WebRTC学习笔记一简单示例

　　一、捕获本地媒体流getUserMedia1.index.html                                      开始录制         停止录制                               
　　部署本地服务器后，chrome中访问 192.168.11.129:9050  会报错: navigator.getUserMedia is not a function  。原因是，Chrome 47以后，getUserMedia API只能允许来自＂安全可信＂的客户端的视频音频请求，如HTTPS和本地的Localhost。所以将访问地址改成localhost:9050即可。 二、同网页示例
　　例子来源：https://codelabs.developers.google.com/codelabs/webrtc-web/#4 1.index.html                                                     
Realtime communication with WebRTC
                                                        开始              调用              挂断                                          2.main.js
　　【腾讯文档】FFmpegWebRTCRTMPRTSPHLSRTP播放器-音视频流媒体高级开发-资料领取FFmpegWebRTCRTMPRTSPHLSRTP鎾斁鍣�-闊宠棰戞祦濯掍綋楂樼骇寮€鍙�-璧勬枡棰嗗彇＂use strict＂;  //log function trace(text) {   text = text.trim();   const now = (window.performance.now() / 1000).toFixed(3);   console.log(now, text); }  // 设置两个video，分别显示本地视频流和远端视频流 const localVideo = document.getElementById(＂localVideo＂); const remoteVideo = document.getElementById(＂remoteVideo＂);  localVideo.addEventListener(＂loadedmetadata＂, logVideoLoaded); remoteVideo.addEventListener(＂loadedmetadata＂, logVideoLoaded); remoteVideo.addEventListener(＂onresize＂, logResizedVideo);  function logVideoLoaded(event) {   const video = event.target;   trace(`${video.id} videoWidth: ${video.videoWidth}px, ` +         `videoHeight: ${video.videoHeight}px.`); }  function logResizedVideo(event) {   logVideoLoaded(event);   if (startTime) {     const elapsedTime = window.performance.now() - startTime;     startTime = null;     trace(`Setup time: ${elapsedTime.toFixed(3)}ms.`);   } }  let startTime = null; let localStream; let remoteStream; // 建立两个对等连接对象，分表代表本地和远端 let localPeerConnection; let remotePeerConnection;  const startButton = document.getElementById(＂startButton＂); const callButton = document.getElementById(＂callButton＂); const hangupButton = document.getElementById(＂hangupButton＂); callButton.disabled = true; hangupButton.disabled = true; startButton.addEventListener(＂click＂, startAction); callButton.addEventListener(＂click＂, callAction); hangupButton.addEventListener(＂click＂, hangupAction);  // 传输视频，不传输音频 const mediaStreamConstraints = {   video: true,   audio: false }; //开始事件，采集摄像头到本地 function startAction() {   startButton.disabled = true;   navigator.getUserMedia(mediaStreamConstraints, gotLocalMediaStream, handleLocalMediaStreamError)   trace(＂Requesting local stream.＂); } function gotLocalMediaStream(mediaStream) {   localVideo.srcObject = mediaStream;   localStream = mediaStream;   trace(＂Received local stream.＂);   callButton.disabled = false;  } function handleLocalMediaStreamError(error) {   trace(`navigator.getUserMedia error: ${error.toString()}.`); }  // 设置只交换视频 const offerOptions = {   offerToReceiveVideo: 1, };  // 创建对等连接 function callAction() {   callButton.disabled = true;   hangupButton.disabled = false;    trace(＂Starting call.＂);   startTime = window.performance.now();    const videoTracks = localStream.getVideoTracks();   const audioTracks = localStream.getAudioTracks();   if (videoTracks.length > 0) {     trace(`Using video device: ${videoTracks[0].label}.`);   }   if (audioTracks.length > 0) {     trace(`Using audio device: ${audioTracks[0].label}.`);   }    // 服务器配置   const servers = null;    localPeerConnection = new RTCPeerConnection(servers);   trace(＂Created local peer connection object localPeerConnection.＂);   localPeerConnection.addEventListener(＂icecandidate＂, handleConnection);   localPeerConnection.addEventListener(＂iceconnectionstatechange＂, handleConnectionChange);    remotePeerConnection = new RTCPeerConnection(servers);   trace(＂Created remote peer connection object remotePeerConnection.＂);   remotePeerConnection.addEventListener(＂icecandidate＂, handleConnection);   remotePeerConnection.addEventListener(＂iceconnectionstatechange＂, handleConnectionChange);   remotePeerConnection.addEventListener(＂addstream＂, gotRemoteMediaStream);    localPeerConnection.addStream(localStream);   trace(＂Added local stream to localPeerConnection.＂);    trace(＂localPeerConnection createOffer start.＂);   localPeerConnection.createOffer(offerOptions)     .then(createdOffer).catch(setSessionDescriptionError); }  function getOtherPeer(peerConnection) {   return (peerConnection === localPeerConnection) ?       remotePeerConnection : localPeerConnection; }  function getPeerName(peerConnection) {   return (peerConnection === localPeerConnection) ?       ＂localPeerConnection＂ : ＂remotePeerConnection＂; }  function handleConnection(event) {   const peerConnection = event.target;   const iceCandidate = event.candidate;    if (iceCandidate) {     const newIceCandidate = new RTCIceCandidate(iceCandidate);     const otherPeer = getOtherPeer(peerConnection);      otherPeer.addIceCandidate(newIceCandidate)       .then(() => {         handleConnectionSuccess(peerConnection);       }).catch((error) => {         handleConnectionFailure(peerConnection, error);       });      trace(`${getPeerName(peerConnection)} ICE candidate: ` +           `${event.candidate.candidate}.`);   } }  function handleConnectionSuccess(peerConnection) {   trace(`${getPeerName(peerConnection)} addIceCandidate success.`); };  function handleConnectionFailure(peerConnection, error) {   trace(`${getPeerName(peerConnection)} failed to add ICE Candidate: `+         `${error.toString()}.`); }  function handleConnectionChange(event) {   const peerConnection = event.target;   console.log(＂ICE state change event: ＂, event);   trace(`${getPeerName(peerConnection)} ICE state: ` +         `${peerConnection.iceConnectionState}.`); }  function gotRemoteMediaStream(event) {   const mediaStream = event.stream;   remoteVideo.srcObject = mediaStream;   remoteStream = mediaStream;   trace(＂Remote peer connection received remote stream.＂); }  function createdOffer(description) {   trace(`Offer from localPeerConnection: ${description.sdp}`);    trace(＂localPeerConnection setLocalDescription start.＂);   localPeerConnection.setLocalDescription(description)     .then(() => {       setLocalDescriptionSuccess(localPeerConnection);     }).catch(setSessionDescriptionError);    trace(＂remotePeerConnection setRemoteDescription start.＂);   remotePeerConnection.setRemoteDescription(description)     .then(() => {       setRemoteDescriptionSuccess(remotePeerConnection);     }).catch(setSessionDescriptionError);    trace(＂remotePeerConnection createAnswer start.＂);   remotePeerConnection.createAnswer()     .then(createdAnswer)     .catch(setSessionDescriptionError); }  function createdAnswer(description) {   trace(`Answer from remotePeerConnection: ${description.sdp}.`);    trace(＂remotePeerConnection setLocalDescription start.＂);   remotePeerConnection.setLocalDescription(description)     .then(() => {       setLocalDescriptionSuccess(remotePeerConnection);     }).catch(setSessionDescriptionError);    trace(＂localPeerConnection setRemoteDescription start.＂);   localPeerConnection.setRemoteDescription(description)     .then(() => {       setRemoteDescriptionSuccess(localPeerConnection);     }).catch(setSessionDescriptionError); }  function setSessionDescriptionError(error) {   trace(`Failed to create session description: ${error.toString()}.`); }   function setLocalDescriptionSuccess(peerConnection) {   setDescriptionSuccess(peerConnection, ＂setLocalDescription＂); }  function setRemoteDescriptionSuccess(peerConnection) {   setDescriptionSuccess(peerConnection, ＂setRemoteDescription＂); }  function setDescriptionSuccess(peerConnection, functionName) {   const peerName = getPeerName(peerConnection);   trace(`${peerName} ${functionName} complete.`); }  //断掉 function hangupAction() {   localPeerConnection.close();   remotePeerConnection.close();   localPeerConnection = null;   remotePeerConnection = null;   hangupButton.disabled = true;   callButton.disabled = false;   trace(＂Ending call.＂); }3.源码分析
　　点击开始，触发startAction没什么好说的。点击调用，直接看callAction： (1)首先使用 new RTCPeerConnection  创建了两个connection   const servers = null;    localPeerConnection = new RTCPeerConnection(servers);
　　servers在这个例子中并没有用，是用来配置STUN and TURN s服务器的，先忽略。
　　(2)添加事件侦听，先忽略 //也可以使用onicecandidate这种写法 addEventListener(＂icecandidate＂, handleConnection); addEventListener(＂iceconnectionstatechange＂, handleConnectionChange);
　　(3)然后就是addStream和createOffer   localPeerConnection.addStream(localStream);   trace(＂Added local stream to localPeerConnection.＂);    trace(＂localPeerConnection createOffer start.＂);   localPeerConnection.createOffer(offerOptions)     .then(createdOffer).catch(setSessionDescriptionError);
　　其中createOffer需要一个Options // 设置只交换视频 const offerOptions = {   offerToReceiveVideo: 1, };
　　这里我的理解是，createOffer为了产生SDP描述，要先使用addStream把视频流加载进去才能解析。 A创建一个RTCPeerConnection对象。 A使用RTCPeerConnection .createOffer()方法产生一个offer（一个SDP会话描述）。 A用生成的offer调用setLocalDescription()，设置成自己的本地会话描述。 A将offer通过信令机制发送给B。 B用A的offer调用setRemoteDescription()，设置成自己的远端会话描述，以便他的RTCPeerConnection知道A的设置。 B调用createAnswer()生成answer B通过调用setLocalDescription()将其answer设置为本地会话描述。 B然后使用信令机制将他的answer发回给A。 A使用setRemoteDescription()将B的应答设置为远端会话描述。
　　上述过程可以在源码createOffer和createAnser中看到。
　　(4)icecandidate事件 参考https://developer.mozilla.org/zh-CN/docs/Web/API/RTCPeerConnection/icecandidate_event
　　当 RTCPeerConnection通过RTCPeerConnection.setLocalDescription() (en-US)方法更改本地描述之后，该RTCPeerConnection会抛出icecandidate事件。该事件的监听器需要将更改后的描述信息传送给远端RTCPeerConnection，以更新远端的备选源。
　　意思就是setLocalDescription被调用后，触发icecandidate事件，这一点可以在示例的console中得到验证。
　　image.png 4.来张流程图，转自https://segmentfault.com/a/1190000037513346
　　image.png
　　(5)addTrack,addTransceiver addStream() 已过时，官方不推荐使用.将一个MediaStream音频或视频的本地源，添加到WebRTC对等连接流对象中。官方推荐我们使用另外一个方法addTrack   remotePeerConnection.ontrack = function(evt) {       const mediaStream = evt.streams[0];     remoteVideo.srcObject = mediaStream;     remoteStream = mediaStream;     trace(＂Remote peer connection received remote stream.＂); }  localStream.getTracks().forEach(track => {     localPeerConnection.addTrack(track, localStream);     // localPeerConnection.addTransceiver(track, {streams: [localStream]}); // 这个也可以 });
　　如果你是做音视频聊天相关的产品，那么addTrack 刚好能满足你的需求，毕竟需要使用到用户的摄像头、麦克风（浏览器会询问用户是否授权）。但是你只想建立音视频轨道，并不需要使用摄像头、麦克风，那我们应该怎么去做呢？
　　addTransceiver创建一个新的RTCRtpTransceiver并将其添加到与关联的收发器集中RTCPeerConnection。每个收发器都代表一个双向流，并带有RTCRtpSender和RTCRtpReceiver。 let rtcTransceiver = RTCPeerConnection .addTransceiver（trackOrKind，init）;
　　(a)trackOrKind: MediaStreamTrack以与所述收发器相关联，或者一个DOMString被用作kind接收器的的track。这里视频轨道就传＂video＂，音频轨道就传＂audio＂ (b)init: 可选参数。如下： direction：收发器的首选方向性。此值用于初始化新RTCRtpTransceiver对象的*RTCRtpTransceiver.direction属性。 sendEncodings:从中发送RTP媒体时允许的编码列表RTCRtpSender。每个条目都是类型RTCRtpEncodingParameters。 streams: MediaStream要添加到收发器的对象列表RTCRtpReceiver；当远程对等方RTCPeerConnection的track事件发生时，这些是将由该事件指定的流。
　　举个例子: 添加一个单向的音视频流收发器             this.rtcPeerConnection.addTransceiver(＂video＂, {                 direction: ＂recvonly＂             });             this.rtcPeerConnection.addTransceiver(＂audio＂, {                 direction: ＂recvonly＂             });
　　上述代码只会接收对端发过来的音视频流，不会将自己的音视频流传输给对端。direction:
　　image.png 三、网络1V1示例
　　源码参见https://github.com/wuyawei/webrtc-stream
　　这个例子不再是同一个网页，所以需要借助socket.io通讯。 房间相关逻辑暂时忽略，看一下创建offer部分： socket.on(＂apply＂, data => { // 你点同意的地方     ...     this.$confirm(data.self + ＂ 向你请求视频通话, 是否同意?＂, ＂提示＂, {         confirmButtonText: ＂同意＂,         cancelButtonText: ＂拒绝＂,         type: ＂warning＂     }).then(async () => {         await this.createP2P(data); // 同意之后创建自己的 peer 等待对方的 offer         ... // 这里不发 offer     })     ... }); socket.on(＂reply＂, async data =>{ // 对方知道你点了同意的地方     switch (data.type) {         case ＂1＂: // 只有这里发 offer             await this.createP2P(data); // 对方同意之后创建自己的 peer             this.createOffer(data); // 并给对方发送 offer             break;         ...     } });
　　本例采取的是呼叫方发送 Offer，这个地方一定得注意，只要有一方创建 Offer 就可以了，因为一旦连接就是双向的。
　　和微信等视频通话一样，双方都需要进行媒体流输出，因为你们都要看见对方。所以这里和之前本地对等连接的区别就是都需要给自己的 RTCPeerConnection 实例添加媒体流，然后连接后各自都能拿到对方的视频流。在 初始化 RTCPeerConnection 时，记得加上 onicecandidate 函数，用以给对方发送 ICE 候选。 async createP2P(data) {     this.loading = true; // loading动画     this.loadingText = ＂正在建立通话连接＂;     await this.createMedia(data); }, async createMedia(data) {     ... // 获取并将本地流赋值给 video  同之前     this.initPeer(data); // 获取到媒体流后，调用函数初始化 RTCPeerConnection }, initPeer(data) {     // 创建输出端 PeerConnection     ...     this.peer.addStream(this.localstream); // 都需要添加本地流     this.peer.onicecandidate = (event) => {     // 监听ICE候选信息 如果收集到，就发送给对方         if (event.candidate) { // 发送 ICE 候选             socket.emit(＂1v1ICE＂,             {account: data.self, self: this.account, sdp: event.candidate});         }     };     this.peer.onaddstream = (event) => {     // 监听是否有媒体流接入，如果有就赋值给 rtcB 的 src，改变相应loading状态，赋值省略         this.isToPeer = true;         this.loading = false;         ...     }; }
　　createOffer 等信息交换和之前一样，只是需要通过 Socket 转发给对应的客户端。然后各自接收到消息后分别采取对应的措施。 socket.on(＂1v1answer＂, (data) =>{ // 接收到 answer     this.onAnswer(data); }); socket.on(＂1v1ICE＂, (data) =>{ // 接收到 ICE     this.onIce(data); }); socket.on(＂1v1offer＂, (data) =>{ // 接收到 offer     this.onOffer(data); });  async createOffer(data) { // 创建并发送 offer     try {         // 创建offer         let offer = await this.peer.createOffer(this.offerOption);         // 呼叫端设置本地 offer 描述         await this.peer.setLocalDescription(offer);         // 给对方发送 offer         socket.emit(＂1v1offer＂, {account: data.self, self: this.account, sdp: offer});     } catch (e) {         console.log(＂createOffer: ＂, e);     } }, async onOffer(data) { // 接收offer并发送 answer     try {         // 接收端设置远程 offer 描述         await this.peer.setRemoteDescription(data.sdp);         // 接收端创建 answer         let answer = await this.peer.createAnswer();         // 接收端设置本地 answer 描述         await this.peer.setLocalDescription(answer);         // 给对方发送 answer         socket.emit(＂1v1answer＂, {account: data.self, self: this.account, sdp: answer});     } catch (e) {         console.log(＂onOffer: ＂, e);     } }, async onAnswer(data) { // 接收answer     try {         await this.peer.setRemoteDescription(data.sdp); // 呼叫端设置远程 answer 描述     } catch (e) {         console.log(＂onAnswer: ＂, e);     } }, async onIce(data) { // 接收 ICE 候选     try {         await this.peer.addIceCandidate(data.sdp); // 设置远程 ICE     } catch (e) {         console.log(＂onAnswer: ＂, e);     } }
　　挂断的思路依然是将各自的 peer 关闭，但是这里挂断方还需要借助 Socket 告诉对方，你已经挂电话了，不然对方还在痴痴地等。 hangup() { // 挂断通话 并做相应处理 对方收到消息后一样需要关闭连接     socket.emit(＂1v1hangup＂, {account: this.isCall, self: this.account});     this.peer.close();     this.peer = null;     this.isToPeer = false;     this.isCall = false; }