[asterisk-users] Converting GSM calls to SIP

[Vahan Yerkanian]

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 <jbrower at signalogic.com 
> <mailto:jbrower at signalogic.com>> wrote:
>
>     > On Wed, Apr 14, 2010 at 10:33 AM, Tonty T <tonty2 at gmail.com
>     <mailto:tonty2 at gmail.com>> 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
>
>
>
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