AWC and QoS 1

[slapin]

Looks like AWC application is not marking UDP/RTP packets with EF DSCP=46 marks. Users, who call the bridge over our private network from remote locations are having voice quality issues. I have opened a trouble ticket, but maybe someone knows how to enable it in the application or which range of UDP ports is being used for voice traffic. So far I was observing UDP ports 120xx. It would be nice to know for sure, becuse I have included this into our QoS policy on the edge router. I quickly scanned through documentation and could not find the answer.

 

[IrwinMFletcher]

The ports start at 12000 and increment by 2 up to the max number of ports, which I believe is 96.

 

[IrwinMFletcher]

There is also no way to change the DSCP value.

 

[slapin]

This is kind of odd as everybody knows that VoIP doesn't work without it. They cannot expect that people will not try to use it in their private networks. Marking traffic on the router using the range of ports helps, so I will live with it for a while.

 

[andy4uk]

I have a customer who is using AWC - they are also using Teleworker handsets, calls from the Teleworker handsets work ok...... but when they call the AWC the voice quality seems to be very poor.

Has anyone else experienced this problem?

Andy

 

[slapin]

This all is not straight forward like connecting two wires. You need to describe your environment. Where is ICP, where is TW where is AWC and how it is all interconnected.

 

[IrwinMFletcher]

Agreed, the way it's all connected is important information. My guess however is that it's an internal routing issue.

 

[Jakt99]

Chaps

Correct me if I'm wrong.

AWC utilises SIP connectivity between MAS and Mitel controller thus if you config the MAS LAN NIC and controller to be on same subnet (vlan and assuming both devices reside in same comp room) and configure the L2 switch(es) to also give the same dscp QOS priority to SIP port packets (5060/1 i think), this should improve voice quality would it not??

Am no L2 switch guru but have seen programming to give same QOS priority to dscp and SIP packets in HP procurves before. No idea about cisco world...

 

[slapin]

The direction of thinking is correct, but in details wrong L2 QoS works with CoS and it is effective on trunk (passing multiple VLANs) ports only. If you have a computer connected through the phone and they are in different VLANs, then on ingress (from computer to the switch) the phone will prioritize ethernet frames. On egress (from the switch to the computer, CoS configured on the port will control frames priority. L3 packets are encapsulated in ethernet frames and they have DSCP marks in their headers. Those marks are invisible for L2 ethernet ports and can work when packets crossing IP interfaces. Considering all this, easy to see that it is important to know details about environment.

 

[IrwinMFletcher]

Giving priority to SIP packets only ensures that the signaling is responsive, it does nothing for the RTP streaming, i.e. voice quality.

If teleworker to teleworker sounds fine, then it's an internal network problem since the tw-tw case is a loopback on the TW box. I'd put a phone on the same subnet as the AWC and make a call to a TW set from it and observe the quality. Actually, first, I'd get more information as to what 'very poor' quality actually means....