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Cisco 9300 queues and output drops

FuadG
Level 1
Level 1

Hello everyone,

I am trying to troubleshoot VOIP related problems when voice during calls is intermittent

The phones are connected to separate switches and there is Cisco 9300 in between those switches.

I have read through this article:

https://www.cisco.com/c/en/us/support/docs/switches/catalyst-9300-switch/216236-troubleshoot-output-drops-on-catalyst-90.html#anc13

And increased qos queue-softmax-multiplier 1200 

Now drops are not increasing anymore in this table, but TH0/TH1/TH2 are still increasing

AQM Global counters
GlobalHardLimit: 5176 | GlobalHardBufCount: 0
GlobalSoftLimit: 14672 | GlobalSoftBufCount: 0

----------------------------------------------------------------------------------------------
High Watermark Soft Buffers: Port Monitor Disabled
----------------------------------------------------------------------------------------------
Asic:0 Core:1 DATA Port:15 Hardware Enqueue Counters
-------------------------------------------------------------------------
Q Buffers Enqueue-TH0 Enqueue-TH1 Enqueue-TH2
(Count) (Bytes) (Bytes) (Bytes)
-- ------- -------------------- -------------------- --------------------
0 0 15626884274 32907700571 2516556011
1 0 0 0 7226027781
2 0 0 0 0
3 0 0 0 0
4 0 0 0 0
5 0 0 0 0
6 0 0 0 0
7 0 0 0 0
Asic:0 Core:1 DATA Port:15 Hardware Drop Counters
-----------------------------------------------------------------------------------------------------------
Q Drop-TH0 Drop-TH1 Drop-TH2 SBufDrop QebDrop
(Bytes) (Bytes) (Bytes) (Bytes) (Bytes)
-- -------------------- -------------------- -------------------- -------------------- --------------------
0 14586413831 614569504 0 0 0
1 0 0 15293510335 0 0
2 0 0 0 0 0
3 0 0 0 0 0
4 0 0 0 0 0
5 0 0 0 0 0
6 0 0 0 0 0
7 0 0 0 0 0

Can this queuing affect voice traffic? 

41 Replies 41

Hello,

All uplinks are Gig, there are Cisco 78xx series (which has 100mbs port I believe) that connects to Gig port on the switch. 

Show int reflects Full Duplex 100Mbs

I dont think Cisco phone and the switch are not capable of auto neg, considering the fact that it was working fine for several years

 

You have bottle neck

Huge data pass through 100 M 

Try use 1 G and see 

MHM

Did a bit more of investigation, seems like only 7811/21 phones with 100mbs are affected.

I examined Total Output Drops on Switches A and B and noticed that these Drops only occur on interfaces connected to these models. All other IP phone models (like 7931 for example, which also have 100mbs) has 0 Output Drops. PC on 100mbs also has 0 Output drops

Are you aware of any bug related to these models and firmware sip78xx.11-5-1SR1-1

Of other not effect check duplex also.

This issue not QoS either duplex/speed mismatch or bug I think

MHM

There are no specific configuration for duplex/speed, all auto neg.

Can these Total Output drops be the reason of the voice problems? Am I digging in right direction?

You meaning voice quality?

If yes then sure Output drop effect voice quality.

How many phone you have for same model?

MHM

We have plenty of them, not all of them are complaining though. 

However, those who have voice quality problems are located in different buildings, so there is no common switch or any other common factor - this is what gives me doubts about a bug, but I can not come up with any other reason for now

 


@MHM Cisco World wrote:

You have bottle neck

Huge data pass through 100 M 

Try use 1 G and see 

MHM


Yup, a FE will often bottleneck before a gig interface, but moving from FE to gig doesn't guarantee VoIP quality.

Generally, you need a correct QoS implementation to meet service guarantees, regardless of interface bandwidth assuming the interface can be oversubscribed.  If it cannot be oversubscribed, congestion cannot form.

Regarding the QoS

Do I configure auto qos voip cisco-phone on the port connected to Cisco Phone and auto qos voip trust on uplinks?

As far as I understand, I need to configure it on all switches along the path, am I correct?

And the last question - how can I make sure that Cisco Phone is marking its packets a correct DSCP value? Web interface of the phone is not showing this info

Hi @FuadG - you require QoS configured end to end so where the IP Endpoint / Phone connects to the switch enable auto qos voip cisco-phone on the switchport.  Then on your uplinks enable auto qos voip trust.  I believe on newer models switches QoS is enabled by default but back in the day you had to enable QoS by entering mls qos from configure mode.  You'll also need to enable QoS where your CUCM on-prem servers are on the network and also where your voice gateway / CUBE connects to the network.

BTW, I don't recommend (current) AutoQoS because it provides a QoS that imposes a complex QoS model which may good, neutral or bad for YOUR usage case.

Some (not so much Cisco, itself) might argue with Cisco behind the model it's ideal.  Well, I think we're now up to version 4 of what the model is, and it's "easy" to modify, both of which sort of negates it's not necessarily ideal.

In practice, it usually does well for real-time traffic, like VoIP, and usually isn't adverse to other traffic.

Although AutoQoS is easy to modify, how it should be modified isn't so easy, IMO.

I believe the QoS model being used should be one specific for your goals.

Oh, the later gen Cisco switches (starting with 3650/3850?) actually implement a simple QoS model by default.


@FuadG wrote:

When I use this formula:

Percentage of output drops = Total output drops / Total output bytes X 100 I get an extremely low value.

Where you get that formula?

First, you need determine whether you want to count drops in bytes or packets.

Second, using either bytes or packets  drop % = total drops / (total drops + transmitted) * 100

Often when taken from interface counters, you're using values over hours, days, weeks, even months, so the percentage can be much, much (much) lower than during a short interval.


Drops count are no longer increasing on 9300 switch in the table from my initial post, but Enqueue numbers are increasing constantly - is it something abnormal or it is as it should be?

Before packets are dropped, they should be enqueued.  Drops are overflows of queues.

Usually enqueuing a packet is better than dropping a packet, but not always.  Enqueuing adds latency.  For VoIP, both drops and latency (and jitter) are bad.

With VoIP, a dropped packet loses some actual voice sound.  A too delayed packet, might be ignored (logically dropped), but the delayed packet can further delay other VoIP packets, which may cause those to be ignored too.

In contrast to real-time VoIP or video, steaming voice or video initially delays/buffers a few seconds.  This initial pause, assumes, most of the time, the sender can send faster than needed, but temporary transmission delays are made up by the buffered data already received.  For such applications, delay and jitter are covered by the buffer, but packets drops are still problematic.

BTW, any of you phones generate MOS scores?

If your VoIP is properly directed to a PQ queue, it shouldn't normally be dropped or enqueued.

What I don't know is whether your VoIP traffic is properly classified and treated.

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