Steps to a Successful VOIP System Implementation

 
 
By Tim McCreery  |  Posted 2008-12-23 Email Print this article Print
 
 
 
 
 
 
 


The following five steps are critical to a successful VOIP system implementation:

Step #1: Establish a thorough baseline

Establish a thorough baseline of current network activity on all segments that will host VOIP. Failing to understand the degree to which latency, jitter and packet loss affect your network before deploying VOIP is nothing less than negligent. You must understand current network load and behavior, including any areas where latency is elevated or highly variable.

Network segments that are prone to packet corruption and loss must be diagnosed and healed. In many networks, traffic loads may vary substantially over time. As loads increase, inconsistent packet delivery rates are probable. Thus, increasing loads form the foundation for excessive latency and jitter-which are two of the most prevalent inhibitors for consistent VOIP performance.

When collecting baseline metrics, remember that network behavior varies widely as various business activities occur. Be sure to create a baseline that reflects all major phases and facets of your network's activities. There is no acceptable reason for enduring a poor VOIP implementation when a solid, proactive baseline could have predicted inferior performance.      

Step #2: Analyze store-and-forward and queuing congestion

Analyze store-and-forward and queuing congestion in switches and routers. Congestion can lead to packets spacing unpredictably and thus resulting in jitter. Keep in mind that the more hops a packet has to travel, the worse the jitter. Try to reduce hops as much as possible. Latency due to distance is a fixed fact of physics; there is nothing that can be done to change the time needed for a packet to travel a given number of meters or miles. However, devices that segment networks impose latency that is often highly variable. Jitter is primarily caused by these device-related latency variations. As a device becomes busier, packets must be queued. If those packets happen to be VOIP audio data, jitter is introduced into the audio stream and audio quality declines. 

While many switch and router vendors will advertise that their products can handle a certain level of throughput, few talk about the volume of packets that can be processed during periods of peak utilization. For example, even though a switch might be able to accommodate a full line rate traffic stream when all packets are nearly maximum size, it may not be able to manage the same aggregate throughput when the stream is composed of many more minimum-sized packets. Since most Real-Time Protocol (RTP) audio packets are relatively small (just over 200 bytes for G.711), a device's ability to process packets of that size at full line rate must be assured. Understanding how a device reacts to traffic streams characterized by many short bursts of many packets is also important.



 
 
 
 
Tim McCreery is President and CEO of WildPackets, Inc. Tim has over 25 years of experience in the networking industry. Tim co-founded WildPackets as AG Group in 1990. Tim's past positions include Founder and President of Kinetics, Founder and CEO of SilkStream Corporation, and VP of Marketing and Business Development at Excelan, Inc. Tim has also been on the board of other startups, including Clear Ink Corporation and Tut Systems. Tim is also President of the Board of Youth Homes, Inc., a non-profit agency serving at-risk foster care children in Contra Costa County, Calif. Tim graduated from the University of California, Berkeley with degrees in Mathematics, Computer Science and Psychology. Tim taught undergraduate Computer Science at U.C. Berkeley while obtaining a Master's degree in Electrical Engineering and Computer Sciences. He can be reached at tim.mccreery@wildpackets.com.
 
 
 
 
 
 
 

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