RTP sources

The contributing and synchronization sources of the media a receiver received.

class webrtc.models.rtp_source.RTCRtpContributingSource(timestamp, source, rtp_timestamp, audio_level=None)

A contributing source (CSRC) of the media an webrtc.RTCRtpReceiver received in the last 10 seconds.

Also the base of webrtc.RTCRtpSynchronizationSource.

See MDN: RTCRtpReceiver/getContributingSources.

Parameters:

• timestamp (float) – When the latest packet from the source arrived, in milliseconds since the Unix epoch, on the same clock as webrtc.RTCStats timestamps.

• source (int) – The CSRC or SSRC identifier of the source.

• rtp_timestamp (int) – The RTP timestamp of the latest packet from the source.

• audio_level (float, optional) – The audio level of the latest packet, linear from 0 (silence) to 1 (0 dBov). It’s converted from the level header extension, and is None if the packet had none.

class webrtc.models.rtp_source.RTCRtpSynchronizationSource(timestamp, source, rtp_timestamp, audio_level=None)

A synchronization source (SSRC) of the media an webrtc.RTCRtpReceiver received in the last 10 s.

It has the members of webrtc.RTCRtpContributingSource.

See MDN: RTCRtpReceiver/getSynchronizationSources.