Others are providing you generally good answers. I would add one thing
from my experience.
Your upload speed of 512k is the really limiting factor in your setup.
A G.711 encoded calls is going to consume 80 kbps each inbound and
outbound. Two calls = 160 kbps, etc. To ensure perfect voice quality
you will end up severing off some of your outbound bandwidth to the
voip traffic pretty much on a full time basis. If you assign 200k, to
be safe, to the voip traffic you'll be living with 312k or less for the
rest of you upload activity.
Be very certain that you measure your actual upload data rate. Often
it's less than quoted and even a little misconfiguration will result in
poor quality calls.
In the end I switched to using G.729a coded calling to the ITSPs that I
use. That drops the per leg data rate to 32 kbps including ethernet
overhead. The call quality is still very good. And I only slice off 128
kbps to allow four simultaneous calls into my Asterisk server.
By reducing the data requirement for voip you ease the requirement to
balance the traffic management so very carefully. I makes for a more
robust setup where the occasional busy day doesn't bring that
one-call-too-many that illustrates to your callers that you're using
voip over limited bandwidth.
On Tue, 16 Jan 2007 23:52:34 -0000, padraig at premierbb dot ie wrote:
>I have posted this question on several bulletin boards, but all suggestions failed to fix my
problem. Basically I have a dsl connection, then monowall, then the lan network consisting of 60+
machines. What I am trying to do is prioritise port 5060-5062 for voip traffic only. When the
network is congested, I want the voip quality to be as near as perfect as possible. could someone
please show me the "true" way of setting this up? BTW the dsl connection is 6 meg down, 512k up.
Michael Graves mgraves at pixelpower dot com
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. mgraves at mstvp dot com