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P2P dial for hosted pbx

Posted by conraddewet on Wed, 11/10/2010

Hi guys, we host a Thirdlane PBX for a clients and the question came up.. Is it possible for a phone or dialler to directly call (peer 2 peer) while registered to the pbx?

in other words... the call goes of to thirdlane, as say... ext-101@192.168.0.1, thirdlane then knows that ext-101 is in fact the desk next to you and connects the two devices directly.

Why you ask... Well if the thirdlane server was on the other side of an ADSL line say, then the clients only has the ability to have a handful of simultaneous calls, as each channel that's open will use bandwidth.... however if the actual audio stream was sent peer to peer, the normal 100/1000 network would be limitation on the number of simultaneous. calls... quite a lot more.

In some research that I did a while ago I used PJSIP to establish an inter office call, but that was only when there was no registration. I also see that Asterisk does actually know where the extension is sitting locally in the office if you do a sip show peer ext-101...

Addr->IP : 196.210.XXX.XXX Port XXXXXX
Reg. Contact : sip:101-Euphoria@192.168.1.127:5062

I was reading about "canreinvite"... it seems similare.. or is that just for video etc?

Any thoughts on this... or is this wishful thinking?


Submitted by eeman on Wed, 11/10/2010 Permalink

canreinvite is not the solution you are looking for. It will reign hell on you if one endpoint is behind NAT, additionally your CDR will show very short calls which will mess up your billing. In 1.8 they've changed this around a bit more and added directmedia and directrtpsetup options.

Submitted by k3leland on Tue, 12/07/2010 Permalink

We use canreinvite so that phones on a lan can send their audio to each other directly over the lan and not use up the wan resources. This also generally results in less audio quality issues. NAT is certainly an issue when it comes to re-invites so to solve the nat problem eeman mentions we use the sipproxy eeman sells ;)

Submitted by eeman on Tue, 12/07/2010 Permalink

in 1.8 there are better ways to do connected media instead of reinvites. directmedia= .. the RTP is direct but the sip signalling remains between the pbx and phone.

Submitted by ipfreely on Tue, 12/14/2010 Permalink

We use an Ingate Sbc Infront of asterisk. It tries to keep media direct between between 2 endpoints behind the same natted firewall.

Cheers,
Chris