Traditional telephone services have typically gained a reputation of providing excellent voice quality and superior reliability. Consequently, users take for granted that their phone systems will provide high quality with virtually no downtime. Yet many voice over IP (VOIP) installations fail to meet these expectations, primarily because organizations have not adequately evaluated their network infrastructure to determine whether it can adequately support applications that are very sensitive to latency, packet loss, jitter and other similar performance factors.
VOIP requires a steady, predictable packet delivery rate in order to maintain quality. Jitter, which is variation in packet delivery timing, is the most common culprit that reduces call quality in VOIP systems. Jitter causes the audio stream to become broken, uneven or irregular. As a result, the listener's experience becomes unpleasant or intolerable.
The end results of packet loss are similar to those of jitter but are typically more severe when the rate of packet loss is high. Excessive latency can result in unnatural conversation flow where there is a delay between words that one speaks versus words that one hears. Latency can cause callers to talk over one another and can also result in echoes on the line. Hence, jitter, packet loss and latency can have dramatic consequences in maintaining normal and expected call quality.
Some VOIP systems are all but unusable from the time they are implemented because nobody took the time to properly profile the existing network. Before implementing VOIP, a thorough application impact study should be completed.