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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>How are the calls being transferred from
Box A to Box B?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>On what box is the receptionist registered
too?<o:p></o:p></span></font></p>
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10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Daniel Cole<br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, December 11, 2007
9:00 PM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] Call
Quality Issues With 2 Trixbox's - RouterIssue?</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>Hello
Everyone,<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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).<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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.<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>Any
thoughts/ideas are most welcome.<o:p></o:p></span></font></p>
<p><font size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>Thank
you<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><strong><b><font
size=3 color="#666666" face=Arial><span style='font-size:12.0pt;font-family:
Arial;color:#666666'>Daniel Cole </span></font></b></strong><strong><b><font
size=1 color="#666666" face=Arial><span style='font-size:8.0pt;font-family:
Arial;color:#666666'>(CCNA)</span></font></b></strong><strong><b><font
color="#666666" face=Arial><span style='font-family:Arial;color:#666666'> </span></font></b></strong><o:p></o:p></p>
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