import JsSIP from "jssip";
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class SipService {
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constructor() {
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this.ua = null;
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this.currentSession = null;
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this.onStatusChange = null;
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this.onCallStatusChange = null;
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this.onIncomingCall = null;
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}
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init(config) {
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try {
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this.updateStatus("connecting", "连接中;...");
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this.ua = new JsSIP.UA({
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sockets: [new JsSIP.WebSocketInterface(config.wsUrl)],
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uri: config.sipUri,
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password: config.password,
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display_name: config.displayName,
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iceServers: [],
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register: true,
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sessionExpires: 1800,
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minSessionExpires: 90,
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register_expires: 300,
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});
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this.ua.start();
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// 事件监听
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this.ua.on("registered", () =>
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this.updateStatus("registered", "已注册56")
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);
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this.ua.on("registrationFailed", (e) =>
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this.updateStatus("failed", `注册失败11: ${e.cause}`)
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);
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this.ua.on("disconnected", () =>
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this.updateStatus("disconnected", "连接断开")
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);
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this.ua.on("connected", () =>
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this.updateStatus("connecting", "重新连接中...")
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);
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this.ua.on("newRTCSession", (data) =>
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this.handleIncomingCall(data.session)
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);
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} catch (error) {
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this.updateStatus("failed", `初始化失败: ${error.message}`);
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console.error("SIP初始化失败:", error);
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throw error;
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}
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}
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makeCall(targetNumber) {
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return new Promise((resolve, reject) => {
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try {
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if (!this.ua) {
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throw new Error("SIP客户端未初始化");
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}
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if (!this.ua.isRegistered()) {
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throw new Error("SIP未注册，无法呼叫");
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}
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const options = {
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sessionTimers: true, // 启用会话计时器
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sessionTimersExpires: 150,
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extraHeaders: ["Accept: application/sdp"],
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mediaConstraints: { audio: true, video: false },
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rtcOfferConstraints: {
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offerToReceiveAudio: true,
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offerToReceiveVideo: false,
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},
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eventHandlers: {
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progress: () => this.updateCallStatus("calling", "呼叫中..."),
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failed: (e) => {
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this.handleCallFailure(e, reject);
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},
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ended: () => this.updateCallStatus("ended", "通话结束"),
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confirmed: () => {
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this.updateCallStatus("connected", "通话已接通");
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resolve();
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},
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},
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};
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this.currentSession = this.ua.call(
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`sip:${targetNumber}@192.168.10.124`,
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options
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);
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this.setupPeerConnection(this.currentSession);
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this.setupAudio(this.currentSession);
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} catch (error) {
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this.updateCallStatus("failed", `呼叫失败22: ${error.message}`);
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reject(error);
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}
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});
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}
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setupPeerConnection(session) {
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session.on("peerconnection", (pc) => {
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const originalCreateOffer = pc.createOffer.bind(pc);
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pc.createOffer = async (offerOptions) => {
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try {
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const offer = await originalCreateOffer(offerOptions);
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return this.normalizeSDP(offer);
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} catch (error) {
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console.error("创建Offer失败:", error);
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throw error;
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}
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};
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});
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}
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normalizeSDP(offer) {
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let sdp = offer.sdp;
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// 标准化SDP
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sdp = sdp.replace(/c=IN IP4.*\r\n/, "c=IN IP4 0.0.0.0\r\n");
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sdp = sdp.replace(
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/m=audio \d+.*\r\n/,
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"m=audio 9 UDP/TLS/RTP/SAVPF 0 8\r\n"
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);
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// 确保包含基本编解码器
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if (!sdp.includes("PCMU/8000")) sdp += "a=rtpmap:0 PCMU/8000\r\n";
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if (!sdp.includes("PCMA/8000")) sdp += "a=rtpmap:8 PCMA/8000\r\n";
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// 添加必要属性
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sdp += "a=rtcp-mux\r\n";
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sdp += "a=sendrecv\r\n";
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console.log("标准化后的SDP:", sdp);
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return new RTCSessionDescription({
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type: offer.type,
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sdp: sdp,
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});
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}
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handleCallFailure(e, reject) {
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if (e.response?.status_code === 422) {
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const serverMinSE = e.response.headers["Min-SE"]?.[0]?.raw || "未知";
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console.error(`服务器要求 Min-SE ≤ ${serverMinSE}，当前设置: 120`);
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}
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console.error("呼叫失败详情:", {
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cause: e.cause,
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message: e.message,
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response: e.response && {
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status: e.response.status_code,
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reason: e.response.reason_phrase,
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},
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});
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let errorMessage = "呼叫失败";
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switch (e.cause) {
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case "Incompatible SDP":
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errorMessage = "媒体协商失败，请检查编解码器配置";
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break;
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case "488":
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case "606":
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errorMessage = "对方设备不支持当前媒体配置";
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break;
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case "422":
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errorMessage = "会话参数不满足服务器要求";
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break;
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default:
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errorMessage = `呼叫失败: ${e.cause || e.message}`;
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}
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this.updateCallStatus("failed55", errorMessage);
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reject(new Error(errorMessage));
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}
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setupAudio(session) {
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session.connection.addEventListener("addstream", (e) => {
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const audioElement = document.getElementById("remoteAudio");
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if (audioElement) {
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audioElement.srcObject = e.stream;
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}
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});
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}
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endCall() {
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if (this.currentSession) {
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this.currentSession.terminate();
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this.updateCallStatus("ended", "通话已结束");
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this.currentSession = null;
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}
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}
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updateStatus(type, text) {
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console.log(`SIP状态更新: ${type} - ${text}`);
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if (this.onStatusChange) {
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this.onStatusChange({ type, text });
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}
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}
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updateCallStatus(type, text) {
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console.log(`通话状态更新: ${type} - ${text}`);
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if (this.onCallStatusChange) {
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this.onCallStatusChange({ type, text });
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}
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}
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handleIncomingCall(session) {
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if (session.direction === "incoming") {
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console.log("来电:", session.remote_identity.uri.toString());
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if (this.onIncomingCall) {
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this.onIncomingCall(session);
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}
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}
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}
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}
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export default new SipService();
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