@chapter Protocol Options
 @c man begin 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 -@var{option} @var{value} in the
 FFmpeg tools, or by setting the value explicitly in the
 @code{AVFormatContext} options or using the @file{libavutil/opt.h} API
 for programmatic use.
 

 The list of supported options follows:
 
 @table @option
 @item protocol_whitelist @var{list} (@emph{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.
 @end table
 
 @c man end PROTOCOL OPTIONS
 

 @chapter Protocols
 @c man begin 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=@var{PROTOCOL}", or you can disable a
 particular protocol using the option
 "--disable-protocol=@var{PROTOCOL}".
 
 The option "-protocols" of the ff* tools will display the list of

 supported protocols.

 All protocols accept the following options:
 
 @table @option
 @item rw_timeout
 Maximum time to wait for (network) read/write operations to complete,
 in microseconds.
 @end table
 

 A description of the currently available protocols follows.
 

 @section amqp
 
 Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based
 publish-subscribe communication protocol.
 
 FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate
 AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ.
 
 After starting the broker, an FFmpeg client may stream data to the broker using
 the command:
 
 @example
 ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@@]hostname[:port]
 @end example
 
 Where hostname and port (default is 5672) is the address of the broker. The
 client may also set a user/password for authentication. The default for both
 fields is "guest".
 
 Muliple subscribers may stream from the broker using the command:
 @example
 ffplay amqp://[[user]:[password]@@]hostname[:port]
 @end example
 
 In RabbitMQ all data published to the broker flows through a specific exchange,
 and each subscribing client has an assigned queue/buffer. When a packet arrives
 at an exchange, it may be copied to a client's queue depending on the exchange
 and routing_key fields.
 
 The following options are supported:
 
 @table @option
 
 @item exchange
 Sets the exchange to use on the broker. RabbitMQ has several predefined
 exchanges: "amq.direct" is the default exchange, where the publisher and
 subscriber must have a matching routing_key; "amq.fanout" is the same as a
 broadcast operation (i.e. the data is forwarded to all queues on the fanout
 exchange independent of the routing_key); and "amq.topic" is similar to
 "amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ
 documentation).
 
 @item routing_key
 Sets the routing key. The default value is "amqp". The routing key is used on
 the "amq.direct" and "amq.topic" exchanges to decide whether packets are written
 to the queue of a subscriber.
 
 @item pkt_size
 Maximum size of each packet sent/received to the broker. Default is 131072.
 Minimum is 4096 and max is any large value (representable by an int). When
 receiving packets, this sets an internal buffer size in FFmpeg. It should be
 equal to or greater than the size of the published packets to the broker. Otherwise
 the received message may be truncated causing decoding errors.
 
 @item connection_timeout
 The timeout in seconds during the initial connection to the broker. The
 default value is rw_timeout, or 5 seconds if rw_timeout is not set.
 
 @end table
 

 @section async
 
 Asynchronous data filling wrapper for input stream.
 
 Fill data in a background thread, to decouple I/O operation from demux thread.
 
 @example
 async:@var{URL}
 async:http://host/resource
 async:cache:http://host/resource
 @end example
 

 @section bluray
 
 Read BluRay playlist.
 
 The accepted options are:
 @table @option
 
 @item angle
 BluRay angle
 
 @item chapter
 Start chapter (1...N)
 
 @item playlist
 Playlist to read (BDMV/PLAYLIST/?????.mpls)
 
 @end table
 
 Examples:
 
 Read longest playlist from BluRay mounted to /mnt/bluray:
 @example
 bluray:/mnt/bluray
 @end example
 
 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
 @example
 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
 @end example
 

 @section cache
 
 Caching wrapper for input stream.
 

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

 
 @example
 cache:@var{URL}
 @end example
 

 @section 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:

 @example
 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
 @end example
 
 where @var{URL1}, @var{URL2}, ..., @var{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 @file{split1.mpeg},

 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the

 command:
 @example
 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 @end example
 
 Note that you may need to escape the character "|" which is special for
 many shells.
 

 @section crypto
 
 AES-encrypted stream reading protocol.
 
 The accepted options are:
 @table @option
 @item key
 Set the AES decryption key binary block from given hexadecimal representation.
 
 @item iv
 Set the AES decryption initialization vector binary block from given hexadecimal representation.
 @end table
 
 Accepted URL formats:
 @example
 crypto:@var{URL}
 crypto+@var{URL}
 @end example
 

 @section data
 
 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
 
 For example, to convert a GIF file given inline with @command{ffmpeg}:
 @example
 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
 @end example
 

 @section file
 
 File access protocol.
 

 Read from or write to a file.

 A file URL can have the form:
 @example
 file:@var{filename}
 @end example
 
 where @var{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 @file{input.mpeg} with @command{ffmpeg}

 use the command:
 @example
 ffmpeg -i file:input.mpeg output.mpeg
 @end example
 

 This protocol accepts the following options:
 
 @table @option
 @item truncate
 Truncate existing files on write, if set to 1. A value of 0 prevents
 truncating. Default value is 1.
 
 @item blocksize
 Set I/O operation maximum block size, in bytes. Default value is
 @code{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.

 
 @item 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).

 
 @item seekable
 Controls if seekability is advertised on the file. 0 means non-seekable, -1
 means auto (seekable for normal files, non-seekable for named pipes).
 
 Many demuxers handle seekable and non-seekable resources differently,
 overriding this might speed up opening certain files at the cost of losing some
 features (e.g. accurate seeking).

 @end table
 

 @section ftp
 

 FTP (File Transfer Protocol).

 Read from or write to remote resources using FTP protocol.

 
 Following syntax is required.
 @example
 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
 @end example
 
 This protocol accepts the following options.
 
 @table @option
 @item 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.
 

 @item ftp-user
 Set a user to be used for authenticating to the FTP server. This is overridden by the
 user in the FTP URL.
 
 @item ftp-password
 Set a password to be used for authenticating to the FTP server. This is overridden by
 the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
 

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

 @section gopher
 
 Gopher protocol.
 

 @section 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
 "+@var{proto}" after the hls URI scheme name, where @var{proto}
 is either "file" or "http".
 
 @example
 hls+http://host/path/to/remote/resource.m3u8
 hls+file://path/to/local/resource.m3u8
 @end example
 

 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.
 

 @section http
 

 HTTP (Hyper Text Transfer Protocol).

 This protocol accepts the following options:

 
 @table @option
 @item 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.

 
 @item chunked_post

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

 @item content_type

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

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

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

 @item multiple_requests
 Use persistent connections if set to 1, default is 0.
 
 @item post_data
 Set custom HTTP post data.
 

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

 @item user_agent

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

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

 @item 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.
 

 @item reconnect_at_eof

 If set then eof is treated like an error and causes reconnection, this is useful

 for live / endless streams.
 
 @item reconnect_streamed
 If set then even streamed/non seekable streams will be reconnected on errors.
 

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

 @item mime_type

 Export the MIME type.

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

 @item 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 @option{icy_metadata_headers} and @option{icy_metadata_packet} options.

 The default is 1.

 
 @item icy_metadata_headers

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

 
 @item icy_metadata_packet
 If the server supports ICY metadata, and @option{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.

 @item 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.

 @item offset
 Set initial byte offset.
 
 @item end_offset
 Try to limit the request to bytes preceding this offset.

 @item 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.
 

 @item 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.

 @example
 # Server side (sending):
 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
 
 # Client side (receiving):
 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
 
 # Client can also be done with wget:
 wget http://@var{server}:@var{port} -O somefile.ogg
 
 # Server side (receiving):
 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
 
 # Client side (sending):
 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
 
 # Client can also be done with wget:
 wget --post-file=somefile.ogg http://@var{server}:@var{port}
 @end example
 

 @item send_expect_100
 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
 value is -1.
 

 @end table
 

 @subsection HTTP Cookies
 
 Some HTTP requests will be denied unless cookie values are passed in with the
 request. The @option{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:
 @example
 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
 @end example
 

 @section Icecast
 

 Icecast protocol (stream to Icecast servers)

 
 This protocol accepts the following options:

 
 @table @option
 @item ice_genre

 Set the stream genre.

 
 @item ice_name

 Set the stream name.

 
 @item ice_description

 Set the stream description.

 
 @item ice_url

 Set the stream website URL.

 
 @item ice_public
 Set if the stream should be public.

 The default is 0 (not public).

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

 @item password
 Set the Icecast mountpoint password.

 
 @item content_type

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

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

 
 @end table
 
 @example
 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
 @end example
 

 @section mmst
 
 MMS (Microsoft Media Server) protocol over TCP.
 

 @section mmsh
 
 MMS (Microsoft Media Server) protocol over HTTP.
 
 The required syntax is:
 @example
 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
 @end example
 

 @section 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.
 @example

 # 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:
 @end example
 

 Note that some formats (typically MOV) require the output protocol to

 be seekable, so they will fail with the MD5 output protocol.
 
 @section pipe
 
 UNIX pipe access protocol.
 

 Read and write from UNIX pipes.

 
 The accepted syntax is:
 @example
 pipe:[@var{number}]
 @end example
 
 @var{number} is the number corresponding to the file descriptor of the

 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).  If @var{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 @command{ffmpeg}:

 @example
 cat test.wav | ffmpeg -i pipe:0

 # ...this is the same as...

 cat test.wav | ffmpeg -i pipe:
 @end example
 

 For writing to stdout with @command{ffmpeg}:

 @example
 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
 @end example
 

 This protocol accepts the following options:
 
 @table @option
 @item blocksize
 Set I/O operation maximum block size, in bytes. Default value is
 @code{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.
 @end table
 

 Note that some formats (typically MOV), require the output protocol to

 be seekable, so they will fail with the pipe output protocol.
 

 @section 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 @code{rtp_mpegts} muxer and
 the @code{rtp} protocol.
 
 The required syntax is:
 @example
 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
 @end example
 
 The destination UDP ports are @code{port + 2} for the column FEC stream
 and @code{port + 4} for the row FEC stream.
 
 This protocol accepts the following options:
 @table @option
 
 @item l=@var{n}
 The number of columns (4-20, LxD <= 100)
 
 @item d=@var{n}
 The number of rows (4-20, LxD <= 100)
 
 @end table
 
 Example usage:
 
 @example
 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
 @end example
 

 @section 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:
 @example

 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]

 @end example
 

 The accepted parameters are:

 @table @option
 

 @item username
 An optional username (mostly for publishing).
 
 @item password
 An optional password (mostly for publishing).
 

 @item server

 The address of the RTMP server.

 
 @item port

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

 
 @item 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. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
 the value parsed from the URI through the @code{rtmp_app} option, too.

 
 @item playpath
 It is the path or name of the resource to play with reference to the

 application specified in @var{app}, may be prefixed by "mp4:". You
 can override the value parsed from the URI through the @code{rtmp_playpath}
 option, too.
 

 @item listen
 Act as a server, listening for an incoming connection.
 
 @item timeout
 Maximum time to wait for the incoming connection. Implies listen.

 @end table
 
 Additionally, the following parameters can be set via command line options
 (or in code via @code{AVOption}s):
 @table @option
 
 @item rtmp_app
 Name of application to connect on the RTMP server. This option
 overrides the parameter specified in the URI.
 

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

 @item rtmp_conn
 Extra arbitrary AMF connection parameters, parsed from a string,
 e.g. like @code{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. @code{NB:myFlag:1}). This option may be used multiple
 times to construct arbitrary AMF sequences.
 

 @item 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>).)

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

 @item rtmp_live
 Specify that the media is a live stream. No resuming or seeking in
 live streams is possible. The default value is @code{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 @code{live} and
 @code{recorded}.
 

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

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

 @item 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.
 

 @item rtmp_swfhash
 SHA256 hash of the decompressed SWF file (32 bytes).
 
 @item rtmp_swfsize
 Size of the decompressed SWF file, required for SWFVerification.
 

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

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

 @item rtmp_tcurl

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

 
 @end table
 

 For example to read with @command{ffplay} a multimedia resource named

 "sample" from the application "vod" from an RTMP server "myserver":
 @example
 ffplay rtmp://myserver/vod/sample
 @end example
 

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

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

 @end example
 

 @section 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.
 

 @section 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.
 

 @section 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.
 

 @section 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.
 

 @section 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.
 

 @section libsmbclient
 

 libsmbclient permits one to manipulate CIFS/SMB network resources.

 
 Following syntax is required.
 
 @example
 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
 @end example
 
 This protocol accepts the following options.
 
 @table @option
 @item 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.
 
 @item truncate
 Truncate existing files on write, if set to 1. A value of 0 prevents
 truncating. Default value is 1.
 
 @item workgroup
 Set the workgroup used for making connections. By default workgroup is not specified.
 
 @end table
 
 For more information see: @url{http://www.samba.org/}.
 

 @section libssh
 
 Secure File Transfer Protocol via libssh
 

 Read from or write to remote resources using SFTP protocol.

 
 Following syntax is required.
 
 @example
 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
 @end example
 
 This protocol accepts the following options.
 
 @table @option
 @item 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.
 
 @item truncate
 Truncate existing files on write, if set to 1. A value of 0 prevents
 truncating. Default value is 1.
 

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

 @end table
 
 Example: Play a file stored on remote server.
 
 @example
 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
 @end example
 

 @section 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:
 @example
 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
 @end example
 
 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
 @var{server}, @var{port}, @var{app} and @var{playpath} have the same

 meaning as specified for the RTMP native protocol.

 @var{options} contains a list of space-separated options of the form
 @var{key}=@var{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

 @command{ffmpeg}:

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

 To play the same stream using @command{ffplay}:

 @example
 ffplay "rtmp://myserver/live/mystream live=1"
 @end example
 
 @section rtp
 

 Real-time Transport Protocol.

 The required syntax for an RTP URL is:
 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
 
 @var{port} specifies the RTP port to use.
 
 The following URL options are supported:
 
 @table @option
 
 @item ttl=@var{n}

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

 
 @item rtcpport=@var{n}
 Set the remote RTCP port to @var{n}.
 
 @item localrtpport=@var{n}
 Set the local RTP port to @var{n}.
 
 @item localrtcpport=@var{n}'
 Set the local RTCP port to @var{n}.
 
 @item pkt_size=@var{n}
 Set max packet size (in bytes) to @var{n}.
 
 @item connect=0|1
 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
 to 0).
 
 @item sources=@var{ip}[,@var{ip}]
 List allowed source IP addresses.
 
 @item block=@var{ip}[,@var{ip}]
 List disallowed (blocked) source IP addresses.
 
 @item 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).
 

 @item localport=@var{n}
 Set the local RTP port to @var{n}.
 
 This is a deprecated option. Instead, @option{localrtpport} should be
 used.

 
 @end table
 
 Important notes:
 
 @enumerate
 
 @item

 If @option{rtcpport} is not set the RTCP port will be set to the RTP

 port value plus 1.
 
 @item

 If @option{localrtpport} (the local RTP port) is not set any available

 port will be used for the local RTP and RTCP ports.
 
 @item
 If @option{localrtcpport} (the local RTCP port) is not set it will be

 set to the local RTP port value plus 1.

 @end enumerate
 

 @section 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

 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).

 
 The required syntax for a RTSP url is:
 @example

 rtsp://@var{hostname}[:@var{port}]/@var{path}

 @end example
 

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

 The following options are supported.

 @table @option

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

 @item rtsp_transport

 Set RTSP transport protocols.

 
 It accepts the following values:
 @table @samp

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

 @item udp_multicast

 Use UDP multicast as lower transport protocol.
 
 @item http
 Use HTTP tunneling as lower transport protocol, which is useful for
 passing proxies.
 @end table
 
 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 @samp{tcp} and @samp{udp} options are supported.

 @item rtsp_flags
 Set RTSP flags.

 The following values are accepted:
 @table @samp

 @item filter_src
 Accept packets only from negotiated peer address and port.

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

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

 @end table
 

 Default value is @samp{none}.
 
 @item allowed_media_types
 Set media types to accept from the server.
 
 The following flags are accepted:
 @table @samp
 @item video
 @item audio
 @item data
 @end table
 
 By default it accepts all media types.
 
 @item min_port
 Set minimum local UDP port. Default value is 5000.
 
 @item max_port
 Set maximum local UDP port. Default value is 65000.
 
 @item timeout
 Set maximum timeout (in seconds) to wait for incoming connections.
 

 A value of -1 means infinite (default). This option implies the

 @option{rtsp_flags} set to @samp{listen}.
 
 @item reorder_queue_size
 Set number of packets to buffer for handling of reordered packets.
 
 @item stimeout

 Set socket TCP I/O timeout in microseconds.

 
 @item user-agent

 Override User-Agent header. If not specified, it defaults to the

 libavformat identifier string.
 @end table
 

 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 @code{max_delay} field of AVFormatContext).

 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the

 streams to display can be chosen with @code{-vst} @var{n} and
 @code{-ast} @var{n} for video and audio respectively, and can be switched
 on the fly by pressing @code{v} and @code{a}.
 

 @subsection Examples

 The following examples all make use of the @command{ffplay} and
 @command{ffmpeg} tools.

 @itemize
 @item
 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:

 @example

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

 @end example
 

 @item
 Watch a stream tunneled over HTTP:

 @example

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

 @end example
 

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

 @example
 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
 @end example
 

 @item
 Receive a stream in realtime:

 @example

 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}

 @end example

 @end itemize

 @section 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.
 

 @subsection Muxer
 

 The syntax for a SAP url given to the muxer is:
 @example
 sap://@var{destination}[:@var{port}][?@var{options}]
 @end example
 
 The RTP packets are sent to @var{destination} on port @var{port},
 or to port 5004 if no port is specified.
 @var{options} is a @code{&}-separated list. The following options
 are supported:
 
 @table @option
 
 @item announce_addr=@var{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 @var{destination} is an IPv6 address.
 
 @item announce_port=@var{port}
 Specify the port to send the announcements on, defaults to
 9875 if not specified.
 
 @item ttl=@var{ttl}
 Specify the time to live value for the announcements and RTP packets,
 defaults to 255.
 
 @item same_port=@var{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.

 @end table
 
 Example command lines follow.
 
 To broadcast a stream on the local subnet, for watching in VLC:
 
 @example
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
 @end example
 

 Similarly, for watching in @command{ffplay}:

 
 @example
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
 @end example
 

 And for watching in @command{ffplay}, over IPv6:

 
 @example
 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
 @end example
 
 @subsection Demuxer
 
 The syntax for a SAP url given to the demuxer is:
 @example
 sap://[@var{address}][:@var{port}]
 @end example
 
 @var{address} is the multicast address to listen for announcements on,
 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{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:
 
 @example
 ffplay sap://
 @end example
 
 To play back the first stream announced on one the default IPv6 SAP multicast address:
 
 @example
 ffplay sap://[ff0e::2:7ffe]
 @end example
 

 @section sctp
 
 Stream Control Transmission Protocol.
 
 The accepted URL syntax is:
 @example
 sctp://@var{host}:@var{port}[?@var{options}]
 @end example
 
 The protocol accepts the following options:
 @table @option
 @item listen
 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
 
 @item max_streams
 Set the maximum number of streams. By default no limit is set.
 @end table
 

 @section srt
 
 Haivision Secure Reliable Transport Protocol via libsrt.
 
 The supported syntax for a SRT URL is:
 @example
 srt://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
 @var{options} contains a list of &-separated options of the form
 @var{key}=@var{val}.
 
 or
 
 @example
 @var{options} srt://@var{hostname}:@var{port}
 @end example
 
 @var{options} contains a list of '-@var{key} @var{val}'
 options.
 
 This protocol accepts the following options.
 
 @table @option

 @item connect_timeout=@var{milliseconds}

 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).
 
 @item ffs=@var{bytes}
 Flight Flag Size (Window Size), in bytes. FFS is actually an
 internal parameter and you should set it to not less than
 @option{recv_buffer_size} and @option{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.
 
 @item inputbw=@var{bytes/seconds}
 Sender nominal input rate, in bytes per seconds. Used along with
 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
 calculate maximum sending rate when recovery packets are sent
 along with the main media stream:
 @option{inputbw} * (100 + @option{oheadbw}) / 100
 if @option{inputbw} is not set while @option{maxbw} is set to
 relative (0), the actual input rate is evaluated inside
 the library. Default value is 0.
 
 @item iptos=@var{tos}
 IP Type of Service. Applies to sender only. Default value is 0xB8.
 
 @item ipttl=@var{ttl}
 IP Time To Live. Applies to sender only. Default value is 64.
 

 @item latency=@var{microseconds}

 Timestamp-based Packet Delivery Delay.
 Used to absorb bursts of missed packet retransmissions.
 This flag sets both @option{rcvlatency} and @option{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 @option{peerlatency},
 when side is sender and @option{rcvlatency}
 when side is receiver, and the bidirectional stream
 sending is not supported.
 

 @item listen_timeout=@var{microseconds}

 Set socket listen timeout.
 
 @item maxbw=@var{bytes/seconds}
 Maximum sending bandwidth, in bytes per seconds.
 -1 infinite (CSRTCC limit is 30mbps)
 0 relative to input rate (see @option{inputbw})
 >0 absolute limit value
 Default value is 0 (relative)
 
 @item mode=@var{caller|listener|rendezvous}
 Connection mode.
 @option{caller} opens client connection.
 @option{listener} starts server to listen for incoming connections.
 @option{rendezvous} use Rendez-Vous connection mode.
 Default value is caller.
 
 @item mss=@var{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.
 
 @item nakreport=@var{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.
 
 @item oheadbw=@var{percents}
 Recovery bandwidth overhead above input rate, in percents.
 See @option{inputbw}. Default value is 25%.
 
 @item passphrase=@var{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 @option{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).
 

 @item enforced_encryption=@var{1|0}
 If true, both connection parties must have the same password
 set (including empty, that is, with no encryption). If the
 password doesn't match or only one side is unencrypted,
 the connection is rejected. Default is true.
 
 @item kmrefreshrate=@var{packets}
 The number of packets to be transmitted after which the
 encryption key is switched to a new key. Default is -1.
 -1 means auto (0x1000000 in srt library). The range for
 this option is integers in the 0 - @code{INT_MAX}.
 
 @item kmpreannounce=@var{packets}
 The interval between when a new encryption key is sent and
 when switchover occurs. This value also applies to the
 subsequent interval between when switchover occurs and
 when the old encryption key is decommissioned. Default is -1.
 -1 means auto (0x1000 in srt library). The range for
 this option is integers in the 0 - @code{INT_MAX}.
 

 @item payload_size=@var{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.
 

 @item pkt_size=@var{bytes}
 Alias for @samp{payload_size}.
 

 @item peerlatency=@var{microseconds}

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

 @item pbkeylen=@var{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.
 

 @item rcvlatency=@var{microseconds}

 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 @option{peerlatency}
 set by the peer side. Before version 1.3.0 this option
 is only available as @option{latency}.
 

 @item recv_buffer_size=@var{bytes}

 Set UDP receive buffer size, expressed in bytes.

 
 @item send_buffer_size=@var{bytes}

 Set UDP send buffer size, expressed in bytes.

 @item timeout=@var{microseconds}

 Set raise error timeouts for read, write and connect operations. Note that the
 SRT library has internal timeouts which can be controlled separately, the
 value set here is only a cap on those.

 
 @item tlpktdrop=@var{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.
 

 @item sndbuf=@var{bytes}
 Set send buffer size, expressed in bytes.
 
 @item rcvbuf=@var{bytes}
 Set receive buffer size, expressed in bytes.
 
 Receive buffer must not be greater than @option{ffs}.
 
 @item lossmaxttl=@var{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.
 
 @item 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.
 
 @item streamid=@var{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 doesn’t make sense in Rendezvous connection; the result
 might be that simply one side will override the value from the other
 side and it’s the matter of luck which one would win
 
 @item smoother=@var{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.
 
 @item messageapi=@var{1|0}
 When set, this socket uses the Message API, otherwise it uses Buffer
 API. Note that in live mode (see @option{transtype}) there’s 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.
 
 @item transtype=@var{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 @option{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 @option{messageapi}
 for further explanations
 

 @item linger=@var{seconds}
 The number of seconds that the socket waits for unsent data when closing.
 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
 seconds in file mode). The range for this option is integers in the
 0 - @code{INT_MAX}.
 

 @end table
 
 For more information see: @url{https://github.com/Haivision/srt}.
 

 @section srtp
 
 Secure Real-time Transport Protocol.
 
 The accepted options are:
 @table @option
 @item srtp_in_suite
 @item srtp_out_suite
 Select input and output encoding suites.
 
 Supported values:
 @table @samp
 @item AES_CM_128_HMAC_SHA1_80
 @item SRTP_AES128_CM_HMAC_SHA1_80
 @item AES_CM_128_HMAC_SHA1_32
 @item SRTP_AES128_CM_HMAC_SHA1_32
 @end table
 
 @item srtp_in_params
 @item 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.
 @end table
 

 @section subfile
 
 Virtually extract a segment of a file or another stream.
 The underlying stream must be seekable.
 
 Accepted options:
 @table @option
 @item start
 Start offset of the extracted segment, in bytes.
 @item end
 End offset of the extracted segment, in bytes.

 If set to 0, extract till end of file.

 @end table
 
 Examples:
 
 Extract a chapter from a DVD VOB file (start and end sectors obtained
 externally and multiplied by 2048):
 @example
 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
 @end example
 
 Play an AVI file directly from a TAR archive:

 @example

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

 @end example

 Play a MPEG-TS file from start offset till end:
 @example
 subfile,,start,32815239,end,0,,:video.ts
 @end example
 

 @section tee
 

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

 by |
 
 @example
 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
 @end example
 

 @section tcp
 

 Transmission Control Protocol.

 The required syntax for a TCP url is:
 @example
 tcp://@var{hostname}:@var{port}[?@var{options}]
 @end example
 

 @var{options} contains a list of &-separated options of the form
 @var{key}=@var{val}.

 The list of supported options follows.
 
 @table @option
 @item listen=@var{1|0}
 Listen for an incoming connection. Default value is 0.

 @item timeout=@var{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.
 

 @item listen_timeout=@var{milliseconds}
 Set listen timeout, expressed in milliseconds.

 
 @item recv_buffer_size=@var{bytes}
 Set receive buffer size, expressed bytes.
 
 @item send_buffer_size=@var{bytes}
 Set send buffer size, expressed bytes.

 
 @item tcp_nodelay=@var{1|0}
 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

 
 @item tcp_mss=@var{bytes}
 Set maximum segment size for outgoing TCP packets, expressed in bytes.

 @end table
 
 The following example shows how to setup a listening TCP connection
 with @command{ffmpeg}, which is then accessed with @command{ffplay}:

 @example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
 @end example
 

 @section tls
 

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

 
 The required syntax for a TLS/SSL url is:
 @example
 tls://@var{hostname}:@var{port}[?@var{options}]
 @end example
 

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

 @table @option
 

 @item ca_file, cafile=@var{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.

 
 @item tls_verify=@var{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.
 

 @item cert_file, cert=@var{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.)
 

 @item key_file, key=@var{filename}

 A file containing the private key for the certificate.
 
 @item listen=@var{1|0}
 If enabled, listen for connections on the provided port, and assume
 the server role in the handshake instead of the client role.

 
 @end table
 
 Example command lines:
 
 To create a TLS/SSL server that serves an input stream.
 
 @example
 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
 @end example
 
 To play back a stream from the TLS/SSL server using @command{ffplay}:
 
 @example
 ffplay tls://@var{hostname}:@var{port}
 @end example
 

 @section udp
 
 User Datagram Protocol.
 

 The required syntax for an UDP URL is:

 @example
 udp://@var{hostname}:@var{port}[?@var{options}]
 @end example
 

 @var{options} contains a list of &-separated options of the form @var{key}=@var{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 @var{fifo_size} and
 @var{overrun_nonfatal} options are related to this buffer.
 

 The list of supported options follows.

 
 @table @option
 @item buffer_size=@var{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 32 KB for output, 384 KB for input.  See also @var{fifo_size}.

 @item bitrate=@var{bitrate}
 If set to nonzero, the output will have the specified constant bitrate if the
 input has enough packets to sustain it.
 
 @item burst_bits=@var{bits}
 When using @var{bitrate} this specifies the maximum number of bits in
 packet bursts.

 @item localport=@var{port}

 Override the local UDP port to bind with.

 @item localaddr=@var{addr}

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

 @item pkt_size=@var{size}

 Set the size in bytes of UDP packets.

 
 @item reuse=@var{1|0}

 Explicitly allow or disallow reusing UDP sockets.

 
 @item ttl=@var{ttl}

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

 
 @item connect=@var{1|0}
 Initialize the UDP socket with @code{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.

 
 @item sources=@var{address}[,@var{address}]

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

 
 @item block=@var{address}[,@var{address}]

 Ignore packets sent from the specified addresses. In case of multicast, also
 exclude the source addresses in the multicast subscription.

 
 @item fifo_size=@var{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.
 
 @item overrun_nonfatal=@var{1|0}
 Survive in case of UDP receiving circular buffer overrun. Default
 value is 0.

 
 @item timeout=@var{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.

 
 @item broadcast=@var{1|0}
 Explicitly allow or disallow UDP broadcasting.
 
 Note that broadcasting may not work properly on networks having
 a broadcast storm protection.

 @end table
 

 @subsection Examples

 @itemize
 @item
 Use @command{ffmpeg} to stream over UDP to a remote endpoint:

 @example
 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
 @end example
 

 @item
 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
 sized UDP packets, using a large input buffer:

 @example
 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
 @end example
 

 @item
 Use @command{ffmpeg} to receive over UDP from a remote endpoint:

 @example

 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...

 @end example

 @end itemize

 @section unix
 
 Unix local socket
 
 The required syntax for a Unix socket URL is:
 
 @example
 unix://@var{filepath}
 @end example
 
 The following parameters can be set via command line options
 (or in code via @code{AVOption}s):
 
 @table @option
 @item timeout
 Timeout in ms.
 @item listen
 Create the Unix socket in listening mode.
 @end table
 

 @section zmq
 
 ZeroMQ asynchronous messaging using the libzmq library.
 
 This library supports unicast streaming to multiple clients without relying on
 an external server.
 
 The required syntax for streaming or connecting to a stream is:
 @example
 zmq:tcp://ip-address:port
 @end example
 
 Example:
 Create a localhost stream on port 5555:
 @example
 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
 @end example
 
 Multiple clients may connect to the stream using:
 @example
 ffplay zmq:tcp://127.0.0.1:5555
 @end example
 
 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
 The server side binds to a port and publishes data. Clients connect to the
 server (via IP address/port) and subscribe to the stream. The order in which
 the server and client start generally does not matter.
 
 ffmpeg must be compiled with the --enable-libzmq option to support
 this protocol.
 
 Options can be set on the @command{ffmpeg}/@command{ffplay} command
 line. The following options are supported:
 
 @table @option
 
 @item pkt_size
 Forces the maximum packet size for sending/receiving data. The default value is

 131,072 bytes. On the server side, this sets the maximum size of sent packets

 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
 packets. Note that pkt_size on the clients should be equal to or greater than
 pkt_size on the server. Otherwise the received message may be truncated causing
 decoding errors.
 
 @end table
 
 

 @c man end PROTOCOLS