diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc
index a19f2cc8a4..67b4efbc73 100644
--- a/webrtc/api/peerconnection_unittest.cc
+++ b/webrtc/api/peerconnection_unittest.cc
@@ -1846,14 +1846,7 @@ TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
 // This test sets up a Jsep call with SCTP DataChannel and verifies the
 // negotiation is completed without error.
 #ifdef HAVE_SCTP
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_CreateOfferWithSctpDataChannel \
-  DISABLED_CreateOfferWithSctpDataChannel
-#else
-#define MAYBE_CreateOfferWithSctpDataChannel CreateOfferWithSctpDataChannel
-#endif
-TEST_F(P2PTestConductor, MAYBE_CreateOfferWithSctpDataChannel) {
+TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints constraints;
   constraints.SetMandatory(
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index 51968e6cbf..14a067995b 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -1701,16 +1701,10 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
   EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_ReceiveFireFoxOffer DISABLED_ReceiveFireFoxOffer
-#else
-#define MAYBE_ReceiveFireFoxOffer ReceiveFireFoxOffer
-#endif
 // Test that we can create a session description from an SDP string from
 // FireFox, use it as a remote session description, generate an answer and use
 // the answer as a local description.
-TEST_F(PeerConnectionInterfaceTest, MAYBE_ReceiveFireFoxOffer) {
+TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
@@ -2094,19 +2088,11 @@ TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
   EXPECT_EQ(0u, observer_.remote_streams()->count());
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SdpWithoutMsidCreatesDefaultStream \
-  DISABLED_SdpWithoutMsidCreatesDefaultStream
-#else
-#define MAYBE_SdpWithoutMsidCreatesDefaultStream \
-  SdpWithoutMsidCreatesDefaultStream
-#endif
 // This tests that a default MediaStream is created if a remote session
 // description doesn't contain any streams and no MSID support.
 // It also tests that the default stream is updated if a video m-line is added
 // in a subsequent session description.
-TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithoutMsidCreatesDefaultStream) {
+TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
@@ -2132,18 +2118,10 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithoutMsidCreatesDefaultStream) {
             remote_stream->GetVideoTracks()[0]->state());
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \
-  DISABLED_SendOnlySdpWithoutMsidCreatesDefaultStream
-#else
-#define MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream \
-  SendOnlySdpWithoutMsidCreatesDefaultStream
-#endif
 // This tests that a default MediaStream is created if a remote session
 // description doesn't contain any streams and media direction is send only.
 TEST_F(PeerConnectionInterfaceTest,
-       MAYBE_SendOnlySdpWithoutMsidCreatesDefaultStream) {
+       SendOnlySdpWithoutMsidCreatesDefaultStream) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
@@ -2175,19 +2153,11 @@ TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) {
   // No crash is a pass.
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \
-  DISABLED_SdpWithoutMsidAndStreamsCreatesDefaultStream
-#else
-#define MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream \
-  SdpWithoutMsidAndStreamsCreatesDefaultStream
-#endif
 // This tests that a default MediaStream is created if the remote session
 // description doesn't contain any streams and don't contain an indication if
 // MSID is supported.
 TEST_F(PeerConnectionInterfaceTest,
-       MAYBE_SdpWithoutMsidAndStreamsCreatesDefaultStream) {
+       SdpWithoutMsidAndStreamsCreatesDefaultStream) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
@@ -2200,17 +2170,9 @@ TEST_F(PeerConnectionInterfaceTest,
   EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SdpWithMsidDontCreatesDefaultStream \
-  DISABLED_SdpWithMsidDontCreatesDefaultStream
-#else
-#define MAYBE_SdpWithMsidDontCreatesDefaultStream \
-  SdpWithMsidDontCreatesDefaultStream
-#endif
 // This tests that a default MediaStream is not created if the remote session
 // description doesn't contain any streams but does support MSID.
-TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithMsidDontCreatesDefaultStream) {
+TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
@@ -2219,19 +2181,11 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_SdpWithMsidDontCreatesDefaultStream) {
   EXPECT_EQ(0u, observer_.remote_streams()->count());
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_DefaultTracksNotDestroyedAndRecreated \
-  DISABLED_DefaultTracksNotDestroyedAndRecreated
-#else
-#define MAYBE_DefaultTracksNotDestroyedAndRecreated \
-  DefaultTracksNotDestroyedAndRecreated
-#endif
 // This tests that when setting a new description, the old default tracks are
 // not destroyed and recreated.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
 TEST_F(PeerConnectionInterfaceTest,
-       MAYBE_DefaultTracksNotDestroyedAndRecreated) {
+       DefaultTracksNotDestroyedAndRecreated) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
@@ -2266,17 +2220,11 @@ TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
   EXPECT_EQ(0u, observer_.remote_streams()->count());
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_LocalDescriptionChanged DISABLED_LocalDescriptionChanged
-#else
-#define MAYBE_LocalDescriptionChanged LocalDescriptionChanged
-#endif
 // This tests that an RtpSender is created when the local description is set
 // after adding a local stream.
 // TODO(deadbeef): This test and the one below it need to be updated when
 // an RtpSender's lifetime isn't determined by when a local description is set.
-TEST_F(PeerConnectionInterfaceTest, MAYBE_LocalDescriptionChanged) {
+TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
@@ -2312,18 +2260,10 @@ TEST_F(PeerConnectionInterfaceTest, MAYBE_LocalDescriptionChanged) {
   EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1]));
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \
-  DISABLED_AddLocalStreamAfterLocalDescriptionChanged
-#else
-#define MAYBE_AddLocalStreamAfterLocalDescriptionChanged \
-  AddLocalStreamAfterLocalDescriptionChanged
-#endif
 // This tests that an RtpSender is created when the local description is set
 // before adding a local stream.
 TEST_F(PeerConnectionInterfaceTest,
-       MAYBE_AddLocalStreamAfterLocalDescriptionChanged) {
+       AddLocalStreamAfterLocalDescriptionChanged) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
@@ -2349,18 +2289,10 @@ TEST_F(PeerConnectionInterfaceTest,
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \
-  DISABLED_ChangeSsrcOnTrackInLocalSessionDescription
-#else
-#define MAYBE_ChangeSsrcOnTrackInLocalSessionDescription \
-  ChangeSsrcOnTrackInLocalSessionDescription
-#endif
 // This tests that the expected behavior occurs if the SSRC on a local track is
 // changed when SetLocalDescription is called.
 TEST_F(PeerConnectionInterfaceTest,
-       MAYBE_ChangeSsrcOnTrackInLocalSessionDescription) {
+       ChangeSsrcOnTrackInLocalSessionDescription) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
@@ -2404,18 +2336,10 @@ TEST_F(PeerConnectionInterfaceTest,
   // changed.
 }
 
-// Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659
-#if defined(WEBRTC_WIN) && defined(_DEBUG)
-#define MAYBE_SignalSameTracksInSeparateMediaStream \
-  DISABLED_SignalSameTracksInSeparateMediaStream
-#else
-#define MAYBE_SignalSameTracksInSeparateMediaStream \
-  SignalSameTracksInSeparateMediaStream
-#endif
 // This tests that the expected behavior occurs if a new session description is
 // set with the same tracks, but on a different MediaStream.
 TEST_F(PeerConnectionInterfaceTest,
-       MAYBE_SignalSameTracksInSeparateMediaStream) {
+       SignalSameTracksInSeparateMediaStream) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);