[arm-allstar] Audio codec g711
kb4fxc at inttek.net
Sun Jan 7 14:54:04 EST 2018
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!
Email-Chris at redcrosscommunications.org
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