Hi all, I have a sipX deployed behind NAT. I configured it according to http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal.
Now there are three phones, one of them is behind the same NAT with sipX. The other two phones are connected to internet through different ADSLs, both of them are behind NAT. When the call is established between the phone in local NAT and remote user, RTPs are routed through sipX relay. But when the call is established between remote users( NAT behind ADSLs ) RTPs are not routed through media relay, so there is no audio on both sides. Why it does not route RTPs through media relay? What should I do in order to enable media relay in this case? Regards, Arman
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