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

[Alexander Lopez]

How are the calls being transferred from Box A to Box B?

 

On what box is the receptionist registered too?

 

 

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel
Cole
Sent: Tuesday, December 11, 2007 9:00 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Call Quality Issues With 2 Trixbox's -
RouterIssue?

 

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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