[asterisk-users] Call quality - how to debug

[Adrian Marsh]

Yeah, I know,  but when I last tried an upgrade to 1.4.18 it broke the
whole IAX connectivity and I was forced to drop back.

I'll go:

1) Memory upgrade first
2) Clone the machine, and upgrade to latest 1.4.x

However - my question would still stand, how exactly would I be able to
debug whats going on in the RTP stream? And why its stuttering
(sometimes halfway through a call).

Any tips or tricks for actually debugging within Asterisk ?

Thanks,

Adrian

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Darrick
Hartman
Sent: 02 June 2009 15:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug

Do you have any idea the number of bugs that have been fixed since 
1.4.15?  Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug
this.

On 06/02/2009 08:58 AM, Adrian Marsh wrote:
> 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,

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