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VoIP subscriber gateways
3.1.4.1.1.
Common settings
SIP Configuration:
–
STUN enable
– STUN (Session Traversal Utilities for NAT) is used during initialization of STUN server
in the network to determine public address (the device external gateway address);
–
STUN server address (:port)
–
IP address or domain name of STUN server. Alternative
server port can be assigned after colon (the default value is 3478);
–
STUN request sending interval (sec)
–
STUN request sending interval. The less polling
interval then higher speed of reaction on the public address changes;
–
Public I
P
– the parameter is used as external device address during work on NAT (on gateway). This
parameter is used as a public address of gateway (NAT) WAN interface on which TAU-8.IP is set up.
At that, SIP and RTP port forwarding is required (these ports are used by TAU-8.IP);
–
Disable NAPTR DNS queries
– in some cases, when DNS operates incorrectly, NAPTR queries
(Naming authority pointer) may cause negative result. When flag is set, these queries will be
disabled;
–
Disable SRV DNS requests (STUN request sending interval)
– in some cases, when DNS server
operates incorrectly, SRV requires may cause negative result. When flag is set, automatic queries
will be disabled;
–
Invite initial timeout (ms)
–
time interval (in milliseconds) between the first INVITE message
transfer and the second INVITE message transfer when the first message is unanswered. This
interval will be doubled for the next INVITEs (third, fourth and etc.).(For example, if the second
INVITE will be transferred after 300 ms, the third will be transmitted after 600 ms, the fourth –
after 1200 ms and etc.);
–
Retransmission interval for nonINVITE requests (ms)
–
time interval in milliseconds between the
first nonINVITE
message transfer and the second nonINVITE message transfer when the first
message is unanswered. This interval will be doubled for the next message transfers (third, fourth
and etc.).(For example, if the second nonINVITE will be transferred after 300 ms, the third will be
transmitted after 600 ms, the fourth – after 1200 ms and etc., up to value of INVITE initial
timeout);
–
Invite total timeout (ms)
– total timeout of INVITE message transmission, in milliseconds. Upon
timeout of INVITE message transmission (in milliseconds) the selected direction will be not
available. It is used to limit INVITE message retranslation including determination of SIP-proxy
accessibility
;
–
Transport
– selecting a transport layer protocol that is used to receive and transmit SIP messages:
–
UDP(preferred), TCP
– receiving via UDP and TCP. TCP is used for packet sending with size
more than 1300 bytes, UDP- for packets with size up to 1300 bytes;
–
TCP(preferred), UDP
– reception via UDP and TCP. Transmission via TCP. If connection is not
established via TCP, the transmission will be performed via UDP;