I have multiple users using the same phone. I have some users that speak into the handset “LOUDLY”. And some users that speak into the handset “softly”.
I would love to use AGC Automatic Gain Control inside of FreePBX. Perhaps within PJSIP.
Does anyone have a suggestion on a good way to implement this?
That’s really the job of the telephone. If people talk too loud, they will over-range the codec, and distort. If they talk too softly, there will be a loss of dynamic range, and therefore higher background noise. The phone needs to adjust the levels so that the full range of the codec is used subject to sufficient margin to prevent clipping. That can’t be done properly once the dynamic range has been limited by the codec.
Nonetheless, there is a dialplan function to set AGC, although it is really intended for analogue lines.
However, it looks like it may only work with the speex codec, even though the documentation doesn’t mention such a restriction.
See also:
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