[OpenSIPS-Users] Nat traversal and conntrack

[Grygoriy Dobrovolskyy]

Good day list, from day one i have one unresolved question about Nat
traversal, i know it works, but i dont know how.

Here is the classic situation:

10 phones > Nat > B2BUA

Let's say they are the Aastra phones, local sip is 5060, and local rtp is
3000 by default. Nat is intellegent one, it's a micro linux distribution,
there is absolutely no port forwarding done from Nat to phones, but the
voice works!
How ??

I can understand when the phone calls asterisk, so the rtp goes from phone
to B2BUA so it is outgoing, nat restrictions not apply. But when incoming
call goes in, how Nat know where to send voice ?
My only guess would be conntrack, and B2BUA tag somehow the data.
Can onlyone explain this to me ?

Thank you
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