| Harald Alvestrand | 168935a | 2015-05-13 08:23:14 | [diff] [blame^] | 1 | <!doctype html> |
| 2 | <!-- |
| 3 | This test uses no media, and thus does not require fake media devices. |
| 4 | --> |
| 5 | |
| 6 | <html> |
| 7 | <head> |
| 8 | <meta http-equiv="Content-Type" content="text/html; charset=UTF-8"> |
| 9 | <title>RTCPeerConnection No-Media Connection Test</title> |
| 10 | </head> |
| 11 | <body> |
| 12 | <div id="log"></div> |
| 13 | <h2>iceConnectionState info</h2> |
| 14 | <div id="stateinfo"> |
| 15 | </div> |
| 16 | |
| 17 | <!-- These files are in place when executing on W3C. --> |
| 18 | <script src="/resources/testharness.js"></script> |
| 19 | <script src="/resources/testharnessreport.js"></script> |
| 20 | <script src="/common/vendor-prefix.js" |
| 21 | data-prefixed-objects= |
| 22 | '[{"ancestors":["window"], "name":"RTCPeerConnection"}, |
| 23 | {"ancestors":["window"], "name":"RTCSessionDescription"}, |
| 24 | {"ancestors":["window"], "name":"RTCIceCandidate"}]' |
| 25 | > |
| 26 | </script> |
| 27 | <script type="text/javascript"> |
| 28 | var test = async_test('Can set up a basic WebRTC call with no data using promises.'); |
| 29 | |
| 30 | var gFirstConnection = null; |
| 31 | var gSecondConnection = null; |
| 32 | |
| 33 | var onOfferCreated = test.step_func(function(offer) { |
| 34 | // TODO: Switch to promise-based interface. |
| 35 | gFirstConnection.setLocalDescription(offer) |
| 36 | .catch(failed('setLocalDescription first')); |
| 37 | |
| 38 | // This would normally go across the application's signaling solution. |
| 39 | // In our case, the "signaling" is to call this function. |
| 40 | receiveCall(offer.sdp); |
| 41 | }); |
| 42 | |
| 43 | function receiveCall(offerSdp) { |
| 44 | |
| 45 | var parsedOffer = new RTCSessionDescription({ type: 'offer', |
| 46 | sdp: offerSdp }); |
| 47 | gSecondConnection.setRemoteDescription(parsedOffer) |
| 48 | .then(function() { |
| 49 | gSecondConnection.createAnswer().then(onAnswerCreated, |
| 50 | failed('createAnswer')); |
| 51 | }, |
| 52 | failed('setRemoteDescription second')); |
| 53 | }; |
| 54 | |
| 55 | var onAnswerCreated = test.step_func(function(answer) { |
| 56 | gSecondConnection.setLocalDescription(answer) |
| 57 | .catch(failed('setLocalDescription second')); |
| 58 | |
| 59 | // Similarly, this would go over the application's signaling solution. |
| 60 | handleAnswer(answer.sdp); |
| 61 | }); |
| 62 | |
| 63 | function handleAnswer(answerSdp) { |
| 64 | var parsedAnswer = new RTCSessionDescription({ type: 'answer', |
| 65 | sdp: answerSdp }); |
| 66 | gFirstConnection.setRemoteDescription(parsedAnswer) |
| 67 | .catch(failed('setRemoteDescription first')); |
| 68 | }; |
| 69 | |
| 70 | var onIceCandidateToFirst = test.step_func(function(event) { |
| 71 | // If event.candidate is null = no more candidates. |
| 72 | if (event.candidate) { |
| 73 | var candidate = new RTCIceCandidate(event.candidate); |
| 74 | gSecondConnection.addIceCandidate(candidate); |
| 75 | } |
| 76 | }); |
| 77 | |
| 78 | var onIceCandidateToSecond = test.step_func(function(event) { |
| 79 | if (event.candidate) { |
| 80 | var candidate = new RTCIceCandidate(event.candidate); |
| 81 | gFirstConnection.addIceCandidate(candidate); |
| 82 | } |
| 83 | }); |
| 84 | |
| 85 | var onRemoteStream = test.step_func(function(event) { |
| 86 | assert_unreached('WebRTC received a stream when there was none'); |
| 87 | }); |
| 88 | |
| 89 | var onIceConnectionStateChange = test.step_func(function(event) { |
| 90 | assert_equals(event.type, 'iceconnectionstatechange'); |
| 91 | var stateinfo = document.getElementById('stateinfo'); |
| 92 | stateinfo.innerHTML = 'First: ' + gFirstConnection.iceConnectionState |
| 93 | + '<br>Second: ' + gSecondConnection.iceConnectionState; |
| 94 | // Note: All these combinations are legal states indicating that the |
| 95 | // call has connected. All browsers should end up in completed/completed, |
| 96 | // but as of this moment, we've chosen to terminate the test early. |
| 97 | // TODO: Revise test to ensure completed/completed is reached. |
| 98 | if (gFirstConnection.iceConnectionState == 'connected' && |
| 99 | gSecondConnection.iceConnectionState == 'connected') { |
| 100 | test.done() |
| 101 | } |
| 102 | if (gFirstConnection.iceConnectionState == 'connected' && |
| 103 | gSecondConnection.iceConnectionState == 'completed') { |
| 104 | test.done() |
| 105 | } |
| 106 | if (gFirstConnection.iceConnectionState == 'completed' && |
| 107 | gSecondConnection.iceConnectionState == 'connected') { |
| 108 | test.done() |
| 109 | } |
| 110 | if (gFirstConnection.iceConnectionState == 'completed' && |
| 111 | gSecondConnection.iceConnectionState == 'completed') { |
| 112 | test.done() |
| 113 | } |
| 114 | }); |
| 115 | |
| 116 | // Returns a suitable error callback. |
| 117 | function failed(function_name) { |
| 118 | return test.step_func(function() { |
| 119 | assert_unreached('WebRTC called error callback for ' + function_name); |
| 120 | }); |
| 121 | } |
| 122 | |
| 123 | // Returns a suitable do-nothing. |
| 124 | function ignoreSuccess(function_name) { |
| 125 | } |
| 126 | |
| 127 | // This function starts the test. |
| 128 | test.step(function() { |
| 129 | gFirstConnection = new RTCPeerConnection(null); |
| 130 | gFirstConnection.onicecandidate = onIceCandidateToFirst; |
| 131 | gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange; |
| 132 | |
| 133 | gSecondConnection = new RTCPeerConnection(null); |
| 134 | gSecondConnection.onicecandidate = onIceCandidateToSecond; |
| 135 | gSecondConnection.onaddstream = onRemoteStream; |
| 136 | gSecondConnection.oniceconnectionstatechange = onIceConnectionStateChange; |
| 137 | |
| 138 | // The offerToReceiveVideo is necessary and sufficient to make |
| 139 | // an actual connection. |
| 140 | // TODO: Use a promise-based API. This is legacy. |
| 141 | gFirstConnection.createOffer({offerToReceiveVideo: true}) |
| 142 | .then(onOfferCreated, failed('createOffer')); |
| 143 | |
| 144 | }); |
| 145 | </script> |
| 146 | |
| 147 | </body> |
| 148 | </html> |
