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<!doctype html>
<!--
To quickly iterate when developing this test, use --use-fake-ui-for-media-stream
for Chrome and set the media.navigator.permission.disabled property to true in
Firefox. You must either have a webcam/mic available on the system or use for
instance --use-fake-device-for-media-stream for Chrome.
-->
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
<title>RTCPeerConnection Connection Test</title>
</head>
<body>
<div id="log"></div>
<div>
<video id="local-view" autoplay="autoplay"></video>
<video id="remote-view" autoplay="autoplay"/>
</video>
</div>
<!-- These files are in place when executing on W3C. -->
<script src="/resources/testharness.js"></script>
<script src="/resources/testharnessreport.js"></script>
<script src="/common/vendor-prefix.js"
data-prefixed-objects=
'[{"ancestors":["navigator"], "name":"getUserMedia"}]'
data-prefixed-prototypes=
'[{"ancestors":["HTMLMediaElement"],"name":"srcObject"}]'>
</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 RTCPeerConnection(null);
gFirstConnection.onicecandidate = onIceCandidateToFirst;
gFirstConnection.addStream(localStream);
gFirstConnection.createOffer(onOfferCreated, failed('createOffer'));
var videoTag = document.getElementById('local-view');
videoTag.srcObject = 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 RTCPeerConnection(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();
};
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) {
var videoTag = document.getElementById('remote-view');
videoTag.srcObject = 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.getUserMedia({ video: true, audio: true },
getUserMediaOkCallback,
failed('getUserMedia'));
});
</script>
</body>
</html>
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