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
packetsDiscarded/s
The average number of RTP packets discarded per second by the jitter buffer due to late or early-arrival.
Description
Real number
The average number of RTP packets discarded per second by the jitter buffer due to late or early-arrival, i.e., these packets are not played out.
When using a codec like Opus with RED encapsulation, discarded samples typically correspond to redundant frames that were received but not required.
See also
- inbound-rtp->packetsDiscarded
- inbound-rtp->packetsDiscarded(%)
- inbound-rtp->packetsReceived/s
