Reported in a handler function for CallEvents.QualityIssueHighMediaLatency Latency is calculated based on rtt (round trip time) and jitter buffer delay. Latency refers to the time it takes a voice/video packet to reach its destination plus the time it waits in a jitter buffer. Sufficient latency causes call participants to speak over the top of each other. The issue level may vary during the call. Possible reasons:
- Network congestion/delays.
- Lack of bandwidth.Only in Chrome