SIP NAT

From Yate Documentation
Revision as of 16:33, 12 November 2012 by Monica (Talk | contribs)

Jump to: navigation, search

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.


SIP NAT.png


A better explanation can be found at: http://freshmeat.net/articles/view/2079/

Retrieved from "https://docs.yate.ro/wiki/index.php?title=SIP_NAT&oldid=1905"
Personal tools
Namespaces

Variants
Actions
Preface
Configuration
Administrators
Developers