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Re: The Qos PipeDream [Was: RE: Two Tiered Internet]
Joe Maimon wrote:
VoIP by design will have high PPS per connection as opposed to data flows.Chris Woodfield wrote:In that vein, and not quite on this topic, it would be real nice if voip applications made an effort to stop abusing networks with unneccessarily large pps.
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.
Something about intelligent edges? The payload length of voip applications often has a lot to do with rtt. Adapting payload length to the actuall average rtt could have a positive effect on pps throughput.I'm not sure why you say the payload length has much to do with RTT. Serialization delay on slow edge links could increase RTT, but this would worsen substantially with longer samples (assuming the same CODEC and compression). Payload length is a factor of the sample length and compression algorithm. More efficient compression will result in smaller payloads but overhead becomes a higher percentage of the overall flow. Only lengthier samples will reduce PPS, and the added latency in a two-way conversation will substantially reduce call quality.
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