[Asterisk-Users] Problem with chan_iax.c implimentation causesbadaudio?

[Adam Robins]

We upgraded all five servers to 1.2.4.  We tried trunking/notrunking.  

End users use an IAX2 softphone on their desktop PCs.  Agents are VLANed
and all IAX2 traffic is QoS'd on all LAN and WAN legs.  Calls flow from
the agents to the local Asterisk server as IAX2/ulaw.  Then they went
over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well).
Calls get to the PSTN from the central site via PRI on TE410P cards.

Point is that it worked fine for 6-9 months before the Asterisk 1.2.4
upgrade.



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, March 21, 2006 10:21 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentation
causesbadaudio?

On Tuesday 21 March 2006 09:47, Adam Robins wrote:
> We have three remote call center Asterisk servers communicating with 
> two central Asterisk boxes over a private IP-VPN with QoS.  All 
> systems were running Asterisk 1.0.7 communicating via IAX2 with little

> or no quality issues at all.
>
> Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was
horrific.
> We tried with/without jitterbuffer.  We messed with every jitterbuffer

> parameter.  We tried G729/ilbc/ulaw.  It was a total mess.

Did you upgrade all three boxes?  Did you try disabling trunking?  What
was your "last mile" solution?  (i.e. what did the end-users speak into,
and how did their calls get to the PSTN?)  If it was to a far-end
Asterisk box, what version where they running?  Were you communicating
using IAX2 to them too?  
Did they upgrade to 1.2.4 as well?

I am running SVN trunk with IAX2 and SIP and have *zero* issues.

-A.
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