[Asterisk-Users] voice over atlantic

[David Hajek]

Yep. Thats G729, not GSM.

Btw, GSM codec implemented in Asterisk is EFR?

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Wiley Siler
> Sent: Friday, September 09, 2005 12:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] voice over atlantic
> 
> 
> http://www.digium.com/index.php?menu=product_detail&category=e
> xtras&prod
> uct=G729
> 
>  
> 
> -----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 2:51 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] voice over atlantic
> 
> 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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