ZyXEL Communications 2002 Series - Manual (Page 74)

Prestige 2002 Series User’s Guide
74 Chapter 11 Troubleshooting
11.8 Problems with Voice Service
Table 34 Troubleshooting Voice Service
PROBLEM CORRECTIVE ACTION
After the VoIP is
configured and
working, others are
unable to call you
or you lose your
connection during a
call. There is a NAT
router between the
Prestige and the
SIP server.
This could be caused by a short NAT UDP session timeout on the NAT router.
When the SIP session’s entry in the NAT table times out, the NAT router does not
have any record to use for forwarding VoIP traffic to the Prestige.
If possible, set the NAT router to use a longer NAT UDP session timeout.
Otherwise, try one of the following:
•
Shorten the registration expiration period (see the Expiration Duration field
in the VoIP Advanced screen) in order to cause the Prestige to re-register
with the SIP register server more frequently. Note that this will not help if the
SIP register server enforces a long registration expiration period (since the
Prestige will also use the period set by the SIP register server).
• Use STUN. If your VoIP service provider does not have a STUN server, you
can still enable STUN and enter the IP address and port number of the SIP
server in the STUN server fields. This causes the Prestige to send STUN
requests to the SIP server. While this will not make STUN work (since there
won’t be any responses to the STUN requests), it should keep the NAT UDP
session in the NAT router.