[Asterisk-Users] Call quality problems

[Michael Welter]

Doug Lytle wrote:
> Michael Welter wrote:
>> Doug Lytle wrote:
>>> Michael Welter wrote:
>> The machine is totally idle.
>>
>> The T1 vendor noticed 2% packet loss during a ping flood originating
>> from outside.  We changed the Cisco IAD, and there is no longer packet
> 
> I've noted from employees that the volumes levels on the phones 
> themselves, when set too high will cause crackling.  Does the crackling 
> coincide with talking on the local side?
> 
> What firmware are you running on the Polycoms?

I'm not on site, but I remember 1.6.4.

It's not really crackling or popping that's the problem.  The problem is
with dropouts.  It also seems that the trailing edge of each word will
sometimes be lost (possibly a dropout).

If you're familiar with the WWV time signal (303-499-7111), for the
first 45 minutes of each hour there is a tone interrupted by a click
every second (during the last 15 minutes it's just the clicks).  When I
listen to this on the Asterisk system, the tone only lasts for a
fraction of a second and then silence until the next click.

Is the phone (or Asterisk) performing echo suppression that drops the
last part of the tone?

Also, there are no ZAP cards in the system.  What timing source does SIP
use to play the incoming media stream?

Thanks for your comments, Doug.

-- 
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike at TelecomMatters.net
www.TelecomMatters.net