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
jitterBufferTargetDelay
Total delay accumulated by the jitter buffer across all samples.
Description
Real number; in seconds
This value is increased by the target jitter buffer delay every time a sample is emitted by the jitter buffer. The added target is the target delay, in seconds, at the time that the sample was emitted from the jitter buffer. To get the average target delay, divide by jitterBufferEmittedCount.
This value is an accumulator.
See also
- inbound-rtp->jitter
- inbound-rtp->jitterBufferDelay
- inbound-rtp->jitterBufferEmittedCount
- inbound-rtp->jitterBufferMinimumDelay
- WebRTC Statistics Specification
