Unable to limit WebRTC P2P Multi-participant Receiving Bandwidth

Hope that helps The bandwidth requirements are almost the same as the bandwidth requirement for opus and vp8. Real time audio typically has a bitrate of 40-200kbit/s. Video requires at least 200 kbit/s (500kbit/s if you want to see people's faces).
According to webrtc-experiment the minimum bandwidth for opus is 6kbit/s and for vp8 100kbits/s. So in total that makes 106kbit/s but when you account for the overhead of the webrtc protocol stack and constantly varying network conditions I would guess that 200kbit/s is the minimum if one wants stable video and audio.

it should still fix some issue
What is the maximum of KB/s you can send and receive with WebRTC datachannel.

var pc = new RTCPeerConnection();
var channel = pc.createDataChannel('dc', {
  ordered: true,
  maxPacketLifeTime: 1000,
  maxRetransmits: 5
});

With these it helps Hmm...
@peter-cordes makes a good point about (device) cache. Doing a Google search returns https://www.techspot.com/review/984-intel-ssd-750-series/ which says the following:

With these it helps Remove the second parameters.encodings = [{}]; It accidentally erases all properties of the first encoding in encodings, some of which are meant to be read-only and aren't allowed to be modified.
The pc.setParameters() method requires its input to have come from pc.getParameters() effectively, and only certain encodings properties are allowed to be modified.

const parameters = pc.getParemeters();
// modify certain parameters only
await pc.setParemeters(parameters);

  if (!parameters.encodings) {
    parameters.encodings = [{}];
  }