SIP Router

Hello all.

We’re running FreePBX Distro on 7 machines spread out around our state. Sites are interconnected via VPN. IAX trunks created between all boxes, as well, and run through the VPN connection. VPN Connections are made over standard cable business internet connections. About 100 users or so at the moment.

Right now, we’re in extension mode. People need to take their phone with them if they want to visit a different office and keep receiving calls. They want to enable user & devices mode, and log into phones at whatever location they travel to. However, we need to keep the 7 different systems as they have local numbers via a PRI in each location.

We have a central location with a stronger server setup; more redundancy, better network, etc. If we provisioned everything from there, and routed all calls from the PRIs at the satellite offices to the central location, then back out to whichever user needed the call, what sort of network traffic would we be looking at? Is this a feasible setup?

I guess I was less curious about the number of bytes/sec being used and more about how the traffic will be routed around. If I’m in office A, and I want to call someone in office B, but the central office is office C, does all my media traffic get routed through office C (like A->C->B) or is RTP smart, and link A->B after figuring out where everybody is and establishing a call?

Does this make sense?

It makes sense - and it’s an excellent question.

From what I understand, RTP is designed to be used exactly like you are hoping.

Depending on what options you use, RTP is generally pretty smart. My experience with this is that the system by default is configured to pass the RTP traffic off to be direct. Of course, the options you choose along the way can mess that up, so some trial-and-error may be involved.