One of our customers wants us to implement QoS for VoIP on their WAN link.
The customer end of it seems to be easy to set up (Cisco 881), but I have no idea how I should do this on our 6500 that is responsible for terminating customer WAN Links on layer 3 (using subinterfaces).
It seems like the 6500 doesn't support the policy-map command "priority" on subinterfaces.
See the attached drawing for topology, equipment and config details.
Does anyone have any suggestions to how I should implement QoS for VoIP on customer WAN links with the equipment I have today?
You would have to place the policy-map on the physical interface, not the sub-interface. You can match against an extended ACL for the priority classes. This would allow you to only admit the appropriate client traffic into that class.
As a side note: Layer two WAN designs, such as the one you've shown above, have a critical flaw with QoS. The provider edge switch (and all the equipment within the carrier's cloud) is not running QoS. This leaves open the possibility that oversubscription/delay/jitter/loss may occur FROM the provider TO your customer edge equipment. All you can control is egress control FROM you TO the provider. Two easy examples:
You have a 100 customers terminating on that physical interface from the provider. Each customer has 20Mbps of bandwidth from the provider and the link is running at 100Mbps. What happens when half the customers max out the link? The provider has no mechanisms to ensure that the priority traffic of other customers still makes it to you.
The customer site has a 10Mbps CIR on a 100Mbps access rate Ethernet interface. What happens when users try to pull more than 10Mbps worth of traffic across the WAN? The provider has no way to protect voice/priority packets to the customer edge device where the link is constrained to only 10Mbps.
Then of course there's the provider cloud. Congestion can occur there as well.
Layer three WAN designs, typically MPLS these days, is more conducive to QoS.
The bandwidth reservation amounts are a design decision. You need to understand how many calls are involved; the Cisco UC SRND has a section on Bandwidth Provisioning that can help you understand how much to assign. There is also a Voice Codec Bandwidth Calculator available.
As for sample configurations: the QoS SRND has a section on the 6500 which includes sample configurations. While that section is intended for the access-layer it will show you the MQC configuration and ways the supervisor/ASICs function. The WAN Aggregator section more directly maps to your use case.
PS- Yes, I know it's a lot of documentation. QoS - particularly at the head-end - is requires a lot of thought.
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