summaryrefslogtreecommitdiffstats
path: root/testing/web-platform/tests/webrtc/simplecall.html
diff options
context:
space:
mode:
authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /testing/web-platform/tests/webrtc/simplecall.html
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
downloadUXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.gz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.lz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.xz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.zip
Add m-esr52 at 52.6.0
Diffstat (limited to 'testing/web-platform/tests/webrtc/simplecall.html')
-rw-r--r--testing/web-platform/tests/webrtc/simplecall.html118
1 files changed, 118 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc/simplecall.html b/testing/web-platform/tests/webrtc/simplecall.html
new file mode 100644
index 000000000..698586894
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/simplecall.html
@@ -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>