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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}
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// 初始化SIP客户端
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init(config) {
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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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realm: config.realm,
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ha1: config.ha1,
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register: true
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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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console.log('SIP注册成功')
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})
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this.ua.on('registrationFailed', (e) => {
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console.error('SIP注册失败:', e)
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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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}
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// 一键拨号
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makeCall(targetNumber) {
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if (!this.ua) {
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console.error('SIP客户端未初始化')
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return
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}
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const options = {
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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: { audio: true, video: false },
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rtcOfferConstraints: { offerToReceiveAudio: 1 }
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}
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this.currentSession = this.ua.call(`sip:${targetNumber}`, options)
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this.setupAudio(this.currentSession)
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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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// 处理音频流
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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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// 处理来电
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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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// 这里可以触发UI通知
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}
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}
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}
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export default new SipService()
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