[asterisk-users] Bad audio quality on SIP

[Michelle Dupuis]

If it's a random phone on the SIP side, we have to look further upstream.
While jitterbuffers may help, in my opinion they mask a problem.

What type of connection do you have to the internet?  Have you done
tracert's to your voip provider?  What do they look like?

When you say that you do QoS - how?  What device and settings/app helper?

MD

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Asterisk
Sent: Monday, February 12, 2007 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bad audio quality on SIP

Well, the PSTN side is complaining about a random phone on the SIP side.
Yes, they do hear choppiness.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Monday, February 12, 2007 2:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Bad audio quality on SIP

If the PSTN side is only complaining about conversations with a single phone
on the SIP side, look at the SIP phone.

Check the settings for that SIP phone/PC (VAD disabled, NIC settings,
runaway processes).  Do PSTN callers here choppiness from the SIP phone
caller?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Asterisk
Sent: Monday, February 12, 2007 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Bad audio quality on SIP

Hi guys,

I have the following configuration:

10 SIP softphones <--> Asterisk <--> PSTN

Audio is always good on SIP softphone side, but callers from PSTN side
*sometimes* complain that the audio quality is bad (and volume low). The QoS
is turned on on the computers where SIP softphone is installed, and the tos
setting in sip.conf is set to 0x18.

The interesting thing is that usually only one SIP softphone user (but each
time someone else) would get complaints like that ... others seem to work
okay.

What could be wrong?

Thanx,
Alex

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