Setup QoS on Network
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
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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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"
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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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
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
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 !!!
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 !!!
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.
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.
You can't create a VLAN for voice while using softphones...
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.
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.
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?
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?
That's correct - I stand corrected - it is SIP and not QOS tagging.
There's big $$ in that for you down the road shepimport. My friend Kerry used to work for HB Fuller, his only function was QOS as they worked hand in hand with Cisco in the very early days of Cisco VOIP through 2004.
Now he consults for HB Fuller but has partnered into a company called 3Key Logic. He's doing much better now but he was in the QOS trenches for years and was able to spend quality time in places like Guatemala, Honduras, El Salvadore.
I need to keep remembering SIP - We are replacing our current Cisco 7750 with a new Shoretel. I am not really supportive of the venture but the Director of IT hates Cisco as a company so he's trashing the Cisco system. I should have SIP on the brain by now.
There's big $$ 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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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
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
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
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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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...
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...
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.
If I get a chance I will post some info re: the softphone QOS.
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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