How do you ensure IP telephony QoS?

Hi Chelsea,

It is best if QoS is built into the overall strategy of network design. You will need to ensure that you have end-to-end equipment that can support QoS, and that some time is spent classifying enterprise traffic on the network, and establishing the business rules for prioritization.

Voice traffic is more dependent on latency than bandwidth (very little bandwidth is required per call for a clear call), so it is important that voice traffic be ranked ahead of almost everything else.

As for monitoring, there are tools from vendors such as ActionPacked! Networks (http://www.actionpacked.com/products) that provide visibility into the overall health of a network and allow for easy and centralized QoS configuration. SolarWinds (http://www.solarwinds.com/products/orion/application_monitor/) also provides some excellent tools for network and application performance monitoring.

-ASB: http://xeesm.com/AndrewBaker

Use DSCP, and a separate VLAN for voice traffic.

Hi Chelsea
One would hope that you have people who understand VoIP and QOS requirements to set this up; but just for starters; and points not to miss...
Ensure all your Voice traffic is tagged correctly, either at the device e.g. phone or on the ingress port into your network i.e. the switch port it’s plugged into..
Both for the RTP (voice stream) and the signalling... If your using a SIP based system that is purely doing signalling and the voice streams don't traverse the core equipment ensure that the signalling is prioritised in this core area. Along with media servers and such, so in effect all RTP streams must be tagged in the areas they enter the network.. This goes for analogue ports especially, such as mediatrix box's...
Importantly for remote sites, you must have a good understanding of how much traffic you will be sending down these links, but now and the likelihood of expansion; also not forgetting that you will have to recalculate the bandwidth and re-define the policy on WAN links should you increase the traffic...
The main things that are missed by engineers is ensuring nothing other than voice traffic ends up in the priority queues on your routers, if you have a WAN environment, and you have all devices tagged and trusted correctly...
With very high speed switched networks it’s a bit more forgiving nowadays, but still even in the LAN failing to ensure the above can and does cause problems... I've had fax machines using clear channel codec’s. i.e. not using T.38 that fail even with the slightest of jitter on a very good Cisco network.. It took me a while to find the problem but it was down to a switch without the voice traffic in the priority queue.

So in a nutshell, ensure your network supports end to end QOS, but this is no good without everyone involved agreeing on what policies your doing to deploy... Agreement on what is being classified, agreement on where; agreement on what queue this priority traffic is going to be in on the switches and routers and agreement on how much bandwidth will be required in the WAN if you have one... Defining all these points clearly and with everyone involved will go a long way toward having a robust, secure and high quality VoIP deployment. I’ve seen it go wrong so may times and had to sort it out, and it all bee down to lack of planning and miss configuration that I've then had to try and correct.
Fundamentally most QOS enabled equipment on sale now is more than capable of creating an excellent VoIP network, its all down to the groundwork before you implement it and if there's one thing to keep in mind that’s its...
As for ensuing the QOS is being adhered to then on Cisco equipment its easy to see if traffic is being matched and prioritised, and if traffic is being dropped; using external tools its also easy to determine just from the networks traces if your voice and signalling traffic has the correct QOS markings; simply using Wireshark, this will also give you a very good indication of the voice quality, there's simply no need to listen to calls. Tools such as Observer and Solar winds will give a very accurate overview of the quality of the call, so long as the trace is done correctly..
Have fun

Darren

Hello Chelsea Hartfiel, My name is Jonathan Price and I'm VoIP Certified by Mitel Networks and 3COM as they are 2 of the TOP10 companies making VoIP phone systems in the marketplace. I have switched to Hosted PBX, however, due to wanting to get out of the proprietary market and into the INDUSTRY OPEN STANDARDS BASED market. In answer to your question about QoS, you need to have an edge device (meaning, a device located between your demark router that is providing the bandwidth to your solution, and the network switch (presumably you are using a switch with Power over Ethernet with is the industry standard when deploying a VoIP solution). The edge device acts as the bandwidth gate keeper to manage ALL BANDWIDTH that is used on your network and it prioritizes voice on the network for crystal clear voice calls. The 2 TOP brands are Edgemarc (models varywith solutions and it is license sensitive) and D-Link DIR655 (though TRENDnet makes a VoIP Router as well, the D-Link has been deployed longer by more companies). Though there may be additional brands out there, I have been doing telephones for over 20+ years and we sell SIP phones and gateways on eBay and I ask every single buyer what they are using their device on and which platform, so as a PowerSeller with 100% feedback, integrity and knowledge COUNT. Please email/send message for additional information. Jonathan Price