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140 lines
5.1 KiB
HTML
140 lines
5.1 KiB
HTML
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<!doctype html>
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<!--
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This test uses the legacy callback API with no media, and thus does not require fake media devices.
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-->
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<html>
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<head>
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<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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<title>RTCPeerConnection No-Media Connection Test</title>
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</head>
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<body>
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<div id="log"></div>
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<h2>iceConnectionState info</h2>
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<div id="stateinfo">
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</div>
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<!-- These files are in place when executing on W3C. -->
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script type="text/javascript">
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var test = async_test('Can set up a basic WebRTC call with no data.');
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var gFirstConnection = null;
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var gSecondConnection = null;
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var onOfferCreated = test.step_func(function(offer) {
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gFirstConnection.setLocalDescription(offer, ignoreSuccess,
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failed('setLocalDescription first'));
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// This would normally go across the application's signaling solution.
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// In our case, the "signaling" is to call this function.
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receiveCall(offer.sdp);
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});
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function receiveCall(offerSdp) {
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var parsedOffer = new RTCSessionDescription({ type: 'offer',
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sdp: offerSdp });
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// These functions use the legacy interface extensions to RTCPeerConnection.
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gSecondConnection.setRemoteDescription(parsedOffer,
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function() {
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gSecondConnection.createAnswer(onAnswerCreated,
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failed('createAnswer'));
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},
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failed('setRemoteDescription second'));
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};
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var onAnswerCreated = test.step_func(function(answer) {
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gSecondConnection.setLocalDescription(answer, ignoreSuccess,
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failed('setLocalDescription second'));
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// Similarly, this would go over the application's signaling solution.
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handleAnswer(answer.sdp);
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});
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function handleAnswer(answerSdp) {
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var parsedAnswer = new RTCSessionDescription({ type: 'answer',
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sdp: answerSdp });
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gFirstConnection.setRemoteDescription(parsedAnswer, ignoreSuccess,
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failed('setRemoteDescription first'));
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};
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var onIceCandidateToFirst = test.step_func(function(event) {
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// If event.candidate is null = no more candidates.
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if (event.candidate) {
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gSecondConnection.addIceCandidate(event.candidate);
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}
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});
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var onIceCandidateToSecond = test.step_func(function(event) {
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if (event.candidate) {
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gFirstConnection.addIceCandidate(event.candidate);
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}
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});
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var onRemoteStream = test.step_func(function(event) {
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assert_unreached('WebRTC received a stream when there was none');
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});
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var onIceConnectionStateChange = test.step_func(function(event) {
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assert_equals(event.type, 'iceconnectionstatechange');
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assert_not_equals(gFirstConnection.iceConnectionState, "failed", "iceConnectionState of first connection");
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assert_not_equals(gSecondConnection.iceConnectionState, "failed", "iceConnectionState of second connection");
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var stateinfo = document.getElementById('stateinfo');
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stateinfo.innerHTML = 'First: ' + gFirstConnection.iceConnectionState
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+ '<br>Second: ' + gSecondConnection.iceConnectionState;
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// Note: All these combinations are legal states indicating that the
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// call has connected. All browsers should end up in completed/completed,
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// but as of this moment, we've chosen to terminate the test early.
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// TODO: Revise test to ensure completed/completed is reached.
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if (gFirstConnection.iceConnectionState == 'connected' &&
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gSecondConnection.iceConnectionState == 'connected') {
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test.done()
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}
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if (gFirstConnection.iceConnectionState == 'connected' &&
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gSecondConnection.iceConnectionState == 'completed') {
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test.done()
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}
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if (gFirstConnection.iceConnectionState == 'completed' &&
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gSecondConnection.iceConnectionState == 'connected') {
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test.done()
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}
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if (gFirstConnection.iceConnectionState == 'completed' &&
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gSecondConnection.iceConnectionState == 'completed') {
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test.done()
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}
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});
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// Returns a suitable error callback.
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function failed(function_name) {
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return test.step_func(function() {
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assert_unreached('WebRTC called error callback for ' + function_name);
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});
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}
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// Returns a suitable do-nothing.
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function ignoreSuccess(function_name) {
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}
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// This function starts the test.
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test.step(function() {
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gFirstConnection = new RTCPeerConnection(null);
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gFirstConnection.onicecandidate = onIceCandidateToFirst;
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gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange;
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gSecondConnection = new RTCPeerConnection(null);
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gSecondConnection.onicecandidate = onIceCandidateToSecond;
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gSecondConnection.onaddstream = onRemoteStream;
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gSecondConnection.oniceconnectionstatechange = onIceConnectionStateChange;
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// The offerToReceiveVideo is necessary and sufficient to make
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// an actual connection.
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gFirstConnection.createOffer(onOfferCreated, failed('createOffer'),
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{offerToReceiveVideo: true});
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});
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</script>
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</body>
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</html>
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