[R-C] congestion control in WebRTC

Dear all, Recently I am investigating the feedback mechanism for rate control in webrtc. Does the receiver use RTCP to inform the sender about the information such as available bandwidth, lossy ratio, .., etc.? I saw the packet format by wireshark, and saw that there is extra information appended on the SR (Sender Report) of RTCP, but wireshark cannot recognize it. Does anyone have the same experience? Or is there any document describing the details of the feedback mechanism or the format of the feedback packets in webrtc? Many thanks BRs manphis