webrtc/test/w3c/peerconnection_conformance_test.html - src/ - Git at Google

 <!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">

 <!--
 To quickly iterate when developing this test, make sure you select
 'Always allow this site to use this webcam' option in the dropdown menu of
 Chrome when it's requesting access to your webcam.
 Notice that this requires the site you're browsing to use HTTPS.

 Without that, the test might timeout before you have had the chance to accept
 access to the webcam.
 -->

 <html>
 <head>
   <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
   <title>PeerConnection Connection Test</title>

   <script src="https://w3c-test.org/resources/testharness.js"></script>

   <script type="text/javascript">
   var test = async_test('Can set up a basic WebRTC call.', {timeout: 5000});
   var gFirstConnection = null;
   var gSecondConnection = null;

   function getUserMediaOkCallback(localStream) {
     gFirstConnection = new webkitRTCPeerConnection(null, null);
     gFirstConnection.onicecandidate = onIceCandidateToFirst;
     gFirstConnection.addStream(localStream);
     gFirstConnection.createOffer(onOfferCreated);

     var videoTag = document.getElementById('local-view');
     videoTag.src = webkitURL.createObjectURL(localStream);
   };

   var onOfferCreated = test.step_func(function(offer) {
     gFirstConnection.setLocalDescription(offer);

     // 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) {
     gSecondConnection = new webkitRTCPeerConnection(null, null);
     gSecondConnection.onicecandidate = onIceCandidateToSecond;
     gSecondConnection.onaddstream = onRemoteStream;

     var parsedOffer = new RTCSessionDescription({ type: 'offer',
                                                   sdp: offerSdp });
     gSecondConnection.setRemoteDescription(parsedOffer);

     gSecondConnection.createAnswer(onAnswerCreated,
                                    failed('createAnswer'));
   };

   var onAnswerCreated = test.step_func(function(answer) {
     gSecondConnection.setLocalDescription(answer);

     // 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);

     // Call negotiated: done.
     test.done();
   };

   // Note: the ice candidate handlers are special. We can not wrap them in test
   // steps since that seems to cause some kind of starvation that prevents the
   // call of being set up. Unfortunately we cannot report errors in here.
   var onIceCandidateToFirst = function(event) {
     // If event.candidate is null = no more candidates.
     if (event.candidate) {
       var candidate = new RTCIceCandidate(event.candidate);
       gSecondConnection.addIceCandidate(candidate);
     }
   };

   var onIceCandidateToSecond = function(event) {
     if (event.candidate) {
       var candidate = new RTCIceCandidate(event.candidate);
       gFirstConnection.addIceCandidate(candidate);
     }
   };

   var onRemoteStream = test.step_func(function(event) {
     var videoTag = document.getElementById('remote-view');
     videoTag.src = webkitURL.createObjectURL(event.stream);
   });

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

   // This function starts the test.
   test.step(function() {
     navigator.webkitGetUserMedia({ video: true, audio: true },
         getUserMediaOkCallback,
         failed('getUserMedia'));
   });
 </script>
 </head>

 <body>

 <div>
   <video width="320" height="240" id="remote-view" autoplay="autoplay"></video>
   <video width="320" height="240" id="local-view" autoplay="autoplay"></video>
 </div>
 <div id="log"></div>
 </body>
 </html>
	<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">

	<!--
	To quickly iterate when developing this test, make sure you select
	'Always allow this site to use this webcam' option in the dropdown menu of
	Chrome when it's requesting access to your webcam.
	Notice that this requires the site you're browsing to use HTTPS.

	Without that, the test might timeout before you have had the chance to accept
	access to the webcam.
	-->

	<html>
	<head>
	<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
	<title>PeerConnection Connection Test</title>

	<script src="https://w3c-test.org/resources/testharness.js"></script>

	<script type="text/javascript">
	var test = async_test('Can set up a basic WebRTC call.', {timeout: 5000});
	var gFirstConnection = null;
	var gSecondConnection = null;

	function getUserMediaOkCallback(localStream) {
	gFirstConnection = new webkitRTCPeerConnection(null, null);
	gFirstConnection.onicecandidate = onIceCandidateToFirst;
	gFirstConnection.addStream(localStream);
	gFirstConnection.createOffer(onOfferCreated);

	var videoTag = document.getElementById('local-view');
	videoTag.src = webkitURL.createObjectURL(localStream);
	};

	var onOfferCreated = test.step_func(function(offer) {
	gFirstConnection.setLocalDescription(offer);

	// 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) {
	gSecondConnection = new webkitRTCPeerConnection(null, null);
	gSecondConnection.onicecandidate = onIceCandidateToSecond;
	gSecondConnection.onaddstream = onRemoteStream;

	var parsedOffer = new RTCSessionDescription({ type: 'offer',
	sdp: offerSdp });
	gSecondConnection.setRemoteDescription(parsedOffer);

	gSecondConnection.createAnswer(onAnswerCreated,
	failed('createAnswer'));
	};

	var onAnswerCreated = test.step_func(function(answer) {
	gSecondConnection.setLocalDescription(answer);

	// 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);

	// Call negotiated: done.
	test.done();
	};

	// Note: the ice candidate handlers are special. We can not wrap them in test
	// steps since that seems to cause some kind of starvation that prevents the
	// call of being set up. Unfortunately we cannot report errors in here.
	var onIceCandidateToFirst = function(event) {
	// If event.candidate is null = no more candidates.
	if (event.candidate) {
	var candidate = new RTCIceCandidate(event.candidate);
	gSecondConnection.addIceCandidate(candidate);
	}
	};

	var onIceCandidateToSecond = function(event) {
	if (event.candidate) {
	var candidate = new RTCIceCandidate(event.candidate);
	gFirstConnection.addIceCandidate(candidate);
	}
	};

	var onRemoteStream = test.step_func(function(event) {
	var videoTag = document.getElementById('remote-view');
	videoTag.src = webkitURL.createObjectURL(event.stream);
	});

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

	// This function starts the test.
	test.step(function() {
	navigator.webkitGetUserMedia({ video: true, audio: true },
	getUserMediaOkCallback,
	failed('getUserMedia'));
	});
	</script>
	</head>

	<body>

	<div>
	<video width="320" height="240" id="remote-view" autoplay="autoplay"></video>
	<video width="320" height="240" id="local-view" autoplay="autoplay"></video>
	</div>
	<div id="log"></div>
	</body>
	</html>