SIP NAT

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/

See also

• NAT traversal for the SIP protocol
• SIP_with_NAT_in_Yate