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FFMPEG-PROTOCOLS(1)					   FFMPEG-PROTOCOLS(1)

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.

       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.

       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 or ffserver.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.

       o   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

       o   Watch a stream tunneled over	HTTP:

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

       o   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

       o   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.

   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.

       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

   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.

       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 GnuTLS,
	   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
	   Choose the local IP address.	This is	useful e.g. if sending
	   multicast and the host has multiple interfaces, where the user can
	   choose which	interface to send on by	specifying the IP address of
	   that	interface.

       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 to	the multicast group from one of	the
	   specified sender IP addresses.

       block=address[,address]
	   Ignore packets sent to the multicast	group from the specified
	   sender IP addresses.

       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

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

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

       o   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

       o   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), ffserver(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)

NAME | DESCRIPTION | PROTOCOL OPTIONS | PROTOCOLS | SEE ALSO | AUTHORS

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