I am currently dealing with a client who has a MPLS network with latency of around 15ms and 5ms jitter (I am moving them to a NEW WAN with 10ms and 3 ms Jitter with 0.1% packet loss), but I need to understand do I really need QOS to run Cisco Call Manager 5.x?
The links I have run at 10-30% util with data and never peak. The latency is low as I am sure it needs to be under 150ms to deliver voice. If needed I can run QOS and this is not an issue, but a third party will not install a Cisco IPT as at present on the MPLS network as traffic cannot be tagged (not worth the cost as new WAN will be in withing two months). In two months time I can move the client over and run as much QOS as you can shake a stick at, but the latency is 10ms jitter is 3ms, 0.1% loss and 80% bandwidth unused.
I am a big believer that if it's not needed for simplicity don't over complicate it.
Would Cisco support the above if no QOS was enabled over the network (as long as the stats I have provided are correct)?
I have been sent a doc from the third party (a Cisco doc) that mentions QOS on converged networks, but still not clear if I need it as if the bucket is not getting full or reaching its full capacity; the QOS at the top of the bucket is not going to be used?
Any positive or negative feedback on this would be much appreciated.
I don't know about "Cisco support" with regard to QoS provided or not for Cisco VoIP, but in general the problem is without QoS, service levels are not guaranteed. With your currently stated performance measurements, likely VoIP might work fine most, if not all the time, but it's the "all the time" that can be the issue.
QoS is not required in order to get support. If you can show that the network isn't the problem (packet caps from both sides showing latency/jitter during troubleshooting), then they won't hassle you about it.
That being said, it is recommended to do as you never know when it will bite you for not having implemented QoS. It only takes one bad device to start flooding bogus traffic and then your whole voice system starts to deteriorate. On top of that, it also helps to prevent people from hijacking VOIP ip addresses/subnets and using them to abuse your dedicated voice links as their personal data stream... You can limit the guaranteed traffic to exclusively RTP/RTSP/SIP and prevent these kinds of potential attacks or abuses from ever happening.
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