On our office network (26 people), some users have complained of poor Skype call quality, particularly in the upstream direction. I wanted to ask, how do I identify Skype traffic, considering that it uses a random port, in order that I might prioritise it at the router level?

4 Answers 4


I think you are right; Skype uses random non-common outbound ports (greater than 1024) for each session which makes QoS tagging problematic

But you can use QoS for SIP and provide voice data priority as long as you have SIP Quality of Service-based hardware.

  • How does the SIP QoS hardware prioritise Skype? I can buy whatever hardware is needed!
    – chrism2671
    Oct 29, 2011 at 21:31
  • @chrism2671, you can find Skype's description of QoS here link. Also, SIP protocol tagging is strait forward. Here link is a good article on how to implement the service on Cisco products.
    – Fergus
    Oct 30, 2011 at 3:08
  • Neither article give any indication of how Skype might be identified as SIP traffic. It looks like Skype's documentation applies to using Skype Connect, where the internal traffic is all SIP (not Skype) and can this have QoS applied easily. In these cases, Skype is only used when placing a call to an external system. E.g. SIP Client -> SIP Server -> Skype Connect -> Internet - so the QoS is being applied internally only on SIP traffic.
    – dunxd
    Jun 25, 2012 at 14:14

You can match skype traffic in a class-map using the configuration below on Cisco routers. To do this, the router uses a feature called NBAR (it looks at layer 4 and higher information in the packet to determine the application.)

class-map priority
  match protocol skype

Once matched, you can then give that class higher priority like this:

policy-map outbound
 class priority
  priority 2000 ! Gives a dedicated 2Mbits/sec
interface Gigabit0/1
 description Outside interface
 service-policy output outbound

I'm sure other vendors offer similar functionality, but I can't really say for sure.


If your computers are on a Windows domain , you can use group policy QoS settings to assign DSCP values to traffic generated by the Skype executable file (skype.exe)(actually, you can do the same with local policy on a workgroup computer, you just might need a registry key added too). DSCP 46 I believe is what's usually used for expedited forwarding. Then you'd need to get QoS setup on any switches and routers the traffic will traverse so that it will honor the DSCP ef markings.

  • This solution is interesting. Would you like to improve your answer with references? Apr 14, 2015 at 6:30
  • I found a good walk through of just this found here skype4b.uk/2015/09/07/…
    – Nixphoe
    Feb 17, 2017 at 15:52

Skype for Business already automatically tags its two different kinds of traffic with a Differentiated Services Code Point (DSCP):

  • Audio: EF (Expedited Forwarding, DSCP 46) - low delay, low loss and low jitter
  • Video: AF41 (Assuring Forwarding 41, DSCP 34) - class 4, low drop probability

Classifying by port is wrong

Others might suggest trying to classify Skype traffic by ports (to the point that they might even complain that Skype uses random ports that make it more difficult to find).

Obviously trying to classify Skype traffic by the ports it communicates on is wrong, because then you would mistakenly classify both audio and video into the same priority class - thus defeating the entire purpose of QoS.

Classifying by Skype.exe is wrong

Others might also suggest using QoS policy (gpedit.msc) to apply a DSCP policy to Skype.exe. This, again, is also wrong - as you'll mistakenly classify both audio and video as the same priority.

Skype for Business only

Unfortunately it is only Skype for Business that tags it's different classes of traffic.

Microsoft is petty and petulant, and they don't specifically disable the transparent feature that can only help people in the free version.

Cisco Recommends

For audio and video Cisco recommends:

Traffic DSCP Notes
Voice EF (DSCP 46) No packet loss, 150 ms latency
Interactive video AF41 (DSCP 34) Packet loss 1%, 150 ms latency
Streaming video CS4 (DSCP 32) Packet loss 4-5%, 4-5s latency

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