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

[Paul Hales]

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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