The Qos PipeDream [Was: RE: Two Tiered Internet]

Sun Dec 18 22:00:03 UTC 2005

Joe Maimon wrote:

> Chris Woodfield wrote:
>>
>> One thing to note here is that while VoIP flows are low volume on a  
>> bits-per-second basis, they push substantially more packets per  
>> kilobit than other traffic types - as much as 50pps per 82Kbps flow.  
>> And I have seen cases of older line cards approaching their pps  
>> limits when handling large numbers of VoIP flows even though there's  
>> plenty of throughput headroom. That's not something LLQ or priority  
>> queueing are going to be able to help you mitigate at all.
>>
>> -C
> 
> 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.

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.

> 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.

-- 
Jay Hennigan - CCIE #7880 - Network Administration - jay at west.net
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