测试完成

WXL

2025-08-07 6f1e7a63fa6afc261ba2247bf1d56f59dbb9f4c9

 src/utils/sipService.js

@@ -4,11 +4,11 @@
  constructor() {
    this.ua = null;
    this.currentSession = null;
    this.onStatusChange = null; // 状态变化回调
    this.onCallStatusChange = null; // 新增通话状态回调
    this.onStatusChange = null;
    this.onCallStatusChange = null;
    this.onIncomingCall = null;
  }

  // 初始化SIP客户端
  init(config) {
    try {
      this.updateStatus("connecting", "连接中...");
@@ -18,160 +18,161 @@
        uri: config.sipUri,
        password: config.password,
        display_name: config.displayName,
        iceservers: [],
        // realm: config.realm,
        iceServers: [{ urls: "stun:stun.l.google.com:19302" }],
        register: true,
        session_expires: 180,
        sessionTimersExpires: 300, // 设置 Session-Expires=120（必须 >= Min-SE）
        extraHeaders: [
          "Min-SE: 120", // 可选：显式告诉服务器你支持的最小值
        ],
        register_expires: 300, // 注册有效期(秒)
        connection_recovery_min_interval: 2, // 最小重连间隔
        connection_recovery_max_interval: 30, // 最大重连间隔
        session_expires: 900,
        sessionTimersExpires: 600,
        extraHeaders: ["Min-SE: 120"],
        register_expires: 300,
        connection_recovery_min_interval: 2,
        connection_recovery_max_interval: 30,
        pcConfig: {
          iceTransportPolicy: "all",
          rtcpMuxPolicy: "require",
          bundlePolicy: "max-bundle"
        }
      });

      this.ua.start();

      // 注册事件监听
      this.ua.on("registered", () => {
        this.updateStatus("registered", "已注册");
      });
      // 事件监听
      this.ua.on("registered", () => this.updateStatus("registered", "已注册"));
      this.ua.on("registrationFailed", (e) =>
        this.updateStatus("failed", `注册失败: ${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));

      this.ua.on("registrationFailed", (e) => {
        this.updateStatus("failed", `注册失败: ${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...");
      });

      // 监听来电
      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);
    }
  }
  handleIncomingCall(session) {
    if (session.direction === "incoming") {
      console.log("来电:", session.remote_identity.uri.toString());
      // 可以在这里触发 UI 通知
      if (this.onIncomingCall) {
        this.onIncomingCall(session);
      }
      throw error;
    }
  }

  // 更新状态并通知UI
  updateStatus(type, text) {
    console.log(`SIP状态更新: ${type} - ${text}`);
    if (this.onStatusChange) {
      this.onStatusChange({ type, text });
    }
  }

  // 一键拨号 - 增加注册状态检查
  makeCall(targetNumber) {
    if (!this.ua) {
      throw new Error("SIP客户端未初始化");
    }
    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");
    }
        if (!this.ua.isRegistered()) {
          throw new Error("SIP未注册，无法呼叫");
        }

    const options = {
      sessionTimers: true,
      sessionTimersExpires: 300,
      extraHeaders: [
        "Min-SE: 120",
        "Route: <sip:@192.168.100.6>",
        "Accept: application/sdp",
        "Supported: replaces, timer",
        "Allow: INVITE, ACK, BYE, CANCEL, OPTIONS",
      ],
      eventHandlers: {
        progress: (e) => {
          this.updateCallStatus("calling", "呼叫中...");
        },
        failed: (e) => {
          this.updateCallStatus("ended", `呼叫失败: ${e.cause}`);
        },
        ended: (e) => {
          this.updateCallStatus("ended", "通话结束");
        },
        confirmed: (e) => {
          this.updateCallStatus("connected", "通话已接通");
        },
      },
      mediaConstraints: {
        audio: true,
        video: false,
      },
      rtcOfferConstraints: {
        offerToReceiveAudio: 1,
        offerToReceiveVideo: 0,
        mandatory: {
          OfferToReceiveAudio: true,
          OfferToReceiveVideo: false,
        },
      },
      pcConfig: {
        iceServers: [{ urls: "stun:stun.l.google.com:19302" }],
        iceTransportPolicy: "all",
        bundlePolicy: "balanced",
        rtcpMuxPolicy: "require",
        codecs: {
          audio: [
            { name: "PCMU", clockRate: 8000, payloadType: 0 },
            { name: "PCMA", clockRate: 8000, payloadType: 8 },
        const options = {
          sessionTimers: false, // 暂时禁用以减少兼容性问题
          extraHeaders: [
            "Min-SE: 120",
            "Accept: application/sdp",
            "Supported: outbound"
          ],
          video: [],
        },
      },
    };
          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:${targetNumber}@192.168.100.6`,
      options
    );
    // 在会话创建后修改 SDP
    this.currentSession.on("peerconnection", (pc) => {
       this.updateCallStatus('calling', '呼叫中...');
      pc.createOffer = (offerOptions) => {
        return RTCPeerConnection.prototype.createOffer
          .call(pc, offerOptions)
          .then((offer) => {
            const modifiedSdp = offer.sdp
              .replace(/c=IN IP4 192\.168\.100\.10/g, "c=IN IP4 192.168.100.6")
              .replace(/m=audio \d+ RTP\/AVP.*/, "m=audio 7078 RTP/AVP 0 8");
            return new RTCSessionDescription({
              type: "offer",
              sdp: modifiedSdp,
            });
          });
        this.currentSession = this.ua.call(
          `sip:${targetNumber}@192.168.10.124`,
          options
        );

        this.setupPeerConnection(this.currentSession);
        this.setupAudio(this.currentSession);

      } catch (error) {
        this.updateCallStatus("failed", `呼叫失败1: ${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;
        }
      };
    });
    this.currentSession.on('failed', (e) => {
        this.updateCallStatus('failed', `呼叫失败2: ${e.cause}`);
      });

      this.currentSession.on('ended', () => {
        this.updateCallStatus('ended', '通话已结束');
      });

      this.currentSession.on('confirmed', () => {
        this.updateCallStatus('connected', '通话已接通');
      });
    this.setupAudio(this.currentSession);
  }

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

    // 1. 标准化连接行
    sdp = sdp.replace(/c=IN IP4.*\r\n/, "c=IN IP4 0.0.0.0\r\n");

    // 2. 标准化音频媒体行
    sdp = sdp.replace(/m=audio \d+.*\r\n/,
      "m=audio 9 UDP/TLS/RTP/SAVPF 0 8\r\n");

    // 3. 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";
    }

    // 4. 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) {
    console.error("呼叫失败详情:", {
      cause: e.cause,
      message: e.message,
      response: e.response && e.response.status_code
    });

    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;
      default:
        errorMessage = `呼叫失败: ${e.cause || e.message}`;
    }

    this.updateCallStatus("failed2", errorMessage);
    reject(new Error(errorMessage));
  }

  setupAudio(session) {
    session.connection.addEventListener("addstream", (e) => {
      const audioElement = document.getElementById("remoteAudio");
@@ -180,21 +181,37 @@
      }
    });
  }
  // 挂断当前通话

  endCall() {
  if (this.currentSession) {
    if (this.currentSession) {
      this.currentSession.terminate();
      this.updateCallStatus('ended', '通话已结束');
      this.updateCallStatus("ended", "通话已结束");
      this.currentSession = null;
    }
  }
  // 新增方法：更新通话状态

  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();