asterisk users - Mar 2006 - Problem with chan_iax.c implimentation causesbad audio?

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.

We switched to SIP and instantly all problems disappeared.



-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, March 21, 2006 9:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentation
causesbad audio?

On Tuesday 21 March 2006 07:19, Matt wrote:
> I was going to avoid naming names :P   But anyway.. yes it's
> asterlink.  Guys seem nice enough.. and by golly.. when I switched to 
> SIP the termination is crystal clear... so far I'm happy with the 
> service from Asterlink... just wish I could use IAX2.... oh well..
> it really matters not to me HOW I get the audio stream.. just that it 
> works and is stable.
I don't know why you'd avoid naming names.  Asterlink does have good
service, and as I said they are a smart bunch of guys.  I get troubles
with my SIP registrations to them on occasion but that's it.  I have
absolutely no trouble recommending them to anyone.

-A.
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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.