[arm-allstar] Audio codec g711

[David McGough]

Hi Chris,

The G.711 CODEC won't currently get you any additional audio quality when 
using the current app_rpt channel drivers--these channel drivers are all 
limited to 4KHz bandwidth (8KHz sample rate). You'll get the very best 
AllStar audio quality using the ulaw CODEC.

I added the G.711 CODEC for interconnectivity with Cisco SCCP phones,
where I run G.711 exclusively. And, SCCP can't dynamically change CODECs
like some other protocols, such as SIP.

As for node interconnections, Asterisk will typically transcode from one 
CODEC to another, as needed.

73, David KB4FXC


On Sun, 7 Jan 2018, "chris novara via arm-allstar" wrote:

> I'm reading that allstar supports a higher quality codec of G.711

I know we use ADPCM now. I'm recording about 55 kB of bandwidth in each 
direction now per connection on my hub.

I'm interested to learn if I change my dedicated nodes to this G.711 does 
it work? If so, can someone tell me how to do this?

If I'm successful in upgrading, if a user connects to me using the 
standard adpcm codec, will we be able to talk?

I would love to try this!




Chris Novara
Phone: 541-778-1175
Eugene, Oregon 
Email-Chris at redcrosscommunications.org
_______________________________________________

arm-allstar mailing list
arm-allstar at hamvoip.org
http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar

Visit the BBB and RPi2/3 web page - http://hamvoip.org