Provided by: ffmpeg_4.1.4-1build2_amd64 bug

NAME

       ffmpeg-protocols - FFmpeg protocols

DESCRIPTION

       This document describes the input and output protocols provided by the libavformat
       library.

PROTOCOL OPTIONS

       The libavformat library provides some generic global options, which can be set on all the
       protocols. In addition each protocol may support so-called private options, which are
       specific for that component.

       Options may be set by specifying -option value in the FFmpeg tools, or by setting the
       value explicitly in the "AVFormatContext" options or using the libavutil/opt.h API for
       programmatic use.

       The list of supported options follows:

       protocol_whitelist list (input)
           Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
           prefixed by "-" are disabled.  All protocols are allowed by default but protocols used
           by an another protocol (nested protocols) are restricted to a per protocol subset.

PROTOCOLS

       Protocols are configured elements in FFmpeg that enable access to resources that require
       specific protocols.

       When you configure your FFmpeg build, all the supported protocols are enabled by default.
       You can list all available ones using the configure option "--list-protocols".

       You can disable all the protocols using the configure option "--disable-protocols", and
       selectively enable a protocol using the option "--enable-protocol=PROTOCOL", or you can
       disable a particular protocol using the option "--disable-protocol=PROTOCOL".

       The option "-protocols" of the ff* tools will display the list of supported protocols.

       All protocols accept the following options:

       rw_timeout
           Maximum time to wait for (network) read/write operations to complete, in microseconds.

       A description of the currently available protocols follows.

   async
       Asynchronous data filling wrapper for input stream.

       Fill data in a background thread, to decouple I/O operation from demux thread.

               async:<URL>
               async:http://host/resource
               async:cache:http://host/resource

   bluray
       Read BluRay playlist.

       The accepted options are:

       angle
           BluRay angle

       chapter
           Start chapter (1...N)

       playlist
           Playlist to read (BDMV/PLAYLIST/?????.mpls)

       Examples:

       Read longest playlist from BluRay mounted to /mnt/bluray:

               bluray:/mnt/bluray

       Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:

               -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

   cache
       Caching wrapper for input stream.

       Cache the input stream to temporary file. It brings seeking capability to live streams.

               cache:<URL>

   concat
       Physical concatenation protocol.

       Read and seek from many resources in sequence as if they were a unique resource.

       A URL accepted by this protocol has the syntax:

               concat:<URL1>|<URL2>|...|<URLN>

       where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one
       possibly specifying a distinct protocol.

       For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay
       use the command:

               ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

       Note that you may need to escape the character "|" which is special for many shells.

   crypto
       AES-encrypted stream reading protocol.

       The accepted options are:

       key Set the AES decryption key binary block from given hexadecimal representation.

       iv  Set the AES decryption initialization vector binary block from given hexadecimal
           representation.

       Accepted URL formats:

               crypto:<URL>
               crypto+<URL>

   data
       Data in-line in the URI. See <http://en.wikipedia.org/wiki/Data_URI_scheme>.

       For example, to convert a GIF file given inline with ffmpeg:

               ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

   file
       File access protocol.

       Read from or write to a file.

       A file URL can have the form:

               file:<filename>

       where filename is the path of the file to read.

       An URL that does not have a protocol prefix will be assumed to be a file URL. Depending on
       the build, an URL that looks like a Windows path with the drive letter at the beginning
       will also be assumed to be a file URL (usually not the case in builds for unix-like
       systems).

       For example to read from a file input.mpeg with ffmpeg use the command:

               ffmpeg -i file:input.mpeg output.mpeg

       This protocol accepts the following options:

       truncate
           Truncate existing files on write, if set to 1. A value of 0 prevents truncating.
           Default value is 1.

       blocksize
           Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which
           results in not limiting the requested block size.  Setting this value reasonably low
           improves user termination request reaction time, which is valuable for files on slow
           medium.

   ftp
       FTP (File Transfer Protocol).

       Read from or write to remote resources using FTP protocol.

       Following syntax is required.

               ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

       This protocol accepts the following options.

       timeout
           Set timeout in microseconds of socket I/O operations used by the underlying low level
           operation. By default it is set to -1, which means that the timeout is not specified.

       ftp-anonymous-password
           Password used when login as anonymous user. Typically an e-mail address should be
           used.

       ftp-write-seekable
           Control seekability of connection during encoding. If set to 1 the resource is
           supposed to be seekable, if set to 0 it is assumed not to be seekable. Default value
           is 0.

       NOTE: Protocol can be used as output, but it is recommended to not do it, unless special
       care is taken (tests, customized server configuration etc.). Different FTP servers behave
       in different way during seek operation. ff* tools may produce incomplete content due to
       server limitations.

       This protocol accepts the following options:

       follow
           If set to 1, the protocol will retry reading at the end of the file, allowing reading
           files that still are being written. In order for this to terminate, you either need to
           use the rw_timeout option, or use the interrupt callback (for API users).

   gopher
       Gopher protocol.

   hls
       Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8
       playlists describing the segments can be remote HTTP resources or local files, accessed
       using the standard file protocol.  The nested protocol is declared by specifying "+proto"
       after the hls URI scheme name, where proto is either "file" or "http".

               hls+http://host/path/to/remote/resource.m3u8
               hls+file://path/to/local/resource.m3u8

       Using this protocol is discouraged - the hls demuxer should work just as well (if not,
       please report the issues) and is more complete.  To use the hls demuxer instead, simply
       use the direct URLs to the m3u8 files.

   http
       HTTP (Hyper Text Transfer Protocol).

       This protocol accepts the following options:

       seekable
           Control seekability of connection. If set to 1 the resource is supposed to be
           seekable, if set to 0 it is assumed not to be seekable, if set to -1 it will try to
           autodetect if it is seekable. Default value is -1.

       chunked_post
           If set to 1 use chunked Transfer-Encoding for posts, default is 1.

       content_type
           Set a specific content type for the POST messages or for listen mode.

       http_proxy
           set HTTP proxy to tunnel through e.g. http://example.com:1234

       headers
           Set custom HTTP headers, can override built in default headers. The value must be a
           string encoding the headers.

       multiple_requests
           Use persistent connections if set to 1, default is 0.

       post_data
           Set custom HTTP post data.

       referer
           Set the Referer header. Include 'Referer: URL' header in HTTP request.

       user_agent
           Override the User-Agent header. If not specified the protocol will use a string
           describing the libavformat build. ("Lavf/<version>")

       user-agent
           This is a deprecated option, you can use user_agent instead it.

       timeout
           Set timeout in microseconds of socket I/O operations used by the underlying low level
           operation. By default it is set to -1, which means that the timeout is not specified.

       reconnect_at_eof
           If set then eof is treated like an error and causes reconnection, this is useful for
           live / endless streams.

       reconnect_streamed
           If set then even streamed/non seekable streams will be reconnected on errors.

       reconnect_delay_max
           Sets the maximum delay in seconds after which to give up reconnecting

       mime_type
           Export the MIME type.

       http_version
           Exports the HTTP response version number. Usually "1.0" or "1.1".

       icy If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports
           this, the metadata has to be retrieved by the application by reading the
           icy_metadata_headers and icy_metadata_packet options.  The default is 1.

       icy_metadata_headers
           If the server supports ICY metadata, this contains the ICY-specific HTTP reply
           headers, separated by newline characters.

       icy_metadata_packet
           If the server supports ICY metadata, and icy was set to 1, this contains the last non-
           empty metadata packet sent by the server. It should be polled in regular intervals by
           applications interested in mid-stream metadata updates.

       cookies
           Set the cookies to be sent in future requests. The format of each cookie is the same
           as the value of a Set-Cookie HTTP response field. Multiple cookies can be delimited by
           a newline character.

       offset
           Set initial byte offset.

       end_offset
           Try to limit the request to bytes preceding this offset.

       method
           When used as a client option it sets the HTTP method for the request.

           When used as a server option it sets the HTTP method that is going to be expected from
           the client(s).  If the expected and the received HTTP method do not match the client
           will be given a Bad Request response.  When unset the HTTP method is not checked for
           now. This will be replaced by autodetection in the future.

       listen
           If set to 1 enables experimental HTTP server. This can be used to send data when used
           as an output option, or read data from a client with HTTP POST when used as an input
           option.  If set to 2 enables experimental multi-client HTTP server. This is not yet
           implemented in ffmpeg.c and thus must not be used as a command line option.

                   # Server side (sending):
                   ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>

                   # Client side (receiving):
                   ffmpeg -i http://<server>:<port> -c copy somefile.ogg

                   # Client can also be done with wget:
                   wget http://<server>:<port> -O somefile.ogg

                   # Server side (receiving):
                   ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg

                   # Client side (sending):
                   ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>

                   # Client can also be done with wget:
                   wget --post-file=somefile.ogg http://<server>:<port>

       HTTP Cookies

       Some HTTP requests will be denied unless cookie values are passed in with the request. The
       cookies option allows these cookies to be specified. At the very least, each cookie must
       specify a value along with a path and domain.  HTTP requests that match both the domain
       and path will automatically include the cookie value in the HTTP Cookie header field.
       Multiple cookies can be delimited by a newline.

       The required syntax to play a stream specifying a cookie is:

               ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8

   Icecast
       Icecast protocol (stream to Icecast servers)

       This protocol accepts the following options:

       ice_genre
           Set the stream genre.

       ice_name
           Set the stream name.

       ice_description
           Set the stream description.

       ice_url
           Set the stream website URL.

       ice_public
           Set if the stream should be public.  The default is 0 (not public).

       user_agent
           Override the User-Agent header. If not specified a string of the form "Lavf/<version>"
           will be used.

       password
           Set the Icecast mountpoint password.

       content_type
           Set the stream content type. This must be set if it is different from audio/mpeg.

       legacy_icecast
           This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT
           method but the SOURCE method.

               icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>

   mmst
       MMS (Microsoft Media Server) protocol over TCP.

   mmsh
       MMS (Microsoft Media Server) protocol over HTTP.

       The required syntax is:

               mmsh://<server>[:<port>][/<app>][/<playpath>]

   md5
       MD5 output protocol.

       Computes the MD5 hash of the data to be written, and on close writes this to the
       designated output or stdout if none is specified. It can be used to test muxers without
       writing an actual file.

       Some examples follow.

               # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
               ffmpeg -i input.flv -f avi -y md5:output.avi.md5

               # Write the MD5 hash of the encoded AVI file to stdout.
               ffmpeg -i input.flv -f avi -y md5:

       Note that some formats (typically MOV) require the output protocol to be seekable, so they
       will fail with the MD5 output protocol.

   pipe
       UNIX pipe access protocol.

       Read and write from UNIX pipes.

       The accepted syntax is:

               pipe:[<number>]

       number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1
       for stdout, 2 for stderr).  If number is not specified, by default the stdout file
       descriptor will be used for writing, stdin for reading.

       For example to read from stdin with ffmpeg:

               cat test.wav | ffmpeg -i pipe:0
               # ...this is the same as...
               cat test.wav | ffmpeg -i pipe:

       For writing to stdout with ffmpeg:

               ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
               # ...this is the same as...
               ffmpeg -i test.wav -f avi pipe: | cat > test.avi

       This protocol accepts the following options:

       blocksize
           Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which
           results in not limiting the requested block size.  Setting this value reasonably low
           improves user termination request reaction time, which is valuable if data
           transmission is slow.

       Note that some formats (typically MOV), require the output protocol to be seekable, so
       they will fail with the pipe output protocol.

   prompeg
       Pro-MPEG Code of Practice #3 Release 2 FEC protocol.

       The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2
       Transport Streams sent over RTP.

       This protocol must be used in conjunction with the "rtp_mpegts" muxer and the "rtp"
       protocol.

       The required syntax is:

               -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>

       The destination UDP ports are "port + 2" for the column FEC stream and "port + 4" for the
       row FEC stream.

       This protocol accepts the following options:

       l=n The number of columns (4-20, LxD <= 100)

       d=n The number of rows (4-20, LxD <= 100)

       Example usage:

               -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>

   rtmp
       Real-Time Messaging Protocol.

       The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a
       TCP/IP network.

       The required syntax is:

               rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]

       The accepted parameters are:

       username
           An optional username (mostly for publishing).

       password
           An optional password (mostly for publishing).

       server
           The address of the RTMP server.

       port
           The number of the TCP port to use (by default is 1935).

       app It is the name of the application to access. It usually corresponds to the path where
           the application is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.).
           You can override the value parsed from the URI through the "rtmp_app" option, too.

       playpath
           It is the path or name of the resource to play with reference to the application
           specified in app, may be prefixed by "mp4:". You can override the value parsed from
           the URI through the "rtmp_playpath" option, too.

       listen
           Act as a server, listening for an incoming connection.

       timeout
           Maximum time to wait for the incoming connection. Implies listen.

       Additionally, the following parameters can be set via command line options (or in code via
       "AVOption"s):

       rtmp_app
           Name of application to connect on the RTMP server. This option overrides the parameter
           specified in the URI.

       rtmp_buffer
           Set the client buffer time in milliseconds. The default is 3000.

       rtmp_conn
           Extra arbitrary AMF connection parameters, parsed from a string, e.g. like "B:1
           S:authMe O:1 NN:code:1.23 NS:flag:ok O:0".  Each value is prefixed by a single
           character denoting the type, B for Boolean, N for number, S for string, O for object,
           or Z for null, followed by a colon. For Booleans the data must be either 0 or 1 for
           FALSE or TRUE, respectively.  Likewise for Objects the data must be 0 or 1 to end or
           begin an object, respectively. Data items in subobjects may be named, by prefixing the
           type with 'N' and specifying the name before the value (i.e. "NB:myFlag:1"). This
           option may be used multiple times to construct arbitrary AMF sequences.

       rtmp_flashver
           Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2.
           (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)

       rtmp_flush_interval
           Number of packets flushed in the same request (RTMPT only). The default is 10.

       rtmp_live
           Specify that the media is a live stream. No resuming or seeking in live streams is
           possible. The default value is "any", which means the subscriber first tries to play
           the live stream specified in the playpath. If a live stream of that name is not found,
           it plays the recorded stream. The other possible values are "live" and "recorded".

       rtmp_pageurl
           URL of the web page in which the media was embedded. By default no value will be sent.

       rtmp_playpath
           Stream identifier to play or to publish. This option overrides the parameter specified
           in the URI.

       rtmp_subscribe
           Name of live stream to subscribe to. By default no value will be sent.  It is only
           sent if the option is specified or if rtmp_live is set to live.

       rtmp_swfhash
           SHA256 hash of the decompressed SWF file (32 bytes).

       rtmp_swfsize
           Size of the decompressed SWF file, required for SWFVerification.

       rtmp_swfurl
           URL of the SWF player for the media. By default no value will be sent.

       rtmp_swfverify
           URL to player swf file, compute hash/size automatically.

       rtmp_tcurl
           URL of the target stream. Defaults to proto://host[:port]/app.

       For example to read with ffplay a multimedia resource named "sample" from the application
       "vod" from an RTMP server "myserver":

               ffplay rtmp://myserver/vod/sample

       To publish to a password protected server, passing the playpath and app names separately:

               ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

   rtmpe
       Encrypted Real-Time Messaging Protocol.

       The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia
       content within standard cryptographic primitives, consisting of Diffie-Hellman key
       exchange and HMACSHA256, generating a pair of RC4 keys.

   rtmps
       Real-Time Messaging Protocol over a secure SSL connection.

       The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across
       an encrypted connection.

   rtmpt
       Real-Time Messaging Protocol tunneled through HTTP.

       The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming
       multimedia content within HTTP requests to traverse firewalls.

   rtmpte
       Encrypted Real-Time Messaging Protocol tunneled through HTTP.

       The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for
       streaming multimedia content within HTTP requests to traverse firewalls.

   rtmpts
       Real-Time Messaging Protocol tunneled through HTTPS.

       The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming
       multimedia content within HTTPS requests to traverse firewalls.

   libsmbclient
       libsmbclient permits one to manipulate CIFS/SMB network resources.

       Following syntax is required.

               smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

       This protocol accepts the following options.

       timeout
           Set timeout in milliseconds of socket I/O operations used by the underlying low level
           operation. By default it is set to -1, which means that the timeout is not specified.

       truncate
           Truncate existing files on write, if set to 1. A value of 0 prevents truncating.
           Default value is 1.

       workgroup
           Set the workgroup used for making connections. By default workgroup is not specified.

       For more information see: <http://www.samba.org/>.

   libssh
       Secure File Transfer Protocol via libssh

       Read from or write to remote resources using SFTP protocol.

       Following syntax is required.

               sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

       This protocol accepts the following options.

       timeout
           Set timeout of socket I/O operations used by the underlying low level operation. By
           default it is set to -1, which means that the timeout is not specified.

       truncate
           Truncate existing files on write, if set to 1. A value of 0 prevents truncating.
           Default value is 1.

       private_key
           Specify the path of the file containing private key to use during authorization.  By
           default libssh searches for keys in the ~/.ssh/ directory.

       Example: Play a file stored on remote server.

               ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

   librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
       Real-Time Messaging Protocol and its variants supported through librtmp.

       Requires the presence of the librtmp headers and library during configuration. You need to
       explicitly configure the build with "--enable-librtmp". If enabled this will replace the
       native RTMP protocol.

       This protocol provides most client functions and a few server functions needed to support
       RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and
       tunneled variants of these encrypted types (RTMPTE, RTMPTS).

       The required syntax is:

               <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>

       where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte",
       "rtmpts" corresponding to each RTMP variant, and server, port, app and playpath have the
       same meaning as specified for the RTMP native protocol.  options contains a list of space-
       separated options of the form key=val.

       See the librtmp manual page (man 3 librtmp) for more information.

       For example, to stream a file in real-time to an RTMP server using ffmpeg:

               ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

       To play the same stream using ffplay:

               ffplay "rtmp://myserver/live/mystream live=1"

   rtp
       Real-time Transport Protocol.

       The required syntax for an RTP URL is: rtp://hostname[:port][?option=val...]

       port specifies the RTP port to use.

       The following URL options are supported:

       ttl=n
           Set the TTL (Time-To-Live) value (for multicast only).

       rtcpport=n
           Set the remote RTCP port to n.

       localrtpport=n
           Set the local RTP port to n.

       localrtcpport=n'
           Set the local RTCP port to n.

       pkt_size=n
           Set max packet size (in bytes) to n.

       connect=0|1
           Do a "connect()" on the UDP socket (if set to 1) or not (if set to 0).

       sources=ip[,ip]
           List allowed source IP addresses.

       block=ip[,ip]
           List disallowed (blocked) source IP addresses.

       write_to_source=0|1
           Send packets to the source address of the latest received packet (if set to 1) or to a
           default remote address (if set to 0).

       localport=n
           Set the local RTP port to n.

           This is a deprecated option. Instead, localrtpport should be used.

       Important notes:

       1.  If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.

       2.  If localrtpport (the local RTP port) is not set any available port will be used for
           the local RTP and RTCP ports.

       3.  If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port
           value plus 1.

   rtsp
       Real-Time Streaming Protocol.

       RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The
       demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g.
       Apple and Microsoft) and Real-RTSP (with data transferred over RDT).

       The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it
       (currently Darwin Streaming Server and Mischa Spiegelmock's
       <https://github.com/revmischa/rtsp-server>).

       The required syntax for a RTSP url is:

               rtsp://<hostname>[:<port>]/<path>

       Options can be set on the ffmpeg/ffplay command line, or set in code via "AVOption"s or in
       "avformat_open_input".

       The following options are supported.

       initial_pause
           Do not start playing the stream immediately if set to 1. Default value is 0.

       rtsp_transport
           Set RTSP transport protocols.

           It accepts the following values:

           udp Use UDP as lower transport protocol.

           tcp Use TCP (interleaving within the RTSP control channel) as lower transport
               protocol.

           udp_multicast
               Use UDP multicast as lower transport protocol.

           http
               Use HTTP tunneling as lower transport protocol, which is useful for passing
               proxies.

           Multiple lower transport protocols may be specified, in that case they are tried one
           at a time (if the setup of one fails, the next one is tried).  For the muxer, only the
           tcp and udp options are supported.

       rtsp_flags
           Set RTSP flags.

           The following values are accepted:

           filter_src
               Accept packets only from negotiated peer address and port.

           listen
               Act as a server, listening for an incoming connection.

           prefer_tcp
               Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.

           Default value is none.

       allowed_media_types
           Set media types to accept from the server.

           The following flags are accepted:

           video
           audio
           data

           By default it accepts all media types.

       min_port
           Set minimum local UDP port. Default value is 5000.

       max_port
           Set maximum local UDP port. Default value is 65000.

       timeout
           Set maximum timeout (in seconds) to wait for incoming connections.

           A value of -1 means infinite (default). This option implies the rtsp_flags set to
           listen.

       reorder_queue_size
           Set number of packets to buffer for handling of reordered packets.

       stimeout
           Set socket TCP I/O timeout in microseconds.

       user-agent
           Override User-Agent header. If not specified, it defaults to the libavformat
           identifier string.

       When receiving data over UDP, the demuxer tries to reorder received packets (since they
       may arrive out of order, or packets may get lost totally). This can be disabled by setting
       the maximum demuxing delay to zero (via the "max_delay" field of AVFormatContext).

       When watching multi-bitrate Real-RTSP streams with ffplay, the streams to display can be
       chosen with "-vst" n and "-ast" n for video and audio respectively, and can be switched on
       the fly by pressing "v" and "a".

       Examples

       The following examples all make use of the ffplay and ffmpeg tools.

       ·   Watch a stream over UDP, with a max reordering delay of 0.5 seconds:

                   ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

       ·   Watch a stream tunneled over HTTP:

                   ffplay -rtsp_transport http rtsp://server/video.mp4

       ·   Send a stream in realtime to a RTSP server, for others to watch:

                   ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

       ·   Receive a stream in realtime:

                   ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

   sap
       Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in
       libavformat, it is a muxer and demuxer.  It is used for signalling of RTP streams, by
       announcing the SDP for the streams regularly on a separate port.

       Muxer

       The syntax for a SAP url given to the muxer is:

               sap://<destination>[:<port>][?<options>]

       The RTP packets are sent to destination on port port, or to port 5004 if no port is
       specified.  options is a "&"-separated list. The following options are supported:

       announce_addr=address
           Specify the destination IP address for sending the announcements to.  If omitted, the
           announcements are sent to the commonly used SAP announcement multicast address
           224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.

       announce_port=port
           Specify the port to send the announcements on, defaults to 9875 if not specified.

       ttl=ttl
           Specify the time to live value for the announcements and RTP packets, defaults to 255.

       same_port=0|1
           If set to 1, send all RTP streams on the same port pair. If zero (the default), all
           streams are sent on unique ports, with each stream on a port 2 numbers higher than the
           previous.  VLC/Live555 requires this to be set to 1, to be able to receive the stream.
           The RTP stack in libavformat for receiving requires all streams to be sent on unique
           ports.

       Example command lines follow.

       To broadcast a stream on the local subnet, for watching in VLC:

               ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

       Similarly, for watching in ffplay:

               ffmpeg -re -i <input> -f sap sap://224.0.0.255

       And for watching in ffplay, over IPv6:

               ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

       Demuxer

       The syntax for a SAP url given to the demuxer is:

               sap://[<address>][:<port>]

       address is the multicast address to listen for announcements on, if omitted, the default
       224.2.127.254 (sap.mcast.net) is used. port is the port that is listened on, 9875 if
       omitted.

       The demuxers listens for announcements on the given address and port.  Once an
       announcement is received, it tries to receive that particular stream.

       Example command lines follow.

       To play back the first stream announced on the normal SAP multicast address:

               ffplay sap://

       To play back the first stream announced on one the default IPv6 SAP multicast address:

               ffplay sap://[ff0e::2:7ffe]

   sctp
       Stream Control Transmission Protocol.

       The accepted URL syntax is:

               sctp://<host>:<port>[?<options>]

       The protocol accepts the following options:

       listen
           If set to any value, listen for an incoming connection. Outgoing connection is done by
           default.

       max_streams
           Set the maximum number of streams. By default no limit is set.

   srt
       Haivision Secure Reliable Transport Protocol via libsrt.

       The supported syntax for a SRT URL is:

               srt://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       or

               <options> srt://<hostname>:<port>

       options contains a list of '-key val' options.

       This protocol accepts the following options.

       connect_timeout
           Connection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges)
           with the default connect timeout of 3 seconds. This option applies to the caller and
           rendezvous connection modes. The connect timeout is 10 times the value set for the
           rendezvous mode (which can be used as a workaround for this connection problem with
           earlier versions).

       ffs=bytes
           Flight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and
           you should set it to not less than recv_buffer_size and mss. The default value is
           relatively large, therefore unless you set a very large receiver buffer, you do not
           need to change this option. Default value is 25600.

       inputbw=bytes/seconds
           Sender nominal input rate, in bytes per seconds. Used along with oheadbw, when maxbw
           is set to relative (0), to calculate maximum sending rate when recovery packets are
           sent along with the main media stream: inputbw * (100 + oheadbw) / 100 if inputbw is
           not set while maxbw is set to relative (0), the actual input rate is evaluated inside
           the library. Default value is 0.

       iptos=tos
           IP Type of Service. Applies to sender only. Default value is 0xB8.

       ipttl=ttl
           IP Time To Live. Applies to sender only. Default value is 64.

       latency
           Timestamp-based Packet Delivery Delay.  Used to absorb bursts of missed packet
           retransmissions.  This flag sets both rcvlatency and peerlatency to the same value.
           Note that prior to version 1.3.0 this is the only flag to set the latency, however
           this is effectively equivalent to setting peerlatency, when side is sender and
           rcvlatency when side is receiver, and the bidirectional stream sending is not
           supported.

       listen_timeout
           Set socket listen timeout.

       maxbw=bytes/seconds
           Maximum sending bandwidth, in bytes per seconds.  -1 infinite (CSRTCC limit is 30mbps)
           0 relative to input rate (see inputbw) >0 absolute limit value Default value is 0
           (relative)

       mode=caller|listener|rendezvous
           Connection mode.  caller opens client connection.  listener starts server to listen
           for incoming connections.  rendezvous use Rendez-Vous connection mode.  Default value
           is caller.

       mss=bytes
           Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using
           a packet counter assuming fully filled packets. The smallest MSS between the peers is
           used. This is 1500 by default in the overall internet.  This is the maximum size of
           the UDP packet and can be only decreased, unless you have some unusual dedicated
           network settings. Default value is 1500.

       nakreport=1|0
           If set to 1, Receiver will send `UMSG_LOSSREPORT` messages periodically until a lost
           packet is retransmitted or intentionally dropped. Default value is 1.

       oheadbw=percents
           Recovery bandwidth overhead above input rate, in percents.  See inputbw. Default value
           is 25%.

       passphrase=string
           HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The
           passphrase is the shared secret between the sender and the receiver. It is used to
           generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function).
           It is used only if pbkeylen is non-zero. It is used on the receiver only if the
           received data is encrypted.  The configured passphrase cannot be recovered (write-
           only).

       payload_size=bytes
           Sets the maximum declared size of a packet transferred during the single call to the
           sending function in Live mode. Use 0 if this value isn't used (which is default in
           file mode).  Default is -1 (automatic), which typically means MPEG-TS; if you are
           going to use SRT to send any different kind of payload, such as, for example, wrapping
           a live stream in very small frames, then you can use a bigger maximum frame size,
           though not greater than 1456 bytes.

       pkt_size=bytes
           Alias for payload_size.

       peerlatency
           The latency value (as described in rcvlatency) that is set by the sender side as a
           minimum value for the receiver.

       pbkeylen=bytes
           Sender encryption key length, in bytes.  Only can be set to 0, 16, 24 and 32.  Enable
           sender encryption if not 0.  Not required on receiver (set to 0), key size obtained
           from sender in HaiCrypt handshake.  Default value is 0.

       rcvlatency
           The time that should elapse since the moment when the packet was sent and the moment
           when it's delivered to the receiver application in the receiving function.  This time
           should be a buffer time large enough to cover the time spent for sending, unexpectedly
           extended RTT time, and the time needed to retransmit the lost UDP packet. The
           effective latency value will be the maximum of this options' value and the value of
           peerlatency set by the peer side. Before version 1.3.0 this option is only available
           as latency.

       recv_buffer_size=bytes
           Set UDP receive buffer size, expressed in bytes.

       send_buffer_size=bytes
           Set UDP send buffer size, expressed in bytes.

       rw_timeout
           Set raise error timeout for read/write optations.

           This option is only relevant in read mode: if no data arrived in more than this time
           interval, raise error.

       tlpktdrop=1|0
           Too-late Packet Drop. When enabled on receiver, it skips missing packets that have not
           been delivered in time and delivers the following packets to the application when
           their time-to-play has come. It also sends a fake ACK to the sender. When enabled on
           sender and enabled on the receiving peer, the sender drops the older packets that have
           no chance of being delivered in time. It was automatically enabled in the sender if
           the receiver supports it.

       sndbuf=bytes
           Set send buffer size, expressed in bytes.

       rcvbuf=bytes
           Set receive buffer size, expressed in bytes.

           Receive buffer must not be greater than ffs.

       lossmaxttl=packets
           The value up to which the Reorder Tolerance may grow. When Reorder Tolerance is > 0,
           then packet loss report is delayed until that number of packets come in. Reorder
           Tolerance increases every time a "belated" packet has come, but it wasn't due to
           retransmission (that is, when UDP packets tend to come out of order), with the
           difference between the latest sequence and this packet's sequence, and not more than
           the value of this option. By default it's 0, which means that this mechanism is turned
           off, and the loss report is always sent immediately upon experiencing a "gap" in
           sequences.

       minversion
           The minimum SRT version that is required from the peer. A connection to a peer that
           does not satisfy the minimum version requirement will be rejected.

           The version format in hex is 0xXXYYZZ for x.y.z in human readable form.

       streamid=string
           A string limited to 512 characters that can be set on the socket prior to connecting.
           This stream ID will be able to be retrieved by the listener side from the socket that
           is returned from srt_accept and was connected by a socket with that set stream ID. SRT
           does not enforce any special interpretation of the contents of this string.  This
           option doesnXt make sense in Rendezvous connection; the result might be that simply
           one side will override the value from the other side and itXs the matter of luck which
           one would win

       smoother=live|file
           The type of Smoother used for the transmission for that socket, which is responsible
           for the transmission and congestion control. The Smoother type must be exactly the
           same on both connecting parties, otherwise the connection is rejected.

       messageapi=1|0
           When set, this socket uses the Message API, otherwise it uses Buffer API. Note that in
           live mode (see transtype) thereXs only message API available. In File mode you can
           chose to use one of two modes:

           Stream API (default, when this option is false). In this mode you may send as many
           data as you wish with one sending instruction, or even use dedicated functions that
           read directly from a file. The internal facility will take care of any speed and
           congestion control. When receiving, you can also receive as many data as desired, the
           data not extracted will be waiting for the next call. There is no boundary between
           data portions in the Stream mode.

           Message API. In this mode your single sending instruction passes exactly one piece of
           data that has boundaries (a message). Contrary to Live mode, this message may span
           across multiple UDP packets and the only size limitation is that it shall fit as a
           whole in the sending buffer. The receiver shall use as large buffer as necessary to
           receive the message, otherwise the message will not be given up. When the message is
           not complete (not all packets received or there was a packet loss) it will not be
           given up.

       transtype=live|file
           Sets the transmission type for the socket, in particular, setting this option sets
           multiple other parameters to their default values as required for a particular
           transmission type.

           live: Set options as for live transmission. In this mode, you should send by one
           sending instruction only so many data that fit in one UDP packet, and limited to the
           value defined first in payload_size (1316 is default in this mode). There is no speed
           control in this mode, only the bandwidth control, if configured, in order to not
           exceed the bandwidth with the overhead transmission (retransmitted and control
           packets).

           file: Set options as for non-live transmission. See messageapi for further
           explanations

       For more information see: <https://github.com/Haivision/srt>.

   srtp
       Secure Real-time Transport Protocol.

       The accepted options are:

       srtp_in_suite
       srtp_out_suite
           Select input and output encoding suites.

           Supported values:

           AES_CM_128_HMAC_SHA1_80
           SRTP_AES128_CM_HMAC_SHA1_80
           AES_CM_128_HMAC_SHA1_32
           SRTP_AES128_CM_HMAC_SHA1_32
       srtp_in_params
       srtp_out_params
           Set input and output encoding parameters, which are expressed by a base64-encoded
           representation of a binary block. The first 16 bytes of this binary block are used as
           master key, the following 14 bytes are used as master salt.

   subfile
       Virtually extract a segment of a file or another stream.  The underlying stream must be
       seekable.

       Accepted options:

       start
           Start offset of the extracted segment, in bytes.

       end End offset of the extracted segment, in bytes.  If set to 0, extract till end of file.

       Examples:

       Extract a chapter from a DVD VOB file (start and end sectors obtained externally and
       multiplied by 2048):

               subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB

       Play an AVI file directly from a TAR archive:

               subfile,,start,183241728,end,366490624,,:archive.tar

       Play a MPEG-TS file from start offset till end:

               subfile,,start,32815239,end,0,,:video.ts

   tee
       Writes the output to multiple protocols. The individual outputs are separated by |

               tee:file://path/to/local/this.avi|file://path/to/local/that.avi

   tcp
       Transmission Control Protocol.

       The required syntax for a TCP url is:

               tcp://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       The list of supported options follows.

       listen=1|0
           Listen for an incoming connection. Default value is 0.

       timeout=microseconds
           Set raise error timeout, expressed in microseconds.

           This option is only relevant in read mode: if no data arrived in more than this time
           interval, raise error.

       listen_timeout=milliseconds
           Set listen timeout, expressed in milliseconds.

       recv_buffer_size=bytes
           Set receive buffer size, expressed bytes.

       send_buffer_size=bytes
           Set send buffer size, expressed bytes.

       tcp_nodelay=1|0
           Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

       tcp_mss=bytes
           Set maximum segment size for outgoing TCP packets, expressed in bytes.

       The following example shows how to setup a listening TCP connection with ffmpeg, which is
       then accessed with ffplay:

               ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
               ffplay tcp://<hostname>:<port>

   tls
       Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

       The required syntax for a TLS/SSL url is:

               tls://<hostname>:<port>[?<options>]

       The following parameters can be set via command line options (or in code via "AVOption"s):

       ca_file, cafile=filename
           A file containing certificate authority (CA) root certificates to treat as trusted. If
           the linked TLS library contains a default this might not need to be specified for
           verification to work, but not all libraries and setups have defaults built in.  The
           file must be in OpenSSL PEM format.

       tls_verify=1|0
           If enabled, try to verify the peer that we are communicating with.  Note, if using
           OpenSSL, this currently only makes sure that the peer certificate is signed by one of
           the root certificates in the CA database, but it does not validate that the
           certificate actually matches the host name we are trying to connect to. (With other
           backends, the host name is validated as well.)

           This is disabled by default since it requires a CA database to be provided by the
           caller in many cases.

       cert_file, cert=filename
           A file containing a certificate to use in the handshake with the peer.  (When
           operating as server, in listen mode, this is more often required by the peer, while
           client certificates only are mandated in certain setups.)

       key_file, key=filename
           A file containing the private key for the certificate.

       listen=1|0
           If enabled, listen for connections on the provided port, and assume the server role in
           the handshake instead of the client role.

       Example command lines:

       To create a TLS/SSL server that serves an input stream.

               ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>

       To play back a stream from the TLS/SSL server using ffplay:

               ffplay tls://<hostname>:<port>

   udp
       User Datagram Protocol.

       The required syntax for an UDP URL is:

               udp://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       In case threading is enabled on the system, a circular buffer is used to store the
       incoming data, which allows one to reduce loss of data due to UDP socket buffer overruns.
       The fifo_size and overrun_nonfatal options are related to this buffer.

       The list of supported options follows.

       buffer_size=size
           Set the UDP maximum socket buffer size in bytes. This is used to set either the
           receive or send buffer size, depending on what the socket is used for.  Default is
           64KB.  See also fifo_size.

       bitrate=bitrate
           If set to nonzero, the output will have the specified constant bitrate if the input
           has enough packets to sustain it.

       burst_bits=bits
           When using bitrate this specifies the maximum number of bits in packet bursts.

       localport=port
           Override the local UDP port to bind with.

       localaddr=addr
           Local IP address of a network interface used for sending packets or joining multicast
           groups.

       pkt_size=size
           Set the size in bytes of UDP packets.

       reuse=1|0
           Explicitly allow or disallow reusing UDP sockets.

       ttl=ttl
           Set the time to live value (for multicast only).

       connect=1|0
           Initialize the UDP socket with "connect()". In this case, the destination address
           can't be changed with ff_udp_set_remote_url later.  If the destination address isn't
           known at the start, this option can be specified in ff_udp_set_remote_url, too.  This
           allows finding out the source address for the packets with getsockname, and makes
           writes return with AVERROR(ECONNREFUSED) if "destination unreachable" is received.
           For receiving, this gives the benefit of only receiving packets from the specified
           peer address/port.

       sources=address[,address]
           Only receive packets sent from the specified addresses. In case of multicast, also
           subscribe to multicast traffic coming from these addresses only.

       block=address[,address]
           Ignore packets sent from the specified addresses. In case of multicast, also exclude
           the source addresses in the multicast subscription.

       fifo_size=units
           Set the UDP receiving circular buffer size, expressed as a number of packets with size
           of 188 bytes. If not specified defaults to 7*4096.

       overrun_nonfatal=1|0
           Survive in case of UDP receiving circular buffer overrun. Default value is 0.

       timeout=microseconds
           Set raise error timeout, expressed in microseconds.

           This option is only relevant in read mode: if no data arrived in more than this time
           interval, raise error.

       broadcast=1|0
           Explicitly allow or disallow UDP broadcasting.

           Note that broadcasting may not work properly on networks having a broadcast storm
           protection.

       Examples

       ·   Use ffmpeg to stream over UDP to a remote endpoint:

                   ffmpeg -i <input> -f <format> udp://<hostname>:<port>

       ·   Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP packets, using a
           large input buffer:

                   ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535

       ·   Use ffmpeg to receive over UDP from a remote endpoint:

                   ffmpeg -i udp://[<multicast-address>]:<port> ...

   unix
       Unix local socket

       The required syntax for a Unix socket URL is:

               unix://<filepath>

       The following parameters can be set via command line options (or in code via "AVOption"s):

       timeout
           Timeout in ms.

       listen
           Create the Unix socket in listening mode.

SEE ALSO

       ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)

AUTHORS

       The FFmpeg developers.

       For details about the authorship, see the Git history of the project
       (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in the FFmpeg source
       directory, or browsing the online repository at <http://source.ffmpeg.org>.

       Maintainers for the specific components are listed in the file MAINTAINERS in the source
       code tree.

                                                                              FFMPEG-PROTOCOLS(1)