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SIP NAT STUN Expert needed

Last Modified: 2013-12-21
I'm trying to get a more fundamental understanding of the complications of running SIP phones through NAT to a SIP server that's also behind NAT. More specifically, we're trying to make the 3CX phone client and also a Grandstream GXP-2000 deskphone register with a 3CX SIP server (that uses PSTN exclusively to make calls) that's behind NAT, (and in some cases our clients' 3CX SIP servers are behind double NAT).

Here's some of the questions that come up:

1) If I have port forwarding on the phone/client side setup for the same ports (5060 UDP/TCP, 3478 UDP/TCP, 9000-9019 UDP) as what we use on the server side, will that cause a problem?

2) Do we always need to use STUN? On the client side? Server Side? Are stun servers reliable?

3) NAT seems to really be the big problem. Is is possible to disable NAT on consumer DSL modems such as the Thomson 516 or the Thomson 546?

Any information that can be offered on the fundamentals of NAT, testing SIP with NAT, would be greatly appreciated.

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