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; // 状态变化回调
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}
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// 初始化SIP客户端
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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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// realm: config.realm,
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register: true,
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session_expires: 180,
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sessionTimersExpires: 300, // 设置 Session-Expires=120（必须 >= Min-SE）
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extraHeaders: [
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"Min-SE: 120", // 可选：显式告诉服务器你支持的最小值
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],
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register_expires: 300, // 注册有效期(秒)
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connection_recovery_min_interval: 2, // 最小重连间隔
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connection_recovery_max_interval: 30, // 最大重连间隔
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});
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this.ua.start();
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// 注册事件监听
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this.ua.on("registered", () => {
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this.updateStatus("registered", "已注册");
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});
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this.ua.on("registrationFailed", (e) => {
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this.updateStatus("failed", `注册失败: ${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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// 监听来电
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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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}
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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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// 可以在这里触发 UI 通知
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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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// 更新状态并通知UI
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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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// 一键拨号 - 增加注册状态检查
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makeCall(targetNumber) {
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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,
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sessionTimersExpires: 300,
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extraHeaders: [
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"Min-SE: 120",
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"Route: <sip:@192.168.100.6>",
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"Accept: application/sdp",
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"Supported: replaces, timer",
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"Allow: INVITE, ACK, BYE, CANCEL, OPTIONS",
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],
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eventHandlers: {
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progress: (e) => console.log("呼叫中..."),
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failed: (e) => console.error("呼叫失败:", e),
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ended: (e) => console.log("通话结束"),
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confirmed: (e) => console.log("通话已接通"),
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},
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mediaConstraints: {
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audio: true,
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video: false,
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},
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rtcOfferConstraints: {
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offerToReceiveAudio: 1,
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offerToReceiveVideo: 0,
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mandatory: {
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OfferToReceiveAudio: true,
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OfferToReceiveVideo: false,
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},
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},
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pcConfig: {
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iceServers: [{ urls: "stun:stun.l.google.com:19302" }],
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iceTransportPolicy: "all",
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bundlePolicy: "balanced",
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rtcpMuxPolicy: "require",
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codecs: {
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audio: [
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{ name: "PCMU", clockRate: 8000, payloadType: 0 },
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{ name: "PCMA", clockRate: 8000, payloadType: 8 },
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],
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video: [],
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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}@192.168.100.6`,
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options
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);
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// 在会话创建后修改 SDP
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this.currentSession.on("peerconnection", (pc) => {
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pc.createOffer = (offerOptions) => {
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return RTCPeerConnection.prototype.createOffer
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.call(pc, offerOptions)
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.then((offer) => {
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const modifiedSdp = offer.sdp
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.replace(/c=IN IP4 192\.168\.100\.10/g, "c=IN IP4 192.168.100.6")
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.replace(/m=audio \d+ RTP\/AVP.*/, "m=audio 7078 RTP/AVP 0 8");
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return new RTCSessionDescription({
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type: "offer",
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sdp: modifiedSdp,
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});
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});
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};
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});
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this.setupAudio(this.currentSession);
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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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// 挂断当前通话
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endCall() {
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if (this.currentSession) {
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this.currentSession.terminate();
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this.currentSession = null;
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}
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}
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}
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export default new SipService();
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