[asterisk-users] Converting GSM calls to SIP

[Tonty T]

Thanks!

That is some good info, I will check with them.


On Thu, Apr 15, 2010 at 1:49 AM, Vahan Yerkanian <vahan at arminco.com> 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 <jbrower at signalogic.com>wrote:
>
>>  > 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
>>
>>
>>
>> --
>>  _____________________________________________________________________
>> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100415/091568b9/attachment-0001.htm