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import FIREFOX_52_6_0esr_RELEASE from mozilla-esr52 hg repo
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4
testing/web-platform/tests/webrtc/OWNERS
Normal file
4
testing/web-platform/tests/webrtc/OWNERS
Normal file
|
|
@ -0,0 +1,4 @@
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|||
@dontcallmedom
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||||
@tidoust
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||||
@alvestrand
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||||
@phoglund
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||||
108
testing/web-platform/tests/webrtc/datachannel-emptystring.html
Normal file
108
testing/web-platform/tests/webrtc/datachannel-emptystring.html
Normal file
|
|
@ -0,0 +1,108 @@
|
|||
<!doctype html>
|
||||
<!--
|
||||
This test creates a data channel between two local PeerConnection instances
|
||||
and ensures that an empty string sent by one is received by the second.
|
||||
-->
|
||||
|
||||
<html>
|
||||
<head>
|
||||
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
|
||||
<title>RTCPeerConnection Data Channel Empty String Test</title>
|
||||
</head>
|
||||
<body>
|
||||
<div id="log"></div>
|
||||
<h2>Messages exchanged</h2>
|
||||
<div id="msg"></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 send empty strings across a WebRTC data channel.');
|
||||
|
||||
var gFirstConnection = null;
|
||||
var gSecondConnection = null;
|
||||
var sendChannel = null;
|
||||
var receiveChannel = null;
|
||||
|
||||
var onReceiveChannel = function (event) {
|
||||
receiveChannel = event.channel;
|
||||
receiveChannel.onmessage = onReceiveMessage;
|
||||
};
|
||||
|
||||
|
||||
// When the data channel is open, send an empty string message
|
||||
// followed by a message that contains the string "done".
|
||||
var onSendChannelOpen = test.step_func(function (event) {
|
||||
var msgEl = document.getElementById('msg');
|
||||
sendChannel.send('');
|
||||
msgEl.innerHTML += 'Sent: [empty string]<br>';
|
||||
sendChannel.send('done');
|
||||
msgEl.innerHTML += 'Sent: "done"<br>';
|
||||
});
|
||||
|
||||
// Check the messages received on the other side.
|
||||
// There should be an empty string message followed by a message that
|
||||
// contains the string "done".
|
||||
// Pass/Fail the test according to the messages received
|
||||
var emptyMessageReceived = false;
|
||||
var onReceiveMessage = test.step_func(function (event) {
|
||||
var msgEl = document.getElementById('msg');
|
||||
msgEl.innerHTML += 'Received: ' +
|
||||
(event.data ? '"' + event.data + '"' : '[empty string]') + '<br>';
|
||||
if (emptyMessageReceived) {
|
||||
assert_equals(event.data, 'done', 'The "done" message was not received');
|
||||
test.done();
|
||||
}
|
||||
else {
|
||||
assert_equals(event.data, '', 'Empty message not received');
|
||||
emptyMessageReceived = true;
|
||||
}
|
||||
});
|
||||
|
||||
function exchangeIce(pc) {
|
||||
return function(e) {
|
||||
if (e.candidate) {
|
||||
pc.addIceCandidate(e.candidate);
|
||||
}
|
||||
};
|
||||
}
|
||||
|
||||
function exchangeDescription(pc1, pc2) {
|
||||
return function() {
|
||||
return pc1.setRemoteDescription(pc2.localDescription);
|
||||
};
|
||||
}
|
||||
|
||||
test.step(function() {
|
||||
gFirstConnection = new RTCPeerConnection(null);
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||||
|
||||
gSecondConnection = new RTCPeerConnection(null);
|
||||
|
||||
gFirstConnection.onicecandidate = exchangeIce(gSecondConnection);
|
||||
gSecondConnection.onicecandidate = exchangeIce(gFirstConnection);
|
||||
|
||||
gSecondConnection.ondatachannel = onReceiveChannel;
|
||||
|
||||
// Note the data channel will preserve the order of messages
|
||||
// exchanged over it by default.
|
||||
sendChannel = gFirstConnection.createDataChannel('sendDataChannel');
|
||||
sendChannel.onopen = onSendChannelOpen;
|
||||
|
||||
gFirstConnection.createOffer()
|
||||
.then(gFirstConnection.setLocalDescription.bind(gFirstConnection))
|
||||
.then(exchangeDescription(gSecondConnection, gFirstConnection))
|
||||
.then(function() {
|
||||
return gSecondConnection.createAnswer();
|
||||
})
|
||||
.then(gSecondConnection.setLocalDescription.bind(gSecondConnection))
|
||||
.then(exchangeDescription(gFirstConnection, gSecondConnection))
|
||||
.catch(test.step_func(function(e) {
|
||||
assert_unreached('Error ' + e.name + ': ' + e.message);
|
||||
}));
|
||||
});
|
||||
</script>
|
||||
|
||||
</body>
|
||||
</html>
|
||||
|
||||
139
testing/web-platform/tests/webrtc/no-media-call.html
Normal file
139
testing/web-platform/tests/webrtc/no-media-call.html
Normal file
|
|
@ -0,0 +1,139 @@
|
|||
<!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>
|
||||
121
testing/web-platform/tests/webrtc/promises-call.html
Normal file
121
testing/web-platform/tests/webrtc/promises-call.html
Normal file
|
|
@ -0,0 +1,121 @@
|
|||
<!doctype html>
|
||||
<!--
|
||||
This test uses data only, and thus does not require fake media devices.
|
||||
-->
|
||||
|
||||
<html>
|
||||
<head>
|
||||
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
|
||||
<title>RTCPeerConnection Data-Only Connection Test with Promises</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 only data using promises.');
|
||||
|
||||
var gFirstConnection = null;
|
||||
var gSecondConnection = null;
|
||||
|
||||
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');
|
||||
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()
|
||||
}
|
||||
});
|
||||
|
||||
// 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 createDataChannel is necessary and sufficient to make
|
||||
// sure the ICE connection be attempted.
|
||||
gFirstConnection.createDataChannel('channel');
|
||||
|
||||
var atStep = 'Create offer';
|
||||
|
||||
gFirstConnection.createOffer()
|
||||
.then(function(offer) {
|
||||
atStep = 'Set local description at first';
|
||||
return gFirstConnection.setLocalDescription(offer);
|
||||
})
|
||||
.then(function() {
|
||||
atStep = 'Set remote description at second';
|
||||
return gSecondConnection.setRemoteDescription(
|
||||
gFirstConnection.localDescription);
|
||||
})
|
||||
.then(function() {
|
||||
atStep = 'Create answer';
|
||||
return gSecondConnection.createAnswer();
|
||||
})
|
||||
.then(function(answer) {
|
||||
atStep = 'Set local description at second';
|
||||
return gSecondConnection.setLocalDescription(answer);
|
||||
})
|
||||
.then(function() {
|
||||
atStep = 'Set remote description at first';
|
||||
return gFirstConnection.setRemoteDescription(
|
||||
gSecondConnection.localDescription);
|
||||
})
|
||||
.then(function() {
|
||||
atStep = 'Negotiation completed';
|
||||
})
|
||||
.catch(test.step_func(function(e) {
|
||||
assert_unreached('Error ' + e.name + ': ' + e.message +
|
||||
' happened at step ' + atStep);
|
||||
}));
|
||||
});
|
||||
</script>
|
||||
|
||||
</body>
|
||||
</html>
|
||||
|
|
@ -0,0 +1,107 @@
|
|||
<!doctype html>
|
||||
<html>
|
||||
<head>
|
||||
<meta charset=utf-8>
|
||||
<title>IDL check of RTCPeerConnection</title>
|
||||
<link rel="author" title="Harald Alvestrand" href="mailto:hta@google.com"/>
|
||||
<link rel="help" href="http://w3c.github.io/webrtc-pc/#rtcpeerconnection-interface">
|
||||
</head>
|
||||
<body>
|
||||
|
||||
<h1 class="instructions">Description</h1>
|
||||
<p class="instructions">This test verifies the availability of the RTCPeerConnection interface.</p>
|
||||
<div id='log'></div>
|
||||
<script src=/resources/testharness.js></script>
|
||||
<script src=/resources/testharnessreport.js></script>
|
||||
<script src=/resources/WebIDLParser.js></script>
|
||||
<script src=/resources/idlharness.js></script>
|
||||
|
||||
<!-- The IDL is copied from the 22 September 2015 editors' draft. -->
|
||||
<script type="text/plain">
|
||||
interface EventTarget {
|
||||
// Only a dummy definition is needed here.
|
||||
};
|
||||
[ Constructor (optional RTCConfiguration configuration)]
|
||||
interface RTCPeerConnection : EventTarget {
|
||||
Promise<RTCSessionDescription> createOffer (optional RTCOfferOptions options);
|
||||
Promise<RTCSessionDescription> createAnswer (optional RTCAnswerOptions options);
|
||||
Promise<void> setLocalDescription (RTCSessionDescription description);
|
||||
readonly attribute RTCSessionDescription? localDescription;
|
||||
readonly attribute RTCSessionDescription? currentLocalDescription;
|
||||
readonly attribute RTCSessionDescription? pendingLocalDescription;
|
||||
Promise<void> setRemoteDescription (RTCSessionDescription description);
|
||||
readonly attribute RTCSessionDescription? remoteDescription;
|
||||
readonly attribute RTCSessionDescription? currentRemoteDescription;
|
||||
readonly attribute RTCSessionDescription? pendingRemoteDescription;
|
||||
Promise<void> addIceCandidate (RTCIceCandidate candidate);
|
||||
readonly attribute RTCSignalingState signalingState;
|
||||
readonly attribute RTCIceGatheringState iceGatheringState;
|
||||
readonly attribute RTCIceConnectionState iceConnectionState;
|
||||
readonly attribute boolean? canTrickleIceCandidates;
|
||||
RTCConfiguration getConfiguration ();
|
||||
void setConfiguration (RTCConfiguration configuration);
|
||||
void close ();
|
||||
attribute EventHandler onnegotiationneeded;
|
||||
attribute EventHandler onicecandidate;
|
||||
attribute EventHandler onsignalingstatechange;
|
||||
attribute EventHandler oniceconnectionstatechange;
|
||||
attribute EventHandler onicegatheringstatechange;
|
||||
};
|
||||
|
||||
partial interface RTCPeerConnection {
|
||||
void createOffer (RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback, optional RTCOfferOptions options);
|
||||
void setLocalDescription (RTCSessionDescription description, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
|
||||
void createAnswer (RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback);
|
||||
void setRemoteDescription (RTCSessionDescription description, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
|
||||
void addIceCandidate (RTCIceCandidate candidate, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
|
||||
void getStats (MediaStreamTrack? selector, RTCStatsCallback successCallback, RTCPeerConnectionErrorCallback failureCallback);
|
||||
};
|
||||
|
||||
partial interface RTCPeerConnection {
|
||||
static Promise<RTCCertificate> generateCertificate (AlgorithmIdentifier keygenAlgorithm);
|
||||
};
|
||||
|
||||
partial interface RTCPeerConnection {
|
||||
sequence<RTCRtpSender> getSenders ();
|
||||
sequence<RTCRtpReceiver> getReceivers ();
|
||||
RTCRtpSender addTrack (MediaStreamTrack track, MediaStream... streams);
|
||||
void removeTrack (RTCRtpSender sender);
|
||||
attribute EventHandler ontrack;
|
||||
};
|
||||
|
||||
partial interface RTCPeerConnection {
|
||||
RTCDataChannel createDataChannel ([TreatNullAs=EmptyString] DOMString label, optional RTCDataChannelInit dataChannelDict);
|
||||
attribute EventHandler ondatachannel;
|
||||
};
|
||||
|
||||
partial interface RTCPeerConnection {
|
||||
readonly attribute RTCDTMFSender? dtmf;
|
||||
};
|
||||
|
||||
partial interface RTCPeerConnection {
|
||||
Promise<RTCStatsReport> getStats (optional MediaStreamTrack? selector);
|
||||
};
|
||||
|
||||
partial interface RTCPeerConnection {
|
||||
void setIdentityProvider (DOMString provider, optional DOMString protocol, optional DOMString usernameHint);
|
||||
Promise<DOMString> getIdentityAssertion ();
|
||||
readonly attribute Promise<RTCIdentityAssertion> peerIdentity;
|
||||
readonly attribute DOMString? idpLoginUrl;
|
||||
};
|
||||
|
||||
</script>
|
||||
<script>
|
||||
(function() {
|
||||
var idl_array = new IdlArray();
|
||||
[].forEach.call(document.querySelectorAll("script[type=text\\/plain]"),
|
||||
function(node) {
|
||||
idl_array.add_idls(node.textContent);
|
||||
});
|
||||
window.pc = new RTCPeerConnection(null);
|
||||
idl_array.add_objects({"RTCPeerConnection": ["pc"]});
|
||||
idl_array.test();
|
||||
done();
|
||||
})();
|
||||
</script>
|
||||
</body>
|
||||
</html>
|
||||
118
testing/web-platform/tests/webrtc/simplecall.html
Normal file
118
testing/web-platform/tests/webrtc/simplecall.html
Normal file
|
|
@ -0,0 +1,118 @@
|
|||
<!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>
|
||||
Loading…
Add table
Add a link
Reference in a new issue