[asterisk-users] Asterisk - SIP client latency

Jerry Jones jjones at danrj.com
Fri Aug 18 22:13:27 MST 2006


Such an objective question. Everyone, including different users will  
have a different answer.

Is this within an enterprise? at home? with a paid service? what  
codec? pure IP or TDM mix?

I would say anything over 200 is bad, now how close you get to that.....

We try to engineer our on net to sub 100

of course our echo cans tell us the PRI to the PSTN regularly hit  
over 150ms which is ridiculous, and keep getting worse


On Aug 19, 2006, at 12:04 AM, Freddy Setiawan wrote:

> Heya all,
>
> what is the acceptable latency for VoIP calling? 200ms? 300ms?
>
>
> Best Regards,
>
> Freddy Setiawan
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