protocols.texi

@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.

A description of the currently available protocols follows.

@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.
@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-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.

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

@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 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 mutli-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.
@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

@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 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 miliseconds 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.