[asterisk-users] Split SIP and RTP to different IP addr

[Johan Wilfer]

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