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Merge pull request #435 from mconf/improving-client-checks

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Client check - improvements to the webrtc related tests
This commit is contained in:
Fred Dixon 2014-11-16 17:06:02 -05:00
commit 298a0e4525
8 changed files with 152 additions and 11127 deletions
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2
bbb-client-check/.gitignore vendored
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@ -7,3 +7,5 @@ org.eclipse.ltk.core.refactoring.prefs
FlexPrettyPrintCommand.prefs
index.template.html
conf/config.xml
resources/lib/bbb_webrtc_bridge_sip.js
resources/lib/sip.js

5
bbb-client-check/build.xml
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@ -103,6 +103,11 @@
</target>
<target name="Resolve-Dependency"
description="Generate HTML wrapper">
<copy todir="resources/lib/" >
<fileset file="../bigbluebutton-client/resources/prod/lib/bbb_webrtc_bridge_sip.js" />
<fileset file="../bigbluebutton-client/resources/prod/lib/sip.js" />
</copy>
<get src="${TEST_IMAGE_URL}" dest="${html.output}/test_image.jpg" skipexisting="true" />
<copy file="html-template/index.html"
tofile="${html.output}/index.html"/>

3
bbb-client-check/html-template/index.html
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@ -35,7 +35,8 @@
<!-- END Browser History required section -->
<script type="text/javascript" src="resources/lib/api-bridge.js"></script>
<script type="text/javascript" src="resources/lib/sip-0.6.2.js"></script>
<script type="text/javascript" src="resources/lib/sip.js"></script>
<script type="text/javascript" src="resources/lib/bbb_webrtc_bridge_sip.js"></script>
<script type="text/javascript" src="resources/lib/deployJava.js"></script>
<script type="text/javascript" src="swfobject.js"></script>
<script type="text/javascript">

5
bbb-client-check/html-template/index.template.html
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@ -34,8 +34,10 @@
<script type="text/javascript" src="history/history.js"></script>
<!-- END Browser History required section -->
<script type="text/javascript" src="resources/lib/sip-0.6.2.js"></script>
<script type="text/javascript" src="resources/lib/api-bridge.js"></script>
<script type="text/javascript" src="resources/lib/sip.js"></script>
<script type="text/javascript" src="resources/lib/bbb_webrtc_bridge_sip.js"></script>
<script type="text/javascript" src="resources/lib/deployJava.js"></script>
<script type="text/javascript" src="swfobject.js"></script>
<script type="text/javascript">
// For version detection, set to min. required Flash Player version, or 0 (or 0.0.0), for no version detection.
@ -62,6 +64,7 @@
</script>
</head>
<body>
<div id="deployJavaPluginContainer" style="visibility:hidden; height:0px; "></div>
<!-- SWFObject's dynamic embed method replaces this alternative HTML content with Flash content when enough
JavaScript and Flash plug-in support is available. The div is initially hidden so that it doesn't show
when JavaScript is disabled.

226
bbb-client-check/resources/lib/api-bridge.js
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@ -1,5 +1,6 @@
{
var BBBClientCheck = {};
var BBB = {};
var userAgent;
var userMicMedia;
var currentSession;
@ -184,152 +185,121 @@
}
BBBClientCheck.isWebRTCSupported = function() {
var isWebRTCSupportedInfo = SIP.WebRTC.isSupported();
var isWebRTCSupportedInfo = isWebRTCAvailable();
var swfObj = getSwfObj();
swfObj.isWebRTCSupported(isWebRTCSupportedInfo);
}
BBBClientCheck.webRTCEchoAndSocketTest = function() {
startWebRTCAudioTest();
}
function startWebRTCAudioTest() {
console.log("Starting WebRTC audio test...");
var swfObj = getSwfObj();
var callback = function(message) {
switch(message.status) {
case 'websocketFailed':
console.log("websocketFailed");
swfObj.webRTCSocketTest(false, message.cause);
break;
case 'websocketSuccess':
console.log("websocketSuccess");
swfObj.webRTCSocketTest(true, 'Connected');
break;
case 'failed':
swfObj.webRTCEchoTest(false, message.cause);
console.log("call failed");
break;
case 'ended':
console.log("call ended");
break;
case 'started':
console.log("call started");
swfObj.webRTCEchoTest(true, 'Connected');
break;
case 'mediasuccess':
console.log("call mediasuccess");
break;
case 'mediafail':
console.log("call mediafail");
break;
}
}
var callerIdName = "12345" + "-bbbID-" + "bbbTestUser";
webrtc_call(callerIdName, "9196", callback);
}
function createUA(username, server, callback) {
/* 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: 'ws://' + server + '/ws',
displayName: username,
register: false,
traceSip: false,
userAgentString: "BigBlueButton",
stunServers: "stun:stun.freeswitch.org"
};
console.log("Creating SIP.UA");
userAgent = new SIP.UA(configuration);
userAgent.on('disconnected', function() {
if (userAgent !== undefined) {
userAgent.stop();
userAgent = null;
callback({status: 'websocketFailed', cause: 'Could not make a WebSocket Connection'});
}
});
userAgent.on('connected', function() {
callback({status: 'websocketSuccess'});
});
userAgent.start();
}
function make_call(username, voiceBridge, server, callback) {
var audioContext = new (window.AudioContext || window.webkitAudioContext)();
var silentStream = audioContext.createMediaStreamDestination().stream;
console.log("Setting options.. ");
var options = {
media: {
stream: silentStream,
render: {
remote: {
audio: document.getElementById('remote-media')
}
}
}
};
userMicMedia = silentStream;
console.log("Calling to " + voiceBridge + "....");
currentSession = userAgent.invite('sip:' + voiceBridge + '@' + server, options);
startWebRTCAudioTest();
}
console.log("Call connecting...");
function sendWebRTCEchoTestAnswer(success, errorcode=undefined) {
var swfObj = getSwfObj();
swfObj.webRTCEchoTest(success, errorcode);
currentSession.on('failed', function(response, cause) {
console.log('call failed with the case ' + casuse);
callback({status: 'failed', cause: cause});
});
currentSession.on('bye', function(request) {
console.log('call ended ' + currentSession.endTime);
callback({status: 'ended'})
});
currentSession.on('accepted', function(data) {
console.log('BigBlueClient Test Call started');
callback({status: 'started'});
webrtc_hangup(function() {
console.log("[BBBClientCheck] Handling webRTC hangup callback");
var userAgentTemp = userAgent;
userAgent = null;
userAgentTemp.stop();
});
}
function webrtc_call(username, voiceBridge, callback) {
console.log("webrtc_call started...");
if (!SIP.WebRTC.isSupported()) {
callback({status: "failed", cause: "Browser version not supported" });
BBB.getMyUserInfo = function(callback) {
var obj = {
myUserID: "12345",
myUsername: "bbbTestUser",
myAvatarURL: "undefined",
myRole: "undefined",
amIPresenter: "undefined",
dialNumber: "undefined",
voiceBridge: "undefined",
customdata: "undefined"
}
var server = window.document.location.host;
console.log("webrtc_call server: " + server);
if(userAgent == undefined) {
createUA(username, server, callback);
}
else {
callback({status: 'websocketSuccess'});
}
make_call(username, voiceBridge, server, callback);
callback(obj);
}
function getUserMicMedia(getUserMediaSuccess, getUserMicMediaFail) {
if (userMicMedia == undefined) {
SIP.WebRTC.getUserMedia({audio:true, video:false}, getUserMediaSuccess, getUserMicMediaFail);
} else {
getUserMicMediaSuccess(userMicMedia);
}
// webrtc test callbacks
BBB.webRTCEchoTestFailed = function(errorcode) {
console.log("[BBBClientCheck] Handling webRTCEchoTestFailed");
sendWebRTCEchoTestAnswer(false, errorcode);
}
BBB.webRTCEchoTestEnded = function() {
console.log("[BBBClientCheck] Handling webRTCEchoTestEnded");
}
BBB.webRTCEchoTestStarted = function() {
console.log("[BBBClientCheck] Handling webRTCEchoTestStarted");
sendWebRTCEchoTestAnswer(true, 'Connected');
}
BBB.webRTCEchoTestConnecting = function() {
console.log("[BBBClientCheck] Handling webRTCEchoTestConnecting");
}
BBB.webRTCEchoTestWaitingForICE = function() {
console.log("[BBBClientCheck] Handling webRTCEchoTestWaitingForICE");
}
BBB.webRTCEchoTestWebsocketSucceeded = function() {
console.log("[BBBClientCheck] Handling webRTCEchoTestWebsocketSucceeded");
var swfObj = getSwfObj();
swfObj.webRTCSocketTest(true, 'Connected');
}
BBB.webRTCEchoTestWebsocketFailed = function(errorcode) {
console.log("[BBBClientCheck] Handling webRTCEchoTestWebsocketFailed");
var swfObj = getSwfObj();
swfObj.webRTCSocketTest(false, errorcode);
}
// webrtc callbacks
BBB.webRTCConferenceCallFailed = function(errorcode) {
console.log("[BBBClientCheck] Handling webRTCConferenceCallFailed");
}
BBB.webRTCConferenceCallEnded = function() {
console.log("[BBBClientCheck] Handling webRTCConferenceCallEnded");
}
BBB.webRTCConferenceCallStarted = function() {
console.log("[BBBClientCheck] Handling webRTCConferenceCallStarted");
}
BBB.webRTCConferenceCallConnecting = function() {
console.log("[BBBClientCheck] Handling webRTCConferenceCallConnecting");
}
BBB.webRTCConferenceCallWaitingForICE = function() {
console.log("[BBBClientCheck] Handling webRTCConferenceCallWaitingForICE");
}
BBB.webRTCConferenceCallWebsocketSucceeded = function() {
console.log("[BBBClientCheck] Handling webRTCConferenceCallWebsocketSucceeded");
}
BBB.webRTCConferenceCallWebsocketFailed = function(errorcode) {
console.log("[BBBClientCheck] Handling webRTCConferenceCallWebsocketFailed");
}
BBB.webRTCMediaRequest = function() {
console.log("[BBBClientCheck] Handling webRTCMediaRequest");
}
BBB.webRTCMediaSuccess = function() {
console.log("[BBBClientCheck] Handling webRTCMediaSuccess");
}
BBB.webRTCMediaFail = function() {
console.log("[BBBClientCheck] Handling webRTCMediaFail");
}
}

10989
bbb-client-check/resources/lib/sip-0.6.2.js
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20
bigbluebutton-client/resources/prod/lib/bbb_api_bridge.js
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@ -462,6 +462,16 @@
}
}
BBB.webRTCConferenceCallWebsocketSucceeded = function() {
}
BBB.webRTCConferenceCallWebsocketFailed = function(errorcode) {
var swfObj = getSwfObj();
if (swfObj) {
swfObj.webRTCConferenceCallFailed(errorcode);
}
}
BBB.webRTCEchoTestStarted = function() {
var swfObj = getSwfObj();
if (swfObj) {
@ -498,6 +508,16 @@
}
}
BBB.webRTCEchoTestWebsocketSucceeded = function() {
}
BBB.webRTCEchoTestWebsocketFailed = function(reason) {
var swfObj = getSwfObj();
if (swfObj) {
swfObj.webRTCEchoTestFailed(reason);
}
}
BBB.webRTCMediaRequest = function() {
var swfObj = getSwfObj();
if (swfObj) {

17
bigbluebutton-client/resources/prod/lib/bbb_webrtc_bridge_sip.js
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@ -50,6 +50,12 @@ function joinWebRTCVoiceConference() {
case 'mediafail':
BBB.webRTCMediaFail();
break;
case 'websocketSucceded':
BBB.webRTCConferenceCallWebsocketSucceeded();
break;
case 'websocketFailed':
BBB.webRTCConferenceCallWebsocketFailed(message.errorcode);
break;
}
}
@ -90,6 +96,12 @@ function startWebRTCAudioTest(){
case 'mediafail':
BBB.webRTCMediaFail();
break;
case 'websocketSucceded':
BBB.webRTCEchoTestWebsocketSucceeded();
break;
case 'websocketFailed':
BBB.webRTCEchoTestWebsocketFailed(message.errorcode);
break;
}
}
@ -174,6 +186,7 @@ function createUA(username, server, callback) {
userAgent = new SIP.UA(configuration);
userAgent.on('connected', function() {
uaConnected = true;
callback({'status':'websocketSucceded'});
});
userAgent.on('disconnected', function() {
if (userAgent) {
@ -181,9 +194,9 @@ function createUA(username, server, callback) {
userAgent = null;
if (uaConnected) {
callback({'status':'failed', 'errorcode': 1001}); // WebSocket disconnected
callback({'status':'websocketFailed', 'errorcode': 1001}); // WebSocket disconnected
} else {
callback({'status':'failed', 'errorcode': 1002}); // Could not make a WebSocket connection
callback({'status':'websocketFailed', 'errorcode': 1002}); // Could not make a WebSocket connection
}
}
});