SIP NAT
From Yate Documentation
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*[http://freshmeat.net/articles/view/2079/ NAT traversal for the SIP protocol] | *[http://freshmeat.net/articles/view/2079/ NAT traversal for the SIP protocol] | ||
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Revision as of 16:40, 12 November 2012
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
