Node.js 浏览器中未显示Nodewebrtc视频流

我的目标是使用gstreamer管道和WebRTC在浏览器中显示视频。 我发现它使用了node.js服务器。 我首先修改了其中一个示例（视频合成），以显示来自videotestsrc的gstreamer管道的视频。 以下是（在我看来）这方面的两个重要文件：

server.js

'use strict';

const gstreamer = require("gstreamer-superficial");
const { RTCVideoSource } = require('wrtc').nonstandard;


  const pipeline = new gstreamer.Pipeline("videotestsrc is-live=True ! videoconvert ! videoscale ! clockoverlay shaded-background=true ! video/x-raw,format=I420,width=1920,height=1080,framerate=25/1 ! appsink name=sink");
  
  const appsink = pipeline.findChild('sink');
  const source = new RTCVideoSource();
  const track = source.createTrack();
  peerConnection.addTransceiver(track);

  const frame = {
    width: 1920,
    height: 1080,
    data: null
  };
  var pull = function() {
    appsink.pull(function(buf, caps) {
      if (caps) {
        console.log("CAPS", caps);
      }
      if (buf) {
        // console.log("BUFFER size",buf.length);
        frame.data = new Uint8Array(buf);
        try {
          source.onFrame(frame);
          console.log("ON FRAME");
          console.log(source);
        } catch (e) {console.log(e)}
        pull();
      } else {
        setTimeout(pull, 500);
      }
    });
  };

  pipeline.play();

  pull();


  // NOTE(mroberts): This is a hack so that we can get a callback when the
  // RTCPeerConnection is closed. In the future, we can subscribe to
  // "connectionstatechange" events.
  const { close } = peerConnection;
  peerConnection.close = function() {
    track.stop();
    return close.apply(this, arguments);
  };
}

module.exports = { beforeOffer };

client.js

'use strict';

const createExample = require('../../lib/browser/example');

const description = 'This example uses node-webrtc’s RTCVideoSource and \
 gstreamer-superficial to stream a video though a GStreamer pipeline.';

const remoteVideo = document.createElement('video');
remoteVideo.autoplay = true;
// remoteVideo.width = 1920;
// remoteVideo.height = 1080;

async function beforeAnswer(peerConnection) {

  const remoteStream = new MediaStream(peerConnection.getReceivers().map(receiver => receiver.track));
  console.log(remoteStream);
  remoteVideo.srcObject = remoteStream;
  console.warn(remoteVideo.srcObject);

  // NOTE(mroberts): This is a hack so that we can get a callback when the
  // RTCPeerConnection is closed. In the future, we can subscribe to
  // "connectionstatechange" events.
  const { close } = peerConnection;
  peerConnection.close = function() {
    remoteVideo.srcObject = null;
    return close.apply(this, arguments);
  };
}

createExample('video-compositing', description, { beforeAnswer });

/* const videos = document.createElement('div');
videos.className = 'grid';
videos.appendChild(remoteVideo);
document.body.appendChild(videos); */

document.body.appendChild(remoteVideo);

在服务器的控制台中，我得到了一个看起来相当不错的输出，但是在浏览器中什么都不会显示。 我不知道管道是否有问题，或者我哪里做错了什么

我用了

chrome://webrtc-internals/

在webRTC上检查一点，但这对我也没有帮助。但我仍然包括了在那里给我看的东西。也许“事件”丢失了

如果我仔细看一下收信机，莫尔补充道。它向我展示了以下内容。发送方跟踪和流下是否为空可能是一个问题

有人能帮我吗？提前谢谢