The getSynchronizationSources() method of the RTCRtpReceiver interface returns an array of RTCRtpContributingSource instances, each corresponding to one SSRC (synchronization source) identifier received by the current RTCRtpReceiver in the last ten seconds.
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getSynchronizationSources()
None.
An array of object describing one of the synchronization sources that provided data to the incoming stream in the past ten seconds. These objects contain the following properties:
audioLevel Optional
A double-precision floating-point value between 0.0 and 1.0 specifying the audio level (in dBov) contained in the last RTP packet played from this source.
rtpTimestamp Optional
The RTP timestamp of the media played out at the time indicated by timestamp. This value is a source-generated time value which can be used to help with sequencing and synchronization.
source Optional
A 32-bit unsigned integer value specifying the CSRC identifier of the contributing source.
timestamp Optional
A DOMHighResTimeStamp indicating the most recent time at which a frame originating from this source was delivered to the receiver's MediaStreamTrack.
| Specification |
|---|
| WebRTC: Real-Time Communication in Browsers # dom-rtcrtpreceiver-getsynchronizationsources |
| Desktop | Mobile | |||||||||||
|---|---|---|---|---|---|---|---|---|---|---|---|---|
| Chrome | Edge | Firefox | Internet Explorer | Opera | Safari | WebView Android | Chrome Android | Firefox for Android | Opera Android | Safari on IOS | Samsung Internet | |
getSynchronizationSources |
73 | 79 | 59 | No | 60 | 12.1 | 73 | 73 | 59 | 52 | 12.2 | 11.0 |
audio_tracks_supported |
73 | 79 | 59 | No | 60 | 12.1 | 73 | 73 | 59 | 52 | 12.2 | 11.0 |
video_tracks_supported |
73 | 79 | 68 | No | 60 | 12.1 | 73 | 73 | 68 | 52 | 12.2 | 11.0 |
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https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpReceiver/getSynchronizationSources