given opus as a variable bitrate codec applied to voip rtp, i can verify
that the bitrate really changes
by a few kbps between max and min. as i understood, the bitrate variation
is dependent on the audio
source. are there any other factors which would affect this varying
bitrate? like for example: packet losses,
jitter, latency, etc. Will it automatically shift to lower bitrate /
sampling rate whenever it detects network issues?
Kelvin Chua
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