| <!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> |
