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<P>Hello Everyone,</P>
<P>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.</P>
<P>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).</P>
<P>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.</P>
<P>Now for the Fun part!<BR>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.</P>
<P>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).<BR>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.</P>
<P>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.</P>
<P>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.</P>
<P>Any thoughts/ideas are most welcome.</P>
<P>Thank you</P></FONT></DIV>
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<P class=MsoNormal align=left><STRONG><SPAN
style="COLOR: #666666; FONT-FAMILY: Arial">Daniel Cole<SPAN
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