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.

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

SIP NAT.png


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