132 lines
3.7 KiB
JavaScript
Executable File
132 lines
3.7 KiB
JavaScript
Executable File
import Users from '/imports/api/users';
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import Meetings from '/imports/api/meetings';
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import Auth from '/imports/ui/services/auth';
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import {callServer} from '/imports/ui/services/api';
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import {vertoExitAudio, vertoJoinListenOnly, vertoJoinMicrophone} from '/imports/api/verto';
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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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function getVoiceBridge() {
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return Meetings.findOne({}).voiceConf;
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}
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function amIListenOnly() {
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const uid = Auth.userID;
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return Users.findOne({ userId: uid }).user.listenOnly;
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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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function exitAudio(afterExitCall = () => {}) {
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if (!MEDIA_CONFIG.useSIPAudio) {
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vertoExitAudio();
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return;
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} else {
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// To be called when the hangup is initiated
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const hangupCallback = function () {
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console.log('Exiting Voice Conference');
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};
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// Checks periodically until a call is established so we can successfully end the call
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// 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
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// 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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// notify BBB-apps we are leaving the call call if we are listen only
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if (amIListenOnly()) {
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callServer('listenOnlyRequestToggle', false);
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}
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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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})
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// automatically run function
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(this, afterExitCall);
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return false;
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};
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}
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// join the conference. If listen only send the request to the server
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function joinVoiceCallSIP(options) {
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const extension = getVoiceBridge();
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console.log(options);
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if (MEDIA_CONFIG.useSIPAudio) {
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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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window.BBB = {};
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window.BBB.getMyUserInfo = function (callback) {
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const uid = Auth.userID;
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const result = {
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myUserID: uid,
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myUsername: Users.findOne({ userId: uid }).user.name,
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myInternalUserID: uid,
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myAvatarURL: null,
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myRole: 'getMyRole',
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amIPresenter: 'false',
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voiceBridge: extension,
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dialNumber: null,
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};
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return callback(result);
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};
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const m = Meetings.findOne();
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const st = {
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stun: m.stuns,
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turn: m.turns,
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};
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callIntoConference(extension, function () {}, options.isListenOnly, st);
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return;
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}
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}
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function joinListenOnly() {
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callServer('listenOnlyRequestToggle', true);
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if (MEDIA_CONFIG.useSIPAudio) {
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joinVoiceCallSIP({ isListenOnly: true });
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} else {
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vertoJoinListenOnly();
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}
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}
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function joinMicrophone() {
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if (MEDIA_CONFIG.useSIPAudio) {
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joinVoiceCallSIP({ isListenOnly: false });
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} else {
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vertoJoinMicrophone();
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}
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}
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export { joinListenOnly, joinMicrophone, exitAudio, getVoiceBridge, };
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