QoS for 2821 Router as Voice Gateway

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Aug 17th, 2010
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I am a newbie to the the 2821 as my voice gateway and have never setup QoS on the router. We have moved to a new service provider that is a SIP trunk that connects to our router. The router then has dial-peers that go back to our CallManager 8.0 implementation. Everything works fine, however now I am beginning to experience and hear about some QoS issues with call Quality. Being that I have never set QoS up on a router before are there any suggestions to start the process on this. The router is the entry point for the network as well as the default gateway for the network....Router on a stick config. The SP comes in on Gi0/0 and my internal network is setup as sub-interfaces on Gi0/1.10, .100, .101, .102, .103. the .102 network is my voice network.

Thank you in advance for your input.

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paolo bevilacqua Tue, 08/17/2010 - 13:25
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Which side has quality issues, you hearing others, or vice-versa ?

nshoe18 Tue, 08/17/2010 - 13:29
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From what I can tell it is mainly us hearing them....Some jitter and jumpiness to the call quality.

paolo bevilacqua Tue, 08/17/2010 - 14:08
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Then there is not much you can do, as the problem happens on nodes not controlled by you.

Try limiting internet usage, increasing bandwidth, worst case you will need to dedicate a circuit to voice.

philip.e.denton Wed, 08/18/2010 - 05:02
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Either way, if you need some good QoS templates for your environment start with the Enterprise QoS SRND.  It includes templates for many different devices (router and switch models) and many different port roles (access, uplink, WAN, etc).

Aaron Harrison Wed, 08/18/2010 - 05:23
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Is the link to the SP a private one? i.e. just used for voice and nothing else?

A quick check I would make is that the outgoing interface to the voice SP is set as the correct speed/duplex.

Do a show inteface gig x/x to see whether any errors have occurred on the port...


Steven Holl Wed, 08/18/2010 - 09:06
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You should see what the output of 'sh call active voice br' looks like during a call with a voice quality issue.  If you see non-zero values in the lost:x/x/x field for the call leg pointing to the SP, that means they have jitter/packet loss.

If you have issues with voice quality out to your service provider, you could potentially accomadate that with a shaper to the upload speed and a priority queue for the RTP traffic.  That won't affect voice quality in from the PSTN to the IP phone, though.  That's handled at the service provider's bottleneck.



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