Merge branch 'investigate-ping-pong-timers' into upgrade-to-red5-nov-4-2014
This commit is contained in:
Richard Alam
commit
c30604bb3f
@ -19,6 +19,11 @@
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package org.bigbluebutton.voiceconf.red5.media.transcoder;
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import java.util.Random;
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import java.util.concurrent.BlockingQueue;
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import java.util.concurrent.Executor;
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import java.util.concurrent.Executors;
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import java.util.concurrent.LinkedBlockingQueue;
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import java.util.concurrent.TimeUnit;
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import org.bigbluebutton.voiceconf.red5.media.FlashToSipAudioStream.TranscodedAudioListener;
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import org.red5.app.sip.codecs.Codec;
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@ -36,9 +41,16 @@ public class SpeexFlashToSipTranscoderImp implements FlashToSipTranscoder {
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private Codec audioCodec;
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private long timestamp = 0;
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private final static int TS_INCREMENT = 320; // Determined from PCAP traces.
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private final Executor exec = Executors.newSingleThreadExecutor();
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private Runnable audioDataProcessor;
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private volatile boolean processAudioData = false;
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private BlockingQueue<SpeexRtpAudioData> audioDataQ;
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private TranscodedAudioListener transcodedAudioListener;
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public SpeexFlashToSipTranscoderImp(Codec audioCodec) {
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audioDataQ = new LinkedBlockingQueue<SpeexRtpAudioData>();
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this.audioCodec = audioCodec;
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Random rgen = new Random();
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timestamp = rgen.nextInt(1000);
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@ -49,7 +61,13 @@ public class SpeexFlashToSipTranscoderImp implements FlashToSipTranscoder {
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// Just copy the audio data removing the codec id which is the first-byte
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// represented by the startOffset var.
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System.arraycopy(audioData, startOffset, transcodedAudio, 0, length);
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transcodedAudioListener.handleTranscodedAudioData(transcodedAudio, timestamp += TS_INCREMENT);
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SpeexRtpAudioData srad = new SpeexRtpAudioData(transcodedAudio, timestamp += TS_INCREMENT);
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try {
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audioDataQ.offer(srad, 100, TimeUnit.MILLISECONDS);
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} catch (InterruptedException e) {
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log.warn("Failed to add speex audio data into queue.");
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}
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}
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public int getCodecId() {
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@ -73,14 +91,33 @@ public class SpeexFlashToSipTranscoderImp implements FlashToSipTranscoder {
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public void setTranscodedAudioListener(TranscodedAudioListener transcodedAudioListener) {
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this.transcodedAudioListener = transcodedAudioListener;
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}
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@Override
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public void start() {
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// do nothing. just implement the interface.
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private void processAudioData() {
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while (processAudioData) {
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SpeexRtpAudioData srad;
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try {
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srad = audioDataQ.take();
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transcodedAudioListener.handleTranscodedAudioData(srad.audioData, srad.timestamp);
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} catch (InterruptedException e) {
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// TODO Auto-generated catch block
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e.printStackTrace();
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}
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}
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}
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@Override
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public void start() {
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processAudioData = true;
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audioDataProcessor = new Runnable() {
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public void run() {
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processAudioData();
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}
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};
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exec.execute(audioDataProcessor);
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}
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@Override
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public void stop() {
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// do nothing. just implement the interface.
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}
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public void stop() {
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processAudioData = false;
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}
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}
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@ -0,0 +1,12 @@
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package org.bigbluebutton.voiceconf.red5.media.transcoder;
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public class SpeexRtpAudioData {
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public final byte[] audioData;
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public final long timestamp;
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public SpeexRtpAudioData(byte[] audioData, long timestamp) {
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this.audioData = audioData;
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this.timestamp = timestamp;
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}
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}
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@ -36,6 +36,7 @@ bbb.micSettings.comboMicList.toolTip = Select a microphone
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bbb.micSettings.micRecordVolume.label = Gain
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bbb.micSettings.micRecordVolume.toolTip = Set your microphone gain
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bbb.micSettings.nextButton = Next
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bbb.micSettings.nextButton.toolTip = Start the echo test
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bbb.micSettings.join = Join Audio
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bbb.micSettings.join.toolTip = Join the audio conference
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bbb.micSettings.cancel = Cancel
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@ -49,8 +50,8 @@ bbb.micSettings.webrtc.capableBrowser = Your browser supports WebRTC.
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bbb.micSettings.webrtc.capableBrowser.dontuseit = Click not to use WebRTC
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bbb.micSettings.webrtc.capableBrowser.dontuseit.toolTip = Click here if you don't want to use the WebRTC technology (recommended if you have problems using it).
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bbb.micSettings.webrtc.notCapableBrowser = WebRTC is not supported in your browser. Please use Google Chrome (version 32 or greater); or Mozilla Firefox (version 26 or greater). You will still be able to join the voice conference using the Adobe Flash Platform.
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bbb.micSettings.webrtc.connectingtoecho = Connecting
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bbb.micSettings.webrtc.connectingtocall = Connecting to conference
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bbb.micSettings.webrtc.connecting = Calling
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bbb.micSettings.webrtc.waitingforice = Connecting
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bbb.micSettings.webrtc.endingecho = Joining audio
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bbb.micSettings.webrtc.endedecho = Echo test ended.
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bbb.micPermissions.firefox.title = Firefox Microphone Permissions
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@ -151,6 +152,7 @@ bbb.presentation.fitToWidth.toolTip = Fit Presentation To Width
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bbb.presentation.fitToPage.toolTip = Fit Presentation To Page
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bbb.presentation.uploadPresBtn.toolTip = Upload Presentation
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bbb.presentation.backBtn.toolTip = Previous slide
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bbb.presentation.btnSlideNum.accessibilityName = Slide {0} of {1}
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bbb.presentation.btnSlideNum.toolTip = Select a slide
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bbb.presentation.forwardBtn.toolTip = Next slide
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bbb.presentation.maxUploadFileExceededAlert = Error: The file is bigger than what's allowed.
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@ -448,6 +448,13 @@
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}
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}
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BBB.webRTCConferenceCallWaitingForICE = function() {
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var swfObj = getSwfObj();
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if (swfObj) {
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swfObj.webRTCConferenceCallWaitingForICE();
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}
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}
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BBB.webRTCCallProgressCallback = function(progress) {
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var swfObj = getSwfObj();
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if (swfObj) {
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@ -475,6 +482,14 @@
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swfObj.webRTCEchoTestFailed(reason);
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}
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}
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BBB.webRTCEchoTestWaitingForICE = function() {
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var swfObj = getSwfObj();
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if (swfObj) {
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swfObj.webRTCEchoTestWaitingForICE();
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}
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}
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BBB.webRTCEchoTestEnded = function() {
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var swfObj = getSwfObj();
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@ -1,5 +1,5 @@
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var callerIdName, conferenceVoiceBridge, userAgent, userMicMedia, userWebcamMedia, currentSession, callTimeout, callActive, callFailCounter, callPurposefullyEnded, uaConnected;
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var callerIdName, conferenceVoiceBridge, userAgent, userMicMedia, userWebcamMedia, currentSession, callTimeout, callActive, callICEConnected, callFailCounter, callPurposefullyEnded, uaConnected;
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function callIntoConference(voiceBridge, callback) {
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if (!callerIdName) {
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@ -38,6 +38,9 @@ function joinWebRTCVoiceConference() {
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case 'connecting':
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BBB.webRTCConferenceCallConnecting();
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break;
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case 'waitingforice':
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BBB.webRTCConferenceCallWaitingForICE();
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break;
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case 'mediarequest':
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BBB.webRTCMediaRequest();
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break;
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@ -75,6 +78,9 @@ function startWebRTCAudioTest(){
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case 'connecting':
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BBB.webRTCEchoTestConnecting();
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break;
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case 'waitingforice':
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BBB.webRTCEchoTestWaitingForICE();
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break;
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case 'mediarequest':
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BBB.webRTCMediaRequest();
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break;
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@ -281,6 +287,7 @@ function make_call(username, voiceBridge, server, callback, recall) {
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}, 10000);
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callActive = false;
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callICEConnected = false;
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callPurposefullyEnded = false;
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callFailCounter = 0;
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console.log("Calling to " + voiceBridge + "....");
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@ -340,9 +347,13 @@ function make_call(username, voiceBridge, server, callback, recall) {
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});
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currentSession.on('accepted', function(data){
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callActive = true;
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console.log('BigBlueButton call accepted');
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console.log('BigBlueButton call started');
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callback({'status':'started'});
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if (callICEConnected === true) {
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callback({'status':'started'});
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} else {
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callback({'status':'waitingforice'});
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}
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clearTimeout(callTimeout);
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});
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currentSession.mediaHandler.on('iceFailed', function() {
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@ -355,6 +366,30 @@ function make_call(username, voiceBridge, server, callback, recall) {
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clearTimeout(callTimeout);
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});
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// Some browsers use status of 'connected', others use 'completed', and a couple use both
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currentSession.mediaHandler.on('iceConnected', function() {
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console.log('Received ICE status changed to connected');
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if (callICEConnected === false) {
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callICEConnected = true;
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if (callActive === true) {
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callback({'status':'started'});
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}
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clearTimeout(callTimeout);
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}
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});
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currentSession.mediaHandler.on('iceCompleted', function() {
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console.log('Received ICE status changed to completed');
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if (callICEConnected === false) {
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callICEConnected = true;
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if (callActive === true) {
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callback({'status':'started'});
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}
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clearTimeout(callTimeout);
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}
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});
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}
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function webrtc_hangup(callback) {
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@ -1,5 +1,5 @@
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/*
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* SIP version 0.6.2
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* SIP version 0.6.3
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* Copyright (c) 2014-2014 Junction Networks, Inc <http://www.onsip.com>
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* Homepage: http://sipjs.com
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* License: http://sipjs.com/license/
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@ -37,7 +37,7 @@ module.exports={
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"name": "sip.js",
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"title": "SIP.js",
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"description": "A simple, intuitive, and powerful JavaScript signaling library",
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"version": "0.6.2",
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"version": "0.6.3",
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"main": "src/index.js",
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"browser": {
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"./src/Grammar/index.js": "./src/Grammar/dist/Grammar.js",
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@ -2483,12 +2483,6 @@ var Hacks = module.exports = {
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return typeof mozRTCPeerConnection !== 'undefined';
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},
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cannotHandleRelayCandidates: function (message) {
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if (this.isFirefox() && message.body) {
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message.body = message.body.replace(/relay/g, 'host generation 0');
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}
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},
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cannotHandleExtraWhitespace: function (message) {
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if (this.isFirefox() && message.body) {
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message.body = message.body.replace(/ \r\n/g, "\r\n");
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@ -2532,6 +2526,15 @@ var Hacks = module.exports = {
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}
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}
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return sdp;
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},
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hackCLinInIP: function (sdp) {
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/* Starting in Firefox 34 they have set the c line to 0.0.0.0 which
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* means "hold" according to legacy SIP standards and Freeswitch
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* interprets it according to the SIP standards. We replace the
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* 0.0.0.0 with any other IP so that the call continues.
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*/
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return sdp.replace("c=IN IP4 0.0.0.0", "c=IN IP4 127.0.0.1");
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}
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},
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@ -3524,7 +3527,7 @@ _dereq_('./EventEmitter')(SIP);
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SIP.C = _dereq_('./Constants')(SIP.name, SIP.version);
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SIP.Exceptions = _dereq_('./Exceptions');
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SIP.Timers = _dereq_('./Timers')(environment.timers);
|
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_dereq_('./Transport')(SIP, environment.WebSocket);
|
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SIP.Transport = environment.Transport(SIP, environment.WebSocket);
|
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_dereq_('./Parser')(SIP);
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_dereq_('./SIPMessage')(SIP);
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_dereq_('./URI')(SIP);
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@ -3548,7 +3551,7 @@ SIP.Grammar = _dereq_('./Grammar')(SIP);
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return SIP;
|
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};
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|
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},{"../package.json":1,"./ClientContext":2,"./Constants":3,"./Dialogs":5,"./DigestAuthentication":6,"./EventEmitter":7,"./Exceptions":8,"./Grammar":9,"./Hacks":11,"./LoggerFactory":12,"./MediaHandler":13,"./NameAddrHeader":14,"./Parser":15,"./RegisterContext":16,"./RequestSender":17,"./SIPMessage":19,"./SanityCheck":20,"./ServerContext":21,"./Session":22,"./Subscription":24,"./Timers":25,"./Transactions":26,"./Transport":27,"./UA":28,"./URI":29,"./Utils":30,"./WebRTC":31}],19:[function(_dereq_,module,exports){
|
||||
},{"../package.json":1,"./ClientContext":2,"./Constants":3,"./Dialogs":5,"./DigestAuthentication":6,"./EventEmitter":7,"./Exceptions":8,"./Grammar":9,"./Hacks":11,"./LoggerFactory":12,"./MediaHandler":13,"./NameAddrHeader":14,"./Parser":15,"./RegisterContext":16,"./RequestSender":17,"./SIPMessage":19,"./SanityCheck":20,"./ServerContext":21,"./Session":22,"./Subscription":24,"./Timers":25,"./Transactions":26,"./UA":28,"./URI":29,"./Utils":30,"./WebRTC":31}],19:[function(_dereq_,module,exports){
|
||||
/**
|
||||
* @fileoverview SIP Message
|
||||
*/
|
||||
@ -5346,7 +5349,6 @@ InviteServerContext = function(ua, request) {
|
||||
}
|
||||
|
||||
//TODO: move this into media handler
|
||||
SIP.Hacks.Firefox.cannotHandleRelayCandidates(request);
|
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SIP.Hacks.Firefox.cannotHandleExtraWhitespace(request);
|
||||
SIP.Hacks.AllBrowsers.maskDtls(request);
|
||||
|
||||
@ -5811,7 +5813,6 @@ InviteServerContext.prototype = {
|
||||
if (!this.hasAnswer) {
|
||||
if(request.body && request.getHeader('content-type') === 'application/sdp') {
|
||||
// ACK contains answer to an INVITE w/o SDP negotiation
|
||||
SIP.Hacks.Firefox.cannotHandleRelayCandidates(request);
|
||||
SIP.Hacks.Firefox.cannotHandleExtraWhitespace(request);
|
||||
SIP.Hacks.AllBrowsers.maskDtls(request);
|
||||
|
||||
@ -6044,6 +6045,9 @@ InviteClientContext.prototype = {
|
||||
if (self.isCanceled || self.status === C.STATUS_TERMINATED) {
|
||||
return;
|
||||
}
|
||||
|
||||
offer = SIP.Hacks.Firefox.hackCLinInIP(offer);
|
||||
|
||||
self.hasOffer = true;
|
||||
self.request.body = offer;
|
||||
self.status = C.STATUS_INVITE_SENT;
|
||||
@ -6170,7 +6174,6 @@ InviteClientContext.prototype = {
|
||||
return;
|
||||
}
|
||||
|
||||
SIP.Hacks.Firefox.cannotHandleRelayCandidates(response);
|
||||
SIP.Hacks.Firefox.cannotHandleExtraWhitespace(response);
|
||||
SIP.Hacks.AllBrowsers.maskDtls(response);
|
||||
|
||||
@ -6288,7 +6291,6 @@ InviteClientContext.prototype = {
|
||||
break;
|
||||
}
|
||||
|
||||
SIP.Hacks.Firefox.cannotHandleRelayCandidates(response);
|
||||
SIP.Hacks.Firefox.cannotHandleExtraWhitespace(response);
|
||||
SIP.Hacks.AllBrowsers.maskDtls(response);
|
||||
|
||||
@ -8011,7 +8013,7 @@ Transport.prototype = {
|
||||
};
|
||||
|
||||
Transport.C = C;
|
||||
SIP.Transport = Transport;
|
||||
return Transport;
|
||||
};
|
||||
|
||||
},{}],28:[function(_dereq_,module,exports){
|
||||
@ -8892,6 +8894,7 @@ UA.prototype.loadConfig = function(configuration) {
|
||||
// Hacks
|
||||
hackViaTcp: false,
|
||||
hackIpInContact: false,
|
||||
hackWssInTransport: false,
|
||||
|
||||
//autostarting
|
||||
autostart: true,
|
||||
@ -9023,7 +9026,7 @@ UA.prototype.loadConfig = function(configuration) {
|
||||
this.contact = {
|
||||
pub_gruu: null,
|
||||
temp_gruu: null,
|
||||
uri: new SIP.URI('sip', SIP.Utils.createRandomToken(8), settings.viaHost, null, {transport: 'ws'}),
|
||||
uri: new SIP.URI('sip', SIP.Utils.createRandomToken(8), settings.viaHost, null, {transport: ((settings.hackWssInTransport)?'wss':'ws')}),
|
||||
toString: function(options){
|
||||
options = options || {};
|
||||
|
||||
@ -9033,7 +9036,7 @@ UA.prototype.loadConfig = function(configuration) {
|
||||
contact = '<';
|
||||
|
||||
if (anonymous) {
|
||||
contact += (this.temp_gruu || 'sip:anonymous@anonymous.invalid;transport=ws').toString();
|
||||
contact += (this.temp_gruu || ('sip:anonymous@anonymous.invalid;transport='+(settings.hackWssInTransport)?'wss':'ws')).toString();
|
||||
} else {
|
||||
contact += (this.pub_gruu || this.uri).toString();
|
||||
}
|
||||
@ -9103,6 +9106,7 @@ UA.configuration_skeleton = (function() {
|
||||
"displayName",
|
||||
"hackViaTcp", // false.
|
||||
"hackIpInContact", //false
|
||||
"hackWssInTransport", //false
|
||||
"instanceId",
|
||||
"noAnswerTimeout", // 30 seconds.
|
||||
"password",
|
||||
@ -9278,6 +9282,12 @@ UA.configuration_check = {
|
||||
}
|
||||
},
|
||||
|
||||
hackWssInTransport: function(hackWssInTransport) {
|
||||
if (typeof hackWssInTransport === 'boolean') {
|
||||
return hackWssInTransport;
|
||||
}
|
||||
},
|
||||
|
||||
instanceId: function(instanceId) {
|
||||
if(typeof instanceId !== 'string') {
|
||||
return;
|
||||
@ -10247,6 +10257,8 @@ var MediaHandler = function(session, options) {
|
||||
'userMediaFailed',
|
||||
'iceGathering',
|
||||
'iceComplete',
|
||||
'iceConnected',
|
||||
'iceCompleted',
|
||||
'iceFailed',
|
||||
'iceDisconnected',
|
||||
'iceClosed',
|
||||
@ -10285,7 +10297,9 @@ var MediaHandler = function(session, options) {
|
||||
/* Change 'url' to 'urls' whenever this issue is solved:
|
||||
* https://code.google.com/p/webrtc/issues/detail?id=2096
|
||||
*/
|
||||
servers.push({'url': stunServers});
|
||||
[].concat(stunServers).forEach(function (server) {
|
||||
servers.push({'url': server});
|
||||
});
|
||||
|
||||
length = turnServers.length;
|
||||
for (idx = 0; idx < length; idx++) {
|
||||
@ -10348,6 +10362,14 @@ var MediaHandler = function(session, options) {
|
||||
self.emit('iceClosed', this);
|
||||
}
|
||||
|
||||
if (this.iceConnectionState === 'connected') {
|
||||
self.emit('iceConnected', this);
|
||||
}
|
||||
|
||||
if (this.iceConnectionState === 'completed') {
|
||||
self.emit('iceCompleted', this);
|
||||
}
|
||||
|
||||
//Bria state changes are always connected -> disconnected -> connected on accept, so session gets terminated
|
||||
//normal calls switch from failed to connected in some cases, so checking for failed and terminated
|
||||
/*if (this.iceConnectionState === 'failed') {
|
||||
@ -10767,6 +10789,9 @@ MediaStreamManager.render = function render (streams, elements) {
|
||||
}
|
||||
|
||||
function attachAndPlay (element, stream) {
|
||||
if (typeof element === 'function') {
|
||||
element = element();
|
||||
}
|
||||
(environment.attachMediaStream || attachMediaStream)(element, stream);
|
||||
ensureMediaPlaying(element);
|
||||
}
|
||||
@ -10808,7 +10833,7 @@ MediaStreamManager.render = function render (streams, elements) {
|
||||
};
|
||||
|
||||
MediaStreamManager.prototype = Object.create(SIP.EventEmitter.prototype, {
|
||||
'acquire': {value: function acquire (mediaHint) {
|
||||
'acquire': {writable: true, value: function acquire (mediaHint) {
|
||||
mediaHint = Object.keys(mediaHint || {}).length ? mediaHint : this.mediaHint;
|
||||
|
||||
var saveSuccess = function (isHintStream, streams) {
|
||||
@ -10860,7 +10885,7 @@ MediaStreamManager.prototype = Object.create(SIP.EventEmitter.prototype, {
|
||||
}
|
||||
}},
|
||||
|
||||
'release': {value: function release (streams) {
|
||||
'release': {writable: true, value: function release (streams) {
|
||||
streams = [].concat(streams);
|
||||
streams.forEach(function (stream) {
|
||||
var streamId = MediaStreamManager.streamId(stream);
|
||||
@ -10894,6 +10919,7 @@ function getPrefixedProperty (object, name) {
|
||||
|
||||
module.exports = {
|
||||
WebSocket: global.WebSocket,
|
||||
Transport: _dereq_('./Transport'),
|
||||
open: global.open,
|
||||
Promise: global.Promise,
|
||||
timers: global,
|
||||
@ -10917,7 +10943,7 @@ module.exports = {
|
||||
};
|
||||
|
||||
}).call(this,typeof self !== "undefined" ? self : typeof window !== "undefined" ? window : {})
|
||||
},{}],35:[function(_dereq_,module,exports){
|
||||
},{"./Transport":27}],35:[function(_dereq_,module,exports){
|
||||
module.exports = _dereq_('./SIP')(_dereq_('./environment'));
|
||||
|
||||
},{"./SIP":18,"./environment":34}]},{},[35])
|
||||
|
||||
@ -99,10 +99,12 @@ package org.bigbluebutton.main.api
|
||||
ExternalInterface.addCallback("webRTCConferenceCallConnecting", handleWebRTCConferenceCallConnecting);
|
||||
ExternalInterface.addCallback("webRTCConferenceCallEnded", handleWebRTCConferenceCallEnded);
|
||||
ExternalInterface.addCallback("webRTCConferenceCallFailed", handleWebRTCConferenceCallFailed);
|
||||
ExternalInterface.addCallback("webRTCConferenceCallWaitingForICE", handleWebRTCConferenceCallWaitingForICE);
|
||||
ExternalInterface.addCallback("webRTCEchoTestStarted", handleWebRTCEchoTestStarted);
|
||||
ExternalInterface.addCallback("webRTCEchoTestConnecting", handleWebRTCEchoTestConnecting);
|
||||
ExternalInterface.addCallback("webRTCEchoTestFailed", handleWebRTCEchoTestFailed);
|
||||
ExternalInterface.addCallback("webRTCEchoTestEnded", handleWebRTCEchoTestEnded);
|
||||
ExternalInterface.addCallback("webRTCEchoTestWaitingForICE", handleWebRTCEchoTestWaitingForICE);
|
||||
ExternalInterface.addCallback("webRTCMediaRequest", handleWebRTCMediaRequest);
|
||||
ExternalInterface.addCallback("webRTCMediaSuccess", handleWebRTCMediaSuccess);
|
||||
ExternalInterface.addCallback("webRTCMediaFail", handleWebRTCMediaFail);
|
||||
@ -418,6 +420,11 @@ package org.bigbluebutton.main.api
|
||||
trace(LOG + "handleWebRTCConferenceCallFailed: errorCode=[" + errorCode + "]");
|
||||
_dispatcher.dispatchEvent(new WebRTCCallEvent(WebRTCCallEvent.WEBRTC_CALL_FAILED, errorCode));
|
||||
}
|
||||
|
||||
private function handleWebRTCConferenceCallWaitingForICE():void {
|
||||
trace(LOG + "handleWebRTCConferenceCallWaitingForICE: received");
|
||||
_dispatcher.dispatchEvent(new WebRTCCallEvent(WebRTCCallEvent.WEBRTC_CALL_WAITING_FOR_ICE));
|
||||
}
|
||||
|
||||
private function handleWebRTCEchoTestStarted():void {
|
||||
trace(LOG + "handleWebRTCEchoTestStarted: recieved");
|
||||
@ -438,6 +445,11 @@ package org.bigbluebutton.main.api
|
||||
trace(LOG + "handleWebRTCEchoTestEnded: received");
|
||||
_dispatcher.dispatchEvent(new WebRTCEchoTestEvent(WebRTCEchoTestEvent.WEBRTC_ECHO_TEST_ENDED));
|
||||
}
|
||||
|
||||
private function handleWebRTCEchoTestWaitingForICE():void {
|
||||
trace(LOG + "handleWebRTCEchoTestWaitingForICE: received");
|
||||
_dispatcher.dispatchEvent(new WebRTCEchoTestEvent(WebRTCEchoTestEvent.WEBRTC_ECHO_TEST_WAITING_FOR_ICE));
|
||||
}
|
||||
|
||||
private function handleWebRTCMediaRequest():void {
|
||||
trace(LOG + "handleWebRTCMediaRequest: received");
|
||||
|
||||
@ -30,8 +30,10 @@ with BigBlueButton; if not, see <http://www.gnu.org/licenses/>.
|
||||
<mate:Listener type="{WebRTCEchoTestStartedEvent.WEBRTC_ECHO_TEST_STARTED}" method="handleWebRTCEchoTestStartedEvent" />
|
||||
<mate:Listener type="{WebRTCEchoTestEvent.WEBRTC_ECHO_TEST_ENDED}" method="handleWebRTCEchoTestEndedEvent" />
|
||||
<mate:Listener type="{WebRTCEchoTestEvent.WEBRTC_ECHO_TEST_FAILED}" method="handleWebRTCEchoTestFailedEvent" />
|
||||
<mate:Listener type="{WebRTCEchoTestEvent.WEBRTC_ECHO_TEST_WAITING_FOR_ICE}" method="handleWebRTCEchoTestWaitingForICEEvent" />
|
||||
<mate:Listener type="{WebRTCCallEvent.WEBRTC_CALL_CONNECTING}" method="handleWebRTCCallConnectingEvent" />
|
||||
<mate:Listener type="{WebRTCCallEvent.WEBRTC_CALL_FAILED}" method="handleWebRTCCallFailedEvent" />
|
||||
<mate:Listener type="{WebRTCCallEvent.WEBRTC_CALL_WAITING_FOR_ICE}" method="handleWebRTCCallWaitingForICEEvent" />
|
||||
<mate:Listener type="{WebRTCJoinedVoiceConferenceEvent.JOINED_VOICE_CONFERENCE}" method="handleWebRTCCallStartedEvent" />
|
||||
|
||||
<mx:Script>
|
||||
@ -65,7 +67,7 @@ with BigBlueButton; if not, see <http://www.gnu.org/licenses/>.
|
||||
|
||||
private function onCancelClicked():void {
|
||||
JSLog.debug(LOG + "onCancelClicked .");
|
||||
dotTimer.stop();
|
||||
if (dotTimer) dotTimer.stop();
|
||||
PopUpManager.removePopUp(this);
|
||||
}
|
||||
|
||||
@ -75,7 +77,7 @@ with BigBlueButton; if not, see <http://www.gnu.org/licenses/>.
|
||||
|
||||
private function onCreationComplete():void {
|
||||
setCurrentState("connecting");
|
||||
lblConnectMessage.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.connectingtoecho');
|
||||
lblConnectMessage.text = lblConnectMessageMock.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.connecting');
|
||||
dotTimer = new Timer(200, 0);
|
||||
dotTimer.addEventListener(TimerEvent.TIMER, dotAnimate);
|
||||
dotTimer.start();
|
||||
@ -98,10 +100,11 @@ with BigBlueButton; if not, see <http://www.gnu.org/licenses/>.
|
||||
userClosed = true;
|
||||
trace(LOG + "Echo test passed.");
|
||||
JSLog.debug(LOG + "Echo test passed.");
|
||||
setCurrentState("connecting");
|
||||
lblConnectMessage.text = lblConnectMessageMock.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.endingecho');
|
||||
|
||||
var dispatcher:Dispatcher = new Dispatcher();
|
||||
dispatcher.dispatchEvent(new WebRTCEchoTestEvent(WebRTCEchoTestEvent.WEBRTC_ECHO_TEST_HAS_AUDIO));
|
||||
setCurrentState("connecting");
|
||||
lblConnectMessage.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.endingecho');
|
||||
}
|
||||
|
||||
private function noButtonClicked():void {
|
||||
@ -128,7 +131,7 @@ with BigBlueButton; if not, see <http://www.gnu.org/licenses/>.
|
||||
|
||||
private function webRTCEchoTestEnded():void {
|
||||
setCurrentState("connecting");
|
||||
lblConnectMessage.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.endedecho');
|
||||
lblConnectMessage.text = lblConnectMessageMock.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.endedecho');
|
||||
if (!dotTimer.running) dotTimer.start();
|
||||
|
||||
if (!userClosed) {
|
||||
@ -142,15 +145,27 @@ with BigBlueButton; if not, see <http://www.gnu.org/licenses/>.
|
||||
onCancelClicked();
|
||||
}
|
||||
|
||||
private function handleWebRTCEchoTestWaitingForICEEvent(e:WebRTCEchoTestEvent):void {
|
||||
setCurrentState("connecting");
|
||||
lblConnectMessage.text = lblConnectMessageMock.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.waitingforice');
|
||||
if (!dotTimer.running) dotTimer.start();
|
||||
}
|
||||
|
||||
private function handleWebRTCCallConnectingEvent(e:WebRTCCallEvent):void {
|
||||
setCurrentState("connecting");
|
||||
lblConnectMessage.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.connectingtocall');
|
||||
lblConnectMessage.text = lblConnectMessageMock.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.connecting');
|
||||
if (!dotTimer.running) dotTimer.start();
|
||||
}
|
||||
|
||||
private function handleWebRTCCallFailedEvent(e:WebRTCCallEvent):void {
|
||||
onCancelClicked();
|
||||
}
|
||||
|
||||
private function handleWebRTCCallWaitingForICEEvent(e:WebRTCCallEvent):void {
|
||||
setCurrentState("connecting");
|
||||
lblConnectMessage.text = lblConnectMessageMock.text = ResourceUtil.getInstance().getString('bbb.micSettings.webrtc.waitingforice');
|
||||
if (!dotTimer.running) dotTimer.start();
|
||||
}
|
||||
|
||||
private function webRTCCallStarted():void {
|
||||
JSLog.debug(LOG + "webRTCCallStarted.");
|
||||
@ -193,7 +208,10 @@ with BigBlueButton; if not, see <http://www.gnu.org/licenses/>.
|
||||
<mx:State name="connecting">
|
||||
<mx:AddChild relativeTo="cnvTitle" position="after">
|
||||
<mx:HBox width="100%" height="100%" verticalAlign="middle" horizontalAlign="center">
|
||||
<mx:Label id="lblConnectMessage" textAlign="right" styleName="micSettingsWindowSpeakIntoMicLabelStyle" text="" />
|
||||
<mx:TextArea id="lblConnectMessage" editable="false" textAlign="right" borderSkin="{null}"
|
||||
width="{lblConnectMessageMock.width + 4}" height="{lblConnectDots.height}"
|
||||
styleName="micSettingsWindowSpeakIntoMicLabelStyle" />
|
||||
<mx:Text id="lblConnectMessageMock" visible="false" includeInLayout="false" />
|
||||
<mx:Label id="lblConnectDots" width="20" textAlign="left" styleName="micSettingsWindowSpeakIntoMicLabelStyle" text="" />
|
||||
</mx:HBox>
|
||||
</mx:AddChild>
|
||||
|
||||
@ -8,6 +8,7 @@ package org.bigbluebutton.modules.phone.events
|
||||
public static const WEBRTC_CALL_CONNECTING:String = "WEBRTC_CALL_CONNECTING";
|
||||
public static const WEBRTC_CALL_ENDED:String = "WEBRTC_CALL_ENDED";
|
||||
public static const WEBRTC_CALL_FAILED:String = "WEBRTC_CALL_FAILED";
|
||||
public static const WEBRTC_CALL_WAITING_FOR_ICE:String = "WEBRTC_CALL_WAITING_FOR_ICE";
|
||||
|
||||
public var errorCode:Number;
|
||||
|
||||
|
||||
@ -11,6 +11,7 @@ package org.bigbluebutton.modules.phone.events
|
||||
public static const WEBRTC_ECHO_TEST_HAS_AUDIO:String = "WEBRTC_ECHO_TEST_HAS_AUDIO";
|
||||
public static const WEBRTC_ECHO_TEST_NO_AUDIO:String = "WEBRTC_ECHO_TEST_NO_AUDIO";
|
||||
public static const WEBRTC_ECHO_TEST_ENDED_UNEXPECTEDLY:String = "WEBRTC_ECHO_TEST_ENDED_UNEXPECTEDLY";
|
||||
public static const WEBRTC_ECHO_TEST_WAITING_FOR_ICE:String = "WEBRTC_ECHO_TEST_WAITING_FOR_ICE";
|
||||
|
||||
public var errorCode:Number;
|
||||
|
||||
|
||||
Loading…
Reference in New Issue
Block a user