diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index ef5294ffcc..245e89d216 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -986,15 +986,23 @@ RTCError PeerConnection::ValidateConfiguration(
 
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::local_streams() {
+  RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
+                                 "Plan SdpSemantics. Please use GetSenders "
+                                 "instead.";
   return local_streams_;
 }
 
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::remote_streams() {
+  RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
+                                 "Plan SdpSemantics. Please use GetReceivers "
+                                 "instead.";
   return remote_streams_;
 }
 
 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
+  RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
+                                 "SdpSemantics. Please use AddTrack instead.";
   TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
   if (IsClosed()) {
     return false;
@@ -1028,6 +1036,9 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
 }
 
 void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
+  RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
+                                 "Plan SdpSemantics. Please use RemoveTrack "
+                                 "instead.";
   TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
   if (!IsClosed()) {
     for (const auto& track : local_stream->GetAudioTracks()) {
@@ -1261,11 +1272,8 @@ RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
 PeerConnection::AddTransceiver(
     rtc::scoped_refptr<MediaStreamTrackInterface> track,
     const RtpTransceiverInit& init) {
-  if (!IsUnifiedPlan()) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INTERNAL_ERROR,
-        "AddTransceiver only supported when Unified Plan is enabled.");
-  }
+  RTC_CHECK(IsUnifiedPlan())
+      << "AddTransceiver is only available with Unified Plan SdpSemantics";
   if (!track) {
     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
   }
@@ -1289,11 +1297,8 @@ PeerConnection::AddTransceiver(cricket::MediaType media_type) {
 RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
 PeerConnection::AddTransceiver(cricket::MediaType media_type,
                                const RtpTransceiverInit& init) {
-  if (!IsUnifiedPlan()) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INTERNAL_ERROR,
-        "AddTransceiver only supported when Unified Plan is enabled.");
-  }
+  RTC_CHECK(IsUnifiedPlan())
+      << "AddTransceiver is only available with Unified Plan SdpSemantics";
   if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
         media_type == cricket::MEDIA_TYPE_VIDEO)) {
     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
@@ -1427,6 +1432,9 @@ rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
 rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
     const std::string& kind,
     const std::string& stream_id) {
+  RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
+                                 "Plan SdpSemantics. Please use AddTransceiver "
+                                 "instead.";
   TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
   if (IsClosed()) {
     return nullptr;
@@ -1506,7 +1514,8 @@ PeerConnection::GetReceiversInternal() const {
 
 std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
 PeerConnection::GetTransceivers() const {
-  RTC_DCHECK(IsUnifiedPlan());
+  RTC_CHECK(IsUnifiedPlan())
+      << "GetTransceivers is only supported with Unified Plan SdpSemantics.";
   std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
   for (auto transceiver : transceivers_) {
     all_transceivers.push_back(transceiver);
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 312407cbad..30dbdc8288 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -2326,11 +2326,9 @@ TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
 
   // Get the remote audio track created on the receiver, so they can be used as
   // GetStats filters.
-  StreamCollectionInterface* remote_streams = callee()->remote_streams();
-  ASSERT_EQ(1u, remote_streams->count());
-  ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size());
-  MediaStreamTrackInterface* remote_audio_track =
-      remote_streams->at(0)->GetAudioTracks()[0];
+  auto receivers = callee()->pc()->GetReceivers();
+  ASSERT_EQ(1u, receivers.size());
+  auto remote_audio_track = receivers[0]->track();
 
   // Get the audio output level stats. Note that the level is not available
   // until an RTCP packet has been received.
diff --git a/pc/test/peerconnectiontestwrapper.cc b/pc/test/peerconnectiontestwrapper.cc
index b34d7f80c5..e2f606756a 100644
--- a/pc/test/peerconnectiontestwrapper.cc
+++ b/pc/test/peerconnectiontestwrapper.cc
@@ -263,9 +263,8 @@ rtc::scoped_refptr<webrtc::MediaStreamInterface>
     PeerConnectionTestWrapper::GetUserMedia(
         bool audio, const webrtc::FakeConstraints& audio_constraints,
         bool video, const webrtc::FakeConstraints& video_constraints) {
-  std::string label = kStreamLabelBase +
-      rtc::ToString<int>(
-          static_cast<int>(peer_connection_->local_streams()->count()));
+  std::string label =
+      kStreamLabelBase + rtc::ToString(num_get_user_media_calls_++);
   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
       peer_connection_factory_->CreateLocalMediaStream(label);
 
diff --git a/pc/test/peerconnectiontestwrapper.h b/pc/test/peerconnectiontestwrapper.h
index aaf9408ed3..338b31df57 100644
--- a/pc/test/peerconnectiontestwrapper.h
+++ b/pc/test/peerconnectiontestwrapper.h
@@ -107,6 +107,7 @@ class PeerConnectionTestWrapper
       peer_connection_factory_;
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
   std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
+  int num_get_user_media_calls_ = 0;
 };
 
 #endif  // PC_TEST_PEERCONNECTIONTESTWRAPPER_H_