@chapter Protocols
@c man begin PROTOCOLS

Protocols are configured elements in Libav which allow to access
resources which require the use of a particular protocol.

When you configure your Libav 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 applehttp

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.
HTTP is default, specific protocol can be declared by specifying
"+@var{proto}" after the applehttp URI scheme name, where @var{proto}
is either "file" or "http".

@example
applehttp://host/path/to/remote/resource.m3u8
applehttp+http://host/path/to/remote/resource.m3u8
applehttp+file://path/to/local/resource.m3u8
@end example

@section concat

Physical concatenation protocol.

Allow to read and seek from many resource 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 @file{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 file

File access protocol.

Allow to read from or read to a file.

For example to read from a file @file{input.mpeg} with @file{ffmpeg}
use the command:
@example
ffmpeg -i file:input.mpeg output.mpeg
@end example

The ff* tools default to the file protocol, that is a resource
specified with the name "FILE.mpeg" is interpreted as the URL
"file:FILE.mpeg".

@section gopher

Gopher protocol.

@section http

HTTP (Hyper Text Transfer Protocol).

@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.

Allow to 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 @file{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 @file{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

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 multime‐
dia content across a TCP/IP network.

The required syntax is:
@example
rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
@end example

The accepted parameters are:
@table @option

@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.).

@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:".

@end table

For example to read with @file{ffplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
ffplay rtmp://myserver/vod/sample
@end example

@section 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 explicitely 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
@file{ffmpeg}:
@example
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
@end example

To play the same stream using @file{ffplay}:
@example
ffplay "rtmp://myserver/live/mystream live=1"
@end example

@section rtp

Real-Time Protocol.

@section rtsp

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
RTSP server, @url{http://github.com/revmischa/rtsp-server}).

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

@var{options} is a @code{&}-separated list. The following options
are supported:

@table @option

@item udp
Use UDP as lower transport protocol.

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

@item multicast
Use UDP multicast as lower transport protocol.

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

@item filter_src
Accept packets only from negotiated peer address and port.
@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 @code{tcp} and @code{udp} options are supported.

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). In
order for this to be enabled, a maximum delay must be specified in the
@code{max_delay} field of AVFormatContext.

When watching multi-bitrate Real-RTSP streams with @file{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}.

Example command lines:

To watch a stream over UDP, with a max reordering delay of 0.5 seconds:

@example
ffplay -max_delay 500000 rtsp://server/video.mp4?udp
@end example

To watch a stream tunneled over HTTP:

@example
ffplay rtsp://server/video.mp4?http
@end example

To 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

@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 ffplay:

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

And for watching in 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 tcp

Trasmission Control Protocol.

@section udp

User Datagram Protocol.

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

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

@table @option

@item buffer_size=@var{size}
set the UDP buffer size in bytes

@item localport=@var{port}
override the local UDP port to bind with

@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.
@end table

Some usage examples of the udp protocol with @file{ffmpeg} follow.

To stream over UDP to a remote endpoint:
@example
ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
@end example

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

To receive over UDP from a remote endpoint:
@example
ffmpeg -i udp://[@var{multicast-address}]:@var{port}
@end example

@c man end PROTOCOLS