That's is all the overhead I am trying to avoid. What I need is a DID with unlimited channel, but they do not offer DIDs in that country. I wanted to know for example when I get a DID from lets say Vitelity, with unlimited channel, what are they using to forward the calls via SIP or IAX to my server? If I knew the details of the process, I could probably tell them to used this method and route the short code to me via SIP. And if it requires hardware I could invest in it myself and have them host it.
On Wed, Apr 14, 2010 at 2:46 PM, Jeff Brower <[email protected]> wrote: > > On Wed, Apr 14, 2010 at 10:33 AM, Tonty T <[email protected]> wrote: > > > >> This is a solution they proposed, using GSM gateways, but it wont let me > >> handle 1000 simultaneous calls, the other option was using an E1 but the > >> cost would be too much to deploy 35 E1s to support that many calls. > There > >> might be a better way of doing it. > >> > >> > > If you are planning on having 1000 simultaneous calls, you're going to be > > looking at a hefty price tag one way or the other. Things to consider - > if > > you're going to have 1000 concurrent calls going out over VoIP trunks > (SIP / > > IAX / whatever), you need to have enough bandwidth to comfortably handle > > that many calls (each g729 is 8Kb/s bandwidth (but you need to pay a > license > > fee for each channel of g729), each g711alaw is 64Kb/s, etc). That amount > > of bandwidth won't be cheap, plus the cost of the ITSP giving your 1000 > > concurrent channels to call on. On the other hand, if you have a bank of > > E1's, which support (I think) at max 30 concurrent voice channels, you'd > > need 34 available E1 spans. I'm not sure if you can get 34 spans working > in > > a single asterisk server (there was some discussion about this recently > on > > this list), and you'd have the cost of 34 E1 spans as well. > > All good points. It might be worth mentioning that including IP/UDP/RTP > packet overhead, actual bandwidth is 40 kbps > for G729 and 96 kbps for G711. > > -Jeff > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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