CastExtension Impl for Chromoting Host

remoting/host/cast_extension_session.cc

+// Copyright 2014 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "remoting/host/cast_extension_session.h"
+
+#include "base/bind.h"
+#include "base/json/json_reader.h"
+#include "base/json/json_writer.h"
+#include "base/logging.h"
+#include "base/synchronization/waitable_event.h"
+#include "net/url_request/url_request_context_getter.h"
+#include "remoting/host/cast_video_capturer_adapter.h"
+#include "remoting/host/chromium_port_allocator_factory.h"
+#include "remoting/host/client_session.h"
+#include "remoting/proto/control.pb.h"
+#include "remoting/protocol/client_stub.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
+#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h"
+#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h"
+
+namespace remoting {
+
+// Used as the type attribute of all Cast protocol::ExtensionMessages.
+const char kExtensionMessageType[] = "cast_message";
+
+// Top-level keys used in all extension messages between host and client.
+// Must keep synced with webapp.
+const char kTopLevelData[] = "chromoting_data";
+const char kTopLevelSubject[] = "subject";
+
+// Keys used to describe the subject of a cast extension message. WebRTC-related
+// message subjects are prepended with "webrtc_".
+// Must keep synced with webapp.
+const char kSubjectReady[] = "ready";
+const char kSubjectTest[] = "test";
+const char kSubjectNewCandidate[] = "webrtc_candidate";
+const char kSubjectOffer[] = "webrtc_offer";
+const char kSubjectAnswer[] = "webrtc_answer";
+
+// WebRTC headers used inside cast extension messages.

[Wez, 2014/08/14 19:22:03]

Inside messages with subject = webrtc_* ?

[aiguha, 2014/08/15 04:11:39]

Done.

+const char kWebRtcCandidate[] = "candidate";
+const char kWebRtcSessionDescType[] = "type";
+const char kWebRtcSessionDescSDP[] = "sdp";
+const char kWebRtcSDPMid[] = "sdpMid";
+const char kWebRtcSDPMLineIndex[] = "sdpMLineIndex";
+
+// Media labels used over the PeerConnection.
+const char kVideoLabel[] = "cast_video_label";
+const char kStreamLabel[] = "stream_label";
+
+// Default STUN server used to construct
+// webrtc::PeerConnectionInterface::RTCConfiguration for the PeerConnection.
+const char kDefaultStunURI[] = "stun:stun.l.google.com:19302";
+
+const char kWorkerThreadName[] = "CastExtensionSessionWorkerThread";
+
+// Interval between each call to PollPeerConnectionStats().
+const int kStatsLogIntervalSec = 10;
+
+// Minimum frame rate for video streaming over the PeerConnection in frames per
+// second, added as a media constraint when constructing the video source for
+// the Peer Connection.
+const int kMinFrameRate = 5;

[Wez, 2014/08/14 19:22:03]

nit: kMinimumFps or kMinFramesPerSecond

[aiguha, 2014/08/15 04:11:39]

Done.

+
+// A webrtc::SetSessionDescriptionObserver implementation used to receive the
+// results of setting local and remote descriptions of the PeerConnection.
+class CastSetSessionDescriptionObserver
+    : public webrtc::SetSessionDescriptionObserver {
+ public:
+  static CastSetSessionDescriptionObserver* Create() {
+    return new rtc::RefCountedObject<CastSetSessionDescriptionObserver>();
+  }
+  virtual void OnSuccess() OVERRIDE {
+    VLOG(1) << "Setting session description succeeded.";
+  }
+  virtual void OnFailure(const std::string& error) OVERRIDE {
+    LOG(ERROR) << "Setting session description failed: " << error;
+  }
+
+ protected:
+  CastSetSessionDescriptionObserver() {}
+  virtual ~CastSetSessionDescriptionObserver() {}
+
+  DISALLOW_COPY_AND_ASSIGN(CastSetSessionDescriptionObserver);
+};
+
+// A webrtc::CreateSessionDescriptionObserver implementation used to receive the
+// results of creating descriptions for this end of the PeerConnection.
+class CastCreateSessionDescriptionObserver
+    : public webrtc::CreateSessionDescriptionObserver {
+ public:
+  static CastCreateSessionDescriptionObserver* Create(
+      CastExtensionSession* session) {
+    return new rtc::RefCountedObject<CastCreateSessionDescriptionObserver>(
+        session);
+  }
+  virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc) OVERRIDE {
+    if (session_ == NULL) {
+      LOG(ERROR) << "No Session, cannot create session description.";
+      return;
+    }
+    session_->OnCreateSessionDescription(desc);
+  }
+  virtual void OnFailure(const std::string& error) OVERRIDE {
+    if (session_ == NULL) {
+      LOG(ERROR) << "No Session, cannot create session description.";
+      return;
+    }
+    session_->OnCreateSessionDescriptionFailure(error);
+  }
+
+ protected:
+  explicit CastCreateSessionDescriptionObserver(CastExtensionSession* session)
+      : session_(session) {}
+  virtual ~CastCreateSessionDescriptionObserver() {}
+
+ private:
+  CastExtensionSession* session_;
+
+  DISALLOW_COPY_AND_ASSIGN(CastCreateSessionDescriptionObserver);
+};
+
+// A webrtc::StatsObserver implementation used to receive statistics about the
+// current PeerConnection.
+class CastStatsObserver : public webrtc::StatsObserver {
+ public:
+  static CastStatsObserver* Create() {
+    return new rtc::RefCountedObject<CastStatsObserver>();
+  }
+
+  virtual void OnComplete(
+      const std::vector<webrtc::StatsReport>& reports) OVERRIDE {
+    typedef webrtc::StatsReport::Values::iterator ValuesIterator;
+
+    VLOG(1) << "Received " << reports.size() << " new StatsReports.";
+
+    int index;
+    std::vector<webrtc::StatsReport>::const_iterator it;
+    for (it = reports.begin(), index = 0; it != reports.end(); ++it, ++index) {
+      webrtc::StatsReport::Values v = it->values;
+      VLOG(1) << "Report " << index << ":";
+      for (ValuesIterator vIt = v.begin(); vIt != v.end(); ++vIt) {
+        VLOG(1) << "Stat: " << vIt->name << "=" << vIt->value << ".";
+      }
+    }
+  }
+
+ protected:
+  CastStatsObserver() {}
+  virtual ~CastStatsObserver() {}
+
+  DISALLOW_COPY_AND_ASSIGN(CastStatsObserver);
+};
+
+// TODO(aiguha): Fix PeerConnnection-related tear down crash caused by premature
+// destruction of cricket::CaptureManager (which occurs on releasing
+// |peer_conn_factory_|). Tracked by: crbug.com/.

[Wez, 2014/08/14 19:22:03]

Suggest: See crbug.com/<number>

[Wez, 2014/08/14 19:22:03]

Bug #?

[aiguha, 2014/08/15 04:11:39]

Done.

[aiguha, 2014/08/15 04:11:39]

Done.

+CastExtensionSession::~CastExtensionSession() {
+  DCHECK(caller_task_runner_->BelongsToCurrentThread());
+  CleanupPeerConnection();
+}
+
+// static
+scoped_ptr<CastExtensionSession> CastExtensionSession::Create(
+    scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner,
+    scoped_refptr<net::URLRequestContextGetter> url_request_context_getter,
+    const protocol::NetworkSettings& network_settings,
+    ClientSessionControl* client_session_control,
+    protocol::ClientStub* client_stub) {
+  scoped_ptr<CastExtensionSession> cast_extension_session(
+      new CastExtensionSession(caller_task_runner,
+                               url_request_context_getter,
+                               network_settings,
+                               client_session_control,
+                               client_stub));
+  if (!cast_extension_session->WrapTasksAndSave()) {
+    return scoped_ptr<CastExtensionSession>();
+  }
+  if (!cast_extension_session->InitializePeerConnection()) {
+    return scoped_ptr<CastExtensionSession>();
+  }
+  return cast_extension_session.Pass();
+}
+
+void CastExtensionSession::OnCreateSessionDescription(
+    webrtc::SessionDescriptionInterface* desc) {
+  if (!caller_task_runner_->BelongsToCurrentThread()) {
+    caller_task_runner_->PostTask(
+        FROM_HERE,
+        base::Bind(&CastExtensionSession::OnCreateSessionDescription,
+                   base::Unretained(this),
+                   desc));
+    return;
+  }
+
+  peer_connection_->SetLocalDescription(
+      CastSetSessionDescriptionObserver::Create(), desc);
+
+  scoped_ptr<base::DictionaryValue> json(new base::DictionaryValue());
+  json->SetString(kWebRtcSessionDescType, desc->type());
+  std::string subject =
+      (desc->type() == "offer") ? kSubjectOffer : kSubjectAnswer;
+  std::string desc_str;
+  desc->ToString(&desc_str);
+  json->SetString(kWebRtcSessionDescSDP, desc_str);
+  std::string json_str;
+  if (!base::JSONWriter::Write(json.get(), &json_str)) {
+    LOG(ERROR) << "Failed to serialize sdp message.";
+    return;
+  }
+
+  SendMessageToClient(subject.c_str(), json_str);
+}
+
+void CastExtensionSession::OnCreateSessionDescriptionFailure(
+    const std::string& error) {
+  VLOG(1) << "Creating Session Description failed: " << error;
+}
+
+// TODO(aiguha): Support the case(s) where we've grabbed the capturer already,
+// but another extension reset the video pipeline. We should remove the
+// stream from the peer connection here, and then attempt to re-setup the
+// peer connection in the OnRenegotiationNeeded() callback.
+// Tracked by: crbug.com/.

[Wez, 2014/08/14 19:22:03]

nit: See above re "See crbug.com/..."

[aiguha, 2014/08/15 04:11:39]

Done.

+scoped_ptr<webrtc::DesktopCapturer> CastExtensionSession::OnCreateVideoCapturer(
+    scoped_ptr<webrtc::DesktopCapturer> capturer) {
+  if (has_grabbed_capturer_) {
+    LOG(ERROR) << "The video pipeline was reset unexpectedly.";
+    has_grabbed_capturer_ = false;
+    peer_connection_->RemoveStream(stream_.release());
+    return capturer.Pass();
+  }
+
+  if (received_offer_) {
+    has_grabbed_capturer_ = true;
+    if (SetupVideoStream(capturer.Pass())) {
+      peer_connection_->CreateAnswer(
+          CastCreateSessionDescriptionObserver::Create(this), NULL);
+    } else {
+      has_grabbed_capturer_ = false;
+      // Ignore the received offer, since we failed to setup a video stream.
+      received_offer_ = false;
+    }
+    return scoped_ptr<webrtc::DesktopCapturer>();
+  }
+
+  return capturer.Pass();
+}
+
+bool CastExtensionSession::ModifiesVideoPipeline() const {
+  return true;
+}
+
+// Returns true if the |message| is a Cast ExtensionMessage, even if
+// it was badly formed or a resulting action failed. This is done so that
+// the host does not continue to attempt to pass |message| to other
+// HostExtensionSessions.
+bool CastExtensionSession::OnExtensionMessage(
+    ClientSessionControl* client_session_control,
+    protocol::ClientStub* client_stub,
+    const protocol::ExtensionMessage& message) {
+  if (message.type() != kExtensionMessageType) {
+    return false;
+  }
+
+  scoped_ptr<base::Value> value(base::JSONReader::Read(message.data()));
+  base::DictionaryValue* client_message;
+  if (!(value && value->GetAsDictionary(&client_message))) {
+    LOG(ERROR) << "Could not read cast extension message.";
+    return true;
+  }
+
+  std::string subject;
+  if (!client_message->GetString(kTopLevelSubject, &subject)) {
+    LOG(ERROR) << "Invalid Cast Extension Message (missing subject header).";
+    return true;
+  }
+
+  if (subject == kSubjectOffer && !received_offer_) {
+    // Reset the video pipeline so we can grab the screen capturer and setup
+    // a video stream.
+    if (ParseAndSetRemoteDescription(client_message)) {
+      received_offer_ = true;
+      client_session_control_->ResetVideoPipeline();
+    }
+  } else if (subject == kSubjectAnswer) {
+    ParseAndSetRemoteDescription(client_message);
+  } else if (subject == kSubjectNewCandidate) {
+    ParseAndAddICECandidate(client_message);
+  } else {
+    VLOG(1) << "Unexpected CastExtension Message: " << message.data();
+  }
+  return true;
+}
+
+// Private methods ------------------------------------------------------------
+
+CastExtensionSession::CastExtensionSession(
+    scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner,
+    scoped_refptr<net::URLRequestContextGetter> url_request_context_getter,
+    const protocol::NetworkSettings& network_settings,
+    ClientSessionControl* client_session_control,
+    protocol::ClientStub* client_stub)
+    : caller_task_runner_(caller_task_runner),
+      url_request_context_getter_(url_request_context_getter),
+      network_settings_(network_settings),
+      client_session_control_(client_session_control),
+      client_stub_(client_stub),
+      stats_observer_(CastStatsObserver::Create()),
+      received_offer_(false),
+      has_grabbed_capturer_(false),
+      signaling_thread_wrapper_(NULL),
+      worker_thread_wrapper_(NULL),
+      worker_thread_(kWorkerThreadName) {
+  DCHECK(caller_task_runner_->BelongsToCurrentThread());
+  DCHECK(url_request_context_getter_);
+  DCHECK(client_session_control_);
+  DCHECK(client_stub_);
+
+  // The worker thread is created with base::MessageLoop::TYPE_IO because
+  // the PeerConnection performs some port allocation operations on this thread
+  // that require it.

[Wez, 2014/08/14 19:22:03]

Should we file a bug against WebRTC for that behaviour? It seems broken...

[aiguha, 2014/08/15 04:11:39]

Done and linked.

+  base::Thread::Options options(base::MessageLoop::TYPE_IO, 0);
+  worker_thread_.StartWithOptions(options);
+  worker_task_runner_ = worker_thread_.task_runner();
+}
+
+bool CastExtensionSession::ParseAndSetRemoteDescription(
+    base::DictionaryValue* message) {
+  DCHECK(peer_connection_.get() != NULL);
+
+  base::DictionaryValue* message_data;
+  if (!message->GetDictionary(kTopLevelData, &message_data)) {
+    LOG(ERROR) << "Invalid Cast Extension Message (missing data).";
+    return false;
+  }
+
+  std::string webrtc_type;
+  if (!message_data->GetString(kWebRtcSessionDescType, &webrtc_type)) {
+    LOG(ERROR)
+        << "Invalid Cast Extension Message (missing webrtc type header).";
+    return false;
+  }
+
+  std::string sdp;
+  if (!message_data->GetString(kWebRtcSessionDescSDP, &sdp)) {
+    LOG(ERROR) << "Invalid Cast Extension Message (missing webrtc sdp header).";
+    return false;
+  }
+
+  webrtc::SdpParseError error;
+  webrtc::SessionDescriptionInterface* session_description(
+      webrtc::CreateSessionDescription(webrtc_type, sdp, &error));
+
+  if (!session_description) {
+    LOG(ERROR) << "Invalid Cast Extension Message (could not parse sdp).";
+    VLOG(1) << "SdpParseError was: " << error.description;
+    return false;
+  }
+
+  peer_connection_->SetRemoteDescription(
+      CastSetSessionDescriptionObserver::Create(), session_description);
+  return true;
+}
+
+bool CastExtensionSession::ParseAndAddICECandidate(
+    base::DictionaryValue* message) {
+  DCHECK(peer_connection_.get() != NULL);
+
+  base::DictionaryValue* message_data;
+  if (!message->GetDictionary(kTopLevelData, &message_data)) {
+    LOG(ERROR) << "Invalid Cast Extension Message (missing data).";
+    return false;
+  }
+
+  std::string candidate_str;
+  std::string sdp_mid;
+  int sdp_mlineindex = 0;
+  if (!message_data->GetString(kWebRtcSDPMid, &sdp_mid) ||
+      !message_data->GetInteger(kWebRtcSDPMLineIndex, &sdp_mlineindex) ||
+      !message_data->GetString(kWebRtcCandidate, &candidate_str)) {
+    LOG(ERROR) << "Invalid Cast Extension Message (could not parse).";
+    return false;
+  }
+
+  rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate(
+      webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate_str));
+  if (!candidate.get()) {
+    LOG(ERROR)
+        << "Invalid Cast Extension Message (could not create candidate).";
+    return false;
+  }
+
+  if (!peer_connection_->AddIceCandidate(candidate.get())) {
+    LOG(ERROR) << "Failed to apply received ICE Candidate to PeerConnection.";
+    return false;
+  }
+
+  VLOG(1) << "Received and Added ICE Candidate: " << candidate_str;
+
+  return true;
+}
+
+bool CastExtensionSession::SendMessageToClient(const std::string& subject,
+                                               const std::string& data) {
+  DCHECK(caller_task_runner_->BelongsToCurrentThread());
+
+  if (client_stub_ == NULL) {
+    LOG(ERROR) << "No Client Stub. Cannot send message to client.";
+    return false;
+  }
+
+  base::DictionaryValue message_dict;
+  message_dict.SetString(kTopLevelSubject, subject);
+  message_dict.SetString(kTopLevelData, data);
+  std::string message_json;
+
+  if (!base::JSONWriter::Write(&message_dict, &message_json)) {
+    LOG(ERROR) << "Failed to serialize JSON message.";
+    return false;
+  }
+
+  protocol::ExtensionMessage message;
+  message.set_type(kExtensionMessageType);
+  message.set_data(message_json);
+  client_stub_->DeliverHostMessage(message);
+  return true;
+}
+
+void CastExtensionSession::EnsureTaskAndSetSend(rtc::Thread** ptr,
+                                                base::WaitableEvent* event) {
+  jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
+  jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
+  *ptr = jingle_glue::JingleThreadWrapper::current();
+
+  if (event != NULL) {
+    event->Signal();
+  }
+}
+
+bool CastExtensionSession::WrapTasksAndSave() {
+  DCHECK(caller_task_runner_->BelongsToCurrentThread());
+
+  EnsureTaskAndSetSend(&signaling_thread_wrapper_);
+  if (signaling_thread_wrapper_ == NULL)
+    return false;
+
+  base::WaitableEvent wrap_worker_thread_event(true, false);
+  worker_task_runner_->PostTask(
+      FROM_HERE,
+      base::Bind(&CastExtensionSession::EnsureTaskAndSetSend,
+                 base::Unretained(this),
+                 &worker_thread_wrapper_,
+                 &wrap_worker_thread_event));
+  wrap_worker_thread_event.Wait();
+
+  return (worker_thread_wrapper_ != NULL);
+}
+
+bool CastExtensionSession::InitializePeerConnection() {
+  DCHECK(caller_task_runner_->BelongsToCurrentThread());
+  DCHECK(!peer_conn_factory_);
+  DCHECK(!peer_connection_);
+  DCHECK(worker_thread_wrapper_ != NULL);
+  DCHECK(signaling_thread_wrapper_ != NULL);
+
+  peer_conn_factory_ = webrtc::CreatePeerConnectionFactory(
+      worker_thread_wrapper_, signaling_thread_wrapper_, NULL, NULL, NULL);
+
+  if (!peer_conn_factory_.get()) {
+    CleanupPeerConnection();
+    return false;
+  }
+
+  VLOG(1) << "Created PeerConnectionFactory successfully.";
+
+  webrtc::PeerConnectionInterface::IceServers servers;
+  webrtc::PeerConnectionInterface::IceServer server;
+  server.uri = kDefaultStunURI;
+  servers.push_back(server);
+  webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
+  rtc_config.servers = servers;
+
+  // DTLS-SRTP is the preferred encryption method. If set to kValueFalse, the
+  // peer connection uses SDES. Disabling SDES as well will cause the peer
+  // connection to fail to connect.
+  // Note: For protection and unprotection of SRTP packets, the libjingle
+  // ENABLE_EXTERNAL_AUTH flag must not be set.
+  webrtc::FakeConstraints constraints;
+  constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
+                           webrtc::MediaConstraintsInterface::kValueTrue);
+
+  rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
+      port_allocator_factory = ChromiumPortAllocatorFactory::Create(
+          network_settings_, url_request_context_getter_);
+
+  peer_connection_ = peer_conn_factory_->CreatePeerConnection(
+      rtc_config, &constraints, port_allocator_factory, NULL, this);
+
+  if (!peer_connection_.get()) {
+    CleanupPeerConnection();
+    return false;
+  }
+
+  VLOG(1) << "Created PeerConnection successfully.";
+
+  // Send a test message to the client. Then, notify the client to start
+  // webrtc offer/answer negotiation.
+  if (!SendMessageToClient(kSubjectTest, "Hello, client.") ||
+      !SendMessageToClient(kSubjectReady, "Host ready to receive offers.")) {
+    LOG(ERROR) << "Failed to send messages to client.";
+    return false;
+  }
+
+  return true;
+}
+
+bool CastExtensionSession::SetupVideoStream(
+    scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) {
+  DCHECK(caller_task_runner_->BelongsToCurrentThread());
+  DCHECK(desktop_capturer);
+
+  if (stream_) {
+    VLOG(1) << "Already added MediaStream. Aborting Setup.";
+    return false;
+  }
+
+  scoped_ptr<CastVideoCapturerAdapter> cast_video_capturer_adapter(
+      new CastVideoCapturerAdapter(desktop_capturer.Pass()));
+
+  // Set video stream constraints.
+  webrtc::FakeConstraints video_constraints;
+  video_constraints.AddMandatory(
+      webrtc::MediaConstraintsInterface::kMinFrameRate, kMinFrameRate);
+
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track =
+      peer_conn_factory_->CreateVideoTrack(
+          kVideoLabel,
+          peer_conn_factory_->CreateVideoSource(
+              cast_video_capturer_adapter.release(), &video_constraints));
+
+  stream_ = peer_conn_factory_->CreateLocalMediaStream(kStreamLabel);
+
+  if (!stream_->AddTrack(video_track) ||
+      !peer_connection_->AddStream(stream_, NULL)) {
+    return false;
+  }
+
+  VLOG(1) << "Setup video stream successfully.";
+
+  return true;
+}
+
+void CastExtensionSession::PollPeerConnectionStats() {
+  if (!connection_active()) {
+    VLOG(1) << "Cannot poll stats while PeerConnection is inactive.";
+  }
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> video_track =
+      stream_->FindVideoTrack(kVideoLabel);
+  peer_connection_->GetStats(
+      stats_observer_,
+      video_track.release(),
+      webrtc::PeerConnectionInterface::kStatsOutputLevelStandard);
+}
+
+void CastExtensionSession::CleanupPeerConnection() {
+  peer_connection_->Close();
+  peer_connection_ = NULL;
+  stream_ = NULL;
+  peer_conn_factory_ = NULL;
+  worker_thread_.Stop();
+}
+
+bool CastExtensionSession::connection_active() const {
+  return peer_connection_.get() != NULL;
+}
+
+// webrtc::PeerConnectionObserver implementation -------------------------------
+
+void CastExtensionSession::OnError() {
+  VLOG(1) << "PeerConnectionObserver: an error occurred.";
+}
+
+void CastExtensionSession::OnSignalingChange(
+    webrtc::PeerConnectionInterface::SignalingState new_state) {
+  VLOG(1) << "PeerConnectionObserver: SignalingState changed to:" << new_state;
+}
+
+void CastExtensionSession::OnStateChange(
+    webrtc::PeerConnectionObserver::StateType state_changed) {
+  VLOG(1) << "PeerConnectionObserver: StateType changed to: " << state_changed;
+}
+
+void CastExtensionSession::OnAddStream(webrtc::MediaStreamInterface* stream) {
+  VLOG(1) << "PeerConnectionObserver: stream added: " << stream->label();
+}
+
+void CastExtensionSession::OnRemoveStream(
+    webrtc::MediaStreamInterface* stream) {
+  VLOG(1) << "PeerConnectionObserver: stream removed: " << stream->label();
+}
+
+void CastExtensionSession::OnDataChannel(
+    webrtc::DataChannelInterface* data_channel) {
+  VLOG(1) << "PeerConnectionObserver: data channel: " << data_channel->label();
+}
+
+void CastExtensionSession::OnRenegotiationNeeded() {
+  VLOG(1) << "PeerConnectionObserver: renegotiation needed.";
+}
+
+void CastExtensionSession::OnIceConnectionChange(
+    webrtc::PeerConnectionInterface::IceConnectionState new_state) {
+  VLOG(1) << "PeerConnectionObserver: IceConnectionState changed to: "
+          << new_state;
+
+  // TODO(aiguha): Maybe start timer only if enabled by command-line flag or
+  // at a particular verbosity level.
+  if (!stats_polling_timer_.IsRunning() &&
+      new_state == webrtc::PeerConnectionInterface::kIceConnectionConnected) {
+    stats_polling_timer_.Start(
+        FROM_HERE,
+        base::TimeDelta::FromSeconds(kStatsLogIntervalSec),
+        this,
+        &CastExtensionSession::PollPeerConnectionStats);
+  }
+}
+
+void CastExtensionSession::OnIceGatheringChange(
+    webrtc::PeerConnectionInterface::IceGatheringState new_state) {
+  VLOG(1) << "PeerConnectionObserver: IceGatheringState changed to: "
+          << new_state;
+}
+
+void CastExtensionSession::OnIceComplete() {
+  VLOG(1) << "PeerConnectionObserver: all ICE candidates found.";
+}
+
+void CastExtensionSession::OnIceCandidate(
+    const webrtc::IceCandidateInterface* candidate) {
+  std::string candidate_str;
+  if (!candidate->ToString(&candidate_str)) {
+    LOG(ERROR) << "PeerConnectionObserver: failed to serialize candidate.";
+    return;
+  }
+  scoped_ptr<base::DictionaryValue> json(new base::DictionaryValue());
+  json->SetString(kWebRtcSDPMid, candidate->sdp_mid());
+  json->SetInteger(kWebRtcSDPMLineIndex, candidate->sdp_mline_index());
+  json->SetString(kWebRtcCandidate, candidate_str);
+  std::string json_str;
+  if (!base::JSONWriter::Write(json.get(), &json_str)) {
+    LOG(ERROR) << "Failed to serialize candidate message.";
+    return;
+  }
+  SendMessageToClient(kSubjectNewCandidate, json_str);
+}
+
+}  // namespace remoting