[asterisk-users] Call quality - how to debug

[Adrian Marsh]

Hi Dave,

You're quite right, it's a dedicated down and uplink to my ISP, and
Gradwell also has fibre connection into that ISP (so short hop to them)

The reason I don't think it's the fiber link, is that Asterisk recorded
the conversation as two channels. IN (from Gradwell), and OUT (from the
Cisco phone, that's on the same LAN as the asterisk server).  And I hear
distortion on both sides, at the same time.  As thats what asterisk
"hears", and that part of the call is a same-LAN RTP stream, pre-ISP,
then that's why I don't think it's the IAX link.

That said, I've not got complaints from users making internal calls.  So
my thinking was maybe its an IAX/SIP conversion thing

As a test, I've switched my account, and the problem account to inbound
SIP, to see if that makes a difference. That makes it 100% SIP.

Next step, memory upgrade and the A*k upgrade.

Thanks,

Adrian

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 16:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug

Unless I've misunderstood and you're not running ANYTHING but voice over
that internet uplink?

<snip>
So theres no web browsing etc on that 2mb circuit.
</snip>

In which case, I stand corrected and you don't need QOS.

-Dave


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug

2mb is small potatoes... unless you mean MegaBytes instead of
Megabits...

I am assuming you've already implemented QOS? That is likely the problem
if the intermittent quality issue is only on calls between internal and
external parties.

If someone tries to access the yahoo homepage while someone else is on
the phone, without QOS, they are really going to be fighting for that
bandwidth.

-Dave

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Tuesday, June 02, 2009 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

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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