@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 av* 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 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 @command{avplay} use the
command:
@example
avplay 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 @command{avconv}
use the command:
@example
avconv -i file:input.mpeg output.mpeg
@end example

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

This protocol accepts the following options:

@table @option
@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).

@end table

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

@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 offset
Set initial byte offset.

@item end_offset
Try to limit the request to bytes preceding this offset.
@end table

@section Icecast

Icecast (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 or not.
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

@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.
avconv -i input.flv -f avi -y md5:output.avi.md5

# Write the MD5 hash of the encoded AVI file to stdout.
avconv -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 @command{avconv}:
@example
cat test.wav | avconv -i pipe:0
# ...this is the same as...
cat test.wav | avconv -i pipe:
@end example

For writing to stdout with @command{avconv}:
@example
avconv -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
avconv -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 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{avplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
avplay rtmp://myserver/vod/sample
@end example

To publish to a password protected server, passing the playpath and
app names separately:
@example
avconv -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 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{avconv}:
@example
avconv -re -i myfile -f flv rtmp://myserver/live/mystream
@end example

To play the same stream using @command{avplay}:
@example
avplay "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
@uref{http://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

The following options (set on the @command{avconv}/@command{avplay} command
line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
are supported:

Flags for @code{rtsp_transport}:

@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 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 @code{tcp} and @code{udp} options are supported.

Flags for @code{rtsp_flags}:

@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
@item listen
Act as a server, listening for an incoming connection.
@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{avplay}, 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
avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example

To watch a stream tunneled over HTTP:

@example
avplay -rtsp_transport http rtsp://server/video.mp4
@end example

To send a stream in realtime to a RTSP server, for others to watch:

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

To receive a stream in realtime:

@example
avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
@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
avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
@end example

Similarly, for watching in avplay:

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

And for watching in avplay, over IPv6:

@example
avconv -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
avplay sap://
@end example

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

@example
avplay sap://[ff0e::2:7ffe]
@end example

@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
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
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
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 payloadsize=@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 value is for 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 peerlatency
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
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 receive buffer size, expressed in bytes.

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

@item rw_timeout
Set raise error timeout for read/write optations.

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

@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.

@end table

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

@section tcp

Transmission Control Protocol.

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

@table @option

@item listen
Listen for an incoming connection

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

@end table

@section tls

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

The required syntax for a TLS url is:
@example
tls://@var{hostname}:@var{port}
@end example

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

@table @option

@item ca_file
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.

@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 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.

@item cert_file
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
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

@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 &-separated 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 localaddr=@var{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.

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

@item block=@var{address}[,@var{address}]
Ignore packets sent to the multicast group from the specified
sender IP addresses.
@end table

Some usage examples of the udp protocol with @command{avconv} follow.

To stream over UDP to a remote endpoint:
@example
avconv -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
avconv -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
avconv -i udp://[@var{multicast-address}]:@var{port}
@end example

@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

@c man end PROTOCOLS