+@section srt
+
+Haivision Secure Reliable Transport Protocol via libsrt.
+
+The supported syntax for a SRT URL is:
+@example
+srt://@var{hostname}:@var{port}[?@var{options}]
+@end example
+
+@var{options} contains a list of &-separated options of the form
+@var{key}=@var{val}.
+
+or
+
+@example
+@var{options} srt://@var{hostname}:@var{port}
+@end example
+
+@var{options} contains a list of '-@var{key} @var{val}'
+options.
+
+This protocol accepts the following options.
+
+@table @option
+@item connect_timeout=@var{milliseconds}
+Connection timeout; SRT cannot connect for RTT > 1500 msec
+(2 handshake exchanges) with the default connect timeout of
+3 seconds. This option applies to the caller and rendezvous
+connection modes. The connect timeout is 10 times the value
+set for the rendezvous mode (which can be used as a
+workaround for this connection problem with earlier versions).
+
+@item ffs=@var{bytes}
+Flight Flag Size (Window Size), in bytes. FFS is actually an
+internal parameter and you should set it to not less than
+@option{recv_buffer_size} and @option{mss}. The default value
+is relatively large, therefore unless you set a very large receiver buffer,
+you do not need to change this option. Default value is 25600.
+
+@item inputbw=@var{bytes/seconds}
+Sender nominal input rate, in bytes per seconds. Used along with
+@option{oheadbw}, when @option{maxbw} is set to relative (0), to
+calculate maximum sending rate when recovery packets are sent
+along with the main media stream:
+@option{inputbw} * (100 + @option{oheadbw}) / 100
+if @option{inputbw} is not set while @option{maxbw} is set to
+relative (0), the actual input rate is evaluated inside
+the library. Default value is 0.
+
+@item iptos=@var{tos}
+IP Type of Service. Applies to sender only. Default value is 0xB8.
+
+@item ipttl=@var{ttl}
+IP Time To Live. Applies to sender only. Default value is 64.
+
+@item latency=@var{microseconds}
+Timestamp-based Packet Delivery Delay.
+Used to absorb bursts of missed packet retransmissions.
+This flag sets both @option{rcvlatency} and @option{peerlatency}
+to the same value. Note that prior to version 1.3.0
+this is the only flag to set the latency, however
+this is effectively equivalent to setting @option{peerlatency},
+when side is sender and @option{rcvlatency}
+when side is receiver, and the bidirectional stream
+sending is not supported.
+
+@item listen_timeout=@var{microseconds}
+Set socket listen timeout.
+
+@item maxbw=@var{bytes/seconds}
+Maximum sending bandwidth, in bytes per seconds.
+-1 infinite (CSRTCC limit is 30mbps)
+0 relative to input rate (see @option{inputbw})
+>0 absolute limit value
+Default value is 0 (relative)
+
+@item mode=@var{caller|listener|rendezvous}
+Connection mode.
+@option{caller} opens client connection.
+@option{listener} starts server to listen for incoming connections.
+@option{rendezvous} use Rendez-Vous connection mode.
+Default value is caller.
+
+@item mss=@var{bytes}
+Maximum Segment Size, in bytes. Used for buffer allocation
+and rate calculation using a packet counter assuming fully
+filled packets. The smallest MSS between the peers is
+used. This is 1500 by default in the overall internet.
+This is the maximum size of the UDP packet and can be
+only decreased, unless you have some unusual dedicated
+network settings. Default value is 1500.
+
+@item nakreport=@var{1|0}
+If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
+periodically until a lost packet is retransmitted or
+intentionally dropped. Default value is 1.
+
+@item oheadbw=@var{percents}
+Recovery bandwidth overhead above input rate, in percents.
+See @option{inputbw}. Default value is 25%.
+
+@item passphrase=@var{string}
+HaiCrypt Encryption/Decryption Passphrase string, length
+from 10 to 79 characters. The passphrase is the shared
+secret between the sender and the receiver. It is used
+to generate the Key Encrypting Key using PBKDF2
+(Password-Based Key Derivation Function). It is used
+only if @option{pbkeylen} is non-zero. It is used on
+the receiver only if the received data is encrypted.
+The configured passphrase cannot be recovered (write-only).
+
+@item enforced_encryption=@var{1|0}
+If true, both connection parties must have the same password
+set (including empty, that is, with no encryption). If the
+password doesn't match or only one side is unencrypted,
+the connection is rejected. Default is true.
+
+@item kmrefreshrate=@var{packets}
+The number of packets to be transmitted after which the
+encryption key is switched to a new key. Default is -1.
+-1 means auto (0x1000000 in srt library). The range for
+this option is integers in the 0 - @code{INT_MAX}.
+
+@item kmpreannounce=@var{packets}
+The interval between when a new encryption key is sent and
+when switchover occurs. This value also applies to the
+subsequent interval between when switchover occurs and
+when the old encryption key is decommissioned. Default is -1.
+-1 means auto (0x1000 in srt library). The range for
+this option is integers in the 0 - @code{INT_MAX}.
+
+@item payload_size=@var{bytes}
+Sets the maximum declared size of a packet transferred
+during the single call to the sending function in Live
+mode. Use 0 if this value isn't used (which is default in
+file mode).
+Default is -1 (automatic), which typically means MPEG-TS;
+if you are going to use SRT
+to send any different kind of payload, such as, for example,
+wrapping a live stream in very small frames, then you can
+use a bigger maximum frame size, though not greater than
+1456 bytes.
+
+@item pkt_size=@var{bytes}
+Alias for @samp{payload_size}.
+
+@item peerlatency=@var{microseconds}
+The latency value (as described in @option{rcvlatency}) that is
+set by the sender side as a minimum value for the receiver.
+
+@item pbkeylen=@var{bytes}
+Sender encryption key length, in bytes.
+Only can be set to 0, 16, 24 and 32.
+Enable sender encryption if not 0.
+Not required on receiver (set to 0),
+key size obtained from sender in HaiCrypt handshake.
+Default value is 0.
+
+@item rcvlatency=@var{microseconds}
+The time that should elapse since the moment when the
+packet was sent and the moment when it's delivered to
+the receiver application in the receiving function.
+This time should be a buffer time large enough to cover
+the time spent for sending, unexpectedly extended RTT
+time, and the time needed to retransmit the lost UDP
+packet. The effective latency value will be the maximum
+of this options' value and the value of @option{peerlatency}
+set by the peer side. Before version 1.3.0 this option
+is only available as @option{latency}.
+
+@item recv_buffer_size=@var{bytes}
+Set UDP receive buffer size, expressed in bytes.
+
+@item send_buffer_size=@var{bytes}
+Set UDP send buffer size, expressed in bytes.
+
+@item timeout=@var{microseconds}
+Set raise error timeouts for read, write and connect operations. Note that the
+SRT library has internal timeouts which can be controlled separately, the
+value set here is only a cap on those.
+
+@item tlpktdrop=@var{1|0}
+Too-late Packet Drop. When enabled on receiver, it skips
+missing packets that have not been delivered in time and
+delivers the following packets to the application when
+their time-to-play has come. It also sends a fake ACK to
+the sender. When enabled on sender and enabled on the
+receiving peer, the sender drops the older packets that
+have no chance of being delivered in time. It was
+automatically enabled in the sender if the receiver
+supports it.
+
+@item sndbuf=@var{bytes}
+Set send buffer size, expressed in bytes.
+
+@item rcvbuf=@var{bytes}
+Set receive buffer size, expressed in bytes.
+
+Receive buffer must not be greater than @option{ffs}.
+
+@item lossmaxttl=@var{packets}
+The value up to which the Reorder Tolerance may grow. When
+Reorder Tolerance is > 0, then packet loss report is delayed
+until that number of packets come in. Reorder Tolerance
+increases every time a "belated" packet has come, but it
+wasn't due to retransmission (that is, when UDP packets tend
+to come out of order), with the difference between the latest
+sequence and this packet's sequence, and not more than the
+value of this option. By default it's 0, which means that this
+mechanism is turned off, and the loss report is always sent
+immediately upon experiencing a "gap" in sequences.
+
+@item minversion
+The minimum SRT version that is required from the peer. A connection
+to a peer that does not satisfy the minimum version requirement
+will be rejected.
+
+The version format in hex is 0xXXYYZZ for x.y.z in human readable
+form.
+
+@item streamid=@var{string}
+A string limited to 512 characters that can be set on the socket prior
+to connecting. This stream ID will be able to be retrieved by the
+listener side from the socket that is returned from srt_accept and
+was connected by a socket with that set stream ID. SRT does not enforce
+any special interpretation of the contents of this string.
+This option doesn’t make sense in Rendezvous connection; the result
+might be that simply one side will override the value from the other
+side and it’s the matter of luck which one would win
+
+@item smoother=@var{live|file}
+The type of Smoother used for the transmission for that socket, which
+is responsible for the transmission and congestion control. The Smoother
+type must be exactly the same on both connecting parties, otherwise
+the connection is rejected.
+
+@item messageapi=@var{1|0}
+When set, this socket uses the Message API, otherwise it uses Buffer
+API. Note that in live mode (see @option{transtype}) there’s only
+message API available. In File mode you can chose to use one of two modes:
+
+Stream API (default, when this option is false). In this mode you may
+send as many data as you wish with one sending instruction, or even use
+dedicated functions that read directly from a file. The internal facility
+will take care of any speed and congestion control. When receiving, you
+can also receive as many data as desired, the data not extracted will be
+waiting for the next call. There is no boundary between data portions in
+the Stream mode.
+
+Message API. In this mode your single sending instruction passes exactly
+one piece of data that has boundaries (a message). Contrary to Live mode,
+this message may span across multiple UDP packets and the only size
+limitation is that it shall fit as a whole in the sending buffer. The
+receiver shall use as large buffer as necessary to receive the message,
+otherwise the message will not be given up. When the message is not
+complete (not all packets received or there was a packet loss) it will
+not be given up.
+
+@item transtype=@var{live|file}
+Sets the transmission type for the socket, in particular, setting this
+option sets multiple other parameters to their default values as required
+for a particular transmission type.
+
+live: Set options as for live transmission. In this mode, you should
+send by one sending instruction only so many data that fit in one UDP packet,
+and limited to the value defined first in @option{payload_size} (1316 is
+default in this mode). There is no speed control in this mode, only the
+bandwidth control, if configured, in order to not exceed the bandwidth with
+the overhead transmission (retransmitted and control packets).
+
+file: Set options as for non-live transmission. See @option{messageapi}
+for further explanations
+
+@item linger=@var{seconds}
+The number of seconds that the socket waits for unsent data when closing.
+Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
+seconds in file mode). The range for this option is integers in the
+0 - @code{INT_MAX}.
+
+@end table
+
+For more information see: @url{https://github.com/Haivision/srt}.
+
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