SIP NAT
From Yate Documentation
In order to solve the common problems with lossing voice when passing through NAT, YATE uses the RTP stream's IP+Port instead of the RTP IP+Port declared in SDP in 200 OK. This small hack allows most of the SIP CLIENTS when Yate is the SERVER to pass the voice over NAT.
A better explanation can be found at: http://freshmeat.net/articles/view/2079/
