WebSocket与WebRTC在实时音视频通信中的协作应用

var signalingServer = new WebSocket('ws://example.com/signaling');

signalingServer.onmessage = function(event){
    var message = JSON.parse(event.data);
    // 处理信令消息
    handleSignalingMessage(message);
};

function handleSignalingMessage(message){
    switch(message.type){
        case 'offer':
            // 处理offer消息，创建WebRTC连接并回复answer
            handleOfferMessage(message);
            break;
        case 'answer':
            // 处理answer消息，设置远程描述
            handleAnswerMessage(message);
            break;
        case 'candidate':
            // 处理candidate消息，添加候选者
            handleCandidateMessage(message);
            break;
        default:
            break;
    }
}

function handleOfferMessage(message){
    var peerConnection = new RTCPeerConnection();
    // 添加本地媒体流
    navigator.mediaDevices.getUserMedia({ audio: true, video: true }).then(function(stream){
        peerConnection.addStream(stream);
    });
    // 设置远程描述
    peerConnection.setRemoteDescription(new RTCSessionDescription(message));
    // 创建answer并发送
    peerConnection.createAnswer().then(function(answer){
        peerConnection.setLocalDescription(answer);
        signalingServer.send(jsON.stringify(answer));
    });
}

function handleAnswerMessage(message){
    peerConnection.setRemoteDescription(new RTCSessionDescription(message));
}

function handleCandidateMessage(message){
    var candidate = new RTCIceCandidate({
        sdpMLineIndex: message.label,
        candidate: message.candidate
    });
    peerConnection.addIceCandidate(candidate);
}