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📜  WebRTC-语音演示

📅  最后修改于: 2020-10-17 05:50:05             🧑  作者: Mango

在本章中，我们将构建一个客户端应用程序，该客户端应用程序允许两个用户在不同的设备上使用WebRTC音频流进行通信。我们的应用程序将有两页。一个用于登录，另一个用于与另一个用户进行音频通话。

登录和音频呼叫页面。

这两个页面将是div标签。大多数输入是通过简单的事件处理程序完成的。

信令服务器

要创建WebRTC连接，客户端必须能够在不使用WebRTC对等连接的情况下传输消息。在这里我们将使用HTML5 WebSockets-两个端点之间的双向套接字连接-Web服务器和Web浏览器。现在让我们开始使用WebSocket库。创建server.js文件并插入以下代码-

//require our websocket library 
var WebSocketServer = require('ws').Server; 

//creating a websocket server at port 9090 
var wss = new WebSocketServer({port: 9090});
  
//when a user connects to our sever 
wss.on('connection', function(connection) { 
   console.log("user connected"); 
    
   //when server gets a message from a connected user 
   connection.on('message', function(message) { 
      console.log("Got message from a user:", message); 
   }); 
    
   connection.send("Hello from server"); 
});

第一行需要我们已经安装的WebSocket库。然后，我们在端口9090上创建一个套接字服务器。接下来，我们监听连接事件。当用户与服务器建立WebSocket连接时，将执行此代码。然后，我们收听用户发送的任何消息。最后，我们向连接的用户发送响应，说“来自服务器的问候”。

在信令服务器中，我们将为每个连接使用基于字符串的用户名，以便我们知道将消息发送到哪里。让我们稍微改变一下连接处理程序-

connection.on('message', function(message) { 
   var data; 
    
   //accepting only JSON messages 
   try { 
      data = JSON.parse(message); 
   } catch (e) { 
      console.log("Invalid JSON");
      data = {}; 
   } 
});

这样，我们仅接受JSON消息。接下来，我们需要将所有连接的用户存储在某个地方。我们将使用一个简单的Javascript对象。更改文件的顶部-

//require our websocket library 
var WebSocketServer = require('ws').Server; 

//creating a websocket server at port 9090 
var wss = new WebSocketServer({port: 9090}); 

//all connected to the server users 
var users = {};

我们将为来自客户端的每条消息添加一个类型字段。例如，如果用户要登录，则他发送登录类型消息。让我们定义它-

connection.on('message', function(message) {
  
   var data; 
   //accepting only JSON messages 
   try { 
      data = JSON.parse(message); 
   } catch (e) { 
      console.log("Invalid JSON"); 
      data = {}; 
   } 
    
   //switching type of the user message 
   switch (data.type) { 
      //when a user tries to login 
      case "login": 
         console.log("User logged:", data.name); 
            
         //if anyone is logged in with this username then refuse 
         if(users[data.name]) { 
            sendTo(connection, { 
               type: "login",
               success: false 
            }); 
         } else { 
            //save user connection on the server 
            users[data.name] = connection; 
            connection.name = data.name; 
                
            sendTo(connection, { 
               type: "login", 
               success: true 
            }); 
         } 
            
         break;
            
      default: 
         sendTo(connection, { 
            type: "error", 
            message: "Command no found: " + data.type 
         }); 
            
         break; 
   }  
});

如果用户使用登录类型发送消息，我们-

  • 检查是否有人已经使用该用户名登录。
  • 如果是这样，则告诉用户他尚未成功登录。
  • 如果没有人使用此用户名，则将用户名添加为连接对象的键。
  • 如果无法识别命令，我们将发送错误。

以下代码是用于将消息发送到连接的帮助程序函数。将其添加到server.js文件-

function sendTo(connection, message) { 
   connection.send(JSON.stringify(message)); 
}

当用户断开连接时，我们应该清理其连接。当关闭事件被触发时，我们可以删除用户。将以下代码添加到连接处理程序-

connection.on("close", function() { 
   if(connection.name) { 
      delete users[connection.name]; 
   } 
});

成功登录后，用户要呼叫另一个。他应该向其他用户提出要约。添加报价处理程序-

case "offer": 
   //for ex. UserA wants to call UserB 
   console.log("Sending offer to: ", data.name); 
    
   //if UserB exists then send him offer details 
   var conn = users[data.name]; 
    
   if(conn != null) { 
      //setting that UserA connected with UserB 
      connection.otherName = data.name; 
      sendTo(conn, { 
         type: "offer", 
         offer: data.offer, 
         name: connection.name 
      });
   }        
    
   break;

首先，我们获得了要呼叫的用户的连接。如果存在，我们会向他发送报价详细信息。我们还将otherName添加到连接对象。这样做是为了以后找到它的简单性。

对响应进行回答的方式与要约处理程序中使用的方式类似。我们的服务器刚刚通过的所有消息的答案给其他用户。添加报价处理后，将下面的代码-

case "answer": 
   console.log("Sending answer to: ", data.name); 
   //for ex. UserB answers UserA
   var conn = users[data.name]; 
    
   if(conn != null) { 
      connection.otherName = data.name;
        
      sendTo(conn, { 
         type: "answer", 
         answer: data.answer 
      }); 
   } 
    
   break;

最后一部分是处理用户之间的ICE候选者。我们使用相同的技术只是在用户之间传递消息。主要区别在于候选消息可能以任何顺序在每个用户身上多次出现。添加候选处理程序-

case "candidate": 
   console.log("Sending candidate to:",data.name); 
   var conn = users[data.name];
    
   if(conn != null) { 
      sendTo(conn, { 
         type: "candidate", 
         candidate: data.candidate 
      }); 
   } 
    
   break;

为了允许我们的用户与另一个用户断开连接，我们应该实现挂断函数。它还将告诉服务器删除所有用户引用。添加请假处理程序-

case "leave": 
   console.log("Disconnecting from", data.name); 
   var conn = users[data.name]; 
   conn.otherName = null; 
    
   //notify the other user so he can disconnect his peer connection 
   if(conn != null) { 
      sendTo(conn, {
         type: "leave" 
      }); 
   }  
    
   break;

这还将向其他用户发送请假事件，以便他可以相应地断开其对等连接。当用户从信令服务器断开连接时，我们也应该处理这种情况。让我们修改关闭处理程序-

connection.on("close", function() { 

   if(connection.name) { 
      delete users[connection.name]; 
        
      if(connection.otherName) { 
         console.log("Disconnecting from ", connection.otherName); 
         var conn = users[connection.otherName]; 
         conn.otherName = null;
            
         if(conn != null) { 
            sendTo(conn, { 
               type: "leave" 
            }); 
         }
            
      } 
   } 
});

以下是我们的信令服务器的完整代码-

//require our websocket library 
var WebSocketServer = require('ws').Server; 

//creating a websocket server at port 9090 
var wss = new WebSocketServer({port: 9090}); 

//all connected to the server users 
var users = {};

//when a user connects to our sever 
wss.on('connection', function(connection) {
  
   console.log("User connected");
    
   //when server gets a message from a connected user 
   connection.on('message', function(message) { 
    
      var data;
        
      //accepting only JSON messages 
      try { 
         data = JSON.parse(message); 
      } catch (e) { 
         console.log("Invalid JSON"); 
         data = {}; 
      }
        
      //switching type of the user message 
      switch (data.type) { 
         //when a user tries to login 
         case "login": 
            console.log("User logged", data.name); 
                
            //if anyone is logged in with this username then refuse 
            if(users[data.name]) { 
               sendTo(connection, { 
                  type: "login", 
                  success: false 
               }); 
            } else { 
               //save user connection on the server 
               users[data.name] = connection; 
               connection.name = data.name;
                    
               sendTo(connection, { 
                  type: "login", 
                  success: true 
               }); 
            } 
                
            break;
                
         case "offer": 
            //for ex. UserA wants to call UserB 
            console.log("Sending offer to: ", data.name); 
                
            //if UserB exists then send him offer details 
            var conn = users[data.name]; 
                
            if(conn != null) { 
               //setting that UserA connected with UserB 
               connection.otherName = data.name; 
                    
               sendTo(conn, { 
                  type: "offer", 
                  offer: data.offer, 
                  name: connection.name 
               }); 
            } 
                
            break;
                
         case "answer": 
            console.log("Sending answer to: ", data.name); 
            //for ex. UserB answers UserA 
            var conn = users[data.name]; 
                
            if(conn != null) { 
               connection.otherName = data.name; 
               sendTo(conn, { 
                  type: "answer", 
                  answer: data.answer 
               });
            } 
                
            break;
                
         case "candidate": 
            console.log("Sending candidate to:",data.name); 
            var conn = users[data.name];  
                
            if(conn != null) { 
               sendTo(conn, { 
                  type: "candidate", 
                  candidate: data.candidate 
               }); 
            } 
                
            break;
                
         case "leave": 
            console.log("Disconnecting from", data.name); 
            var conn = users[data.name]; 
            conn.otherName = null; 
                
            //notify the other user so he can disconnect his peer connection 
            if(conn != null) { 
               sendTo(conn, { 
                  type: "leave" 
               }); 
            }  
                
            break;
                
         default: 
            sendTo(connection, { 
               type: "error", 
               message: "Command not found: " + data.type 
            });
                
            break; 
      }  
   });
    
   //when user exits, for example closes a browser window 
   //this may help if we are still in "offer","answer" or "candidate" state 
   connection.on("close", function() { 
    
      if(connection.name) { 
         delete users[connection.name]; 
            
         if(connection.otherName) { 
            console.log("Disconnecting from ", connection.otherName); 
            var conn = users[connection.otherName]; 
            conn.otherName = null;  
                
            if(conn != null) { 
               sendTo(conn, { 
                  type: "leave" 
              }); 
            }  
         } 
      } 
   });  
    
   connection.send("Hello world"); 
}); 
 
function sendTo(connection, message) { 
   connection.send(JSON.stringify(message)); 
}

客户申请

测试该应用程序的一种方法是打开两个浏览器选项卡，然后尝试进行音频通话。

首先，我们需要安装引导程序库。 Bootstrap是用于开发Web应用程序的前端框架。您可以在http://getbootstrap.com/上了解更多信息。创建一个名为“ audiochat”的文件夹。这将是我们的根应用程序文件夹。在此文件夹中创建一个文件package.json （管理npm依赖项是必需的）并添加以下内容-

{ 
   "name": "webrtc-audiochat", 
   "version": "0.1.0", 
   "description": "webrtc-audiochat", 
   "author": "Author", 
   "license": "BSD-2-Clause" 
}

然后运行npm install bootstrap 。这会将引导程序库安装在audiochat / node_modules文件夹中。

现在，我们需要创建一个基本的HTML页面。使用以下代码在根文件夹中创建index.html文件-

WebRTC Voice Demo 
       
   
 
   
    
    
      
 
        
         
 
            

                
               
WebRTC Voice Demo. Please sign in

                
                
                
                
            
 
         
 
            
      

        
      

        
         
 
            
            
 
               Local audio:  
            

                
            
 
               Remote audio:

您应该熟悉此页面。我们添加了引导css文件。我们还定义了两个页面。最后，我们创建了几个文本字段和按钮，用于从用户那里获取信息。您应该看到本地和远程音频流的两个音频元素。注意，我们已经添加了指向client.js文件的链接。

现在，我们需要与我们的信令服务器建立连接。使用以下代码在根文件夹中创建client.js文件-

//our username 
var name; 
var connectedUser;
  
//connecting to our signaling server 
var conn = new WebSocket('ws://localhost:9090');
  
conn.onopen = function () { 
   console.log("Connected to the signaling server"); 
}; 
 
//when we got a message from a signaling server 
conn.onmessage = function (msg) { 
   console.log("Got message", msg.data);  
   var data = JSON.parse(msg.data);  
    
   switch(data.type) { 
      case "login": 
         handleLogin(data.success); 
         break; 
      //when somebody wants to call us 
      case "offer": 
         handleOffer(data.offer, data.name); 
         break; 
      case "answer": 
         handleAnswer(data.answer); 
         break; 
      //when a remote peer sends an ice candidate to us 
      case "candidate": 
         handleCandidate(data.candidate); 
         break;
      case "leave": 
         handleLeave(); 
         break; 
      default: 
         break; 
   } 
};
  
conn.onerror = function (err) { 
   console.log("Got error", err); 
};
  
//alias for sending JSON encoded messages 
function send(message) { 
   //attach the other peer username to our messages
   if (connectedUser) { 
      message.name = connectedUser; 
   } 
    
   conn.send(JSON.stringify(message)); 
};

现在通过节点服务器运行我们的信令服务器。然后，在根文件夹中运行static命令并在浏览器中打开页面。您应该看到以下控制台输出-

运行信令服务器

下一步是使用唯一的用户名实现用户登录。我们只需将用户名发送到服务器，然后服务器告诉我们是否已使用该用户名。将以下代码添加到client.js文件中-

//****** 
//UI selectors block 
//******

var loginPage = document.querySelector('#loginPage'); 
var usernameInput = document.querySelector('#usernameInput'); 
var loginBtn = document.querySelector('#loginBtn');
 
var callPage = document.querySelector('#callPage'); 
var callToUsernameInput = document.querySelector('#callToUsernameInput');
var callBtn = document.querySelector('#callBtn');
 
var hangUpBtn = document.querySelector('#hangUpBtn');
  
callPage.style.display = "none";
  
// Login when the user clicks the button 
loginBtn.addEventListener("click", function (event) { 
   name = usernameInput.value;
    
   if (name.length > 0) { 
      send({
         type: "login", 
         name: name 
      }); 
   } 
    
}); 
 
function handleLogin(success) { 
   if (success === false) { 
      alert("Ooops...try a different username"); 
   } else { 
      loginPage.style.display = "none"; 
      callPage.style.display = "block"; 
        
      //********************** 
      //Starting a peer connection 
      //**********************
                 
   } 
    
};

首先，我们选择一些对页面上元素的引用。我们隐藏了呼叫页面。然后，我们在登录按钮上添加一个事件侦听器。当用户单击它时，我们会将其用户名发送到服务器。最后，我们实现handleLogin回调。如果登录成功，我们将显示呼叫页面并开始建立对等连接。

要启动对等连接，我们需要-

  • 从麦克风获取音频流
  • 创建RTCPeerConnection对象

将以下代码添加到“ UI选择器块”-

var localAudio = document.querySelector('#localAudio'); 
var remoteAudio = document.querySelector('#remoteAudio'); 

var yourConn; 
var stream;

修改handleLogin函数-

function handleLogin(success) { 
   if (success === false) { 
      alert("Ooops...try a different username"); 
   } else { 
      loginPage.style.display = "none"; 
      callPage.style.display = "block";
        
      //********************** 
      //Starting a peer connection 
      //********************** 
        
      //getting local audio stream 
      navigator.webkitGetUserMedia({ video: false, audio: true }, function (myStream) { 
         stream = myStream; 
            
         //displaying local audio stream on the page
         localAudio.src = window.URL.createObjectURL(stream);
            
         //using Google public stun server 
         var configuration = { 
            "iceServers": [{ "url": "stun:stun2.1.google.com:19302" }] 
         }; 
            
         yourConn = new webkitRTCPeerConnection(configuration); 
            
         // setup stream listening 
         yourConn.addStream(stream); 
            
         //when a remote user adds stream to the peer connection, we display it 
         yourConn.onaddstream = function (e) { 
            remoteAudio.src = window.URL.createObjectURL(e.stream); 
         }; 
            
         // Setup ice handling 
         yourConn.onicecandidate = function (event) { 
            if (event.candidate) { 
               send({ 
                  type: "candidate", 
               }); 
            } 
         };  
            
      }, function (error) { 
         console.log(error); 
      }); 
        
   } 
};

现在，如果您运行代码，则该页面应允许您登录并在页面上显示本地音频流。

允许登录

现在我们准备发起呼叫。首先，我们向其他用户发送报价。用户获得报价后，他将创建答案并开始交易ICE候选人。将以下代码添加到client.js文件-

//initiating a call 
callBtn.addEventListener("click", function () { 
   var callToUsername = callToUsernameInput.value; 
    
   if (callToUsername.length > 0) { 
      connectedUser = callToUsername; 
        
      // create an offer 
      yourConn.createOffer(function (offer) { 
         send({ 
            type: "offer", 
            offer: offer 
         }); 
            
         yourConn.setLocalDescription(offer); 
            
      }, function (error) { 
         alert("Error when creating an offer"); 
      }); 
   } 
    
});
 
//when somebody sends us an offer 
function handleOffer(offer, name) { 
   connectedUser = name; 
   yourConn.setRemoteDescription(new RTCSessionDescription(offer)); 
    
   //create an answer to an offer 
   yourConn.createAnswer(function (answer) { 
      yourConn.setLocalDescription(answer); 
        
      send({ 
         type: "answer",
         answer: answer 
      }); 
        
   }, function (error) { 
      alert("Error when creating an answer"); 
   }); 
    
};
 
//when we got an answer from a remote user 
function handleAnswer(answer) { 
   yourConn.setRemoteDescription(new RTCSessionDescription(answer)); 
};
 
//when we got an ice candidate from a remote user 
function handleCandidate(candidate) { 
   yourConn.addIceCandidate(new RTCIceCandidate(candidate)); 
};

我们添加一个单击处理程序呼叫按钮，从而启动的报价。然后，我们实现onmessage处理程序期望的几个处理程序。它们将被异步处理，直到两个用户都建立了连接。

最后一步是实现挂断功能。这将停止传输数据，并告诉其他用户关闭呼叫。添加以下代码-

//hang up 
hangUpBtn.addEventListener("click", function () { 
   send({ 
      type: "leave" 
   });  
    
   handleLeave(); 
});
  
function handleLeave() { 
   connectedUser = null; 
   remoteAudio.src = null;
    
   yourConn.close(); 
   yourConn.onicecandidate = null; 
   yourConn.onaddstream = null;
};

当用户单击“挂断”按钮时-

  • 它将向其他用户发送“离开”消息
  • 它将关闭RTCPeerConnection并在本地破坏连接

现在运行代码。您应该能够使用两个浏览器选项卡登录到服务器。然后，您可以对选项卡进行音频呼叫并挂断电话。

登录页面通话和挂断页面

以下是整个client.js文件-

//our username 
var name; 
var connectedUser;
 
//connecting to our signaling server 
var conn = new WebSocket('ws://localhost:9090');
 
conn.onopen = function () { 
   console.log("Connected to the signaling server"); 
};
 
//when we got a message from a signaling server 
conn.onmessage = function (msg) { 
   console.log("Got message", msg.data); 
   var data = JSON.parse(msg.data); 
    
   switch(data.type) { 
      case "login": 
         handleLogin(data.success); 
         break; 
      //when somebody wants to call us 
      case "offer": 
         handleOffer(data.offer, data.name); 
         break; 
      case "answer": 
         handleAnswer(data.answer); 
         break; 
      //when a remote peer sends an ice candidate to us 
      case "candidate": 
         handleCandidate(data.candidate); 
         break; 
      case "leave": 
         handleLeave(); 
         break; 
      default: 
         break; 
   } 
}; 

conn.onerror = function (err) { 
   console.log("Got error", err); 
};
 
//alias for sending JSON encoded messages 
function send(message) { 
   //attach the other peer username to our messages 
   if (connectedUser) { 
      message.name = connectedUser; 
   } 
    
   conn.send(JSON.stringify(message)); 
};
 
//****** 
//UI selectors block 
//****** 

var loginPage = document.querySelector('#loginPage'); 
var usernameInput = document.querySelector('#usernameInput'); 
var loginBtn = document.querySelector('#loginBtn');

var callPage = document.querySelector('#callPage'); 
var callToUsernameInput = document.querySelector('#callToUsernameInput');
var callBtn = document.querySelector('#callBtn'); 

var hangUpBtn = document.querySelector('#hangUpBtn'); 
var localAudio = document.querySelector('#localAudio'); 
var remoteAudio = document.querySelector('#remoteAudio'); 

var yourConn; 
var stream; 

callPage.style.display = "none";
 
// Login when the user clicks the button 
loginBtn.addEventListener("click", function (event) { 
   name = usernameInput.value; 
    
   if (name.length > 0) { 
      send({ 
         type: "login", 
         name: name 
      }); 
   } 
    
});
 
function handleLogin(success) { 
   if (success === false) { 
      alert("Ooops...try a different username"); 
   } else { 
      loginPage.style.display = "none"; 
      callPage.style.display = "block"; 
        
      //********************** 
      //Starting a peer connection 
      //********************** 
        
      //getting local audio stream 
      navigator.webkitGetUserMedia({ video: false, audio: true }, function (myStream) { 
         stream = myStream; 
            
         //displaying local audio stream on the page 
         localAudio.src = window.URL.createObjectURL(stream);
            
         //using Google public stun server 
         var configuration = { 
            "iceServers": [{ "url": "stun:stun2.1.google.com:19302" }] 
         }; 
            
         yourConn = new webkitRTCPeerConnection(configuration); 
            
         // setup stream listening 
         yourConn.addStream(stream); 
            
         //when a remote user adds stream to the peer connection, we display it 
         yourConn.onaddstream = function (e) { 
            remoteAudio.src = window.URL.createObjectURL(e.stream); 
         }; 
            
         // Setup ice handling 
         yourConn.onicecandidate = function (event) { 
            if (event.candidate) { 
               send({ 
                  type: "candidate", 
                  candidate: event.candidate 
               }); 
            } 
         }; 
            
      }, function (error) { 
         console.log(error); 
      }); 
        
   } 
};
 
//initiating a call 
callBtn.addEventListener("click", function () { 
   var callToUsername = callToUsernameInput.value; 
    
   if (callToUsername.length > 0) { 
      connectedUser = callToUsername; 
        
      // create an offer 
      yourConn.createOffer(function (offer) { 
         send({
            type: "offer", 
            offer: offer 
         }); 
            
         yourConn.setLocalDescription(offer); 
      }, function (error) { 
         alert("Error when creating an offer"); 
      }); 
   } 
});
 
//when somebody sends us an offer 
function handleOffer(offer, name) { 
   connectedUser = name; 
   yourConn.setRemoteDescription(new RTCSessionDescription(offer)); 
    
   //create an answer to an offer 
   yourConn.createAnswer(function (answer) { 
      yourConn.setLocalDescription(answer); 
        
      send({ 
         type: "answer", 
         answer: answer 
      });
        
   }, function (error) { 
      alert("Error when creating an answer"); 
   }); 
    
};
 
//when we got an answer from a remote user 
function handleAnswer(answer) { 
   yourConn.setRemoteDescription(new RTCSessionDescription(answer)); 
};
 
//when we got an ice candidate from a remote user 
function handleCandidate(candidate) { 
   yourConn.addIceCandidate(new RTCIceCandidate(candidate)); 
};
 
//hang up
hangUpBtn.addEventListener("click", function () { 
   send({ 
      type: "leave" 
   }); 
    
   handleLeave(); 
});
 
function handleLeave() { 
   connectedUser = null; 
   remoteAudio.src = null; 
    
   yourConn.close(); 
   yourConn.onicecandidate = null; 
   yourConn.onaddstream = null; 
};