third_party/web_platform_tests/webrtc/no-media-call.html - cobalt - Git at Google

 <!doctype html>
 <!--
 This test uses the legacy callback API with no media, and thus does not require fake media devices.
 -->

 <html>
 <head>
   <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
   <title>RTCPeerConnection No-Media Connection Test</title>
 </head>
 <body>
   <div id="log"></div>
   <h2>iceConnectionState info</h2>
   <div id="stateinfo">
   </div>

   <!-- These files are in place when executing on W3C. -->
   <script src="/resources/testharness.js"></script>
   <script src="/resources/testharnessreport.js"></script>
   <script type="text/javascript">
   var test = async_test('Can set up a basic WebRTC call with no data.');

   var gFirstConnection = null;
   var gSecondConnection = null;

   var onOfferCreated = test.step_func(function(offer) {
     gFirstConnection.setLocalDescription(offer, ignoreSuccess,
                                          failed('setLocalDescription first'));

     // This would normally go across the application's signaling solution.
     // In our case, the "signaling" is to call this function.
     receiveCall(offer.sdp);
   });

   function receiveCall(offerSdp) {

     var parsedOffer = new RTCSessionDescription({ type: 'offer',
                                                   sdp: offerSdp });
     // These functions use the legacy interface extensions to RTCPeerConnection.
     gSecondConnection.setRemoteDescription(parsedOffer,
       function() {
         gSecondConnection.createAnswer(onAnswerCreated,
                                        failed('createAnswer'));
       },
       failed('setRemoteDescription second'));
   };

   var onAnswerCreated = test.step_func(function(answer) {
     gSecondConnection.setLocalDescription(answer, ignoreSuccess,
                                           failed('setLocalDescription second'));

     // Similarly, this would go over the application's signaling solution.
     handleAnswer(answer.sdp);
   });

   function handleAnswer(answerSdp) {
     var parsedAnswer = new RTCSessionDescription({ type: 'answer',
                                                    sdp: answerSdp });
     gFirstConnection.setRemoteDescription(parsedAnswer, ignoreSuccess,
                                           failed('setRemoteDescription first'));
   };

   var onIceCandidateToFirst = test.step_func(function(event) {
     // If event.candidate is null = no more candidates.
     if (event.candidate) {
       gSecondConnection.addIceCandidate(event.candidate);
     }
   });

   var onIceCandidateToSecond = test.step_func(function(event) {
     if (event.candidate) {
       gFirstConnection.addIceCandidate(event.candidate);
     }
   });

   var onRemoteStream = test.step_func(function(event) {
     assert_unreached('WebRTC received a stream when there was none');
   });

   var onIceConnectionStateChange = test.step_func(function(event) {
     assert_equals(event.type, 'iceconnectionstatechange');
     assert_not_equals(gFirstConnection.iceConnectionState, "failed", "iceConnectionState of first connection");
     assert_not_equals(gSecondConnection.iceConnectionState, "failed", "iceConnectionState of second connection");
     var stateinfo = document.getElementById('stateinfo');
     stateinfo.innerHTML = 'First: ' + gFirstConnection.iceConnectionState
                         + '<br>Second: ' + gSecondConnection.iceConnectionState;
     // Note: All these combinations are legal states indicating that the
     // call has connected. All browsers should end up in completed/completed,
     // but as of this moment, we've chosen to terminate the test early.
     // TODO: Revise test to ensure completed/completed is reached.
     if (gFirstConnection.iceConnectionState == 'connected' &&
         gSecondConnection.iceConnectionState == 'connected') {
       test.done()
     }
     if (gFirstConnection.iceConnectionState == 'connected' &&
         gSecondConnection.iceConnectionState == 'completed') {
       test.done()
     }
     if (gFirstConnection.iceConnectionState == 'completed' &&
         gSecondConnection.iceConnectionState == 'connected') {
       test.done()
     }
     if (gFirstConnection.iceConnectionState == 'completed' &&
         gSecondConnection.iceConnectionState == 'completed') {
       test.done()
     }
   });

   // Returns a suitable error callback.
   function failed(function_name) {
     return test.step_func(function() {
       assert_unreached('WebRTC called error callback for ' + function_name);
     });
   }

   // Returns a suitable do-nothing.
   function ignoreSuccess(function_name) {
   }

   // This function starts the test.
   test.step(function() {
     gFirstConnection = new RTCPeerConnection(null);
     gFirstConnection.onicecandidate = onIceCandidateToFirst;
     gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange;

     gSecondConnection = new RTCPeerConnection(null);
     gSecondConnection.onicecandidate = onIceCandidateToSecond;
     gSecondConnection.onaddstream = onRemoteStream;
     gSecondConnection.oniceconnectionstatechange = onIceConnectionStateChange;

     // The offerToReceiveVideo is necessary and sufficient to make
     // an actual connection.
     gFirstConnection.createOffer(onOfferCreated, failed('createOffer'),
         {offerToReceiveVideo: true});
   });
 </script>

 </body>
 </html>
	<!doctype html>
	<!--
	This test uses the legacy callback API with no media, and thus does not require fake media devices.
	-->

	<html>
	<head>
	<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
	<title>RTCPeerConnection No-Media Connection Test</title>
	</head>
	<body>
	<div id="log"></div>
	<h2>iceConnectionState info</h2>
	<div id="stateinfo">
	</div>

	<!-- These files are in place when executing on W3C. -->
	<script src="/resources/testharness.js"></script>
	<script src="/resources/testharnessreport.js"></script>
	<script type="text/javascript">
	var test = async_test('Can set up a basic WebRTC call with no data.');

	var gFirstConnection = null;
	var gSecondConnection = null;

	var onOfferCreated = test.step_func(function(offer) {
	gFirstConnection.setLocalDescription(offer, ignoreSuccess,
	failed('setLocalDescription first'));

	// This would normally go across the application's signaling solution.
	// In our case, the "signaling" is to call this function.
	receiveCall(offer.sdp);
	});

	function receiveCall(offerSdp) {

	var parsedOffer = new RTCSessionDescription({ type: 'offer',
	sdp: offerSdp });
	// These functions use the legacy interface extensions to RTCPeerConnection.
	gSecondConnection.setRemoteDescription(parsedOffer,
	function() {
	gSecondConnection.createAnswer(onAnswerCreated,
	failed('createAnswer'));
	},
	failed('setRemoteDescription second'));
	};

	var onAnswerCreated = test.step_func(function(answer) {
	gSecondConnection.setLocalDescription(answer, ignoreSuccess,
	failed('setLocalDescription second'));

	// Similarly, this would go over the application's signaling solution.
	handleAnswer(answer.sdp);
	});

	function handleAnswer(answerSdp) {
	var parsedAnswer = new RTCSessionDescription({ type: 'answer',
	sdp: answerSdp });
	gFirstConnection.setRemoteDescription(parsedAnswer, ignoreSuccess,
	failed('setRemoteDescription first'));
	};

	var onIceCandidateToFirst = test.step_func(function(event) {
	// If event.candidate is null = no more candidates.
	if (event.candidate) {
	gSecondConnection.addIceCandidate(event.candidate);
	}
	});

	var onIceCandidateToSecond = test.step_func(function(event) {
	if (event.candidate) {
	gFirstConnection.addIceCandidate(event.candidate);
	}
	});

	var onRemoteStream = test.step_func(function(event) {
	assert_unreached('WebRTC received a stream when there was none');
	});

	var onIceConnectionStateChange = test.step_func(function(event) {
	assert_equals(event.type, 'iceconnectionstatechange');
	assert_not_equals(gFirstConnection.iceConnectionState, "failed", "iceConnectionState of first connection");
	assert_not_equals(gSecondConnection.iceConnectionState, "failed", "iceConnectionState of second connection");
	var stateinfo = document.getElementById('stateinfo');
	stateinfo.innerHTML = 'First: ' + gFirstConnection.iceConnectionState
	+ '<br>Second: ' + gSecondConnection.iceConnectionState;
	// Note: All these combinations are legal states indicating that the
	// call has connected. All browsers should end up in completed/completed,
	// but as of this moment, we've chosen to terminate the test early.
	// TODO: Revise test to ensure completed/completed is reached.
	if (gFirstConnection.iceConnectionState == 'connected' &&
	gSecondConnection.iceConnectionState == 'connected') {
	test.done()
	}
	if (gFirstConnection.iceConnectionState == 'connected' &&
	gSecondConnection.iceConnectionState == 'completed') {
	test.done()
	}
	if (gFirstConnection.iceConnectionState == 'completed' &&
	gSecondConnection.iceConnectionState == 'connected') {
	test.done()
	}
	if (gFirstConnection.iceConnectionState == 'completed' &&
	gSecondConnection.iceConnectionState == 'completed') {
	test.done()
	}
	});

	// Returns a suitable error callback.
	function failed(function_name) {
	return test.step_func(function() {
	assert_unreached('WebRTC called error callback for ' + function_name);
	});
	}

	// Returns a suitable do-nothing.
	function ignoreSuccess(function_name) {
	}

	// This function starts the test.
	test.step(function() {
	gFirstConnection = new RTCPeerConnection(null);
	gFirstConnection.onicecandidate = onIceCandidateToFirst;
	gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange;

	gSecondConnection = new RTCPeerConnection(null);
	gSecondConnection.onicecandidate = onIceCandidateToSecond;
	gSecondConnection.onaddstream = onRemoteStream;
	gSecondConnection.oniceconnectionstatechange = onIceConnectionStateChange;

	// The offerToReceiveVideo is necessary and sufficient to make
	// an actual connection.
	gFirstConnection.createOffer(onOfferCreated, failed('createOffer'),
	{offerToReceiveVideo: true});
	});
	</script>

	</body>
	</html>