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VoIP QoS - need suggestions

I would kill for some decent QoS right now. Currently, I'm using an Asterisk server behind a Cisco PIX 515e running 6.3(4). The PIX is connected to a Cisco 2620, which is my T1 router. My VoIP provider is the same as the T1 provider, so when network traffic is normal, I have

11ms ping time to my provider. However, when some idiot is downloading videos from CNN.com or something, my VoIP calls suffer greatly. I have 802.1Q working great on the local LAN, with my netgear switch. That's not my problem - the problem is obviously the T1 connection.

I'm looking for a way to throttle back all traffic besides the VoIP, which uses the SIP protocol. I'll be upgrading the PIX to version 7 next week I believe. From what I understand, version 7 may help with my problem - but my concern is that I'll need to upgrade my 2620, which is running 12.1. I'd like to upgrade the 2620, so I had the flash and ram maxed out - but I was told the 2620 can only go up to 12.3.

So, I'm looking for good solid suggestions on how to prioritize my SIP traffic using these devices - and if need be, I'm willing to swap out the 2620 with something more modern that will do a better job. Are there some commands I can use with IOS 12.3 and PIX OS 7 that will solve my problems?

Reply to
PSX
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12.1 should be happy with QoS (although there are lots of features).

Try just prioritising the SIP traffic - or you can pick out RTP. It might be easier to just prioritise everything that goes to the Asterisk server.

some stuff from TAC to get you started:

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The place to concentrate on is the WAN link.

However - only the router at the far end of the link can prioritise stuff sent to you, since it has to manipulate the Q for packets it sends to the serial link.

If this is a typical Internet feed, then most of your traffic is inbound, so that is where the problem is - and there is little set up on your router that is going to fix everything.

So - talk to your ISP about whether they will prioritise the voip.

1 further thought - if you are using fairly cheap broadband links the easiest solution may be another ISP link and make sure it only gets used for voip. After all, no contention means no Q build up, so no congestion and no need for QoS..... >
Reply to
stephen

Well, my main problem is with the outgoing part of the call. We routinely get complaints about the other end not being able to hear us and having us "break up". We can normally hear them fine. My ISP claims they are prioritizing the traffic coming to us, now I need to do it going out. I want to avoid the expense of having a seperate T1 just for voice - we host a web server, mail server, and dns server on this network as well - there just has to be a way to give voice the number one slot and have everything else throttle back.

Reply to
PSX

if it is your end of the link then it sounds like it is your lack of priority for voice (it could be the other site of course).

i would try to find a test that reliably breaks the voice 1st, then you will be able to tell when things get better...

try using LLQ - see:

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a lot of the defaults are probably sensible for you.

not sure if your traffic will be DSCP marked since it is coming thru a firewall - you might need a filter / ACL to identify it as priority traffic. The earlier example i pointed you at used IP addresses of voice gateway.

Reply to
stephen

are you using smoothstone? if you are, only a new provider will fix that.

stephen wrote:

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Reply to
justin_ltg

Try this:

policy-map shaper-pmap class class-default shape average 512000 2048

policy-map voice class voice priority percent 40 service-policy shaper-pmap

Attach this to your ingress and egress interfaces substituting the appropriate bandwidth values.

Reply to
delgrundy

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