WebRTC Metrics
A comprehensive overview of WebRTC statistics, derived calculations, extracted features, and observable signals, to better understand call quality, connectivity, and user experience in rtcStats
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inbound-rtpinboundaudiovideo
totalProcessingDelay/jitterBufferEmittedCount(ms)
Last updated Applies tortcstats.com
The average processing time in milliseconds of a single audio sample or video frame.
Description
Real number; in milliseconds
The average processing time in milliseconds of a single audio sample or video frame.
Processing time is calculated from the time the first RTP packet is received (reception timestamp) and to the time the corresponding sample or frame is decoded (decoded timestamp).
This metric is calculated by dividing the totalProcessingDelay with the jitterBufferEmittedCount.
See also
- inbound-rtp->totalProcessingDelay
- inbound-rtp->jitterBufferEmittedCount
Notes
- We multiply totalProcessingDelay by a 1000 in the calculation here to convert it from seconds to milliseconds
