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 a specific content type for the POST messages.
232 Set custom HTTP headers, can override built in default headers. The
233 value must be a string encoding the headers.
235 @item multiple_requests
236 Use persistent connections if set to 1, default is 0.
239 Set custom HTTP post data.
243 Override the User-Agent header. If not specified the protocol will use a
244 string describing the libavformat build. ("Lavf/<version>")
247 Set timeout of socket I/O operations used by the underlying low level
248 operation. By default it is set to -1, which means that the timeout is
252 Export the MIME type.
255 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
256 supports this, the metadata has to be retrieved by the application by reading
257 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
260 @item icy_metadata_headers
261 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
262 headers, separated by newline characters.
264 @item icy_metadata_packet
265 If the server supports ICY metadata, and @option{icy} was set to 1, this
266 contains the last non-empty metadata packet sent by the server. It should be
267 polled in regular intervals by applications interested in mid-stream metadata
271 Set the cookies to be sent in future requests. The format of each cookie is the
272 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
273 delimited by a newline character.
276 Set initial byte offset.
279 Try to limit the request to bytes preceding this offset.
282 @subsection HTTP Cookies
284 Some HTTP requests will be denied unless cookie values are passed in with the
285 request. The @option{cookies} option allows these cookies to be specified. At
286 the very least, each cookie must specify a value along with a path and domain.
287 HTTP requests that match both the domain and path will automatically include the
288 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
291 The required syntax to play a stream specifying a cookie is:
293 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
298 MMS (Microsoft Media Server) protocol over TCP.
302 MMS (Microsoft Media Server) protocol over HTTP.
304 The required syntax is:
306 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
313 Computes the MD5 hash of the data to be written, and on close writes
314 this to the designated output or stdout if none is specified. It can
315 be used to test muxers without writing an actual file.
317 Some examples follow.
319 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
320 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
322 # Write the MD5 hash of the encoded AVI file to stdout.
323 ffmpeg -i input.flv -f avi -y md5:
326 Note that some formats (typically MOV) require the output protocol to
327 be seekable, so they will fail with the MD5 output protocol.
331 UNIX pipe access protocol.
333 Allow to read and write from UNIX pipes.
335 The accepted syntax is:
340 @var{number} is the number corresponding to the file descriptor of the
341 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
342 is not specified, by default the stdout file descriptor will be used
343 for writing, stdin for reading.
345 For example to read from stdin with @command{ffmpeg}:
347 cat test.wav | ffmpeg -i pipe:0
348 # ...this is the same as...
349 cat test.wav | ffmpeg -i pipe:
352 For writing to stdout with @command{ffmpeg}:
354 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
355 # ...this is the same as...
356 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
359 This protocol accepts the following options:
363 Set I/O operation maximum block size, in bytes. Default value is
364 @code{INT_MAX}, which results in not limiting the requested block size.
365 Setting this value reasonably low improves user termination request reaction
366 time, which is valuable if data transmission is slow.
369 Note that some formats (typically MOV), require the output protocol to
370 be seekable, so they will fail with the pipe output protocol.
374 Real-Time Messaging Protocol.
376 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
377 content across a TCP/IP network.
379 The required syntax is:
381 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
384 The accepted parameters are:
388 An optional username (mostly for publishing).
391 An optional password (mostly for publishing).
394 The address of the RTMP server.
397 The number of the TCP port to use (by default is 1935).
400 It is the name of the application to access. It usually corresponds to
401 the path where the application is installed on the RTMP server
402 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
403 the value parsed from the URI through the @code{rtmp_app} option, too.
406 It is the path or name of the resource to play with reference to the
407 application specified in @var{app}, may be prefixed by "mp4:". You
408 can override the value parsed from the URI through the @code{rtmp_playpath}
412 Act as a server, listening for an incoming connection.
415 Maximum time to wait for the incoming connection. Implies listen.
418 Additionally, the following parameters can be set via command line options
419 (or in code via @code{AVOption}s):
423 Name of application to connect on the RTMP server. This option
424 overrides the parameter specified in the URI.
427 Set the client buffer time in milliseconds. The default is 3000.
430 Extra arbitrary AMF connection parameters, parsed from a string,
431 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
432 Each value is prefixed by a single character denoting the type,
433 B for Boolean, N for number, S for string, O for object, or Z for null,
434 followed by a colon. For Booleans the data must be either 0 or 1 for
435 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
436 1 to end or begin an object, respectively. Data items in subobjects may
437 be named, by prefixing the type with 'N' and specifying the name before
438 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
439 times to construct arbitrary AMF sequences.
442 Version of the Flash plugin used to run the SWF player. The default
443 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
444 <libavformat version>).)
446 @item rtmp_flush_interval
447 Number of packets flushed in the same request (RTMPT only). The default
451 Specify that the media is a live stream. No resuming or seeking in
452 live streams is possible. The default value is @code{any}, which means the
453 subscriber first tries to play the live stream specified in the
454 playpath. If a live stream of that name is not found, it plays the
455 recorded stream. The other possible values are @code{live} and
459 URL of the web page in which the media was embedded. By default no
463 Stream identifier to play or to publish. This option overrides the
464 parameter specified in the URI.
467 Name of live stream to subscribe to. By default no value will be sent.
468 It is only sent if the option is specified or if rtmp_live
472 SHA256 hash of the decompressed SWF file (32 bytes).
475 Size of the decompressed SWF file, required for SWFVerification.
478 URL of the SWF player for the media. By default no value will be sent.
481 URL to player swf file, compute hash/size automatically.
484 URL of the target stream. Defaults to proto://host[:port]/app.
488 For example to read with @command{ffplay} a multimedia resource named
489 "sample" from the application "vod" from an RTMP server "myserver":
491 ffplay rtmp://myserver/vod/sample
494 To publish to a password protected server, passing the playpath and
495 app names separately:
497 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
502 Encrypted Real-Time Messaging Protocol.
504 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
505 streaming multimedia content within standard cryptographic primitives,
506 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
511 Real-Time Messaging Protocol over a secure SSL connection.
513 The Real-Time Messaging Protocol (RTMPS) is used for streaming
514 multimedia content across an encrypted connection.
518 Real-Time Messaging Protocol tunneled through HTTP.
520 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
521 for streaming multimedia content within HTTP requests to traverse
526 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
528 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
529 is used for streaming multimedia content within HTTP requests to traverse
534 Real-Time Messaging Protocol tunneled through HTTPS.
536 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
537 for streaming multimedia content within HTTPS requests to traverse
542 Secure File Transfer Protocol via libssh
544 Allow to read from or write to remote resources using SFTP protocol.
546 Following syntax is required.
549 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
552 This protocol accepts the following options.
556 Set timeout of socket I/O operations used by the underlying low level
557 operation. By default it is set to -1, which means that the timeout
561 Truncate existing files on write, if set to 1. A value of 0 prevents
562 truncating. Default value is 1.
565 Specify the path of the file containing private key to use during authorization.
566 By default libssh searches for keys in the @file{~/.ssh/} directory.
570 Example: Play a file stored on remote server.
573 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
576 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
578 Real-Time Messaging Protocol and its variants supported through
581 Requires the presence of the librtmp headers and library during
582 configuration. You need to explicitly configure the build with
583 "--enable-librtmp". If enabled this will replace the native RTMP
586 This protocol provides most client functions and a few server
587 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
588 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
589 variants of these encrypted types (RTMPTE, RTMPTS).
591 The required syntax is:
593 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
596 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
597 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
598 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
599 meaning as specified for the RTMP native protocol.
600 @var{options} contains a list of space-separated options of the form
603 See the librtmp manual page (man 3 librtmp) for more information.
605 For example, to stream a file in real-time to an RTMP server using
608 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
611 To play the same stream using @command{ffplay}:
613 ffplay "rtmp://myserver/live/mystream live=1"
618 Real-time Transport Protocol.
620 The required syntax for an RTP URL is:
621 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
623 @var{port} specifies the RTP port to use.
625 The following URL options are supported:
630 Set the TTL (Time-To-Live) value (for multicast only).
632 @item rtcpport=@var{n}
633 Set the remote RTCP port to @var{n}.
635 @item localrtpport=@var{n}
636 Set the local RTP port to @var{n}.
638 @item localrtcpport=@var{n}'
639 Set the local RTCP port to @var{n}.
641 @item pkt_size=@var{n}
642 Set max packet size (in bytes) to @var{n}.
645 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
648 @item sources=@var{ip}[,@var{ip}]
649 List allowed source IP addresses.
651 @item block=@var{ip}[,@var{ip}]
652 List disallowed (blocked) source IP addresses.
654 @item write_to_source=0|1
655 Send packets to the source address of the latest received packet (if
656 set to 1) or to a default remote address (if set to 0).
658 @item localport=@var{n}
659 Set the local RTP port to @var{n}.
661 This is a deprecated option. Instead, @option{localrtpport} should be
671 If @option{rtcpport} is not set the RTCP port will be set to the RTP
675 If @option{localrtpport} (the local RTP port) is not set any available
676 port will be used for the local RTP and RTCP ports.
679 If @option{localrtcpport} (the local RTCP port) is not set it will be
680 set to the the local RTP port value plus 1.
685 Real-Time Streaming Protocol.
687 RTSP is not technically a protocol handler in libavformat, it is a demuxer
688 and muxer. The demuxer supports both normal RTSP (with data transferred
689 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
690 data transferred over RDT).
692 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
693 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
694 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
696 The required syntax for a RTSP url is:
698 rtsp://@var{hostname}[:@var{port}]/@var{path}
701 Options can be set on the @command{ffmpeg}/@command{ffplay} command
702 line, or set in code via @code{AVOption}s or in
703 @code{avformat_open_input}.
705 The following options are supported.
709 Do not start playing the stream immediately if set to 1. Default value
713 Set RTSP trasport protocols.
715 It accepts the following values:
718 Use UDP as lower transport protocol.
721 Use TCP (interleaving within the RTSP control channel) as lower
725 Use UDP multicast as lower transport protocol.
728 Use HTTP tunneling as lower transport protocol, which is useful for
732 Multiple lower transport protocols may be specified, in that case they are
733 tried one at a time (if the setup of one fails, the next one is tried).
734 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
739 The following values are accepted:
742 Accept packets only from negotiated peer address and port.
744 Act as a server, listening for an incoming connection.
746 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
749 Default value is @samp{none}.
751 @item allowed_media_types
752 Set media types to accept from the server.
754 The following flags are accepted:
761 By default it accepts all media types.
764 Set minimum local UDP port. Default value is 5000.
767 Set maximum local UDP port. Default value is 65000.
770 Set maximum timeout (in seconds) to wait for incoming connections.
772 A value of -1 mean infinite (default). This option implies the
773 @option{rtsp_flags} set to @samp{listen}.
775 @item reorder_queue_size
776 Set number of packets to buffer for handling of reordered packets.
779 Set socket TCP I/O timeout in micro seconds.
782 Override User-Agent header. If not specified, it default to the
783 libavformat identifier string.
786 When receiving data over UDP, the demuxer tries to reorder received packets
787 (since they may arrive out of order, or packets may get lost totally). This
788 can be disabled by setting the maximum demuxing delay to zero (via
789 the @code{max_delay} field of AVFormatContext).
791 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
792 streams to display can be chosen with @code{-vst} @var{n} and
793 @code{-ast} @var{n} for video and audio respectively, and can be switched
794 on the fly by pressing @code{v} and @code{a}.
798 The following examples all make use of the @command{ffplay} and
799 @command{ffmpeg} tools.
803 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
805 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
809 Watch a stream tunneled over HTTP:
811 ffplay -rtsp_transport http rtsp://server/video.mp4
815 Send a stream in realtime to a RTSP server, for others to watch:
817 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
821 Receive a stream in realtime:
823 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
829 Session Announcement Protocol (RFC 2974). This is not technically a
830 protocol handler in libavformat, it is a muxer and demuxer.
831 It is used for signalling of RTP streams, by announcing the SDP for the
832 streams regularly on a separate port.
836 The syntax for a SAP url given to the muxer is:
838 sap://@var{destination}[:@var{port}][?@var{options}]
841 The RTP packets are sent to @var{destination} on port @var{port},
842 or to port 5004 if no port is specified.
843 @var{options} is a @code{&}-separated list. The following options
848 @item announce_addr=@var{address}
849 Specify the destination IP address for sending the announcements to.
850 If omitted, the announcements are sent to the commonly used SAP
851 announcement multicast address 224.2.127.254 (sap.mcast.net), or
852 ff0e::2:7ffe if @var{destination} is an IPv6 address.
854 @item announce_port=@var{port}
855 Specify the port to send the announcements on, defaults to
856 9875 if not specified.
859 Specify the time to live value for the announcements and RTP packets,
862 @item same_port=@var{0|1}
863 If set to 1, send all RTP streams on the same port pair. If zero (the
864 default), all streams are sent on unique ports, with each stream on a
865 port 2 numbers higher than the previous.
866 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
867 The RTP stack in libavformat for receiving requires all streams to be sent
871 Example command lines follow.
873 To broadcast a stream on the local subnet, for watching in VLC:
876 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
879 Similarly, for watching in @command{ffplay}:
882 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
885 And for watching in @command{ffplay}, over IPv6:
888 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
893 The syntax for a SAP url given to the demuxer is:
895 sap://[@var{address}][:@var{port}]
898 @var{address} is the multicast address to listen for announcements on,
899 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
900 is the port that is listened on, 9875 if omitted.
902 The demuxers listens for announcements on the given address and port.
903 Once an announcement is received, it tries to receive that particular stream.
905 Example command lines follow.
907 To play back the first stream announced on the normal SAP multicast address:
913 To play back the first stream announced on one the default IPv6 SAP multicast address:
916 ffplay sap://[ff0e::2:7ffe]
921 Stream Control Transmission Protocol.
923 The accepted URL syntax is:
925 sctp://@var{host}:@var{port}[?@var{options}]
928 The protocol accepts the following options:
931 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
934 Set the maximum number of streams. By default no limit is set.
939 Secure Real-time Transport Protocol.
941 The accepted options are:
945 Select input and output encoding suites.
949 @item AES_CM_128_HMAC_SHA1_80
950 @item SRTP_AES128_CM_HMAC_SHA1_80
951 @item AES_CM_128_HMAC_SHA1_32
952 @item SRTP_AES128_CM_HMAC_SHA1_32
956 @item srtp_out_params
957 Set input and output encoding parameters, which are expressed by a
958 base64-encoded representation of a binary block. The first 16 bytes of
959 this binary block are used as master key, the following 14 bytes are
965 Virtually extract a segment of a file or another stream.
966 The underlying stream must be seekable.
971 Start offset of the extracted segment, in bytes.
973 End offset of the extracted segment, in bytes.
978 Extract a chapter from a DVD VOB file (start and end sectors obtained
979 externally and multiplied by 2048):
981 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
984 Play an AVI file directly from a TAR archive:
985 subfile,,start,183241728,end,366490624,,:archive.tar
989 Transmission Control Protocol.
991 The required syntax for a TCP url is:
993 tcp://@var{hostname}:@var{port}[?@var{options}]
996 @var{options} contains a list of &-separated options of the form
999 The list of supported options follows.
1002 @item listen=@var{1|0}
1003 Listen for an incoming connection. Default value is 0.
1005 @item timeout=@var{microseconds}
1006 Set raise error timeout, expressed in microseconds.
1008 This option is only relevant in read mode: if no data arrived in more
1009 than this time interval, raise error.
1011 @item listen_timeout=@var{microseconds}
1012 Set listen timeout, expressed in microseconds.
1015 The following example shows how to setup a listening TCP connection
1016 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1018 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1019 ffplay tcp://@var{hostname}:@var{port}
1024 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1026 The required syntax for a TLS/SSL url is:
1028 tls://@var{hostname}:@var{port}[?@var{options}]
1031 The following parameters can be set via command line options
1032 (or in code via @code{AVOption}s):
1036 @item ca_file, cafile=@var{filename}
1037 A file containing certificate authority (CA) root certificates to treat
1038 as trusted. If the linked TLS library contains a default this might not
1039 need to be specified for verification to work, but not all libraries and
1040 setups have defaults built in.
1041 The file must be in OpenSSL PEM format.
1043 @item tls_verify=@var{1|0}
1044 If enabled, try to verify the peer that we are communicating with.
1045 Note, if using OpenSSL, this currently only makes sure that the
1046 peer certificate is signed by one of the root certificates in the CA
1047 database, but it does not validate that the certificate actually
1048 matches the host name we are trying to connect to. (With GnuTLS,
1049 the host name is validated as well.)
1051 This is disabled by default since it requires a CA database to be
1052 provided by the caller in many cases.
1054 @item cert_file, cert=@var{filename}
1055 A file containing a certificate to use in the handshake with the peer.
1056 (When operating as server, in listen mode, this is more often required
1057 by the peer, while client certificates only are mandated in certain
1060 @item key_file, key=@var{filename}
1061 A file containing the private key for the certificate.
1063 @item listen=@var{1|0}
1064 If enabled, listen for connections on the provided port, and assume
1065 the server role in the handshake instead of the client role.
1069 Example command lines:
1071 To create a TLS/SSL server that serves an input stream.
1074 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1077 To play back a stream from the TLS/SSL server using @command{ffplay}:
1080 ffplay tls://@var{hostname}:@var{port}
1085 User Datagram Protocol.
1087 The required syntax for an UDP URL is:
1089 udp://@var{hostname}:@var{port}[?@var{options}]
1092 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1094 In case threading is enabled on the system, a circular buffer is used
1095 to store the incoming data, which allows one to reduce loss of data due to
1096 UDP socket buffer overruns. The @var{fifo_size} and
1097 @var{overrun_nonfatal} options are related to this buffer.
1099 The list of supported options follows.
1102 @item buffer_size=@var{size}
1103 Set the UDP maximum socket buffer size in bytes. This is used to set either
1104 the receive or send buffer size, depending on what the socket is used for.
1105 Default is 64KB. See also @var{fifo_size}.
1107 @item localport=@var{port}
1108 Override the local UDP port to bind with.
1110 @item localaddr=@var{addr}
1111 Choose the local IP address. This is useful e.g. if sending multicast
1112 and the host has multiple interfaces, where the user can choose
1113 which interface to send on by specifying the IP address of that interface.
1115 @item pkt_size=@var{size}
1116 Set the size in bytes of UDP packets.
1118 @item reuse=@var{1|0}
1119 Explicitly allow or disallow reusing UDP sockets.
1122 Set the time to live value (for multicast only).
1124 @item connect=@var{1|0}
1125 Initialize the UDP socket with @code{connect()}. In this case, the
1126 destination address can't be changed with ff_udp_set_remote_url later.
1127 If the destination address isn't known at the start, this option can
1128 be specified in ff_udp_set_remote_url, too.
1129 This allows finding out the source address for the packets with getsockname,
1130 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1131 unreachable" is received.
1132 For receiving, this gives the benefit of only receiving packets from
1133 the specified peer address/port.
1135 @item sources=@var{address}[,@var{address}]
1136 Only receive packets sent to the multicast group from one of the
1137 specified sender IP addresses.
1139 @item block=@var{address}[,@var{address}]
1140 Ignore packets sent to the multicast group from the specified
1141 sender IP addresses.
1143 @item fifo_size=@var{units}
1144 Set the UDP receiving circular buffer size, expressed as a number of
1145 packets with size of 188 bytes. If not specified defaults to 7*4096.
1147 @item overrun_nonfatal=@var{1|0}
1148 Survive in case of UDP receiving circular buffer overrun. Default
1151 @item timeout=@var{microseconds}
1152 Set raise error timeout, expressed in microseconds.
1154 This option is only relevant in read mode: if no data arrived in more
1155 than this time interval, raise error.
1157 @item broadcast=@var{1|0}
1158 Explicitly allow or disallow UDP broadcasting.
1160 Note that broadcasting may not work properly on networks having
1161 a broadcast storm protection.
1164 @subsection Examples
1168 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1170 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1174 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1175 sized UDP packets, using a large input buffer:
1177 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1181 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1183 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1191 The required syntax for a Unix socket URL is:
1194 unix://@var{filepath}
1197 The following parameters can be set via command line options
1198 (or in code via @code{AVOption}s):
1204 Create the Unix socket in listening mode.
1207 @c man end PROTOCOLS