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Re: VoIP QOS best practices
From: Steve Feldman <feldman () twincreeks net>
Date: Mon, 10 Feb 2003 11:17:19 -0800

On Mon, Feb 10, 2003 at 10:34:14AM -0800, Bill Woodcock wrote:

    > QoS isn't necessarily about throwing packets away.  It is more like
    > making voice packets 'go to the head of the line'.  Of course, if you
    > have saturation, some packets will get dropped, but at least the voice
    > packets won't get dropped since they were prioritized higher.

Why bother?  It's a pain in the ass, and doesn't give any noticable

So QoS on the access link can do two things:
 - Reduce jitter on selected packets (by moving them to the head of the queue)
 - Reduce packet loss on selected packets (by preferentially dropping
   non-selected packets, _if_ there is congestion).

So, has anyone done measurements to see if either of these makes
a difference in the real world?

IP phones have jitter buffers to reduce the effects of jitter.
Does reducing packet jitter make a noticable difference?

VoIP can withstand a small amount of packet loss without too much
loss of quality.   Does normal TCP backoff keep the UDP packet loss
low enough in the event of congestions?

It seems that Bill's experience with a real-world deployment indicates
that, _subjectively_, percieved quality without QoS is "good enough".

Anyone have real counter-examples, or real measurements?

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