SIP NAT
From Yate Documentation
(Difference between revisions)
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[[File:SIP_NAT.png]] | [[File:SIP_NAT.png]] | ||
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'''See also''' | '''See also''' | ||
| − | *[http://freshmeat.net/articles/view/2079/ NAT traversal for the SIP protocol] | + | |
| − | *[[SIP_in_Yate#SIP_with_NAT_in_Yate|SIP with NAT in Yate]] | + | * [http://freshmeat.net/articles/view/2079/ NAT traversal for the SIP protocol] |
| + | * [[SIP_in_Yate#SIP_with_NAT_in_Yate|SIP with NAT in Yate]] | ||
| + | * [[Telephony]] | ||
Revision as of 14:03, 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
