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<!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>
|
