[asterisk-users] Converting GSM calls to SIP

[Jeff Brower]

> On Wed, Apr 14, 2010 at 10:33 AM, Tonty T <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