2013-02-15 11:26, Mikhail Lischuk skrev:
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.
You probably wants a SIP Proxy (like Kamailio). This way you can have
SIP signalling over VPN or use TLS, and kamailio can talk with asterisk
over udp.
RTP always flows directly between asterisk and your provider, and sip
will use the proxy:
SIP: Provider <- (vpn/tls) -> Kamailio <- (udp) -> Asterisk
RTP: Provider <- ------------------------------ -> Asterisk
Good luck!
--
Johan Wilfer
--
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