I have several Asterisk VMs running in my own facility, but that doesn't change the fact that a particular provider's media gateway that SIP reinvites me to is somewhere non-local.
----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: "Owen DeLong" <o...@delong.com> To: "Mike Hammett" <na...@ics-il.net> Cc: nanog@nanog.org Sent: Monday, July 20, 2015 8:04:24 AM Subject: Re: SIP trunking providers Why not set up a small Asterisk box in a local datacenter and only trunk out the non-local calls? Owen > On Jul 20, 2015, at 03:36 , Mike Hammett <na...@ics-il.net> wrote: > > I want the gateway in Chicago as well. > > I am Chicago based. The end users are Chicago based. Therefore the > origination would be coming from a Chicago area gateway. Half of the calls > (inbound would be guaranteed to be local as they'd be coming in through a > local tandem anyway. Most of the termination traffic would again be to local > numbers, therefore would again have to be through local tandems. > > > > > ----- > Mike Hammett > Intelligent Computing Solutions > http://www.ics-il.com > > ----- Original Message ----- > > From: "Nathan Anderson" <nath...@fsr.com> > To: "Mike Hammett" <na...@ics-il.net> > Cc: nanog@nanog.org > Sent: Monday, July 20, 2015 4:11:37 AM > Subject: RE: SIP trunking providers > > Maybe I'm missing something here, but what does it matter if the RTP from > your perspective ends in Chicago or not? If it does end in Chicago, that only > means they are proxying the audio before sending it on to the actual media > gateway for that call where it finally drops onto the PSTN. So all that > happens is that the audio latency remains the same (or worse, because of the > additional, unnecessary proxy) AND that the actual media gateway remains > hidden from you. You won't be able to actually test and see the latency to > the MG, and you will be under the (false) impression that latency across all > calls is equally "good" because you are only measuring RTT to a specific and > common media proxy. By sending the audio directly to an MG closer to the > point of exit from IP-land, it is taking a more direct route to the callee > than you are seemingly asking for. > > If you're not talking about adding a proxy to the equation, are you expecting > to find a provider in Chicago that immediately goes from IP to PSTN within > Chicago, regardless of the actual destination of the call? Circuit-switched > TDM is not a no-latency connection. Physics is involved here. The farther > apart the caller is from the callee, the more latency there will be, > regardless of the medium. All other things being equal (similar network path, > etc.), I doubt IP packet switching significantly increases the latency over > and above TDM call trunking. But I'm not an expert, and again, if I'm missing > something here, I would love to be proven wrong. > > -- > Nathan Anderson > First Step Internet, LLC > nath...@fsr.com > > ________________________________________ > From: NANOG [nanog-boun...@nanog.org] On Behalf Of Mike Hammett > [na...@ics-il.net] > Sent: Sunday, July 19, 2015 1:04 PM > Cc: nanog@nanog.org > Subject: Re: SIP trunking providers > > I too am looking for the Chicago area. Low volume. I'm looking for people > whose SIP and RTP hit the end of the road in Chicago. Not interested in > someone whose SIP servers are in LA , but will redirect me to the nearest > gateway... without telling me where said gateway is. > > > > > ----- > Mike Hammett > Intelligent Computing Solutions > http://www.ics-il.com > > ----- Original Message ----- > > From: "Rafael Possamai" <raf...@gav.ufsc.br> > To: nanog@nanog.org > Sent: Friday, June 19, 2015 4:40:48 PM > Subject: SIP trunking providers > > Would anyone in the list be able to recommend a SIP trunk provider in the > Chicago area? Not a VoIP expert, so just looking for someone with previous > experience. > > > Thanks, > Rafael > >
