Voice over IP

VoIP is simply the ability to make telephone calls, send facsimiles, or transmit audio over IP-based data networks with a suitable quality of service (QoS) and a cost/benefit ratio superior to traditional systems. VoIP transmits voice signals in digital form in discrete packets rather than in the protocols of traditional switched circuits.

Three major factors can impact the quality of the service:

• Delay: Significant end-to-end delay in a voice network may result in echo and talker overlap. Echo becomes a problem when one-way network delay is more than 50 milliseconds. VoIP systems must implement some means of echo cancellation. If the round-trip delay is greater than 250 milliseconds, talker overlap--one caller stepping on the other talker's speech--is likely. For adequate quality of service, network delay between endpoints should be less than 50ms.
• Jitter (Delay Variability): Jitter results when data packets arrive at their destination at irregular intervals as a result of variable transmission delay over the network. To remove jitter, the VoIP engine must collect and hold data packets in a buffer long enough for the slowest packet to arrive and be played in sequence. A jitter buffer, however, adds to delay. Jitter of less than 20 ms. between endpoints is normally required.
• Packet Loss: Under peak network loads and periods of congestion, packets may be dropped. Because voice transmission is highly time-sensitive, normal TCP-based re-transmission schemes are not suitable. Methods to compensate for packet loss include interpolation of speech by re-playing the last packet and sending of redundant information. The maximum packet loss between network endpoints should not exceed 0.2%.