[asterisk-users] Nat=route ???

[Brynjolfur Thorvardsson]

Hi, I have a question regarding NAT - I have two Asterisk setups, and a couple of softphones on my laptop to test them. In the first Asterisk I've got nat=yes for all SIP phones. The second setup is identical as far as software is concerned, but the server is running on a VPS with one of the larger VPS hosting services.

On this second setup I was able to phone out from my XLite softphone but when I tried phoning in nothing happened, basically because the phone was always UNREACHABLE. It would register fine with Asterisk but then disappear. After nosing around for quite a bit I found a suggestion that I try setting nat=route for the SIP phone, and suddenly it worked both ways.

Another SIP phone on the same laptop connects to the first Asterisk server setup (which runs on a dedicated box with fixed IP, not in VPS) but I notice that every minute or so Asterisk tells me that the phone is unreachable, then a few seconds later it becomes reachable again.

My laptop is currently sitting at home with typical home-Internet configuration (ADSL, Nat, no fixed IP).

I did see something somewhere about the big VPS providers using some form of "hidden" NAT but I don't know what that could mean.

My question is: Does this difference in behaviour have something to do with the second server running on a VPS - and are there any drawbacks to using nat=route on all client SIP phones?

Best regards

Binni

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