diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
index 28e9e27..b0c9129 100644
--- a/talk/app/webrtc/audiotrack.cc
+++ b/talk/app/webrtc/audiotrack.cc
@@ -31,7 +31,7 @@
 
 namespace webrtc {
 
-const char MediaStreamTrackInterface::kAudioTrackKind[] = "audio";
+static const char kAudioTrackKind[] = "audio";
 
 AudioTrack::AudioTrack(const std::string& label,
                        AudioSourceInterface* audio_source)
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index cac1735..5911e85 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -100,9 +100,6 @@
     kFailed = 3,  // Track negotiation failed.
   };
 
-  static const char kAudioTrackKind[];
-  static const char kVideoTrackKind[];
-
   virtual std::string kind() const = 0;
   virtual std::string id() const = 0;
   virtual bool enabled() const = 0;
diff --git a/talk/app/webrtc/mediastreamprovider.h b/talk/app/webrtc/mediastreamprovider.h
index a78b55a..1c62daf 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -50,8 +50,8 @@
 // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
 // refactored away eventually, as the classes converge.
 
-// This interface is called by AudioRtpSender/Receivers to change the settings
-// of an audio track connected to certain PeerConnection.
+// This interface is called by AudioTrackHandler classes in mediastreamhandler.h
+// to change the settings of an audio track connected to certain PeerConnection.
 class AudioProviderInterface {
  public:
   // Enable/disable the audio playout of a remote audio track with |ssrc|.
@@ -71,8 +71,9 @@
   virtual ~AudioProviderInterface() {}
 };
 
-// This interface is called by VideoRtpSender/Receivers to change the settings
-// of a video track connected to a certain PeerConnection.
+// This interface is called by VideoTrackHandler classes in mediastreamhandler.h
+// to change the settings of a video track connected to a certain
+// PeerConnection.
 class VideoProviderInterface {
  public:
   virtual bool SetCaptureDevice(uint32_t ssrc,
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 138da5b..fecc19b 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -59,7 +59,6 @@
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
-using webrtc::RtpSenderInterface;
 using webrtc::StreamCollection;
 using webrtc::StunConfigurations;
 using webrtc::TurnConfigurations;
@@ -366,15 +365,25 @@
 }
 
 // Add the stream and RTP data channel info to |session_options|.
-void AddSendStreams(
-    cricket::MediaSessionOptions* session_options,
-    const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
-    const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
-        rtp_data_channels) {
+void SetStreams(cricket::MediaSessionOptions* session_options,
+                rtc::scoped_refptr<StreamCollection> streams,
+                const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
+                    rtp_data_channels) {
   session_options->streams.clear();
-  for (const auto& sender : senders) {
-    session_options->AddSendStream(sender->media_type(), sender->id(),
-                                   sender->stream_id());
+  if (streams != nullptr) {
+    for (size_t i = 0; i < streams->count(); ++i) {
+      MediaStreamInterface* stream = streams->at(i);
+      // For each audio track in the stream, add it to the MediaSessionOptions.
+      for (const auto& track : stream->GetAudioTracks()) {
+        session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, track->id(),
+                                       stream->label());
+      }
+      // For each video track in the stream, add it to the MediaSessionOptions.
+      for (const auto& track : stream->GetVideoTracks()) {
+        session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, track->id(),
+                                       stream->label());
+      }
+    }
   }
 
   // Check for data channels.
@@ -659,6 +668,8 @@
   return remote_streams_;
 }
 
+// TODO(deadbeef): Create RtpSenders immediately here, even if local
+// description hasn't yet been set.
 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
   if (IsClosed()) {
     return false;
@@ -669,46 +680,23 @@
 
   local_streams_->AddStream(local_stream);
 
+  // Find tracks that have already been configured in SDP. This can occur if a
+  // local session description that contains the MSID of these tracks is set
+  // before AddLocalStream is called. It can also occur if the local session
+  // description is not changed and RemoveLocalStream  is called and later
+  // AddLocalStream is called again with the same stream.
   for (const auto& track : local_stream->GetAudioTracks()) {
-    auto sender = FindSenderForTrack(track.get());
-    if (sender == senders_.end()) {
-      // Normal case; we've never seen this track before.
-      AudioRtpSender* new_sender = new AudioRtpSender(
-          track.get(), local_stream->label(), session_.get(), stats_.get());
-      senders_.push_back(new_sender);
-      // If the sender has already been configured in SDP, we call SetSsrc,
-      // which will connect the sender to the underlying transport. This can
-      // occur if a local session description that contains the ID of the sender
-      // is set before AddStream is called. It can also occur if the local
-      // session description is not changed and RemoveStream is called, and
-      // later AddStream is called again with the same stream.
-      const TrackInfo* track_info = FindTrackInfo(
-          local_audio_tracks_, local_stream->label(), track->id());
-      if (track_info) {
-        new_sender->SetSsrc(track_info->ssrc);
-      }
-    } else {
-      // We already have a sender for this track, so just change the stream_id
-      // so that it's correct in the next call to CreateOffer.
-      (*sender)->set_stream_id(local_stream->label());
+    const TrackInfo* track_info =
+        FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      CreateAudioSender(local_stream, track.get(), track_info->ssrc);
     }
   }
   for (const auto& track : local_stream->GetVideoTracks()) {
-    auto sender = FindSenderForTrack(track.get());
-    if (sender == senders_.end()) {
-      // Normal case; we've never seen this track before.
-      VideoRtpSender* new_sender = new VideoRtpSender(
-          track.get(), local_stream->label(), session_.get());
-      senders_.push_back(new_sender);
-      const TrackInfo* track_info = FindTrackInfo(
-          local_video_tracks_, local_stream->label(), track->id());
-      if (track_info) {
-        new_sender->SetSsrc(track_info->ssrc);
-      }
-    } else {
-      // We already have a sender for this track, so just change the stream_id
-      // so that it's correct in the next call to CreateOffer.
-      (*sender)->set_stream_id(local_stream->label());
+    const TrackInfo* track_info =
+        FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      CreateVideoSender(local_stream, track.get(), track_info->ssrc);
     }
   }
 
@@ -718,27 +706,21 @@
 }
 
 // TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely, when we have unified plan SDP.
+// indefinitely.
 void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
   for (const auto& track : local_stream->GetAudioTracks()) {
-    auto sender = FindSenderForTrack(track.get());
-    if (sender == senders_.end()) {
-      LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
-                      << " doesn't exist.";
-      continue;
+    const TrackInfo* track_info =
+        FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      DestroyAudioSender(local_stream, track.get(), track_info->ssrc);
     }
-    (*sender)->Stop();
-    senders_.erase(sender);
   }
   for (const auto& track : local_stream->GetVideoTracks()) {
-    auto sender = FindSenderForTrack(track.get());
-    if (sender == senders_.end()) {
-      LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
-                      << " doesn't exist.";
-      continue;
+    const TrackInfo* track_info =
+        FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      DestroyVideoSender(local_stream, track.get());
     }
-    (*sender)->Stop();
-    senders_.erase(sender);
   }
 
   local_streams_->RemoveStream(local_stream);
@@ -769,21 +751,6 @@
   return DtmfSenderProxy::Create(signaling_thread(), sender.get());
 }
 
-rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
-    const std::string& kind) {
-  RtpSenderInterface* new_sender;
-  if (kind == MediaStreamTrackInterface::kAudioTrackKind) {
-    new_sender = new AudioRtpSender(session_.get(), stats_.get());
-  } else if (kind == MediaStreamTrackInterface::kVideoTrackKind) {
-    new_sender = new VideoRtpSender(session_.get());
-  } else {
-    LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
-    return rtc::scoped_refptr<RtpSenderInterface>();
-  }
-  senders_.push_back(new_sender);
-  return RtpSenderProxy::Create(signaling_thread(), new_sender);
-}
-
 std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
     const {
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders;
@@ -1300,6 +1267,49 @@
   }
 }
 
+void PeerConnection::CreateAudioSender(MediaStreamInterface* stream,
+                                       AudioTrackInterface* audio_track,
+                                       uint32_t ssrc) {
+  senders_.push_back(new AudioRtpSender(audio_track, ssrc, session_.get()));
+  stats_->AddLocalAudioTrack(audio_track, ssrc);
+}
+
+void PeerConnection::CreateVideoSender(MediaStreamInterface* stream,
+                                       VideoTrackInterface* video_track,
+                                       uint32_t ssrc) {
+  senders_.push_back(new VideoRtpSender(video_track, ssrc, session_.get()));
+}
+
+// TODO(deadbeef): Keep RtpSenders around even if track goes away in local
+// description.
+void PeerConnection::DestroyAudioSender(MediaStreamInterface* stream,
+                                        AudioTrackInterface* audio_track,
+                                        uint32_t ssrc) {
+  auto it = FindSenderForTrack(audio_track);
+  if (it == senders_.end()) {
+    LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
+                    << " doesn't exist.";
+    return;
+  } else {
+    (*it)->Stop();
+    senders_.erase(it);
+  }
+  stats_->RemoveLocalAudioTrack(audio_track, ssrc);
+}
+
+void PeerConnection::DestroyVideoSender(MediaStreamInterface* stream,
+                                        VideoTrackInterface* video_track) {
+  auto it = FindSenderForTrack(video_track);
+  if (it == senders_.end()) {
+    LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id()
+                    << " doesn't exist.";
+    return;
+  } else {
+    (*it)->Stop();
+    senders_.erase(it);
+  }
+}
+
 void PeerConnection::OnIceConnectionChange(
     PeerConnectionInterface::IceConnectionState new_state) {
   RTC_DCHECK(signaling_thread()->IsCurrent());
@@ -1375,7 +1385,7 @@
     return false;
   }
 
-  AddSendStreams(session_options, senders_, rtp_data_channels_);
+  SetStreams(session_options, local_streams_, rtp_data_channels_);
   // Offer to receive audio/video if the constraint is not set and there are
   // send streams, or we're currently receiving.
   if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
@@ -1408,7 +1418,7 @@
     return false;
   }
 
-  AddSendStreams(session_options, senders_, rtp_data_channels_);
+  SetStreams(session_options, local_streams_, rtp_data_channels_);
   session_options->bundle_enabled =
       session_options->bundle_enabled &&
       (session_options->has_audio() || session_options->has_video() ||
@@ -1430,7 +1440,8 @@
   TrackInfos* current_tracks = GetRemoteTracks(media_type);
 
   // Find removed tracks. I.e., tracks where the track id or ssrc don't match
-  // the new StreamParam.
+  // the
+  // new StreamParam.
   auto track_it = current_tracks->begin();
   while (track_it != current_tracks->end()) {
     const TrackInfo& info = *track_it;
@@ -1632,44 +1643,62 @@
                                       const std::string& track_id,
                                       uint32_t ssrc,
                                       cricket::MediaType media_type) {
-  RtpSenderInterface* sender = FindSenderById(track_id);
-  if (!sender) {
-    LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
-                    << " has been configured in the local description.";
+  MediaStreamInterface* stream = local_streams_->find(stream_label);
+  if (!stream) {
+    LOG(LS_WARNING) << "An unknown local MediaStream with label "
+                    << stream_label << " has been configured.";
     return;
   }
 
-  if (sender->media_type() != media_type) {
-    LOG(LS_WARNING) << "An RtpSender has been configured in the local"
-                    << " description with an unexpected media type.";
-    return;
+  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+    AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
+    if (!audio_track) {
+      LOG(LS_WARNING) << "An unknown local AudioTrack with id , " << track_id
+                      << " has been configured.";
+      return;
+    }
+    CreateAudioSender(stream, audio_track, ssrc);
+  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+    VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
+    if (!video_track) {
+      LOG(LS_WARNING) << "An unknown local VideoTrack with id , " << track_id
+                      << " has been configured.";
+      return;
+    }
+    CreateVideoSender(stream, video_track, ssrc);
+  } else {
+    RTC_DCHECK(false && "Invalid media type");
   }
-
-  sender->set_stream_id(stream_label);
-  sender->SetSsrc(ssrc);
 }
 
 void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
                                          const std::string& track_id,
                                          uint32_t ssrc,
                                          cricket::MediaType media_type) {
-  RtpSenderInterface* sender = FindSenderById(track_id);
-  if (!sender) {
-    // This is the normal case. I.e., RemoveStream has been called and the
+  MediaStreamInterface* stream = local_streams_->find(stream_label);
+  if (!stream) {
+    // This is the normal case. I.e., RemoveLocalStream has been called and the
     // SessionDescriptions has been renegotiated.
     return;
   }
-
-  // A sender has been removed from the SessionDescription but it's still
-  // associated with the PeerConnection. This only occurs if the SDP doesn't
-  // match with the calls to CreateSender, AddStream and RemoveStream.
-  if (sender->media_type() != media_type) {
-    LOG(LS_WARNING) << "An RtpSender has been configured in the local"
-                    << " description with an unexpected media type.";
-    return;
+  // A track has been removed from the SessionDescription but the MediaStream
+  // is still associated with PeerConnection. This only occurs if the SDP
+  // doesn't match with the calls to AddLocalStream and RemoveLocalStream.
+  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+    AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
+    if (!audio_track) {
+      return;
+    }
+    DestroyAudioSender(stream, audio_track, ssrc);
+  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+    VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
+    if (!video_track) {
+      return;
+    }
+    DestroyVideoSender(stream, video_track);
+  } else {
+    RTC_DCHECK(false && "Invalid media type.");
   }
-
-  sender->SetSsrc(0);
 }
 
 void PeerConnection::UpdateLocalRtpDataChannels(
@@ -1887,15 +1916,6 @@
       DataChannelProxy::Create(signaling_thread(), channel));
 }
 
-RtpSenderInterface* PeerConnection::FindSenderById(const std::string& id) {
-  auto it =
-      std::find_if(senders_.begin(), senders_.end(),
-                   [id](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
-                     return sender->id() == id;
-                   });
-  return it != senders_.end() ? it->get() : nullptr;
-}
-
 std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
 PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
   return std::find_if(
diff --git a/talk/app/webrtc/peerconnection.h b/talk/app/webrtc/peerconnection.h
index c233388..2d388ae 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -91,9 +91,6 @@
   rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
       AudioTrackInterface* track) override;
 
-  rtc::scoped_refptr<RtpSenderInterface> CreateSender(
-      const std::string& kind) override;
-
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
       const override;
   std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
@@ -190,6 +187,12 @@
                             AudioTrackInterface* audio_track);
   void DestroyVideoReceiver(MediaStreamInterface* stream,
                             VideoTrackInterface* video_track);
+  void CreateAudioSender(MediaStreamInterface* stream,
+                         AudioTrackInterface* audio_track,
+                         uint32_t ssrc);
+  void CreateVideoSender(MediaStreamInterface* stream,
+                         VideoTrackInterface* video_track,
+                         uint32_t ssrc);
   void DestroyAudioSender(MediaStreamInterface* stream,
                           AudioTrackInterface* audio_track,
                           uint32_t ssrc);
@@ -325,8 +328,6 @@
   void OnDataChannelOpenMessage(const std::string& label,
                                 const InternalDataChannelInit& config);
 
-  RtpSenderInterface* FindSenderById(const std::string& id);
-
   std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
   FindSenderForTrack(MediaStreamTrackInterface* track);
   std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index abe5e1c..3cf66d6 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -96,7 +96,6 @@
 #if !defined(THREAD_SANITIZER)
 static const int kMaxWaitForStatsMs = 3000;
 #endif
-static const int kMaxWaitForActivationMs = 5000;
 static const int kMaxWaitForFramesMs = 10000;
 static const int kEndAudioFrameCount = 3;
 static const int kEndVideoFrameCount = 3;
@@ -251,7 +250,18 @@
         peer_connection_factory_->CreateLocalMediaStream(stream_label);
 
     if (audio && can_receive_audio()) {
-      stream->AddTrack(CreateLocalAudioTrack(stream_label));
+      FakeConstraints constraints;
+      // Disable highpass filter so that we can get all the test audio frames.
+      constraints.AddMandatory(
+          MediaConstraintsInterface::kHighpassFilter, false);
+      rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+          peer_connection_factory_->CreateAudioSource(&constraints);
+      // TODO(perkj): Test audio source when it is implemented. Currently audio
+      // always use the default input.
+      std::string label = stream_label + kAudioTrackLabelBase;
+      rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+          peer_connection_factory_->CreateAudioTrack(label, source));
+      stream->AddTrack(audio_track);
     }
     if (video && can_receive_video()) {
       stream->AddTrack(CreateLocalVideoTrack(stream_label));
@@ -347,35 +357,6 @@
     data_observer_.reset(new MockDataChannelObserver(data_channel_));
   }
 
-  rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
-      const std::string& stream_label) {
-    FakeConstraints constraints;
-    // Disable highpass filter so that we can get all the test audio frames.
-    constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
-    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
-        peer_connection_factory_->CreateAudioSource(&constraints);
-    // TODO(perkj): Test audio source when it is implemented. Currently audio
-    // always use the default input.
-    std::string label = stream_label + kAudioTrackLabelBase;
-    return peer_connection_factory_->CreateAudioTrack(label, source);
-  }
-
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
-      const std::string& stream_label) {
-    // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
-    FakeConstraints source_constraints = video_constraints_;
-    source_constraints.SetMandatoryMaxFrameRate(10);
-
-    cricket::FakeVideoCapturer* fake_capturer =
-        new webrtc::FakePeriodicVideoCapturer();
-    video_capturers_.push_back(fake_capturer);
-    rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
-        peer_connection_factory_->CreateVideoSource(fake_capturer,
-                                                    &source_constraints);
-    std::string label = stream_label + kVideoTrackLabelBase;
-    return peer_connection_factory_->CreateVideoTrack(label, source);
-  }
-
   DataChannelInterface* data_channel() { return data_channel_; }
   const MockDataChannelObserver* data_observer() const {
     return data_observer_.get();
@@ -690,6 +671,22 @@
     return peer_connection_.get() != nullptr;
   }
 
+  rtc::scoped_refptr<webrtc::VideoTrackInterface>
+  CreateLocalVideoTrack(const std::string stream_label) {
+    // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
+    FakeConstraints source_constraints = video_constraints_;
+    source_constraints.SetMandatoryMaxFrameRate(10);
+
+    cricket::FakeVideoCapturer* fake_capturer =
+        new webrtc::FakePeriodicVideoCapturer();
+    video_capturers_.push_back(fake_capturer);
+    rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
+        peer_connection_factory_->CreateVideoSource(
+            fake_capturer, &source_constraints);
+    std::string label = stream_label + kVideoTrackLabelBase;
+    return peer_connection_factory_->CreateVideoTrack(label, source);
+  }
+
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
       webrtc::PortAllocatorFactoryInterface* factory,
       const MediaConstraintsInterface* constraints) {
@@ -960,11 +957,13 @@
       initiating_client_->AddMediaStream(true, true);
     }
     initiating_client_->Negotiate();
+    const int kMaxWaitForActivationMs = 5000;
     // Assert true is used here since next tests are guaranteed to fail and
     // would eat up 5 seconds.
     ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
     VerifySessionDescriptions();
 
+
     int audio_frame_count = kEndAudioFrameCount;
     // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
     if (!initiating_client_->can_receive_audio() ||
@@ -1574,34 +1573,6 @@
   LocalP2PTest();
 }
 
-// This tests that if we negotiate after calling CreateSender but before we
-// have a track, then set a track later, frames from the newly-set track are
-// received end-to-end.
-TEST_F(JsepPeerConnectionP2PTestClient, EarlyWarmupTest) {
-  ASSERT_TRUE(CreateTestClients());
-  auto audio_sender = initializing_client()->pc()->CreateSender("audio");
-  auto video_sender = initializing_client()->pc()->CreateSender("video");
-  initializing_client()->Negotiate();
-  // Wait for ICE connection to complete, without any tracks.
-  // Note that the receiving client WILL (in HandleIncomingOffer) create
-  // tracks, so it's only the initiator here that's doing early warmup.
-  ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
-  VerifySessionDescriptions();
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
-                 initializing_client()->ice_connection_state(),
-                 kMaxWaitForFramesMs);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
-                 receiving_client()->ice_connection_state(),
-                 kMaxWaitForFramesMs);
-  // Now set the tracks, and expect frames to immediately start flowing.
-  EXPECT_TRUE(
-      audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack("")));
-  EXPECT_TRUE(
-      video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack("")));
-  EXPECT_TRUE_WAIT(FramesNotPending(kEndAudioFrameCount, kEndVideoFrameCount),
-                   kMaxWaitForFramesMs);
-}
-
 class IceServerParsingTest : public testing::Test {
  public:
   // Convenience for parsing a single URL.
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index 317e663..77caa9d 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -337,12 +337,6 @@
       AudioTrackInterface* track) = 0;
 
   // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
-  // |kind| must be "audio" or "video".
-  virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
-      const std::string& kind) {
-    return rtc::scoped_refptr<RtpSenderInterface>();
-  }
-
   virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
       const {
     return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index df5c38b..63163fd 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -2020,10 +2020,8 @@
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
 
   // Remove an audio and video track.
-  pc_->RemoveStream(reference_collection_->at(0));
   rtc::scoped_ptr<SessionDescriptionInterface> desc_2;
   CreateSessionDescriptionAndReference(1, 1, desc_2.accept());
-  pc_->AddStream(reference_collection_->at(0));
   EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
   senders = pc_->GetSenders();
   EXPECT_EQ(2u, senders.size());
diff --git a/talk/app/webrtc/peerconnectionproxy.h b/talk/app/webrtc/peerconnectionproxy.h
index 9b446c0..d207fbb 100644
--- a/talk/app/webrtc/peerconnectionproxy.h
+++ b/talk/app/webrtc/peerconnectionproxy.h
@@ -43,9 +43,6 @@
   PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
   PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
                 CreateDtmfSender, AudioTrackInterface*)
-  PROXY_METHOD1(rtc::scoped_refptr<RtpSenderInterface>,
-                CreateSender,
-                const std::string&)
   PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
                      GetSenders)
   PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
index 5b3ad49..3a78f45 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -29,7 +29,6 @@
 
 #include "talk/app/webrtc/localaudiosource.h"
 #include "talk/app/webrtc/videosourceinterface.h"
-#include "webrtc/base/helpers.h"
 
 namespace webrtc {
 
@@ -60,49 +59,34 @@
 }
 
 AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
-                               const std::string& stream_id,
-                               AudioProviderInterface* provider,
-                               StatsCollector* stats)
+                               uint32_t ssrc,
+                               AudioProviderInterface* provider)
     : id_(track->id()),
-      stream_id_(stream_id),
-      provider_(provider),
-      stats_(stats),
       track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
       cached_track_enabled_(track->enabled()),
       sink_adapter_(new LocalAudioSinkAdapter()) {
-  RTC_DCHECK(provider != nullptr);
   track_->RegisterObserver(this);
   track_->AddSink(sink_adapter_.get());
+  Reconfigure();
 }
 
-AudioRtpSender::AudioRtpSender(AudioProviderInterface* provider,
-                               StatsCollector* stats)
-    : id_(rtc::CreateRandomUuid()),
-      stream_id_(rtc::CreateRandomUuid()),
-      provider_(provider),
-      stats_(stats),
-      sink_adapter_(new LocalAudioSinkAdapter()) {}
-
 AudioRtpSender::~AudioRtpSender() {
+  track_->RemoveSink(sink_adapter_.get());
+  track_->UnregisterObserver(this);
   Stop();
 }
 
 void AudioRtpSender::OnChanged() {
-  RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
-    if (can_send_track()) {
-      SetAudioSend();
-    }
+    Reconfigure();
   }
 }
 
 bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
-  if (stopped_) {
-    LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
-    return false;
-  }
-  if (track && track->kind() != MediaStreamTrackInterface::kAudioTrackKind) {
+  if (track->kind() != "audio") {
     LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
                   << " track.";
     return false;
@@ -110,83 +94,36 @@
   AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
 
   // Detach from old track.
-  if (track_) {
-    track_->RemoveSink(sink_adapter_.get());
-    track_->UnregisterObserver(this);
-  }
-
-  if (can_send_track() && stats_) {
-    stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
-  }
+  track_->RemoveSink(sink_adapter_.get());
+  track_->UnregisterObserver(this);
 
   // Attach to new track.
-  bool prev_can_send_track = can_send_track();
   track_ = audio_track;
-  if (track_) {
-    cached_track_enabled_ = track_->enabled();
-    track_->RegisterObserver(this);
-    track_->AddSink(sink_adapter_.get());
-  }
-
-  // Update audio provider.
-  if (can_send_track()) {
-    SetAudioSend();
-    if (stats_) {
-      stats_->AddLocalAudioTrack(track_.get(), ssrc_);
-    }
-  } else if (prev_can_send_track) {
-    cricket::AudioOptions options;
-    provider_->SetAudioSend(ssrc_, false, options, nullptr);
-  }
+  cached_track_enabled_ = track_->enabled();
+  track_->RegisterObserver(this);
+  track_->AddSink(sink_adapter_.get());
+  Reconfigure();
   return true;
 }
 
-void AudioRtpSender::SetSsrc(uint32_t ssrc) {
-  if (stopped_ || ssrc == ssrc_) {
-    return;
-  }
-  // If we are already sending with a particular SSRC, stop sending.
-  if (can_send_track()) {
-    cricket::AudioOptions options;
-    provider_->SetAudioSend(ssrc_, false, options, nullptr);
-    if (stats_) {
-      stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
-    }
-  }
-  ssrc_ = ssrc;
-  if (can_send_track()) {
-    SetAudioSend();
-    if (stats_) {
-      stats_->AddLocalAudioTrack(track_.get(), ssrc_);
-    }
-  }
-}
-
 void AudioRtpSender::Stop() {
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
-  if (stopped_) {
+  if (!provider_) {
     return;
   }
-  if (track_) {
-    track_->RemoveSink(sink_adapter_.get());
-    track_->UnregisterObserver(this);
-  }
-  if (can_send_track()) {
-    cricket::AudioOptions options;
-    provider_->SetAudioSend(ssrc_, false, options, nullptr);
-    if (stats_) {
-      stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
-    }
-  }
-  stopped_ = true;
+  cricket::AudioOptions options;
+  provider_->SetAudioSend(ssrc_, false, options, nullptr);
+  provider_ = nullptr;
 }
 
-void AudioRtpSender::SetAudioSend() {
-  RTC_DCHECK(!stopped_ && can_send_track());
+void AudioRtpSender::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
   cricket::AudioOptions options;
   if (track_->enabled() && track_->GetSource()) {
     // TODO(xians): Remove this static_cast since we should be able to connect
-    // a remote audio track to a peer connection.
+    // a remote audio track to peer connection.
     options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
   }
 
@@ -199,42 +136,35 @@
 }
 
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
-                               const std::string& stream_id,
+                               uint32_t ssrc,
                                VideoProviderInterface* provider)
     : id_(track->id()),
-      stream_id_(stream_id),
-      provider_(provider),
       track_(track),
+      ssrc_(ssrc),
+      provider_(provider),
       cached_track_enabled_(track->enabled()) {
-  RTC_DCHECK(provider != nullptr);
   track_->RegisterObserver(this);
+  VideoSourceInterface* source = track_->GetSource();
+  if (source) {
+    provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
+  }
+  Reconfigure();
 }
 
-VideoRtpSender::VideoRtpSender(VideoProviderInterface* provider)
-    : id_(rtc::CreateRandomUuid()),
-      stream_id_(rtc::CreateRandomUuid()),
-      provider_(provider) {}
-
 VideoRtpSender::~VideoRtpSender() {
+  track_->UnregisterObserver(this);
   Stop();
 }
 
 void VideoRtpSender::OnChanged() {
-  RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
-    if (can_send_track()) {
-      SetVideoSend();
-    }
+    Reconfigure();
   }
 }
 
 bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
-  if (stopped_) {
-    LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
-    return false;
-  }
-  if (track && track->kind() != MediaStreamTrackInterface::kVideoTrackKind) {
+  if (track->kind() != "video") {
     LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
                   << " track.";
     return false;
@@ -242,72 +172,30 @@
   VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
 
   // Detach from old track.
-  if (track_) {
-    track_->UnregisterObserver(this);
-  }
+  track_->UnregisterObserver(this);
 
   // Attach to new track.
-  bool prev_can_send_track = can_send_track();
   track_ = video_track;
-  if (track_) {
-    cached_track_enabled_ = track_->enabled();
-    track_->RegisterObserver(this);
-  }
-
-  // Update video provider.
-  if (can_send_track()) {
-    VideoSourceInterface* source = track_->GetSource();
-    // TODO(deadbeef): If SetTrack is called with a disabled track, and the
-    // previous track was enabled, this could cause a frame from the new track
-    // to slip out. Really, what we need is for SetCaptureDevice and
-    // SetVideoSend
-    // to be combined into one atomic operation, all the way down to
-    // WebRtcVideoSendStream.
-    provider_->SetCaptureDevice(ssrc_,
-                                source ? source->GetVideoCapturer() : nullptr);
-    SetVideoSend();
-  } else if (prev_can_send_track) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
-    provider_->SetVideoSend(ssrc_, false, nullptr);
-  }
+  cached_track_enabled_ = track_->enabled();
+  track_->RegisterObserver(this);
+  Reconfigure();
   return true;
 }
 
-void VideoRtpSender::SetSsrc(uint32_t ssrc) {
-  if (stopped_ || ssrc == ssrc_) {
-    return;
-  }
-  // If we are already sending with a particular SSRC, stop sending.
-  if (can_send_track()) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
-    provider_->SetVideoSend(ssrc_, false, nullptr);
-  }
-  ssrc_ = ssrc;
-  if (can_send_track()) {
-    VideoSourceInterface* source = track_->GetSource();
-    provider_->SetCaptureDevice(ssrc_,
-                                source ? source->GetVideoCapturer() : nullptr);
-    SetVideoSend();
-  }
-}
-
 void VideoRtpSender::Stop() {
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
-  if (stopped_) {
+  if (!provider_) {
     return;
   }
-  if (track_) {
-    track_->UnregisterObserver(this);
-  }
-  if (can_send_track()) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
-    provider_->SetVideoSend(ssrc_, false, nullptr);
-  }
-  stopped_ = true;
+  provider_->SetCaptureDevice(ssrc_, nullptr);
+  provider_->SetVideoSend(ssrc_, false, nullptr);
+  provider_ = nullptr;
 }
 
-void VideoRtpSender::SetVideoSend() {
-  RTC_DCHECK(!stopped_ && can_send_track());
+void VideoRtpSender::Reconfigure() {
+  if (!provider_) {
+    return;
+  }
   const cricket::VideoOptions* options = nullptr;
   VideoSourceInterface* source = track_->GetSource();
   if (track_->enabled() && source) {
diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h
index d5f88a9..3741909 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/talk/app/webrtc/rtpsender.h
@@ -36,7 +36,6 @@
 
 #include "talk/app/webrtc/mediastreamprovider.h"
 #include "talk/app/webrtc/rtpsenderinterface.h"
-#include "talk/app/webrtc/statscollector.h"
 #include "talk/media/base/audiorenderer.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
@@ -71,15 +70,9 @@
 class AudioRtpSender : public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
-  // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
-  // at the appropriate times.
   AudioRtpSender(AudioTrackInterface* track,
-                 const std::string& stream_id,
-                 AudioProviderInterface* provider,
-                 StatsCollector* stats);
-
-  // Randomly generates id and stream_id.
-  AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
+                 uint32_t ssrc,
+                 AudioProviderInterface* provider);
 
   virtual ~AudioRtpSender();
 
@@ -92,37 +85,18 @@
     return track_.get();
   }
 
-  void SetSsrc(uint32_t ssrc) override;
-
-  uint32_t ssrc() const override { return ssrc_; }
-
-  cricket::MediaType media_type() const override {
-    return cricket::MEDIA_TYPE_AUDIO;
-  }
-
   std::string id() const override { return id_; }
 
-  void set_stream_id(const std::string& stream_id) override {
-    stream_id_ = stream_id;
-  }
-  std::string stream_id() const override { return stream_id_; }
-
   void Stop() override;
 
  private:
-  bool can_send_track() const { return track_ && ssrc_; }
-  // Helper function to construct options for
-  // AudioProviderInterface::SetAudioSend.
-  void SetAudioSend();
+  void Reconfigure();
 
   std::string id_;
-  std::string stream_id_;
-  AudioProviderInterface* provider_;
-  StatsCollector* stats_;
   rtc::scoped_refptr<AudioTrackInterface> track_;
-  uint32_t ssrc_ = 0;
-  bool cached_track_enabled_ = false;
-  bool stopped_ = false;
+  uint32_t ssrc_;
+  AudioProviderInterface* provider_;
+  bool cached_track_enabled_;
 
   // Used to pass the data callback from the |track_| to the other end of
   // cricket::AudioRenderer.
@@ -133,12 +107,9 @@
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
   VideoRtpSender(VideoTrackInterface* track,
-                 const std::string& stream_id,
+                 uint32_t ssrc,
                  VideoProviderInterface* provider);
 
-  // Randomly generates id and stream_id.
-  explicit VideoRtpSender(VideoProviderInterface* provider);
-
   virtual ~VideoRtpSender();
 
   // ObserverInterface implementation
@@ -150,36 +121,18 @@
     return track_.get();
   }
 
-  void SetSsrc(uint32_t ssrc) override;
-
-  uint32_t ssrc() const override { return ssrc_; }
-
-  cricket::MediaType media_type() const override {
-    return cricket::MEDIA_TYPE_VIDEO;
-  }
-
   std::string id() const override { return id_; }
 
-  void set_stream_id(const std::string& stream_id) override {
-    stream_id_ = stream_id;
-  }
-  std::string stream_id() const override { return stream_id_; }
-
   void Stop() override;
 
  private:
-  bool can_send_track() const { return track_ && ssrc_; }
-  // Helper function to construct options for
-  // VideoProviderInterface::SetVideoSend.
-  void SetVideoSend();
+  void Reconfigure();
 
   std::string id_;
-  std::string stream_id_;
-  VideoProviderInterface* provider_;
   rtc::scoped_refptr<VideoTrackInterface> track_;
-  uint32_t ssrc_ = 0;
-  bool cached_track_enabled_ = false;
-  bool stopped_ = false;
+  uint32_t ssrc_;
+  VideoProviderInterface* provider_;
+  bool cached_track_enabled_;
 };
 
 }  // namespace webrtc
diff --git a/talk/app/webrtc/rtpsenderinterface.h b/talk/app/webrtc/rtpsenderinterface.h
index f54e8ca..fca98f2 100644
--- a/talk/app/webrtc/rtpsenderinterface.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -35,7 +35,6 @@
 
 #include "talk/app/webrtc/proxy.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/session/media/mediasession.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
@@ -48,24 +47,10 @@
   virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
   virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
 
-  // Used to set the SSRC of the sender, once a local description has been set.
-  // If |ssrc| is 0, this indiates that the sender should disconnect from the
-  // underlying transport (this occurs if the sender isn't seen in a local
-  // description).
-  virtual void SetSsrc(uint32_t ssrc) = 0;
-  virtual uint32_t ssrc() const = 0;
-
-  // Audio or video sender?
-  virtual cricket::MediaType media_type() const = 0;
-
   // Not to be confused with "mid", this is a field we can temporarily use
   // to uniquely identify a receiver until we implement Unified Plan SDP.
   virtual std::string id() const = 0;
 
-  // TODO(deadbeef): Support one sender having multiple stream ids.
-  virtual void set_stream_id(const std::string& stream_id) = 0;
-  virtual std::string stream_id() const = 0;
-
   virtual void Stop() = 0;
 
  protected:
@@ -76,12 +61,7 @@
 BEGIN_PROXY_MAP(RtpSender)
 PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
-PROXY_METHOD1(void, SetSsrc, uint32_t)
-PROXY_CONSTMETHOD0(uint32_t, ssrc)
-PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
 PROXY_CONSTMETHOD0(std::string, id)
-PROXY_METHOD1(void, set_stream_id, const std::string&)
-PROXY_CONSTMETHOD0(std::string, stream_id)
 PROXY_METHOD0(void, Stop)
 END_PROXY()
 
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
index c987186..c9d7e00 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -48,9 +48,7 @@
 static const char kVideoTrackId[] = "video_1";
 static const char kAudioTrackId[] = "audio_1";
 static const uint32_t kVideoSsrc = 98;
-static const uint32_t kVideoSsrc2 = 100;
 static const uint32_t kAudioSsrc = 99;
-static const uint32_t kAudioSsrc2 = 101;
 
 namespace webrtc {
 
@@ -122,10 +120,8 @@
     audio_track_ = AudioTrack::Create(kAudioTrackId, NULL);
     EXPECT_TRUE(stream_->AddTrack(audio_track_));
     EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-    audio_rtp_sender_ =
-        new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
-                           &audio_provider_, nullptr);
-    audio_rtp_sender_->SetSsrc(kAudioSsrc);
+    audio_rtp_sender_ = new AudioRtpSender(stream_->GetAudioTracks()[0],
+                                           kAudioSsrc, &audio_provider_);
   }
 
   void CreateVideoRtpSender() {
@@ -134,8 +130,7 @@
                     kVideoSsrc, video_track_->GetSource()->GetVideoCapturer()));
     EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
     video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
-                                           stream_->label(), &video_provider_);
-    video_rtp_sender_->SetSsrc(kVideoSsrc);
+                                           kVideoSsrc, &video_provider_);
   }
 
   void DestroyAudioRtpSender() {
@@ -285,206 +280,4 @@
   DestroyAudioRtpReceiver();
 }
 
-// Test that provider methods aren't called without both a track and an SSRC.
-TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(&audio_provider_, nullptr);
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  EXPECT_TRUE(sender->SetTrack(track));
-  EXPECT_TRUE(sender->SetTrack(nullptr));
-  sender->SetSsrc(kAudioSsrc);
-  sender->SetSsrc(0);
-  // Just let it get destroyed and make sure it doesn't call any methods on the
-  // provider interface.
-}
-
-// Test that provider methods aren't called without both a track and an SSRC.
-TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) {
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(&video_provider_);
-  EXPECT_TRUE(sender->SetTrack(video_track_));
-  EXPECT_TRUE(sender->SetTrack(nullptr));
-  sender->SetSsrc(kVideoSsrc);
-  sender->SetSsrc(0);
-  // Just let it get destroyed and make sure it doesn't call any methods on the
-  // provider interface.
-}
-
-// Test that an audio sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set first.
-TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) {
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(&audio_provider_, nullptr);
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  sender->SetSsrc(kAudioSsrc);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  sender->SetTrack(track);
-
-  // Calls expected from destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-}
-
-// Test that an audio sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set last.
-TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) {
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(&audio_provider_, nullptr);
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  sender->SetTrack(track);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  sender->SetSsrc(kAudioSsrc);
-
-  // Calls expected from destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-}
-
-// Test that a video sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set first.
-TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(&video_provider_);
-  sender->SetSsrc(kVideoSsrc);
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
-  sender->SetTrack(video_track_);
-
-  // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
-}
-
-// Test that a video sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set last.
-TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(&video_provider_);
-  sender->SetTrack(video_track_);
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
-  sender->SetSsrc(kVideoSsrc);
-
-  // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
-}
-
-// Test that the sender is disconnected from the provider when its SSRC is
-// set to 0.
-TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) {
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
-  sender->SetSsrc(kAudioSsrc);
-
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-  sender->SetSsrc(0);
-
-  // Make sure it's SetSsrc that called methods on the provider, and not the
-  // destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
-}
-
-// Test that the sender is disconnected from the provider when its SSRC is
-// set to 0.
-TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
-  sender->SetSsrc(kVideoSsrc);
-
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
-  sender->SetSsrc(0);
-
-  // Make sure it's SetSsrc that called methods on the provider, and not the
-  // destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
-  EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
-}
-
-TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
-  sender->SetSsrc(kAudioSsrc);
-
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-  EXPECT_TRUE(sender->SetTrack(nullptr));
-
-  // Make sure it's SetTrack that called methods on the provider, and not the
-  // destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
-}
-
-TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
-  sender->SetSsrc(kVideoSsrc);
-
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
-  EXPECT_TRUE(sender->SetTrack(nullptr));
-
-  // Make sure it's SetTrack that called methods on the provider, and not the
-  // destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
-  EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
-}
-
-TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
-  rtc::scoped_refptr<AudioTrackInterface> track =
-      AudioTrack::Create(kAudioTrackId, nullptr);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
-  rtc::scoped_refptr<AudioRtpSender> sender =
-      new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
-  sender->SetSsrc(kAudioSsrc);
-
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, true, _, _)).Times(1);
-  sender->SetSsrc(kAudioSsrc2);
-
-  // Calls expected from destructor.
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, false, _, _)).Times(1);
-}
-
-TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
-  rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
-  sender->SetSsrc(kVideoSsrc);
-
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc2,
-                               video_track_->GetSource()->GetVideoCapturer()));
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
-  sender->SetSsrc(kVideoSsrc2);
-
-  // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
-  EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
-}
-
 }  // namespace webrtc
diff --git a/talk/app/webrtc/test/fakemediastreamsignaling.h b/talk/app/webrtc/test/fakemediastreamsignaling.h
new file mode 100644
index 0000000..562c4ad
--- /dev/null
+++ b/talk/app/webrtc/test/fakemediastreamsignaling.h
@@ -0,0 +1,140 @@
+/*
+ * libjingle
+ * Copyright 2013 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
+#define TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
+
+#include "talk/app/webrtc/audiotrack.h"
+#include "talk/app/webrtc/mediastreamsignaling.h"
+#include "talk/app/webrtc/videotrack.h"
+
+static const char kStream1[] = "stream1";
+static const char kVideoTrack1[] = "video1";
+static const char kAudioTrack1[] = "audio1";
+
+static const char kStream2[] = "stream2";
+static const char kVideoTrack2[] = "video2";
+static const char kAudioTrack2[] = "audio2";
+
+class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
+                                 public webrtc::MediaStreamSignalingObserver {
+ public:
+  explicit FakeMediaStreamSignaling(cricket::ChannelManager* channel_manager) :
+    webrtc::MediaStreamSignaling(rtc::Thread::Current(), this,
+                                 channel_manager) {
+  }
+
+  void SendAudioVideoStream1() {
+    ClearLocalStreams();
+    AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
+  }
+
+  void SendAudioVideoStream2() {
+    ClearLocalStreams();
+    AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
+  }
+
+  void SendAudioVideoStream1And2() {
+    ClearLocalStreams();
+    AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
+    AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
+  }
+
+  void SendNothing() {
+    ClearLocalStreams();
+  }
+
+  void UseOptionsAudioOnly() {
+    ClearLocalStreams();
+    AddLocalStream(CreateStream(kStream2, kAudioTrack2, ""));
+  }
+
+  void UseOptionsVideoOnly() {
+    ClearLocalStreams();
+    AddLocalStream(CreateStream(kStream2, "", kVideoTrack2));
+  }
+
+  void ClearLocalStreams() {
+    while (local_streams()->count() != 0) {
+      RemoveLocalStream(local_streams()->at(0));
+    }
+  }
+
+  // Implements MediaStreamSignalingObserver.
+  virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {}
+  virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {}
+  virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {}
+  virtual void OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream,
+                                    webrtc::AudioTrackInterface* audio_track,
+                                    uint32_t ssrc) {}
+  virtual void OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream,
+                                    webrtc::VideoTrackInterface* video_track,
+                                    uint32_t ssrc) {}
+  virtual void OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream,
+                                     webrtc::AudioTrackInterface* audio_track,
+                                     uint32_t ssrc) {}
+  virtual void OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream,
+                                     webrtc::VideoTrackInterface* video_track,
+                                     uint32_t ssrc) {}
+  virtual void OnRemoveRemoteAudioTrack(
+      webrtc::MediaStreamInterface* stream,
+      webrtc::AudioTrackInterface* audio_track) {}
+  virtual void OnRemoveRemoteVideoTrack(
+      webrtc::MediaStreamInterface* stream,
+      webrtc::VideoTrackInterface* video_track) {}
+  virtual void OnRemoveLocalAudioTrack(webrtc::MediaStreamInterface* stream,
+                                       webrtc::AudioTrackInterface* audio_track,
+                                       uint32_t ssrc) {}
+  virtual void OnRemoveLocalVideoTrack(
+      webrtc::MediaStreamInterface* stream,
+      webrtc::VideoTrackInterface* video_track) {}
+  virtual void OnRemoveLocalStream(webrtc::MediaStreamInterface* stream) {}
+
+ private:
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateStream(
+      const std::string& stream_label,
+      const std::string& audio_track_id,
+      const std::string& video_track_id) {
+    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
+        webrtc::MediaStream::Create(stream_label));
+
+    if (!audio_track_id.empty()) {
+      rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+          webrtc::AudioTrack::Create(audio_track_id, NULL));
+      stream->AddTrack(audio_track);
+    }
+
+    if (!video_track_id.empty()) {
+      rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
+          webrtc::VideoTrack::Create(video_track_id, NULL));
+      stream->AddTrack(video_track);
+    }
+    return stream;
+  }
+};
+
+#endif  // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
diff --git a/talk/app/webrtc/videotrack.cc b/talk/app/webrtc/videotrack.cc
index 3222735..7c78aea 100644
--- a/talk/app/webrtc/videotrack.cc
+++ b/talk/app/webrtc/videotrack.cc
@@ -31,7 +31,7 @@
 
 namespace webrtc {
 
-const char MediaStreamTrackInterface::kVideoTrackKind[] = "video";
+static const char kVideoTrackKind[] = "video";
 
 VideoTrack::VideoTrack(const std::string& label,
                        VideoSourceInterface* video_source)
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 24454dc..95abeab 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -1747,6 +1747,7 @@
       cricket::GetFirstVideoContent(desc);
   if ((!video_info || video_info->rejected) && video_channel_) {
     SignalVideoChannelDestroyed();
+    const std::string content_name = video_channel_->content_name();
     channel_manager_->DestroyVideoChannel(video_channel_.release());
   }
 
@@ -1754,6 +1755,7 @@
       cricket::GetFirstAudioContent(desc);
   if ((!voice_info || voice_info->rejected) && voice_channel_) {
     SignalVoiceChannelDestroyed();
+    const std::string content_name = voice_channel_->content_name();
     channel_manager_->DestroyVoiceChannel(voice_channel_.release());
   }
 
@@ -1761,6 +1763,7 @@
       cricket::GetFirstDataContent(desc);
   if ((!data_info || data_info->rejected) && data_channel_) {
     SignalDataChannelDestroyed();
+    const std::string content_name = data_channel_->content_name();
     channel_manager_->DestroyDataChannel(data_channel_.release());
   }
 }
diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp
index a8bf57d..41b38b3 100755
--- a/talk/libjingle_tests.gyp
+++ b/talk/libjingle_tests.gyp
@@ -215,6 +215,7 @@
         'app/webrtc/test/fakeconstraints.h',
         'app/webrtc/test/fakedatachannelprovider.h',
         'app/webrtc/test/fakedtlsidentitystore.h',
+        'app/webrtc/test/fakemediastreamsignaling.h',
         'app/webrtc/test/fakeperiodicvideocapturer.h',
         'app/webrtc/test/fakevideotrackrenderer.h',
         'app/webrtc/test/mockpeerconnectionobservers.h',
diff --git a/webrtc/base/helpers.cc b/webrtc/base/helpers.cc
index 1ad5d0e..8e59b64 100644
--- a/webrtc/base/helpers.cc
+++ b/webrtc/base/helpers.cc
@@ -164,20 +164,16 @@
   int seed_;
 };
 
-namespace {
-
 // TODO: Use Base64::Base64Table instead.
-static const char kBase64[64] = {
-    'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M',
-    'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z',
-    'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm',
-    'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z',
-    '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'};
+static const char BASE64[64] = {
+  'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M',
+  'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z',
+  'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm',
+  'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z',
+  '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'
+};
 
-static const char kHex[16] = {'0', '1', '2', '3', '4', '5', '6', '7',
-                              '8', '9', 'a', 'b', 'c', 'd', 'e', 'f'};
-
-static const char kUuidDigit17[4] = {'8', '9', 'a', 'b'};
+namespace {
 
 // This round about way of creating a global RNG is to safe-guard against
 // indeterminant static initialization order.
@@ -236,7 +232,7 @@
 }
 
 bool CreateRandomString(size_t len, std::string* str) {
-  return CreateRandomString(len, kBase64, 64, str);
+  return CreateRandomString(len, BASE64, 64, str);
 }
 
 bool CreateRandomString(size_t len, const std::string& table,
@@ -245,41 +241,6 @@
                             static_cast<int>(table.size()), str);
 }
 
-// Version 4 UUID is of the form:
-// xxxxxxxx-xxxx-4xxx-yxxx-xxxxxxxxxxxx
-// Where 'x' is a hex digit, and 'y' is 8, 9, a or b.
-std::string CreateRandomUuid() {
-  std::string str;
-  scoped_ptr<uint8_t[]> bytes(new uint8_t[31]);
-  if (!Rng().Generate(bytes.get(), 31)) {
-    LOG(LS_ERROR) << "Failed to generate random string!";
-    return str;
-  }
-  str.reserve(36);
-  for (size_t i = 0; i < 8; ++i) {
-    str.push_back(kHex[bytes[i] % 16]);
-  }
-  str.push_back('-');
-  for (size_t i = 8; i < 12; ++i) {
-    str.push_back(kHex[bytes[i] % 16]);
-  }
-  str.push_back('-');
-  str.push_back('4');
-  for (size_t i = 12; i < 15; ++i) {
-    str.push_back(kHex[bytes[i] % 16]);
-  }
-  str.push_back('-');
-  str.push_back(kUuidDigit17[bytes[15] % 4]);
-  for (size_t i = 16; i < 19; ++i) {
-    str.push_back(kHex[bytes[i] % 16]);
-  }
-  str.push_back('-');
-  for (size_t i = 19; i < 31; ++i) {
-    str.push_back(kHex[bytes[i] % 16]);
-  }
-  return str;
-}
-
 uint32_t CreateRandomId() {
   uint32_t id;
   if (!Rng().Generate(&id, sizeof(id))) {
diff --git a/webrtc/base/helpers.h b/webrtc/base/helpers.h
index 0e79373..102c08b 100644
--- a/webrtc/base/helpers.h
+++ b/webrtc/base/helpers.h
@@ -39,9 +39,6 @@
 bool CreateRandomString(size_t length, const std::string& table,
                         std::string* str);
 
-// Generates a (cryptographically) random UUID version 4 string.
-std::string CreateRandomUuid();
-
 // Generates a random id.
 uint32_t CreateRandomId();
 
diff --git a/webrtc/base/helpers_unittest.cc b/webrtc/base/helpers_unittest.cc
index 83cc685..6ea0167 100644
--- a/webrtc/base/helpers_unittest.cc
+++ b/webrtc/base/helpers_unittest.cc
@@ -43,23 +43,16 @@
   EXPECT_EQ(256U, random2.size());
 }
 
-TEST_F(RandomTest, TestCreateRandomUuid) {
-  std::string random = CreateRandomUuid();
-  EXPECT_EQ(36U, random.size());
-}
-
 TEST_F(RandomTest, TestCreateRandomForTest) {
   // Make sure we get the output we expect.
   SetRandomTestMode(true);
   EXPECT_EQ(2154761789U, CreateRandomId());
   EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
-  EXPECT_EQ("41706e92-cdd3-46d9-a22d-8ff1737ffb11", CreateRandomUuid());
 
   // Reset and make sure we get the same output.
   SetRandomTestMode(true);
   EXPECT_EQ(2154761789U, CreateRandomId());
   EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
-  EXPECT_EQ("41706e92-cdd3-46d9-a22d-8ff1737ffb11", CreateRandomUuid());
 
   // Test different character sets.
   SetRandomTestMode(true);