Packet Loss & Jitter - VoIP

Packet loss and VoIP

Telephony is all UDP based, and packets may not arrive at the destination, or get discarded if they arrive delayed or contain errors. This results in missing audio information at the destination.
The industry has adopted the Mean Opinion Score (MOS) as the universal metric to measure and classify the conversation quality that happens over a network. As the name suggests, it is based on the opinion of the user and ranges from 1.0 to 5.0 with the following classifications:

MOS Quality Impairment
5 Excellent Imperceptible
4 Good Perceptible but not annoying
3 Fair Slightly annoying
2 Poor Annoying
1 Bad Very annoying

More details at: https://en.wikipedia.org/wiki/Packet_loss

Jitter and VoIP
Jitter is the variation in the delay of received packets. High jitter results in choppy voice or temporary glitches. VoIP devices implement jitter buffering algorithms to compensate packets that arrive at high timing variations, and packets can even get dropped when they arrive with excessive delay.
More details at: https://en.wikipedia.org/wiki/Jitter

Latency and VoIP
Audio latency consists of two parts: the time it takes to encode the audio and the travel time of the packet. The latency itself doesn’t affect the quality of the delivered audio, but it can affect the interaction between the two end users. At 100 ms of latency, the users start talking on top of each other, and at 300 ms, the conversation becomes impossible to follow.
More details at: https://en.wikipedia.org/wiki/Latency_(audio)