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Originally Posted by Mallycat
I have read this thread, and am very interested in HOW to set up my ATA so it correctly routes the call bypassing VoXaLot once the SIP session has been initiated. This question was asked several times in the thread posted above, but has not been answered as yet. I assume the SIP message takes one path (via VoXaLot) and the UDP voice message takes a more direct path. I have not been able to experience this, so I assume my ATA is not set up as well as it could be.
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If your ATA is behind a NAT (ie, it is not sitting on a public IP address), you need some way for the ATA to find out what the public IP address is, so it can put that address in the outgoing SIP messages.
One way is to use STUN, the other way would be to manually input the IP address. In addition, you would need to port forward some UDP ports for the incoming RTP stream (as well as UDP port 5060 for the incoming SIP INVITES).
Edited: more detail -
Basically, you need to get your ATA to send SIP INVITE messages out as if it was sitting on your public IP address.
Then, so you can actually receive the incoming audio stream from the other party when they answer, you need to either set port forwarding in your router, or use STUN to "punch a hole" for incoming RTP stream to get through - I think that's how STUN works.