[asterisk-users] asterisk 1.4 with around 230 SIP connections

[dave cantera]

speaking of multi-casting voice.  since it isn't likely to get the ip 
phones changed, could an app_multicast do the job?
has anyone thought of doing that?
daveC

Kristian Kielhofner wrote:
> 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.
>
>   

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