text: /*
   *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
   *
   *  Use of this source code is governed by a BSD-style license
   *  that can be found in the LICENSE file in the root of the source
   *  tree. An additional intellectual property rights grant can be found
   *  in the file PATENTS.  All contributing project authors may
   *  be found in the AUTHORS file in the root of the source tree.
   */
  
  // Disable for TSan v2, see
  // https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
  #if !defined(THREAD_SANITIZER)
  
  #include <stdio.h>
  
  #include <functional>
  #include <list>
  #include <map>
  #include <memory>
  #include <utility>
  #include <vector>
  
  #include "absl/algorithm/container.h"
  #include "absl/memory/memory.h"
  #include "api/audio_codecs/builtin_audio_decoder_factory.h"
  #include "api/audio_codecs/builtin_audio_encoder_factory.h"
  #include "api/media_stream_interface.h"
  #include "api/peer_connection_interface.h"
  #include "api/peer_connection_proxy.h"
  #include "api/rtp_receiver_interface.h"
  #include "api/test/loopback_media_transport.h"
  #include "api/uma_metrics.h"
  #include "api/video_codecs/builtin_video_decoder_factory.h"
  #include "api/video_codecs/builtin_video_encoder_factory.h"
  #include "api/video_codecs/sdp_video_format.h"
  #include "call/call.h"
  #include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
  #include "logging/rtc_event_log/rtc_event_log_factory.h"
  #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
  #include "media/engine/fake_webrtc_video_engine.h"
  #include "media/engine/webrtc_media_engine.h"
  #include "modules/audio_processing/include/audio_processing.h"
  #include "p2p/base/mock_async_resolver.h"
  #include "p2p/base/p2p_constants.h"
  #include "p2p/base/port_interface.h"
  #include "p2p/base/test_stun_server.h"
  #include "p2p/base/test_turn_customizer.h"
  #include "p2p/base/test_turn_server.h"
  #include "p2p/client/basic_port_allocator.h"
  #include "pc/dtmf_sender.h"
  #include "pc/local_audio_source.h"
  #include "pc/media_session.h"
  #include "pc/peer_connection.h"
  #include "pc/peer_connection_factory.h"
  #include "pc/rtp_media_utils.h"
  #include "pc/session_description.h"
  #include "pc/test/fake_audio_capture_module.h"
  #include "pc/test/fake_periodic_video_track_source.h"
  #include "pc/test/fake_rtc_certificate_generator.h"
  #include "pc/test/fake_video_track_renderer.h"
  #include "pc/test/mock_peer_connection_observers.h"
  #include "rtc_base/fake_network.h"
  #include "rtc_base/firewall_socket_server.h"
  #include "rtc_base/gunit.h"
  #include "rtc_base/numerics/safe_conversions.h"
  #include "rtc_base/test_certificate_verifier.h"
  #include "rtc_base/time_utils.h"
  #include "rtc_base/virtual_socket_server.h"
  #include "system_wrappers/include/metrics.h"
  #include "test/gmock.h"
  
  namespace webrtc {
  namespace {
  
  using ::cricket::ContentInfo;
  using ::cricket::StreamParams;
  using ::rtc::SocketAddress;
  using ::testing::_;
  using ::testing::Combine;
  using ::testing::Contains;
  using ::testing::ElementsAre;
  using ::testing::NiceMock;
  using ::testing::Return;
  using ::testing::SetArgPointee;
  using ::testing::UnorderedElementsAreArray;
  using ::testing::Values;
  using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
  
  static const int kDefaultTimeout = 10000;
  static const int kMaxWaitForStatsMs = 3000;
  static const int kMaxWaitForActivationMs = 5000;
  static const int kMaxWaitForFramesMs = 10000;
  // Default number of audio/video frames to wait for before considering a test
  // successful.
  static const int kDefaultExpectedAudioFrameCount = 3;
  static const int kDefaultExpectedVideoFrameCount = 3;
  
  static const char kDataChannelLabel[] = "data_channel";
  
  // SRTP cipher name negotiated by the tests. This must be updated if the
  // default changes.
  static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_80;
  static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;
  
  static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0);
  
  // Helper function for constructing offer/answer options to initiate an ICE
  // restart.
  PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() {
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.ice_restart = true;
    return options;
  }
  
  // Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic"
  // attribute from received SDP, simulating a legacy endpoint.
  void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) {
    for (ContentInfo& content : desc->contents()) {
      content.media_description()->mutable_streams().clear();
    }
    desc->set_msid_supported(false);
    desc->set_msid_signaling(0);
  }
  
  // Removes all stream information besides the stream ids, simulating an
  // endpoint that only signals a=msid lines to convey stream_ids.
  void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
    for (ContentInfo& content : desc->contents()) {
      std::string track_id;
      std::vector<std::string> stream_ids;
      if (!content.media_description()->streams().empty()) {
        const StreamParams& first_stream =
            content.media_description()->streams()[0];
        track_id = first_stream.id;
        stream_ids = first_stream.stream_ids();
      }
      content.media_description()->mutable_streams().clear();
      StreamParams new_stream;
      new_stream.id = track_id;
      new_stream.set_stream_ids(stream_ids);
      content.media_description()->AddStream(new_stream);
    }
  }
  
  int FindFirstMediaStatsIndexByKind(
      const std::string& kind,
      const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
          media_stats_vec) {
    for (size_t i = 0; i < media_stats_vec.size(); i++) {
      if (media_stats_vec[i]->kind.ValueToString() == kind) {
        return i;
      }
    }
    return -1;
  }
  
  class SignalingMessageReceiver {
   public:
    virtual void ReceiveSdpMessage(SdpType type, const std::string& msg) = 0;
    virtual void ReceiveIceMessage(const std::string& sdp_mid,
                                   int sdp_mline_index,
                                   const std::string& msg) = 0;
  
   protected:
    SignalingMessageReceiver() {}
    virtual ~SignalingMessageReceiver() {}
  };
  
  class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
   public:
    explicit MockRtpReceiverObserver(cricket::MediaType media_type)
        : expected_media_type_(media_type) {}
  
    void OnFirstPacketReceived(cricket::MediaType media_type) override {
      ASSERT_EQ(expected_media_type_, media_type);
      first_packet_received_ = true;
    }
  
    bool first_packet_received() const { return first_packet_received_; }
  
    virtual ~MockRtpReceiverObserver() {}
  
   private:
    bool first_packet_received_ = false;
    cricket::MediaType expected_media_type_;
  };
  
  // Helper class that wraps a peer connection, observes it, and can accept
  // signaling messages from another wrapper.
  //
  // Uses a fake network, fake A/V capture, and optionally fake
  // encoders/decoders, though they aren't used by default since they don't
  // advertise support of any codecs.
  // TODO(steveanton): See how this could become a subclass of
  // PeerConnectionWrapper defined in peerconnectionwrapper.h.
  class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
                                public SignalingMessageReceiver {
   public:
    // Different factory methods for convenience.
    // TODO(deadbeef): Could use the pattern of:
    //
    // PeerConnectionWrapper =
    //     WrapperBuilder.WithConfig(...).WithOptions(...).build();
    //
    // To reduce some code duplication.
    static PeerConnectionWrapper* CreateWithDtlsIdentityStore(
        const std::string& debug_name,
        std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
        rtc::Thread* network_thread,
        rtc::Thread* worker_thread) {
      PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
      webrtc::PeerConnectionDependencies dependencies(nullptr);
      dependencies.cert_generator = std::move(cert_generator);
      if (!client->Init(nullptr, nullptr, std::move(dependencies), network_thread,
                        worker_thread, nullptr,
                        /*media_transport_factory=*/nullptr)) {
        delete client;
        return nullptr;
      }
      return client;
    }
  
    webrtc::PeerConnectionFactoryInterface* pc_factory() const {
      return peer_connection_factory_.get();
    }
  
    webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
  
    // If a signaling message receiver is set (via ConnectFakeSignaling), this
    // will set the whole offer/answer exchange in motion. Just need to wait for
    // the signaling state to reach "stable".
    void CreateAndSetAndSignalOffer() {
      auto offer = CreateOffer();
      ASSERT_NE(nullptr, offer);
      EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer)));
    }
  
    // Sets the options to be used when CreateAndSetAndSignalOffer is called, or
    // when a remote offer is received (via fake signaling) and an answer is
    // generated. By default, uses default options.
    void SetOfferAnswerOptions(
        const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
      offer_answer_options_ = options;
    }
  
    // Set a callback to be invoked when SDP is received via the fake signaling
    // channel, which provides an opportunity to munge (modify) the SDP. This is
    // used to test SDP being applied that a PeerConnection would normally not
    // generate, but a non-JSEP endpoint might.
    void SetReceivedSdpMunger(
        std::function<void(cricket::SessionDescription*)> munger) {
      received_sdp_munger_ = std::move(munger);
    }
  
    // Similar to the above, but this is run on SDP immediately after it's
    // generated.
    void SetGeneratedSdpMunger(
        std::function<void(cricket::SessionDescription*)> munger) {
      generated_sdp_munger_ = std::move(munger);
    }
  
    // Set a callback to be invoked when a remote offer is received via the fake
    // signaling channel. This provides an opportunity to change the
    // PeerConnection state before an answer is created and sent to the caller.
    void SetRemoteOfferHandler(std::function<void()> handler) {
      remote_offer_handler_ = std::move(handler);
    }
  
    void SetRemoteAsyncResolver(rtc::MockAsyncResolver* resolver) {
      remote_async_resolver_ = resolver;
    }
  
    // Every ICE connection state in order that has been seen by the observer.
    std::vector<PeerConnectionInterface::IceConnectionState>
    ice_connection_state_history() const {
      return ice_connection_state_history_;
    }
    void clear_ice_connection_state_history() {
      ice_connection_state_history_.clear();
    }
  
    // Every PeerConnection state in order that has been seen by the observer.
    std::vector<PeerConnectionInterface::PeerConnectionState>
    peer_connection_state_history() const {
      return peer_connection_state_history_;
    }
  
    // Every ICE gathering state in order that has been seen by the observer.
    std::vector<PeerConnectionInterface::IceGatheringState>
    ice_gathering_state_history() const {
      return ice_gathering_state_history_;
    }
  
    void AddAudioVideoTracks() {
      AddAudioTrack();
      AddVideoTrack();
    }
  
    rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack() {
      return AddTrack(CreateLocalAudioTrack());
    }
  
    rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack() {
      return AddTrack(CreateLocalVideoTrack());
    }
  
    rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
      cricket::AudioOptions options;
      // Disable highpass filter so that we can get all the test audio frames.
      options.highpass_filter = false;
      rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
          peer_connection_factory_->CreateAudioSource(options);
      // TODO(perkj): Test audio source when it is implemented. Currently audio
      // always use the default input.
      return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(),
                                                        source);
    }
  
    rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
      webrtc::FakePeriodicVideoSource::Config config;
      config.timestamp_offset_ms = rtc::TimeMillis();
      return CreateLocalVideoTrackInternal(config);
    }
  
    rtc::scoped_refptr<webrtc::VideoTrackInterface>
    CreateLocalVideoTrackWithConfig(
        webrtc::FakePeriodicVideoSource::Config config) {
      return CreateLocalVideoTrackInternal(config);
    }
  
    rtc::scoped_refptr<webrtc::VideoTrackInterface>
    CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
      webrtc::FakePeriodicVideoSource::Config config;
      config.rotation = rotation;
      config.timestamp_offset_ms = rtc::TimeMillis();
      return CreateLocalVideoTrackInternal(config);
    }
  
    rtc::scoped_refptr<RtpSenderInterface> AddTrack(
        rtc::scoped_refptr<MediaStreamTrackInterface> track,
        const std::vector<std::string>& stream_ids = {}) {
      auto result = pc()->AddTrack(track, stream_ids);
      EXPECT_EQ(RTCErrorType::NONE, result.error().type());
      return result.MoveValue();
    }
  
    std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType(
        cricket::MediaType media_type) {
      std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
      for (const auto& receiver : pc()->GetReceivers()) {
        if (receiver->media_type() == media_type) {
          receivers.push_back(receiver);
        }
      }
      return receivers;
    }
  
    rtc::scoped_refptr<RtpTransceiverInterface> GetFirstTransceiverOfType(
        cricket::MediaType media_type) {
      for (auto transceiver : pc()->GetTransceivers()) {
        if (transceiver->receiver()->media_type() == media_type) {
          return transceiver;
        }
      }
      return nullptr;
    }
  
    bool SignalingStateStable() {
      return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
    }
  
    void CreateDataChannel() { CreateDataChannel(nullptr); }
  
    void CreateDataChannel(const webrtc::DataChannelInit* init) {
      CreateDataChannel(kDataChannelLabel, init);
    }
  
    void CreateDataChannel(const std::string& label,
                           const webrtc::DataChannelInit* init) {
      data_channel_ = pc()->CreateDataChannel(label, init);
      ASSERT_TRUE(data_channel_.get() != nullptr);
      data_observer_.reset(new MockDataChannelObserver(data_channel_));
    }
  
    DataChannelInterface* data_channel() { return data_channel_; }
    const MockDataChannelObserver* data_observer() const {
      return data_observer_.get();
    }
  
    int audio_frames_received() const {
      return fake_audio_capture_module_->frames_received();
    }
  
    // Takes minimum of video frames received for each track.
    //
    // Can be used like:
    // EXPECT_GE(expected_frames, min_video_frames_received_per_track());
    //
    // To ensure that all video tracks received at least a certain number of
    // frames.
    int min_video_frames_received_per_track() const {
      int min_frames = INT_MAX;
      if (fake_video_renderers_.empty()) {
        return 0;
      }
  
      for (const auto& pair : fake_video_renderers_) {
        min_frames = std::min(min_frames, pair.second->num_rendered_frames());
      }
      return min_frames;
    }
  
    // Returns a MockStatsObserver in a state after stats gathering finished,
    // which can be used to access the gathered stats.
    rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack(
        webrtc::MediaStreamTrackInterface* track) {
      rtc::scoped_refptr<MockStatsObserver> observer(
          new rtc::RefCountedObject<MockStatsObserver>());
      EXPECT_TRUE(peer_connection_->GetStats(
          observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
      EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
      return observer;
    }
  
    // Version that doesn't take a track "filter", and gathers all stats.
    rtc::scoped_refptr<MockStatsObserver> OldGetStats() {
      return OldGetStatsForTrack(nullptr);
    }
  
    // Synchronously gets stats and returns them. If it times out, fails the test
    // and returns null.
    rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
      rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
          new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
      peer_connection_->GetStats(callback);
      EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
      return callback->report();
    }
  
    int rendered_width() {
      EXPECT_FALSE(fake_video_renderers_.empty());
      return fake_video_renderers_.empty()
                 ? 0
                 : fake_video_renderers_.begin()->second->width();
    }
  
    int rendered_height() {
      EXPECT_FALSE(fake_video_renderers_.empty());
      return fake_video_renderers_.empty()
                 ? 0
                 : fake_video_renderers_.begin()->second->height();
    }
  
    double rendered_aspect_ratio() {
      if (rendered_height() == 0) {
        return 0.0;
      }
      return static_cast<double>(rendered_width()) / rendered_height();
    }
  
    webrtc::VideoRotation rendered_rotation() {
      EXPECT_FALSE(fake_video_renderers_.empty());
      return fake_video_renderers_.empty()
                 ? webrtc::kVideoRotation_0
                 : fake_video_renderers_.begin()->second->rotation();
    }
  
    int local_rendered_width() {
      return local_video_renderer_ ? local_video_renderer_->width() : 0;
    }
  
    int local_rendered_height() {
      return local_video_renderer_ ? local_video_renderer_->height() : 0;
    }
  
    double local_rendered_aspect_ratio() {
      if (local_rendered_height() == 0) {
        return 0.0;
      }
      return static_cast<double>(local_rendered_width()) /
             local_rendered_height();
    }
  
    size_t number_of_remote_streams() {
      if (!pc()) {
        return 0;
      }
      return pc()->remote_streams()->count();
    }
  
    StreamCollectionInterface* remote_streams() const {
      if (!pc()) {
        ADD_FAILURE();
        return nullptr;
      }
      return pc()->remote_streams();
    }
  
    StreamCollectionInterface* local_streams() {
      if (!pc()) {
        ADD_FAILURE();
        return nullptr;
      }
      return pc()->local_streams();
    }
  
    webrtc::PeerConnectionInterface::SignalingState signaling_state() {
      return pc()->signaling_state();
    }
  
    webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
      return pc()->ice_connection_state();
    }
  
    webrtc::PeerConnectionInterface::IceConnectionState
    standardized_ice_connection_state() {
      return pc()->standardized_ice_connection_state();
    }
  
    webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
      return pc()->ice_gathering_state();
    }
  
    // Returns a MockRtpReceiverObserver for each RtpReceiver returned by
    // GetReceivers. They're updated automatically when a remote offer/answer
    // from the fake signaling channel is applied, or when
    // ResetRtpReceiverObservers below is called.
    const std::vector<std::unique_ptr<MockRtpReceiverObserver>>&
    rtp_receiver_observers() {
      return rtp_receiver_observers_;
    }
  
    void ResetRtpReceiverObservers() {
      rtp_receiver_observers_.clear();
      for (const rtc::scoped_refptr<RtpReceiverInterface>& receiver :
           pc()->GetReceivers()) {
        std::unique_ptr<MockRtpReceiverObserver> observer(
            new MockRtpReceiverObserver(receiver->media_type()));
        receiver->SetObserver(observer.get());
        rtp_receiver_observers_.push_back(std::move(observer));
      }
    }
  
    rtc::FakeNetworkManager* network() const {
      return fake_network_manager_.get();
    }
    cricket::PortAllocator* port_allocator() const { return port_allocator_; }
  
    webrtc::FakeRtcEventLogFactory* event_log_factory() const {
      return event_log_factory_;
    }
  
   private:
    explicit PeerConnectionWrapper(const std::string& debug_name)
        : debug_name_(debug_name) {}
  
    bool Init(
        const PeerConnectionFactory::Options* options,
        const PeerConnectionInterface::RTCConfiguration* config,
        webrtc::PeerConnectionDependencies dependencies,
        rtc::Thread* network_thread,
        rtc::Thread* worker_thread,
        std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
        std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) {
      // There's an error in this test code if Init ends up being called twice.
      RTC_DCHECK(!peer_connection_);
      RTC_DCHECK(!peer_connection_factory_);
  
      fake_network_manager_.reset(new rtc::FakeNetworkManager());
      fake_network_manager_->AddInterface(kDefaultLocalAddress);
  
      std::unique_ptr<cricket::PortAllocator> port_allocator(
          new cricket::BasicPortAllocator(fake_network_manager_.get()));
      port_allocator_ = port_allocator.get();
      fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
      if (!fake_audio_capture_module_) {
        return false;
      }
      rtc::Thread* const signaling_thread = rtc::Thread::Current();
  
      webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies;
      pc_factory_dependencies.network_thread = network_thread;
      pc_factory_dependencies.worker_thread = worker_thread;
      pc_factory_dependencies.signaling_thread = signaling_thread;
      pc_factory_dependencies.media_engine =
          cricket::WebRtcMediaEngineFactory::Create(
              rtc::scoped_refptr<webrtc::AudioDeviceModule>(
                  fake_audio_capture_module_),
              webrtc::CreateBuiltinAudioEncoderFactory(),
              webrtc::CreateBuiltinAudioDecoderFactory(),
              webrtc::CreateBuiltinVideoEncoderFactory(),
              webrtc::CreateBuiltinVideoDecoderFactory(), nullptr,
              webrtc::AudioProcessingBuilder().Create());
      pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();
      if (event_log_factory) {
        event_log_factory_ = event_log_factory.get();
        pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
      } else {
        pc_factory_dependencies.event_log_factory =
            webrtc::CreateRtcEventLogFactory();
      }
      if (media_transport_factory) {
        pc_factory_dependencies.media_transport_factory =
            std::move(media_transport_factory);
      }
      peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
          std::move(pc_factory_dependencies));
  
      if (!peer_connection_factory_) {
        return false;
      }
      if (options) {
        peer_connection_factory_->SetOptions(*options);
      }
      if (config) {
        sdp_semantics_ = config->sdp_semantics;
      }
  
      dependencies.allocator = std::move(port_allocator);
      peer_connection_ = CreatePeerConnection(config, std::move(dependencies));
      return peer_connection_.get() != nullptr;
    }
  
    rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
        const PeerConnectionInterface::RTCConfiguration* config,
        webrtc::PeerConnectionDependencies dependencies) {
      PeerConnectionInterface::RTCConfiguration modified_config;
      // If |config| is null, this will result in a default configuration being
      // used.
      if (config) {
        modified_config = *config;
      }
      // Disable resolution adaptation; we don't want it interfering with the
      // test results.
      // TODO(deadbeef): Do something more robust. Since we're testing for aspect
      // ratios and not specific resolutions, is this even necessary?
      modified_config.set_cpu_adaptation(false);
  
      dependencies.observer = this;
      return peer_connection_factory_->CreatePeerConnection(
          modified_config, std::move(dependencies));
    }
  
    void set_signaling_message_receiver(
        SignalingMessageReceiver* signaling_message_receiver) {
      signaling_message_receiver_ = signaling_message_receiver;
    }
  
    void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; }
  
    void set_signal_ice_candidates(bool signal) {
      signal_ice_candidates_ = signal;
    }
  
    rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
        webrtc::FakePeriodicVideoSource::Config config) {
      // Set max frame rate to 10fps to reduce the risk of test flakiness.
      // TODO(deadbeef): Do something more robust.
      config.frame_interval_ms = 100;
  
      video_track_sources_.emplace_back(
          new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>(
              config, false /* remote */));
      rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
          peer_connection_factory_->CreateVideoTrack(
              rtc::CreateRandomUuid(), video_track_sources_.back()));
      if (!local_video_renderer_) {
        local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
      }
      return track;
    }
  
    void HandleIncomingOffer(const std::string& msg) {
      RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer";
      std::unique_ptr<SessionDescriptionInterface> desc =
          webrtc::CreateSessionDescription(SdpType::kOffer, msg);
      if (received_sdp_munger_) {
        received_sdp_munger_(desc->description());
      }
  
      EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
      // Setting a remote description may have changed the number of receivers,
      // so reset the receiver observers.
      ResetRtpReceiverObservers();
      if (remote_offer_handler_) {
        remote_offer_handler_();
      }
      auto answer = CreateAnswer();
      ASSERT_NE(nullptr, answer);
      EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer)));
    }
  
    void HandleIncomingAnswer(const std::string& msg) {
      RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer";
      std::unique_ptr<SessionDescriptionInterface> desc =
          webrtc::CreateSessionDescription(SdpType::kAnswer, msg);
      if (received_sdp_munger_) {
        received_sdp_munger_(desc->description());
      }
  
      EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
      // Set the RtpReceiverObserver after receivers are created.
      ResetRtpReceiverObservers();
    }
  
    // Returns null on failure.
    std::unique_ptr<SessionDescriptionInterface> CreateOffer() {
      rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
          new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
      pc()->CreateOffer(observer, offer_answer_options_);
      return WaitForDescriptionFromObserver(observer);
    }
  
    // Returns null on failure.
    std::unique_ptr<SessionDescriptionInterface> CreateAnswer() {
      rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
          new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
      pc()->CreateAnswer(observer, offer_answer_options_);
      return WaitForDescriptionFromObserver(observer);
    }
  
    std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver(
        MockCreateSessionDescriptionObserver* observer) {
      EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
      if (!observer->result()) {
        return nullptr;
      }
      auto description = observer->MoveDescription();
      if (generated_sdp_munger_) {
        generated_sdp_munger_(description->description());
      }
      return description;
    }
  
    // Setting the local description and sending the SDP message over the fake
    // signaling channel are combined into the same method because the SDP
    // message needs to be sent as soon as SetLocalDescription finishes, without
    // waiting for the observer to be called. This ensures that ICE candidates
    // don't outrace the description.
    bool SetLocalDescriptionAndSendSdpMessage(
        std::unique_ptr<SessionDescriptionInterface> desc) {
      rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
          new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
      RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage";
      SdpType type = desc->GetType();
      std::string sdp;
      EXPECT_TRUE(desc->ToString(&sdp));
      pc()->SetLocalDescription(observer, desc.release());
      if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
        RemoveUnusedVideoRenderers();
      }
      // As mentioned above, we need to send the message immediately after
      // SetLocalDescription.
      SendSdpMessage(type, sdp);
      EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
      return true;
    }
  
    bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) {
      rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
          new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
      RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription";
      pc()->SetRemoteDescription(observer, desc.release());
      if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
        RemoveUnusedVideoRenderers();
      }
      EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
      return observer->result();
    }
  
    // This is a work around to remove unused fake_video_renderers from
    // transceivers that have either stopped or are no longer receiving.
    void RemoveUnusedVideoRenderers() {
      auto transceivers = pc()->GetTransceivers();
      for (auto& transceiver : transceivers) {
        if (transceiver->receiver()->media_type() != cricket::MEDIA_TYPE_VIDEO) {
          continue;
        }
        // Remove fake video renderers from any stopped transceivers.
        if (transceiver->stopped()) {
          auto it =
              fake_video_renderers_.find(transceiver->receiver()->track()->id());
          if (it != fake_video_renderers_.end()) {
            fake_video_renderers_.erase(it);
          }
        }
        // Remove fake video renderers from any transceivers that are no longer
        // receiving.
        if ((transceiver->current_direction() &&
             !webrtc::RtpTransceiverDirectionHasRecv(
                 *transceiver->current_direction()))) {
          auto it =
              fake_video_renderers_.find(transceiver->receiver()->track()->id());
          if (it != fake_video_renderers_.end()) {
            fake_video_renderers_.erase(it);
          }
        }
      }
    }
  
    // Simulate sending a blob of SDP with delay |signaling_delay_ms_| (0 by
    // default).
    void SendSdpMessage(SdpType type, const std::string& msg) {
      if (signaling_delay_ms_ == 0) {
        RelaySdpMessageIfReceiverExists(type, msg);
      } else {
        invoker_.AsyncInvokeDelayed<void>(
            RTC_FROM_HERE, rtc::Thread::Current(),
            rtc::Bind(&PeerConnectionWrapper::RelaySdpMessageIfReceiverExists,
                      this, type, msg),
            signaling_delay_ms_);
      }
    }
  
    void RelaySdpMessageIfReceiverExists(SdpType type, const std::string& msg) {
      if (signaling_message_receiver_) {
        signaling_message_receiver_->ReceiveSdpMessage(type, msg);
      }
    }
  
    // Simulate trickling an ICE candidate with delay |signaling_delay_ms_| (0 by
    // default).
    void SendIceMessage(const std::string& sdp_mid,
                        int sdp_mline_index,
                        const std::string& msg) {
      if (signaling_delay_ms_ == 0) {
        RelayIceMessageIfReceiverExists(sdp_mid, sdp_mline_index, msg);
      } else {
        invoker_.AsyncInvokeDelayed<void>(
            RTC_FROM_HERE, rtc::Thread::Current(),
            rtc::Bind(&PeerConnectionWrapper::RelayIceMessageIfReceiverExists,
                      this, sdp_mid, sdp_mline_index, msg),
            signaling_delay_ms_);
      }
    }
  
    void RelayIceMessageIfReceiverExists(const std::string& sdp_mid,
                                         int sdp_mline_index,
                                         const std::string& msg) {
      if (signaling_message_receiver_) {
        signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index,
                                                       msg);
      }
    }
  
    // SignalingMessageReceiver callbacks.
    void ReceiveSdpMessage(SdpType type, const std::string& msg) override {
      if (type == SdpType::kOffer) {
        HandleIncomingOffer(msg);
      } else {
        HandleIncomingAnswer(msg);
      }
    }
  
    void ReceiveIceMessage(const std::string& sdp_mid,
                           int sdp_mline_index,
                           const std::string& msg) override {
      RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage";
      std::unique_ptr<webrtc::IceCandidateInterface> candidate(
          webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
      EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
    }
  
    // PeerConnectionObserver callbacks.
    void OnSignalingChange(
        webrtc::PeerConnectionInterface::SignalingState new_state) override {
      EXPECT_EQ(pc()->signaling_state(), new_state);
    }
    void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
                    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
                        streams) override {
      if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
        rtc::scoped_refptr<VideoTrackInterface> video_track(
            static_cast<VideoTrackInterface*>(receiver->track().get()));
        ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
                    fake_video_renderers_.end());
        fake_video_renderers_[video_track->id()] =
            absl::make_unique<FakeVideoTrackRenderer>(video_track);
      }
    }
    void OnRemoveTrack(
        rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
      if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
        auto it = fake_video_renderers_.find(receiver->track()->id());
        RTC_DCHECK(it != fake_video_renderers_.end());
        fake_video_renderers_.erase(it);
      }
    }
    void OnRenegotiationNeeded() override {}
    void OnIceConnectionChange(
        webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
      EXPECT_EQ(pc()->ice_connection_state(), new_state);
      ice_connection_state_history_.push_back(new_state);
    }
    void OnConnectionChange(
        webrtc::PeerConnectionInterface::PeerConnectionState new_state) override {
      peer_connection_state_history_.push_back(new_state);
    }
  
    void OnIceGatheringChange(
        webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
      EXPECT_EQ(pc()->ice_gathering_state(), new_state);
      ice_gathering_state_history_.push_back(new_state);
    }
    void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
      RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate";
  
      if (remote_async_resolver_) {
        const auto& local_candidate = candidate->candidate();
        const auto& mdns_responder = network()->GetMdnsResponderForTesting();
        if (local_candidate.address().IsUnresolvedIP()) {
          RTC_DCHECK(local_candidate.type() == cricket::LOCAL_PORT_TYPE);
          rtc::SocketAddress resolved_addr(local_candidate.address());
          const auto resolved_ip = mdns_responder->GetMappedAddressForName(
              local_candidate.address().hostname());
          RTC_DCHECK(!resolved_ip.IsNil());
          resolved_addr.SetResolvedIP(resolved_ip);
          EXPECT_CALL(*remote_async_resolver_, GetResolvedAddress(_, _))
              .WillOnce(DoAll(SetArgPointee<1>(resolved_addr), Return(true)));
          EXPECT_CALL(*remote_async_resolver_, Destroy(_));
        }
      }
  
      std::string ice_sdp;
      EXPECT_TRUE(candidate->ToString(&ice_sdp));
      if (signaling_message_receiver_ == nullptr || !signal_ice_candidates_) {
        // Remote party may be deleted.
        return;
      }
      SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
    }
    void OnDataChannel(
        rtc::scoped_refptr<DataChannelInterface> data_channel) override {
      RTC_LOG(LS_INFO) << debug_name_ << ": OnDataChannel";
      data_channel_ = data_channel;
      data_observer_.reset(new MockDataChannelObserver(data_channel));
    }
  
    std::string debug_name_;
  
    std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
  
    rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
        peer_connection_factory_;
  
    cricket::PortAllocator* port_allocator_;
    // Needed to keep track of number of frames sent.
    rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
    // Needed to keep track of number of frames received.
    std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
        fake_video_renderers_;
    // Needed to ensure frames aren't received for removed tracks.
    std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
        removed_fake_video_renderers_;
  
    // For remote peer communication.
    SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
    int signaling_delay_ms_ = 0;
    bool signal_ice_candidates_ = true;
  
    // Store references to the video sources we've created, so that we can stop
    // them, if required.
    std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>>
        video_track_sources_;
    // |local_video_renderer_| attached to the first created local video track.
    std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
  
    SdpSemantics sdp_semantics_;
    PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
    std::function<void(cricket::SessionDescription*)> received_sdp_munger_;
    std::function<void(cricket::SessionDescription*)> generated_sdp_munger_;
    std::function<void()> remote_offer_handler_;
    rtc::MockAsyncResolver* remote_async_resolver_ = nullptr;
    rtc::scoped_refptr<DataChannelInterface> data_channel_;
    std::unique_ptr<MockDataChannelObserver> data_observer_;
  
    std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;
  
    std::vector<PeerConnectionInterface::IceConnectionState>
        ice_connection_state_history_;
    std::vector<PeerConnectionInterface::PeerConnectionState>
        peer_connection_state_history_;
    std::vector<PeerConnectionInterface::IceGatheringState>
        ice_gathering_state_history_;
  
    webrtc::FakeRtcEventLogFactory* event_log_factory_;
  
    rtc::AsyncInvoker invoker_;
  
    friend class PeerConnectionIntegrationBaseTest;
  };
  
  class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
   public:
    virtual ~MockRtcEventLogOutput() = default;
    MOCK_CONST_METHOD0(IsActive, bool());
    MOCK_METHOD1(Write, bool(const std::string&));
  };
  
  // This helper object is used for both specifying how many audio/video frames
  // are expected to be received for a caller/callee. It provides helper functions
  // to specify these expectations. The object initially starts in a state of no
  // expectations.
  class MediaExpectations {
   public:
    enum ExpectFrames {
      kExpectSomeFrames,
      kExpectNoFrames,
      kNoExpectation,
    };
  
    void ExpectBidirectionalAudioAndVideo() {
      ExpectBidirectionalAudio();
      ExpectBidirectionalVideo();
    }
  
    void ExpectBidirectionalAudio() {
      CallerExpectsSomeAudio();
      CalleeExpectsSomeAudio();
    }
  
    void ExpectNoAudio() {
      CallerExpectsNoAudio();
      CalleeExpectsNoAudio();
    }
  
    void ExpectBidirectionalVideo() {
      CallerExpectsSomeVideo();
      CalleeExpectsSomeVideo();
    }
  
    void ExpectNoVideo() {
      CallerExpectsNoVideo();
      CalleeExpectsNoVideo();
    }
  
    void CallerExpectsSomeAudioAndVideo() {
      CallerExpectsSomeAudio();
      CallerExpectsSomeVideo();
    }
  
    void CalleeExpectsSomeAudioAndVideo() {
      CalleeExpectsSomeAudio();
      CalleeExpectsSomeVideo();
    }
  
    // Caller's audio functions.
    void CallerExpectsSomeAudio(
        int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
      caller_audio_expectation_ = kExpectSomeFrames;
      caller_audio_frames_expected_ = expected_audio_frames;
    }
  
    void CallerExpectsNoAudio() {
      caller_audio_expectation_ = kExpectNoFrames;
      caller_audio_frames_expected_ = 0;
    }
  
    // Caller's video functions.
    void CallerExpectsSomeVideo(
        int expected_video_frames = kDefaultExpectedVideoFrameCount) {
      caller_video_expectation_ = kExpectSomeFrames;
      caller_video_frames_expected_ = expected_video_frames;
    }
  
    void CallerExpectsNoVideo() {
      caller_video_expectation_ = kExpectNoFrames;
      caller_video_frames_expected_ = 0;
    }
  
    // Callee's audio functions.
    void CalleeExpectsSomeAudio(
        int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
      callee_audio_expectation_ = kExpectSomeFrames;
      callee_audio_frames_expected_ = expected_audio_frames;
    }
  
    void CalleeExpectsNoAudio() {
      callee_audio_expectation_ = kExpectNoFrames;
      callee_audio_frames_expected_ = 0;
    }
  
    // Callee's video functions.
    void CalleeExpectsSomeVideo(
        int expected_video_frames = kDefaultExpectedVideoFrameCount) {
      callee_video_expectation_ = kExpectSomeFrames;
      callee_video_frames_expected_ = expected_video_frames;
    }
  
    void CalleeExpectsNoVideo() {
      callee_video_expectation_ = kExpectNoFrames;
      callee_video_frames_expected_ = 0;
    }
  
    ExpectFrames caller_audio_expectation_ = kNoExpectation;
    ExpectFrames caller_video_expectation_ = kNoExpectation;
    ExpectFrames callee_audio_expectation_ = kNoExpectation;
    ExpectFrames callee_video_expectation_ = kNoExpectation;
    int caller_audio_frames_expected_ = 0;
    int caller_video_frames_expected_ = 0;
    int callee_audio_frames_expected_ = 0;
    int callee_video_frames_expected_ = 0;
  };
  
  // Tests two PeerConnections connecting to each other end-to-end, using a
  // virtual network, fake A/V capture and fake encoder/decoders. The
  // PeerConnections share the threads/socket servers, but use separate versions
  // of everything else (including "PeerConnectionFactory"s).
  class PeerConnectionIntegrationBaseTest : public testing::Test {
   public:
    explicit PeerConnectionIntegrationBaseTest(SdpSemantics sdp_semantics)
        : sdp_semantics_(sdp_semantics),
          ss_(new rtc::VirtualSocketServer()),
          fss_(new rtc::FirewallSocketServer(ss_.get())),
          network_thread_(new rtc::Thread(fss_.get())),
          worker_thread_(rtc::Thread::Create()),
          loopback_media_transports_(network_thread_.get()) {
      network_thread_->SetName("PCNetworkThread", this);
      worker_thread_->SetName("PCWorkerThread", this);
      RTC_CHECK(network_thread_->Start());
      RTC_CHECK(worker_thread_->Start());
      webrtc::metrics::Reset();
    }
  
    ~PeerConnectionIntegrationBaseTest() {
      // The PeerConnections should deleted before the TurnCustomizers.
      // A TurnPort is created with a raw pointer to a TurnCustomizer. The
      // TurnPort has the same lifetime as the PeerConnection, so it's expected
      // that the TurnCustomizer outlives the life of the PeerConnection or else
      // when Send() is called it will hit a seg fault.
      if (caller_) {
        caller_->set_signaling_message_receiver(nullptr);
        delete SetCallerPcWrapperAndReturnCurrent(nullptr);
      }
      if (callee_) {
        callee_->set_signaling_message_receiver(nullptr);
        delete SetCalleePcWrapperAndReturnCurrent(nullptr);
      }
  
      // If turn servers were created for the test they need to be destroyed on
      // the network thread.
      network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
        turn_servers_.clear();
        turn_customizers_.clear();
      });
    }
  
    bool SignalingStateStable() {
      return caller_->SignalingStateStable() && callee_->SignalingStateStable();
    }
  
    bool DtlsConnected() {
      // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
      // are connected. This is an important distinction. Once we have separate
      // ICE and DTLS state, this check needs to use the DTLS state.
      return (callee()->ice_connection_state() ==
                  webrtc::PeerConnectionInterface::kIceConnectionConnected ||
              callee()->ice_connection_state() ==
                  webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
             (caller()->ice_connection_state() ==
                  webrtc::PeerConnectionInterface::kIceConnectionConnected ||
              caller()->ice_connection_state() ==
                  webrtc::PeerConnectionInterface::kIceConnectionCompleted);
    }
  
    // When |event_log_factory| is null, the default implementation of the event
    // log factory will be used.
    std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWrapper(
        const std::string& debug_name,
        const PeerConnectionFactory::Options* options,
        const RTCConfiguration* config,
        webrtc::PeerConnectionDependencies dependencies,
        std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
        std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) {
      RTCConfiguration modified_config;
      if (config) {
        modified_config = *config;
      }
      modified_config.sdp_semantics = sdp_semantics_;
      if (!dependencies.cert_generator) {
        dependencies.cert_generator =
            absl::make_unique<FakeRTCCertificateGenerator>();
      }
      std::unique_ptr<PeerConnectionWrapper> client(
          new PeerConnectionWrapper(debug_name));
  
      if (!client->Init(options, &modified_config, std::move(dependencies),
                        network_thread_.get(), worker_thread_.get(),
                        std::move(event_log_factory),
                        std::move(media_transport_factory))) {
        return nullptr;
      }
      return client;
    }
  
    std::unique_ptr<PeerConnectionWrapper>
    CreatePeerConnectionWrapperWithFakeRtcEventLog(
        const std::string& debug_name,
        const PeerConnectionFactory::Options* options,
        const RTCConfiguration* config,
        webrtc::PeerConnectionDependencies dependencies) {
      std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory(
          new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current()));
      return CreatePeerConnectionWrapper(debug_name, options, config,
                                         std::move(dependencies),
                                         std::move(event_log_factory),
                                         /*media_transport_factory=*/nullptr);
    }
  
    bool CreatePeerConnectionWrappers() {
      return CreatePeerConnectionWrappersWithConfig(
          PeerConnectionInterface::RTCConfiguration(),
          PeerConnectionInterface::RTCConfiguration());
    }
  
    bool CreatePeerConnectionWrappersWithSdpSemantics(
        SdpSemantics caller_semantics,
        SdpSemantics callee_semantics) {
      // Can't specify the sdp_semantics in the passed-in configuration since it
      // will be overwritten by CreatePeerConnectionWrapper with whatever is
      // stored in sdp_semantics_. So get around this by modifying the instance
      // variable before calling CreatePeerConnectionWrapper for the caller and
      // callee PeerConnections.
      SdpSemantics original_semantics = sdp_semantics_;
      sdp_semantics_ = caller_semantics;
      caller_ = CreatePeerConnectionWrapper(
          "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
          nullptr, /*media_transport_factory=*/nullptr);
      sdp_semantics_ = callee_semantics;
      callee_ = CreatePeerConnectionWrapper(
          "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
          nullptr, /*media_transport_factory=*/nullptr);
      sdp_semantics_ = original_semantics;
      return caller_ && callee_;
    }
  
    bool CreatePeerConnectionWrappersWithConfig(
        const PeerConnectionInterface::RTCConfiguration& caller_config,
        const PeerConnectionInterface::RTCConfiguration& callee_config) {
      caller_ = CreatePeerConnectionWrapper(
          "Caller", nullptr, &caller_config,
          webrtc::PeerConnectionDependencies(nullptr), nullptr,
          /*media_transport_factory=*/nullptr);
      callee_ = CreatePeerConnectionWrapper(
          "Callee", nullptr, &callee_config,
          webrtc::PeerConnectionDependencies(nullptr), nullptr,
          /*media_transport_factory=*/nullptr);
      return caller_ && callee_;
    }
  
    bool CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
        const PeerConnectionInterface::RTCConfiguration& caller_config,
        const PeerConnectionInterface::RTCConfiguration& callee_config,
        std::unique_ptr<webrtc::MediaTransportFactory> caller_factory,
        std::unique_ptr<webrtc::MediaTransportFactory> callee_factory) {
      caller_ =
          CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
                                      webrtc::PeerConnectionDependencies(nullptr),
                                      nullptr, std::move(caller_factory));
      callee_ =
          CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
                                      webrtc::PeerConnectionDependencies(nullptr),
                                      nullptr, std::move(callee_factory));
      return caller_ && callee_;
    }
  
    bool CreatePeerConnectionWrappersWithConfigAndDeps(
        const PeerConnectionInterface::RTCConfiguration& caller_config,
        webrtc::PeerConnectionDependencies caller_dependencies,
        const PeerConnectionInterface::RTCConfiguration& callee_config,
        webrtc::PeerConnectionDependencies callee_dependencies) {
      caller_ =
          CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
                                      std::move(caller_dependencies), nullptr,
                                      /*media_transport_factory=*/nullptr);
      callee_ =
          CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
                                      std::move(callee_dependencies), nullptr,
                                      /*media_transport_factory=*/nullptr);
      return caller_ && callee_;
    }
  
    bool CreatePeerConnectionWrappersWithOptions(
        const PeerConnectionFactory::Options& caller_options,
        const PeerConnectionFactory::Options& callee_options) {
      caller_ = CreatePeerConnectionWrapper(
          "Caller", &caller_options, nullptr,
          webrtc::PeerConnectionDependencies(nullptr), nullptr,
          /*media_transport_factory=*/nullptr);
      callee_ = CreatePeerConnectionWrapper(
          "Callee", &callee_options, nullptr,
          webrtc::PeerConnectionDependencies(nullptr), nullptr,
          /*media_transport_factory=*/nullptr);
      return caller_ && callee_;
    }
  
    bool CreatePeerConnectionWrappersWithFakeRtcEventLog() {
      PeerConnectionInterface::RTCConfiguration default_config;
      caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
          "Caller", nullptr, &default_config,
          webrtc::PeerConnectionDependencies(nullptr));
      callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
          "Callee", nullptr, &default_config,
          webrtc::PeerConnectionDependencies(nullptr));
      return caller_ && callee_;
    }
  
    std::unique_ptr<PeerConnectionWrapper>
    CreatePeerConnectionWrapperWithAlternateKey() {
      std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
          new FakeRTCCertificateGenerator());
      cert_generator->use_alternate_key();
  
      webrtc::PeerConnectionDependencies dependencies(nullptr);
      dependencies.cert_generator = std::move(cert_generator);
      return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
                                         std::move(dependencies), nullptr,
                                         /*media_transport_factory=*/nullptr);
    }
  
    cricket::TestTurnServer* CreateTurnServer(
        rtc::SocketAddress internal_address,
        rtc::SocketAddress external_address,
        cricket::ProtocolType type = cricket::ProtocolType::PROTO_UDP,
        const std::string& common_name = "test turn server") {
      rtc::Thread* thread = network_thread();
      std::unique_ptr<cricket::TestTurnServer> turn_server =
          network_thread()->Invoke<std::unique_ptr<cricket::TestTurnServer>>(
              RTC_FROM_HERE,
              [thread, internal_address, external_address, type, common_name] {
                return absl::make_unique<cricket::TestTurnServer>(
                    thread, internal_address, external_address, type,
                    /*ignore_bad_certs=*/true, common_name);
              });
      turn_servers_.push_back(std::move(turn_server));
      // Interactions with the turn server should be done on the network thread.
      return turn_servers_.back().get();
    }
  
    cricket::TestTurnCustomizer* CreateTurnCustomizer() {
      std::unique_ptr<cricket::TestTurnCustomizer> turn_customizer =
          network_thread()->Invoke<std::unique_ptr<cricket::TestTurnCustomizer>>(
              RTC_FROM_HERE,
              [] { return absl::make_unique<cricket::TestTurnCustomizer>(); });
      turn_customizers_.push_back(std::move(turn_customizer));
      // Interactions with the turn customizer should be done on the network
      // thread.
      return turn_customizers_.back().get();
    }
  
    // Checks that the function counters for a TestTurnCustomizer are greater than
    // 0.
    void ExpectTurnCustomizerCountersIncremented(
        cricket::TestTurnCustomizer* turn_customizer) {
      unsigned int allow_channel_data_counter =
          network_thread()->Invoke<unsigned int>(
              RTC_FROM_HERE, [turn_customizer] {
                return turn_customizer->allow_channel_data_cnt_;
              });
      EXPECT_GT(allow_channel_data_counter, 0u);
      unsigned int modify_counter = network_thread()->Invoke<unsigned int>(
          RTC_FROM_HERE,
          [turn_customizer] { return turn_customizer->modify_cnt_; });
      EXPECT_GT(modify_counter, 0u);
    }
  
    // Once called, SDP blobs and ICE candidates will be automatically signaled
    // between PeerConnections.
    void ConnectFakeSignaling() {
      caller_->set_signaling_message_receiver(callee_.get());
      callee_->set_signaling_message_receiver(caller_.get());
    }
  
    // Once called, SDP blobs will be automatically signaled between
    // PeerConnections. Note that ICE candidates will not be signaled unless they
    // are in the exchanged SDP blobs.
    void ConnectFakeSignalingForSdpOnly() {
      ConnectFakeSignaling();
      SetSignalIceCandidates(false);
    }
  
    void SetSignalingDelayMs(int delay_ms) {
      caller_->set_signaling_delay_ms(delay_ms);
      callee_->set_signaling_delay_ms(delay_ms);
    }
  
    void SetSignalIceCandidates(bool signal) {
      caller_->set_signal_ice_candidates(signal);
      callee_->set_signal_ice_candidates(signal);
    }
  
    // Messages may get lost on the unreliable DataChannel, so we send multiple
    // times to avoid test flakiness.
    void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc,
                                const std::string& data,
                                int retries) {
      for (int i = 0; i < retries; ++i) {
        dc->Send(DataBuffer(data));
      }
    }
  
    rtc::Thread* network_thread() { return network_thread_.get(); }
  
    rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
  
    webrtc::MediaTransportPair* loopback_media_transports() {
      return &loopback_media_transports_;
    }
  
    PeerConnectionWrapper* caller() { return caller_.get(); }
  
    // Set the |caller_| to the |wrapper| passed in and return the
    // original |caller_|.
    PeerConnectionWrapper* SetCallerPcWrapperAndReturnCurrent(
        PeerConnectionWrapper* wrapper) {
      PeerConnectionWrapper* old = caller_.release();
      caller_.reset(wrapper);
      return old;
    }
  
    PeerConnectionWrapper* callee() { return callee_.get(); }
  
    // Set the |callee_| to the |wrapper| passed in and return the
    // original |callee_|.
    PeerConnectionWrapper* SetCalleePcWrapperAndReturnCurrent(
        PeerConnectionWrapper* wrapper) {
      PeerConnectionWrapper* old = callee_.release();
      callee_.reset(wrapper);
      return old;
    }
  
    void SetPortAllocatorFlags(uint32_t caller_flags, uint32_t callee_flags) {
      network_thread()->Invoke<void>(
          RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::set_flags,
                                   caller()->port_allocator(), caller_flags));
      network_thread()->Invoke<void>(
          RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::set_flags,
                                   callee()->port_allocator(), callee_flags));
    }
  
    rtc::FirewallSocketServer* firewall() const { return fss_.get(); }
  
    // Expects the provided number of new frames to be received within
    // kMaxWaitForFramesMs. The new expected frames are specified in
    // |media_expectations|. Returns false if any of the expectations were
    // not met.
    bool ExpectNewFrames(const MediaExpectations& media_expectations) {
      // First initialize the expected frame counts based upon the current
      // frame count.
      int total_caller_audio_frames_expected = caller()->audio_frames_received();
      if (media_expectations.caller_audio_expectation_ ==
          MediaExpectations::kExpectSomeFrames) {
        total_caller_audio_frames_expected +=
            media_expectations.caller_audio_frames_expected_;
      }
      int total_caller_video_frames_expected =
          caller()->min_video_frames_received_per_track();
      if (media_expectations.caller_video_expectation_ ==
          MediaExpectations::kExpectSomeFrames) {
        total_caller_video_frames_expected +=
            media_expectations.caller_video_frames_expected_;
      }
      int total_callee_audio_frames_expected = callee()->audio_frames_received();
      if (media_expectations.callee_audio_expectation_ ==
          MediaExpectations::kExpectSomeFrames) {
        total_callee_audio_frames_expected +=
            media_expectations.callee_audio_frames_expected_;
      }
      int total_callee_video_frames_expected =
          callee()->min_video_frames_received_per_track();
      if (media_expectations.callee_video_expectation_ ==
          MediaExpectations::kExpectSomeFrames) {
        total_callee_video_frames_expected +=
            media_expectations.callee_video_frames_expected_;
      }
  
      // Wait for the expected frames.
      EXPECT_TRUE_WAIT(caller()->audio_frames_received() >=
                               total_caller_audio_frames_expected &&
                           caller()->min_video_frames_received_per_track() >=
                               total_caller_video_frames_expected &&
                           callee()->audio_frames_received() >=
                               total_callee_audio_frames_expected &&
                           callee()->min_video_frames_received_per_track() >=
                               total_callee_video_frames_expected,
                       kMaxWaitForFramesMs);
      bool expectations_correct =
          caller()->audio_frames_received() >=
              total_caller_audio_frames_expected &&
          caller()->min_video_frames_received_per_track() >=
              total_caller_video_frames_expected &&
          callee()->audio_frames_received() >=
              total_callee_audio_frames_expected &&
          callee()->min_video_frames_received_per_track() >=
              total_callee_video_frames_expected;
  
      // After the combined wait, print out a more detailed message upon
      // failure.
      EXPECT_GE(caller()->audio_frames_received(),
                total_caller_audio_frames_expected);
      EXPECT_GE(caller()->min_video_frames_received_per_track(),
                total_caller_video_frames_expected);
      EXPECT_GE(callee()->audio_frames_received(),
                total_callee_audio_frames_expected);
      EXPECT_GE(callee()->min_video_frames_received_per_track(),
                total_callee_video_frames_expected);
  
      // We want to make sure nothing unexpected was received.
      if (media_expectations.caller_audio_expectation_ ==
          MediaExpectations::kExpectNoFrames) {
        EXPECT_EQ(caller()->audio_frames_received(),
                  total_caller_audio_frames_expected);
        if (caller()->audio_frames_received() !=
            total_caller_audio_frames_expected) {
          expectations_correct = false;
        }
      }
      if (media_expectations.caller_video_expectation_ ==
          MediaExpectations::kExpectNoFrames) {
        EXPECT_EQ(caller()->min_video_frames_received_per_track(),
                  total_caller_video_frames_expected);
        if (caller()->min_video_frames_received_per_track() !=
            total_caller_video_frames_expected) {
          expectations_correct = false;
        }
      }
      if (media_expectations.callee_audio_expectation_ ==
          MediaExpectations::kExpectNoFrames) {
        EXPECT_EQ(callee()->audio_frames_received(),
                  total_callee_audio_frames_expected);
        if (callee()->audio_frames_received() !=
            total_callee_audio_frames_expected) {
          expectations_correct = false;
        }
      }
      if (media_expectations.callee_video_expectation_ ==
          MediaExpectations::kExpectNoFrames) {
        EXPECT_EQ(callee()->min_video_frames_received_per_track(),
                  total_callee_video_frames_expected);
        if (callee()->min_video_frames_received_per_track() !=
            total_callee_video_frames_expected) {
          expectations_correct = false;
        }
      }
      return expectations_correct;
    }
  
    void TestNegotiatedCipherSuite(
        const PeerConnectionFactory::Options& caller_options,
        const PeerConnectionFactory::Options& callee_options,
        int expected_cipher_suite) {
      ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
                                                          callee_options));
      ConnectFakeSignaling();
      caller()->AddAudioVideoTracks();
      callee()->AddAudioVideoTracks();
      caller()->CreateAndSetAndSignalOffer();
      ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
      EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
                     caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
      // TODO(bugs.webrtc.org/9456): Fix it.
      EXPECT_EQ(1, webrtc::metrics::NumEvents(
                       "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
                       expected_cipher_suite));
    }
  
    void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
                                           bool remote_gcm_enabled,
                                           int expected_cipher_suite) {
      PeerConnectionFactory::Options caller_options;
      caller_options.crypto_options.srtp.enable_gcm_crypto_suites =
          local_gcm_enabled;
      PeerConnectionFactory::Options callee_options;
      callee_options.crypto_options.srtp.enable_gcm_crypto_suites =
          remote_gcm_enabled;
      TestNegotiatedCipherSuite(caller_options, callee_options,
                                expected_cipher_suite);
    }
  
   protected:
    SdpSemantics sdp_semantics_;
  
   private:
    // |ss_| is used by |network_thread_| so it must be destroyed later.
    std::unique_ptr<rtc::VirtualSocketServer> ss_;
    std::unique_ptr<rtc::FirewallSocketServer> fss_;
    // |network_thread_| and |worker_thread_| are used by both
    // |caller_| and |callee_| so they must be destroyed
    // later.
    std::unique_ptr<rtc::Thread> network_thread_;
    std::unique_ptr<rtc::Thread> worker_thread_;
    // The turn servers and turn customizers should be accessed & deleted on the
    // network thread to avoid a race with the socket read/write that occurs
    // on the network thread.
    std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
    std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_;
    webrtc::MediaTransportPair loopback_media_transports_;
    std::unique_ptr<PeerConnectionWrapper> caller_;
    std::unique_ptr<PeerConnectionWrapper> callee_;
  };
  
  class PeerConnectionIntegrationTest
      : public PeerConnectionIntegrationBaseTest,
        public ::testing::WithParamInterface<SdpSemantics> {
   protected:
    PeerConnectionIntegrationTest()
        : PeerConnectionIntegrationBaseTest(GetParam()) {}
  };
  
  class PeerConnectionIntegrationTestPlanB
      : public PeerConnectionIntegrationBaseTest {
   protected:
    PeerConnectionIntegrationTestPlanB()
        : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB) {}
  };
  
  class PeerConnectionIntegrationTestUnifiedPlan
      : public PeerConnectionIntegrationBaseTest {
   protected:
    PeerConnectionIntegrationTestUnifiedPlan()
        : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
  };
  
  // Test the OnFirstPacketReceived callback from audio/video RtpReceivers.  This
  // includes testing that the callback is invoked if an observer is connected
  // after the first packet has already been received.
  TEST_P(PeerConnectionIntegrationTest,
         RtpReceiverObserverOnFirstPacketReceived) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    // Start offer/answer exchange and wait for it to complete.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Should be one receiver each for audio/video.
    EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
    EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
    // Wait for all "first packet received" callbacks to be fired.
    EXPECT_TRUE_WAIT(
        absl::c_all_of(caller()->rtp_receiver_observers(),
                       [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                         return o->first_packet_received();
                       }),
        kMaxWaitForFramesMs);
    EXPECT_TRUE_WAIT(
        absl::c_all_of(callee()->rtp_receiver_observers(),
                       [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                         return o->first_packet_received();
                       }),
        kMaxWaitForFramesMs);
    // If new observers are set after the first packet was already received, the
    // callback should still be invoked.
    caller()->ResetRtpReceiverObservers();
    callee()->ResetRtpReceiverObservers();
    EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
    EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
    EXPECT_TRUE(
        absl::c_all_of(caller()->rtp_receiver_observers(),
                       [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                         return o->first_packet_received();
                       }));
    EXPECT_TRUE(
        absl::c_all_of(callee()->rtp_receiver_observers(),
                       [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                         return o->first_packet_received();
                       }));
  }
  
  class DummyDtmfObserver : public DtmfSenderObserverInterface {
   public:
    DummyDtmfObserver() : completed_(false) {}
  
    // Implements DtmfSenderObserverInterface.
    void OnToneChange(const std::string& tone) override {
      tones_.push_back(tone);
      if (tone.empty()) {
        completed_ = true;
      }
    }
  
    const std::vector<std::string>& tones() const { return tones_; }
    bool completed() const { return completed_; }
  
   private:
    bool completed_;
    std::vector<std::string> tones_;
  };
  
  // Assumes |sender| already has an audio track added and the offer/answer
  // exchange is done.
  void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender,
                                    PeerConnectionWrapper* receiver) {
    // We should be able to get a DTMF sender from the local sender.
    rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender =
        sender->pc()->GetSenders().at(0)->GetDtmfSender();
    ASSERT_TRUE(dtmf_sender);
    DummyDtmfObserver observer;
    dtmf_sender->RegisterObserver(&observer);
  
    // Test the DtmfSender object just created.
    EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
    EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));
  
    EXPECT_TRUE_WAIT(observer.completed(), kDefaultTimeout);
    std::vector<std::string> tones = {"1", "a", ""};
    EXPECT_EQ(tones, observer.tones());
    dtmf_sender->UnregisterObserver();
    // TODO(deadbeef): Verify the tones were actually received end-to-end.
  }
  
  // Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each
  // direction).
  TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Only need audio for DTMF.
    caller()->AddAudioTrack();
    callee()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // DTLS must finish before the DTMF sender can be used reliably.
    ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
    TestDtmfFromSenderToReceiver(caller(), callee());
    TestDtmfFromSenderToReceiver(callee(), caller());
  }
  
  // Basic end-to-end test, verifying media can be encoded/transmitted/decoded
  // between two connections, using DTLS-SRTP.
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    // Do normal offer/answer and wait for some frames to be received in each
    // direction.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
    EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                            webrtc::kEnumCounterKeyProtocolDtls));
    EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                            webrtc::kEnumCounterKeyProtocolSdes));
  }
  
  // Uses SDES instead of DTLS for key agreement.
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
    PeerConnectionInterface::RTCConfiguration sdes_config;
    sdes_config.enable_dtls_srtp.emplace(false);
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config));
    ConnectFakeSignaling();
  
    // Do normal offer/answer and wait for some frames to be received in each
    // direction.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
    EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                            webrtc::kEnumCounterKeyProtocolSdes));
    EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                            webrtc::kEnumCounterKeyProtocolDtls));
  }
  
  // Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
  // certificate once the DTLS handshake has finished.
  TEST_P(PeerConnectionIntegrationTest,
         GetRemoteAudioSSLCertificateReturnsExchangedCertificate) {
    auto GetRemoteAudioSSLCertificate = [](PeerConnectionWrapper* wrapper) {
      auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
      auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
      return pc->GetRemoteAudioSSLCertificate();
    };
    auto GetRemoteAudioSSLCertChain = [](PeerConnectionWrapper* wrapper) {
      auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
      auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
      return pc->GetRemoteAudioSSLCertChain();
    };
  
    auto caller_cert = rtc::RTCCertificate::FromPEM(kRsaPems[0]);
    auto callee_cert = rtc::RTCCertificate::FromPEM(kRsaPems[1]);
  
    // Configure each side with a known certificate so they can be compared later.
    PeerConnectionInterface::RTCConfiguration caller_config;
    caller_config.enable_dtls_srtp.emplace(true);
    caller_config.certificates.push_back(caller_cert);
    PeerConnectionInterface::RTCConfiguration callee_config;
    callee_config.enable_dtls_srtp.emplace(true);
    callee_config.certificates.push_back(callee_cert);
    ASSERT_TRUE(
        CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));
    ConnectFakeSignaling();
  
    // When first initialized, there should not be a remote SSL certificate (and
    // calling this method should not crash).
    EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller()));
    EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee()));
    EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(caller()));
    EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(callee()));
  
    caller()->AddAudioTrack();
    callee()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
  
    // Once DTLS has been connected, each side should return the other's SSL
    // certificate when calling GetRemoteAudioSSLCertificate.
  
    auto caller_remote_cert = GetRemoteAudioSSLCertificate(caller());
    ASSERT_TRUE(caller_remote_cert);
    EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
              caller_remote_cert->ToPEMString());
  
    auto callee_remote_cert = GetRemoteAudioSSLCertificate(callee());
    ASSERT_TRUE(callee_remote_cert);
    EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
              callee_remote_cert->ToPEMString());
  
    auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller());
    ASSERT_TRUE(caller_remote_cert_chain);
    ASSERT_EQ(1U, caller_remote_cert_chain->GetSize());
    auto remote_cert = &caller_remote_cert_chain->Get(0);
    EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
              remote_cert->ToPEMString());
  
    auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee());
    ASSERT_TRUE(callee_remote_cert_chain);
    ASSERT_EQ(1U, callee_remote_cert_chain->GetSize());
    remote_cert = &callee_remote_cert_chain->Get(0);
    EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
              remote_cert->ToPEMString());
  }
  
  // This test sets up a call between two parties with a source resolution of
  // 1280x720 and verifies that a 16:9 aspect ratio is received.
  TEST_P(PeerConnectionIntegrationTest,
         Send1280By720ResolutionAndReceive16To9AspectRatio) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    // Add video tracks with 16:9 aspect ratio, size 1280 x 720.
    webrtc::FakePeriodicVideoSource::Config config;
    config.width = 1280;
    config.height = 720;
    config.timestamp_offset_ms = rtc::TimeMillis();
    caller()->AddTrack(caller()->CreateLocalVideoTrackWithConfig(config));
    callee()->AddTrack(callee()->CreateLocalVideoTrackWithConfig(config));
  
    // Do normal offer/answer and wait for at least one frame to be received in
    // each direction.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                         callee()->min_video_frames_received_per_track() > 0,
                     kMaxWaitForFramesMs);
  
    // Check rendered aspect ratio.
    EXPECT_EQ(16.0 / 9, caller()->local_rendered_aspect_ratio());
    EXPECT_EQ(16.0 / 9, caller()->rendered_aspect_ratio());
    EXPECT_EQ(16.0 / 9, callee()->local_rendered_aspect_ratio());
    EXPECT_EQ(16.0 / 9, callee()->rendered_aspect_ratio());
  }
  
  // This test sets up an one-way call, with media only from caller to
  // callee.
  TEST_P(PeerConnectionIntegrationTest, OneWayMediaCall) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudioAndVideo();
    media_expectations.CallerExpectsNoAudio();
    media_expectations.CallerExpectsNoVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // This test sets up a audio call initially, with the callee rejecting video
  // initially. Then later the callee decides to upgrade to audio/video, and
  // initiates a new offer/answer exchange.
  TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Initially, offer an audio/video stream from the caller, but refuse to
    // send/receive video on the callee side.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioTrack();
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      PeerConnectionInterface::RTCOfferAnswerOptions options;
      options.offer_to_receive_video = 0;
      callee()->SetOfferAnswerOptions(options);
    } else {
      callee()->SetRemoteOfferHandler([this] {
        callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
      });
    }
    // Do offer/answer and make sure audio is still received end-to-end.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    {
      MediaExpectations media_expectations;
      media_expectations.ExpectBidirectionalAudio();
      media_expectations.ExpectNoVideo();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
    // Sanity check that the callee's description has a rejected video section.
    ASSERT_NE(nullptr, callee()->pc()->local_description());
    const ContentInfo* callee_video_content =
        GetFirstVideoContent(callee()->pc()->local_description()->description());
    ASSERT_NE(nullptr, callee_video_content);
    EXPECT_TRUE(callee_video_content->rejected);
  
    // Now negotiate with video and ensure negotiation succeeds, with video
    // frames and additional audio frames being received.
    callee()->AddVideoTrack();
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      PeerConnectionInterface::RTCOfferAnswerOptions options;
      options.offer_to_receive_video = 1;
      callee()->SetOfferAnswerOptions(options);
    } else {
      callee()->SetRemoteOfferHandler(nullptr);
      caller()->SetRemoteOfferHandler([this] {
        // The caller creates a new transceiver to receive video on when receiving
        // the offer, but by default it is send only.
        auto transceivers = caller()->pc()->GetTransceivers();
        ASSERT_EQ(3U, transceivers.size());
        ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
                  transceivers[2]->receiver()->media_type());
        transceivers[2]->sender()->SetTrack(caller()->CreateLocalVideoTrack());
        transceivers[2]->SetDirection(RtpTransceiverDirection::kSendRecv);
      });
    }
    callee()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    {
      // Expect additional audio frames to be received after the upgrade.
      MediaExpectations media_expectations;
      media_expectations.ExpectBidirectionalAudioAndVideo();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
  }
  
  // Simpler than the above test; just add an audio track to an established
  // video-only connection.
  TEST_P(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Do initial offer/answer with just a video track.
    caller()->AddVideoTrack();
    callee()->AddVideoTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Now add an audio track and do another offer/answer.
    caller()->AddAudioTrack();
    callee()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Ensure both audio and video frames are received end-to-end.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // This test sets up a call that's transferred to a new caller with a different
  // DTLS fingerprint.
  TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Keep the original peer around which will still send packets to the
    // receiving client. These SRTP packets will be dropped.
    std::unique_ptr<PeerConnectionWrapper> original_peer(
        SetCallerPcWrapperAndReturnCurrent(
            CreatePeerConnectionWrapperWithAlternateKey().release()));
    // TODO(deadbeef): Why do we call Close here? That goes against the comment
    // directly above.
    original_peer->pc()->Close();
  
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Wait for some additional frames to be transmitted end-to-end.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // This test sets up a call that's transferred to a new callee with a different
  // DTLS fingerprint.
  TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Keep the original peer around which will still send packets to the
    // receiving client. These SRTP packets will be dropped.
    std::unique_ptr<PeerConnectionWrapper> original_peer(
        SetCalleePcWrapperAndReturnCurrent(
            CreatePeerConnectionWrapperWithAlternateKey().release()));
    // TODO(deadbeef): Why do we call Close here? That goes against the comment
    // directly above.
    original_peer->pc()->Close();
  
    ConnectFakeSignaling();
    callee()->AddAudioVideoTracks();
    caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Wait for some additional frames to be transmitted end-to-end.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // This test sets up a non-bundled call and negotiates bundling at the same
  // time as starting an ICE restart. When bundling is in effect in the restart,
  // the DTLS-SRTP context should be successfully reset.
  TEST_P(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    // Remove the bundle group from the SDP received by the callee.
    callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
      desc->RemoveGroupByName("BUNDLE");
    });
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    {
      MediaExpectations media_expectations;
      media_expectations.ExpectBidirectionalAudioAndVideo();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
    // Now stop removing the BUNDLE group, and trigger an ICE restart.
    callee()->SetReceivedSdpMunger(nullptr);
    caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Expect additional frames to be received after the ICE restart.
    {
      MediaExpectations media_expectations;
      media_expectations.ExpectBidirectionalAudioAndVideo();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
  }
  
  // Test CVO (Coordination of Video Orientation). If a video source is rotated
  // and both peers support the CVO RTP header extension, the actual video frames
  // don't need to be encoded in different resolutions, since the rotation is
  // communicated through the RTP header extension.
  TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Add rotated video tracks.
    caller()->AddTrack(
        caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
    callee()->AddTrack(
        callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
  
    // Wait for video frames to be received by both sides.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                         callee()->min_video_frames_received_per_track() > 0,
                     kMaxWaitForFramesMs);
  
    // Ensure that the aspect ratio is unmodified.
    // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
    // not just assumed.
    EXPECT_EQ(4.0 / 3, caller()->local_rendered_aspect_ratio());
    EXPECT_EQ(4.0 / 3, caller()->rendered_aspect_ratio());
    EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio());
    EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio());
    // Ensure that the CVO bits were surfaced to the renderer.
    EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation());
    EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation());
  }
  
  // Test that when the CVO extension isn't supported, video is rotated the
  // old-fashioned way, by encoding rotated frames.
  TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Add rotated video tracks.
    caller()->AddTrack(
        caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
    callee()->AddTrack(
        callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
  
    // Remove the CVO extension from the offered SDP.
    callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
      cricket::VideoContentDescription* video =
          GetFirstVideoContentDescription(desc);
      video->ClearRtpHeaderExtensions();
    });
    // Wait for video frames to be received by both sides.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                         callee()->min_video_frames_received_per_track() > 0,
                     kMaxWaitForFramesMs);
  
    // Expect that the aspect ratio is inversed to account for the 90/270 degree
    // rotation.
    // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
    // not just assumed.
    EXPECT_EQ(3.0 / 4, caller()->local_rendered_aspect_ratio());
    EXPECT_EQ(3.0 / 4, caller()->rendered_aspect_ratio());
    EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio());
    EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio());
    // Expect that each endpoint is unaware of the rotation of the other endpoint.
    EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation());
    EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation());
  }
  
  // Test that if the answerer rejects the audio m= section, no audio is sent or
  // received, but video still can be.
  TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      // Only add video track for callee, and set offer_to_receive_audio to 0, so
      // it will reject the audio m= section completely.
      PeerConnectionInterface::RTCOfferAnswerOptions options;
      options.offer_to_receive_audio = 0;
      callee()->SetOfferAnswerOptions(options);
    } else {
      // Stopping the audio RtpTransceiver will cause the media section to be
      // rejected in the answer.
      callee()->SetRemoteOfferHandler([this] {
        callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)->Stop();
      });
    }
    callee()->AddTrack(callee()->CreateLocalVideoTrack());
    // Do offer/answer and wait for successful end-to-end video frames.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalVideo();
    media_expectations.ExpectNoAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  
    // Sanity check that the callee's description has a rejected audio section.
    ASSERT_NE(nullptr, callee()->pc()->local_description());
    const ContentInfo* callee_audio_content =
        GetFirstAudioContent(callee()->pc()->local_description()->description());
    ASSERT_NE(nullptr, callee_audio_content);
    EXPECT_TRUE(callee_audio_content->rejected);
    if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
      // The caller's transceiver should have stopped after receiving the answer.
      EXPECT_TRUE(caller()
                      ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
                      ->stopped());
    }
  }
  
  // Test that if the answerer rejects the video m= section, no video is sent or
  // received, but audio still can be.
  TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      // Only add audio track for callee, and set offer_to_receive_video to 0, so
      // it will reject the video m= section completely.
      PeerConnectionInterface::RTCOfferAnswerOptions options;
      options.offer_to_receive_video = 0;
      callee()->SetOfferAnswerOptions(options);
    } else {
      // Stopping the video RtpTransceiver will cause the media section to be
      // rejected in the answer.
      callee()->SetRemoteOfferHandler([this] {
        callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
      });
    }
    callee()->AddTrack(callee()->CreateLocalAudioTrack());
    // Do offer/answer and wait for successful end-to-end audio frames.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudio();
    media_expectations.ExpectNoVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  
    // Sanity check that the callee's description has a rejected video section.
    ASSERT_NE(nullptr, callee()->pc()->local_description());
    const ContentInfo* callee_video_content =
        GetFirstVideoContent(callee()->pc()->local_description()->description());
    ASSERT_NE(nullptr, callee_video_content);
    EXPECT_TRUE(callee_video_content->rejected);
    if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
      // The caller's transceiver should have stopped after receiving the answer.
      EXPECT_TRUE(caller()
                      ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
                      ->stopped());
    }
  }
  
  // Test that if the answerer rejects both audio and video m= sections, nothing
  // bad happens.
  // TODO(deadbeef): Test that a data channel still works. Currently this doesn't
  // test anything but the fact that negotiation succeeds, which doesn't mean
  // much.
  TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      // Don't give the callee any tracks, and set offer_to_receive_X to 0, so it
      // will reject both audio and video m= sections.
      PeerConnectionInterface::RTCOfferAnswerOptions options;
      options.offer_to_receive_audio = 0;
      options.offer_to_receive_video = 0;
      callee()->SetOfferAnswerOptions(options);
    } else {
      callee()->SetRemoteOfferHandler([this] {
        // Stopping all transceivers will cause all media sections to be rejected.
        for (const auto& transceiver : callee()->pc()->GetTransceivers()) {
          transceiver->Stop();
        }
      });
    }
    // Do offer/answer and wait for stable signaling state.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Sanity check that the callee's description has rejected m= sections.
    ASSERT_NE(nullptr, callee()->pc()->local_description());
    const ContentInfo* callee_audio_content =
        GetFirstAudioContent(callee()->pc()->local_description()->description());
    ASSERT_NE(nullptr, callee_audio_content);
    EXPECT_TRUE(callee_audio_content->rejected);
    const ContentInfo* callee_video_content =
        GetFirstVideoContent(callee()->pc()->local_description()->description());
    ASSERT_NE(nullptr, callee_video_content);
    EXPECT_TRUE(callee_video_content->rejected);
  }
  
  // This test sets up an audio and video call between two parties. After the
  // call runs for a while, the caller sends an updated offer with video being
  // rejected. Once the re-negotiation is done, the video flow should stop and
  // the audio flow should continue.
  TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    {
      MediaExpectations media_expectations;
      media_expectations.ExpectBidirectionalAudioAndVideo();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
    // Renegotiate, rejecting the video m= section.
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      caller()->SetGeneratedSdpMunger(
          [](cricket::SessionDescription* description) {
            for (cricket::ContentInfo& content : description->contents()) {
              if (cricket::IsVideoContent(&content)) {
                content.rejected = true;
              }
            }
          });
    } else {
      caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
    }
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
  
    // Sanity check that the caller's description has a rejected video section.
    ASSERT_NE(nullptr, caller()->pc()->local_description());
    const ContentInfo* caller_video_content =
        GetFirstVideoContent(caller()->pc()->local_description()->description());
    ASSERT_NE(nullptr, caller_video_content);
    EXPECT_TRUE(caller_video_content->rejected);
    // Wait for some additional audio frames to be received.
    {
      MediaExpectations media_expectations;
      media_expectations.ExpectBidirectionalAudio();
      media_expectations.ExpectNoVideo();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
  }
  
  // Do one offer/answer with audio, another that disables it (rejecting the m=
  // section), and another that re-enables it. Regression test for:
  // bugs.webrtc.org/6023
  TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    // Add audio track, do normal offer/answer.
    rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
        caller()->CreateLocalAudioTrack();
    rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
        caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Remove audio track, and set offer_to_receive_audio to false to cause the
    // m= section to be completely disabled, not just "recvonly".
    caller()->pc()->RemoveTrack(sender);
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 0;
    caller()->SetOfferAnswerOptions(options);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Add the audio track again, expecting negotiation to succeed and frames to
    // flow.
    sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
    options.offer_to_receive_audio = 1;
    caller()->SetOfferAnswerOptions(options);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio();
    EXPECT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Basic end-to-end test, but without SSRC/MSID signaling. This functionality
  // is needed to support legacy endpoints.
  // TODO(deadbeef): When we support the MID extension and demuxing on MID, also
  // add a test for an end-to-end test without MID signaling either (basically,
  // the minimum acceptable SDP).
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Add audio and video, testing that packets can be demuxed on payload type.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    // Remove SSRCs and MSIDs from the received offer SDP.
    callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Basic end-to-end test, without SSRC signaling. This means that the track
  // was created properly and frames are delivered when the MSIDs are communicated
  // with a=msid lines and no a=ssrc lines.
  TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
         EndToEndCallWithoutSsrcSignaling) {
    const char kStreamId[] = "streamId";
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Add just audio tracks.
    caller()->AddTrack(caller()->CreateLocalAudioTrack(), {kStreamId});
    callee()->AddAudioTrack();
  
    // Remove SSRCs from the received offer SDP.
    callee()->SetReceivedSdpMunger(RemoveSsrcsAndKeepMsids);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Tests that video flows between multiple video tracks when SSRCs are not
  // signaled. This exercises the MID RTP header extension which is needed to
  // demux the incoming video tracks.
  TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
         EndToEndCallWithTwoVideoTracksAndNoSignaledSsrc) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddVideoTrack();
    caller()->AddVideoTrack();
    callee()->AddVideoTrack();
    callee()->AddVideoTrack();
  
    caller()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
    callee()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_EQ(2u, caller()->pc()->GetReceivers().size());
    ASSERT_EQ(2u, callee()->pc()->GetReceivers().size());
  
    // Expect video to be received in both directions on both tracks.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalVideo();
    EXPECT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLinePresent) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioTrack();
    caller()->AddVideoTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    auto callee_receivers = callee()->pc()->GetReceivers();
    ASSERT_EQ(2u, callee_receivers.size());
    EXPECT_TRUE(callee_receivers[0]->stream_ids().empty());
    EXPECT_TRUE(callee_receivers[1]->stream_ids().empty());
  }
  
  TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLineMissing) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioTrack();
    caller()->AddVideoTrack();
    callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    auto callee_receivers = callee()->pc()->GetReceivers();
    ASSERT_EQ(2u, callee_receivers.size());
    ASSERT_EQ(1u, callee_receivers[0]->stream_ids().size());
    ASSERT_EQ(1u, callee_receivers[1]->stream_ids().size());
    EXPECT_EQ(callee_receivers[0]->stream_ids()[0],
              callee_receivers[1]->stream_ids()[0]);
    EXPECT_EQ(callee_receivers[0]->streams()[0],
              callee_receivers[1]->streams()[0]);
  }
  
  // Test that if two video tracks are sent (from caller to callee, in this test),
  // they're transmitted correctly end-to-end.
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Add one audio/video stream, and one video-only stream.
    caller()->AddAudioVideoTracks();
    caller()->AddVideoTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_EQ(3u, callee()->pc()->GetReceivers().size());
  
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) {
    bool first = true;
    for (cricket::ContentInfo& content : desc->contents()) {
      if (first) {
        first = false;
        continue;
      }
      content.bundle_only = true;
    }
    first = true;
    for (cricket::TransportInfo& transport : desc->transport_infos()) {
      if (first) {
        first = false;
        continue;
      }
      transport.description.ice_ufrag.clear();
      transport.description.ice_pwd.clear();
      transport.description.connection_role = cricket::CONNECTIONROLE_NONE;
      transport.description.identity_fingerprint.reset(nullptr);
    }
  }
  
  // Test that if applying a true "max bundle" offer, which uses ports of 0,
  // "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and
  // "a=ice-pwd" for all but the audio "m=" section, negotiation still completes
  // successfully and media flows.
  // TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works.
  // TODO(deadbeef): Won't need this test once we start generating actual
  // standards-compliant SDP.
  TEST_P(PeerConnectionIntegrationTest,
         EndToEndCallWithSpecCompliantMaxBundleOffer) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    // Do the equivalent of setting the port to 0, adding a=bundle-only, and
    // removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all
    // but the first m= section.
    callee()->SetReceivedSdpMunger(MakeSpecCompliantMaxBundleOffer);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Test that we can receive the audio output level from a remote audio track.
  // TODO(deadbeef): Use a fake audio source and verify that the output level is
  // exactly what the source on the other side was configured with.
  TEST_P(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Just add an audio track.
    caller()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Get the audio output level stats. Note that the level is not available
    // until an RTCP packet has been received.
    EXPECT_TRUE_WAIT(callee()->OldGetStats()->AudioOutputLevel() > 0,
                     kMaxWaitForFramesMs);
  }
  
  // Test that an audio input level is reported.
  // TODO(deadbeef): Use a fake audio source and verify that the input level is
  // exactly what the source was configured with.
  TEST_P(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Just add an audio track.
    caller()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Get the audio input level stats. The level should be available very
    // soon after the test starts.
    EXPECT_TRUE_WAIT(caller()->OldGetStats()->AudioInputLevel() > 0,
                     kMaxWaitForStatsMs);
  }
  
  // Test that we can get incoming byte counts from both audio and video tracks.
  TEST_P(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    // Do offer/answer, wait for the callee to receive some frames.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  
    // Get a handle to the remote tracks created, so they can be used as GetStats
    // filters.
    for (const auto& receiver : callee()->pc()->GetReceivers()) {
      // We received frames, so we definitely should have nonzero "received bytes"
      // stats at this point.
      EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(),
                0);
    }
  }
  
  // Test that we can get outgoing byte counts from both audio and video tracks.
  TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    auto audio_track = caller()->CreateLocalAudioTrack();
    auto video_track = caller()->CreateLocalVideoTrack();
    caller()->AddTrack(audio_track);
    caller()->AddTrack(video_track);
    // Do offer/answer, wait for the callee to receive some frames.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  
    // The callee received frames, so we definitely should have nonzero "sent
    // bytes" stats at this point.
    EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0);
    EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
  }
  
  // Test that we can get capture start ntp time.
  TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioTrack();
  
    callee()->AddAudioTrack();
  
    // Do offer/answer, wait for the callee to receive some frames.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Get the remote audio track created on the receiver, so they can be used as
    // GetStats filters.
    auto receivers = callee()->pc()->GetReceivers();
    ASSERT_EQ(1u, receivers.size());
    auto remote_audio_track = receivers[0]->track();
  
    // Get the audio output level stats. Note that the level is not available
    // until an RTCP packet has been received.
    EXPECT_TRUE_WAIT(
        callee()->OldGetStatsForTrack(remote_audio_track)->CaptureStartNtpTime() >
            0,
        2 * kMaxWaitForFramesMs);
  }
  
  // Test that the track ID is associated with all local and remote SSRC stats
  // using the old GetStats() and more than 1 audio and more than 1 video track.
  // This is a regression test for crbug.com/906988
  TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
         OldGetStatsAssociatesTrackIdForManyMediaSections) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    auto audio_sender_1 = caller()->AddAudioTrack();
    auto video_sender_1 = caller()->AddVideoTrack();
    auto audio_sender_2 = caller()->AddAudioTrack();
    auto video_sender_2 = caller()->AddVideoTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudioAndVideo();
    ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);
  
    std::vector<std::string> track_ids = {
        audio_sender_1->track()->id(), video_sender_1->track()->id(),
        audio_sender_2->track()->id(), video_sender_2->track()->id()};
  
    auto caller_stats = caller()->OldGetStats();
    EXPECT_THAT(caller_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
    auto callee_stats = callee()->OldGetStats();
    EXPECT_THAT(callee_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
  }
  
  // Test that the new GetStats() returns stats for all outgoing/incoming streams
  // with the correct track IDs if there are more than one audio and more than one
  // video senders/receivers.
  TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    auto audio_sender_1 = caller()->AddAudioTrack();
    auto video_sender_1 = caller()->AddVideoTrack();
    auto audio_sender_2 = caller()->AddAudioTrack();
    auto video_sender_2 = caller()->AddVideoTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudioAndVideo();
    ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);
  
    std::vector<std::string> track_ids = {
        audio_sender_1->track()->id(), video_sender_1->track()->id(),
        audio_sender_2->track()->id(), video_sender_2->track()->id()};
  
    rtc::scoped_refptr<const webrtc::RTCStatsReport> caller_report =
        caller()->NewGetStats();
    ASSERT_TRUE(caller_report);
    auto outbound_stream_stats =
        caller_report->GetStatsOfType<webrtc::RTCOutboundRTPStreamStats>();
    ASSERT_EQ(4u, outbound_stream_stats.size());
    std::vector<std::string> outbound_track_ids;
    for (const auto& stat : outbound_stream_stats) {
      ASSERT_TRUE(stat->bytes_sent.is_defined());
      EXPECT_LT(0u, *stat->bytes_sent);
      ASSERT_TRUE(stat->track_id.is_defined());
      const auto* track_stat =
          caller_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
      ASSERT_TRUE(track_stat);
      outbound_track_ids.push_back(*track_stat->track_identifier);
    }
    EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids));
  
    rtc::scoped_refptr<const webrtc::RTCStatsReport> callee_report =
        callee()->NewGetStats();
    ASSERT_TRUE(callee_report);
    auto inbound_stream_stats =
        callee_report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
    ASSERT_EQ(4u, inbound_stream_stats.size());
    std::vector<std::string> inbound_track_ids;
    for (const auto& stat : inbound_stream_stats) {
      ASSERT_TRUE(stat->bytes_received.is_defined());
      EXPECT_LT(0u, *stat->bytes_received);
      ASSERT_TRUE(stat->track_id.is_defined());
      const auto* track_stat =
          callee_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
      ASSERT_TRUE(track_stat);
      inbound_track_ids.push_back(*track_stat->track_identifier);
    }
    EXPECT_THAT(inbound_track_ids, UnorderedElementsAreArray(track_ids));
  }
  
  // Test that we can get stats (using the new stats implementation) for
  // unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
  // SDP.
  TEST_P(PeerConnectionIntegrationTest,
         GetStatsForUnsignaledStreamWithNewStatsApi) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioTrack();
    // Remove SSRCs and MSIDs from the received offer SDP.
    callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio(1);
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  
    // We received a frame, so we should have nonzero "bytes received" stats for
    // the unsignaled stream, if stats are working for it.
    rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
        callee()->NewGetStats();
    ASSERT_NE(nullptr, report);
    auto inbound_stream_stats =
        report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
    ASSERT_EQ(1U, inbound_stream_stats.size());
    ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
    ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
    ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined());
  }
  
  // Same as above but for the legacy stats implementation.
  TEST_P(PeerConnectionIntegrationTest,
         GetStatsForUnsignaledStreamWithOldStatsApi) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioTrack();
    // Remove SSRCs and MSIDs from the received offer SDP.
    callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Note that, since the old stats implementation associates SSRCs with tracks
    // using SDP, when SSRCs aren't signaled in SDP these stats won't have an
    // associated track ID. So we can't use the track "selector" argument.
    //
    // Also, we use "EXPECT_TRUE_WAIT" because the stats collector may decide to
    // return cached stats if not enough time has passed since the last update.
    EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0,
                     kDefaultTimeout);
  }
  
  // Test that we can successfully get the media related stats (audio level
  // etc.) for the unsignaled stream.
  TEST_P(PeerConnectionIntegrationTest,
         GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    // Remove SSRCs and MSIDs from the received offer SDP.
    callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio(1);
    media_expectations.CalleeExpectsSomeVideo(1);
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  
    rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
        callee()->NewGetStats();
    ASSERT_NE(nullptr, report);
  
    auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
    auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats);
    ASSERT_GE(audio_index, 0);
    EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined());
  }
  
  // Helper for test below.
  void ModifySsrcs(cricket::SessionDescription* desc) {
    for (ContentInfo& content : desc->contents()) {
      for (StreamParams& stream :
           content.media_description()->mutable_streams()) {
        for (uint32_t& ssrc : stream.ssrcs) {
          ssrc = rtc::CreateRandomId();
        }
      }
    }
  }
  
  // Test that the "RTCMediaSteamTrackStats"  object is updated correctly when
  // SSRCs are unsignaled, and the SSRC of the received (audio) stream changes.
  // This should result in two "RTCInboundRTPStreamStats", but only one
  // "RTCMediaStreamTrackStats", whose counters go up continuously rather than
  // being reset to 0 once the SSRC change occurs.
  //
  // Regression test for this bug:
  // https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
  //
  // The bug causes the track stats to only represent one of the two streams:
  // whichever one has the higher SSRC. So with this bug, there was a 50% chance
  // that the track stat counters would reset to 0 when the new stream is
  // received, and a 50% chance that they'll stop updating (while
  // "concealed_samples" continues increasing, due to silence being generated for
  // the inactive stream).
  TEST_P(PeerConnectionIntegrationTest,
         TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioTrack();
    // Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint
    // that doesn't signal SSRCs (from the callee's perspective).
    callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Wait for 50 audio frames (500ms of audio) to be received by the callee.
    {
      MediaExpectations media_expectations;
      media_expectations.CalleeExpectsSomeAudio(50);
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
    // Some audio frames were received, so we should have nonzero "samples
    // received" for the track.
    rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
        callee()->NewGetStats();
    ASSERT_NE(nullptr, report);
    auto track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
    ASSERT_EQ(1U, track_stats.size());
    ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
    ASSERT_GT(*track_stats[0]->total_samples_received, 0U);
    // uint64_t prev_samples_received = *track_stats[0]->total_samples_received;
  
    // Create a new offer and munge it to cause the caller to use a new SSRC.
    caller()->SetGeneratedSdpMunger(ModifySsrcs);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Wait for 25 more audio frames (250ms of audio) to be received, from the new
    // SSRC.
    {
      MediaExpectations media_expectations;
      media_expectations.CalleeExpectsSomeAudio(25);
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
  
    report = callee()->NewGetStats();
    ASSERT_NE(nullptr, report);
    track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
    ASSERT_EQ(1U, track_stats.size());
    ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
    // The "total samples received" stat should only be greater than it was
    // before.
    // TODO(deadbeef): Uncomment this assertion once the bug is completely fixed.
    // Right now, the new SSRC will cause the counters to reset to 0.
    // EXPECT_GT(*track_stats[0]->total_samples_received, prev_samples_received);
  
    // Additionally, the percentage of concealed samples (samples generated to
    // conceal packet loss) should be less than 50%. If it's greater, that's a
    // good sign that we're seeing stats from the old stream that's no longer
    // receiving packets, and is generating concealed samples of silence.
    constexpr double kAcceptableConcealedSamplesPercentage = 0.50;
    ASSERT_TRUE(track_stats[0]->concealed_samples.is_defined());
    EXPECT_LT(*track_stats[0]->concealed_samples,
              *track_stats[0]->total_samples_received *
                  kAcceptableConcealedSamplesPercentage);
  
    // Also ensure that we have two "RTCInboundRTPStreamStats" as expected, as a
    // sanity check that the SSRC really changed.
    // TODO(deadbeef): This isn't working right now, because we're not returning
    // *any* stats for the inactive stream. Uncomment when the bug is completely
    // fixed.
    // auto inbound_stream_stats =
    //     report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
    // ASSERT_EQ(2U, inbound_stream_stats.size());
  }
  
  // Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
    PeerConnectionFactory::Options dtls_10_options;
    dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
                                                        dtls_10_options));
    ConnectFakeSignaling();
    // Do normal offer/answer and wait for some frames to be received in each
    // direction.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated.
  TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
    PeerConnectionFactory::Options dtls_10_options;
    dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
                                                        dtls_10_options));
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
                         caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
                     kDefaultTimeout);
    EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                   caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
    // TODO(bugs.webrtc.org/9456): Fix it.
    EXPECT_EQ(1, webrtc::metrics::NumEvents(
                     "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
                     kDefaultSrtpCryptoSuite));
  }
  
  // Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
  TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
    PeerConnectionFactory::Options dtls_12_options;
    dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options,
                                                        dtls_12_options));
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
                         caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
                     kDefaultTimeout);
    EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                   caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
    // TODO(bugs.webrtc.org/9456): Fix it.
    EXPECT_EQ(1, webrtc::metrics::NumEvents(
                     "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
                     kDefaultSrtpCryptoSuite));
  }
  
  // Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
  // callee only supports 1.0.
  TEST_P(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
    PeerConnectionFactory::Options caller_options;
    caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
    PeerConnectionFactory::Options callee_options;
    callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
    ASSERT_TRUE(
        CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
    ConnectFakeSignaling();
    // Do normal offer/answer and wait for some frames to be received in each
    // direction.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the
  // callee supports 1.2.
  TEST_P(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
    PeerConnectionFactory::Options caller_options;
    caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
    PeerConnectionFactory::Options callee_options;
    callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
    ASSERT_TRUE(
        CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
    ConnectFakeSignaling();
    // Do normal offer/answer and wait for some frames to be received in each
    // direction.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // The three tests below verify that "enable_aes128_sha1_32_crypto_cipher"
  // works as expected; the cipher should only be used if enabled by both sides.
  TEST_P(PeerConnectionIntegrationTest,
         Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
    PeerConnectionFactory::Options caller_options;
    caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
    PeerConnectionFactory::Options callee_options;
    callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
        false;
    int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
    TestNegotiatedCipherSuite(caller_options, callee_options,
                              expected_cipher_suite);
  }
  
  TEST_P(PeerConnectionIntegrationTest,
         Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
    PeerConnectionFactory::Options caller_options;
    caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
        false;
    PeerConnectionFactory::Options callee_options;
    callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
    int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
    TestNegotiatedCipherSuite(caller_options, callee_options,
                              expected_cipher_suite);
  }
  
  TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
    PeerConnectionFactory::Options caller_options;
    caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
    PeerConnectionFactory::Options callee_options;
    callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
    int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32;
    TestNegotiatedCipherSuite(caller_options, callee_options,
                              expected_cipher_suite);
  }
  
  // Test that a non-GCM cipher is used if both sides only support non-GCM.
  TEST_P(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
    bool local_gcm_enabled = false;
    bool remote_gcm_enabled = false;
    int expected_cipher_suite = kDefaultSrtpCryptoSuite;
    TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                      expected_cipher_suite);
  }
  
  // Test that a GCM cipher is used if both ends support it.
  TEST_P(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) {
    bool local_gcm_enabled = true;
    bool remote_gcm_enabled = true;
    int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm;
    TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                      expected_cipher_suite);
  }
  
  // Test that GCM isn't used if only the offerer supports it.
  TEST_P(PeerConnectionIntegrationTest,
         NonGcmCipherUsedWhenOnlyCallerSupportsGcm) {
    bool local_gcm_enabled = true;
    bool remote_gcm_enabled = false;
    int expected_cipher_suite = kDefaultSrtpCryptoSuite;
    TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                      expected_cipher_suite);
  }
  
  // Test that GCM isn't used if only the answerer supports it.
  TEST_P(PeerConnectionIntegrationTest,
         NonGcmCipherUsedWhenOnlyCalleeSupportsGcm) {
    bool local_gcm_enabled = false;
    bool remote_gcm_enabled = true;
    int expected_cipher_suite = kDefaultSrtpCryptoSuite;
    TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                      expected_cipher_suite);
  }
  
  // Verify that media can be transmitted end-to-end when GCM crypto suites are
  // enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported,
  // only verify that a GCM cipher is negotiated, and not necessarily that SRTP
  // works with it.
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
    PeerConnectionFactory::Options gcm_options;
    gcm_options.crypto_options.srtp.enable_gcm_crypto_suites = true;
    ASSERT_TRUE(
        CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options));
    ConnectFakeSignaling();
    // Do normal offer/answer and wait for some frames to be received in each
    // direction.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // This test sets up a call between two parties with audio, video and an RTP
  // data channel.
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.enable_rtp_data_channel = true;
    rtc_config.enable_dtls_srtp = false;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
    ConnectFakeSignaling();
    // Expect that data channel created on caller side will show up for callee as
    // well.
    caller()->CreateDataChannel();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Ensure the existence of the RTP data channel didn't impede audio/video.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_NE(nullptr, callee()->data_channel());
    EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    // Ensure data can be sent in both directions.
    std::string data = "hello world";
    SendRtpDataWithRetries(caller()->data_channel(), data, 5);
    EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                   kDefaultTimeout);
    SendRtpDataWithRetries(callee()->data_channel(), data, 5);
    EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                   kDefaultTimeout);
  }
  
  // Ensure that an RTP data channel is signaled as closed for the caller when
  // the callee rejects it in a subsequent offer.
  TEST_P(PeerConnectionIntegrationTest,
         RtpDataChannelSignaledClosedInCalleeOffer) {
    // Same procedure as above test.
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.enable_rtp_data_channel = true;
    rtc_config.enable_dtls_srtp = false;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
    ConnectFakeSignaling();
    caller()->CreateDataChannel();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_NE(nullptr, callee()->data_channel());
    ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    // Close the data channel on the callee, and do an updated offer/answer.
    callee()->data_channel()->Close();
    callee()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    EXPECT_FALSE(caller()->data_observer()->IsOpen());
    EXPECT_FALSE(callee()->data_observer()->IsOpen());
  }
  
  // Tests that data is buffered in an RTP data channel until an observer is
  // registered for it.
  //
  // NOTE: RTP data channels can receive data before the underlying
  // transport has detected that a channel is writable and thus data can be
  // received before the data channel state changes to open. That is hard to test
  // but the same buffering is expected to be used in that case.
  TEST_P(PeerConnectionIntegrationTest,
         DataBufferedUntilRtpDataChannelObserverRegistered) {
    // Use fake clock and simulated network delay so that we predictably can wait
    // until an SCTP message has been delivered without "sleep()"ing.
    rtc::ScopedFakeClock fake_clock;
    // Some things use a time of "0" as a special value, so we need to start out
    // the fake clock at a nonzero time.
    // TODO(deadbeef): Fix this.
    fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
    virtual_socket_server()->set_delay_mean(5);  // 5 ms per hop.
    virtual_socket_server()->UpdateDelayDistribution();
  
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.enable_rtp_data_channel = true;
    rtc_config.enable_dtls_srtp = false;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
    ConnectFakeSignaling();
    caller()->CreateDataChannel();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE(caller()->data_channel() != nullptr);
    ASSERT_TRUE_SIMULATED_WAIT(callee()->data_channel() != nullptr,
                               kDefaultTimeout, fake_clock);
    ASSERT_TRUE_SIMULATED_WAIT(caller()->data_observer()->IsOpen(),
                               kDefaultTimeout, fake_clock);
    ASSERT_EQ_SIMULATED_WAIT(DataChannelInterface::kOpen,
                             callee()->data_channel()->state(), kDefaultTimeout,
                             fake_clock);
  
    // Unregister the observer which is normally automatically registered.
    callee()->data_channel()->UnregisterObserver();
    // Send data and advance fake clock until it should have been received.
    std::string data = "hello world";
    caller()->data_channel()->Send(DataBuffer(data));
    SIMULATED_WAIT(false, 50, fake_clock);
  
    // Attach data channel and expect data to be received immediately. Note that
    // EXPECT_EQ_WAIT is used, such that the simulated clock is not advanced any
    // further, but data can be received even if the callback is asynchronous.
    MockDataChannelObserver new_observer(callee()->data_channel());
    EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout,
                             fake_clock);
    // Closing the PeerConnections destroys the ports before the ScopedFakeClock.
    // If this is not done a DCHECK can be hit in ports.cc, because a large
    // negative number is calculated for the rtt due to the global clock changing.
    caller()->pc()->Close();
    callee()->pc()->Close();
  }
  
  // This test sets up a call between two parties with audio, video and but only
  // the caller client supports RTP data channels.
  TEST_P(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
    PeerConnectionInterface::RTCConfiguration rtc_config_1;
    rtc_config_1.enable_rtp_data_channel = true;
    // Must disable DTLS to make negotiation succeed.
    rtc_config_1.enable_dtls_srtp = false;
    PeerConnectionInterface::RTCConfiguration rtc_config_2;
    rtc_config_2.enable_dtls_srtp = false;
    rtc_config_2.enable_dtls_srtp = false;
    ASSERT_TRUE(
        CreatePeerConnectionWrappersWithConfig(rtc_config_1, rtc_config_2));
    ConnectFakeSignaling();
    caller()->CreateDataChannel();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // The caller should still have a data channel, but it should be closed, and
    // one should ever have been created for the callee.
    EXPECT_TRUE(caller()->data_channel() != nullptr);
    EXPECT_FALSE(caller()->data_observer()->IsOpen());
    EXPECT_EQ(nullptr, callee()->data_channel());
  }
  
  // This test sets up a call between two parties with audio, and video. When
  // audio and video is setup and flowing, an RTP data channel is negotiated.
  TEST_P(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.enable_rtp_data_channel = true;
    rtc_config.enable_dtls_srtp = false;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
    ConnectFakeSignaling();
    // Do initial offer/answer with audio/video.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Create data channel and do new offer and answer.
    caller()->CreateDataChannel();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_NE(nullptr, callee()->data_channel());
    EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
    // Ensure data can be sent in both directions.
    std::string data = "hello world";
    SendRtpDataWithRetries(caller()->data_channel(), data, 5);
    EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                   kDefaultTimeout);
    SendRtpDataWithRetries(callee()->data_channel(), data, 5);
    EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                   kDefaultTimeout);
  }
  
  #ifdef HAVE_SCTP
  
  // This test sets up a call between two parties with audio, video and an SCTP
  // data channel.
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Expect that data channel created on caller side will show up for callee as
    // well.
    caller()->CreateDataChannel();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Ensure the existence of the SCTP data channel didn't impede audio/video.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
    // Caller data channel should already exist (it created one). Callee data
    // channel may not exist yet, since negotiation happens in-band, not in SDP.
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    // Ensure data can be sent in both directions.
    std::string data = "hello world";
    caller()->data_channel()->Send(DataBuffer(data));
    EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                   kDefaultTimeout);
    callee()->data_channel()->Send(DataBuffer(data));
    EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                   kDefaultTimeout);
  }
  
  // Ensure that when the callee closes an SCTP data channel, the closing
  // procedure results in the data channel being closed for the caller as well.
  TEST_P(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) {
    // Same procedure as above test.
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->CreateDataChannel();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    // Close the data channel on the callee side, and wait for it to reach the
    // "closed" state on both sides.
    callee()->data_channel()->Close();
    EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
  }
  
  TEST_P(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    webrtc::DataChannelInit init;
    init.id = 53;
    init.maxRetransmits = 52;
    caller()->CreateDataChannel("data-channel", &init);
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_EQ(init.id, callee()->data_channel()->id());
    EXPECT_EQ("data-channel", callee()->data_channel()->label());
    EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
    EXPECT_FALSE(callee()->data_channel()->negotiated());
  }
  
  // Test usrsctp's ability to process unordered data stream, where data actually
  // arrives out of order using simulated delays. Previously there have been some
  // bugs in this area.
  TEST_P(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
    // Introduce random network delays.
    // Otherwise it's not a true "unordered" test.
    virtual_socket_server()->set_delay_mean(20);
    virtual_socket_server()->set_delay_stddev(5);
    virtual_socket_server()->UpdateDelayDistribution();
    // Normal procedure, but with unordered data channel config.
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    webrtc::DataChannelInit init;
    init.ordered = false;
    caller()->CreateDataChannel(&init);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    static constexpr int kNumMessages = 100;
    // Deliberately chosen to be larger than the MTU so messages get fragmented.
    static constexpr size_t kMaxMessageSize = 4096;
    // Create and send random messages.
    std::vector<std::string> sent_messages;
    for (int i = 0; i < kNumMessages; ++i) {
      size_t length =
          (rand() % kMaxMessageSize) + 1;  // NOLINT (rand_r instead of rand)
      std::string message;
      ASSERT_TRUE(rtc::CreateRandomString(length, &message));
      caller()->data_channel()->Send(DataBuffer(message));
      callee()->data_channel()->Send(DataBuffer(message));
      sent_messages.push_back(message);
    }
  
    // Wait for all messages to be received.
    EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
                   caller()->data_observer()->received_message_count(),
                   kDefaultTimeout);
    EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
                   callee()->data_observer()->received_message_count(),
                   kDefaultTimeout);
  
    // Sort and compare to make sure none of the messages were corrupted.
    std::vector<std::string> caller_received_messages =
        caller()->data_observer()->messages();
    std::vector<std::string> callee_received_messages =
        callee()->data_observer()->messages();
    absl::c_sort(sent_messages);
    absl::c_sort(caller_received_messages);
    absl::c_sort(callee_received_messages);
    EXPECT_EQ(sent_messages, caller_received_messages);
    EXPECT_EQ(sent_messages, callee_received_messages);
  }
  
  // This test sets up a call between two parties with audio, and video. When
  // audio and video are setup and flowing, an SCTP data channel is negotiated.
  TEST_P(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Do initial offer/answer with audio/video.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Create data channel and do new offer and answer.
    caller()->CreateDataChannel();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Caller data channel should already exist (it created one). Callee data
    // channel may not exist yet, since negotiation happens in-band, not in SDP.
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
    // Ensure data can be sent in both directions.
    std::string data = "hello world";
    caller()->data_channel()->Send(DataBuffer(data));
    EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                   kDefaultTimeout);
    callee()->data_channel()->Send(DataBuffer(data));
    EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                   kDefaultTimeout);
  }
  
  // Set up a connection initially just using SCTP data channels, later upgrading
  // to audio/video, ensuring frames are received end-to-end. Effectively the
  // inverse of the test above.
  // This was broken in M57; see https://crbug.com/711243
  TEST_P(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Do initial offer/answer with just data channel.
    caller()->CreateDataChannel();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Wait until data can be sent over the data channel.
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    // Do subsequent offer/answer with two-way audio and video. Audio and video
    // should end up bundled on the DTLS/ICE transport already used for data.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) {
    cricket::DataContentDescription* dcd_offer =
        GetFirstDataContentDescription(desc);
    ASSERT_TRUE(dcd_offer);
    dcd_offer->set_use_sctpmap(false);
    dcd_offer->set_protocol("UDP/DTLS/SCTP");
  }
  
  // Test that the data channel works when a spec-compliant SCTP m= section is
  // offered (using "a=sctp-port" instead of "a=sctpmap", and using
  // "UDP/DTLS/SCTP" as the protocol).
  TEST_P(PeerConnectionIntegrationTest,
         DataChannelWorksWhenSpecCompliantSctpOfferReceived) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->CreateDataChannel();
    caller()->SetGeneratedSdpMunger(MakeSpecCompliantSctpOffer);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    // Ensure data can be sent in both directions.
    std::string data = "hello world";
    caller()->data_channel()->Send(DataBuffer(data));
    EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                   kDefaultTimeout);
    callee()->data_channel()->Send(DataBuffer(data));
    EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                   kDefaultTimeout);
  }
  
  #endif  // HAVE_SCTP
  
  // This test sets up a call between two parties with a media transport data
  // channel.
  TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelEndToEnd) {
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.use_media_transport_for_data_channels = true;
    rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
        rtc_config, rtc_config, loopback_media_transports()->first_factory(),
        loopback_media_transports()->second_factory()));
    ConnectFakeSignaling();
  
    // Expect that data channel created on caller side will show up for callee as
    // well.
    caller()->CreateDataChannel();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Ensure that the media transport is ready.
    loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
    loopback_media_transports()->FlushAsyncInvokes();
  
    // Caller data channel should already exist (it created one). Callee data
    // channel may not exist yet, since negotiation happens in-band, not in SDP.
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    // Ensure data can be sent in both directions.
    std::string data = "hello world";
    caller()->data_channel()->Send(DataBuffer(data));
    EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                   kDefaultTimeout);
    callee()->data_channel()->Send(DataBuffer(data));
    EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                   kDefaultTimeout);
  }
  
  // Ensure that when the callee closes a media transport data channel, the
  // closing procedure results in the data channel being closed for the caller
  // as well.
  TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelCalleeCloses) {
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.use_media_transport_for_data_channels = true;
    rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
        rtc_config, rtc_config, loopback_media_transports()->first_factory(),
        loopback_media_transports()->second_factory()));
    ConnectFakeSignaling();
  
    // Create a data channel on the caller and signal it to the callee.
    caller()->CreateDataChannel();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Ensure that the media transport is ready.
    loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
    loopback_media_transports()->FlushAsyncInvokes();
  
    // Data channels exist and open on both ends of the connection.
    ASSERT_NE(nullptr, caller()->data_channel());
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
  
    // Close the data channel on the callee side, and wait for it to reach the
    // "closed" state on both sides.
    callee()->data_channel()->Close();
    EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
  }
  
  TEST_P(PeerConnectionIntegrationTest,
         MediaTransportDataChannelConfigSentToOtherSide) {
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.use_media_transport_for_data_channels = true;
    rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
        rtc_config, rtc_config, loopback_media_transports()->first_factory(),
        loopback_media_transports()->second_factory()));
    ConnectFakeSignaling();
  
    // Create a data channel with a non-default configuration and signal it to the
    // callee.
    webrtc::DataChannelInit init;
    init.id = 53;
    init.maxRetransmits = 52;
    caller()->CreateDataChannel("data-channel", &init);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Ensure that the media transport is ready.
    loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
    loopback_media_transports()->FlushAsyncInvokes();
  
    // Ensure that the data channel exists on the callee with the correct
    // configuration.
    ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
    EXPECT_EQ(init.id, callee()->data_channel()->id());
    EXPECT_EQ("data-channel", callee()->data_channel()->label());
    EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
    EXPECT_FALSE(callee()->data_channel()->negotiated());
  }
  
  TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalAudio) {
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.use_media_transport = true;
    rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
        rtc_config, rtc_config, loopback_media_transports()->first_factory(),
        loopback_media_transports()->second_factory()));
    ConnectFakeSignaling();
  
    caller()->AddAudioTrack();
    callee()->AddAudioTrack();
    // Start offer/answer exchange and wait for it to complete.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Ensure that the media transport is ready.
    loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
    loopback_media_transports()->FlushAsyncInvokes();
  
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  
    webrtc::MediaTransportPair::Stats first_stats =
        loopback_media_transports()->FirstStats();
    webrtc::MediaTransportPair::Stats second_stats =
        loopback_media_transports()->SecondStats();
  
    EXPECT_GT(first_stats.received_audio_frames, 0);
    EXPECT_GE(second_stats.sent_audio_frames, first_stats.received_audio_frames);
  
    EXPECT_GT(second_stats.received_audio_frames, 0);
    EXPECT_GE(first_stats.sent_audio_frames, second_stats.received_audio_frames);
  }
  
  TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalVideo) {
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.use_media_transport = true;
    rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
        rtc_config, rtc_config, loopback_media_transports()->first_factory(),
        loopback_media_transports()->second_factory()));
    ConnectFakeSignaling();
  
    caller()->AddVideoTrack();
    callee()->AddVideoTrack();
    // Start offer/answer exchange and wait for it to complete.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Ensure that the media transport is ready.
    loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
    loopback_media_transports()->FlushAsyncInvokes();
  
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  
    webrtc::MediaTransportPair::Stats first_stats =
        loopback_media_transports()->FirstStats();
    webrtc::MediaTransportPair::Stats second_stats =
        loopback_media_transports()->SecondStats();
  
    EXPECT_GT(first_stats.received_video_frames, 0);
    EXPECT_GE(second_stats.sent_video_frames, first_stats.received_video_frames);
  
    EXPECT_GT(second_stats.received_video_frames, 0);
    EXPECT_GE(first_stats.sent_video_frames, second_stats.received_video_frames);
  }
  
  TEST_P(PeerConnectionIntegrationTest,
         MediaTransportDataChannelUsesRtpBidirectionalVideo) {
    PeerConnectionInterface::RTCConfiguration rtc_config;
    rtc_config.use_media_transport = false;
    rtc_config.use_media_transport_for_data_channels = true;
    rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
        rtc_config, rtc_config, loopback_media_transports()->first_factory(),
        loopback_media_transports()->second_factory()));
    ConnectFakeSignaling();
  
    caller()->AddVideoTrack();
    callee()->AddVideoTrack();
    // Start offer/answer exchange and wait for it to complete.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Test that the ICE connection and gathering states eventually reach
  // "complete".
  TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Do normal offer/answer.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
                   caller()->ice_gathering_state(), kMaxWaitForFramesMs);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
                   callee()->ice_gathering_state(), kMaxWaitForFramesMs);
    // After the best candidate pair is selected and all candidates are signaled,
    // the ICE connection state should reach "complete".
    // TODO(deadbeef): Currently, the ICE "controlled" agent (the
    // answerer/"callee" by default) only reaches "connected". When this is
    // fixed, this test should be updated.
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                   caller()->ice_connection_state(), kDefaultTimeout);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                   callee()->ice_connection_state(), kDefaultTimeout);
  }
  
  constexpr int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
                                  cricket::PORTALLOCATOR_DISABLE_RELAY |
                                  cricket::PORTALLOCATOR_DISABLE_TCP;
  
  // Use a mock resolver to resolve the hostname back to the original IP on both
  // sides and check that the ICE connection connects.
  TEST_P(PeerConnectionIntegrationTest,
         IceStatesReachCompletionWithRemoteHostname) {
    auto caller_resolver_factory =
        absl::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
    auto callee_resolver_factory =
        absl::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
    NiceMock<rtc::MockAsyncResolver> callee_async_resolver;
    NiceMock<rtc::MockAsyncResolver> caller_async_resolver;
  
    // This also verifies that the injected AsyncResolverFactory is used by
    // P2PTransportChannel.
    EXPECT_CALL(*caller_resolver_factory, Create())
        .WillOnce(Return(&caller_async_resolver));
    webrtc::PeerConnectionDependencies caller_deps(nullptr);
    caller_deps.async_resolver_factory = std::move(caller_resolver_factory);
  
    EXPECT_CALL(*callee_resolver_factory, Create())
        .WillOnce(Return(&callee_async_resolver));
    webrtc::PeerConnectionDependencies callee_deps(nullptr);
    callee_deps.async_resolver_factory = std::move(callee_resolver_factory);
  
    PeerConnectionInterface::RTCConfiguration config;
    config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
    config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
  
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
        config, std::move(caller_deps), config, std::move(callee_deps)));
  
    caller()->SetRemoteAsyncResolver(&callee_async_resolver);
    callee()->SetRemoteAsyncResolver(&caller_async_resolver);
  
    // Enable hostname candidates with mDNS names.
    caller()->network()->CreateMdnsResponder(network_thread());
    callee()->network()->CreateMdnsResponder(network_thread());
  
    SetPortAllocatorFlags(kOnlyLocalPorts, kOnlyLocalPorts);
  
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                   caller()->ice_connection_state(), kDefaultTimeout);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                   callee()->ice_connection_state(), kDefaultTimeout);
  
    EXPECT_EQ(1, webrtc::metrics::NumEvents(
                     "WebRTC.PeerConnection.CandidatePairType_UDP",
                     webrtc::kIceCandidatePairHostNameHostName));
  }
  
  // Test that firewalling the ICE connection causes the clients to identify the
  // disconnected state and then removing the firewall causes them to reconnect.
  class PeerConnectionIntegrationIceStatesTest
      : public PeerConnectionIntegrationBaseTest,
        public ::testing::WithParamInterface<
            std::tuple<SdpSemantics, std::tuple<std::string, uint32_t>>> {
   protected:
    PeerConnectionIntegrationIceStatesTest()
        : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())) {
      port_allocator_flags_ = std::get<1>(std::get<1>(GetParam()));
    }
  
    void StartStunServer(const SocketAddress& server_address) {
      stun_server_.reset(
          cricket::TestStunServer::Create(network_thread(), server_address));
    }
  
    bool TestIPv6() {
      return (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6);
    }
  
    void SetPortAllocatorFlags() {
      PeerConnectionIntegrationBaseTest::SetPortAllocatorFlags(
          port_allocator_flags_, port_allocator_flags_);
    }
  
    std::vector<SocketAddress> CallerAddresses() {
      std::vector<SocketAddress> addresses;
      addresses.push_back(SocketAddress("1.1.1.1", 0));
      if (TestIPv6()) {
        addresses.push_back(SocketAddress("1111:0:a:b:c:d:e:f", 0));
      }
      return addresses;
    }
  
    std::vector<SocketAddress> CalleeAddresses() {
      std::vector<SocketAddress> addresses;
      addresses.push_back(SocketAddress("2.2.2.2", 0));
      if (TestIPv6()) {
        addresses.push_back(SocketAddress("2222:0:a:b:c:d:e:f", 0));
      }
      return addresses;
    }
  
    void SetUpNetworkInterfaces() {
      // Remove the default interfaces added by the test infrastructure.
      caller()->network()->RemoveInterface(kDefaultLocalAddress);
      callee()->network()->RemoveInterface(kDefaultLocalAddress);
  
      // Add network addresses for test.
      for (const auto& caller_address : CallerAddresses()) {
        caller()->network()->AddInterface(caller_address);
      }
      for (const auto& callee_address : CalleeAddresses()) {
        callee()->network()->AddInterface(callee_address);
      }
    }
  
   private:
    uint32_t port_allocator_flags_;
    std::unique_ptr<cricket::TestStunServer> stun_server_;
  };
  
  // Tests that the PeerConnection goes through all the ICE gathering/connection
  // states over the duration of the call. This includes Disconnected and Failed
  // states, induced by putting a firewall between the peers and waiting for them
  // to time out.
  TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyIceStates) {
    // TODO(bugs.webrtc.org/8295): When using a ScopedFakeClock, this test will
    // sometimes hit a DCHECK in platform_thread.cc about the PacerThread being
    // too busy. For now, revert to running without a fake clock.
  
    const SocketAddress kStunServerAddress =
        SocketAddress("99.99.99.1", cricket::STUN_SERVER_PORT);
    StartStunServer(kStunServerAddress);
  
    PeerConnectionInterface::RTCConfiguration config;
    PeerConnectionInterface::IceServer ice_stun_server;
    ice_stun_server.urls.push_back(
        "stun:" + kStunServerAddress.HostAsURIString() + ":" +
        kStunServerAddress.PortAsString());
    config.servers.push_back(ice_stun_server);
  
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
    ConnectFakeSignaling();
    SetPortAllocatorFlags();
    SetUpNetworkInterfaces();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
  
    // Initial state before anything happens.
    ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew,
              caller()->ice_gathering_state());
    ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
              caller()->ice_connection_state());
    ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
              caller()->standardized_ice_connection_state());
  
    // Start the call by creating the offer, setting it as the local description,
    // then sending it to the peer who will respond with an answer. This happens
    // asynchronously so that we can watch the states as it runs in the
    // background.
    caller()->CreateAndSetAndSignalOffer();
  
    ASSERT_EQ(PeerConnectionInterface::kIceConnectionCompleted,
              caller()->ice_connection_state());
    ASSERT_EQ(PeerConnectionInterface::kIceConnectionConnected,
              caller()->standardized_ice_connection_state());
  
    // Verify that the observer was notified of the intermediate transitions.
    EXPECT_THAT(caller()->ice_connection_state_history(),
                ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
                            PeerConnectionInterface::kIceConnectionConnected,
                            PeerConnectionInterface::kIceConnectionCompleted));
    EXPECT_THAT(
        caller()->peer_connection_state_history(),
        ElementsAre(PeerConnectionInterface::PeerConnectionState::kConnecting,
                    PeerConnectionInterface::PeerConnectionState::kConnected));
    EXPECT_THAT(caller()->ice_gathering_state_history(),
                ElementsAre(PeerConnectionInterface::kIceGatheringGathering,
                            PeerConnectionInterface::kIceGatheringComplete));
  
    // Block connections to/from the caller and wait for ICE to become
    // disconnected.
    for (const auto& caller_address : CallerAddresses()) {
      firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
    }
    RTC_LOG(LS_INFO) << "Firewall rules applied";
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
                   caller()->ice_connection_state(), kDefaultTimeout);
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
                   caller()->standardized_ice_connection_state(),
                   kDefaultTimeout);
  
    // Let ICE re-establish by removing the firewall rules.
    firewall()->ClearRules();
    RTC_LOG(LS_INFO) << "Firewall rules cleared";
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                   caller()->ice_connection_state(), kDefaultTimeout);
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                   caller()->standardized_ice_connection_state(),
                   kDefaultTimeout);
  
    // According to RFC7675, if there is no response within 30 seconds then the
    // peer should consider the other side to have rejected the connection. This
    // is signaled by the state transitioning to "failed".
    constexpr int kConsentTimeout = 30000;
    for (const auto& caller_address : CallerAddresses()) {
      firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
    }
    RTC_LOG(LS_INFO) << "Firewall rules applied again";
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
                   caller()->ice_connection_state(), kConsentTimeout);
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
                   caller()->standardized_ice_connection_state(),
                   kConsentTimeout);
  }
  
  // Tests that the best connection is set to the appropriate IPv4/IPv6 connection
  // and that the statistics in the metric observers are updated correctly.
  TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    SetPortAllocatorFlags();
    SetUpNetworkInterfaces();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
  
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // TODO(bugs.webrtc.org/9456): Fix it.
    const int num_best_ipv4 = webrtc::metrics::NumEvents(
        "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
    const int num_best_ipv6 = webrtc::metrics::NumEvents(
        "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
    if (TestIPv6()) {
      // When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
      // connection.
      EXPECT_EQ(0, num_best_ipv4);
      EXPECT_EQ(1, num_best_ipv6);
    } else {
      EXPECT_EQ(1, num_best_ipv4);
      EXPECT_EQ(0, num_best_ipv6);
    }
  
    EXPECT_EQ(0, webrtc::metrics::NumEvents(
                     "WebRTC.PeerConnection.CandidatePairType_UDP",
                     webrtc::kIceCandidatePairHostHost));
    EXPECT_EQ(1, webrtc::metrics::NumEvents(
                     "WebRTC.PeerConnection.CandidatePairType_UDP",
                     webrtc::kIceCandidatePairHostPublicHostPublic));
  }
  
  constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
                                        cricket::PORTALLOCATOR_DISABLE_STUN |
                                        cricket::PORTALLOCATOR_DISABLE_RELAY;
  constexpr uint32_t kFlagsIPv6NoStun =
      cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_STUN |
      cricket::PORTALLOCATOR_ENABLE_IPV6 | cricket::PORTALLOCATOR_DISABLE_RELAY;
  constexpr uint32_t kFlagsIPv4Stun =
      cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_RELAY;
  
  INSTANTIATE_TEST_SUITE_P(
      PeerConnectionIntegrationTest,
      PeerConnectionIntegrationIceStatesTest,
      Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
              Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
                     std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
                     std::make_pair("IPv4 with STUN", kFlagsIPv4Stun))));
  
  // This test sets up a call between two parties with audio and video.
  // During the call, the caller restarts ICE and the test verifies that
  // new ICE candidates are generated and audio and video still can flow, and the
  // ICE state reaches completed again.
  TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Do normal offer/answer and wait for ICE to complete.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                   caller()->ice_connection_state(), kMaxWaitForFramesMs);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                   callee()->ice_connection_state(), kMaxWaitForFramesMs);
  
    // To verify that the ICE restart actually occurs, get
    // ufrag/password/candidates before and after restart.
    // Create an SDP string of the first audio candidate for both clients.
    const webrtc::IceCandidateCollection* audio_candidates_caller =
        caller()->pc()->local_description()->candidates(0);
    const webrtc::IceCandidateCollection* audio_candidates_callee =
        callee()->pc()->local_description()->candidates(0);
    ASSERT_GT(audio_candidates_caller->count(), 0u);
    ASSERT_GT(audio_candidates_callee->count(), 0u);
    std::string caller_candidate_pre_restart;
    ASSERT_TRUE(
        audio_candidates_caller->at(0)->ToString(&caller_candidate_pre_restart));
    std::string callee_candidate_pre_restart;
    ASSERT_TRUE(
        audio_candidates_callee->at(0)->ToString(&callee_candidate_pre_restart));
    const cricket::SessionDescription* desc =
        caller()->pc()->local_description()->description();
    std::string caller_ufrag_pre_restart =
        desc->transport_infos()[0].description.ice_ufrag;
    desc = callee()->pc()->local_description()->description();
    std::string callee_ufrag_pre_restart =
        desc->transport_infos()[0].description.ice_ufrag;
  
    // Have the caller initiate an ICE restart.
    caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                   caller()->ice_connection_state(), kMaxWaitForFramesMs);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                   callee()->ice_connection_state(), kMaxWaitForFramesMs);
  
    // Grab the ufrags/candidates again.
    audio_candidates_caller = caller()->pc()->local_description()->candidates(0);
    audio_candidates_callee = callee()->pc()->local_description()->candidates(0);
    ASSERT_GT(audio_candidates_caller->count(), 0u);
    ASSERT_GT(audio_candidates_callee->count(), 0u);
    std::string caller_candidate_post_restart;
    ASSERT_TRUE(
        audio_candidates_caller->at(0)->ToString(&caller_candidate_post_restart));
    std::string callee_candidate_post_restart;
    ASSERT_TRUE(
        audio_candidates_callee->at(0)->ToString(&callee_candidate_post_restart));
    desc = caller()->pc()->local_description()->description();
    std::string caller_ufrag_post_restart =
        desc->transport_infos()[0].description.ice_ufrag;
    desc = callee()->pc()->local_description()->description();
    std::string callee_ufrag_post_restart =
        desc->transport_infos()[0].description.ice_ufrag;
    // Sanity check that an ICE restart was actually negotiated in SDP.
    ASSERT_NE(caller_candidate_pre_restart, caller_candidate_post_restart);
    ASSERT_NE(callee_candidate_pre_restart, callee_candidate_post_restart);
    ASSERT_NE(caller_ufrag_pre_restart, caller_ufrag_post_restart);
    ASSERT_NE(callee_ufrag_pre_restart, callee_ufrag_post_restart);
  
    // Ensure that additional frames are received after the ICE restart.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Verify that audio/video can be received end-to-end when ICE renomination is
  // enabled.
  TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
    PeerConnectionInterface::RTCConfiguration config;
    config.enable_ice_renomination = true;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
    ConnectFakeSignaling();
    // Do normal offer/answer and wait for some frames to be received in each
    // direction.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Sanity check that ICE renomination was actually negotiated.
    const cricket::SessionDescription* desc =
        caller()->pc()->local_description()->description();
    for (const cricket::TransportInfo& info : desc->transport_infos()) {
      ASSERT_THAT(info.description.transport_options, Contains("renomination"));
    }
    desc = callee()->pc()->local_description()->description();
    for (const cricket::TransportInfo& info : desc->transport_infos()) {
      ASSERT_THAT(info.description.transport_options, Contains("renomination"));
    }
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // With a max bundle policy and RTCP muxing, adding a new media description to
  // the connection should not affect ICE at all because the new media will use
  // the existing connection.
  TEST_P(PeerConnectionIntegrationTest,
         AddMediaToConnectedBundleDoesNotRestartIce) {
    PeerConnectionInterface::RTCConfiguration config;
    config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
    config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(
        config, PeerConnectionInterface::RTCConfiguration()));
    ConnectFakeSignaling();
  
    caller()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                   caller()->ice_connection_state(), kDefaultTimeout);
  
    caller()->clear_ice_connection_state_history();
  
    caller()->AddVideoTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    EXPECT_EQ(0u, caller()->ice_connection_state_history().size());
  }
  
  // This test sets up a call between two parties with audio and video. It then
  // renegotiates setting the video m-line to "port 0", then later renegotiates
  // again, enabling video.
  TEST_P(PeerConnectionIntegrationTest,
         VideoFlowsAfterMediaSectionIsRejectedAndRecycled) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    // Do initial negotiation, only sending media from the caller. Will result in
    // video and audio recvonly "m=" sections.
    caller()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Negotiate again, disabling the video "m=" section (the callee will set the
    // port to 0 due to offer_to_receive_video = 0).
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      PeerConnectionInterface::RTCOfferAnswerOptions options;
      options.offer_to_receive_video = 0;
      callee()->SetOfferAnswerOptions(options);
    } else {
      callee()->SetRemoteOfferHandler([this] {
        callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
      });
    }
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Sanity check that video "m=" section was actually rejected.
    const ContentInfo* answer_video_content = cricket::GetFirstVideoContent(
        callee()->pc()->local_description()->description());
    ASSERT_NE(nullptr, answer_video_content);
    ASSERT_TRUE(answer_video_content->rejected);
  
    // Enable video and do negotiation again, making sure video is received
    // end-to-end, also adding media stream to callee.
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      PeerConnectionInterface::RTCOfferAnswerOptions options;
      options.offer_to_receive_video = 1;
      callee()->SetOfferAnswerOptions(options);
    } else {
      // The caller's transceiver is stopped, so we need to add another track.
      auto caller_transceiver =
          caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
      EXPECT_TRUE(caller_transceiver->stopped());
      caller()->AddVideoTrack();
    }
    callee()->AddVideoTrack();
    callee()->SetRemoteOfferHandler(nullptr);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Verify the caller receives frames from the newly added stream, and the
    // callee receives additional frames from the re-enabled video m= section.
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio();
    media_expectations.ExpectBidirectionalVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // 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(PeerConnectionIntegrationTestPlanB,
         MediaFlowsAfterEarlyWarmupWithCreateSender) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    auto caller_audio_sender =
        caller()->pc()->CreateSender("audio", "caller_stream");
    auto caller_video_sender =
        caller()->pc()->CreateSender("video", "caller_stream");
    auto callee_audio_sender =
        callee()->pc()->CreateSender("audio", "callee_stream");
    auto callee_video_sender =
        callee()->pc()->CreateSender("video", "callee_stream");
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
    // Wait for ICE to complete, without any tracks being set.
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                   caller()->ice_connection_state(), kMaxWaitForFramesMs);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                   callee()->ice_connection_state(), kMaxWaitForFramesMs);
    // Now set the tracks, and expect frames to immediately start flowing.
    EXPECT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
    EXPECT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
    EXPECT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
    EXPECT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // This tests that if we negotiate after calling AddTransceiver but before we
  // have a track, then set a track later, frames from the newly-set tracks are
  // received end-to-end.
  TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
         MediaFlowsAfterEarlyWarmupWithAddTransceiver) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    auto audio_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
    ASSERT_EQ(RTCErrorType::NONE, audio_result.error().type());
    auto caller_audio_sender = audio_result.MoveValue()->sender();
    auto video_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
    ASSERT_EQ(RTCErrorType::NONE, video_result.error().type());
    auto caller_video_sender = video_result.MoveValue()->sender();
    callee()->SetRemoteOfferHandler([this] {
      ASSERT_EQ(2u, callee()->pc()->GetTransceivers().size());
      callee()->pc()->GetTransceivers()[0]->SetDirection(
          RtpTransceiverDirection::kSendRecv);
      callee()->pc()->GetTransceivers()[1]->SetDirection(
          RtpTransceiverDirection::kSendRecv);
    });
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
    // Wait for ICE to complete, without any tracks being set.
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                   caller()->ice_connection_state(), kMaxWaitForFramesMs);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                   callee()->ice_connection_state(), kMaxWaitForFramesMs);
    // Now set the tracks, and expect frames to immediately start flowing.
    auto callee_audio_sender = callee()->pc()->GetSenders()[0];
    auto callee_video_sender = callee()->pc()->GetSenders()[1];
    ASSERT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
    ASSERT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
    ASSERT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
    ASSERT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // This test verifies that a remote video track can be added via AddStream,
  // and sent end-to-end. For this particular test, it's simply echoed back
  // from the caller to the callee, rather than being forwarded to a third
  // PeerConnection.
  TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    // Just send a video track from the caller.
    caller()->AddVideoTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
    ASSERT_EQ(1U, callee()->remote_streams()->count());
  
    // Echo the stream back, and do a new offer/anwer (initiated by callee this
    // time).
    callee()->pc()->AddStream(callee()->remote_streams()->at(0));
    callee()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
  
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Test that we achieve the expected end-to-end connection time, using a
  // fake clock and simulated latency on the media and signaling paths.
  // We use a TURN<->TURN connection because this is usually the quickest to
  // set up initially, especially when we're confident the connection will work
  // and can start sending media before we get a STUN response.
  //
  // With various optimizations enabled, here are the network delays we expect to
  // be on the critical path:
  // 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then
  //                       signaling answer (with DTLS fingerprint).
  // 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when
  //                  using TURN<->TURN pair, and DTLS exchange is 4 packets,
  //                  the first of which should have arrived before the answer.
  TEST_P(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
    rtc::ScopedFakeClock fake_clock;
    // Some things use a time of "0" as a special value, so we need to start out
    // the fake clock at a nonzero time.
    // TODO(deadbeef): Fix this.
    fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
  
    static constexpr int media_hop_delay_ms = 50;
    static constexpr int signaling_trip_delay_ms = 500;
    // For explanation of these values, see comment above.
    static constexpr int required_media_hops = 9;
    static constexpr int required_signaling_trips = 2;
    // For internal delays (such as posting an event asychronously).
    static constexpr int allowed_internal_delay_ms = 20;
    static constexpr int total_connection_time_ms =
        media_hop_delay_ms * required_media_hops +
        signaling_trip_delay_ms * required_signaling_trips +
        allowed_internal_delay_ms;
  
    static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
                                                                   3478};
    static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
                                                                   0};
    static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
                                                                   3478};
    static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
                                                                   0};
    cricket::TestTurnServer* turn_server_1 = CreateTurnServer(
        turn_server_1_internal_address, turn_server_1_external_address);
  
    cricket::TestTurnServer* turn_server_2 = CreateTurnServer(
        turn_server_2_internal_address, turn_server_2_external_address);
    // Bypass permission check on received packets so media can be sent before
    // the candidate is signaled.
    network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_1] {
      turn_server_1->set_enable_permission_checks(false);
    });
    network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_2] {
      turn_server_2->set_enable_permission_checks(false);
    });
  
    PeerConnectionInterface::RTCConfiguration client_1_config;
    webrtc::PeerConnectionInterface::IceServer ice_server_1;
    ice_server_1.urls.push_back("turn:88.88.88.0:3478");
    ice_server_1.username = "test";
    ice_server_1.password = "test";
    client_1_config.servers.push_back(ice_server_1);
    client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
    client_1_config.presume_writable_when_fully_relayed = true;
  
    PeerConnectionInterface::RTCConfiguration client_2_config;
    webrtc::PeerConnectionInterface::IceServer ice_server_2;
    ice_server_2.urls.push_back("turn:99.99.99.0:3478");
    ice_server_2.username = "test";
    ice_server_2.password = "test";
    client_2_config.servers.push_back(ice_server_2);
    client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
    client_2_config.presume_writable_when_fully_relayed = true;
  
    ASSERT_TRUE(
        CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
    // Set up the simulated delays.
    SetSignalingDelayMs(signaling_trip_delay_ms);
    ConnectFakeSignaling();
    virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
    virtual_socket_server()->UpdateDelayDistribution();
  
    // Set "offer to receive audio/video" without adding any tracks, so we just
    // set up ICE/DTLS with no media.
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 1;
    options.offer_to_receive_video = 1;
    caller()->SetOfferAnswerOptions(options);
    caller()->CreateAndSetAndSignalOffer();
    EXPECT_TRUE_SIMULATED_WAIT(DtlsConnected(), total_connection_time_ms,
                               fake_clock);
    // Closing the PeerConnections destroys the ports before the ScopedFakeClock.
    // If this is not done a DCHECK can be hit in ports.cc, because a large
    // negative number is calculated for the rtt due to the global clock changing.
    caller()->pc()->Close();
    callee()->pc()->Close();
  }
  
  // Verify that a TurnCustomizer passed in through RTCConfiguration
  // is actually used by the underlying TURN candidate pair.
  // Note that turnport_unittest.cc contains more detailed, lower-level tests.
  TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
    static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
                                                                   3478};
    static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
                                                                   0};
    static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
                                                                   3478};
    static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
                                                                   0};
    CreateTurnServer(turn_server_1_internal_address,
                     turn_server_1_external_address);
    CreateTurnServer(turn_server_2_internal_address,
                     turn_server_2_external_address);
  
    PeerConnectionInterface::RTCConfiguration client_1_config;
    webrtc::PeerConnectionInterface::IceServer ice_server_1;
    ice_server_1.urls.push_back("turn:88.88.88.0:3478");
    ice_server_1.username = "test";
    ice_server_1.password = "test";
    client_1_config.servers.push_back(ice_server_1);
    client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
    auto* customizer1 = CreateTurnCustomizer();
    client_1_config.turn_customizer = customizer1;
  
    PeerConnectionInterface::RTCConfiguration client_2_config;
    webrtc::PeerConnectionInterface::IceServer ice_server_2;
    ice_server_2.urls.push_back("turn:99.99.99.0:3478");
    ice_server_2.username = "test";
    ice_server_2.password = "test";
    client_2_config.servers.push_back(ice_server_2);
    client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
    auto* customizer2 = CreateTurnCustomizer();
    client_2_config.turn_customizer = customizer2;
  
    ASSERT_TRUE(
        CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
    ConnectFakeSignaling();
  
    // Set "offer to receive audio/video" without adding any tracks, so we just
    // set up ICE/DTLS with no media.
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 1;
    options.offer_to_receive_video = 1;
    caller()->SetOfferAnswerOptions(options);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
  
    ExpectTurnCustomizerCountersIncremented(customizer1);
    ExpectTurnCustomizerCountersIncremented(customizer2);
  }
  
  // Verifies that you can use TCP instead of UDP to connect to a TURN server and
  // send media between the caller and the callee.
  TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
    static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                                 3478};
    static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
  
    // Enable TCP for the fake turn server.
    CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                     cricket::PROTO_TCP);
  
    webrtc::PeerConnectionInterface::IceServer ice_server;
    ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
    ice_server.username = "test";
    ice_server.password = "test";
  
    PeerConnectionInterface::RTCConfiguration client_1_config;
    client_1_config.servers.push_back(ice_server);
    client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
  
    PeerConnectionInterface::RTCConfiguration client_2_config;
    client_2_config.servers.push_back(ice_server);
    client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
  
    ASSERT_TRUE(
        CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
  
    // Do normal offer/answer and wait for ICE to complete.
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                   callee()->ice_connection_state(), kMaxWaitForFramesMs);
  
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    EXPECT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Verify that a SSLCertificateVerifier passed in through
  // PeerConnectionDependencies is actually used by the underlying SSL
  // implementation to determine whether a certificate presented by the TURN
  // server is accepted by the client. Note that openssladapter_unittest.cc
  // contains more detailed, lower-level tests.
  TEST_P(PeerConnectionIntegrationTest,
         SSLCertificateVerifierUsedForTurnConnections) {
    static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                                 3478};
    static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
  
    // Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
    // that host name verification passes on the fake certificate.
    CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                     cricket::PROTO_TLS, "88.88.88.0");
  
    webrtc::PeerConnectionInterface::IceServer ice_server;
    ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
    ice_server.username = "test";
    ice_server.password = "test";
  
    PeerConnectionInterface::RTCConfiguration client_1_config;
    client_1_config.servers.push_back(ice_server);
    client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
  
    PeerConnectionInterface::RTCConfiguration client_2_config;
    client_2_config.servers.push_back(ice_server);
    // Setting the type to kRelay forces the connection to go through a TURN
    // server.
    client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
  
    // Get a copy to the pointer so we can verify calls later.
    rtc::TestCertificateVerifier* client_1_cert_verifier =
        new rtc::TestCertificateVerifier();
    client_1_cert_verifier->verify_certificate_ = true;
    rtc::TestCertificateVerifier* client_2_cert_verifier =
        new rtc::TestCertificateVerifier();
    client_2_cert_verifier->verify_certificate_ = true;
  
    // Create the dependencies with the test certificate verifier.
    webrtc::PeerConnectionDependencies client_1_deps(nullptr);
    client_1_deps.tls_cert_verifier =
        std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
    webrtc::PeerConnectionDependencies client_2_deps(nullptr);
    client_2_deps.tls_cert_verifier =
        std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
  
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
        client_1_config, std::move(client_1_deps), client_2_config,
        std::move(client_2_deps)));
    ConnectFakeSignaling();
  
    // Set "offer to receive audio/video" without adding any tracks, so we just
    // set up ICE/DTLS with no media.
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 1;
    options.offer_to_receive_video = 1;
    caller()->SetOfferAnswerOptions(options);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
  
    EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
    EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
  }
  
  TEST_P(PeerConnectionIntegrationTest,
         SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection) {
    static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                                 3478};
    static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
  
    // Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
    // that host name verification passes on the fake certificate.
    CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                     cricket::PROTO_TLS, "88.88.88.0");
  
    webrtc::PeerConnectionInterface::IceServer ice_server;
    ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
    ice_server.username = "test";
    ice_server.password = "test";
  
    PeerConnectionInterface::RTCConfiguration client_1_config;
    client_1_config.servers.push_back(ice_server);
    client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
  
    PeerConnectionInterface::RTCConfiguration client_2_config;
    client_2_config.servers.push_back(ice_server);
    // Setting the type to kRelay forces the connection to go through a TURN
    // server.
    client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
  
    // Get a copy to the pointer so we can verify calls later.
    rtc::TestCertificateVerifier* client_1_cert_verifier =
        new rtc::TestCertificateVerifier();
    client_1_cert_verifier->verify_certificate_ = false;
    rtc::TestCertificateVerifier* client_2_cert_verifier =
        new rtc::TestCertificateVerifier();
    client_2_cert_verifier->verify_certificate_ = false;
  
    // Create the dependencies with the test certificate verifier.
    webrtc::PeerConnectionDependencies client_1_deps(nullptr);
    client_1_deps.tls_cert_verifier =
        std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
    webrtc::PeerConnectionDependencies client_2_deps(nullptr);
    client_2_deps.tls_cert_verifier =
        std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
  
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
        client_1_config, std::move(client_1_deps), client_2_config,
        std::move(client_2_deps)));
    ConnectFakeSignaling();
  
    // Set "offer to receive audio/video" without adding any tracks, so we just
    // set up ICE/DTLS with no media.
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 1;
    options.offer_to_receive_video = 1;
    caller()->SetOfferAnswerOptions(options);
    caller()->CreateAndSetAndSignalOffer();
    bool wait_res = true;
    // TODO(bugs.webrtc.org/9219): When IceConnectionState is implemented
    // properly, should be able to just wait for a state of "failed" instead of
    // waiting a fixed 10 seconds.
    WAIT_(DtlsConnected(), kDefaultTimeout, wait_res);
    ASSERT_FALSE(wait_res);
  
    EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
    EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
  }
  
  // Test that audio and video flow end-to-end when codec names don't use the
  // expected casing, given that they're supposed to be case insensitive. To test
  // this, all but one codec is removed from each media description, and its
  // casing is changed.
  //
  // In the past, this has regressed and caused crashes/black video, due to the
  // fact that code at some layers was doing case-insensitive comparisons and
  // code at other layers was not.
  TEST_P(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
  
    // Remove all but one audio/video codec (opus and VP8), and change the
    // casing of the caller's generated offer.
    caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) {
      cricket::AudioContentDescription* audio =
          GetFirstAudioContentDescription(description);
      ASSERT_NE(nullptr, audio);
      auto audio_codecs = audio->codecs();
      audio_codecs.erase(std::remove_if(audio_codecs.begin(), audio_codecs.end(),
                                        [](const cricket::AudioCodec& codec) {
                                          return codec.name != "opus";
                                        }),
                         audio_codecs.end());
      ASSERT_EQ(1u, audio_codecs.size());
      audio_codecs[0].name = "OpUs";
      audio->set_codecs(audio_codecs);
  
      cricket::VideoContentDescription* video =
          GetFirstVideoContentDescription(description);
      ASSERT_NE(nullptr, video);
      auto video_codecs = video->codecs();
      video_codecs.erase(std::remove_if(video_codecs.begin(), video_codecs.end(),
                                        [](const cricket::VideoCodec& codec) {
                                          return codec.name != "VP8";
                                        }),
                         video_codecs.end());
      ASSERT_EQ(1u, video_codecs.size());
      video_codecs[0].name = "vP8";
      video->set_codecs(video_codecs);
    });
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Verify frames are still received end-to-end.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Wait for one audio frame to be received by the callee.
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio(1);
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
    ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
    auto receiver = callee()->pc()->GetReceivers()[0];
    ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
    auto sources = receiver->GetSources();
    ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
    EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
              sources[0].source_id());
    EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
  }
  
  TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddVideoTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Wait for one video frame to be received by the callee.
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeVideo(1);
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
    ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
    auto receiver = callee()->pc()->GetReceivers()[0];
    ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_VIDEO);
    auto sources = receiver->GetSources();
    ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
    EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
              sources[0].source_id());
    EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
  }
  
  // Test that if a track is removed and added again with a different stream ID,
  // the new stream ID is successfully communicated in SDP and media continues to
  // flow end-to-end.
  // TODO(webrtc.bugs.org/8734): This test does not work for Unified Plan because
  // it will not reuse a transceiver that has already been sending. After creating
  // a new transceiver it tries to create an offer with two senders of the same
  // track ids and it fails.
  TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    // Add track using stream 1, do offer/answer.
    rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
        caller()->CreateLocalAudioTrack();
    rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
        caller()->AddTrack(track, {"stream_1"});
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    {
      MediaExpectations media_expectations;
      media_expectations.CalleeExpectsSomeAudio(1);
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
    // Remove the sender, and create a new one with the new stream.
    caller()->pc()->RemoveTrack(sender);
    sender = caller()->AddTrack(track, {"stream_2"});
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    // Wait for additional audio frames to be received by the callee.
    {
      MediaExpectations media_expectations;
      media_expectations.CalleeExpectsSomeAudio();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
  }
  
  TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    auto output = absl::make_unique<testing::NiceMock<MockRtcEventLogOutput>>();
    ON_CALL(*output, IsActive()).WillByDefault(testing::Return(true));
    ON_CALL(*output, Write(::testing::_)).WillByDefault(testing::Return(true));
    EXPECT_CALL(*output, Write(::testing::_)).Times(::testing::AtLeast(1));
    EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
        std::move(output), webrtc::RtcEventLog::kImmediateOutput));
  
    caller()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  }
  
  // Test that if candidates are only signaled by applying full session
  // descriptions (instead of using AddIceCandidate), the peers can connect to
  // each other and exchange media.
  TEST_P(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    // Each side will signal the session descriptions but not candidates.
    ConnectFakeSignalingForSdpOnly();
  
    // Add audio video track and exchange the initial offer/answer with media
    // information only. This will start ICE gathering on each side.
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
    caller()->CreateAndSetAndSignalOffer();
  
    // Wait for all candidates to be gathered on both the caller and callee.
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                   caller()->ice_gathering_state(), kDefaultTimeout);
    ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                   callee()->ice_gathering_state(), kDefaultTimeout);
  
    // The candidates will now be included in the session description, so
    // signaling them will start the ICE connection.
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Ensure that media flows in both directions.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Test that SetAudioPlayout can be used to disable audio playout from the
  // start, then later enable it. This may be useful, for example, if the caller
  // needs to play a local ringtone until some event occurs, after which it
  // switches to playing the received audio.
  TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    // Set up audio-only call where audio playout is disabled on caller's side.
    caller()->pc()->SetAudioPlayout(false);
    caller()->AddAudioTrack();
    callee()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Pump messages for a second.
    WAIT(false, 1000);
    // Since audio playout is disabled, the caller shouldn't have received
    // anything (at the playout level, at least).
    EXPECT_EQ(0, caller()->audio_frames_received());
    // As a sanity check, make sure the callee (for which playout isn't disabled)
    // did still see frames on its audio level.
    ASSERT_GT(callee()->audio_frames_received(), 0);
  
    // Enable playout again, and ensure audio starts flowing.
    caller()->pc()->SetAudioPlayout(true);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  double GetAudioEnergyStat(PeerConnectionWrapper* pc) {
    auto report = pc->NewGetStats();
    auto track_stats_list =
        report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
    const webrtc::RTCMediaStreamTrackStats* remote_track_stats = nullptr;
    for (const auto* track_stats : track_stats_list) {
      if (track_stats->remote_source.is_defined() &&
          *track_stats->remote_source) {
        remote_track_stats = track_stats;
        break;
      }
    }
  
    if (!remote_track_stats->total_audio_energy.is_defined()) {
      return 0.0;
    }
    return *remote_track_stats->total_audio_energy;
  }
  
  // Test that if audio playout is disabled via the SetAudioPlayout() method, then
  // incoming audio is still processed and statistics are generated.
  TEST_P(PeerConnectionIntegrationTest,
         DisableAudioPlayoutStillGeneratesAudioStats) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    // Set up audio-only call where playout is disabled but audio-processing is
    // still active.
    caller()->AddAudioTrack();
    callee()->AddAudioTrack();
    caller()->pc()->SetAudioPlayout(false);
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Wait for the callee to receive audio stats.
    EXPECT_TRUE_WAIT(GetAudioEnergyStat(caller()) > 0, kMaxWaitForFramesMs);
  }
  
  // Test that SetAudioRecording can be used to disable audio recording from the
  // start, then later enable it. This may be useful, for example, if the caller
  // wants to ensure that no audio resources are active before a certain state
  // is reached.
  TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
  
    // Set up audio-only call where audio recording is disabled on caller's side.
    caller()->pc()->SetAudioRecording(false);
    caller()->AddAudioTrack();
    callee()->AddAudioTrack();
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Pump messages for a second.
    WAIT(false, 1000);
    // Since caller has disabled audio recording, the callee shouldn't have
    // received anything.
    EXPECT_EQ(0, callee()->audio_frames_received());
    // As a sanity check, make sure the caller did still see frames on its
    // audio level since audio recording is enabled on the calle side.
    ASSERT_GT(caller()->audio_frames_received(), 0);
  
    // Enable audio recording again, and ensure audio starts flowing.
    caller()->pc()->SetAudioRecording(true);
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Test that after closing PeerConnections, they stop sending any packets (ICE,
  // DTLS, RTP...).
  TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
    // Set up audio/video/data, wait for some frames to be received.
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
  #ifdef HAVE_SCTP
    caller()->CreateDataChannel();
  #endif
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
    // Close PeerConnections.
    caller()->pc()->Close();
    callee()->pc()->Close();
    // Pump messages for a second, and ensure no new packets end up sent.
    uint32_t sent_packets_a = virtual_socket_server()->sent_packets();
    WAIT(false, 1000);
    uint32_t sent_packets_b = virtual_socket_server()->sent_packets();
    EXPECT_EQ(sent_packets_a, sent_packets_b);
  }
  
  // Test that transport stats are generated by the RTCStatsCollector for a
  // connection that only involves data channels. This is a regression test for
  // crbug.com/826972.
  #ifdef HAVE_SCTP
  TEST_P(PeerConnectionIntegrationTest,
         TransportStatsReportedForDataChannelOnlyConnection) {
    ASSERT_TRUE(CreatePeerConnectionWrappers());
    ConnectFakeSignaling();
    caller()->CreateDataChannel();
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
  
    auto caller_report = caller()->NewGetStats();
    EXPECT_EQ(1u, caller_report->GetStatsOfType<RTCTransportStats>().size());
    auto callee_report = callee()->NewGetStats();
    EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
  }
  #endif  // HAVE_SCTP
  
  TEST_P(PeerConnectionIntegrationTest,
         IceEventsGeneratedAndLoggedInRtcEventLog) {
    ASSERT_TRUE(CreatePeerConnectionWrappersWithFakeRtcEventLog());
    ConnectFakeSignaling();
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 1;
    caller()->SetOfferAnswerOptions(options);
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
    ASSERT_NE(nullptr, caller()->event_log_factory());
    ASSERT_NE(nullptr, callee()->event_log_factory());
    webrtc::FakeRtcEventLog* caller_event_log =
        static_cast<webrtc::FakeRtcEventLog*>(
            caller()->event_log_factory()->last_log_created());
    webrtc::FakeRtcEventLog* callee_event_log =
        static_cast<webrtc::FakeRtcEventLog*>(
            callee()->event_log_factory()->last_log_created());
    ASSERT_NE(nullptr, caller_event_log);
    ASSERT_NE(nullptr, callee_event_log);
    int caller_ice_config_count = caller_event_log->GetEventCount(
        webrtc::RtcEvent::Type::IceCandidatePairConfig);
    int caller_ice_event_count = caller_event_log->GetEventCount(
        webrtc::RtcEvent::Type::IceCandidatePairEvent);
    int callee_ice_config_count = callee_event_log->GetEventCount(
        webrtc::RtcEvent::Type::IceCandidatePairConfig);
    int callee_ice_event_count = callee_event_log->GetEventCount(
        webrtc::RtcEvent::Type::IceCandidatePairEvent);
    EXPECT_LT(0, caller_ice_config_count);
    EXPECT_LT(0, caller_ice_event_count);
    EXPECT_LT(0, callee_ice_config_count);
    EXPECT_LT(0, callee_ice_event_count);
  }
  
  INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest,
                           PeerConnectionIntegrationTest,
                           Values(SdpSemantics::kPlanB,
                                  SdpSemantics::kUnifiedPlan));
  
  // Tests that verify interoperability between Plan B and Unified Plan
  // PeerConnections.
  class PeerConnectionIntegrationInteropTest
      : public PeerConnectionIntegrationBaseTest,
        public ::testing::WithParamInterface<
            std::tuple<SdpSemantics, SdpSemantics>> {
   protected:
    // Setting the SdpSemantics for the base test to kDefault does not matter
    // because we specify not to use the test semantics when creating
    // PeerConnectionWrappers.
    PeerConnectionIntegrationInteropTest()
        : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB),
          caller_semantics_(std::get<0>(GetParam())),
          callee_semantics_(std::get<1>(GetParam())) {}
  
    bool CreatePeerConnectionWrappersWithSemantics() {
      return CreatePeerConnectionWrappersWithSdpSemantics(caller_semantics_,
                                                          callee_semantics_);
    }
  
    const SdpSemantics caller_semantics_;
    const SdpSemantics callee_semantics_;
  };
  
  TEST_P(PeerConnectionIntegrationInteropTest, NoMediaLocalToNoMediaRemote) {
    ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
    ConnectFakeSignaling();
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  }
  
  TEST_P(PeerConnectionIntegrationInteropTest, OneAudioLocalToNoMediaRemote) {
    ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
    ConnectFakeSignaling();
    auto audio_sender = caller()->AddAudioTrack();
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Verify that one audio receiver has been created on the remote and that it
    // has the same track ID as the sending track.
    auto receivers = callee()->pc()->GetReceivers();
    ASSERT_EQ(1u, receivers.size());
    EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, receivers[0]->media_type());
    EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id());
  
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  TEST_P(PeerConnectionIntegrationInteropTest, OneAudioOneVideoToNoMediaRemote) {
    ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
    ConnectFakeSignaling();
    auto video_sender = caller()->AddVideoTrack();
    auto audio_sender = caller()->AddAudioTrack();
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Verify that one audio and one video receiver have been created on the
    // remote and that they have the same track IDs as the sending tracks.
    auto audio_receivers =
        callee()->GetReceiversOfType(cricket::MEDIA_TYPE_AUDIO);
    ASSERT_EQ(1u, audio_receivers.size());
    EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id());
    auto video_receivers =
        callee()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO);
    ASSERT_EQ(1u, video_receivers.size());
    EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id());
  
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  TEST_P(PeerConnectionIntegrationInteropTest,
         OneAudioOneVideoLocalToOneAudioOneVideoRemote) {
    ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
    ConnectFakeSignaling();
    caller()->AddAudioVideoTracks();
    callee()->AddAudioVideoTracks();
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  TEST_P(PeerConnectionIntegrationInteropTest,
         ReverseRolesOneAudioLocalToOneVideoRemote) {
    ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
    ConnectFakeSignaling();
    caller()->AddAudioTrack();
    callee()->AddVideoTrack();
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Verify that only the audio track has been negotiated.
    EXPECT_EQ(0u, caller()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO).size());
    // Might also check that the callee's NegotiationNeeded flag is set.
  
    // Reverse roles.
    callee()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    MediaExpectations media_expectations;
    media_expectations.CallerExpectsSomeVideo();
    media_expectations.CalleeExpectsSomeAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  INSTANTIATE_TEST_SUITE_P(
      PeerConnectionIntegrationTest,
      PeerConnectionIntegrationInteropTest,
      Values(std::make_tuple(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
             std::make_tuple(SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB)));
  
  // Test that if the Unified Plan side offers two video tracks then the Plan B
  // side will only see the first one and ignore the second.
  TEST_F(PeerConnectionIntegrationTestPlanB, TwoVideoUnifiedPlanToNoMediaPlanB) {
    ASSERT_TRUE(CreatePeerConnectionWrappersWithSdpSemantics(
        SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB));
    ConnectFakeSignaling();
    auto first_sender = caller()->AddVideoTrack();
    caller()->AddVideoTrack();
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  
    // Verify that there is only one receiver and it corresponds to the first
    // added track.
    auto receivers = callee()->pc()->GetReceivers();
    ASSERT_EQ(1u, receivers.size());
    EXPECT_TRUE(receivers[0]->track()->enabled());
    EXPECT_EQ(first_sender->track()->id(), receivers[0]->track()->id());
  
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  
  // Test that if the initial offer tagged BUNDLE section is rejected due to its
  // associated RtpTransceiver being stopped and another transceiver is added,
  // then renegotiation causes the callee to receive the new video track without
  // error.
  // This is a regression test for bugs.webrtc.org/9954
  TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
         ReOfferWithStoppedBundleTaggedTransceiver) {
    RTCConfiguration config;
    config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
    ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
    ConnectFakeSignaling();
    auto audio_transceiver_or_error =
        caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
    ASSERT_TRUE(audio_transceiver_or_error.ok());
    auto audio_transceiver = audio_transceiver_or_error.MoveValue();
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    {
      MediaExpectations media_expectations;
      media_expectations.CalleeExpectsSomeAudio();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
  
    audio_transceiver->Stop();
    caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack());
  
    caller()->CreateAndSetAndSignalOffer();
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    {
      MediaExpectations media_expectations;
      media_expectations.CalleeExpectsSomeVideo();
      ASSERT_TRUE(ExpectNewFrames(media_expectations));
    }
  }
  
  }  // namespace
  }  // namespace webrtc
  
  #endif  // if !defined(THREAD_SANITIZER)
  