Asterisk relay between multiple subnets

asteriskvoip

I wonder what's the best way to go when you have phones on multiple networks which are not directly reachable. I have 3 networks

10.3.x.x
10.6.x.x
10.17.x.x

My asterisk server resides on the 10.3.0.5 IP. The machines from the 10.6 and 10.17 networks are routed here through VPN tunnels. At this point we don't talk about NAT anywhere on the network just pure routing. Since the 10.3.0.5 PBX has routes back to all the subnet's it has no problem to communicate with softphones/hardphones from these ranges.

The problem comes from that Asterisk (as far as I understand) only responsible for the SIP communication part not the Audio/Video transmission which is in P2P fashion done between the devices.

So although a client using sipdroid from 10.6.x.x is able to connect to the pbx (10.3.0.5) and dial a bria client on the 10.17.x.x network once the phone rings out and the call establishes no audio will be transmitted simply because it has no way to directly connect there.

For this there are multiple solutions described in this text:

http://msdn.microsoft.com/en-us/library/ee480411%28v=winembedded.60%29.aspx

What I would prefer is to keep these networks segregated as they are now. What would be the best solution? Is it possible to actually relay through all the audio/video information through the Asterisk server? That would be the best in my case, I using Astlinux there which has a lot of other parts.

Thanks

Best Answer

The default behaviour of Asterisk is to use reINVITEs to connect media between the SIP endpoints directly. The name for this setting has changed between versions of Asterisk, currently it is directmedia. Disabling this should cause Asterisk to proxy the RTP and act as a B2BUA.

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