[asterisk-users] Bad audio quality on SIP

[Michelle Dupuis]

The problem will be on the "outside" of your Asterisk PBX.  In other words,
your asterisk server's external NIC (or if just one NIC), connection to your
firewall/router,  to you voip provider.

You need to run tracert's from your Asterisk box to your voip provider.  

QoS on the windows clients is useless (and doesn't matter in this case).
QoS is often misunderstood.  Without knowledge of the protocols you are
running, your network admin could not have setup the router/firewall to
shape traffic properly.  Prioritizing based on QoS bits offers minimal
benefits.

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 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bad audio quality on SIP

It's all in the local LAN network - client computers (with SIP
softphones) are connected and registered at Asterisk SIP proxy via 100 MB
connection each.

The QoS is enabled under TCP/IP protocol in LAN connection in Windows (cause
SIP softphones are running in Windows environment), and tos in sip.conf is
set to 0x18. Unfortunately I don't have access to switch to tell you how
it's set up there, but the network technicians said it is enabled.

-----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 3:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Bad audio quality on SIP

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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