What you need to know about your network and VoIP traffic
To ensure that VoIP will work well on your network, you need to take several things into account
before you implement VoIP or begin making network changes.
First, keep in mind that VoIP is a mission-critical network application. Perhaps, until this point, your most critical company application was email. Although email is very critical, it typically demands little of a network because it is forgiving of network delays. Email was designed to be queued and slowly travel the Internet. Thus, when your network has slowdowns, it doesn't mind.
Second, VoIP traffic needs a solid infrastructure. Those old hubs in the wiring closet just aren't going to cut it if your new VoIP phones are to provide high-quality calls. You will need a network infrastructure that is fast and reliable to speed along that VoIP traffic with minimal delays. Also, ensure that your network is secure -- free of viruses, worms and intrusions. You don't want this mission-critical application brought down by a security hole.
Third, you will need to implement Quality of Service (QoS) on your network.VoIP traffic cannot be competing with someone downloading a 100 MB PowerPoint presentation from the file server or someone listening to streaming audio from the Internet. The network must be designed so that VoIP has higher priority than just about every other application currently on the network. That is because VoIP is more delay-sensitive than 99% of other applications typically run on your network. In fact, audio is more sensitive to delay than video.
Five changes you should make to adjust your network for VoIP traffic
1. 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.
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. 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. Find a troubleshooting tool for VoIP traffic. Network troubleshooting tools especially for VoIP are available from Network Instruments, NetIQ, Wireshark 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?
4. 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.
More on adjusting networks for VoIP
5. 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.
About the author:
David Davis (CCIE #9369, CWNA, MCSE, CISSP, Linux+, CEH) has been in the IT industry for 15 years. Currently, he manages a group of systems/network administrators for a privately owned retail company and authors IT-related material in his spare time. He has written more than 50 articles, eight practice tests and three video courses, and has co-authored one book. His website is HappyRouter.com.
Ask the expert: Will more bandwidth fix latency and jitter in your network?
This was first published in April 2007