SIP NAT
From Yate Documentation
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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. | + | 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. |
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| + | 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/ | A better explanation can be found at: http://freshmeat.net/articles/view/2079/ | ||
Revision as of 13:03, 31 May 2013
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
