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reverted jssip back to 0.3.7 and fixed chrome webrtc

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This commit is contained in:
Chad Pilkey 2014-07-21 10:25:08 -07:00
parent 930c6d881b
commit 5c79202ade
4 changed files with 19338 additions and 21545 deletions
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24
bigbluebutton-client/resources/prod/BigBlueButton.html
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@ -43,7 +43,7 @@
<script src="lib/bbb_blinker.js" language="javascript"></script>
<script src="lib/bbb_deskshare.js" language="javascript"></script>
<script src="lib/bbb_api_bridge.js" language="javascript"></script>
<script src="lib/jssip-0.4.0-devel.js" language="javascript"></script>
<script src="lib/jssip-0.3.7.js" language="javascript"></script>
<script src="lib/bbb_webrtc_bridge_jssip.js" language="javascript"></script>
<!-- <script src="lib/sipml5api.js" language="javascript"></script> -->
<!-- <script src="lib/bbb_webrtc_bridge_sipml5.js" language="javascript"></script> -->
@ -101,7 +101,7 @@
<script type="text/javascript">
var fsServer = null;
var fsServer = null;
var callIntoConference = function(destination, callback) {
if (!isWebrtcCapable()) {
@ -159,8 +159,8 @@
webrtc_hangup(callback);
}
var startWebrtcAudioTest = function(){
console.log("Testing webrtc audio");
var startWebrtcAudioTest = function(){
console.log("Testing webrtc audio");
var callback = function(message) {
if (message.status == 'failed') {
BBB.webRtcEchoTestFailed(message.cause);
@ -171,15 +171,15 @@
}
}
webrtc_call("tester", "9196", fsServer, callback);
}
webrtc_call("tester", "9196", fsServer, callback);
}
var stopWebrtcAudioTest = function(){
console.log("Stopping webrtc audio test");
webrtc_hangup(function(){
console.log("Webrtc audio test stopped");
});
}
var stopWebrtcAudioTest = function(){
console.log("Stopping webrtc audio test");
webrtc_hangup(function(){
console.log("Webrtc audio test stopped");
});
}
</script>

4
bigbluebutton-client/resources/prod/lib/bbb_webrtc_bridge_jssip.js
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@ -34,7 +34,7 @@ function webrtc_call(username, voiceBridge, server, callback) {
register: false,
// register_expires: null,
// no_answer_timeout: null,
trace_sip: true,
trace_sip: false,
stun_servers: "stun:74.125.134.127:19302",
// turn_servers: null,
// use_preloaded_route: null,
@ -91,7 +91,7 @@ function webrtc_call(username, voiceBridge, server, callback) {
remoteView.src = window.URL.createObjectURL(rtcSession.getRemoteStreams()[0]);
remoteStream = true;
}
callback({'status':'started', 'localStream': localStream, 'remoteStream': remoteStream});
callback({'status':'started', 'localStream': localStream, 'remoteStream': remoteStream});
}
};

40843
bigbluebutton-client/resources/prod/lib/jssip-0.4.0-devel.js → bigbluebutton-client/resources/prod/lib/jssip-0.3.7.js
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File diff suppressed because it is too large Load Diff

12
bigbluebutton-client/src/org/bigbluebutton/modules/phone/managers/WebRtcCallManager.as
View File

@ -2,7 +2,11 @@ package org.bigbluebutton.modules.phone.managers
{
import com.asfusion.mate.events.Dispatcher;
import flash.events.TimerEvent;
import flash.external.ExternalInterface;
import flash.utils.Timer;
import flexlib.scheduling.timelineClasses.TimeRangeDescriptorUtil;
import mx.controls.Alert;
import mx.events.CloseEvent;
@ -107,6 +111,8 @@ package org.bigbluebutton.modules.phone.managers
dispatcher.dispatchEvent(new UseFlashModeCommand());
}
private var t:Timer;
public function handleWebRtcEchoTestHasAudioEvent():void {
state = STOP_ECHO_THEN_JOIN_CONF;
endEchoTest();
@ -115,7 +121,11 @@ package org.bigbluebutton.modules.phone.managers
* after. (richard mar 28, 2014)
*/
//echoTestDone = true;
joinVoiceConference();
t = new Timer(1000, 1);
t.addEventListener(TimerEvent.TIMER, function(e:TimerEvent):void {
joinVoiceConference();
});
t.start();
}
public function handleWebRtcConfCallStartedEvent():void {