[Asterisk-Users] voice over atlantic

[David Hajek]

Nice. Thanks.

What Asterisk version? Can you lookup jitterbuffer settings?

Thanks a lot.

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> asterisk groups
> Sent: Thursday, September 08, 2005 7:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] voice over atlantic
> 
> David,
> I have IAX trunks running between the US and S. America using 
> the GSM codec and jitterbuffer=yes and the quality seems very 
> good to my ears.
> Don't have the details of the jitterbuffer parameters right 
> now but hopefully this will give you some useful feedback.
> 
> Good luck.
> 
> 
> On Thu, 2005-09-08 at 16:49 -0400, David Hajek wrote:
> > Hi-
> > 
> > I'm using IAX between two boxes, where one box is located in US and 
> > the second in Europe. I'm trying to achieve the best voice 
> quality and 
> > mainly reliability between these boxes and looking for hints and 
> > experience of others.
> > 
> > Facts:
> > - Asterisk 1.0.7
> > - RTT varies from 130-170 ms, depends on time and actual Internet 
> > throughput
> > 
> > Questions:
> > - What is the sugested codec for such setup? Now I'm using 
> ULAW, but 
> > realizing it may not be the best choice. Sometimes I can 
> hear broken 
> > audio. Maybe speex is better choice?
> > - Jitter buffer, yes/no? What are the suggested values. 
> Currently I'm 
> > using these values:
> > jitterbuffer=yes
> > dropcount=10
> > maxjitterbuffer=500
> > maxexcessbuffer=300
> > minexcessbuffer=20
> > jittershrinkrate=2
> > - Trunking? Is it reliable enough?
> > 
> 
> 
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