[asterisk-users] Call quality - how to debug

[Adrian Marsh]

Hi,

It's a 2mb dedicated leased fibre line, with <50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other channel), which is on the same LAN as the
server, would have the same problem at the same time.

Gut feeling is that A*k was CPU overloaded, or the local LAN was, but
none of the stats show that going on at all, although my CPU stats are
5min samples - so that might hide a 60s of intense CPU activity.

It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram.
Only runs Asterisk.

Adrian


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug


On 2 Jun 2009, at 14:14, Adrian Marsh wrote:

> Hi All,
>
> I've a 1.4.15 A*k server supporting several users (approx 80 total,  
> but <10 sim calls usually).  I've one user who complains of  
> intermittent bad calls, though I suspect the bad calls are across  
> the board, but intermittent.
>
> Inbound calls are via in IAX trunk from Gradwell. CPU stats say that  
> Asterisk never uses more than 4-5% cpu, systems idle besides that.  
> Memory seems ok too. Network utilisation is < 300kbps.  The voice  
> network (clients + server) sit on their own dedicated 100Mb  
> switches.  Stats from the switch say its lightly loaded.
>
> I've turned on voicefile recording.  What we hear, when there is a  
> bad call, is stuttered speech, from BOTH sides (so local SIP client,  
> and remote IAX inbound call).
> Debug from asterisk just shows the call inbound, answered and then  
> hung up as per normal.
>
> I'm at a loss of how to debug the voice issue further, without  
> putting a wireshark PC on the switch, port-mirroring the server and  
> then capturing all of the traffic in a round-robin-type capture and  
> even then I'm not sure what that will achieve.
>
> I'm going to switch from IAX to SIP for the inbound calls for that  
> user and see if that helps.
>
> Any ideas welcome,
>

What internet connection do you have...
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