diff --git a/webrtc/media/base/streamparams.h b/webrtc/media/base/streamparams.h
index 75d82809cf..5d7feafbe3 100644
--- a/webrtc/media/base/streamparams.h
+++ b/webrtc/media/base/streamparams.h
@@ -252,6 +252,13 @@ const StreamParams* GetStream(const StreamParamsVec& streams,
   return found == streams.end() ? nullptr : &(*found);
 }
 
+template <class Condition>
+StreamParams* GetStream(StreamParamsVec& streams, Condition condition) {
+  StreamParamsVec::iterator found =
+      std::find_if(streams.begin(), streams.end(), condition);
+  return found == streams.end() ? nullptr : &(*found);
+}
+
 inline const StreamParams* GetStreamBySsrc(const StreamParamsVec& streams,
                                            uint32_t ssrc) {
   return GetStream(streams,
@@ -261,6 +268,14 @@ inline const StreamParams* GetStreamBySsrc(const StreamParamsVec& streams,
 inline const StreamParams* GetStreamByIds(const StreamParamsVec& streams,
                                           const std::string& groupid,
                                           const std::string& id) {
+  return GetStream(streams, [&groupid, &id](const StreamParams& sp) {
+    return sp.groupid == groupid && sp.id == id;
+  });
+}
+
+inline StreamParams* GetStreamByIds(StreamParamsVec& streams,
+                                    const std::string& groupid,
+                                    const std::string& id) {
   return GetStream(streams,
       [&groupid, &id](const StreamParams& sp) {
         return sp.groupid == groupid && sp.id == id;
diff --git a/webrtc/pc/mediasession.cc b/webrtc/pc/mediasession.cc
index dbd32566d5..0df65297db 100644
--- a/webrtc/pc/mediasession.cc
+++ b/webrtc/pc/mediasession.cc
@@ -448,8 +448,7 @@ static bool AddStreamParams(MediaType media_type,
     if (stream_it->type != media_type)
       continue;  // Wrong media type.
 
-    const StreamParams* param =
-        GetStreamByIds(*current_streams, "", stream_it->id);
+    StreamParams* param = GetStreamByIds(*current_streams, "", stream_it->id);
     // groupid is empty for StreamParams generated using
     // MediaSessionDescriptionFactory.
     if (!param) {
@@ -500,6 +499,10 @@ static bool AddStreamParams(MediaType media_type,
       // This is necessary so that we can use the CNAME for other media types.
       current_streams->push_back(stream_param);
     } else {
+      // Use existing generated SSRCs/groups, but update the sync_label if
+      // necessary. This may be needed if a MediaStreamTrack was moved from one
+      // MediaStream to another.
+      param->sync_label = stream_it->sync_label;
       content_description->AddStream(*param);
     }
   }
diff --git a/webrtc/pc/peerconnection_integrationtest.cc b/webrtc/pc/peerconnection_integrationtest.cc
index d818300c23..49771f60a4 100644
--- a/webrtc/pc/peerconnection_integrationtest.cc
+++ b/webrtc/pc/peerconnection_integrationtest.cc
@@ -217,6 +217,10 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
     return client;
   }
 
+  webrtc::PeerConnectionFactoryInterface* pc_factory() const {
+    return peer_connection_factory_.get();
+  }
+
   webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
 
   // If a signaling message receiver is set (via ConnectFakeSignaling), this
@@ -2773,6 +2777,38 @@ TEST_F(PeerConnectionIntegrationTest, GetSources) {
             contributing_sources[0].source_id());
 }
 
+// Test that if a track is removed and added again with a different stream ID,
+// the new stream ID is successfully communicated in SDP and media continues to
+// flow end-to-end.
+TEST_F(PeerConnectionIntegrationTest, RemoveAndAddTrackWithNewStreamId) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+
+  rtc::scoped_refptr<MediaStreamInterface> stream_1 =
+      caller()->pc_factory()->CreateLocalMediaStream("stream_1");
+  rtc::scoped_refptr<MediaStreamInterface> stream_2 =
+      caller()->pc_factory()->CreateLocalMediaStream("stream_2");
+
+  // Add track using stream 1, do offer/answer.
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+      caller()->CreateLocalAudioTrack();
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
+      caller()->pc()->AddTrack(track, {stream_1.get()});
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait for one audio frame to be received by the callee.
+  ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs);
+
+  // Remove the sender, and create a new one with the new stream.
+  caller()->pc()->RemoveTrack(sender);
+  sender = caller()->pc()->AddTrack(track, {stream_2.get()});
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait for additional audio frames to be received by the callee.
+  ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0,
+                                  kMaxWaitForFramesMs);
+}
+
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)