[Asterisk-Users] voice over atlantic

David Hajek david.hajek at systinet.com
Thu Sep 8 14:50:39 MST 2005


Probably missing something here. Never heard of GSM commercial licence
for asterisk.

Do you have any URLs?

Thanks.

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Wiley Siler
> Sent: Thursday, September 08, 2005 11:09 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] voice over atlantic
> 
> Pay the license fee and get the GSM codec would probably be best.
> The fee is nominal and the codec is a good one...
> $0.02
> 
> W
> 
>  
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> David Hajek
> Sent: Thursday, September 08, 2005 1:50 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] voice over atlantic
> 
> 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?
> 
> Thanks for any hints.
> 
> --
> David
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