diff --git a/pc/peerconnection_jsep_unittest.cc b/pc/peerconnection_jsep_unittest.cc
index ac17df28c2..f77579bc51 100644
--- a/pc/peerconnection_jsep_unittest.cc
+++ b/pc/peerconnection_jsep_unittest.cc
@@ -1228,4 +1228,19 @@ TEST_F(PeerConnectionJsepTest,
 
 // TODO(bugs.webrtc.org/7932): Also test multi-stream case.
 
+// Test that if an RtpTransceiver with a current_direction set is stopped, then
+// current_direction is changed to null.
+TEST_F(PeerConnectionJsepTest, CurrentDirectionResetWhenRtpTransceiverStopped) {
+  auto caller = CreatePeerConnection();
+  auto callee = CreatePeerConnection();
+
+  auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+
+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
+
+  ASSERT_TRUE(transceiver->current_direction());
+  transceiver->Stop();
+  EXPECT_FALSE(transceiver->current_direction());
+}
+
 }  // namespace webrtc
diff --git a/pc/rtptransceiver.cc b/pc/rtptransceiver.cc
index 1f1a850c9f..d4a67883c3 100644
--- a/pc/rtptransceiver.cc
+++ b/pc/rtptransceiver.cc
@@ -207,6 +207,7 @@ void RtpTransceiver::Stop() {
     receiver->internal()->Stop();
   }
   stopped_ = true;
+  current_direction_ = rtc::nullopt;
 }
 
 void RtpTransceiver::SetCodecPreferences(