Audio quality depends on jitter timing. Too low and you get dropouts, too high and you get uncanny valley with the delays and pauses. You should never have to twiddle these settings since internet routing is so dynamic. But things to look at include jitter sample size (20ms is a good start), codec selection (lower bandwidth), and server resource usage (more cpu or ram). I don’t use mumble but ran an asterix server for a decade and these where the three things that mattered most to reduce conference call latency
In this case, it's mostly getting the levels set up and cutoff set up correctly since by default it cuts off when you aren't talking (and thankfully doesn't have AGC). There is a push to talk mode, but most people that I know don't use it.
