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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

jitter(ms)

Packet Jitter measured in milliseconds for this SSRC.

Description

Real number; in milliseconds

Packet Jitter for this SSRC, originally measured in seconds by WebRTC's getStats() API and converted to milliseconds by rtcStats for easier readability. Calculated as defined in RFC3550 section 6.4.1.

It is an estimate of the statistical variance of the RTP data packet interarrival time.

See also

Notes

  • The higher the jitter value is, the worse the media quality is expected to be and the bigger the playout delay will need to be to account for it
  • WebRTC getStats() returns jitter in seconds. For the purpose of visualization, we convert it to milliseconds in rtcStats