QoS config for 3750X

[Zaaf Aba]

Hi all

I have gone through the QoS documentation for Cisco C3750X and other posts on the form.  Managed to find answers to some of my questions but there are still few which need clarification please. We have non Cisco voice solution (Gateway and phones) running over Cisco LAN and L3 cross site connection.

Questions are listed below as individual line item for ease of answering. Feel free to answer any number of questions. Your assistance will be highly appreciated.

- sh ver shows

Cisco IOS Software, C3750E Software (C3750E-IPBASEK9-M), Version 15.0(2)

License Level                   : lanbase

The above seems to be conflicting to me e.g IPBASEK9 and lanbase? on Cisco C3750X edge (stacked with 32G stack ring bandwidth).

- Keeping security in mind what should be choose, auto qos voip cisco-phone or auto qos trust cos or auto qos trust cos and then tweak the generated config if needed? Or mark voice traffic  manually in conjunction of limiting the bandwidth ingressing the access port using classmap?

- If I trust cos or dscp ingressing access port then do I need to trust cos or dscp for the egressing etherchannel trunk (or physical ports in etherchannel) at access switch or I need to set the DSCP again for voice data egressing trunk etherchannel? Should a second classmap be used (with bandwidth or priority command with 2mbps) because the one used for access port will have lower data rate e.g 128kbps?

- As QoS is end to end, what to do for the voice ingressing the trunk etherchannel on the Cisco Core and similarly for egressing voice data on trunk etherchannel of the core?

- Most of the example I have seen QoS config is applied to the outbound traffic. In which cases can the config be applied to the input traffic as it will be FIFO be default? Will there be added  benefit of applying config to the inbound traffic?

- My understanding is that NBAR isn't available on 3750x? so should I configure extended acl and match  dest ip and UDP port numbers because SIP/SDP/RTP don’t have their own ip protocol number as they are covered under UDP (protocol number 17). Hence can't match SIP or RTP in the acl using protocol. Instead it is easier to match port number for sip (UPD5060,5061,5060) whereas rtp has random high port numbers.  How can that be achieved?

- People mentioned RSVP is old way of configuration whereas in some examples bandwidth and priority command is used for reserving bandwidth. Do both of these commands use something else other than RSVP?

Thanks for your assistance in advance.

[Dennis Mink]

Carlos, see my 10 cents worth below:

- Keeping security in mind what should be choose, auto qos voip cisco-phone or auto qos trust cos or auto qos trust cos and then tweak the generated config if needed? Or mark voice traffic  manually in conjunction of limiting the bandwidth ingressing the access port using classmap?

If your phones are configured to set a certain cos value, then the access port can be set to trust

- If I trust cos or dscp ingressing access port then do I need to trust cos or dscp for the egressing etherchannel trunk (or physical ports in etherchannel) at access switch or I need to set the DSCP again for voice data egressing trunk etherchannel? Should a second classmap be used (with bandwidth or priority command with 2mbps) because the one used for access port will have lower data rate e.g 128kbps?

either or, but I would choose to trust. you could set then manually on the qos Policing L3 device though.

- As QoS is end to end, what to do for the voice ingressing the trunk etherchannel on the Cisco Core and similarly for egressing voice data on trunk etherchannel of the core?

- Most of the example I have seen QoS config is applied to the outbound traffic. In which cases can the config be applied to the input traffic as it will be FIFO be default? Will there be added  benefit of applying config to the inbound traffic?

- My understanding is that NBAR isn't available on 3750x? so should I configure extended acl and match  dest ip and UDP port numbers because SIP/SDP/RTP don’t have their own ip protocol number as they are covered under UDP (protocol number 17). Hence can't match SIP or RTP in the acl using protocol. Instead it is easier to match port number for sip (UPD5060,5061,5060) whereas rtp has random high port numbers.  How can that be achieved?

If you trust dscp values af41 and ef, you dont need to define RTP and signalling, as realitime traffic and signalling can be defined through dscp/cos values and not acls

- People mentioned RSVP is old way of configuration whereas in some examples bandwidth and priority command is used for reserving bandwidth. Do both of these commands use something else other than RSVP?

RSVP is an end to end protocol, QoS is neither a protocol nor is it end to end, so you cant compare these two.  QoS is only good on the device you configure it on and only kicks in when there is contention.

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