WebRTC 远程视频流不工作

我添加了一个简单的 webRTC 应用程序，它将浏览器窗口连接到自身，从我的相机流式传输视频数据。最终目标是在页面上获取两个视频流，一个直接来自摄像头，另一个来自浏览器本地建立的 WebRTC 连接。

不幸的是，远程视频流没有显示。知道为什么吗？

<video id="yours" autoplay></video>
<video id="theirs" autoplay></video>

这是 JavaScript

function hasUserMedia() {
     navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia ||
        navigator.msGetUserMedia;

     return !!navigator.getUserMedia;
    }

    function hasRTCPeerConnection() {
     window.RTCPeerConnection = window.RTCPeerConnection || window.webkitRTCPeerConnection 
        || window.mozRTCPeerConnection;

     return !!window.RTCPeerConnection;
    }

    var yourVideo = document.querySelector('#yours'),
     theirVideo = document.querySelector('#theirs'),
     yourConnection, theirConnection;


    if (hasUserMedia()) {
        navigator.getUserMedia({ video: true, audio: false }, function(stream) {
         yourVideo.src = window.URL.createObjectURL(stream);
         if (hasRTCPeerConnection()) {
            startPeerConnection(stream);
         } else {
         alert("Sorry, your browser does not support WebRTC.");
         }
         }, function (error) {
         console.log(error);
         });
        }else{
            alert("Sorry, your browser does not support WebRTC.");
        }


    function startPeerConnection(stream){
        var configuration = {
            "iceServers": [{ "url": "stun:stun.1.google.com:19302"
            }]
        };

        yourConnection = new RTCPeerConnection(configuration);
        theirConnection = new RTCPeerConnection(configuration);



         // Setup stream listening
         yourConnection.addStream(stream);

         theirConnection.onaddstream = function (event) {
            theirVideo.src = window.URL.createObjectURL(event.stream);
            console.log('stream added');
         };

         // console.log(yourConnection);
          //console.log(theirConnection);

         // Setup ice handling
         yourConnection.onicecandidate = function (event) {
         if (event.candidate) {
                 theirConnection.addIceCandidate(new RTCIceCandidate(event.
                candidate));
             }
         };
         theirConnection.onicecandidate = function (event) {
             if (event.candidate) {
                 yourConnection.addIceCandidate(new RTCIceCandidate(event.
                candidate));
             }
         };

          // Begin the offer
         yourConnection.createOffer(function (offer) {
            yourConnection.setLocalDescription(offer);
            theirConnection.setRemoteDescription(offer);

            theirConnection.createAnswer(function (offer) {
                theirConnection.setLocalDescription(offer);
                yourConnection.setRemoteDescription(offer);
            });
         });
    };

我正在关注 Dan Ristic 关于 WebRTC 的书，并了解他对编码做了什么。不幸的是，远程视频没有显示。

Add 失败回调使其发挥作用。否则你不仅不会看到错误，而且这样做实际上会让它工作，因为一个非常奇怪的原因：

您是 WebIDL 超载的受害者。 发生的情况是 WebRTC API 有两个版本，而您将它们混合在一起。

有一个现代承诺API, e.g.:

pc.createOffer(options).then(successCallback, failureCallback);

and a 已弃用的回调版本, e.g.:

pc.createOffer(successCallback, failureCallback, options);

换句话说，有两个createOffer采用不同数量参数的函数。

不幸的是，你击中了第一个createOffer因为你只传递一个参数！首先createOffer期望一个options不幸的是，WebIDL 中的对象与函数无法区分。因此，它被视为有效参数（空选项对象）。即使这已经引起了TypeError，它不会导致异常，因为 Promise API 会拒绝返回的 Promise，而不是抛出异常：

pc.createOffer(3).catch(e => console.log("Here: "+ e.name)); // Here: TypeError

您也没有检查返回的承诺，因此错误会丢失。

这是一个工作版本（https 小提琴对于 Chrome）：

navigator.getUserMedia = navigator.getUserMedia ||
                         navigator.webkitGetUserMedia ||
                         navigator.mozGetUserMedia;
window.RTCPeerConnection = window.RTCPeerConnection ||
                           window.webkitRTCPeerConnection;

var yourConnection, theirConnection;

navigator.getUserMedia({ video: true, audio: false }, function(stream) {
    yourVideo.src = window.URL.createObjectURL(stream);

    var config = { "iceServers": [{ "urls": "stun:stun.1.google.com:19302"}] };
    yourConnection = new RTCPeerConnection(config);
    theirConnection = new RTCPeerConnection(config);

    yourConnection.addStream(stream);

    theirConnection.onaddstream = function (event) {
        theirVideo.src = window.URL.createObjectURL(event.stream);
    };

    yourConnection.onicecandidate = function (e) {
        if (e.candidate) {
            theirConnection.addIceCandidate(new RTCIceCandidate(e.candidate),
                                            success, failure);
         }
     };
     theirConnection.onicecandidate = function (e) {
         if (e.candidate) {
             yourConnection.addIceCandidate(new RTCIceCandidate(e.candidate),
                                            success, failure);
         }
     };

     yourConnection.createOffer(function (offer) {
         yourConnection.setLocalDescription(offer, success, failure);
         theirConnection.setRemoteDescription(offer, success, failure);
         theirConnection.createAnswer(function (offer) {
             theirConnection.setLocalDescription(offer, success, failure);
             yourConnection.setRemoteDescription(offer, success, failure);
         }, failure);
     }, failure);
}, failure);

function success() {};
function failure(e) { console.log(e); };

<video id="yourVideo" width="160" height="120" autoplay></video>
<video id="theirVideo" width="160" height="120" autoplay></video>

但回调很费力。我强烈推荐新的 Promise API（https对于 Chrome）：

var pc1 = new RTCPeerConnection(), pc2 = new RTCPeerConnection();

navigator.mediaDevices.getUserMedia({video: true, audio: true})
  .then(stream => pc1.addStream(video1.srcObject = stream))
  .catch(log);

var add = (pc, can) => pc.addIceCandidate(can).catch(log);
pc1.onicecandidate = e => add(pc2, e.candidate);
pc2.onicecandidate = e => add(pc1, e.candidate);

pc2.ontrack = e => video2.srcObject = e.streams[0];
pc1.oniceconnectionstatechange = e => log(pc1.iceConnectionState);
pc1.onnegotiationneeded = e =>
  pc1.createOffer().then(d => pc1.setLocalDescription(d))
  .then(() => pc2.setRemoteDescription(pc1.localDescription))
  .then(() => pc2.createAnswer()).then(d => pc2.setLocalDescription(d))
  .then(() => pc1.setRemoteDescription(pc2.localDescription))
  .catch(log);

var log = msg => console.log(msg);

<video id="video1" height="120" width="160" autoplay muted></video>
<video id="video2" height="120" width="160" autoplay></video><br>
<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>