I am pleased to announce that the Opus codec has been finally enabled on Asterisk SIP, Avaya, and Cisco services here on PhoneNet.
The reason for the long delay doing this is there were some strange bugs that I had to solve primarily around DTMF reception and digits being detected multiple times from Cisco and Avaya devices but I have resolved these.
If your device supports Opus and you’re calling from an Opus enabled endpoint you should now hear the higher quality audio. The bitrate on PhoneNet Radio (791-9799) has been increased from SLN16 to SLN48 and sounds quite frankly rather glorious, and the main PhoneNet IVR (791-0000) is also now Opus.
I still have some more testing to do on IAX as the last time I enabled it on IAX there were some issues with inbound calls from older Asterisk switches so for now it’s SIP only.
Your device will not always use Opus as I am also rolling out common codec negotiation to remove unnecessary transcoding on the network. For example, if you receive a call that’s using G711 then that’s the codec we’re going to present to your device as the preferred one.
As of now the official codecs for PhoneNet are:
- Opus (up to 48K between switches, except for Avaya which is 24K. Internal calls on CUCM can go higher)
- G722
- G711μ
- G711A
We do not support low bandwidth codecs such as G729 or ILBC.