Question regarding QoS Configuration for SIP Trunks (and understanding of SIP Trunks)

I am hoping someone can provide some clarification and recommended QoS treatment for SIP Trunks.

In my environment I have several remote sites that leverage SIP trunks over the WAN between remote PBX/gateways and a centrally hosted Voice Mail system. We are now having conversations about how to properly classify and mark the SIP trunk traffic.

Where my confusion lies, is SIP is a signaling protocol, and further to that point Cisco’s recommendation from a QoS perspective is to mark it as cs3, with other signaling traffic.
Why this confuses me, how does the voice bearing traffic (to/from the vmail server) get transported.

- Does the call to the vmail via the SIP trunk, initiate another flow for the voice bearing traffic (i.e. rtp flow),
- Are those packets voice bearing packets somehow encapsulated within SIP ?

If the latter, then is it common to place SIP traffic in a LLQ ?

Any feedback/insight would be greatly appreciated.


  • Hi 1000baseT,

    in my daily work experience SIP traffic is never prioritized with LLQ.

    it is classified and treated with other signaling protocols like Skinny for example since it is still used by some old Cisco devices.

    In reality SIP does not carry voice packets. It just initiate the call procedure to a remote site. This is because Voice packets are transported by RTP protocol and RTP uses several ports for communication, while SIP uses well known UDP port 5060 to initiate the call. You should see them as a two separated phases of the phone call setup. What really should be prioritized in my opinion is the RTP protocol.



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