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Introduction
audioLevelbytesReceivedconcealedSamplesconcealedSamplesfecBytesReceivedfecPacketsDiscardedfecPacketsReceivedfirCountframeHeightframesDecodedframesDroppedframesPerSecondframesRenderedframeWidthfreezeCountheaderBytesReceivedinsertedSamplesForDecelerationjitterjitterBufferDelayjitterBufferEmittedCountjitterBufferTargetDelayjitterBufferTargetDelaykeyFramesDecodedpacketsDiscardedpacketsLostpacketsReceivedpacketsReceivedWithCepacketsReceivedWithEct1packetsReportedAsLostpacketsReportedAsLostButRecoveredpauseCountpliCountpliCountqpSumremovedSamplesForAccelerationsilentconcealedSamplestotalAudioEnergytotalDecodeTimetotalFreezesDurationtotalInterFrameDelaytotalPausesDurationtotalProcessingDelaytotalSamplesDurationtotalSamplesReceivedtotalSquaredInterFrameDelay
totalPlayoutDelaytotalSamplesCount
qualityLimitationDurations

jitter

inbound-rtpinboundaudiovideo

Packet Jitter measured in seconds for this SSRC.

Description

Real number

Packet Jitter measured in seconds for this SSRC. Calculated as defined in RFC3550 section 6.4.1.

It is an estimate of the statistical variance of the RTP data packet interarrival time, measured in timestamp units.

See also

  • WebRTC Statistics SpecificationW3C

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
rtcStats logortcStats

Troubleshoot your WebRTC application with ease.

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