Best QoS type for VoIP?

cross6

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Jun 16, 2005
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We have 3 toshiba IP systems linked by T1 and cisco routers. The phone people are saying we are getting echo's because we don't have QoS turned on.

But there are so many options, Fair Queueing, Weighted FQ, IP precedence - etc etc etc

Which type is generally best for VoIP?

We also use the T1 for data as well - (Terminal services clients on thin clients).

Thanks.
 

spidey07

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Aug 4, 2000
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class of service 5 or DSCP EF. You really shouldn't have to mark it (the IP phones should already be tagging the traffic), just setup the routers.

Voice is supposed to go into a low latency priority queue and have adequate bandwidth provisioned for it. As for data, class based waited fair queue is fine...later down the road you can start using it to give better service to certain data applications.

There are tons of configurations on cisco.com
 

dphantom

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Jan 14, 2005
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Separate VLANs for voice and daa. We use both 802.1p and 802.1q to tag and prioritize traffic. We currently set aside discrete amounts of bandwidth for IP voice traffic.

We use FQ for the last 4 years with no problems.
 

dphantom

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Originally posted by: spidey07
class of service 5 or DSCP EF. You really shouldn't have to mark it (the IP phones should already be tagging the traffic), just setup the routers.

Voice is supposed to go into a low latency priority queue and have adequate bandwidth provisioned for it. As for data, class based waited fair queue is fine...later down the road you can start using it to give better service to certain data applications.

There are tons of configurations on cisco.com

Good point spidey. We will be changing from FQ to class of service later next year. I am replacing 1750 class routers with 2821's later this year and next as part of that process among other needs for upgraded routers.
 

cross6

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Originally posted by: spidey07
class of service 5 or DSCP EF. You really shouldn't have to mark it (the IP phones should already be tagging the traffic), just setup the routers.

Voice is supposed to go into a low latency priority queue and have adequate bandwidth provisioned for it. As for data, class based waited fair queue is fine...later down the road you can start using it to give better service to certain data applications.

There are tons of configurations on cisco.com

Yeah this is what the phone people email'd me with:

We have our QoS settings in the phone system set to: Critical/ECP, Low Delay, High Throughput, and High Reliability
 

spidey07

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also, if you're running it on the LAN as well it is highly recommended you do QoS on the LAN switches as well.
 

randal

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please remember that you have to turn on qos on both sides of every WAN link!
 

cross6

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All the wan interfaces were set to FIFO,

would just moving to weighted fair queueing alone help things?

Or do I still need to go on with setting up priority queues?
 

spidey07

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Originally posted by: cross6
All the wan interfaces were set to FIFO,

would just moving to weighted fair queueing alone help things?

Or do I still need to go on with setting up priority queues?

priority queue is a requirement. There are very basic configs on cisco.com to do exactly what you need to do.

Don't forget to do the same on all the switches.

You simply cannot run voice without QoS end to end on every LAN switch, every LAN router, every WAN router.
 

cross6

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Originally posted by: spidey07
Originally posted by: cross6
All the wan interfaces were set to FIFO,

would just moving to weighted fair queueing alone help things?

Or do I still need to go on with setting up priority queues?

priority queue is a requirement. There are very basic configs on cisco.com to do exactly what you need to do.

Don't forget to do the same on all the switches.

You simply cannot run voice without QoS end to end on every LAN switch, every LAN router, every WAN router.

What should I be searching for?

I've been on and off cisco.com all day and all I see are rediculously complex qos setups or qos examples for ip over atm.
 

cross6

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Originally posted by: spidey07
http://www.cisco.com/en/US/customer/tec...logies_tech_note09186a0080094660.shtml

Start there. Not to harp or anything, but this kinda stuff should be known before doing VoIP.

One cannot simply slam in QoS without thinking about what is required and how to approach it.

I'm not much of a config guy, you could try forum.cisco.com.



Preaching to the choir my man! :)

I just got hired on - and this place's network is a CLUSTER FvCK.


PS thanks for the help

 

cmetz

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Nov 13, 2001
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cross6, do a ping from one end of the T1 to the other end of the T1. Does the latency jitter much or stay the same? Do you have packet loss?

Any chance you could get the data off of one of those T1 temporarily and check that VoIP only works fine?

QoS fundamentally is about telling devices how to deal with a resource scarcity. If that's not your problem, QoS won't help it. So the way to test that is to make the resource not scarce (e.g., an otherwise idle T1) and make sure *that* works.

Echos are typically not a QoS problem.
 

spidey07

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Originally posted by: cmetz
cross6, do a ping from one end of the T1 to the other end of the T1. Does the latency jitter much or stay the same? Do you have packet loss?

Any chance you could get the data off of one of those T1 temporarily and check that VoIP only works fine?

QoS fundamentally is about telling devices how to deal with a resource scarcity. If that's not your problem, QoS won't help it. So the way to test that is to make the resource not scarce (e.g., an otherwise idle T1) and make sure *that* works.

Echos are typically not a QoS problem.

Hmmm.....how best to approach this without offending one of our most knowlegible members.


It only takes a little bit.....a little bit of jitter/latency that cannot really be measured by a ping. It all comes down to queing/scheduling as far as controlling jitter/latency. With todays modern IP stacks it is common to have a window size of 32 or 64K. Going from a LAN interface to a (low speed, T1, etc) interface will no doubt fill up the trasmit ring/buffer with a simple web request.

That's the whole aspect behind LLQ and voice. It is utterly guarnteed, no matter what, that you will get served and maintain consitant/measurable jitter/latency.

I can't tell you how many LARGE enterprises I've consuslted on with the (and pretty good) logic of "I have plenty of bandwidth, there is no need for QoS".

You've got echo, meaning your QoS isn't setup right.

Again, much respect to Cmetz but I have this kind of debate with customers all the time. "we've got bandwith out of our ears, how in the world can QoS help?"

Just try my suggestions and see what happens, if it doesn't work then you can call me an idiot and I'll eat my hat and you won't be charged.
 

cross6

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Jun 16, 2005
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Originally posted by: cmetz
cross6, do a ping from one end of the T1 to the other end of the T1. Does the latency jitter much or stay the same? Do you have packet loss?

Any chance you could get the data off of one of those T1 temporarily and check that VoIP only works fine?

QoS fundamentally is about telling devices how to deal with a resource scarcity. If that's not your problem, QoS won't help it. So the way to test that is to make the resource not scarce (e.g., an otherwise idle T1) and make sure *that* works.

Echos are typically not a QoS problem.


Well, it's an elusive problem. It only happens randomly, the echo that is. And it's very far and few between.

It just creeped up, after a recent phone system upgrade - so I think the phone people are trying to shift some burden. Since they haven't changed anything to our network in about a year.

But QoS is a must for VoIP regardless of problems right?
 

cross6

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Jun 16, 2005
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Hmmm - after reading about the features we need, it appears not all of them are supported on our routers? (1700 series)

Cisco's white papers keep saying (3000 series and above) for things like DSCP and LLQ support?
 

dphantom

Diamond Member
Jan 14, 2005
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1750s do support that. It is what we use currently, but are in the process of moving to the 28XX platform.
 

dphantom

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Jan 14, 2005
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Originally posted by: cross6
Originally posted by: cmetz
cross6, do a ping from one end of the T1 to the other end of the T1. Does the latency jitter much or stay the same? Do you have packet loss?

Any chance you could get the data off of one of those T1 temporarily and check that VoIP only works fine?

QoS fundamentally is about telling devices how to deal with a resource scarcity. If that's not your problem, QoS won't help it. So the way to test that is to make the resource not scarce (e.g., an otherwise idle T1) and make sure *that* works.

Echos are typically not a QoS problem.


Well, it's an elusive problem. It only happens randomly, the echo that is. And it's very far and few between.

It just creeped up, after a recent phone system upgrade - so I think the phone people are trying to shift some burden. Since they haven't changed anything to our network in about a year.

But QoS is a must for VoIP regardless of problems right?

Yes, QoS is a must do for VoIP regardless of any other underlying problem.

Cisco Echo Link
 

cmetz

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Nov 13, 2001
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spidey07, if you have infinite bandwidth (and a non brain-dead technology), your latency is fixed, and QoS is not needed. QoS comes into play when you have a resource scarcity, even transiently. One packet is going down the line, the next packet - or packets, have to sit in a queue and wait their turn. This creates a variable delay - jitter - which creates many problems for VoIP. If you have a situation where this is happenning, then QoS techniques will help. A low-latency queue will help reduce the time a packet has to wait, at least letting it jump in front of the bulk data packets, and tends to keep your jitter down to the delay caused by a MTU-sized packet up ahead plus however many packets are waiting in the LLQ already. If you still have too much jitter, then more aggressive techniques like dividing into TDM channels might be a good idea.

But QoS is only a cure for a specific problem, and VoIP has *lots* of potential problems. Unfortunately, the VoIP arena brings together people who are new to networking with people who are new to voice, mixes them up, and creates lots of opportunities to be confused. Many people treat QoS as a sort of magic pixie dust for VoIP networking, configure it correctly (ah, but what is correctly? that itself is a long discussion ;) and everything will Just Work. Well, it doesn't work that way. QoS only solves a certain set of problems, those of queueing delay/jitter and drop priority. Your phone guys might not really know networking, and so to them, sure, QoS is the solution - they don't have to do anything, and it's mentioned in the documentation that you should set it up, so obviously that must be the problem, right?

Echo is a great example of a problem that can come from many many sources. In my experience, echo is not usually due to any of the problems QoS would solve. Echo is easist to introduce whenever there is an analog to digital (or back) conversion, due to the way analog phone systems work. Echo can also be created by mismatched echo cancellers, sometimes they help each other and sometimes they hurt each other. There are other causes, too.

The way I like to debug problems is to isolate into testable parts. In this case, the way to test whether QoS is the solution to this echo problem is straightforward (if not simple in the particular environment of the original poster) - get the T1 line to be otherwise idle, and make a call. If it works great when idle, and echos when there's data on it, then QoS is likely to be the solution to your problem. If it continues to have the echo problem when idle, then the problem is somewhere else, and QoS is *not* the solution to your problem.

QoS is still a good idea in a VoIP installation - a simple low-latency queue is a big help. But before you go off and reconfigure your network to do it, make sure the exercise is going to solve today's problem.
 

cmetz

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The Cisco document spidey07 posts is a pretty good one on where echo comes from and how to debug it. I'd definitely suggest cross6 give it a read and use it to help debug.

Qouth that document: "QoS can help in other ways (for example, packet loss and jitter), but it cannot, by itself, eliminate echo."
 

spidey07

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Originally posted by: cmetz
The Cisco document spidey07 posts is a pretty good one on where echo comes from and how to debug it. I'd definitely suggest cross6 give it a read and use it to help debug.

Qouth that document: "QoS can help in other ways (for example, packet loss and jitter), but it cannot, by itself, eliminate echo."

Completely agree.

QoS won't solve all problems, if you're out of bandwidth, you are out of bandwidth. But it will however ensure voice quality is achieving the best it can from a network no matter what congestion is encountered. congested or not.

Also DPphantoms link on echo is pretty dang good. I'm more of a data guy who has dabbled in TDM voice.
 

cross6

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Jun 16, 2005
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Yeah, I have a background in EE, so the whitepaper on why echo is usually leakage on the analog side makes sense to me.

But we had zero qos turned on, not even WFQ. So I'm definately looking into setting this up at least minimally.

Thanks for all the help and input guys.

And thanks for that cisco echo paper, I can show my boss since he probably won't believe me when I say the echo is most likely not our networks fault. (Our T1 is idle 99.9 percent of the time)