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RTCInboundRtpStreamStats: jitter property - Web APIs

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1. Web
2. Web APIs
3. RTCInboundRtpStreamStats
4. jitter

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• Specifications
• Browser compatibility

1. WebRTC API
2. RTCInboundRtpStreamStats
3. Instance properties
   1. audioLevel
   2. bytesReceived
   3. codecId
   4. concealedSamples
   5. concealmentEvents
   6. estimatedPlayoutTimestamp
   7. fecPacketsDiscarded
   8. fecPacketsReceived
   9. frameHeight
   10. framesAssembledFromMultiplePackets
   11. framesDecoded
   12. framesPerSecond
   13. framesReceived
   14. frameWidth
   15. freezeCount
   16. headerBytesReceived
   17. id
   18. insertedSamplesForDeceleration
   19. jitter
   20. jitterBufferDelay
   21. jitterBufferEmittedCount
   22. jitterBufferMinimumDelay
   23. jitterBufferTargetDelay
   24. keyFramesDecoded
   25. kind
   26. lastPacketReceivedTimestamp
   27. mid
   28. nackCount
   29. packetsDiscarded
   30. packetsLost
   31. packetsReceived
   32. pauseCount
   33. playoutId
   34. qpSum
   35. remoteId
   36. removedSamplesForAcceleration
   37. silentConcealedSamples
   38. ssrc
   39. timestamp
   40. totalAssemblyTime
   41. totalAudioEnergy
   42. totalDecodeTime
   43. totalFreezesDuration
   44. totalInterFrameDelay
   45. totalPausesDuration
   46. totalProcessingDelay
   47. totalSamplesDuration
   48. totalSamplesReceived
   49. totalSquaredInterFrameDelay
   50. trackIdentifier
   51. transportId
   52. type
4. Related pages for WebRTC
   1. MediaDevices.getUserMedia()
   2. Navigator.mediaDevices
   3. RTCAudioSourceStats
   4. RTCCertificate
   5. RTCCodecStats
   6. RTCDTMFSender
   7. RTCDTMFToneChangeEvent
   8. RTCDataChannel
   9. RTCDataChannelEvent
   10. RTCDataChannelStats
   11. RTCDtlsTransport
   12. RTCEncodedAudioFrame
   13. RTCEncodedVideoFrame
   14. RTCErrorEvent
   15. RTCIceCandidate
   16. RTCIceCandidatePair
   17. RTCIceCandidatePairStats
   18. RTCIceCandidateStats
   19. RTCIceParameters
   20. RTCIceTransport
   21. RTCOutboundRtpStreamStats
   22. RTCPeerConnection
   23. RTCPeerConnectionIceErrorEvent
   24. RTCPeerConnectionIceEvent
   25. RTCPeerConnectionStats
   26. RTCRemoteInboundRtpStreamStats
   27. RTCRemoteOutboundRtpStreamStats
   28. RTCRtpReceiver
   29. RTCRtpReceiver.transform
   30. RTCRtpScriptTransform
   31. RTCRtpScriptTransformer
   32. RTCRtpSender
   33. RTCRtpSender.transform
   34. RTCRtpTransceiver
   35. RTCSctpTransport
   36. RTCSessionDescription
   37. RTCStatsReport
   38. RTCTrackEvent
   39. RTCTransformEvent
   40. RTCVideoSourceStats
5. Guides
   1. Introduction to WebRTC protocols
   2. Introduction to the Real-time Transport Protocol (RTP)
   3. WebRTC connectivity
   4. Establishing a connection: The WebRTC perfect negotiation pattern
   5. Lifetime of a WebRTC session
   6. Signaling and video calling
   7. Using WebRTC data channels
   8. Using DTMF with WebRTC
   9. Using WebRTC Encoded Transforms
   10. A simple RTCDataChannel sample
   11. Building an Internet-Connected Phone with PeerJS

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• Specifications
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Baseline Widely available

This feature is well established and works across many devices and browser versions. It’s been available across browsers since March 2020.

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The jitter property of the RTCInboundRtpStreamStats dictionary indicates the packet interarrival jitter for this synchronization source (SSRC), in seconds.

The packet jitter is calculated as defined in RFC 3550, section 6.4.1.

A positive number, in seconds.

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This page was last modified on Aug 15, 2025 by MDN contributors.

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