DBL Technology GoIP User Manual page 29

9. Signaling NAT Traversal
10. Advanced Timings

No Answer Expiry
(32-180s)

NICT Expiry (2-180s)

ICT Expiry (5-360s)

Retransmit T1
(200-2000ms)

Retransmit T2
(20000-8000ms)
11. GSM-SIP Code Map
http://www.dbltek.com
This setting is not required if the target SIP server / PBX supports NAT traversal.
However, if your ISP blocks VoIP traffics, you could try to use Relay Proxy setting.
Depending on how your ISP blocks VoIP traffics, the Relay Server method may or may
not work in your network environment.
Two NAT Traversal methods are supported:
1. Stun Server – An external Stun Server is required. This allows the device to
obtain the public IP of the network used.
2. Relay Proxy – This is a proprietary method developed by DBL Technology. DBL's
Relay Proxy server must be used. A free copy of the Relay Proxy can be
download from DBL's website (www.dbltek.com). Please contact
support@dbltek.com for further assistance if needed.
This section consists of 5 basic timers in the SIP protocol. Configure them carefully so
that they are compatible with the SIP Server and your requirements.
This timer specifies the timeout for an unanswered call. A SIP 408 Request Timeout
command is sent to the SIP Server when this timer expires.
Note: The default value is set the maximum value so that it will not interfere with the
call unanswered timeout at SIP Proxy or PBX.
NICT: Non Invite Client Transaction (RFC 3261 Section 17.1.2)
ICT : Invite Client Transaction (RFC 3261 Section 17.1.1)
Round Trip Time (RTT) estimate (RFC 3261 Section 17.1.1)
This timer applies to the following timeout timer.
1. INVITE request retransmission interval, for UDP only
2. Non-INVITE request retransmission interval, UDP only
3. INVITE response retransmission interval.
The maximum retransmit interval for non-INVITE requests and INVITE
responses.
When making a GSM call, a GSM specific cause defined in GSM04.08 Annex H is
returned from the network for call control. This GSM cause must be returned back to
the SIP server for call handling and control. In order to meet various application
requirements, the GSM-SIP Code Map allows the user to define the corresponding SIP
message for each GSM cause listed below .
GoIP User Manual
None
180
2
5
200
2000
28