Note: this article originally appeared at www.searchnetworking.com

#1 – Implement QoS on your Routers and Switches

Implement QoS before you implement VoIP. Preferably, this should be end-to-end QoS, which means that voice traffic must be given higher priority than data traffic on every link. I recommend that you use IEEE 802.1p/q enabled switches. When using 802.1q VLANs, the 802.1p priority tag resides in the 802.1q traffic header. Routers should be DiffServ code points (DSCP). 802.1p & q work together.

By implementing this QoS traffic prioritization, you are ensuring timely VoIP packet delivery and providing the best call quality.

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As part of this QoS plan, you should break voice traffic off into its own VLANs to get it away from the data traffic. These VLANs will put the phones on their own broadcast domains and IP subnets.

#2 – Measure VoIP Quality on the Network

Find a way to measure the quality that the VoIP traffic receives on your network. This may be done with a tool provided by your vendor, or it may be included in your VoIP troubleshooting tool, which we talk about next. Without a tool to measure VoIP quality, you will have to take your users’ word for it. It would be better to have a measuring tool that can give you a real metric of VoIP performance.

#3 – Identify your VoIP Troubleshooting Tool – you will need it!

Find a troubleshooting tool for VoIP traffic. Network troubleshooting tools especially for VoIP are available from Network Instruments, NetIQ and many others. When you go to implement a VoIP troubleshooting tool, consider the following questions:

• Where will you deploy analysis tools — on the LAN or WAN, on each VLAN, or at the VoIP call management server?

• What will you measure?

• Can you find a way to be proactively notified if call quality begins to deteriorate?

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#4 – Develop a Network Baseline

Before implementing VoIP, conduct analysis to determine network bottlenecks — develop a baseline. How good is network performance today, before VoIP traffic gets on the network? This baseline will also point out some critical issues about your LAN and WAN. For example, do you have enough bandwidth on the WAN links that the VoIP traffic will be traversing? Even with the estimated maximum number of simultaneous calls, given the codec you have selected, and the other data traffic on the link? For more information, see VoIP Bandwidth: Calculate consumption.

#5 – How to keep VoIP “happy”

Besides knowing your network, know what VoIP needs in order to be a “happy application.” Here are some things you should know:

• Ideally, jitter should be 20 ms or less. VoIP phones use jitter buffers to try to reduce the effect of jitter on the voice call. More severe jitter can result in packet loss. Other causes of packet loss are problems on the physical network. Without QoS, a burst of data traffic can cause packet loss on voice calls. Packet loss should be 1% or less between end-to-end connections, but a VoIP call may tolerate up to 3%.

• Delay (or latency) should be between 80 and 180 ms to get toll-quality voice.

• The standard for measuring quality of VoIP traffic on your network is called PESQ. Various white papers are available on this topic.

I recommend you take a look at my favorite Cisco CCNA / CCDA video training! Click here to see a free Demo!