Getting Started with WebRTC [note]

Getting Started with WebRTC

原文

RTCPeerConnection

1.caller和callee互相发现彼此

2.并且交换capabilities信息

3.初始化session

4.开始实时交换数据

名词解释:

信令:在客户端之间传递控制信息,通过控制信息处理客户端之间的发现、连接建立、连接维护和连接关闭等任务的机制。

function initialize() {
console.log("Initializing; room=99688636.");
card = document.getElementById("card");
localVideo = document.getElementById("localVideo");
miniVideo = document.getElementById("miniVideo");
remoteVideo = document.getElementById("remoteVideo");
resetStatus();
openChannel('AHRlWrqvgCpvbd9B-Gl5vZ2F1BlpwFv0xBUwRgLF/* ...*/');/*room token 由Google App Engine app 提供*/
doGetUserMedia();//确认浏览器是否支持getUserMedia API 如果支持则调用onUserMediaSuccess
}
/* 建立通道过程
1.客户端A生成一个唯一的ID
2.客户端A把ID传给App Engine app,请求获得Channel token
3.App Engine app 把ID传给 Channel API,请求获得一个channel和token
4.App把token传给客户端A
5.客户端A打开socket,监听channel
*/
function openChannel(channelToken) {
console.log("Opening channel.");
var channel = new goog.appengine.Channel(channelToken);
var handler = {
'onopen': onChannelOpened,
'onmessage': onChannelMessage,
'onerror': onChannelError,
'onclose': onChannelClosed
};
socket = channel.open(handler);
}
/*Sending a message works like this: 1.Client B makes a POST request to the App Engine app with an update.
2.The App Engine app passes a request to the channel.
3.The channel carries a message to Client A.
4.Client A's onmessage callback is called.
*/
//如果浏览器支持getUserMedia,则函数被调用
function onUserMediaSuccess(stream) {
console.log("User has granted access to local media.");
// Call the polyfill wrapper to attach the media stream to this element.
attachMediaStream(localVideo, stream);//localVideo.src = ... localViedo代表一个标签
localVideo.style.opacity = 1;
localStream = stream;
// Caller creates PeerConnection.
if (initiator) maybeStart();//initiator 已经被设置为1,直到caller的session终止 所以这里会调用maybeStart
}
//connection只会被建立一次
//建立前提1.第一次建立 2.localStream已经准备好了,即本地视频 3.信令通道准备好了
function maybeStart() {
if (!started && localStream && channelReady) {
// ...调用func,使用STUN创建RTCPeerConnection(pc),设置各种事件监听函数
createPeerConnection();
// ...
pc.addStream(localStream);
started = true;
// Caller initiates offer to peer.
if (initiator)
doCall();
}
}
//被maybeStart调用
//主要目的是使用STUN服务器和回调函数onIceCandidata来建立connection
//为每一个RTCPeerConnection事件建立handlers
function createPeerConnection() {
var pc_config = {"iceServers": [{"url": "stun:stun.l.google.com:19302"}]};
try {
// Create an RTCPeerConnection via the polyfill (adapter.js).
pc = new RTCPeerConnection(pc_config);//在adapter.js中被包装过了
pc.onicecandidate = onIceCandidate;
console.log("Created RTCPeerConnnection with config:\n" + " \"" +
JSON.stringify(pc_config) + "\".");
} catch (e) {
console.log("Failed to create PeerConnection, exception: " + e.message);
alert("Cannot create RTCPeerConnection object; WebRTC is not supported by this browser.");
return;
} pc.onconnecting = onSessionConnecting;//log status messages作用
pc.onopen = onSessionOpened; //log status messages作用
pc.onaddstream = onRemoteStreamAdded; //log status messages作用
pc.onremovestream = onRemoteStreamRemoved;//为remoteVideo标签设置内容
}
//handler
function onRemoteStreamAdded(event) {
// ...
miniVideo.src = localVideo.src;
attachMediaStream(remoteVideo, event.stream);
remoteStream = event.stream;
waitForRemoteVideo();
}
//在maybeStart()调用createPeerConnection()之后, a call is intitiated by creating and offer and sending it to the callee
function doCall() {
console.log("Sending offer to peer.");
pc.createOffer(setLocalAndSendMessage, null, mediaConstraints);
}
//创建offer的过程和非信令的例子(caller callee都在一个浏览器内)类似。
//不同点:message被发送到远端(remote peer),giving a serialized SessionDescription
//不同点的功能有setLocalAndMessage()完成
//客户端配置信息叫做Session Description
function setLocalAndSendMessage(sessionDescription) {
// Set Opus as the preferred codec in SDP if Opus is present.
sessionDescription.sdp = preferOpus(sessionDescription.sdp);
pc.setLocalDescription(sessionDescription);
sendMessage(sessionDescription);
}
/*signaling with the Channel API*/
/*当createPeerConnection()成功创建RTCPeerConnetion后 回调函数onIceCandidate被调用:
发送收集来的candidates的信息
*/
function onIceCandidate(event) {
if (event.candidate) {
//使用XHR请求,客户端向服务器发送出站信息
sendMessage({type: 'candidate',
label: event.candidate.sdpMLineIndex,
id: event.candidate.sdpMid,
candidate: event.candidate.candidate});
} else {
console.log("End of candidates.");
}
}
//使用XHR请求,从客户端向服务器发送出站消息(Outbound messaging)
function sendMessage(message) {
var msgString = JSON.stringify(message);
console.log('C->S: ' + msgString);
path = '/message?r=99688636' + '&u=92246248';
var xhr = new XMLHttpRequest();
xhr.open('POST', path, true);
xhr.send(msgString);
}
/*
客户端->服务器发送信令消息:使用XHR
服务器->客户端发送信令消息:使用Google App Engine Channel API
*/
//处理由App Engine server发送来的消息
function processSignalingMessage(message) {
var msg = JSON.parse(message); if (msg.type === 'offer') {
// Callee creates PeerConnection
if (!initiator && !started)//initiator代表session是否创建 RTCPeerConnection是否被创建
maybeStart(); pc.setRemoteDescription(new RTCSessionDescription(msg));
doAnswer();
} else if (msg.type === 'answer' && started) {
pc.setRemoteDescription(new RTCSessionDescription(msg));
} else if (msg.type === 'candidate' && started) {
var candidate = new RTCIceCandidate({sdpMLineIndex:msg.label,
candidate:msg.candidate});
pc.addIceCandidate(candidate);//??
} else if (msg.type === 'bye' && started) {
onRemoteHangup();
}
}
function doAnswer() {
console.log("Sending answer to peer.");
pc.createAnswer(setLocalAndSendMessage, null, mediaConstraints);
}
//在哪里设置msg.type?
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