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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

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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.

  • Hello. Thanks for your response. The documentation says -if I understand correctly- that the voip clients still have to be able to reach eachother: Both approaches come with the caveat that the endpoints must be able to address each other directly, so it can't be that one endpoint is behind NAT on a private network that only Asterisk can see and the other endpoint cannot. But if that's taken care of, or you have a far-end NAT traversal solution in place to go with it, then you can do media release on Asterisk. – immoune Aug 20 '14 at 15:06
  • That is not correct, the clients do not need to be able to reach eachother unless directmedia is enabled. Asterisk will use reinvites to get out of the way to reduce load and decrease media latency. I have edited the original comment though as it seems with the change to using directmedia, it is default for it to be enabled – brent Aug 20 '14 at 15:30

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