Hi I am having some sound quality issues with Asterisk V10.12.2-1 and confbridge calls.
A sip trunk is built from and Avaya Communication Manager and the Asterisk box. I have restricted the codec down to G.711A 20ms
Both systems are on the same subnet and the TOS value is set and matches the switches.
The audio is however cutting out and sounds as if there is some packet loss. I get this issue in my lab as well but had always thought it was related to my ESXI server running multiple servers at the same time.
Does anyone have a guide to improving voice quality or know of any common mistakes which relate to sound quality issues of this type?
Any assistance is greatly received