[asterisk-users] Native bridging to SIP endpoints on the same NAT'd network

[Richard Webb]

Hi,

I have the following setup:

Asterisk <-> Nat <-> Internet <-> Nat <-> 2 x SIP endpoints

With directmedia=no I can make a call between the two SIP endpoints; the RTP
stream being passed through the Asterisk box.

Obviously, this is sub-optimal. I attempted to enable bridging of the call
between the 2 endpoints directly, given that they are on the same
non-routeable private net.

With directmedia=nonat, I see Asterisk report the bridging of the calls but
both sides of the call are routed to the originating endpoint so
effectively, the call becomes an echo-loop. There is no audio on the second
end-point although the call remains up.

I assume this is some sort of firewall/nat/routing issue. Could someone
explain what is possibly going on and perhaps offer a solution?

Cheers,

Richard.
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