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RTCRemoteInboundRtpStreamStats: jitter property

Limited availability

This feature is not Baseline because it does not work in some of the most widely-used browsers.

The jitter property of the RTCRemoteInboundRtpStreamStats dictionary returns the packet jitter for the synchronization source (SSRC) as measured by the remote endpoint.

High packet jitter values indicate that packet arrival rates vary significantly, which may degrade video, audio, and other real-time communications over WebRTC.

Value

Packet jitter, in seconds.

The value is calculated using the "interarrival jitter" algorithm described in RFC 3550, section 6.4.1.

Specifications

Specification
Identifiers for WebRTC's Statistics API>
# dom-rtcreceivedrtpstreamstats-jitter>

Browser compatibility

Desktop Mobile
Chrome Edge Firefox Opera Safari Chrome Android Firefox for Android Opera Android Safari on IOS Samsung Internet WebView Android WebView on iOS
jitter 80 80 72 67 No 80 79 57 No 13.0 80 No

See also

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https://developer.mozilla.org/en-US/docs/Web/API/RTCRemoteInboundRtpStreamStats/jitter