86 lines
2.7 KiB
JavaScript
86 lines
2.7 KiB
JavaScript
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import BaseAudioBridge from './base';
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import { makeCall } from '/imports/ui/services/api';
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const APP_CONFIG = Meteor.settings.public.app;
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const MEDIA_CONFIG = Meteor.settings.public.media;
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let triedHangup = false;
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export default class SIPBridge extends BaseAudioBridge {
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constructor(userData) {
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super();
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this.userData = userData;
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}
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joinListenOnly(callbackFromManager) {
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makeCall('listenOnlyToggle', true);
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this._joinVoiceCallSIP({ isListenOnly: true }, callbackFromManager);
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}
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joinMicrophone(callbackFromManager) {
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this._joinVoiceCallSIP({ isListenOnly: false }, callbackFromManager);
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}
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// Periodically check the status of the WebRTC call, when a call has been established attempt to
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// hangup, retry if a call is in progress, send the leave voice conference message to BBB
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exitAudio(isListenOnly, afterExitCall = () => {}) {
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// To be called when the hangup is confirmed
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const hangupCallback = function () {
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console.log(`Exited Voice Conference, listenOnly=${isListenOnly}`);
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// notify BBB-apps we are leaving the call if we are in listen only mode
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if (isListenOnly) {
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makeCall('listenOnlyToggle', false);
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}
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};
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// Checks periodically until a call is established so we can successfully
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// end the call clean state
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triedHangup = false;
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// function to initiate call
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const checkToHangupCall = ((context, afterExitCall = () => {}) => {
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// if an attempt to hang up the call is made when the current session is not yet finished,
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// the request has no effect keep track in the session if we haven't tried a hangup
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if (window.getCallStatus() != null && !triedHangup) {
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console.log('Attempting to hangup on WebRTC call');
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window.webrtc_hangup(hangupCallback);
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// we have hung up, prevent retries
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triedHangup = true;
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if (afterExitCall) {
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afterExitCall(this, APP_CONFIG.listenOnly);
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}
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} else {
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console.log('RETRYING hangup on WebRTC call in ' +
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`${MEDIA_CONFIG.WebRTCHangupRetryInterval} ms`);
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// try again periodically
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setTimeout(checkToHangupCall, MEDIA_CONFIG.WebRTCHangupRetryInterval);
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}
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})(this, afterExitCall);
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return false;
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}
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// join the conference. If listen only send the request to the server
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_joinVoiceCallSIP(options, callbackFromManager) {
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const extension = this.userData.voiceBridge;
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console.log(options);
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// create voice call params
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const joinCallback = function (message) {
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console.log('Beginning WebRTC Conference Call');
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};
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const stunsAndTurns = {
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stun: this.userData.stuns,
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turn: this.userData.turns,
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};
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callIntoConference(extension, callbackFromManager, options.isListenOnly, stunsAndTurns);
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}
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}
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