen=@var{1|0}
1017 If enabled, listen for connections on the provided port, and assume
1018 the server role in the handshake instead of the client role.
1022 Example command lines:
1024 To create a TLS/SSL server that serves an input stream.
1027 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1030 To play back a stream from the TLS/SSL server using @command{ffplay}:
1033 ffplay tls://@var{hostname}:@var{port}
1038 User Datagram Protocol.
1040 The required syntax for an UDP URL is:
1042 udp://@var{hostname}:@var{port}[?@var{options}]
1045 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1047 In case threading is enabled on the system, a circular buffer is used
1048 to store the incoming data, which allows to reduce loss of data due to
1049 UDP socket buffer overruns. The @var{fifo_size} and
1050 @var{overrun_nonfatal} options are related to this buffer.
1052 The list of supported options follows.
1055 @item buffer_size=@var{size}
1056 Set the UDP socket buffer size in bytes. This is used both for the
1057 receiving and the sending buffer size.
1059 @item localport=@var{port}
1060 Override the local UDP port to bind with.
1062 @item localaddr=@var{addr}
1063 Choose the local IP address. This is useful e.g. if sending multicast
1064 and the host has multiple interfaces, where the user can choose
1065 which interface to send on by specifying the IP address of that interface.
1067 @item pkt_size=@var{size}
1068 Set the size in bytes of UDP packets.
1070 @item reuse=@var{1|0}
1071 Explicitly allow or disallow reusing UDP sockets.
1074 Set the time to live value (for multicast only).
1076 @item connect=@var{1|0}
1077 Initialize the UDP socket with @code{connect()}. In this case, the
1078 destination address can't be changed with ff_udp_set_remote_url later.
1079 If the destination address isn't known at the start, this option can
1080 be specified in ff_udp_set_remote_url, too.
1081 This allows finding out the source address for the packets with getsockname,
1082 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1083 unreachable" is received.
1084 For receiving, this gives the benefit of only receiving packets from
1085 the specified peer address/port.
1087 @item sources=@var{address}[,@var{address}]
1088 Only receive packets sent to the multicast group from one of the
1089 specified sender IP addresses.
1091 @item block=@var{address}[,@var{address}]
1092 Ignore packets sent to the multicast group from the specified
1093 sender IP addresses.
1095 @item fifo_size=@var{units}
1096 Set the UDP receiving circular buffer size, expressed as a number of
1097 packets with size of 188 bytes. If not specified defaults to 7*4096.
1099 @item overrun_nonfatal=@var{1|0}
1100 Survive in case of UDP receiving circular buffer overrun. Default
1103 @item timeout=@var{microseconds}
1104 Set raise error timeout, expressed in microseconds.
1106 This option is only relevant in read mode: if no data arrived in more
1107 than this time interval, raise error.
1110 @subsection Examples
1114 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1116 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1120 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1121 sized UDP packets, using a large input buffer:
1123 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1127 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1129 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1137 The required syntax for a Unix socket URL is:
1140 unix://@var{filepath}
1143 The following parameters can be set via command line options
1144 (or in code via @code{AVOption}s):
1150 Create the Unix socket in listening mode.
1153 @c man end PROTOCOLS