Jay Hennigan wrote:
VoIP by design will have high PPS per connection as opposed to data flows.
At 20 ms sample rates you have 50 pps regardless of the CODEC or algorithm.
Increasing the time per sample to 40 ms would cut this in half but the
added
latency would result in degraded quality. In addition, longer sample times
would suffer much more degradation if there is packet loss.
My point is that sampling length should take into effect the rtt. A rtt
of 200ms tolerates far shorter sampling slices than does 20ms.
