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qos for voice

hi to everyone,

i am searching for the following feature in a ciso router:

i have a llq-queue in which i want to forward voip traffic.

we have a three layered modell - core-distribution and access layer.

on the access-router i have no problem regarding to reserve the necassery bandwith for voice traffic

because these offices are rather small and i will to guarantee 80kb/s of bandwith - thats enough and ok.

but a distribution router has about 30 to 40 access-router connected (we are talking about

15 distribution router and about 450 access router).

the problem is now if i want to reserve for 40 accessrouter the bandwith for voice - i will

have to make a reservation for 40*80kb/s=3,2 mb/s only for voice but i only have 2 mb/s

from distribution to the core.

i also know that there will be never a situation, where every wants to phone that means that

i really need the whole 3,2 mb/s for voice.

so we reserved 800kb/s for voice - in the most situation it is ok


the problem ist if the voice-llq queue gets full, ALL voice connections go bad not only

that voice connection who brings the llq queue over the 800kb/s.

i am searching for the following "feature":

a llq queue should act so that if it goes to the configured and guaranteed bandwith and a new

voice stream comes in and the router realizes that - it should drop all packets - but only for that

new placed call - so the phone user gets busy.

the calls which were placed before the llq queue goes full would so have the full qos

and are ok.

i know such thing could be done with rsvp agent, but we have 450 1760 routers in our network

which do not support rsvp agent.

thats the big problem.

the next solution i was thinking about would b WRED - but all voice pakets have the same ip-prec setting.

thanks for any answer !!!


Re: qos for voice

This document describes Quality of Service (QoS) for Voice over IP (VoIP) in a LAN environment by mapping Type of Service (ToS) to Class of Service (CoS) parameters.

Hall of Fame Super Silver

Re: qos for voice

You might think about implementing Call Admission Control (CAC) on CCM or GK if you have multiple clusters of CCM.

There is no mechanism to drop only one voip conversation packets, the router will drop random RTP packets and choppy voice will occur as you descried. With CAC you can control this, and allow only certain number of calls to/from site.

For example assume you have a site with 512kbps of CIR to data center, and your LLQ is prioritizing 33% of voice, this allows you to guarantee 170 kbps for voice, which will be approximately 7 G.729 calls. So for that site location you define 170 kbps. If you have AAR implemented the 8th call will be redirected through PSTN.


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