This feature is not Baseline because it does not work in some of the most widely-used browsers.
The jitterBufferMinimumDelay property of the RTCInboundRtpStreamStats dictionary indicates the minimum jitter buffer delay that might be achieved given only the network characteristics such as jitter and packet loss.
The jitter buffer delay may be impacted by user settings such as RTCRtpReceiver.jitterBufferTarget, and WebRTC mechanisms such as AV synchronization. jitterBufferMinimumDelay can be compared to the jitterBufferTargetDelay to examine the effect of these external factors on the delay.
The property is updated when jitterBufferEmittedCount is updated.
A positive number, in seconds.
| Specification |
|---|
| Identifiers for WebRTC's Statistics API> # dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay> |
| Desktop | Mobile | |||||||||||
|---|---|---|---|---|---|---|---|---|---|---|---|---|
| Chrome | Edge | Firefox | Opera | Safari | Chrome Android | Firefox for Android | Opera Android | Safari on IOS | Samsung Internet | WebView Android | WebView on iOS | |
jitterBufferMinimumDelay |
105 | 105 | 142 | 91 | No | 105 | 142 | 72 | No | 20.0 | 105 | No |
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https://developer.mozilla.org/en-US/docs/Web/API/RTCInboundRtpStreamStats/jitterBufferMinimumDelay