Steps to a Successful VOIP System Implementation
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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.
