[arm-allstar] Audio codec g711
chris at redcrosscommunications.org
Sun Jan 7 16:30:56 EST 2018
Thanks for the info!
I love it now, just wondered
Email-Chris at redcrosscommunications.org
> On Jan 7, 2018, at 11:54 AM, David McGough <kb4fxc at inttek.net> wrote:
> 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
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