SIP trunk provider distance/location

I figured this topic had been answered but could not locate within search.


I setup a Freepbx distro (applies to all packages) package here in our call center in Tulsa, Ok… Our SIP trunk provider is somewhere on the east coast (800 miles away). I realize that the audio never goes through the trunk provider like a proxy, but instead goes directly from our PBX to the receiving party of our call. But I find myself wondering if there is any need, as far as improved reliability goes, in finding a local SIP provider. Or does it really even matter? Assume for now that the reliability we have now is fine, even though it really hasn’t been tested all that much.


Well I have a trunk setup with SIPSTATION. I’m in London (England) and SIPSTATION is in the US. That seems to work fine.

I’ve got a couple of SIP trunks from a local provider, but the internet path from downtown Vicksburg to the provider’s data center two miles away passes through Atlanta, GA. So really, as Lee says, physical proximity to the data center really is not an issue.

I’d say what does matter is that you might want to go with SIPSTATION, because they support FreePbx, or go local if you can to support local business!

It is not the distance as much as what is in between. Note sip station does not proxy media. Your call audio doesnt have to travel to us then back out so the portion of the call where distance matters (rtp) isnt really a factor.

I think the concept of non-unified Signalling and media is pretty abstract. I think it’s thought of as a “pipe” that you purchase with a specific capacity.

The fact that each call stands on it’s own and can be “handed off” is tough.

There is a great article in the NY Times about how the phone network in the US is degrading. The bottom line is that deregulation has brought a lot of choices and the choices are not equivalent.

SIPStation has multiple “signalling” servers and is peered with tens of thousands of gateways around the world. If you do an sip show channelstats in Asterisk you can grab the RTP address, that’s the media gateway. Do a traceroute and you can see how close it is to where you are calling. It’s an interesting exercise in visualizing SIP call flow.