SIP NAT
From Yate Documentation
(Difference between revisions)
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* [[SIP_in_Yate#SIP_with_NAT_in_Yate|SIP with NAT in Yate]] | * [[SIP_in_Yate#SIP_with_NAT_in_Yate|SIP with NAT in Yate]] | ||
* [[Telephony]] | * [[Telephony]] | ||
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| + | [[Category:SIP]] [[Category:NAT]] [[Category:Server]] | ||
Latest revision as of 12:04, 4 November 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
