WebKit export of https://bugs.webkit.org/show_bug.cgi?id=189040
diff --git a/webrtc/no-media-call.html b/webrtc/no-media-call.html index c4979e8..c973932 100644 --- a/webrtc/no-media-call.html +++ b/webrtc/no-media-call.html
@@ -37,9 +37,9 @@ var parsedOffer = new RTCSessionDescription({ type: 'offer', sdp: offerSdp }); // These functions use the legacy interface extensions to RTCPeerConnection. - gSecondConnection.setRemoteDescription(parsedOffer, + gSecondConnection.setRemoteDescription(parsedOffer).then( function() { - gSecondConnection.createAnswer(onAnswerCreated, + gSecondConnection.createAnswer().then(onAnswerCreated, failed('createAnswer')); }, failed('setRemoteDescription second')); @@ -56,7 +56,7 @@ function handleAnswer(answerSdp) { var parsedAnswer = new RTCSessionDescription({ type: 'answer', sdp: answerSdp }); - gFirstConnection.setRemoteDescription(parsedAnswer, ignoreSuccess, + gFirstConnection.setRemoteDescription(parsedAnswer).then(ignoreSuccess, failed('setRemoteDescription first')); }; @@ -125,7 +125,7 @@ // The offerToReceiveVideo is necessary and sufficient to make // an actual connection. - gFirstConnection.createOffer(onOfferCreated, failed('createOffer'), + gFirstConnection.createOffer().then(onOfferCreated, failed('createOffer'), {offerToReceiveVideo: true}); }); </script> 