Latency
This is the time delay in moving the voice packets from the source to the destination. In general this measure should not exceed 150ms in one direction to prevent deterioration of call quality.
Jitter
This is essentially the variability in packet delay. As far as the source endpoint is concerned, the packets have been sent in a continuous stream
But since each packet may take a different route to its destination, network congestion or improper configuration can result in significant variations in packet delay.
Jitter that exceeds 40ms will cause severe deterioration in call quality. High levels of jitter is usually a consequence of slow speeds or congested networks.
Jitter measurement
Jitter may be measured in a number of different ways, several of which are detailed in various IETF standards for RTP such as RFC 3550 and RFC 3611. Some of these methods are Mean packet to packet delay variation, Mean absolute packet delay variation, Packet delay variation histograms and Y.1541 IPDV Parameter.
Very Good RFC reading for RTP
https://tools.ietf.org/html/rfc3550#page-94
https://www.sciencedirect.com/topics/computer-science/jitter => Jitter various things
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