SIP NAT
From Yate Documentation
(Difference between revisions)
| Line 7: | Line 7: | ||
'''See also''' | '''See also''' | ||
| − | [http://freshmeat.net/articles/view/2079/ NAT traversal for the SIP protocol] | + | *[http://freshmeat.net/articles/view/2079/ NAT traversal for the SIP protocol] |
| − | [[http://docs.yate.ro/wiki/SIP_in_Yate#SIP_with_NAT_in_Yate]] | + | *[[http://docs.yate.ro/wiki/SIP_in_Yate#SIP_with_NAT_in_Yate|SIP_with_NAT_in_Yate]] |
Revision as of 17:40, 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
