Test for a no-media call. This test verifies that a connection is established for a call where no media is added, but the initiator offers to receive video. The iceconnectionstate = connected indicator is used to determine that the connection is established.
diff --git a/webrtc/no-media-call.html b/webrtc/no-media-call.html new file mode 100644 index 0000000..3db73c5 --- /dev/null +++ b/webrtc/no-media-call.html
@@ -0,0 +1,130 @@ +<!doctype html> +<!-- +This test uses 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> + <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"}, + {"ancestors":["window"], "name":"RTCPeerConnection"}, + {"ancestors":["window"], "name":"RTCSessionDescription"}, + {"ancestors":["window"], "name":"RTCIceCandidate"}]' + data-prefixed-prototypes= + '[{"ancestors":["HTMLMediaElement"],"name":"srcObject"}]'> + </script> + <script type="text/javascript"> + var test = async_test('Can set up a basic WebRTC call with no data.', + {timeout: 5000}); + + var gFirstConnection = null; + var gSecondConnection = null; + + 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) { + + 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); + }; + + // Note: the ice candidate handlers are special. We can not wrap them in test + // steps since that seems to cause some kind of starvation that prevents the + // call of being set up. Unfortunately we cannot report errors in here. + var onIceCandidateToFirst = function(event) { + // If event.candidate is null = no more candidates. + if (event.candidate) { + var candidate = new RTCIceCandidate(event.candidate); + gSecondConnection.addIceCandidate(candidate); + } + }; + + var onIceCandidateToSecond = function(event) { + if (event.candidate) { + var candidate = new RTCIceCandidate(event.candidate); + gFirstConnection.addIceCandidate(candidate); + } + }; + + var onRemoteStream = test.step_func(function(event) { + assert_unreached('WebRTC received a stream when there was none'); + }); + + function onIceConnectionStateChange(event) { + assert_equals(event.type, 'iceconnectionstatechange'); + if (gFirstConnection.iceConnectionState == 'completed' && + gSecondConnection.iceConnectionState == 'connected') { + test.done() + } + // Note: This should have been as below. + 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); + }); + } + + // This function starts the test. + test.step(function() { + gFirstConnection = new RTCPeerConnection(null, null); + gFirstConnection.onicecandidate = onIceCandidateToFirst; + gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange; + + gSecondConnection = new RTCPeerConnection(null, 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> 