How to increase the bitrate of webrtc?

Question

I'm using webrtc to send a 1080p video stream from one tab to another tab on the same computer(windows10, chrome 76). And the receiver's video quality is not as good as sender's. The bitrate is only about 2400kbps(300kb/s), no difference between 1080p and 720p. Video resolution also become lower when the camera moving.
How can I improve the quality of webrtc video stream?

I have tried to modify sdp to increase bitrate. http://www.rtcbits.com/2016/11/controlling-bandwidth-usage-in-webrtc.html

set x-google-max-bitrate

peer.createAnswer().then(sdp => {
  var arr = sdp.sdp.split('\r\n');
  arr.forEach((str, i) => {
    if (/^a=fmtp:\d*/.test(str)) {
      arr[i] = str + ';x-google-max-bitrate=28000;x-google-min-bitrate=0;x-google-start-bitrate=20000';
    }
  });
  sdp = new RTCSessionDescription({
    type: 'answer',
    sdp: arr.join('\r\n'),
  })
  peer.setLocalDescription(sdp);
  socket.emit('message_send', { type: 'answer', sdp: sdp.sdp });
});

output receive rate (kb/s)

var prevReport = null;
var t = setInterval(function() {
  if (!peer) {
    prevReport = null;
    return;
  }
  peer.getStats(null).then(reporter => {
    reporter.forEach(report => {
      if (report.type === 'inbound-rtp' && report.mediaType === 'video') {
        if (!prevReport) {
          prevReport = report;
        } else {
          console.log((report.bytesReceived - prevReport.bytesReceived) / (report.timestamp - prevReport.timestamp));
        }
      }
    });
  });
}, 1000);

I hope that the bitrate of 1080p could be obviously greater than of 720p.
Is there a way to let webrtc transport lossless or low-loss video stream?

---

The 300kb/s limit only exists when a chrome tab sends video to another chrome tab. When a chrome tab sends video to a firefox tab, the x-google-max-bitrate works.

Answer 1

I tried to set b=AS:10000 and it works.

peer.createAnswer().then(sdp => {
  var arr = sdp.sdp.split('\r\n');
  arr.forEach((str, i) => {
    if (/^a=fmtp:\d*/.test(str)) {
      arr[i] = str + ';x-google-max-bitrate=10000;x-google-min-bitrate=0;x-google-start-bitrate=6000';
    } else if (/^a=mid:(1|video)/.test(str)) {
      arr[i] += '\r\nb=AS:10000';
    }
  });
  sdp = new RTCSessionDescription({
    type: 'answer',
    sdp: arr.join('\r\n'),
  })
  peer.setLocalDescription(sdp);
  socket.emit('message_send', { type: 'answer', sdp: sdp.sdp });
});

Answer 2

Correct way for adjusting audio and video bandwidth limitation is shown in the official sample https://webrtc.github.io/samples/src/content/peerconnection/bandwidth

Here is simplified example for setting video bandwidth limitation 1 Mbps with maxBitrate property of RTCRtpEncodingParameters:

const pc1 = new RTCPeerConnection(servers);

...

const sender = pc1.getSenders()[0];
const parameters = sender.getParameters();
parameters.encodings[0].maxBitrate = 1 * 1000 * 100;
sender.setParameters(parameters);

For adjusting audio bandwidth limitation corresponding sender should be updated, e.g.

const sender = pc1.getSenders()[1];