RE: VoIP QOS best practices

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.

Ray Burkholder

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
benefit.

                                -Bill

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.

Thats what I meant too...

To qualify further on where it needs to be deployed, its required on whatever
the slowest link in the typical path to "the Internet". What I mean is that if
you download your email you will utilise the whole bandwidth of the slowest link
in the chain, this may be a dialup modem but more likely in the office to be
your T1, you dont want this full utilisation of the link (which will occur in
small bursts of a few seconds, dont forget with voice we are interested in per
second traffic volumes not 5 minute averages!) to affect the jitter you need to
implement priorities at this point.

Steve

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?
  Steve