Voice Dropping After 40 Seconds?

[num025]

Simple question guys: what would be a probable cause for normal IP/SIP calls to drop their voice signal after either 40 seconds or 1 minute and 20 seconds into the call if intra-region direct ip-to-ip audio is enabled for their ip-network-region? Let's assume there's a firewall, what do we need to look into the firewall for answers?

 

[wildcard100001]

When the call shuffles to direct IP to IP one of the phones can't reach the destination... Just a thought...

Wildcard

 

[num025]

So it could be a routing problem? We don't experience those issues if direct IP-to-IP is disabled which uses the gateway resource during the duration of an IP call which is what we don't want to happen. By the way, the gateway IP was NATed to a public IP while the rest of the workstations here were Nated to a many-to-one public IP.

 

[wildcard100001]

I don't think direct IP to IP calls will work traversing a NAT.

Wildcard

 

[num025]

Thanks Wildcard! Why whould that be the case? So even if there's an SBC it won't still work since the SBC is also NAting the IPs? Correct me if I'm wrong at anything.

 

[wildcard100001]

I've read that the NAT should be H.323 aware as addresses are included in the body of the packet (not only the header - which is why it won't work without additional configuration effort).

Search for NAT on this forum to get some further ideas on how to proceed... in theory you should be able to get it to work.

Sorry I could not be of further assistance.

Wildcard

 

[num025]

No problem, Wildcard. Your inputs are more than welcome!

Anyway, looks like we finally fixed the problem after patching the firewall. We have yet to experience any drop calls for more than an hour now for call shuffled calls.