Greetings!
I have an Asterisk 1.4 box and due to hardware limitations I cannot upgrade atm. So, as long as I understood from different posts, SIP-TLS is not available for 1.4 Then I set up VPN and route all inter-Asterisk traffic into VPN. But for some reason, with all the RTP inside the VPN I start getting packet losses up to 30%. Maybe CPU is too weak, that is yet to be discovered. What I want to ask is - how can I split SIP and RTP traffic? Say, SIP goes via VPN, but after the call is initiated, servers reinvite each other with real IPs. Is that possible at all? Searching on Internet didn't give me a clue. -- With Best Regards Mikhail Lischuk
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