diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
index b0c91296f9..63bd87cb41 100644
--- a/talk/app/webrtc/audiotrack.cc
+++ b/talk/app/webrtc/audiotrack.cc
@@ -31,7 +31,7 @@
 
 namespace webrtc {
 
-static const char kAudioTrackKind[] = "audio";
+const char MediaStreamTrackInterface::kAudioKind[] = "audio";
 
 AudioTrack::AudioTrack(const std::string& label,
                        AudioSourceInterface* audio_source)
@@ -40,7 +40,7 @@ AudioTrack::AudioTrack(const std::string& label,
 }
 
 std::string AudioTrack::kind() const {
-  return kAudioTrackKind;
+  return kAudioKind;
 }
 
 rtc::scoped_refptr<AudioTrack> AudioTrack::Create(
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index 5911e85e8e..00782cb2a5 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -100,6 +100,9 @@ class MediaStreamTrackInterface : public rtc::RefCountInterface,
     kFailed = 3,  // Track negotiation failed.
   };
 
+  static const char kAudioKind[];
+  static const char kVideoKind[];
+
   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 1c62daf9f1..a78b55a68c 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -50,8 +50,8 @@ namespace webrtc {
 // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
 // refactored away eventually, as the classes converge.
 
-// This interface is called by AudioTrackHandler classes in mediastreamhandler.h
-// to change the settings of an audio track connected to certain PeerConnection.
+// This interface is called by AudioRtpSender/Receivers 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|.
@@ -71,9 +71,8 @@ class AudioProviderInterface {
   virtual ~AudioProviderInterface() {}
 };
 
-// This interface is called by VideoTrackHandler classes in mediastreamhandler.h
-// to change the settings of a video track connected to a certain
-// PeerConnection.
+// This interface is called by VideoRtpSender/Receivers 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 caa892d00b..f0db1397aa 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -60,6 +60,7 @@ using webrtc::DataChannel;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
+using webrtc::RtpSenderInterface;
 using webrtc::StreamCollection;
 using webrtc::StunConfigurations;
 using webrtc::TurnConfigurations;
@@ -400,25 +401,15 @@ bool IsValidOfferToReceiveMedia(int value) {
 }
 
 // Add the stream and RTP data channel info to |session_options|.
-void SetStreams(cricket::MediaSessionOptions* session_options,
-                rtc::scoped_refptr<StreamCollection> streams,
-                const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
-                    rtp_data_channels) {
+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) {
   session_options->streams.clear();
-  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());
-      }
-    }
+  for (const auto& sender : senders) {
+    session_options->AddSendStream(sender->media_type(), sender->id(),
+                                   sender->stream_id());
   }
 
   // Check for data channels.
@@ -733,8 +724,6 @@ PeerConnection::remote_streams() {
   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;
@@ -745,23 +734,46 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
 
   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()) {
-    const TrackInfo* track_info =
-        FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
-    if (track_info) {
-      CreateAudioSender(local_stream, track.get(), track_info->ssrc);
+    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());
     }
   }
   for (const auto& track : local_stream->GetVideoTracks()) {
-    const TrackInfo* track_info =
-        FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
-    if (track_info) {
-      CreateVideoSender(local_stream, track.get(), track_info->ssrc);
+    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());
     }
   }
 
@@ -771,21 +783,27 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
 }
 
 // TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely.
+// indefinitely, when we have unified plan SDP.
 void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
   for (const auto& track : local_stream->GetAudioTracks()) {
-    const TrackInfo* track_info =
-        FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
-    if (track_info) {
-      DestroyAudioSender(local_stream, track.get(), track_info->ssrc);
+    auto sender = FindSenderForTrack(track.get());
+    if (sender == senders_.end()) {
+      LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+                      << " doesn't exist.";
+      continue;
     }
+    (*sender)->Stop();
+    senders_.erase(sender);
   }
   for (const auto& track : local_stream->GetVideoTracks()) {
-    const TrackInfo* track_info =
-        FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
-    if (track_info) {
-      DestroyVideoSender(local_stream, track.get());
+    auto sender = FindSenderForTrack(track.get());
+    if (sender == senders_.end()) {
+      LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+                      << " doesn't exist.";
+      continue;
     }
+    (*sender)->Stop();
+    senders_.erase(sender);
   }
 
   local_streams_->RemoveStream(local_stream);
@@ -816,6 +834,21 @@ rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
   return DtmfSenderProxy::Create(signaling_thread(), sender.get());
 }
 
+rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
+    const std::string& kind) {
+  RtpSenderInterface* new_sender;
+  if (kind == MediaStreamTrackInterface::kAudioKind) {
+    new_sender = new AudioRtpSender(session_.get(), stats_.get());
+  } else if (kind == MediaStreamTrackInterface::kVideoKind) {
+    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;
@@ -1324,49 +1357,6 @@ void PeerConnection::DestroyVideoReceiver(MediaStreamInterface* stream,
   }
 }
 
-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());
@@ -1442,7 +1432,7 @@ bool PeerConnection::GetOptionsForOffer(
     return false;
   }
 
-  SetStreams(session_options, local_streams_, rtp_data_channels_);
+  AddSendStreams(session_options, senders_, 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) {
@@ -1475,7 +1465,7 @@ bool PeerConnection::GetOptionsForAnswer(
     return false;
   }
 
-  SetStreams(session_options, local_streams_, rtp_data_channels_);
+  AddSendStreams(session_options, senders_, rtp_data_channels_);
   session_options->bundle_enabled =
       session_options->bundle_enabled &&
       (session_options->has_audio() || session_options->has_video() ||
@@ -1503,8 +1493,7 @@ void PeerConnection::UpdateRemoteStreamsList(
   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;
@@ -1706,62 +1695,44 @@ void PeerConnection::OnLocalTrackSeen(const std::string& stream_label,
                                       const std::string& track_id,
                                       uint32_t ssrc,
                                       cricket::MediaType media_type) {
-  MediaStreamInterface* stream = local_streams_->find(stream_label);
-  if (!stream) {
-    LOG(LS_WARNING) << "An unknown local MediaStream with label "
-                    << stream_label << " has been configured.";
+  RtpSenderInterface* sender = FindSenderById(track_id);
+  if (!sender) {
+    LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
+                    << " has been configured in the local description.";
     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");
+  if (sender->media_type() != media_type) {
+    LOG(LS_WARNING) << "An RtpSender has been configured in the local"
+                    << " description with an unexpected media type.";
+    return;
   }
+
+  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) {
-  MediaStreamInterface* stream = local_streams_->find(stream_label);
-  if (!stream) {
-    // This is the normal case. I.e., RemoveLocalStream has been called and the
+  RtpSenderInterface* sender = FindSenderById(track_id);
+  if (!sender) {
+    // This is the normal case. I.e., RemoveStream has been called and the
     // SessionDescriptions has been renegotiated.
     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.");
+
+  // 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;
   }
+
+  sender->SetSsrc(0);
 }
 
 void PeerConnection::UpdateLocalRtpDataChannels(
@@ -1979,6 +1950,15 @@ void PeerConnection::OnDataChannelOpenMessage(
       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 cd6f67121f..2588020350 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -101,6 +101,9 @@ class PeerConnection : public PeerConnectionInterface,
   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()
@@ -197,12 +200,6 @@ class PeerConnection : public PeerConnectionInterface,
                             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);
@@ -342,6 +339,8 @@ class PeerConnection : public PeerConnectionInterface,
   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 6307bfc6d8..5aba2460c5 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -98,6 +98,7 @@ static const int kMaxWaitMs = 10000;
 #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;
@@ -288,18 +289,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
         peer_connection_factory_->CreateLocalMediaStream(stream_label);
 
     if (audio && can_receive_audio()) {
-      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);
+      stream->AddTrack(CreateLocalAudioTrack(stream_label));
     }
     if (video && can_receive_video()) {
       stream->AddTrack(CreateLocalVideoTrack(stream_label));
@@ -401,6 +391,35 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     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();
@@ -715,22 +734,6 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     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) {
@@ -1011,7 +1014,6 @@ class MAYBE_JsepPeerConnectionP2PTestClient : public testing::Test {
       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);
@@ -1653,6 +1655,34 @@ TEST_F(MAYBE_JsepPeerConnectionP2PTestClient,
   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(MAYBE_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 b7814e5830..723d643784 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -333,6 +333,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
       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 13fcbc8f14..e3935f7410 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -2037,8 +2037,10 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
   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 d207fbbdd8..9b446c057b 100644
--- a/talk/app/webrtc/peerconnectionproxy.h
+++ b/talk/app/webrtc/peerconnectionproxy.h
@@ -43,6 +43,9 @@ BEGIN_PROXY_MAP(PeerConnection)
   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 3a78f4598a..ea10b7b33f 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -29,6 +29,7 @@
 
 #include "talk/app/webrtc/localaudiosource.h"
 #include "talk/app/webrtc/videosourceinterface.h"
+#include "webrtc/base/helpers.h"
 
 namespace webrtc {
 
@@ -59,34 +60,49 @@ void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
 }
 
 AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
-                               uint32_t ssrc,
-                               AudioProviderInterface* provider)
+                               const std::string& stream_id,
+                               AudioProviderInterface* provider,
+                               StatsCollector* stats)
     : id_(track->id()),
-      track_(track),
-      ssrc_(ssrc),
+      stream_id_(stream_id),
       provider_(provider),
+      stats_(stats),
+      track_(track),
       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();
-    Reconfigure();
+    if (can_send_track()) {
+      SetAudioSend();
+    }
   }
 }
 
 bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
-  if (track->kind() != "audio") {
+  if (stopped_) {
+    LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
+    return false;
+  }
+  if (track && track->kind() != MediaStreamTrackInterface::kAudioKind) {
     LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
                   << " track.";
     return false;
@@ -94,36 +110,83 @@ bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
   AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
 
   // Detach from old track.
-  track_->RemoveSink(sink_adapter_.get());
-  track_->UnregisterObserver(this);
+  if (track_) {
+    track_->RemoveSink(sink_adapter_.get());
+    track_->UnregisterObserver(this);
+  }
+
+  if (can_send_track() && stats_) {
+    stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
+  }
 
   // Attach to new track.
+  bool prev_can_send_track = can_send_track();
   track_ = audio_track;
-  cached_track_enabled_ = track_->enabled();
-  track_->RegisterObserver(this);
-  track_->AddSink(sink_adapter_.get());
-  Reconfigure();
+  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);
+  }
   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 (!provider_) {
+  if (stopped_) {
     return;
   }
-  cricket::AudioOptions options;
-  provider_->SetAudioSend(ssrc_, false, options, nullptr);
-  provider_ = nullptr;
+  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;
 }
 
-void AudioRtpSender::Reconfigure() {
-  if (!provider_) {
-    return;
-  }
+void AudioRtpSender::SetAudioSend() {
+  RTC_DCHECK(!stopped_ && can_send_track());
   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 peer connection.
+    // a remote audio track to a peer connection.
     options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
   }
 
@@ -136,35 +199,42 @@ void AudioRtpSender::Reconfigure() {
 }
 
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
-                               uint32_t ssrc,
+                               const std::string& stream_id,
                                VideoProviderInterface* provider)
     : id_(track->id()),
-      track_(track),
-      ssrc_(ssrc),
+      stream_id_(stream_id),
       provider_(provider),
+      track_(track),
       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();
-    Reconfigure();
+    if (can_send_track()) {
+      SetVideoSend();
+    }
   }
 }
 
 bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
-  if (track->kind() != "video") {
+  if (stopped_) {
+    LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
+    return false;
+  }
+  if (track && track->kind() != MediaStreamTrackInterface::kVideoKind) {
     LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
                   << " track.";
     return false;
@@ -172,30 +242,72 @@ bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
   VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
 
   // Detach from old track.
-  track_->UnregisterObserver(this);
+  if (track_) {
+    track_->UnregisterObserver(this);
+  }
 
   // Attach to new track.
+  bool prev_can_send_track = can_send_track();
   track_ = video_track;
-  cached_track_enabled_ = track_->enabled();
-  track_->RegisterObserver(this);
-  Reconfigure();
+  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);
+  }
   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 (!provider_) {
+  if (stopped_) {
     return;
   }
-  provider_->SetCaptureDevice(ssrc_, nullptr);
-  provider_->SetVideoSend(ssrc_, false, nullptr);
-  provider_ = nullptr;
+  if (track_) {
+    track_->UnregisterObserver(this);
+  }
+  if (can_send_track()) {
+    provider_->SetCaptureDevice(ssrc_, nullptr);
+    provider_->SetVideoSend(ssrc_, false, nullptr);
+  }
+  stopped_ = true;
 }
 
-void VideoRtpSender::Reconfigure() {
-  if (!provider_) {
-    return;
-  }
+void VideoRtpSender::SetVideoSend() {
+  RTC_DCHECK(!stopped_ && can_send_track());
   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 3741909323..d5f88a941a 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/talk/app/webrtc/rtpsender.h
@@ -36,6 +36,7 @@
 
 #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"
@@ -70,9 +71,15 @@ class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
 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,
-                 uint32_t ssrc,
-                 AudioProviderInterface* provider);
+                 const std::string& stream_id,
+                 AudioProviderInterface* provider,
+                 StatsCollector* stats);
+
+  // Randomly generates id and stream_id.
+  AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
 
   virtual ~AudioRtpSender();
 
@@ -85,18 +92,37 @@ class AudioRtpSender : public ObserverInterface,
     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:
-  void Reconfigure();
+  bool can_send_track() const { return track_ && ssrc_; }
+  // Helper function to construct options for
+  // AudioProviderInterface::SetAudioSend.
+  void SetAudioSend();
 
   std::string id_;
-  rtc::scoped_refptr<AudioTrackInterface> track_;
-  uint32_t ssrc_;
+  std::string stream_id_;
   AudioProviderInterface* provider_;
-  bool cached_track_enabled_;
+  StatsCollector* stats_;
+  rtc::scoped_refptr<AudioTrackInterface> track_;
+  uint32_t ssrc_ = 0;
+  bool cached_track_enabled_ = false;
+  bool stopped_ = false;
 
   // Used to pass the data callback from the |track_| to the other end of
   // cricket::AudioRenderer.
@@ -107,9 +133,12 @@ class VideoRtpSender : public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
   VideoRtpSender(VideoTrackInterface* track,
-                 uint32_t ssrc,
+                 const std::string& stream_id,
                  VideoProviderInterface* provider);
 
+  // Randomly generates id and stream_id.
+  explicit VideoRtpSender(VideoProviderInterface* provider);
+
   virtual ~VideoRtpSender();
 
   // ObserverInterface implementation
@@ -121,18 +150,36 @@ class VideoRtpSender : public ObserverInterface,
     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:
-  void Reconfigure();
+  bool can_send_track() const { return track_ && ssrc_; }
+  // Helper function to construct options for
+  // VideoProviderInterface::SetVideoSend.
+  void SetVideoSend();
 
   std::string id_;
-  rtc::scoped_refptr<VideoTrackInterface> track_;
-  uint32_t ssrc_;
+  std::string stream_id_;
   VideoProviderInterface* provider_;
-  bool cached_track_enabled_;
+  rtc::scoped_refptr<VideoTrackInterface> track_;
+  uint32_t ssrc_ = 0;
+  bool cached_track_enabled_ = false;
+  bool stopped_ = false;
 };
 
 }  // namespace webrtc
diff --git a/talk/app/webrtc/rtpsenderinterface.h b/talk/app/webrtc/rtpsenderinterface.h
index fca98f21db..f54e8ca090 100644
--- a/talk/app/webrtc/rtpsenderinterface.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -35,6 +35,7 @@
 
 #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"
 
@@ -47,10 +48,24 @@ class RtpSenderInterface : public rtc::RefCountInterface {
   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:
@@ -61,7 +76,12 @@ class RtpSenderInterface : public rtc::RefCountInterface {
 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 c9d7e008c3..c9871864bd 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -48,7 +48,9 @@ static const char kStreamLabel1[] = "local_stream_1";
 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 {
 
@@ -120,8 +122,10 @@ class RtpSenderReceiverTest : public testing::Test {
     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],
-                                           kAudioSsrc, &audio_provider_);
+    audio_rtp_sender_ =
+        new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
+                           &audio_provider_, nullptr);
+    audio_rtp_sender_->SetSsrc(kAudioSsrc);
   }
 
   void CreateVideoRtpSender() {
@@ -130,7 +134,8 @@ class RtpSenderReceiverTest : public testing::Test {
                     kVideoSsrc, video_track_->GetSource()->GetVideoCapturer()));
     EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
     video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
-                                           kVideoSsrc, &video_provider_);
+                                           stream_->label(), &video_provider_);
+    video_rtp_sender_->SetSsrc(kVideoSsrc);
   }
 
   void DestroyAudioRtpSender() {
@@ -280,4 +285,206 @@ TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) {
   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/videotrack.cc b/talk/app/webrtc/videotrack.cc
index 7c78aea91f..f138240068 100644
--- a/talk/app/webrtc/videotrack.cc
+++ b/talk/app/webrtc/videotrack.cc
@@ -31,7 +31,7 @@
 
 namespace webrtc {
 
-static const char kVideoTrackKind[] = "video";
+const char MediaStreamTrackInterface::kVideoKind[] = "video";
 
 VideoTrack::VideoTrack(const std::string& label,
                        VideoSourceInterface* video_source)
@@ -47,7 +47,7 @@ VideoTrack::~VideoTrack() {
 }
 
 std::string VideoTrack::kind() const {
-  return kVideoTrackKind;
+  return kVideoKind;
 }
 
 void VideoTrack::AddRenderer(VideoRendererInterface* renderer) {
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 420477dbed..ecc7c605ea 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -1739,7 +1739,6 @@ void WebRtcSession::RemoveUnusedChannels(const SessionDescription* desc) {
       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());
   }
 
@@ -1747,7 +1746,6 @@ void WebRtcSession::RemoveUnusedChannels(const SessionDescription* desc) {
       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());
   }
 
@@ -1755,7 +1753,6 @@ void WebRtcSession::RemoveUnusedChannels(const SessionDescription* desc) {
       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/webrtc/base/helpers.cc b/webrtc/base/helpers.cc
index 8e59b6410c..1ad5d0e12b 100644
--- a/webrtc/base/helpers.cc
+++ b/webrtc/base/helpers.cc
@@ -164,17 +164,21 @@ class TestRandomGenerator : public RandomGenerator {
   int seed_;
 };
 
-// TODO: Use Base64::Base64Table instead.
-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', '+', '/'
-};
-
 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 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'};
+
 // This round about way of creating a global RNG is to safe-guard against
 // indeterminant static initialization order.
 scoped_ptr<RandomGenerator>& GetGlobalRng() {
@@ -232,7 +236,7 @@ bool CreateRandomString(size_t len,
 }
 
 bool CreateRandomString(size_t len, std::string* str) {
-  return CreateRandomString(len, BASE64, 64, str);
+  return CreateRandomString(len, kBase64, 64, str);
 }
 
 bool CreateRandomString(size_t len, const std::string& table,
@@ -241,6 +245,41 @@ bool CreateRandomString(size_t len, const std::string& table,
                             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 102c08bd0d..0e7937362a 100644
--- a/webrtc/base/helpers.h
+++ b/webrtc/base/helpers.h
@@ -39,6 +39,9 @@ bool CreateRandomString(size_t length, std::string* str);
 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 6ea0167e98..83cc685919 100644
--- a/webrtc/base/helpers_unittest.cc
+++ b/webrtc/base/helpers_unittest.cc
@@ -43,16 +43,23 @@ TEST_F(RandomTest, TestCreateRandomString) {
   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);