X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=doc%2Fprotocols.texi;h=fad6c44c24558df85e4c24673d3bcee5f40469db;hb=de33b3e457a656230fc6d544a1889218d77a5b3c;hp=c24dc74505239ad3ae5c26926735e9769dae810d;hpb=c1c720d5279aa2e8e0518e2900f5f2b267ee974a;p=ffmpeg

diff --git a/doc/protocols.texi b/doc/protocols.texi
index c24dc745052..fad6c44c245 100644
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -1155,6 +1155,184 @@ If set to any value, listen for an incoming connection. Outgoing connection is d
 Set the maximum number of streams. By default no limit is set.
 @end table
 
+@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 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
+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 srtp
 
 Secure Real-time Transport Protocol.
@@ -1257,6 +1435,9 @@ Set send buffer size, expressed bytes.
 
 @item tcp_nodelay=@var{1|0}
 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
+
+@item tcp_mss=@var{bytes}
+Set maximum segment size for outgoing TCP packets, expressed in bytes.
 @end table
 
 The following example shows how to setup a listening TCP connection