Codecs and WebRTC

Does anyone know if Asterisk would use the lowest band Codec for all participants in a conference or would it use the highest level with each participant independently?
Here’s the setup:
Asterisk is configured to use G.722 followed by G. 711;
Three WebRTC users are connected in a conference. Two of them are configured to use G. 722 followed by G. 711, and the third one is configured to use G. 711 only.
What would the users experience be - will all three users’ devises be using G. 711 or would two of them be using G. 722 and one be using G. 711?

Audio is mixed in signed linear, the conference would be running at 16kHz, the two G.722 participants would get G.722 and the other participant would get G.711.

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