| <!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> |
