Different voice packets can end up with different amounts of delay. One voice packet may be delayed by 100ms, and another by 300ms. We call this phenomenon "jitter," "packet jitter," or "delay jitter."It turns out that jitter is problematic for VoIP, because it interferes with reconstructing the analog voice stream. With large jitter, we end up with more delayed packets that end up getting discarded, and that can lead to a gap in the audio. Too many dropped sequential packets can make the audio unintelligible. Because the human ear is pretty intolerant of audio gaps, audio gaps should ideally be kept below 30ms, but depending on the type of voice codec used and other factors, audio gaps between 30 to 75ms can be acceptable.The main VoIP application mechanism for mitigating jitter is maintaining a buffer, called the "jitter buffer" or the "play-out buffer." This mechanism helps to smooth out and hide the variation in delay between different received packets, by buffering them and playing them out for decoding at a steady rate. There's a tradeoff here, though. A longer jitter buffer reduces the number of packets that are discarded because they were received too late, but that adds to the end-to-end delay. A shorter jitter buffer will not add to the end-to-end delay as much, but that can lead to more dropped packets, which reduces the speech quality.
