WebRTC Metrics
A comprehensive overview of WebRTC statistics, derived indicators, and observable signals, to better understand call quality, connectivity, and user experience in rtcStats
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inbound-rtpinboundaudiovideo
jitterBufferDelay/jitterBufferEmittedCount(ms)
The average time, in milliseconds, each audio sample or a video frame waits in the jitter buffer.
Description
Real number; in milliseconds
This is the average jitter buffer delay: The calculation of the average time each audio sample or video frame waits in the jitter buffer before taken out for playout.
This adds up to the latency observed by the user. We strive for this to be as close as possible to the average jitter on the network.
This metric is calculated by dividing the jitterBufferDelay with the jitterBufferEmittedCount.
See also
- inbound-rtp->jitter
- inbound-rtp->jitterBufferDelay
- inbound-rtp->jitterBufferEmittedCount
Notes
- An audio sample refers to having a sample in any channel of an audio track - if multiple audio channels are used, metrics based on samples do not increment at a higher rate, simultaneously having samples in multiple channels counts as a single sample
- We strive to have this delay as small as possible
