 carpRejected join:2002-10-30 Reviews:
·RoadRunner Cable
| Priority Queue - Threshold other traffic denied service. Been trying to find some recommended thresholds for priority queues as percent of total bandwidth. The threshold is the point where other traffic is now at risk to be severely slowed or flat out denied service.
So, what's your opinion?
I have a situation where someone wants to prioritize over 43% of a DS3 for IP phone traffic. This about 3 times the amount where I get a little nervous. 3845 and 7206VXR handle the interface.
There is other interactive traffic over the link. It's a call center so slowing down a call center workstation while they are on the phone with a customer cannot happen. |
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 Bink join:2006-05-14 Denver, CO kudos:4 | Depends on the number of calls, bandwidth used per call and latency thats required... While no specifics have been provided, this request might not be out of line if a couple hundred call are active at one time
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 carpRejected join:2002-10-30 Reviews:
·RoadRunner Cable
| reply to carp 20Mbps is being allocated to the queue. 90.1kbps per call. Peaks at 168 calls, ~221 max allowed in the 20Mbps.
Latency, well you know how upper management can be. Essentially not allowed for interactive traffic. They can't wait seconds for screen pops with a customer on the phone. |
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 nosx join:2004-12-27 00000 kudos:5 | QOS profiles are dependant on the traffic utilizing the circuit. I have tables full of real profiles and how they chop things up.
I have also seen several providers pull a stunt: "priority percent 90" for EF traffic. This works well because the priority LLQ mechanism is IOS is congestion aware. It only flips on if the interface is "experiencing congestion".
Why? Voice (RTP) is important, and everything else will retransmit. If a circuit is congested a good percent of the time, it needs to be upgraded. For the instances where traffic bursts and problems crop up, the VOIP will be protected. |
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 | reply to carp How big is the pipe in question?
Would you be able to supply your current QOS configs, if not the entire config, for us to review?
Second all that has been said so far. Do you have any sort of monitoring in place that shows you how much bandwidth is being used by what apps? If you don't have any sort of measurements in place, how are you going to tell what's going on?
Regards |
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 carpRejected join:2002-10-30 Reviews:
·RoadRunner Cable
| reply to carp Not looking for help working out a policy. Asking if anyone has seen documentation or has an opinion/experience of when PQs in CBWFQing become troublesome or DOS other classes when the PQ has too high of a bandwidth reservation. I get the impression it's merely create policy, monitor, tweak. Repeat until time to add capacity.
Yes, we know all the applications on the link, what they use, current PQ/CBWFQqueue , etc. There's a "policy/politics" part of the problem since it's rare to get buy in to a traffic policy that makes sense. Can't even be restrictive on email traffic or remote file transfers in most cases.
The answer usually given is "then you need to increase bandwidth if you want everything on a level playing field". It's actually the smart answer in this situation anyways. It's a bigger location with a call center and no redundancy to begin with. It would be a rather busy DS3 when near peak calls. Peak calls + Peak Data will put the link over 1o0% utilization and for some good chunks of time.
Thanks for the thoughts. |
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 CovenantPremium,MVM join:2003-07-01 England | "Only a guru or a fool would break the 33% rule!"
That is the general consensus when provisioning voice bandwidth against Cisco's design recommendations as stated in the SRND.
It all depends on how one perceives oneself. 
I am assuming you are talking about low latency queueing here and not another priority mechanism such as "ip rtp priority" but the consensus is the same either way.
You can, of course, deploy any amount of bandwidth you want for LLQ as have many others in the past and so will in the future however you should only breach the Cisco design guidelines if the circuit is relatively low bw (yours isn't) and if you have performed traffic analysis and checked out the jitter, delay and loss for any non-VoIP critical traffic (IP SLA is good for that) with both the 33% profile and the 43% profile. This will also include checking the router's stats.
Don't forget that any call signalling protocols which may also traverse the link will also suffer giving rise to call setup, call tear down, and any mid-call features delay and registration issues if the phones need to register to their CUCM across the WAN.
Here is some bed time reading which is the CUCM 8.x SRND however the rule is true across all previous versions of CUCM:
»www.cisco.com/en/US/docs/voice_i···p1165070 -- A word to the wise ain't necessary, it's the stupid ones who need the advice! |
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 nosx join:2004-12-27 00000 kudos:5 | Would you provision 33% EF priority queue for a call center with hundreds of agents and an expectation of 120+mbps of VOIP on a 155mbps OC3?
The correct policy is entirely dependant on the traffic and calculated required bandwidth protection for realtime traffic coming from a given location.
So what if EF starves out other queues? If that is legitimate voip, would you rather be dropping phone calls or slowing down youtube videos? |
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 CovenantPremium,MVM join:2003-07-01 England | I guess that is directed at me... That is a very general question with no real detail to be honest and thus I will make some assumptions. I would use a lower bandwidth codec (iLBC is a very good example) AND not put all my eggs in one basket in the context of having just a single cct from a single carrier to carry all my voice calls but have geographically diverse ccts from different carriers.
In your example, it looks like this is a pure voice cct with data as a "distraction" and in my opinion there is no point in having data across on that. You might as well as provision a 20 Mbps circuit with no QoS just for data and leave the OC3 for the voice. It could also in times of outages, be a backup for critical calls outbound.
You mentioned YouTube... I guess you are being cheeky as I am sure the environment you are thinking of does not just have people talking on the phone endlessly with no aim or way to log their calls, and if they are not, they are on YouTube. Depending on the topology, there are CRM applications, CTI control of the agents' desktops, intranet, third party partner websites, etc, which if starved, can also lead to issues.
So in your example, if you can't reach the 33% and are closer to 80% voice traffic, then you can either use a low bw codec + compression to get it below that (assuming you are using G.711u/a) or just move the data to another circuit.
We found that separation at levels above 40% on circuits bigger than your OC3 was the optimum level HOWEVER any request for over 33% needs a traffic profile built up first with 6 monthly checks as to the jitter, delay and packet loss for the 6 class QoS model we have.
Hope that helps. -- A word to the wise ain't necessary, it's the stupid ones who need the advice! |
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 nosx join:2004-12-27 00000 kudos:5 4 edits | While I was being overly cynical with the youtube example (it was late and i hadnt been fed yet) , generally speaking the "other" traffic on any circuit (including agent applications like the crm & cti examples you provided) are all TCP based, and while far from optimal can continue functioning at lower bandwidth due to congestion management techniques such as WRED kicking in.
We run our VOIP at G.729 with 30ms sampling for a good mix of compression without killing our MOS. While a contact center is one example, there are plenty of VOIP heavy enterprises and telcos out there. I have some good examples ranging from call centers, to conferencing bridges, to sales offices. Each requires a different policy based on both circuit speed and intended utilization.
That 33% "rule" does not coincide with either industry best practices or other Cisco advice. In fact, the "using no more than 75% of the total available bandwidth" does not appear to coincide with the service most providers are provisioning today. Taking Verizon for example, their PE-P links were all 50% ef, 50% split of other queues. The most common policy for VOIP on PE-CE circuits was either 50% ef or 90% ef, with the remaining bandwidth split among the other queues in one of several different ways. You will find similar configuration (with varying degrees of flexibility) from both AT&T and Sprint.
==Interesting Edit== So i tracked down the source of the 33% rule, and here is the beef: It was crafted by Cisco Technical Marketing back in 2005, and it has to do with STRICT priority queueing. With LLQ as configured in my posts the queueing is not strict and the policer is congestion aware. This seems to be a carry over from the olden days. ==End of edit===
I attached a snippit of a QOS policy table used for service policy templates approved for provisioning by the architecture gurus.
Below is one of the provisioning system templates that uses policies depending on how the service is ordered. The biggest change to policy based on circuit size is the number of packets in each WRED queue. This is the queue depth calculated for circuits ranging from 1meg to 120meg. The percentages work out fairly well from 1meg to 10gig.
$SPEED >=1 <= 120
policy-map DSCP_1M_TO_120M_$PERCENTVOIPEF_$PROFILE_6Q
class DSCP_EF_CS5
priority percent $PERCENTVOIP
class DSCP_AF4X
bandwidth remaining percent $Q1PERCENT
queue-limit 1024 packets
random-detect dscp-based
random-detect dscp cs4 171 500 1
random-detect dscp af41 171 500 1
random-detect dscp af42 52 171 1
random-detect dscp af43 52 171 1
class DSCP_AF3X_CS6_CS7
bandwidth remaining percent $Q2PERCENT
queue-limit 2048 packets
random-detect dscp-based
random-detect dscp cs3 250 500 1
random-detect dscp af31 250 500 1
random-detect dscp af32 52 250 1
random-detect dscp af33 52 250 1
random-detect dscp cs6 511 512 10
random-detect dscp cs7 511 512 10
class DSCP_AF2X
bandwidth remaining percent $Q3PERCENT
queue-limit 1024 packets
random-detect dscp-based
random-detect dscp cs2 171 500 1
random-detect dscp af21 171 500 1
random-detect dscp af22 52 171 1
random-detect dscp af23 52 171 1
class DSCP_AF1X
bandwidth remaining percent $Q4PERCENT
queue-limit 1024 packets
random-detect dscp-based
random-detect dscp cs1 171 500 1
random-detect dscp af11 171 500 1
random-detect dscp af12 52 171 1
random-detect dscp af13 52 171 1
class IP_BEST_EFFORT_BE
bandwidth remaining percent $Q5PERCENT
queue-limit 512 packets
random-detect dscp-based
random-detect dscp 0 171 500 1
Note how also while they are not in their own class, routing and control traffic (CS6 & CS7) are still protected by the specific WRED config of this example.
Also, on the new SP gear Cisco is changing the way queueing is calculated. Today its either in bytes or packets depending on the platform, but the goodies down in the lab only queues based on milleseconds. That helps people understand the jitter introduced by buffers knowing that a maximum of 30ms traffic will be buffered. The calculation internally ensures that the time based buffering will be as accurate for a 100meg circuit as it is for a 10gig circuit. That way we wont have to keep adjusting WRED queue depth based on circuit size. |
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 | reply to carp said by carp:Asking if anyone has seen documentation or has an opinion/experience of when PQs in CBWFQing become troublesome or DOS other classes when the PQ has too high of a bandwidth reservation. I get the impression it's merely create policy, monitor, tweak. Repeat until time to add capacity. Don't think I've seen documentation before, like you said, design, create, monitor, tweak. The obvious thing is to watch your policy output counters for drops, but the flip side of that is that if there are no drops then it's a real pain in the arse to figure out what's going on -- I've had to deal with this kind of situation once or twice before.
said by carp:It's a bigger location with a call center and no redundancy to begin with. It would be a rather busy DS3 when near peak calls. Peak calls + Peak Data will put the link over 1o0% utilization and for some good chunks of time. 1) no redundancy == BAD for a major location
2) if you got peak times near 24 x 7 then that's a business case to upgrade right there; you've taken the engineering as far as you can for now, I think.
@Covenant / nosx Thanks for the 'light' reading, not really my field, but makes for afterhours reading 
Regards |
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 CovenantPremium,MVM join:2003-07-01 England | reply to nosx said by nosx:==Interesting Edit== So i tracked down the source of the 33% rule, and here is the beef: It was crafted by Cisco Technical Marketing back in 2005, and it has to do with STRICT priority queueing. With LLQ as configured in my posts the queueing is not strict and the policer is congestion aware. This seems to be a carry over from the olden days. ==End of edit=== I find that hard to believe seeing that LLQ was around BEFORE 2005 so excuse me if I don't take your word for it.
In addition, it mentions LLQ explicitly and not strict priority queuing.
It isn't carry over as Cisco updates its recommendations. However on our P-P links, we don't set a limit we just priority queue the voice on our MPLS core on the Ciscos.
Quoting VZB QoS when you are a customer of theirs does not mean you are party to how their QoS is designed or implemented but thanks for showing me what the competition have done for you.
said by nosx:That 33% "rule" does not coincide with either industry best practices or other Cisco advice. OMG! For our P to CE links, we can even do 95% as our clients are getting a slice of the GigE or SDH, thus 95% of their bandwidth is in fact, only 2% of the bearer interface. The other way round, we recommend customers to only set 33% as that is the Cisco recommendation for Enterprises. If you want to see QoS for SPs, then I suggest you have a look at the SP SRND. That is Cisco's advice, just disregarding it because you haven't done it does not mean it is not true.
You seem to have taken this personally however I can't help that and the OP wanted some reference guidelines and us who do VoIP full time and not as an adjunct to data know that when designing voice networks within SP and enterprise, the Cisco CUCM SRND is the bible. That may be unusual for people in the data world however that is how networks are designed in the voice world.
So the recommendation still stands as it is a Cisco recommendation and not a personal one. Show me a design guideline about voice and a different percentage allocation and I will be happy to reconsider. In the mean time, propagate good design guidelines and maybe have a beer. 
Ps. It is sooo good to be back... forgot how people are on here! -- A word to the wise ain't necessary, it's the stupid ones who need the advice! |
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 nosx join:2004-12-27 00000 kudos:5 1 edit | I was not speaking as a customer of a provider, i was speaking as an engineer from the provider that implemented the services. I have the engineering documentation behind alot of that research and deployment as well. Cisco (AS & CE) have been engaged periodically and on the last call they refered to the architecture as "very strong".
The 33% number doesnt hold true unless you are stuck in 2005 on a T1. The right answer is to construct a service policy based on the actual traffic that will use the circuit and must be protected.
If you want an SRND example, see www.cisco.com/warp/public/cc/so/neso/vpn/vpnsp/spqsd_wp.pdf where Cisco explicitly exceeds 33% priority queueing for realtime traffic. |
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 CovenantPremium,MVM join:2003-07-01 England 1 edit | Well thank you for that insider knowledge. I was under the impression that VZB like numerous other SPs actually have an in house algorithm which is based on Cisco (after a NDA) providing the inner workings of its QoS algorithm as the WRED thresholds can be complicated based on a 6 or more QoS model but it's nice to know they still do things the old fashioned way.
That SRND is dated 2002 AND is not relevant to the OP. Why... because we are talking about the 33% rule in context of CE to PE, not PE to CE. You are still acting like a Data engineer and not a VoIP engineer. The methodologies and way of doing things are different. Try spending time in a VoIP specific forum and you will see there is more to VoIP than QoS or just trying things out.
Actually, I realised I am talking to a Data engineer and not a VoIP engineer so you look at it from a SP point of view, I will look at it from a VoIP point of view, and let it be.
The SRND posted for VoIP is the one and only reference for Cisco VoIP in an enterprise environment.
So let's not complicate matters for the OP by throwing in SRNDs which are dated and are not relevant to them as the OP is asking from an enterprise background and not SP. Let me re-iterate that again, there is a difference in QoS between SP and Enterprise and if that can't be appreciated, well, I know for sure that I will never be out of a job and I look forward to many years of consultancy work when I get out of this game.
In addition, if this old hat, why is it in the Voice CCIE? I rest my case now.
Think the OP has long gone cos we scared him off... LOL!
-- Edit: Too blunt for my own good! |
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 aryobaPremium,MVM join:2002-08-22 kudos:1 | reply to Covenant
Re: Priority Queue - Threshold other traffic denied service. said by Covenant:there is a difference in QoS between SP and Enterprise
In addition, if this old hat, why is it in the Voice CCIE? I rest my case now. It is not the first time Voice and Data guys don't agree 
It is not the first time either that SP and Enterprise requirements conflict. This is why in some organizations, there are multiple MPLS networks within the same organization which each network is custom tailored into specific needs.
said by Covenant:Think the OP has long gone cos we scared him off... LOL!
Hopefully nobody is scared off since this is one of a rarely-discussed topic in this forum. Covenant , if you can invite some of the Voice guys in VoIP forum to engage this discussion; it would be appreciated  |
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 carpRejected join:2002-10-30 Reviews:
·RoadRunner Cable
| reply to carp I'm still here.
I reset the counters of the current policy last Thursday and have dropped ~110000 packets in the policy since then with only 13% of the bandwidth for LLQ, which isn't ustilized much since the voice is currently on separate T1's from the data Most of the drops come from the class with the biggest chunk of BW and a little assured forwarding (only AF41). |
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