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 IP+Port instead of the RTP IP+Port declared in SDP in 200 OK. 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 200 OK. This small hack allows most of the SIP CLIENTS when Yate is the SERVER to pass the voice over NAT. |
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[[File:SIP_NAT.png]] | [[File:SIP_NAT.png]] | ||
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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 17:35, 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 200 OK. 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/
