[nsp] CB-LLQ question

jlewis at lewis.org jlewis at lewis.org
Sat Apr 5 11:52:19 EST 2003


On Sat, 5 Apr 2003, Kostas Anagnopoulos wrote:

> I'd like to ask about CBWFQ with LLQ,is there a limit
> to the LLQ traffic a customer should sent if we (as a provider)
> want to preserve delay/jitter quarantees in "voice" class and in the 
> other classes as well.
> For example if the customer admits 90% of the link speed in the LLQ
> and the remaining 10% goes to bussiness class,what will be the end 
> result in terms of delay and jitter in both classes?will they be preserved
> or such a high precentage of LLQ traffic will starve the other
> class.

If you use CBWFQ/LLQ and set a voice class with priority queueing, can you 
still set any sort of QoS for other classes without the various classes 
impacting each other?

>From what I've seen, using QoS to guarantee a portion of the available
bandwidth to a certain class of traffic doesn't guarantee an absence of
jitter/delay.  I recently did some work setting up QoS for VOIP over both
frame and DSL (both aggregated on an ATM DS3).  LLQ on its own did fine
for the VOIP over frame.  On the ADSL, even with LLQ, there was too much
delay if voice packets got stuck behind data packets, and I had to reduce
the MTU considerably to get rid of jitter.

BTW...while setting this up, I found the examples on CIO to be relatively
unhelpful.  They did introduce the concepts necessary for configuring LLQ
for VOIP, but in my real world, things didn't work like the examples and
final configs ended up quite different from the examples.  I found MLPPP
to be supported but non-functional (in some cases even causing the router
to crash) in some (12.1T versions), functional, but not reliable in other
versions, and functional, but not supported on ATM pvcs in the final
version I ended up running (12.2.14S).

Since I'm only providing the circuits and routers, I had no access to the 
VOIP hardware and don't know if they could have adjusted a jitter buffer.  
I do know that the codec in use was not a "good one" and uses about 10x as 
much bandwidth as the codec I use in my own VOIP (which also seems to be 
relatively immune to jitter).
 
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