diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index d9a29ec4db..745d1f5dcb 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -8,10 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// Disable for TSan v2, see
-// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
-#if !defined(THREAD_SANITIZER)
-
 #include <stdio.h>
 
 #include <algorithm>
@@ -3717,6 +3713,9 @@ TEST_P(PeerConnectionIntegrationTest,
   EXPECT_FALSE(callee()->data_observer()->IsOpen());
 }
 
+#if !defined(THREAD_SANITIZER)
+// This test provokes TSAN errors. See bugs.webrtc.org/11282
+
 // Tests that data is buffered in an RTP data channel until an observer is
 // registered for it.
 //
@@ -3763,6 +3762,8 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock,
                            FakeClock());
 }
 
+#endif  // !defined(THREAD_SANITIZER)
+
 // This test sets up a call between two parties with audio, video and but only
 // the caller client supports RTP data channels.
 TEST_P(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
@@ -4073,6 +4074,9 @@ TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
                  callee()->ice_connection_state(), kDefaultTimeout);
 }
 
+#if !defined(THREAD_SANITIZER)
+// This test provokes TSAN errors. See bugs.webrtc.org/3608
+
 constexpr int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
                                 cricket::PORTALLOCATOR_DISABLE_RELAY |
                                 cricket::PORTALLOCATOR_DISABLE_TCP;
@@ -4133,6 +4137,8 @@ TEST_P(PeerConnectionIntegrationTest,
                           webrtc::kIceCandidatePairHostNameHostName));
 }
 
+#endif  // !defined(THREAD_SANITIZER)
+
 // Test that firewalling the ICE connection causes the clients to identify the
 // disconnected state and then removing the firewall causes them to reconnect.
 class PeerConnectionIntegrationIceStatesTest
@@ -4201,6 +4207,9 @@ class PeerConnectionIntegrationIceStatesTestWithFakeClock
     : public FakeClockForTest,
       public PeerConnectionIntegrationIceStatesTest {};
 
+#if !defined(THREAD_SANITIZER)
+// This test provokes TSAN errors. bugs.webrtc.org/11282
+
 // Tests that the PeerConnection goes through all the ICE gathering/connection
 // states over the duration of the call. This includes Disconnected and Failed
 // states, induced by putting a firewall between the peers and waiting for them
@@ -4327,6 +4336,8 @@ TEST_P(PeerConnectionIntegrationIceStatesTestWithFakeClock,
                            kConsentTimeout, FakeClock());
 }
 
+#endif  // !defined(THREAD_SANITIZER)
+
 // Tests that the best connection is set to the appropriate IPv4/IPv6 connection
 // and that the statistics in the metric observers are updated correctly.
 TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
@@ -4683,6 +4694,9 @@ TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }
 
+#if !defined(THREAD_SANITIZER)
+// This test provokes TSAN errors. bugs.webrtc.org/11282
+
 // Test that we achieve the expected end-to-end connection time, using a
 // fake clock and simulated latency on the media and signaling paths.
 // We use a TURN<->TURN connection because this is usually the quickest to
@@ -4773,6 +4787,8 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock,
   ClosePeerConnections();
 }
 
+#endif  // !defined(THREAD_SANITIZER)
+
 // Verify that a TurnCustomizer passed in through RTCConfiguration
 // is actually used by the underlying TURN candidate pair.
 // Note that turnport_unittest.cc contains more detailed, lower-level tests.
@@ -5195,6 +5211,9 @@ TEST_P(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }
 
+#if !defined(THREAD_SANITIZER)
+// These tests provokes TSAN errors. See bugs.webrtc.org/11305.
+
 // Test that SetAudioPlayout can be used to disable audio playout from the
 // start, then later enable it. This may be useful, for example, if the caller
 // needs to play a local ringtone until some event occurs, after which it
@@ -5265,6 +5284,8 @@ TEST_P(PeerConnectionIntegrationTest,
   EXPECT_TRUE_WAIT(GetAudioEnergyStat(caller()) > 0, kMaxWaitForFramesMs);
 }
 
+#endif  // !defined(THREAD_SANITIZER)
+
 // Test that SetAudioRecording can be used to disable audio recording from the
 // start, then later enable it. This may be useful, for example, if the caller
 // wants to ensure that no audio resources are active before a certain state
@@ -6097,5 +6118,3 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
 
 }  // namespace
 }  // namespace webrtc
-
-#endif  // if !defined(THREAD_SANITIZER)