QoS for VOIP Freepbx Setup

[speakercon]

Evening everyone.

 

I'm currently experimenting with setting up a Voip system to replace our old pbx. I've used freepbx at home for a few years so I'm pretty familiar with setting that up, provisioning the phones, dialplans ext...

 

So far I've installed the Distro on an old server, and setup 10 Cisco 7941 phones (in SIP mode) and all is working well.

 

My main concern now is the QoS across the LAN (we will still be using our ISDN lines for external via a Patton Gateway for now). At the moment the call quality between extensions is excellent, however what with the school being almost empty the voice traffic doesn't have much to battle against.

 

Our network has a 1G backbone, and a mixture of 100mb/1G switches (Cisco & HP). We have a number of VLANs across the site.

 

My understanding is that the phones & Asterisk by default (in sip.conf & on the cisco phone status page) add the CoS value (5) and the DSCP code (cs3) to the packets they transmit, so is it a case of making sure 802.1p is enabled on all managed switches? Or is it necessary to setup a new Vlan just for voice and create a trunk between each phone & the switch (tagged voice vlan & untagged data vlan for the pc passthrough)?

 

Any experience of your own way of doing this would be interesting to hear, along with suggestions on the best method in this situation.

 

Many many thanks in advanced!

[MatthewL]

In the past when I've done deployments of these mainly been Avaya or in some instances managed Cisco the phones have always sat on their own VLAN and PC's on another.

 

That way traffic is separate and QoS can be ramped up on that VLAN.

[speakercon]

Thanks for the input, looks like both our HP's & Cisco switches support voice vlans so will probably go down that route, and use 802.1p for QoS.