asterisk users - Dec 2007 - asterisk 1.4 with around 230 SIP connections

Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 
64 bit 4200+ box
would there be any noticable lag or delay to bring each one of them into
a PAGE mode. so one speaker can talk out on all 230 SIP clients for a 
message.

Would this work?

Thanks,

Jerry

Jerry Geis wrote:
> Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 
> 64 bit 4200+ box
> would there be any noticable lag or delay to bring each one of them into
> a PAGE mode. so one speaker can talk out on all 230 SIP clients for a 
> message.
>
> Would this work?
>
> Thanks,
>
> Jerry
>
>   
 I would have some serious reservations throwing this many clients into 
an app_meetme room which is the foundation layer for the page 
functionality.

-- 
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

On Dec 10, 2007 1:17 PM, Jerry Geis <geisj at pagestation.com>
wrote:
> Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a
> 64 bit 4200+ box
> would there be any noticable lag or delay to bring each one of them into
> a PAGE mode. so one speaker can talk out on all 230 SIP clients for a
> message.
>
> Would this work?
>
> Thanks,
>
> Jerry
>
I would also be really concerned about the ability for the NIC to
serve up all of those RTP streams...

50pps x 230 = 11,500pps

It would be nice to have some support for RTP multicast or something.
Obviously this would require changes in Asterisk AND support in each
phone, but it would be really cool.  I think I've seen some
Linksys/Sipura devices support it.

-- 
Kristian Kielhofner