SIP NAT

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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 the 200 OK response. This small hack allows most of the SIP CLIENTS when Yate is the SERVER to pass the voice over NAT.

SIP NAT.png

A better explanation can be found at: http://freshmeat.net/articles/view/2079/

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