Alternatively, can anyone describe for me in plain English what the steps are for formulating a QOS policy?
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Basic QOS policy for Huawei router
Hi experts
I know sod-all about VOIP, so I'm hoping you can help, here. I have just had one Avaya IPOffice 406v2 system installed in our main office, and another in our branch office, which is at the end of a dedicated kilostream link. This link carries data between our networked workstations and servers at both sites (all are on a single domain and subnet). The link is serviced by 2 Huawei Quidway AR28-09 routers (one at each end) acting as a bridge.
I've allocated the 2 ipOffice boxes ip addresses within the network subnet, and they talk across the link just fine. However, I'm aware that I should be trying to control the voice traffic with a QOS policy, and I believe my routers will support it, but I don't know how to structure the policy, and therefore what commands to implement. I am currently talking to the routers via their CON port using hyperterminal, and the command set from the Huawei manuals which I managed to begborrowsteal for them.
I will upload the manual for the QOS command set for the routers to ee-stuff, and I can provide output from any commands, eg
[Display Current-configuration]
Now create configuration...
Current configuration
!
version 1.74
firewall enable
aaa-enable
aaa accounting-scheme optional
bridge enable
bridge 1 stp ieee
!
interface Aux0
async mode flow
link-protocol ppp
!
bridge-set 1
!
interface Serial0
clock DTECLK1
link-protocol ppp
bridge-set 1
!
return
[Display Interfaces]
Aux0 current state:up, line protocol current state:up (spoofing)
The Maximum Transmit Unit is 1500
physical layer is asynchronous, baudrate is 9600 bps
Link-protocol is PPP
LCP initial, IPCP initial, IPXCP initial, CCP initial, BRIDGECP initial
Input queue : (size/max/drops) 0/50/0
FIFO queueing: FIFO
(Outbound queue:Size/Length/Discards
FIFO: 0/75/0
Last 5 minutes input rate 0.00 bytes/sec, 0.00 packets/sec
Last 5 minutes output rate 0.00 bytes/sec, 0.00 packets/sec
Input: 0 packets, 0 bytes
0 broadcasts, 0 multicasts
0 errors, 0 runts, 0 giants
0 CRC,0 frame errors, 0 overrunners
0 aborted sequences 0 no buffers
0 packets with dribble condition detected
Output:1 packets, 27 bytes
0 broadcasts, 0 multicasts
0 errors, 0 underruns, 0 collisions
0 packets had been deferred
Ethernet0 current state:up, line protocol current state:up
Description: Ethernet interface, Ethernet0 Interface
The Maximum Transmit Unit is 1500
IP Sending Frames' Format is Ethernet_II,Hardware address is 00-e0-fc-2a-86-0a
Media type is twisted pair, loopback is not set, promiscuous mode set
Full-duplex mode, 100Mbps-speed mode, link type is autonegotiation
Input queue : (size/max/drops) 0/200/0
FIFO queueing: FIFO
(Outbound queue:Size/Length/Discards
FIFO: 0/75/0
Last 5 minutes input rate 1090.44 bytes/sec, 3.15 packets/sec
Last 5 minutes output rate 583.94 bytes/sec, 3.13 packets/sec
Input: 14822890 packets, 2188455972 bytes
515594 broadcasts, 14330 multicasts
0 errors, 0 runts, 0 giants
0 CRC,0 frame errors, 0 overrunners
0 aborted sequences 0 no buffers
0 packets with dribble condition detected
Output:14505838 packets, 2237588944 bytes
0 broadcasts, 0 multicasts
0 errors, 0 underruns, 0 collisions
0 packets had been deferred
Serial0 current state:up, line protocol current state:up
The Maximum Transmit Unit is 1500
physical layer is synchronous
interface is DTE, clock is DTECLK1, cable type is V35
Link-protocol is PPP
LCP opened, IPCP initial, IPXCP initial, CCP initial, BRIDGECP opened
Input queue : (size/max/drops) 0/50/0
FIFO queueing: FIFO
(Outbound queue:Size/Length/Discards
FIFO: 0/75/47
Last 5 minutes input rate 576.16 bytes/sec, 2.83 packets/sec
Last 5 minutes output rate 1131.27 bytes/sec, 3.85 packets/sec
Input: 13520295 packets, 2237239417 bytes
0 broadcasts, 0 multicasts
31 errors, 0 runts, 0 giants
17 CRC,7 frame errors, 7 overrunners
0 aborted sequences 0 no buffers
0 packets with dribble condition detected
Output:17122363 packets, 2349279100 bytes
0 broadcasts, 0 multicasts
0 errors, 0 underruns, 0 collisions
0 packets had been deferred
DCD=UP DTR=UP DSR=UP RTS=UP CTS=UP
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Answers
Are you noticing problems on calls between sites? If not, this may not even be worth the headache to look into.
The trick to QoS is that you have to mark packets pretty much at the endpoint and use the routers to track and prioritize marked traffic accordingly. So, theory first.
Let's say your IP phones mark packets with a precedence of 5 where your data has a priority of 1, the router looks at that packet and says "ah ok, you're voice RTP you go in the high priority queue" or "oh, you're just a data packet, low priority". then you use something like WRED to handle packets accordingly.
the trick is that if you're Avaya system isn't marking packets properly, it won't make a difference since it's traffic will be seen as data. If that IP phone or PBX doesn't first say "hey, I'm voice" then the router won't be able to tell the difference.
There are different ways to approach this problem. You'll probably want to do a mix of them, depends on your setup. For example, say your use interface serial 0/1 go to that interface then use a command like 'qos wred 5 10 50 10' whcih says create a qos using weighted random early detection so that anything with IP precedence 5 is treated with a certain discard probability. None of that makes any sense in clear English. Nor will it until you dig through the different options you have for QoS.
In general the steps are
1. end device marks packets (tosbit, diffserv, dscp, whatever)
2. router interface looks at the packet on ingress and does one of a few different things.
a. puts it in a queue for egress and waits until bandwidth is available
b. discards the packet because it's all congested
c. puts it in a priority queue and sends the packet out because there's plenty of bandwidth.
Keep in mind this has to be done at both routers so that traffic flowing in both directions is prioritized.
Thanks for your response, rp.
You're right, of course - I don't actually have a problem at the moment, and I hope I'm not going to have to do anything at all. I'm letting test volumes of calls go down my data link at the moment, but I suspect if I let everyone use it hell-for-leather, my data will suffer. I know QOS policies exist to allow voice to be prioritised so that it can be understood clearly over the line, but if voice needs so much bandwidth on my link that my data transfer suffers, then its bye-bye to the VOIP, I'm afraid.
I understand the theory, so I guess I'm going to have to wing it if and when I decide to fiddle about with my router configs. My Avaya supplier assures me that "his" packets are properly marked, and I'm currently only using the ethernet0 interface on the routers, so there's certainly scope for having a good old muck around with it!
I'll leave this Q open for a week or so in case anyone else can offer any suggestions, then close it out.
couple of things to keep in mind. Because QoS can be implemented in a couple of different ways, find out exactly how your Avaya system is marking packets. There are different interpretations of standards that can lead to issues. If, for example, your vendor says that call signal should be marked 5 and RTP marked 3 where another link marks in the other way around, you'll potentially run into the same issues.
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by: jayneePosted on 2007-06-27 at 09:18:54ID: 19373692
Here is the direct link for the qos command set manual:
http://www.ee-stuff.com/Ex
Thanks
jaynee