CNCEC_SUBQHSE_WUHUAN/SGGL/FineUIPro.Web/res/js/mixrtc.min.js

class WebRTC {
  constructor(config) {
    this.ChangeStatus = "";
    this.options = {
      mediaConstraints: { audio: true, video: false },
      pcConfig: {
        iceServers: [
          { urls: ["stun:stun.mixcom.cn:3478"] },
          {
            urls: "turn:stun.mixcom.cn:3478",
            username: "mixcom",
            credential: "MikangyunST",
          },
        ],
      },
    };
    this.audioConf = {
      audio: "",
      ringout: "",
      ringin: "",
      hangup: "",
    };
    if (config == undefined) {
      console.log("[RTC] : RTC is missing configuration information");
      return false;
    }
    if (
      config.audioConfigs == undefined ||
      typeof config.audioConfigs.audio != "object"
    ) {
      console.log("[RTC] : audio is not configured");
    } else {
      this.audioConf = { ...this.audioConf, ...config.audioConfigs };
    }
    if (config.changeStatus) {
      this.ChangeStatus = config.changeStatus;
    }
    if (config.options !== undefined) {
      this.options = config.options;
    }
    if (config.debug !== undefined && config.debug === true) {
      JsSIP.debug.enable("JsSIP:*");
    }
  }

  // 初始化录音播放
  audioInit() {
    // 默认是循环播放
    const audio = this.audioConf.audio;
    audio.srcObject = null;
    audio.currentTime = 0;
    audio.loop = false;
    audio.src = "";
  }

  // 播放录音
  audioPlay(src) {
    const audio = this.audioConf.audio;
    this.audioInit();
    if (src == "ringin" || src == "ringout") {
      audio.loop = true;
    }
    audio.src = this.audioConf[src];
    audio.load();
    audio.play();
  }

  // 连接WebSocket
  WebSocket(url) {
    this.socket = new JsSIP.WebSocketInterface(url);
  }

  // 注册分机
  register(config) {
    console.log("注册分机");
    this.configuration = {
      sockets: [this.socket],
      uri: "sip:" + config.extension + "@" + config.sipserver,
      password: config.password,
      contact_uri: "sip:" + config.extension + "@" + config.sipserver,
    };
    this.ua = new JsSIP.UA(this.configuration);
    // Websocket连接成功
    this.ua.on("connected", function () {
      console.log("websocket连接成功");
    });

    // Websocket连接断开
    this.ua.on("disconnected", function () {
      console.log("websocket连接断开");
      // 重连
      if (this.ua) {
          this.ua.start();
      }
    });

    var that = this;
    // 呼入或者呼出时触发的事件
    this.ua.on("newRTCSession", function (e) {
      const { session, originator } = e;

      that.session = session;

      switch (originator) {
        // out
        case "local":
          // 此处为呼出前的操作
          that.ChangeStatus("CallOut");
          break;
        case "remote":
          that.audioPlay("ringin");
          that.ChangeStatus("CallIn", session.remote_identity.display_name);
          // 此处为呼入接通前的操作
          break;
        default:
      }

      // 连接中
      session.on("connecting", () => {
        that.ChangeStatus("Connecting");
      });
      // 连接已接受
      session.on("accepted", () => {
        that.ChangeStatus("Connected");
      });

      // 通话中
      session.on("confirmed", () => {
        that.ChangeStatus("Calling");
        const stream = new MediaStream();
        const receivers = session.connection.getReceivers();
        if (receivers) {
          receivers.forEach((receiver) => stream.addTrack(receiver.track));
        }
        audio.srcObject = stream;
        // 最后都要播放
        audio.play();
      });

      // 播放回铃音
      session.on("progress", () => {
        // 判断呼出 播放回铃
        if (originator === "local") {
          that.audioInit();
          const stream = new MediaStream();
          const receivers = session.connection.getReceivers();
          if (receivers) {
            receivers.forEach((receiver) => stream.addTrack(receiver.track));
          }
          audio.srcObject = stream;
          audio.play();
        }
      });

      // 结束
      session.on("ended", () => {
        console.log("挂断了");
        that.ChangeStatus("Ended", "");
        that.audioPlay("hangup");
      });
      // 失败
      session.on("failed", (e) => {
        console.log("失败了");
        that.ChangeStatus("Failed", "");
        that.audioPlay("hangup");
      });

      // ICE 交互连通建立事件 加快通讯速度
      session.on('icecandidate', function (event) {
        if (
          event.candidate.type === 'srflx' &&
          event.candidate.relatedAddress !== null &&
          event.candidate.relatedPort !== null
        ) {
          event.ready();
        }
      });
    });

    this.ua.on("registered", () => {
      console.log("[SIP Phone] : Registered (ON LINE)");
    });
    this.ua.on("unregistered", () => {
      console.log("[SIP Phone] : Unregistered (OFF LINE)");
    });
    this.ua.on("registrationFailed", (e) => {
      console.log("[SIP Phone] : Registration Failed (OFF LINE)");
    });

    this.ua.start();
  }

  // 注销
  unregister() {
    this.ua.unregister(this.options);
    this.ua.stop();
    this.ua = null;
    console.log("[SIP Phone] : unregistered")
  }

  // 呼出
  call(phone) {
    this.ua.call(phone, this.options);
    this.audioPlay("ringout");
  }

  // 挂断
  hangup() {
    this.ua.terminateSessions();
  }

  // 接通
  answer() {
    console.log(this.session)
    this.session.answer(this.options);
  }

  // 拒接
  reject() {
    this.ua.terminateSessions();
    // this.seesion.terminate(this.options);
  }
}