# 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`](../interfaces/rtc_rtp_receiver.md#webrtc.interfaces.rtc_rtp_receiver.RTCRtpReceiver) received in the last 10 seconds.

Also the base of [`webrtc.RTCRtpSynchronizationSource`](#webrtc.models.rtp_source.RTCRtpSynchronizationSource).

See [MDN: RTCRtpReceiver/getContributingSources](https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpReceiver/getContributingSources).

* **Parameters:**
  * **timestamp** ([`float`](https://docs.python.org/3/builtins/functions.html#float)) – When the latest packet from the source arrived, in milliseconds since
    the Unix epoch, on the same clock as [`webrtc.RTCStats`](rtc_stats.md#webrtc.models.rtc_stats.RTCStats) timestamps.
  * **source** ([`int`](https://docs.python.org/3/builtins/functions.html#int)) – The CSRC or SSRC identifier of the source.
  * **rtp_timestamp** ([`int`](https://docs.python.org/3/builtins/functions.html#int)) – The RTP timestamp of the latest packet from the source.
  * **audio_level** ([`float`](https://docs.python.org/3/builtins/functions.html#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`](https://docs.python.org/3/builtins/constants.html#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`](../interfaces/rtc_rtp_receiver.md#webrtc.interfaces.rtc_rtp_receiver.RTCRtpReceiver) received in the last 10 s.

It has the members of [`webrtc.RTCRtpContributingSource`](#webrtc.models.rtp_source.RTCRtpContributingSource).

See [MDN: RTCRtpReceiver/getSynchronizationSources](https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpReceiver/getSynchronizationSources).