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Setup QoS on Network

Posted on 2006-11-09
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Hi experts !

I have a customer who has a network with VoIP. He has tremendous jitter in the phones when they are busy. I think that perhaps I can alleviate the problem by structuring Diffserv (QoS0 on the Netgear switches. But I don't know which DSCP values should be assigned to the "highest" level.  Any thoughts?

Thanks,  Pat
PS - I'm on site now
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Question by:PFSullivan
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harbor235 earned 100 total points
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The question is? is all of the VOIP traffic local so his QOS policies will have an effect?

Instituting QOS is fine but if you do not have management control of the entire end to end
data flow then you are wasting your time. However, if you are using a provider that will
accept your traffic with QOS tagging then it is doable.

Please provide additional info on your provider, if any, do they accept your QOS? etc ......

harbor235 ;}
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by:PFSullivan
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Thanks for the reply

Our ISP is Sprint and the Phone system is Avaya - both asssure me that QoS tagging is accepted.  Since I last wrote, it seems that DSCP = 46 is important to be set to highest.  Is this right ??

Pat
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by:EnclosAdmin
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harbor235
You're not really wasting your time - but you’re correct in that you’re not in complete control over a DSL type ISP.  You should always implement QOS regardless of the type of circuit connection that you have if you're going to attempt to use VOIP.

With properly configured QOS you’re still giving yourself the best possible chance of good sound quality.  Your routers on each end still prioritize the traffic between themselves and DSL/Cable Services are becoming more reliable every year.  While in the wild that traffic is just data but when it reaches your premise it will be prioritized giving you the best possible change of good voice quality.

I have clients that have hundreds of home office users domestically and internationally that use VOIP on Cisco, Avaya, Shoretel etc.  With no QOS settings the calls are choppy, echo and dropped.  With QOS enabled and attempted they are vastly superior.  No they are not perfect but in most circumstances the calls work like a standard telephone call.

Now I have some questions for PFSullivan:
This is a Sprint internet circuit?
Is it “Private IP” Sprints term for internet controlled by them where your calls never leave the Sprint network?
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by:PFSullivan
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Hi All - Let me try and answer these question. Please understand that my background is data networks and the VOIP issue is foreign to me.

The companies that are involved are:
Avaya for the Internal phone systems
Sprint is the DS3 supplier
Voicenet is our ISP (for the data T1 which is separate) - Voicenet also manages our Routers (Cisco 2500) and Firewalls (Netscreen) - But I believe that they never touch the VOIP side so I think they are unimportant.

I believe that the Sprint DS3 handles all Phone traffic . I don't know if that fits your description of a 'Private IP"

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by:shepimport
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How many phones and computers are on the network?

did you check if the switch is in full duplex mode... thats more likely the cause of jitter on a 100 mb network with less than 50 computers...

Half duplex transmitions cause jitter...

also... the diff serve value does not matter... pick a number any number... your standard packets are "untagged" therefore a value of 0... any value above zero will give priority...

you isp does not matter... you are not using IP trunks, you are on a legacy connection...

sprint does strip ToS tagging... but the traffic never hits the network.  All tier 1 carries agreed to strip ToS tagging...
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by:PFSullivan
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Hi Shepimport - Thanks

There are 51 computers using "soft phones" and 11 hard handsets on the net.

I have forced all switches to 100/Full
I have forced all NICs to 100/Full

If I load QoS on each station and configure the NICs for "QoS enable", this won't help??  I'm wasting my time as Harbor325 said?

regards,  Pat
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by:shepimport
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No, first off you are not wasting your time... QoS is a learned skill... not hard just takes some experience... you basically are looking for a congestion point...

First off... are all the users on site? on the LAN?

Softphones can not have ToS values set... due to windows not having a packet classification application

the IP phone can probally...

what is the network structure?  any bottle neck points?

a DS3 for 60 users?  partial?  thats crazy !!!
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by:EnclosAdmin
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Okay from what I understand you are using a T1 provided circuit.  So the ISP does have control over the path.  Your QOS should work just fine as long as it doesn't traverse the internet.

In the Cisco world we create a VLAN, then place voice traffic into that VLAN.  We then prioritize traffic withing that VLAN [Your voice traffic] on the Switches [[Routers can do this too but it creates more overhead on the router.]]

That way when the switches see voice traffic they properly tag those packets as Priority #  and they also immediatly reserver the requested bandwidth - forcing all other data traffic to the lower bandwith table.

For instance, you want 12k of bandwidth per call and you have a 256k circuit.  The minute a person picks up the phone that switch has just reserved 12k of the bandwith.  As long as that person is on the phone that 12k is reserved. The next guy picks up a phone - another 12k is reserved.

All other netowrk traffic competes for the remaining bandwidth.

Important question:
What is your CIR [Commited Information Rate]  What does the ISP guarantee you for bandwidth?

Remember that most companies save money by purchasing 1/2 of their port speed for CIR.  So if they buy a 512k port they generally guarantee a 256k CIR.  They can always use the 512k but the ISP only has to guarantee 256k.  That's becoming much less of a problem these days as ISPs are generally adding bandwidth quarterly.

As long as you know that your not anywhere near over taxing your CIR and it sounds like you are not then you should be able to place your voice packets in a VLAN and prioritize those packets throughout.  

I wish I had a Netgear configuration I could show you for creating that but we don't have any more of them.

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by:shepimport
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You can't create a VLAN for voice while using softphones...
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by:EnclosAdmin
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Correct shepimport but the softphone being a VOIP client has to talk to the Avaya server in this case and it creates its own QOS setup between the client and the server after the clent connection is requested.  Technically a softphone is attempting to use native QOS [depending on the system - Avaya, Cisco etc] even through the internet if the client happens to be remote as in a hotel room etc.  The packets are tagged on both sides and at the mercy of the internet as a transport however they are correctly prioritized when sent/received.  Not 100% effective where the internet is involved but very effective when the circuit is private.

In our experience the internet is getting better but we do have more problems with jitter, drops and lost calls without a doubt as you'd expect.
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by:shepimport
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Not to be a jerk ... but, thats wrong... I will feel free to look at any packet capture from a softphone and there is no ToS tagging... The only "qos" that takes place is error recovery in codec's like GIPS Global IP Sound... like Skype uses.. but, not a standard soft client.

Cisco/ Avaya offer no support for softphones beyond a standard SIP User Agent/ Regestration ... Media streams are than decided only on codec by the SDP section of the SIP message.. so either g.711 or g.729/a if you are liscensed for it.

I work for a QoS division of a harware developer... this is what we do all day... sounds exciting ehh?
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by:EnclosAdmin
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That's correct - I stand corrected - it is SIP  and not QOS tagging.

There's big $$ in that for you down the road shepimport.  My friend Kerry used to work for HB Fuller, his only function was QOS as they worked hand in hand with Cisco in the very early days of Cisco VOIP through 2004.

Now he consults for HB Fuller but has partnered into a company called 3Key Logic.  He's doing much better now but he was in the QOS trenches for years and was able to spend quality time in places like Guatemala, Honduras, El Salvadore.

I need to keep remembering SIP - We are replacing our current Cisco 7750 with a new Shoretel.  I am not really supportive of the venture but the Director of IT hates Cisco as a company so he's trashing the Cisco system.  I should have SIP on the brain by now.

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by:Reid Palmeira
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"The companies that are involved are:
Avaya for the Internal phone systems
Sprint is the DS3 supplier
Voicenet is our ISP (for the data T1 which is separate) - Voicenet also manages our Routers (Cisco 2500) and Firewalls (Netscreen) - But I believe that they never touch the VOIP side so I think they are unimportant."

so first, I'm hoping that's a partial DS3 if you have such a small user base. There are a couple of questions there.
1. is this an Internet data DS3? or a plain TDM DS3 going to the PSTN?
2. Related to that, what's the VoIP setup like? do you have TDM trunks going into an Avaya system somewhere that does the VoIP for the clients? If so then the VoIP part ends at the Avaya equipment. so you only need QoS marking there. And because it's a LAN there should be sufficient bandwidth. You want to make sure the marking is correct so that you get the end to end QoS but there should be sufficient bandwidth on the LAN so that you don't get much jitter from that.
3. You'll probably want to differentiate between the SIP signalling and the RTP. In particular, your jitter is going to come from contention with the RTP packets. So DSCP 46 and 26 for the two respectively. QoS 5 and 3, depending on what the circuit provider uses. Keep in mind this QoS needs to be end to end from the point that the PSTN meets your VoIP provider all the way to your endpoints, whether softphone or handset.
4. you don't want to have the NIC's on the computers with the softphones mark anything. that should be done by the SIP registrar itself and/or the network switch based upon the traffic as it goes through. That way you only prioritze the voice.
5. one other thing to check. And you may not notice this on the normal data side because it's a little more tolerant, but you may want to have someone check all your cabling, as bad cable can lead to all sorts of issues that are hard to track down. Poorly terminated cables may drop a bunch of packeets and while data can tolerate that with retransmissions, your voice will sound horrible because of SIP or RTP drops. It's all UDP traffic not TCP remember. Same with wireless if any of the computers are wireless. Interference will wreak havoc on your RTP streams causing all sorts of issues.

Cheers,
-rp
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by:PFSullivan
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Hi RP -   Thanks for chiming in -

I don't know the answers to all of your questions but I do know a few:

The DS3 only has 8 T1s turned on (rest are for growth)

The balance of this info I will locate.  In the mean time my network is structured like this:

Each group of 15 users (soft phones) go to a Netgear managed switch
All 3 switches use the GB uplink port to one of two HP Procurve switches
The Procurve switches feed server, routers, oubound firewall, and Avaya system
There is no wireless in the building
All 11 IP phones go direct to the Procurve

I will be back on this job site Wed next week but in the mean time I have access to the data side of the system if you need info.  I need to look up a few of your pneumonics.

Thanks again,

regards,  Pat

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by:Reid Palmeira
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Hi Pat,
so those 8 T1's are all traditional TDM telephony T1's going back to  your ILEC or CLEC to the PSTN? or are those IP data T1's going to the internet? I'm guessing those are traditional T1's going to your carrier. The trick here is to deliniate between the VoIP part of your network and the TDM part. if the WAN connections are all old school TDM the jitter isn't coming from there. It's coming from the LAN side. If they're internet T1's going to a VoIP provider somewhere, that's probably where your jitter comes in from.

Cheers,
-rp
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by:jfrady
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Just to chime in - some softphones indeed apply VLAN tagging.  3Com PcXset does.  Some NIC's can also apply tags if appropriately configured.

EnlosAdmin - Is the only reason the 7750 is being trashed is thboss hates Cisco - Did you have issues?  I don't really care for Cisco either but $ is $$.  Picking ShoreTel is what I wouldn't be so sure about.  

Reading through the discussion here it seems both LAN and WAN ideas are being floated.  But no info for sure where the problems lie.  Do the issues happen on LAN calls, WAN calls, or both?  Do you have egress to the PSTN or are you using SIP trunking to get to someone providing dial tone?
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by:shepimport
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Hola -

I talked to some of my voice developers and they have mixed opinions on whether a softphone can support ToS or CoS tagging... and after some research I could not find any setup on it... Everyone but one person said no, it will not work due to NIC's on a windows PC ( except a few) do not support ToS on an application level... some can support it but, it must be for all data vs. per application... which makes it obsolite?  The one who did think it would work said he thinks SJ phone does it... which he wrote some of the SIp stack for it... but, he could not find the setting... but... he is getting back to me next week...

BTW... he has a PSTN connection using standard TDM... so its a LAN issue...

Some thought went into why a small network may be having traffic problems... one your broadcast traffic is causing trouble so.... check for a loop in your lan... other thing... a PC may have a virus and be sending out a flood of broadcast traffic... in which... i would check port statistics...
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by:jfrady
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Thanks for the additional info.  Since it is a LAN issue I would recommend getting Ethereal or any other packet capture program and performing some captures to see utilization, broadcast/multicast level, etc.  

If I get a chance I will post some info re: the softphone QOS.

 
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