[asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?

[Paul Hales]

'iax2 show channels'....maybe....

I have a feeling this is going to be one of those ugly ones where it's
going to be a pain to troubleshoot...

Offhand - have you tested 'trunk=yes' vs 'trunk=no'?

PaulH


On Wed, 2007-12-12 at 17:00 +1100, Daniel Cole wrote:
> Hi Paul,
>  
> Where abouts exactly is the best place to get these figures from?
>  
> I have been checking iax2 show netstats, which does give some figures.
> These appear not to be accurate though, as when there are multiple
> inter-site calls, the result for one channel of audio can show no
> jitter or latency, but another will have some jitter and latency. Or
> is this a weird way for the problem to show its head?
>  
> Thanks,
>  
> Daniel Cole  (CCNA) 
> 
> 
>  P Please consider the environment before you print this e-mail or any
> attachments.
> 
> 
>  
> 
> 
> ______________________________________________________________________
> From: Paul Hales [mailto:phales at asteriskit.com.au] 
> Sent: Wednesday, 12 December 2007 4:40 PM
> To: Daniel Cole
> Subject: RE: [asterisk-users] Call Quality Issues With 2 Trixbox's -
> Router Issue?
> 
> 
> 
> 
> Hmmm......wierd....
> 
> Are you getting an weird jitter/latency figures in the CLI?
> 
> PaulH
> 
> 
> On Wed, 2007-12-12 at 16:37 +1100, Daniel Cole wrote: 
> > G729 All Around. 
> > Daniel Cole  (CCNA) 
> > 
> >  P Please consider the environment before you print this e-mail or
> > any attachments.
> > 
> > 
> > 
> > 
> > ____________________________________________________________________
> > 
> > From: Paul Hales [mailto:phales at asteriskit.com.au] 
> > Sent: Wednesday, 12 December 2007 4:10 PM
> > To: Daniel Cole
> > Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's -
> > Router Issue?
> > 
> > 
> > 
> > 
> > What codec are you using?
> > 
> > PaulH
> > 
> > 
> > On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote: 
> > > Hello Everyone,
> > > 
> > > We have recently installed a pair of Trixbox servers in for a
> > > client of our. They have two locations, with one server each. The
> > > servers terminate 3 standard POTS lines into a Sangoma A200D card.
> > > The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD,
> > > Dual Core Xenon Processors). We are using Trixbox 2.2, and G729
> > > all around.
> > > 
> > > Each site has two (2) 512k/512k ADSL connections terminating into
> > > a Cisco 877W router (using an additional 'dumb' modem in a
> > > separate VLAN for the extra dsl connection). Using policy based
> > > routing, all Voice Data goes over one DSL connection (the one that
> > > terminates directly into the router), and all other traffic (e.g.
> > > Web and VPN) goes out the second connection (the bridged dumb dsl
> > > modem).
> > > 
> > > We are also the ISP for this client, and as thus we have full
> > > monitoring of our Layer 2 and Layer 3 networks. From our analysis,
> > > it doesn't appear that there is any issue in these networks. We
> > > have other customers using the VoIP service, who have not
> > > complained of these issues.
> > > 
> > > Now for the Fun part!
> > > The client is complaining of issues with inter-site calls. They
> > > are reporting issues with crackly and broken speech, and horrible
> > > jitter (or packet loss). This presents a huge issues, because they
> > > have one receptionist answering all calls for both sites. So if a
> > > call comes in from the other site, it automatically an inter-site
> > > call, and the quality falls out of it. If the call is then
> > > transfered back to the originating site, the audio 'bounces'
> > > between the two sites, which add to the call quality degradation.
> > > 
> > > We have been monitoring the router while these incidents have been
> > > reported, and it does not appear to be a bandwidth issue. The DSL
> > > tail used for Voice gets to no more then 120k in each direction
> > > (we have tested the links, and can pull data at 53k/s between
> > > sites). CPU usage floats at around 20-25% under load. The router
> > > has only shows major packet loss (that we can tell) when REALLY
> > > pushing it in testing (e.g. 10+ calls between sites).
> > > We have enabled the SIP jitter buffer, as well as the IAX jitter
> > > buffer, which appeared to make a huge difference, but the issue is
> > > still ongoing.
> > > 
> > > These issues have also been reported with some outbound VoIP
> > > calls. Internal calls, and calls directly in or out of the Sangoma
> > > card are clear, with no issues reported.
> > > 
> > > Does anyone have any thoughts on what could be causing these
> > > issues? We have been racking our brains here, and have tried
> > > everything that we can think of. These system is a million times
> > > better then what is what when it was first installed, but it is
> > > still not where it should be in terms of quality.
> > > 
> > > Any thoughts/ideas are most welcome.
> > > 
> > > Thank you
> > > 
> > > 
> > > Daniel Cole  (CCNA) 
> > > 
> > > 
> > > 
> > > 
> > >  P Please consider the environment before you print this e-mail or
> > > any attachments.
> > > 
> > > 
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