Android IOS WebRTC 音视频开发总结（七）-- 基于浏览器的开发

前面写的一系列总结都是讲webrtc如何下载,编译,开发的,有些人可能有点云里雾里了,WEBRTC不是用来搞跨浏览器开发的吗,怎么我讲的这些跟浏览器扯不上任何关系,其实看看下面这个架构图,你就明白了，转载请说明出处（博客园RTC.Blacker）

我前面讲的这些内容都封装在browser里面了,如音视频的采集,编码,传输,回声消除,丢包重传.所以如果你想将这些功能集成到你的产品里面就必须理解这些东西.

如果你只想做基于浏览器的视频通话功能,上面这些你可以不理解,更不需要去下载编译WEBRTC代码,因为实现这些功能所需要的JS接口浏览器已经帮你实现了,你只需要简单调用即可,我们先看看实现下面这样一个功能主要涉及哪些步骤?

1,信令交互:开始视频通话前发起端和接收端需要一些交互,如通知对方开始视频,接收视频,视频参数协商(SDP信息),NAT地址交换,这个过程我们称之为信令交互,WEBRTC没有定义标准信令格式,既可以使用SIP也可以使用XMPP,还可以使用自定义的信令格式,最简单的方式就是使用websocket或XMLHttpRequest,自定义格式完成信令交互过程.

2,获取本地视频流:navigator.getUserMedia(constraints, successCallback, errorCallback);

navigator.getUserMedia = navigator.getUserMedia ||
  navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
// Callback to be called in case of success...
function successCallback(gotStream) {
video.src = window.URL.createObjectURL(stream);
  // Start playing video
  video.play();
}

// Callback to be called in case of failure...
function errorCallback(error){ console.log("navigator.getUserMedia error: ", error);
}

// Constraints object for low resolution video
var qvgaConstraints = { video: {
    mandatory: {
      maxWidth: 320,
      maxHeight: 240
} }
};

// Constraints object for standard resolution video
var vgaConstraints = { video: {
    mandatory: {
      maxWidth: 640,
      maxHeight: 480
} }
};

// Constraints object for high resolution video
var hdConstraints = { video: {
    mandatory: {
      minWidth: 1280,
      minHeight: 960
} }
};

function getMedia(constraints){
if (!!stream) { video.src = null; stream.stop();
}
  navigator.getUserMedia(constraints, successCallback, errorCallback);
}

3,SDP协商:createOffer,createAnswer.

localPeerConnection.createOffer(gotLocalDescription, onSignalingError);

// Handler to be called when the 'local' SDP becomes available
function gotLocalDescription(description){
// Add the local description to the local PeerConnection 
　　localPeerConnection.setLocalDescription(description); 
　　log("Offer from localPeerConnection: \n" + description.sdp);
// ...do the same with the 'pseudoremote' PeerConnection
// Note: this is the part that will have to be changed if you want // the communicating peers to become remote
// (which calls for the setup of a proper signaling channel) remotePeerConnection.setRemoteDescription(description);
  // Create the Answer to the received Offer based on the 'local' description
  remotePeerConnection.createAnswer(gotRemoteDescription, onSignalingError);
}
// Handler to be called when the remote SDP becomes available
function gotRemoteDescription(description){
// Set the remote description as the local description of the
// remote PeerConnection. 
　　remotePeerConnection.setLocalDescription(description);
log("Answer from remotePeerConnection: \n" + description.sdp);
// Conversely, set the remote description as the remote description of the // local PeerConnection 
　　localPeerConnection.setRemoteDescription(description);
}

4,ICE协商:

// Add a handler associated with ICE protocol events 
localPeerConnection.onicecandidate = gotLocalIceCandidate;

 4 remotePeerConnection.onicecandidate = gotRemoteIceCandidate;

// Handler to be called whenever a new local ICE candidate becomes available
function gotLocalIceCandidate(event){ if (event.candidate) {
            // Add candidate to the remote PeerConnection
　　　　　　 remotePeerConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
       　　 log("Local ICE candidate: \n" + event.candidate.candidate);
      }
}

// Handler to be called whenever a new remote ICE candidate becomes available
function gotRemoteIceCandidate(event){ if (event.candidate) {
            // Add candidate to the local PeerConnection
　　　　　　localPeerConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
          log("Remote ICE candidate: \n " + event.candidate.candidate);
      }
}

 

5,使用RTCPeerConnection对象在浏览器之间交换媒体流数据.

function call() {
  log("Starting call");

  // Note well: getVideoTracks() and getAudioTracks() are not currently supported in Firefox...
  // ...just use them with Chrome
  if (navigator.webkitGetUserMedia) {
      // Log info about video and audio device in use
      if (localStream.getVideoTracks().length > 0) {
        log('Using video device: ' + localStream.getVideoTracks()[0].label);
      }
      if (localStream.getAudioTracks().length > 0) {
        log('Using audio device: ' + localStream.getAudioTracks()[0].label);
      }
  }
  
  // Chrome
  if (navigator.webkitGetUserMedia) {
      RTCPeerConnection = webkitRTCPeerConnection;
  // Firefox
  }else if(navigator.mozGetUserMedia){
      RTCPeerConnection = mozRTCPeerConnection;
      RTCSessionDescription = mozRTCSessionDescription;
      RTCIceCandidate = mozRTCIceCandidate;
  }
  log("RTCPeerConnection object: " + RTCPeerConnection);
  
  // This is an optional configuration string, associated with NAT traversal setup
  var servers = null;

  // Create the local PeerConnection object
  localPeerConnection = new RTCPeerConnection(servers);
  log("Created local peer connection object localPeerConnection");
  // Add a handler associated with ICE protocol events
  localPeerConnection.onicecandidate = gotLocalIceCandidate;

  // Create the remote PeerConnection object
  remotePeerConnection = new RTCPeerConnection(servers);
  log("Created remote peer connection object remotePeerConnection");
  // Add a handler associated with ICE protocol events...
  remotePeerConnection.onicecandidate = gotRemoteIceCandidate;
  // ...and a second handler to be activated as soon as the remote stream becomes available 
  remotePeerConnection.onaddstream = gotRemoteStream;

  // Add the local stream (as returned by getUserMedia() to the local PeerConnection
  localPeerConnection.addStream(localStream);
  log("Added localStream to localPeerConnection");
  
  // We're all set! Create an Offer to be 'sent' to the callee as soon as the local SDP is ready
  localPeerConnection.createOffer(gotLocalDescription, onSignalingError);
}

function onSignalingError(error) {
    console.log('Failed to create signaling message : ' + error.name);
}

// Handler to be called when the 'local' SDP becomes available
function gotLocalDescription(description){
  // Add the local description to the local PeerConnection
  localPeerConnection.setLocalDescription(description);
  log("Offer from localPeerConnection: \n" + description.sdp);
  
  // ...do the same with the 'pseudo-remote' PeerConnection
  // Note well: this is the part that will have to be changed if you want the communicating peers to become
  // remote (which calls for the setup of a proper signaling channel)
  remotePeerConnection.setRemoteDescription(description);
  
  // Create the Answer to the received Offer based on the 'local' description
  remotePeerConnection.createAnswer(gotRemoteDescription, onSignalingError);
}

// Handler to be called when the 'remote' SDP becomes available
function gotRemoteDescription(description){
  // Set the 'remote' description as the local description of the remote PeerConnection
  remotePeerConnection.setLocalDescription(description);
  log("Answer from remotePeerConnection: \n" + description.sdp);
  // Conversely, set the 'remote' description as the remote description of the local PeerConnection 
  localPeerConnection.setRemoteDescription(description);
}

// Handler to be called as soon as the remote stream becomes available
function gotRemoteStream(event){    
  // Associate the remote video element with the retrieved stream
  if (window.URL) {
      // Chrome
      remoteVideo.src = window.URL.createObjectURL(event.stream);
  } else {
      // Firefox
      remoteVideo.src = event.stream;
  }  
  log("Received remote stream");
}

// Handler to be called whenever a new local ICE candidate becomes available
function gotLocalIceCandidate(event){
  if (event.candidate) {
    // Add candidate to the remote PeerConnection 
    remotePeerConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
    log("Local ICE candidate: \n" + event.candidate.candidate);
  }
}

// Handler to be called whenever a new 'remote' ICE candidate becomes available
function gotRemoteIceCandidate(event){
  if (event.candidate) {
    // Add candidate to the local PeerConnection      
    localPeerConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
    log("Remote ICE candidate: \n " + event.candidate.candidate);
  }

上面基本上就是浏览器上视频通话涉及的主要对象.

对应到手机端就是webrtc编译成功后的appRTCDemo.apk.