How do I relate WebRTC's RTCP packets to the respective RTP streams?

Question

How do RTP packets relate to RTCP?

I expected them to share ssrc (synchronization source identifier) with the RTP stream it describes, but it sometimes matches (for Sender Reports) and other times not.

Which RFC or draft should I read for this?

Answer 1

RTCP packets, like RTP, are typically encrypted in WebRTC.

For RTCP Sender Reports one can relate it to RTP steams by SSRC (which is unencrypted). But for Receiver Reports and Generic Feedback the relevant SSRC is in encrypted part, so one should decrypt it first.