[asterisk-users] Asterisk as SoftSwitch - Hardware

Sammy Govind govoiper at gmail.com
Tue Nov 8 13:23:39 CST 2011


Hey Sunny,

I think your initial post on what you're looking for don't really tells
much. I think initially you were looking at a different architecture than
now i.e Kamailio+RTPproxy, this changes a lot of things.

If you dont want transcoding and thinking on using Kam+Rtpproxy then I
think asterisk isn't required any more. If that's not the case then for
1200 CCs you'll be required to put in multiple asterisk servers behind
Kamailio/RTpproxy Server.

Share some more details and I'm expecting that your design is going to
change.

Regards.
Sammy.

On Tue, Nov 8, 2011 at 9:31 PM, Sunny <no7find at gmail.com> wrote:

> Jeff,
>
> Kamailio + rtpproxy
> Do you know how to make these configuration work?
>
> I know this is not the best place to ask that question.
>
> Thanks,
> Sunny
>
>
> On 3 November 2011 19:09, Jeff Brower <jbrower at signalogic.com> wrote:
>
>> Sunny-
>>
>> > I was thinking in Kamailio, but this sip proxy handles only the
>> > SIP signalling traffic, no media processing.
>>
>> Kamailio + rtpproxy.
>>
>> -Jeff
>>
>> > On 3 November 2011 17:07, Nick Khamis <symack at gmail.com> wrote:
>> >
>> >> Shouldn't you be using a Proxy?
>> >>
>> >> Nick.
>> >>
>> >> On Thu, Nov 3, 2011 at 1:04 PM, Sunny <no7find at gmail.com> wrote:
>> >> > Hi list,
>> >> > Could anyone tell me what is the "recommended" hardware to a system
>> for
>> >> > following configuration:
>> >> > SBC --> Asterisk (SS) --> Carrier GW
>> >> > Asterisk should work as a Class 4 SoftSwitch, with following
>> >> functionalists:
>> >> > -> Do the IP Authentication
>> >> > -> All communications on RTP/G729 (no transcoding required)
>> >> > -> Load of 1200 concurrent call sessions
>> >> > -> No call routing required
>> >> > Thanks in advance,
>> >> > Sunny
>>
>>
>
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