import JsSIP from "jssip";

class SipService {
  constructor() {
    this.ua = null;
    this.currentSession = null;
    this.onStatusChange = null;
    this.onCallStatusChange = null;
    this.onIncomingCall = null;
  }

  init(config) {
    try {
      this.updateStatus("connecting", "连接中;...");

      this.ua = new JsSIP.UA({
        sockets: [new JsSIP.WebSocketInterface(config.wsUrl)],
        uri: config.sipUri,
        password: config.password,
        display_name: config.displayName,
        iceServers: [],
        register: true,
        sessionExpires: 1800,
        minSessionExpires: 90,
        register_expires: 300,
      });

      this.ua.start();

      // 事件监听
      this.ua.on("registered", () =>
        this.updateStatus("registered", "已注册56")
      );
      this.ua.on("registrationFailed", (e) =>
        this.updateStatus("failed", `注册失败11: ${e.cause}`)
      );
      this.ua.on("disconnected", () =>
        this.updateStatus("disconnected", "连接断开")
      );
      this.ua.on("connected", () =>
        this.updateStatus("connecting", "重新连接中...")
      );
      this.ua.on("newRTCSession", (data) =>
        this.handleIncomingCall(data.session)
      );
    } catch (error) {
      this.updateStatus("failed", `初始化失败: ${error.message}`);
      console.error("SIP初始化失败:", error);
      throw error;
    }
  }

  makeCall(targetNumber) {
    return new Promise((resolve, reject) => {
      try {
        if (!this.ua) {
          throw new Error("SIP客户端未初始化");
        }

        if (!this.ua.isRegistered()) {
          throw new Error("SIP未注册，无法呼叫");
        }

        const options = {
          sessionTimers: true, // 启用会话计时器
          sessionTimersExpires: 150,
          extraHeaders: ["Accept: application/sdp"],
          mediaConstraints: { audio: true, video: false },
          rtcOfferConstraints: {
            offerToReceiveAudio: true,
            offerToReceiveVideo: false,
          },
          eventHandlers: {
            progress: () => this.updateCallStatus("calling", "呼叫中..."),
            failed: (e) => {
              this.handleCallFailure(e, reject);
            },
            ended: () => this.updateCallStatus("ended", "通话结束"),
            confirmed: () => {
              this.updateCallStatus("connected", "通话已接通");
              resolve();
            },
          },
        };

        this.currentSession = this.ua.call(
          `sip:${targetNumber}@192.168.10.124`,
          options
        );

        this.setupPeerConnection(this.currentSession);
        this.setupAudio(this.currentSession);
      } catch (error) {
        this.updateCallStatus("failed", `呼叫失败22: ${error.message}`);
        reject(error);
      }
    });
  }

  setupPeerConnection(session) {
    session.on("peerconnection", (pc) => {
      const originalCreateOffer = pc.createOffer.bind(pc);

      pc.createOffer = async (offerOptions) => {
        try {
          const offer = await originalCreateOffer(offerOptions);
          return this.normalizeSDP(offer);
        } catch (error) {
          console.error("创建Offer失败:", error);
          throw error;
        }
      };
    });
  }

  normalizeSDP(offer) {
    let sdp = offer.sdp;

    // 标准化SDP
    sdp = sdp.replace(/c=IN IP4.*\r\n/, "c=IN IP4 0.0.0.0\r\n");
    sdp = sdp.replace(
      /m=audio \d+.*\r\n/,
      "m=audio 9 UDP/TLS/RTP/SAVPF 0 8\r\n"
    );

    // 确保包含基本编解码器
    if (!sdp.includes("PCMU/8000")) sdp += "a=rtpmap:0 PCMU/8000\r\n";
    if (!sdp.includes("PCMA/8000")) sdp += "a=rtpmap:8 PCMA/8000\r\n";

    // 添加必要属性
    sdp += "a=rtcp-mux\r\n";
    sdp += "a=sendrecv\r\n";

    console.log("标准化后的SDP:", sdp);
    return new RTCSessionDescription({
      type: offer.type,
      sdp: sdp,
    });
  }

  handleCallFailure(e, reject) {
    if (e.response?.status_code === 422) {
      const serverMinSE = e.response.headers["Min-SE"]?.[0]?.raw || "未知";
      console.error(`服务器要求 Min-SE ≤ ${serverMinSE}，当前设置: 120`);
    }
    console.error("呼叫失败详情:", {
      cause: e.cause,
      message: e.message,
      response: e.response && {
        status: e.response.status_code,
        reason: e.response.reason_phrase,
      },
    });

    let errorMessage = "呼叫失败";
    switch (e.cause) {
      case "Incompatible SDP":
        errorMessage = "媒体协商失败，请检查编解码器配置";
        break;
      case "488":
      case "606":
        errorMessage = "对方设备不支持当前媒体配置";
        break;
      case "422":
        errorMessage = "会话参数不满足服务器要求";
        break;
      default:
        errorMessage = `呼叫失败: ${e.cause || e.message}`;
    }

    this.updateCallStatus("failed55", errorMessage);
    reject(new Error(errorMessage));
  }

  setupAudio(session) {
    session.connection.addEventListener("addstream", (e) => {
      const audioElement = document.getElementById("remoteAudio");
      if (audioElement) {
        audioElement.srcObject = e.stream;
      }
    });
  }

  endCall() {
    if (this.currentSession) {
      this.currentSession.terminate();
      this.updateCallStatus("ended", "通话已结束");
      this.currentSession = null;
    }
  }

  updateStatus(type, text) {
    console.log(`SIP状态更新: ${type} - ${text}`);
    if (this.onStatusChange) {
      this.onStatusChange({ type, text });
    }
  }

  updateCallStatus(type, text) {
    console.log(`通话状态更新: ${type} - ${text}`);
    if (this.onCallStatusChange) {
      this.onCallStatusChange({ type, text });
    }
  }

  handleIncomingCall(session) {
    if (session.direction === "incoming") {
      console.log("来电:", session.remote_identity.uri.toString());
      if (this.onIncomingCall) {
        this.onIncomingCall(session);
      }
    }
  }
}

export default new SipService();