var userID, callerIdName=null, conferenceVoiceBridge, userAgent=null, userMicMedia, userWebcamMedia, currentSession=null, callTimeout, callActive, callICEConnected, iceConnectedTimeout, callFailCounter, callPurposefullyEnded, uaConnected, transferTimeout, iceGatheringTimeout; var inEchoTest = true; var html5StunTurn = null; function webRTCCallback(message) { switch (message.status) { case 'succeded': BBB.webRTCCallSucceeded(); break; case 'failed': if (message.errorcode !== 1004) { message.cause = null; } monitorTracksStop(); BBB.webRTCCallFailed(inEchoTest, message.errorcode, message.cause); break; case 'ended': monitorTracksStop(); BBB.webRTCCallEnded(inEchoTest); break; case 'started': monitorTracksStart(); BBB.webRTCCallStarted(inEchoTest); break; case 'connecting': BBB.webRTCCallConnecting(inEchoTest); break; case 'waitingforice': BBB.webRTCCallWaitingForICE(inEchoTest); break; case 'transferring': BBB.webRTCCallTransferring(inEchoTest); break; case 'mediarequest': BBB.webRTCMediaRequest(); break; case 'mediasuccess': BBB.webRTCMediaSuccess(); break; case 'mediafail': BBB.webRTCMediaFail(); break; } } function callIntoConference(voiceBridge, callback, isListenOnly, stunTurn = null) { // root of the call initiation process from the html5 client // Flash will not pass in the listen only field. For html5 it is optional. Assume NOT listen only if no state passed if (isListenOnly == null) { isListenOnly = false; } // if additional stun configuration is passed, store the information if (stunTurn != null) { html5StunTurn = { stunServers: stunTurn.stun, turnServers: stunTurn.turn, }; } // reset callerIdName callerIdName = null; if (!callerIdName) { BBB.getMyUserInfo(function(userInfo) { console.log("User info callback [myUserID=" + userInfo.myUserID + ",myUsername=" + userInfo.myUsername + ",myAvatarURL=" + userInfo.myAvatarURL + ",myRole=" + userInfo.myRole + ",amIPresenter=" + userInfo.amIPresenter + ",dialNumber=" + userInfo.dialNumber + ",voiceBridge=" + userInfo.voiceBridge + ",isListenOnly=" + isListenOnly + "]."); userID = userInfo.myUserID; callerIdName = userInfo.myUserID + "-bbbID-" + userInfo.myUsername; if (isListenOnly) { //prepend the callerIdName so it is recognized as a global audio user callerIdName = "GLOBAL_AUDIO_" + callerIdName; } conferenceVoiceBridge = userInfo.voiceBridge if (voiceBridge === "9196") { voiceBridge = voiceBridge + conferenceVoiceBridge; } else { voiceBridge = conferenceVoiceBridge; } console.log(callerIdName); webrtc_call(callerIdName, voiceBridge, callback, isListenOnly); }); } else { if (voiceBridge === "9196") { voiceBridge = voiceBridge + conferenceVoiceBridge; } else { voiceBridge = conferenceVoiceBridge; } webrtc_call(callerIdName, voiceBridge, callback, isListenOnly); } } function joinWebRTCVoiceConference() { console.log("Joining to the voice conference directly"); inEchoTest = false; // set proper callbacks to previously created user agent if(userAgent) { setUserAgentListeners(webRTCCallback); } callIntoConference(conferenceVoiceBridge, webRTCCallback); } function leaveWebRTCVoiceConference() { console.log("Leaving the voice conference"); webrtc_hangup(); } function startWebRTCAudioTest(){ console.log("Joining the audio test first"); inEchoTest = true; callIntoConference("9196", webRTCCallback); } function stopWebRTCAudioTest(){ console.log("Stopping webrtc audio test"); webrtc_hangup(); } function stopWebRTCAudioTestJoinConference(){ console.log("Transferring from audio test to conference"); webRTCCallback({'status': 'transferring'}); transferTimeout = setTimeout( function() { console.log("Call transfer failed. No response after 3 seconds"); webRTCCallback({'status': 'failed', 'errorcode': 1008}); currentSession = null; if (userAgent != null) { var userAgentTemp = userAgent; userAgent = null; userAgentTemp.stop(); } }, 5000); BBB.listen("UserJoinedVoiceEvent", userJoinedVoiceHandler); currentSession.dtmf(1); inEchoTest = false; } function userJoinedVoiceHandler(event) { console.log("UserJoinedVoiceHandler - " + event); if (inEchoTest === false && userID === event.userID) { BBB.unlisten("UserJoinedVoiceEvent", userJoinedVoiceHandler); clearTimeout(transferTimeout); webRTCCallback({'status': 'started'}); } } function createUA(username, server, callback, makeCallFunc) { if (userAgent) { console.log("User agent already created"); return; } console.log("Fetching STUN/TURN server info for user agent"); console.log(html5StunTurn); if (html5StunTurn != null) { createUAWithStuns(username, server, callback, html5StunTurn, makeCallFunc); return; } BBB.getSessionToken(function(sessionToken) { $.ajax({ dataType: 'json', url: '/bigbluebutton/api/stuns', data: {sessionToken:sessionToken} }).done(function(data) { var stunsConfig = {}; stunsConfig['stunServers'] = ( data['stunServers'] ? data['stunServers'].map(function(data) { return data['url']; }) : [] ); stunsConfig['turnServers'] = ( data['turnServers'] ? data['turnServers'].map(function(data) { return { 'urls': data['url'], 'username': data['username'], 'password': data['password'] }; }) : [] ); //stunsConfig['remoteIceCandidates'] = ( data['remoteIceCandidates'] ? data['remoteIceCandidates'].map(function(data) { // return data['ip']; //}) : [] ); createUAWithStuns(username, server, callback, stunsConfig, makeCallFunc); }).fail(function(data, textStatus, errorThrown) { BBBLog.error("Could not fetch stun/turn servers", {error: textStatus, user: callerIdName, voiceBridge: conferenceVoiceBridge}); callback({'status':'failed', 'errorcode': 1009}); }); }); } function createUAWithStuns(username, server, callback, stunsConfig, makeCallFunc) { console.log("Creating new user agent"); /* VERY IMPORTANT * - You must escape the username because spaces will cause the connection to fail * - We are connecting to the websocket through an nginx redirect instead of directly to 5066 */ var configuration = { uri: 'sip:' + encodeURIComponent(username) + '@' + server, wsServers: ('https:' == document.location.protocol ? 'wss://' : 'ws://') + server + '/ws', displayName: username, register: false, traceSip: true, autostart: false, userAgentString: "BigBlueButton", stunServers: stunsConfig['stunServers'], turnServers: stunsConfig['turnServers'], //artificialRemoteIceCandidates: stunsConfig['remoteIceCandidates'] }; uaConnected = false; userAgent = new SIP.UA(configuration); setUserAgentListeners(callback, makeCallFunc); userAgent.start(); }; function setUserAgentListeners(callback, makeCallFunc) { console.log("resetting UA callbacks"); userAgent.removeAllListeners('connected'); userAgent.on('connected', function() { uaConnected = true; callback({'status':'succeded'}); makeCallFunc(); }); userAgent.removeAllListeners('disconnected'); userAgent.on('disconnected', function() { if (userAgent) { if (userAgent != null) { var userAgentTemp = userAgent; userAgent = null; userAgentTemp.stop(); } if (uaConnected) { callback({'status':'failed', 'errorcode': 1001}); // WebSocket disconnected } else { callback({'status':'failed', 'errorcode': 1002}); // Could not make a WebSocket connection } } }); }; function getUserMicMedia(getUserMicMediaSuccess, getUserMicMediaFailure) { if (userMicMedia == undefined) { if (SIP.WebRTC.isSupported()) { SIP.WebRTC.getUserMedia({audio:true, video:false}, getUserMicMediaSuccess, getUserMicMediaFailure); } else { console.log("getUserMicMedia: webrtc not supported"); getUserMicMediaFailure("WebRTC is not supported"); } } else { console.log("getUserMicMedia: mic already set"); getUserMicMediaSuccess(userMicMedia); } }; function webrtc_call(username, voiceBridge, callback, isListenOnly) { if (!isWebRTCAvailable()) { callback({'status': 'failed', 'errorcode': 1003}); // Browser version not supported return; } if (isListenOnly == null) { // assume NOT listen only unless otherwise stated isListenOnly = false; } var server = window.document.location.hostname; console.log("user " + username + " calling to " + voiceBridge); var makeCallFunc = function() { // only make the call when both microphone and useragent have been created // for listen only, stating listen only is a viable substitute for acquiring user media control if ((isListenOnly||userMicMedia) && userAgent) make_call(username, voiceBridge, server, callback, false, isListenOnly); }; // Reset userAgent so we can successfully switch between listenOnly and listen+speak modes userAgent = null; if (!userAgent) { createUA(username, server, callback, makeCallFunc); } // if the user requests to proceed as listen only (does not require media) or media is already acquired, // proceed with making the call if (isListenOnly || userMicMedia !== undefined) { makeCallFunc(); } else { callback({'status':'mediarequest'}); getUserMicMedia(function(stream) { console.log("getUserMicMedia: success"); userMicMedia = stream; callback({'status':'mediasuccess'}); makeCallFunc(); }, function(e) { console.error("getUserMicMedia: failure - " + e); callback({'status':'mediafail', 'cause': e}); } ); } } function make_call(username, voiceBridge, server, callback, recall, isListenOnly) { if (isListenOnly == null) { isListenOnly = false; } if (userAgent == null) { console.log("userAgent is still null. Delaying call"); var callDelayTimeout = setTimeout( function() { make_call(username, voiceBridge, server, callback, recall, isListenOnly); }, 100); return; } if (!userAgent.isConnected()) { console.log("Trying to make call, but UserAgent hasn't connected yet. Delaying call"); userAgent.once('connected', function() { console.log("UserAgent has now connected, retrying the call"); make_call(username, voiceBridge, server, callback, recall, isListenOnly); }); return; } if (currentSession) { console.log('Active call detected ignoring second make_call'); return; } // Make an audio/video call: console.log("Setting options.. "); var options = {}; if (isListenOnly) { // create necessary options for a listen only stream var stream = null; // handle the web browser // create a stream object through the browser separated from user media if (typeof webkitMediaStream !== 'undefined') { // Google Chrome stream = new webkitMediaStream; } else { // Firefox audioContext = new window.AudioContext; stream = audioContext.createMediaStreamDestination().stream; } options = { media: { stream: stream, // use the stream created above constraints: { audio: true, video: false }, render: { remote: document.getElementById('remote-media') } }, // a list of our RTC Connection constraints RTCConstraints: { // our constraints are mandatory. We must received audio and must not receive audio mandatory: { OfferToReceiveAudio: true, OfferToReceiveVideo: false } } }; } else { options = { media: { stream: userMicMedia, constraints: { audio: true, video: false }, render: { remote: document.getElementById('remote-media') } } }; } callTimeout = setTimeout(function() { console.log('Ten seconds without updates sending timeout code'); callback({'status':'failed', 'errorcode': 1006}); // Failure on call currentSession = null; if (userAgent != null) { var userAgentTemp = userAgent; userAgent = null; userAgentTemp.stop(); } }, 10000); callActive = false; callICEConnected = false; callPurposefullyEnded = false; callFailCounter = 0; console.log("Calling to " + voiceBridge + "...."); currentSession = userAgent.invite('sip:' + voiceBridge + '@' + server, options); // Only send the callback if it's the first try if (recall === false) { console.log('call connecting'); callback({'status':'connecting'}); } else { console.log('call connecting again'); } /* iceGatheringTimeout = setTimeout(function() { console.log('Thirty seconds without ICE gathering finishing'); callback({'status':'failed', 'errorcode': 1011}); // ICE Gathering Failed currentSession = null; if (userAgent != null) { var userAgentTemp = userAgent; userAgent = null; userAgentTemp.stop(); } }, 30000); */ currentSession.mediaHandler.on('iceGatheringComplete', function() { clearTimeout(iceGatheringTimeout); }); // The connecting event fires before the listener can be added currentSession.on('connecting', function(){ clearTimeout(callTimeout); }); currentSession.on('progress', function(response){ console.log('call progress: ' + response); clearTimeout(callTimeout); }); currentSession.on('failed', function(response, cause){ console.log('call failed with cause: '+ cause); if (currentSession) { if (callActive === false) { callback({'status':'failed', 'errorcode': 1004, 'cause': cause}); // Failure on call currentSession = null; if (userAgent != null) { var userAgentTemp = userAgent; userAgent = null; userAgentTemp.stop(); } } else { callActive = false; //currentSession.bye(); currentSession = null; if (userAgent != null) { userAgent.stop(); } } } clearTimeout(callTimeout); }); currentSession.on('bye', function(request){ callActive = false; if (currentSession) { console.log('call ended ' + currentSession.endTime); if (callPurposefullyEnded === true) { callback({'status':'ended'}); } else { callback({'status':'failed', 'errorcode': 1005}); // Call ended unexpectedly } clearTimeout(callTimeout); currentSession = null; } else { console.log('bye event already received'); } }); currentSession.on('cancel', function(request) { callActive = false; if (currentSession) { console.log('call canceled'); clearTimeout(callTimeout); currentSession = null; } else { console.log('cancel event already received'); } }); currentSession.on('accepted', function(data){ callActive = true; console.log('BigBlueButton call accepted'); if (callICEConnected === true) { callback({'status':'started'}); } else { callback({'status':'waitingforice'}); console.log('Waiting for ICE negotiation'); iceConnectedTimeout = setTimeout(function() { console.log('5 seconds without ICE finishing'); callback({'status':'failed', 'errorcode': 1010}); // ICE negotiation timeout currentSession = null; if (userAgent != null) { var userAgentTemp = userAgent; userAgent = null; userAgentTemp.stop(); } }, 5000); } clearTimeout(callTimeout); }); currentSession.mediaHandler.on('iceConnectionFailed', function() { console.log('received ice negotiation failed'); callback({'status':'failed', 'errorcode': 1007}); // Failure on call currentSession = null; clearTimeout(iceConnectedTimeout); if (userAgent != null) { var userAgentTemp = userAgent; userAgent = null; userAgentTemp.stop(); } clearTimeout(callTimeout); }); // Some browsers use status of 'connected', others use 'completed', and a couple use both currentSession.mediaHandler.on('iceConnectionConnected', function() { console.log('Received ICE status changed to connected'); if (callICEConnected === false) { callICEConnected = true; clearTimeout(iceConnectedTimeout); if (callActive === true) { callback({'status':'started'}); } clearTimeout(callTimeout); } }); currentSession.mediaHandler.on('iceConnectionCompleted', function() { console.log('Received ICE status changed to completed'); if (callICEConnected === false) { callICEConnected = true; clearTimeout(iceConnectedTimeout); if (callActive === true) { callback({'status':'started'}); } clearTimeout(callTimeout); } }); } function webrtc_hangup(callback) { callPurposefullyEnded = true; console.log("Hanging up current session"); if (callback) { currentSession.on('bye', callback); } try { currentSession.bye(); } catch (err) { console.log("Forcing to cancel current session"); currentSession.cancel(); } } function isWebRTCAvailable() { if (bowser.msedge) { return false; } else { return SIP.WebRTC.isSupported(); } } function getCallStatus() { return currentSession; }