通过 Websockets 进行 WebRTC 视频聊天

我正在尝试使用 webRTC 和 WebSockets 进行信号发送来开发视频聊天应用程序。 我的问题是，我不知道创建 RTCPeerConnection 并通过 webSocket 连接两个对等点（2 个浏览器）的过程是什么（至少在本地）。

我知道如何通过 WebSockets 与客户端进行通信，但不知道如何使用 RTCPeerConnection API，您知道如何逐步解释该过程的教程吗？（提供 SDP、答案、ICE，...），另一方面，看起来如何通过 RTCPeerConnection 管理这些客户端的服务器代码？

这是我的服务器 Node.js 代码

"use strict";

// Optional. You will see this name in eg. 'ps' or 'top' command
process.title = 'node-webrtc';

// Port where we'll run the websocket server
var webSocketsServerPort = 1337;

// websocket and http servers
var webSocketServer = require('websocket').server;
var http = require('http');

 /* ---------------------------------

            GLOBAL VARIABLES

  ----------------------------------*/

// latest 100 messages
//var history = [ ];

// list of currently connected clients (users)
var clients = [ ];

 /* ---------------------------------

            HTTP SERVER

  ----------------------------------*/

var server = http.createServer(function(request, response) {
    // Not important for us. We're writing WebSocket server, not HTTP server
});
server.listen(webSocketsServerPort, function() {
    console.log((new Date()) + " Server is listening on port " + webSocketsServerPort);
});

 /* ---------------------------------

            WEBSOCKET SERVER

  ----------------------------------*/

var wsServer = new webSocketServer({
    // WebSocket server is tied to a HTTP server. WebSocket request is just
    // an enhanced HTTP request. For more info http://tools.ietf.org/html/rfc6455#page-6
    httpServer: server
});

// This callback function is called every time someone
// tries to connect to the WebSocket server
wsServer.on('request', function(request) {
    console.log((new Date()) + ' Connection from origin ' + request.origin + '.');

    // accept connection - you should check 'request.origin' to make sure that
    // client is connecting from your website
    // (http://en.wikipedia.org/wiki/Same_origin_policy)
    var connection = request.accept(null, request.origin); 
    // we need to know client index to remove them on 'close' event
    var index = clients.push(connection) - 1;

    console.log((new Date()) + ' Connection accepted.');


    // user sent some message
    connection.on('message', function(message) {
        for (var i=0; i < clients.length; i++) {
            clients[i].sendUTF(message);
        }   
    });


    // user disconnected
    connection.on('close', function(conn) {  
        console.log((new Date()) + " Peer " + conn.remoteAddress + " disconnected.");
        // remove user from the list of connected clients
        clients.splice(index, 1);
    });

});

您可能想看看我为 Google I/O 所做的 Codelab：bitbucket.org/webrtc/codelab https://bitbucket.org/webrtc/codelab.

Step 5 https://bitbucket.org/webrtc/codelab/src/6daf0496dc78e1704cb9b71b9e897e829a347561/complete/step5?at=master展示如何使用 socket.io 设置信令服务器，以及Step 6 https://bitbucket.org/webrtc/codelab/src/6daf0496dc78e1704cb9b71b9e897e829a347561/complete/step6?at=master将其与 RTCPeerConnection 结合起来制作一个简单的视频聊天应用程序。

您可能还想看看easyRTC https://github.com/priologic/easyrtc（全栈）和信号大师 https://github.com/andyet/signalmaster（为以下目的创建的信令服务器简单WebRTC https://github.com/HenrikJoreteg/SimpleWebRTC).

“规范”WebRTC 视频聊天示例位于apprtc.appspot.com http://apprtc.appspot.com使用 XHR 和 Google App Engine Channel API 进行信号发送。