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

[Alex Balashov]

On 06/09/2014 09:45 PM, Mark R Lindsey, ECG wrote:
>
>
>
> On Mon, Jun 9, 2014 at 4:14 PM, Alex Balashov <abalashov at evaristesys.com
> <mailto: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.

That makes sense. But:

1) No matter which packet duration you use, a VoIP conversation is still 
going to generate far more PPS than, say, HTTP, which is the sort of 
application that typically traffics in packets close to the MTU, i.e. 1500.

2) That still means RTP packets are one of the more likely things to be 
discarded.

3) Given that this is the case, a discard of a longer packet would 
affect the conversation more than a discard of a shorter one.

Or not?

-- Alex

-- 
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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