bigbluebutton-Github/bigbluebutton-html5/imports/api/audio/client/bridge/sip.js
2017-05-16 09:22:59 -07:00

128 lines
3.8 KiB
JavaScript

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