ZyXEL Communications 2002 Series User Manual

Prestige 2002 Series User’s Guide

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Chapter 11 Troubleshooting

Table 34   Troubleshooting Voice Service

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.