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<DIV><FONT face=Arial size=2>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006>Scenario:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006>Two Astlinux
servers, main office/branch office. Calls come in via PSTN (ZAP) or
SIP VoIP provider always at the main office. Inbound call will ring a
number of extensions at main office and one phone located at a branch office
site. Calls are routed to the branch office via IAX with a simple
"DIAL(${LocalExtensions},IAX/${BranchOffice}/${ExtNo}@default)".</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006>Problem:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006>Calls answered in
the main office are clear as a bell regardless of source (ZAP/SIP).
However, calls answered at branch office tend to be "choppy" and seem to be
"simplex" instead of "duplex". It is almost as if a large percentage of
packets are being lost in the transfer. And when both parties speak, its a
toss up which voice actually makes it. Have also noted that ZAP calls tend
to have significant echo at the branch office while at the main office this is
not the case.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006>Notes:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006>I noticed early on
when I was experimenting with various Asterisk configurations and VoIP service
providers, that the quality of sound wtih SIP seemed to be much better than with
IAX. When I finally settled on a VoIP provider for production use, I went
with SIP because it seemed to provide better quality.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006></SPAN></FONT><FONT
face=Arial size=2><SPAN class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006>For my "branch
office trunking" needs, I am once again trying to get IAX to work mainly because
of the superior NAT firewall traversal. But am once again confounded by
poor quality voice. <FONT face=Arial size=2><SPAN
class=390100816-15092006>I have played around with "jitter buffers" related to
IAX quite a bit and never really seemed able to resolve the sound quality issues
with IAX. But I am not an expert and may have missed some simple setting
that might have cleared up the problem.</SPAN></FONT></SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006>The internet
connection between the main and branch offices is quite good. Suspect it
is superior to what most folks would use to do this task. The hardware in
play is also superior to what most folks might use with more than enough CPU
& memory to do the job. I cannot imagine the problem could be related
to transcoding issues as the CPU utilitization on both Astlinux machines
is but a blip on the radar while calls are active.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006>I have tried the
scenario with/without VPN and have gotten same results.
</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006>Problem is also
present on outbound calls made from the branch office which are routed to the
main office for completion.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006>Questions:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=390100816-15092006>Have others noticed
this? Has anyone figured out a way to beat it? Should I consider
just switching my branch office trunk to SIP and be done with it or can IAX be
tweaked to properly do this job? Anyone out there have any tips
for me on how to tweak IAX better?</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006>Regards</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=390100816-15092006>G.Hendershot</SPAN></FONT></DIV></FONT></DIV></BODY></HTML>