[asterisk-users] Branch office interconnect - IAX :vs: SIP?

[Gary G. Hendershot]

Scenario:
 
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)".
 
Problem:
 
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.
 
Notes:
 
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.
 
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.  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.
 
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.
 
I have tried the scenario with/without VPN and have gotten same results.  
 
Problem is also present on outbound calls made from the branch office which
are routed to the main office for completion.
 
Questions:
 
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?
 
Regards
 
G.Hendershot
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060915/f0282830/attachment.htm