[VoiceOps] High Quality, Reliable Voice via the Internet / SIPNOC

[PE]

But isn't congestion caused by bytes and not number of packets? So, by that argument, larger packets will fill the queue faster than smaller and thus have a higher propensity to drop? And when it does, it is a bigger chunk of audio so it could actually reduce quality rather than improve it. 

Again, it's all theoretical unless someone has the tools, time, and motivation to publish an article with empirical data. 


On Jun 9, 2014, at 9:45 PM, "Mark R Lindsey, ECG" <lindsey at e-c-group.com> wrote:




> On Mon, Jun 9, 2014 at 4:14 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
>> On 06/09/2014 02:50 PM, Mark R Lindsey wrote:
>> 2. Increase the ptime from 20 ms to 30-40 ms to reduce packet-drop exposure
> 
> Or does this thesis lean on countervailing tendencies, such as overall reduced PPS in a higher ptime scenario?

You're on the right track with ptime. The theory idea is that:

(A) Most packet loss is due to congestion

(B) When congestion occurs the router selects a packet to drop

(C) The routers pick a packet to discard more-or-less at random 

(D) Therefore, A 180 byte packet is just as likely to be dropped as a 1500 byte packet. 

(E) A ptime=20 generates twice the packets as ptime=40, and therefore ptime=20 has twice the exposure to the discards 

(F) You can reduce your exposure to discards by reducing the number of packets you have in the queue.

(G) Reduced discards mean better audio quality.

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