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How to Implement QoS for VoIP/RTP

I'm running R77.30 and Gaia on a cluster of two 4800 series gateways.

My organization recently deployed Cisco IP Communicator softphones to some remote VPN users. However, the other parties to calls often report that the voice of our VPN users is cutting in and out for them. This is caused by delay in our VPN users' RTP voice packets reaching the other party.

To minimize this delay, I'd like to deploy QoS. I've read most of the QoS Admin Guide for R77, so I understand the basic functionality of Check Point QoS, and that I should be using either DiffServ Expedited Forwarding or Low Latency Queuing with priority 1. But I'm not sure how to define RTP as a service when creating a rule in the QoS rulebase. There is no predefined RTP service, so that's not an option. My first thought is to use the actual port range, which is UDP 16384 through 32767. However, my concern is that non-RTP traffic will be matched by this rule and consume bandwidth meant for actual RTP traffic.

I opened a ticket with Check Point support, but this was the response offered,


Also I did some research on my end and to my knowledge there is no way to define the RTP service so that only RTP traffic gets matched by the rule. You can create a custom site for ports 16384 through 32767 but you cannot ensure that only RTP traffic matches the rule.


Is this true? Since SmartView Tracker can break down VoIP calls and even estimate codecs being used, I would think it has to have a way to effectively implement QoS for VoIP, since VoIP/video are two of the biggest needs for it in the first place.

Does anyone have experience successfully configuring QoS for RTP traffic? Any advice is greatly appreciated.

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