Thanks! That is some good info, I will check with them.
On Thu, Apr 15, 2010 at 1:49 AM, Vahan Yerkanian <[email protected]> wrote: > On 4/15/10 1:26 AM, Tonty T wrote: > > 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. > > If their switch doesn't support SIP or doesn't have SIP module installed, > there isn't much you can do to get traffic in pure SIP form. Ask them if > they can and willing to serve you the traffic via multiple E3 or even > better, STM fiber links. STM over fiber is the cheapest way to transport > that much channels by means of cabling - you just need 2 strands for TX/RX > or even 1 strand if you go with WDM. However the carrier crade hardware for > it is *very expensive*. On your side you demux STM link(s) into E3/E1s using > expensive carrier grade equipment like Cisco's $25k+ (used) STM cards for > Cisco 7500 and up models or if you're smart enough to know where to dig, > dirt cheap (~$2K for STM-1 to 24E1) Taiwanese/Chinese media converters. > > Oh and yes, this isn't a task for a single Asterisk server. The most I've > seen a single box capable of is 16 E1s (2 x 8E1 cards) in a single chassis > doing only G711a to SIP conversion. > > HTH, > Vahan > > > 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 >> > > > > -- > _____________________________________________________________________ > -- 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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