[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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