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<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=943565705-12122007>Hi Paul,</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=943565705-12122007></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=943565705-12122007>Where abouts exactly is the best place to get these
figures from?</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=943565705-12122007></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=943565705-12122007>I have been checking <EM>iax2 show netstats</EM>, 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?</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=943565705-12122007></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=943565705-12122007>Thanks,</SPAN></FONT></DIV>
<DIV> </DIV>
<DIV class=Section1>
<P class=MsoNormal align=left><STRONG><SPAN
style="COLOR: #666666; FONT-FAMILY: Arial">Daniel Cole<SPAN
style="mso-spacerun: yes"> </SPAN></SPAN></STRONG><STRONG><SPAN
style="FONT-SIZE: 8pt; COLOR: #666666; FONT-FAMILY: Arial">(CCNA)</SPAN></STRONG><STRONG><SPAN
style="COLOR: #666666; FONT-FAMILY: Arial"> </SPAN></STRONG><SPAN
style="FONT-SIZE: 10pt; COLOR: #666666; FONT-FAMILY: Arial"><BR></SPAN></P>
<P class=MsoNormal><SPAN style="COLOR: black"> </SPAN><SPAN
style="FONT-SIZE: 18pt; COLOR: green; FONT-FAMILY: Webdings; mso-bidi-font-family: Arial">P</SPAN><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"> </SPAN><SPAN
style="FONT-SIZE: 10pt; COLOR: green; FONT-FAMILY: 'Harmony Text'">Please
consider the environment before you print this e-mail or any
attachments.</SPAN></P></DIV>
<DIV> </DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> Paul Hales
[mailto:phales@asteriskit.com.au] <BR><B>Sent:</B> Wednesday, 12 December 2007
4:40 PM<BR><B>To:</B> Daniel Cole<BR><B>Subject:</B> RE: [asterisk-users] Call
Quality Issues With 2 Trixbox's - Router Issue?<BR></FONT><BR></DIV>
<DIV></DIV><BR>Hmmm......wierd....<BR><BR>Are you getting an weird
jitter/latency figures in the CLI?<BR><BR>PaulH<BR><BR><BR>On Wed, 2007-12-12 at
16:37 +1100, Daniel Cole wrote:
<BLOCKQUOTE TYPE="CITE"><FONT size=2><FONT color=#0000ff>G729 All
Around.</FONT></FONT> </BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><FONT color=#000000></FONT> </BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><B><FONT color=#666666>Daniel Cole
(CCNA) </FONT></B><BR><BR><FONT color=#000000> </FONT><FONT
color=#00ff00>P</FONT><FONT color=#000080> </FONT><FONT color=#00ff00>Please
consider the environment before you print this e-mail or any
attachments.</FONT><BR><BR><BR></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><FONT color=#000000></FONT> </BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><BR></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<HR>
<BR><B><FONT size=2><FONT color=#000000>From:</FONT></FONT></B><FONT
color=#000000><FONT size=2> Paul Hales [mailto:phales@asteriskit.com.au]
</FONT></FONT><BR><B><FONT size=2><FONT
color=#000000>Sent:</FONT></FONT></B><FONT color=#000000><FONT size=2>
Wednesday, 12 December 2007 4:10 PM</FONT></FONT><BR><B><FONT size=2><FONT
color=#000000>To:</FONT></FONT></B><FONT color=#000000><FONT size=2> Daniel
Cole</FONT></FONT><BR><B><FONT size=2><FONT
color=#000000>Subject:</FONT></FONT></B><FONT color=#000000><FONT size=2> Re:
[asterisk-users] Call Quality Issues With 2 Trixbox's - Router
Issue?</FONT></FONT><BR><BR><BR></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><BR></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><BR><FONT color=#000000>What codec are you
using?</FONT><BR><BR><FONT color=#000000>PaulH</FONT><BR><BR><BR><FONT
color=#000000>On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote:
</FONT><BR>
<BLOCKQUOTE TYPE="CITE"><FONT size=2><FONT color=#000000>Hello
Everyone,</FONT></FONT><BR><BR><FONT size=2><FONT color=#000000>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.</FONT></FONT><BR><BR><FONT size=2><FONT
color=#000000>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).</FONT></FONT><BR><BR><FONT
size=2><FONT color=#000000>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.</FONT></FONT><BR><BR><FONT size=2><FONT color=#000000>Now for
the Fun part!</FONT></FONT><BR><FONT size=2><FONT color=#000000>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.</FONT></FONT><BR><BR><FONT size=2><FONT color=#000000>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).</FONT></FONT><BR><FONT
size=2><FONT color=#000000>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.</FONT></FONT><BR><BR><FONT size=2><FONT
color=#000000>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.</FONT></FONT><BR><BR><FONT size=2><FONT
color=#000000>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.</FONT></FONT><BR><BR><FONT size=2><FONT color=#000000>Any
thoughts/ideas are most welcome.</FONT></FONT><BR><BR><FONT size=2><FONT
color=#000000>Thank you</FONT></FONT><BR><BR><BR><B><FONT
color=#666666>Daniel Cole
(CCNA) </FONT></B><BR><BR><BR><BR><BR><FONT
color=#000000> </FONT><FONT color=#00ff00>P</FONT><FONT color=#000080>
</FONT><FONT color=#00ff00>Please consider the environment before you print
this e-mail or any attachments.</FONT><BR><BR><BR><PRE><FONT color=#000000>_______________________________________________</FONT>
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