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Newbie: monitoring using Cisco IP SLA

ravi.malghan
Level 1
Level 1

Hi: I have been asked to provide some recommendations on how to monitor a network that is going to be used for VoIP. Our network has two main sites (primary and secondary call manager) and multiple remote sites. Want to know if there are some best practices around how the IP SLA should be configured on the routers. I am thinking I will configure IP SLA monitoring on each remote router to both the main sites and IP SLA monitoring between each the two main sites as well. Send a SNMP trap if

one way Jitter > 30ms,

Latency > 150ms,

Packet Loss > 2%

MOS Scores (3.2 - 4.0)

I am planning to use Prognosis to monitor some stuff at the application layer.

Thanks

Ravi

2 Replies 2

Hi Ravi,

Are these numbers something you've agreed on, or something you're using for best practice?

If it's something you've come up with, I would adjust the numbers a bit. I would probably ditch the MOS score, as it can be misleading and if you're monitoring the 'hard' statistics it will probably be unnecessary. I would also lower packet loss to 1% or even 0.5% if possible.

If you lose an evenly distributed 1-2% of packets you could be fine. But, that's generally not how they're lost. And then the voice sounds awful, because they're grouped together.

Here's some example config that I've used in a lab scenario testing the feature:

ip sla monitor 3

type jitter dest-ipaddr 172.16.1.2 dest-port 16384 codec g711ulaw codec-numpackets 30

tos 184

frequency 10

ip sla monitor schedule 3 start-time now

ip sla monitor 4

type jitter dest-ipaddr 172.16.35.5 dest-port 16386 codec g711ulaw codec-numpackets 20

tos 184

frequency 5

ip sla monitor schedule 4 start-time now

Receiving side:

ip sla monitor responder

R3#sh ip sla moni stat

Round trip time (RTT) Index 3

Latest RTT: 54 ms

Latest operation start time: 00:11:46.823 UTC Tue Jan 1 2008

Latest operation return code: OK

RTT Values

Number Of RTT: 13

RTT Min/Avg/Max: 108/124/153 ms

Latency one-way time milliseconds

Number of one-way Samples: 0

Source to Destination one way Min/Avg/Max: 0/0/0 ms

Destination to Source one way Min/Avg/Max: 0/0/0 ms

Jitter time milliseconds

Number of Jitter Samples: 7

Source to Destination Jitter Min/Avg/Max: 5/6/7 ms

Destination to Source Jitter Min/Avg/Max: 1/1/1 ms

Packet Loss Values

Loss Source to Destination: 17 Loss Destination to Source: 0

Out Of Sequence: 0 Tail Drop: 0 Packet Late Arrival: 0

Voice Score Values

Calculated Planning Impairment Factor (ICPIF): 25

Mean Opinion Score (MOS): 3.50

Number of successes: 20

Number of failures: 0

Operation time to live: 3393 sec

Round trip time (RTT) Index 4

Latest RTT: 2 ms

Latest operation start time: 00:12:01.167 UTC Tue Jan 1 2008

Latest operation return code: OK

RTT Values

Number Of RTT: 20

RTT Min/Avg/Max: 2/2/2 ms

Latency one-way time milliseconds

Number of one-way Samples: 0

Source to Destination one way Min/Avg/Max: 0/0/0 ms

Destination to Source one way Min/Avg/Max: 0/0/0 ms

Jitter time milliseconds

Number of Jitter Samples: 19

Source to Destination Jitter Min/Avg/Max: 0/0/0 ms

Destination to Source Jitter Min/Avg/Max: 0/0/0 ms

Packet Loss Values

Loss Source to Destination: 0 Loss Destination to Source: 0

Out Of Sequence: 0 Tail Drop: 0 Packet Late Arrival: 0

Voice Score Values

Calculated Planning Impairment Factor (ICPIF): 1

Mean Opinion Score (MOS): 4.34

Number of successes: 6

Number of failures: 0

Operation time to live: 3570 sec

hth,

nick

Nick: that you for the suggestions. I am looking for some best practices or settings that have worked for others.

How would you suggest the general architecture. Would you measure these metrics between all sites or just from remote sites to sites which have the call manager?

TIA

Ravi