Moving MediaStreamSignaling logic into PeerConnection.

talk/app/webrtc/webrtcsession_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 13 matching lines @@
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include <vector>
 
 #include "talk/app/webrtc/audiotrack.h"
 #include "talk/app/webrtc/fakemetricsobserver.h"
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
+#include "talk/app/webrtc/peerconnection.h"
 #include "talk/app/webrtc/mediastreamsignaling.h"
+#include "talk/app/webrtc/sctputils.h"
+#include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/test/fakeconstraints.h"
 #include "talk/app/webrtc/test/fakedtlsidentitystore.h"
-#include "talk/app/webrtc/test/fakemediastreamsignaling.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/app/webrtc/webrtcsession.h"
 #include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
 #include "talk/media/base/fakemediaengine.h"
 #include "talk/media/base/fakevideorenderer.h"
 #include "talk/media/base/mediachannel.h"
 #include "webrtc/p2p/base/stunserver.h"
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/base/transportchannel.h"
@@ skipping 23 matching lines @@
 using cricket::DF_PLAY;
 using cricket::DF_SEND;
 using cricket::FakeVoiceMediaChannel;
 using cricket::TransportInfo;
 using rtc::SocketAddress;
 using rtc::scoped_ptr;
 using rtc::Thread;
 using webrtc::CreateSessionDescription;
 using webrtc::CreateSessionDescriptionObserver;
 using webrtc::CreateSessionDescriptionRequest;
+using webrtc::DataChannel;
 using webrtc::DtlsIdentityStoreInterface;
 using webrtc::FakeConstraints;
 using webrtc::FakeMetricsObserver;
 using webrtc::IceCandidateCollection;
+using webrtc::InternalDataChannelInit;
 using webrtc::JsepIceCandidate;
 using webrtc::JsepSessionDescription;
 using webrtc::PeerConnectionFactoryInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::SessionDescriptionInterface;
 using webrtc::StreamCollection;
 using webrtc::WebRtcSession;
 using webrtc::kBundleWithoutRtcpMux;
 using webrtc::kCreateChannelFailed;
 using webrtc::kInvalidSdp;
@@ skipping 54 matching lines @@
     "a=ice-pwd:cMvOlFvQ6ochez1ZOoC2uBEC\r\n"
     "a=mid:video\r\n"
     "a=sendrecv\r\n"
     "a=rtcp-mux\r\n"
     "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
     "inline:5/4N5CDvMiyDArHtBByUM71VIkguH17ZNoX60GrA\r\n"
     "a=rtpmap:0 fake_video_codec/90000\r\n"
     "a=rtpmap:96 rtx/90000\r\n"
     "a=fmtp:96 apt=0\r\n";
 
+static const char kStream1[] = "stream1";
+static const char kVideoTrack1[] = "video1";
+static const char kAudioTrack1[] = "audio1";
+
+static const char kStream2[] = "stream2";
+static const char kVideoTrack2[] = "video2";
+static const char kAudioTrack2[] = "audio2";
+
 enum RTCCertificateGenerationMethod { ALREADY_GENERATED, DTLS_IDENTITY_STORE };
 
 // Add some extra |newlines| to the |message| after |line|.
 static void InjectAfter(const std::string& line,
                         const std::string& newlines,
                         std::string* message) {
   const std::string tmp = line + newlines;
   rtc::replace_substrs(line.c_str(), line.length(), tmp.c_str(), tmp.length(),
                        message);
 }
@@ skipping 60 matching lines @@
   PeerConnectionInterface::IceConnectionState ice_connection_state_;
   PeerConnectionInterface::IceGatheringState ice_gathering_state_;
 };
 
 class WebRtcSessionForTest : public webrtc::WebRtcSession {
  public:
   WebRtcSessionForTest(cricket::ChannelManager* cmgr,
                        rtc::Thread* signaling_thread,
                        rtc::Thread* worker_thread,
                        cricket::PortAllocator* port_allocator,
-                       webrtc::IceObserver* ice_observer,
-                       webrtc::MediaStreamSignaling* mediastream_signaling)
-    : WebRtcSession(cmgr, signaling_thread, worker_thread, port_allocator,
-                    mediastream_signaling) {
+                       webrtc::IceObserver* ice_observer)
+      : WebRtcSession(cmgr, signaling_thread, worker_thread, port_allocator) {
     RegisterIceObserver(ice_observer);
   }
   virtual ~WebRtcSessionForTest() {}
 
   // Note that these methods are only safe to use if the signaling thread
   // is the same as the worker thread
   cricket::TransportChannel* voice_rtp_transport_channel() {
     return rtp_transport_channel(voice_channel());
   }
 
@@ skipping 93 matching lines @@
   void SetSink(Sink* sink) override { sink_ = sink; }
 
   int channel_id() const { return channel_id_; }
   cricket::AudioRenderer::Sink* sink() const { return sink_; }
  private:
   int channel_id_;
   cricket::AudioRenderer::Sink* sink_;
 };
 
 class WebRtcSessionTest
-    : public testing::TestWithParam<RTCCertificateGenerationMethod> {
+    : public testing::TestWithParam<RTCCertificateGenerationMethod>,
+      public sigslot::has_slots<> {
  protected:
   // TODO Investigate why ChannelManager crashes, if it's created
   // after stun_server.
   WebRtcSessionTest()
     : media_engine_(new cricket::FakeMediaEngine()),
       data_engine_(new cricket::FakeDataEngine()),
       channel_manager_(new cricket::ChannelManager(
          media_engine_, data_engine_, new cricket::CaptureManager(),
          rtc::Thread::Current())),
       tdesc_factory_(new cricket::TransportDescriptionFactory()),
       desc_factory_(new cricket::MediaSessionDescriptionFactory(
           channel_manager_.get(), tdesc_factory_.get())),
       pss_(new rtc::PhysicalSocketServer),
       vss_(new rtc::VirtualSocketServer(pss_.get())),
       fss_(new rtc::FirewallSocketServer(vss_.get())),
       ss_scope_(fss_.get()),
       stun_socket_addr_(rtc::SocketAddress(kStunAddrHost,
                                                  cricket::STUN_SERVER_PORT)),
       stun_server_(cricket::TestStunServer::Create(Thread::Current(),
                                                    stun_socket_addr_)),
       turn_server_(Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
-      mediastream_signaling_(channel_manager_.get()),
       metrics_observer_(new rtc::RefCountedObject<FakeMetricsObserver>()) {
     cricket::ServerAddresses stun_servers;
     stun_servers.insert(stun_socket_addr_);
     allocator_.reset(new cricket::BasicPortAllocator(
         &network_manager_,
         stun_servers,
         SocketAddress(), SocketAddress(), SocketAddress()));
     allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
                           cricket::PORTALLOCATOR_DISABLE_RELAY);
     EXPECT_TRUE(channel_manager_->Init());
     desc_factory_->set_add_legacy_streams(false);
     allocator_->set_step_delay(cricket::kMinimumStepDelay);
   }
 
   void AddInterface(const SocketAddress& addr) {
     network_manager_.AddInterface(addr);
   }
 
   // If |dtls_identity_store| != null or |rtc_configuration| contains
   // |certificates| then DTLS will be enabled unless explicitly disabled by
   // |rtc_configuration| options. When DTLS is enabled a certificate will be
   // used if provided, otherwise one will be generated using the
   // |dtls_identity_store|.
   void Init(
       rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
       const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
     ASSERT_TRUE(session_.get() == NULL);
     session_.reset(new WebRtcSessionForTest(
-        channel_manager_.get(), rtc::Thread::Current(),
-        rtc::Thread::Current(), allocator_.get(),
-        &observer_,
-        &mediastream_signaling_));
+        channel_manager_.get(), rtc::Thread::Current(), rtc::Thread::Current(),
+        allocator_.get(), &observer_));
+    session_->SignalDataChannelCreationRequested.connect(
+        this, &WebRtcSessionTest::OnDataChannelCreationRequested);
 
     EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
         observer_.ice_connection_state_);
     EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
         observer_.ice_gathering_state_);
 
     EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
                                      dtls_identity_store.Pass(),
                                      rtc_configuration));
     session_->set_metrics_observer(metrics_observer_);
   }
 
+  void OnDataChannelCreationRequested(const std::string& label,
+                                      const InternalDataChannelInit& config) {
+    last_data_channel_label_ = label;
+    last_data_channel_config_ = config;
+  }
+
   void Init() {
     PeerConnectionInterface::RTCConfiguration configuration;
     Init(nullptr, configuration);
   }
 
   void InitWithIceTransport(
       PeerConnectionInterface::IceTransportsType ice_transport_type) {
     PeerConnectionInterface::RTCConfiguration configuration;
     configuration.type = ice_transport_type;
     Init(nullptr, configuration);
@@ skipping 43 matching lines @@
     // Add kTelephoneEventCodec for dtmf test.
     const cricket::AudioCodec kTelephoneEventCodec(
         106, "telephone-event", 8000, 0, 1, 0);
     std::vector<cricket::AudioCodec> codecs;
     codecs.push_back(kTelephoneEventCodec);
     media_engine_->SetAudioCodecs(codecs);
     desc_factory_->set_audio_codecs(codecs);
     Init();
   }
 
-  // Creates a local offer and applies it. Starts ice.
-  // Call mediastream_signaling_.UseOptionsWithStreamX() before this function
+  void SendAudioVideoStream1() {
+    send_stream_1_ = true;
+    send_stream_2_ = false;
+    send_audio_ = true;
+    send_video_ = true;
+  }
+
+  void SendAudioVideoStream2() {
+    send_stream_1_ = false;
+    send_stream_2_ = true;
+    send_audio_ = true;
+    send_video_ = true;
+  }
+
+  void SendAudioVideoStream1And2() {
+    send_stream_1_ = true;
+    send_stream_2_ = true;
+    send_audio_ = true;
+    send_video_ = true;
+  }
+
+  void SendNothing() {
+    send_stream_1_ = false;
+    send_stream_2_ = false;
+    send_audio_ = false;
+    send_video_ = false;
+  }
+
+  void SendAudioOnlyStream2() {
+    send_stream_1_ = false;
+    send_stream_2_ = true;
+    send_audio_ = true;
+    send_video_ = false;
+  }
+
+  void SendVideoOnlyStream2() {
+    send_stream_1_ = false;
+    send_stream_2_ = true;
+    send_audio_ = false;
+    send_video_ = true;
+  }
+
+  void AddStreamsToOptions(cricket::MediaSessionOptions* session_options) {
+    if (send_stream_1_ && send_audio_) {
+      session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, kAudioTrack1,
+                                     kStream1);
+    }
+    if (send_stream_1_ && send_video_) {
+      session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, kVideoTrack1,
+                                     kStream1);
+    }
+    if (send_stream_2_ && send_audio_) {
+      session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, kAudioTrack2,
+                                     kStream2);
+    }
+    if (send_stream_2_ && send_video_) {
+      session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, kVideoTrack2,
+                                     kStream2);
+    }
+    if (data_channel_ && session_->data_channel_type() == cricket::DCT_RTP) {
+      session_options->AddSendStream(cricket::MEDIA_TYPE_DATA,
+                                     data_channel_->label(),
+                                     data_channel_->label());
+    }
+  }
+
+  void GetOptionsForOffer(
+      const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
+      cricket::MediaSessionOptions* session_options) {
+    AddStreamsToOptions(session_options);
+    ASSERT_TRUE(ConvertRtcOptionsForOffer(rtc_options, session_options));
+
+    if (session_->data_channel_type() == cricket::DCT_SCTP && data_channel_) {
+      session_options->data_channel_type = cricket::DCT_SCTP;
+    }
+  }
+
+  void GetOptionsForAnswer(const webrtc::MediaConstraintsInterface* constraints,
+                           cricket::MediaSessionOptions* session_options) {
+    AddStreamsToOptions(session_options);
+    session_options->recv_audio = false;
+    session_options->recv_video = false;
+    ASSERT_TRUE(ParseConstraintsForAnswer(constraints, session_options));
+
+    if (session_->data_channel_type() == cricket::DCT_SCTP) {
+      session_options->data_channel_type = cricket::DCT_SCTP;
+    }
+  }
+
+  // Creates a local offer and applies it. Starts ICE.
+  // Call SendAudioVideoStreamX() before this function
   // to decide which streams to create.
   void InitiateCall() {
     SessionDescriptionInterface* offer = CreateOffer();
     SetLocalDescriptionWithoutError(offer);
     EXPECT_TRUE_WAIT(PeerConnectionInterface::kIceGatheringNew !=
         observer_.ice_gathering_state_,
         kIceCandidatesTimeout);
   }
 
   SessionDescriptionInterface* CreateOffer() {
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.offer_to_receive_audio =
         RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
 
     return CreateOffer(options);
   }
 
   SessionDescriptionInterface* CreateOffer(
       const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
     rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
         observer = new WebRtcSessionCreateSDPObserverForTest();
-    session_->CreateOffer(observer, options);
+    cricket::MediaSessionOptions session_options;
+    GetOptionsForOffer(options, &session_options);
+    session_->CreateOffer(observer, options, session_options);
     EXPECT_TRUE_WAIT(
         observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
         2000);
     return observer->ReleaseDescription();
   }
 
   SessionDescriptionInterface* CreateAnswer(
       const webrtc::MediaConstraintsInterface* constraints) {
     rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observer
         = new WebRtcSessionCreateSDPObserverForTest();
-    session_->CreateAnswer(observer, constraints);
+    cricket::MediaSessionOptions session_options;
+    GetOptionsForAnswer(constraints, &session_options);
+    session_->CreateAnswer(observer, constraints, session_options);
     EXPECT_TRUE_WAIT(
         observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
         2000);
     return observer->ReleaseDescription();
   }
 
   bool ChannelsExist() const {
     return (session_->voice_channel() != NULL &&
             session_->video_channel() != NULL);
   }
@@ skipping 109 matching lines @@
   }
 
   void SetAndVerifyNumUnsignalledRecvStreams(
       int value_set, int value_expected) {
     constraints_.reset(new FakeConstraints());
     constraints_->AddOptional(
         webrtc::MediaConstraintsInterface::kNumUnsignalledRecvStreams,
         value_set);
     session_.reset();
     Init();
-    mediastream_signaling_.SendAudioVideoStream1();
+    SendAudioVideoStream1();
     SessionDescriptionInterface* offer = CreateOffer();
 
     SetLocalDescriptionWithoutError(offer);
 
     video_channel_ = media_engine_->GetVideoChannel(0);
 
     ASSERT_TRUE(video_channel_ != NULL);
     const cricket::VideoOptions& video_options = video_channel_->options();
     EXPECT_EQ(value_expected,
         video_options.unsignalled_recv_stream_limit.GetWithDefaultIfUnset(-1));
@@ skipping 76 matching lines @@
                                  mod_ufrag.c_str(), mod_ufrag.length(),
                                  sdp);
       rtc::replace_substrs(pwd_line.c_str(), pwd_line.length(),
                                  mod_pwd.c_str(), mod_pwd.length(),
                                  sdp);
     }
   }
 
   // Creates a remote offer and and applies it as a remote description,
   // creates a local answer and applies is as a local description.
-  // Call mediastream_signaling_.UseOptionsWithStreamX() before this function
+  // Call SendAudioVideoStreamX() before this function
   // to decide which local and remote streams to create.
   void CreateAndSetRemoteOfferAndLocalAnswer() {
     SessionDescriptionInterface* offer = CreateRemoteOffer();
     SetRemoteDescriptionWithoutError(offer);
     SessionDescriptionInterface* answer = CreateAnswer(NULL);
     SetLocalDescriptionWithoutError(answer);
   }
   void SetLocalDescriptionWithoutError(SessionDescriptionInterface* desc) {
     EXPECT_TRUE(session_->SetLocalDescription(desc, NULL));
     session_->MaybeStartGathering();
@@ skipping 155 matching lines @@
     rtc::replace_substrs(default_port_str, strlen(default_port_str),
                                new_port_str, strlen(new_port_str),
                                &offer_str);
     JsepSessionDescription* offer = new JsepSessionDescription(
         offer_basis->type());
     delete offer_basis;
     offer->Initialize(offer_str, NULL);
     return offer;
   }
 
-  // Create a remote offer. Call mediastream_signaling_.UseOptionsWithStreamX()
+  // Create a remote offer. Call SendAudioVideoStreamX()
   // before this function to decide which streams to create.
   JsepSessionDescription* CreateRemoteOffer() {
     cricket::MediaSessionOptions options;
-    mediastream_signaling_.GetOptionsForAnswer(NULL, &options);
+    GetOptionsForAnswer(NULL, &options);
     return CreateRemoteOffer(options, session_->remote_description());
   }
 
   JsepSessionDescription* CreateRemoteAnswer(
       const SessionDescriptionInterface* offer,
       cricket::MediaSessionOptions options,
       cricket::SecurePolicy policy) {
     desc_factory_->set_secure(policy);
     const std::string session_id =
         rtc::ToString(rtc::CreateRandomId64());
     JsepSessionDescription* answer(
         new JsepSessionDescription(JsepSessionDescription::kAnswer));
     if (!answer->Initialize(desc_factory_->CreateAnswer(offer->description(),
                                                         options, NULL),
                             session_id, kSessionVersion)) {
       delete answer;
       answer = NULL;
     }
     return answer;
   }
 
   JsepSessionDescription* CreateRemoteAnswer(
       const SessionDescriptionInterface* offer,
       cricket::MediaSessionOptions options) {
       return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
   }
 
-  // Creates an answer session description with streams based on
-  // |mediastream_signaling_|. Call
-  // mediastream_signaling_.UseOptionsWithStreamX() before this function
+  // Creates an answer session description.
+  // Call SendAudioVideoStreamX() before this function
   // to decide which streams to create.
   JsepSessionDescription* CreateRemoteAnswer(
       const SessionDescriptionInterface* offer) {
     cricket::MediaSessionOptions options;
-    mediastream_signaling_.GetOptionsForAnswer(NULL, &options);
+    GetOptionsForAnswer(NULL, &options);
     return CreateRemoteAnswer(offer, options, cricket::SEC_REQUIRED);
   }
 
   void TestSessionCandidatesWithBundleRtcpMux(bool bundle, bool rtcp_mux) {
     AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
     Init();
-    mediastream_signaling_.SendAudioVideoStream1();
+    SendAudioVideoStream1();
 
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.use_rtp_mux = bundle;
 
     SessionDescriptionInterface* offer = CreateOffer(options);
     // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
     // and answer.
     SetLocalDescriptionWithoutError(offer);
 
     rtc::scoped_ptr<SessionDescriptionInterface> answer(
@@ skipping 27 matching lines @@
       EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
     }
   }
   // Tests that we can only send DTMF when the dtmf codec is supported.
   void TestCanInsertDtmf(bool can) {
     if (can) {
       InitWithDtmfCodec();
     } else {
       Init();
     }
-    mediastream_signaling_.SendAudioVideoStream1();
+    SendAudioVideoStream1();
     CreateAndSetRemoteOfferAndLocalAnswer();
     EXPECT_FALSE(session_->CanInsertDtmf(""));
     EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
   }
 
   // Helper class to configure loopback network and verify Best
   // Connection using right IP protocol for TestLoopbackCall
   // method. LoopbackNetworkManager applies firewall rules to block
   // all ping traffic once ICE completed, and remove them to observe
   // ICE reconnected again. This LoopbackNetworkConfiguration struct
@@ skipping 95 matching lines @@
   // the correct sequence of ICE states when a connection is established,
   // broken, and re-established.
   // The Connection state should go:
   // New -> Checking -> (Connected) -> Completed -> Disconnected -> Completed
   //     -> Failed.
   // The Gathering state should go: New -> Gathering -> Completed.
 
   void TestLoopbackCall(const LoopbackNetworkConfiguration& config) {
     LoopbackNetworkManager loopback_network_manager(this, config);
     Init();
-    mediastream_signaling_.SendAudioVideoStream1();
+    SendAudioVideoStream1();
     SessionDescriptionInterface* offer = CreateOffer();
 
     EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
               observer_.ice_gathering_state_);
     SetLocalDescriptionWithoutError(offer);
     EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
               observer_.ice_connection_state_);
     EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringGathering,
                    observer_.ice_gathering_state_,
                    kIceCandidatesTimeout);
@@ skipping 76 matching lines @@
     const cricket::AudioContentDescription* audio_content_desc =
         static_cast<const cricket::AudioContentDescription*>(description);
     ASSERT(audio_content_desc != NULL);
     for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) {
       if (audio_content_desc->codecs()[i].name == "CN")
         return false;
     }
     return true;
   }
 
+  void CreateDataChannel() {
+    webrtc::InternalDataChannelInit dci;
+    dci.reliable = session_->data_channel_type() == cricket::DCT_SCTP;
+    data_channel_ = DataChannel::Create(
+        session_.get(), session_->data_channel_type(), "datachannel", dci);
+  }
+
   void SetLocalDescriptionWithDataChannel() {
-    webrtc::InternalDataChannelInit dci;
-    dci.reliable = false;
-    session_->CreateDataChannel("datachannel", &dci);
+    CreateDataChannel();
     SessionDescriptionInterface* offer = CreateOffer();
     SetLocalDescriptionWithoutError(offer);
   }
 
   void VerifyMultipleAsyncCreateDescription(
       RTCCertificateGenerationMethod cert_gen_method,
       CreateSessionDescriptionRequest::Type type) {
     InitWithDtls(cert_gen_method);
     VerifyMultipleAsyncCreateDescriptionAfterInit(true, type);
   }
@@ skipping 10 matching lines @@
     SetFactoryDtlsSrtp();
     if (type == CreateSessionDescriptionRequest::kAnswer) {
       cricket::MediaSessionOptions options;
       scoped_ptr<JsepSessionDescription> offer(
             CreateRemoteOffer(options, cricket::SEC_DISABLED));
       ASSERT_TRUE(offer.get() != NULL);
       SetRemoteDescriptionWithoutError(offer.release());
     }
 
     PeerConnectionInterface::RTCOfferAnswerOptions options;
+    cricket::MediaSessionOptions session_options;
     const int kNumber = 3;
     rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
         observers[kNumber];
     for (int i = 0; i < kNumber; ++i) {
       observers[i] = new WebRtcSessionCreateSDPObserverForTest();
       if (type == CreateSessionDescriptionRequest::kOffer) {
-        session_->CreateOffer(observers[i], options);
+        session_->CreateOffer(observers[i], options, session_options);
       } else {
-        session_->CreateAnswer(observers[i], NULL);
+        session_->CreateAnswer(observers[i], nullptr, session_options);
       }
     }
 
     WebRtcSessionCreateSDPObserverForTest::State expected_state =
         success ? WebRtcSessionCreateSDPObserverForTest::kSucceeded :
                   WebRtcSessionCreateSDPObserverForTest::kFailed;
 
     for (int i = 0; i < kNumber; ++i) {
       EXPECT_EQ_WAIT(expected_state, observers[i]->state(), 1000);
       if (success) {
@@ skipping 24 matching lines @@
   rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
   rtc::scoped_ptr<rtc::FirewallSocketServer> fss_;
   rtc::SocketServerScope ss_scope_;
   rtc::SocketAddress stun_socket_addr_;
   rtc::scoped_ptr<cricket::TestStunServer> stun_server_;
   cricket::TestTurnServer turn_server_;
   rtc::FakeNetworkManager network_manager_;
   rtc::scoped_ptr<cricket::BasicPortAllocator> allocator_;
   PeerConnectionFactoryInterface::Options options_;
   rtc::scoped_ptr<FakeConstraints> constraints_;
-  FakeMediaStreamSignaling mediastream_signaling_;
   rtc::scoped_ptr<WebRtcSessionForTest> session_;
   MockIceObserver observer_;
   cricket::FakeVideoMediaChannel* video_channel_;
   cricket::FakeVoiceMediaChannel* voice_channel_;
   rtc::scoped_refptr<FakeMetricsObserver> metrics_observer_;
+  // The following flags affect options created for CreateOffer/CreateAnswer.
+  bool send_stream_1_ = false;
+  bool send_stream_2_ = false;
+  bool send_audio_ = false;
+  bool send_video_ = false;
+  rtc::scoped_refptr<DataChannel> data_channel_;
+  // Last values received from data channel creation signal.
+  std::string last_data_channel_label_;
+  InternalDataChannelInit last_data_channel_config_;
 };
 
 TEST_P(WebRtcSessionTest, TestInitializeWithDtls) {
   InitWithDtls(GetParam());
   // SDES is disabled when DTLS is on.
   EXPECT_EQ(cricket::SEC_DISABLED, session_->SdesPolicy());
 }
 
 TEST_F(WebRtcSessionTest, TestInitializeWithoutDtls) {
   Init();
@@ skipping 12 matching lines @@
 }
 
 TEST_F(WebRtcSessionTest, TestSessionCandidatesWithBundleRtcpMux) {
   TestSessionCandidatesWithBundleRtcpMux(true, true);
 }
 
 TEST_F(WebRtcSessionTest, TestMultihomeCandidates) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   AddInterface(rtc::SocketAddress(kClientAddrHost2, kClientAddrPort));
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   InitiateCall();
   EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
   EXPECT_EQ(8u, observer_.mline_0_candidates_.size());
   EXPECT_EQ(8u, observer_.mline_1_candidates_.size());
 }
 
 TEST_F(WebRtcSessionTest, TestStunError) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   AddInterface(rtc::SocketAddress(kClientAddrHost2, kClientAddrPort));
   fss_->AddRule(false,
                 rtc::FP_UDP,
                 rtc::FD_ANY,
                 rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   InitiateCall();
   // Since kClientAddrHost1 is blocked, not expecting stun candidates for it.
   EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
   EXPECT_EQ(6u, observer_.mline_0_candidates_.size());
   EXPECT_EQ(6u, observer_.mline_1_candidates_.size());
 }
 
 // Test session delivers no candidates gathered when constraint set to "none".
 TEST_F(WebRtcSessionTest, TestIceTransportsNone) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   InitWithIceTransport(PeerConnectionInterface::kNone);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   InitiateCall();
   EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
   EXPECT_EQ(0u, observer_.mline_0_candidates_.size());
   EXPECT_EQ(0u, observer_.mline_1_candidates_.size());
 }
 
 // Test session delivers only relay candidates gathered when constaint set to
 // "relay".
 TEST_F(WebRtcSessionTest, TestIceTransportsRelay) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   ConfigureAllocatorWithTurn();
   InitWithIceTransport(PeerConnectionInterface::kRelay);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   InitiateCall();
   EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
   EXPECT_EQ(2u, observer_.mline_0_candidates_.size());
   EXPECT_EQ(2u, observer_.mline_1_candidates_.size());
   for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
     EXPECT_EQ(cricket::RELAY_PORT_TYPE,
               observer_.mline_0_candidates_[i].type());
   }
   for (size_t i = 0; i < observer_.mline_1_candidates_.size(); ++i) {
     EXPECT_EQ(cricket::RELAY_PORT_TYPE,
               observer_.mline_1_candidates_[i].type());
   }
 }
 
 // Test session delivers all candidates gathered when constaint set to "all".
 TEST_F(WebRtcSessionTest, TestIceTransportsAll) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   InitWithIceTransport(PeerConnectionInterface::kAll);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   InitiateCall();
   EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
   // Host + STUN. By default allocator is disabled to gather relay candidates.
   EXPECT_EQ(4u, observer_.mline_0_candidates_.size());
   EXPECT_EQ(4u, observer_.mline_1_candidates_.size());
 }
 
 TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
   Init();
   SessionDescriptionInterface* offer = NULL;
   // Since |offer| is NULL, there's no way to tell if it's an offer or answer.
   std::string unknown_action;
   SetLocalDescriptionExpectError(unknown_action, kInvalidSdp, offer);
   SetRemoteDescriptionExpectError(unknown_action, kInvalidSdp, offer);
 }
 
 // Test creating offers and receive answers and make sure the
 // media engine creates the expected send and receive streams.
 TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
   const std::string session_id_orig = offer->session_id();
   const std::string session_version_orig = offer->session_version();
   SetLocalDescriptionWithoutError(offer);
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_EQ(1u, video_channel_->recv_streams().size());
   EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
 
   ASSERT_EQ(1u, voice_channel_->recv_streams().size());
   EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
 
   ASSERT_EQ(1u, video_channel_->send_streams().size());
   EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
   ASSERT_EQ(1u, voice_channel_->send_streams().size());
   EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
 
   // Create new offer without send streams.
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   offer = CreateOffer();
 
   // Verify the session id is the same and the session version is
   // increased.
   EXPECT_EQ(session_id_orig, offer->session_id());
   EXPECT_LT(rtc::FromString<uint64>(session_version_orig),
             rtc::FromString<uint64>(offer->session_version()));
 
   SetLocalDescriptionWithoutError(offer);
   EXPECT_EQ(0u, video_channel_->send_streams().size());
   EXPECT_EQ(0u, voice_channel_->send_streams().size());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   answer = CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   // Make sure the receive streams have not changed.
   ASSERT_EQ(1u, video_channel_->recv_streams().size());
   EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
   ASSERT_EQ(1u, voice_channel_->recv_streams().size());
   EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
 }
 
 // Test receiving offers and creating answers and make sure the
 // media engine creates the expected send and receive streams.
 TEST_F(WebRtcSessionTest, TestReceiveSdesOfferCreateSdesAnswer) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* offer = CreateOffer();
   VerifyCryptoParams(offer->description());
   SetRemoteDescriptionWithoutError(offer);
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   VerifyCryptoParams(answer->description());
   SetLocalDescriptionWithoutError(answer);
 
   const std::string session_id_orig = answer->session_id();
   const std::string session_version_orig = answer->session_version();
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_EQ(1u, video_channel_->recv_streams().size());
   EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
 
   ASSERT_EQ(1u, voice_channel_->recv_streams().size());
   EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
 
   ASSERT_EQ(1u, video_channel_->send_streams().size());
   EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
   ASSERT_EQ(1u, voice_channel_->send_streams().size());
   EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
 
-  mediastream_signaling_.SendAudioVideoStream1And2();
+  SendAudioVideoStream1And2();
   offer = CreateOffer();
   SetRemoteDescriptionWithoutError(offer);
 
   // Answer by turning off all send streams.
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   answer = CreateAnswer(NULL);
 
   // Verify the session id is the same and the session version is
   // increased.
   EXPECT_EQ(session_id_orig, answer->session_id());
   EXPECT_LT(rtc::FromString<uint64>(session_version_orig),
             rtc::FromString<uint64>(answer->session_version()));
   SetLocalDescriptionWithoutError(answer);
 
   ASSERT_EQ(2u, video_channel_->recv_streams().size());
@@ skipping 93 matching lines @@
   // SetRemoteDescription and SetLocalDescription will take the ownership of
   // the offer.
   SetLocalDescriptionWithoutError(offer);
   SetRemoteDescriptionAnswerExpectError(kSdpWithoutSdesCrypto, answer);
 }
 
 // Test that we accept an offer with a DTLS fingerprint when DTLS is on
 // and that we return an answer with a DTLS fingerprint.
 TEST_P(WebRtcSessionTest, TestReceiveDtlsOfferCreateDtlsAnswer) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   InitWithDtls(GetParam());
   SetFactoryDtlsSrtp();
   cricket::MediaSessionOptions options;
   options.recv_video = true;
   JsepSessionDescription* offer =
       CreateRemoteOffer(options, cricket::SEC_DISABLED);
   ASSERT_TRUE(offer != NULL);
   VerifyFingerprintStatus(offer->description(), true);
   VerifyNoCryptoParams(offer->description(), true);
 
   // SetRemoteDescription will take the ownership of the offer.
   SetRemoteDescriptionWithoutError(offer);
 
   // Verify that we get a crypto fingerprint in the answer.
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   ASSERT_TRUE(answer != NULL);
   VerifyFingerprintStatus(answer->description(), true);
   // Check that we don't have an a=crypto line in the answer.
   VerifyNoCryptoParams(answer->description(), true);
 
   // Now set the local description, which should work, even without a=crypto.
   SetLocalDescriptionWithoutError(answer);
 }
 
 // Test that we set a local offer with a DTLS fingerprint when DTLS is on
 // and then we accept a remote answer with a DTLS fingerprint successfully.
 TEST_P(WebRtcSessionTest, TestCreateDtlsOfferReceiveDtlsAnswer) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   InitWithDtls(GetParam());
   SetFactoryDtlsSrtp();
 
   // Verify that we get a crypto fingerprint in the answer.
   SessionDescriptionInterface* offer = CreateOffer();
   ASSERT_TRUE(offer != NULL);
   VerifyFingerprintStatus(offer->description(), true);
   // Check that we don't have an a=crypto line in the offer.
   VerifyNoCryptoParams(offer->description(), true);
 
@@ skipping 68 matching lines @@
   // SetRemoteDescription and SetLocalDescription will take the ownership of
   // the offer and answer.
   SetLocalDescriptionWithoutError(offer);
   SetRemoteDescriptionAnswerExpectError(
       kSdpWithoutDtlsFingerprint, answer);
 }
 
 // Test that we create a local offer without SDES or DTLS and accept a remote
 // answer without SDES or DTLS when encryption is disabled.
 TEST_P(WebRtcSessionTest, TestCreateOfferReceiveAnswerWithoutEncryption) {
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   options_.disable_encryption = true;
   InitWithDtls(GetParam());
 
   // Verify that we get a crypto fingerprint in the answer.
   SessionDescriptionInterface* offer = CreateOffer();
   ASSERT_TRUE(offer != NULL);
   VerifyFingerprintStatus(offer->description(), false);
   // Check that we don't have an a=crypto line in the offer.
   VerifyNoCryptoParams(offer->description(), false);
 
@@ skipping 35 matching lines @@
   VerifyFingerprintStatus(answer->description(), false);
   // Check that we don't have an a=crypto line in the answer.
   VerifyNoCryptoParams(answer->description(), false);
 
   // Now set the local description, which should work, even without a=crypto.
   SetLocalDescriptionWithoutError(answer);
 }
 
 TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   // SetLocalDescription take ownership of offer.
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
 
   // SetLocalDescription take ownership of offer.
   SessionDescriptionInterface* offer2 = CreateOffer();
   SetLocalDescriptionWithoutError(offer2);
 }
 
 TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   // SetLocalDescription take ownership of offer.
   SessionDescriptionInterface* offer = CreateOffer();
   SetRemoteDescriptionWithoutError(offer);
 
   SessionDescriptionInterface* offer2 = CreateOffer();
   SetRemoteDescriptionWithoutError(offer2);
 }
 
 TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
   offer = CreateOffer();
   SetRemoteDescriptionOfferExpectError(
       "Called in wrong state: STATE_SENTINITIATE", offer);
 }
 
 TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   SessionDescriptionInterface* offer = CreateOffer();
   SetRemoteDescriptionWithoutError(offer);
   offer = CreateOffer();
   SetLocalDescriptionOfferExpectError(
       "Called in wrong state: STATE_RECEIVEDINITIATE", offer);
 }
 
 TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   SetRemoteDescriptionExpectState(offer, BaseSession::STATE_RECEIVEDINITIATE);
 
   JsepSessionDescription* pranswer = static_cast<JsepSessionDescription*>(
       CreateAnswer(NULL));
   pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
   SetLocalDescriptionExpectState(pranswer, BaseSession::STATE_SENTPRACCEPT);
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   JsepSessionDescription* pranswer2 = static_cast<JsepSessionDescription*>(
       CreateAnswer(NULL));
   pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
 
   SetLocalDescriptionExpectState(pranswer2, BaseSession::STATE_SENTPRACCEPT);
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   SetLocalDescriptionExpectState(answer, BaseSession::STATE_SENTACCEPT);
 }
 
 TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
 
   JsepSessionDescription* pranswer =
       CreateRemoteAnswer(session_->local_description());
   pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
 
   SetRemoteDescriptionExpectState(pranswer,
                                   BaseSession::STATE_RECEIVEDPRACCEPT);
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   JsepSessionDescription* pranswer2 =
       CreateRemoteAnswer(session_->local_description());
   pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
 
   SetRemoteDescriptionExpectState(pranswer2,
                                   BaseSession::STATE_RECEIVEDPRACCEPT);
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionExpectState(answer, BaseSession::STATE_RECEIVEDACCEPT);
 }
 
 TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(offer.get());
   SetLocalDescriptionAnswerExpectError("Called in wrong state: STATE_INIT",
                                        answer);
 }
 
 TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(offer.get());
   SetRemoteDescriptionAnswerExpectError(
       "Called in wrong state: STATE_INIT", answer);
 }
 
 TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   cricket::Candidate candidate;
   candidate.set_component(1);
   JsepIceCandidate ice_candidate1(kMediaContentName0, 0, candidate);
 
   // Fail since we have not set a offer description.
   EXPECT_FALSE(session_->ProcessIceMessage(&ice_candidate1));
 
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
@@ skipping 32 matching lines @@
 }
 
 // Test that a remote candidate is added to the remote session description and
 // that it is retained if the remote session description is changed.
 TEST_F(WebRtcSessionTest, TestRemoteCandidatesAddedToSessionDescription) {
   Init();
   cricket::Candidate candidate1;
   candidate1.set_component(1);
   JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
                                   candidate1);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
   const SessionDescriptionInterface* remote_desc =
       session_->remote_description();
   ASSERT_TRUE(remote_desc != NULL);
   ASSERT_EQ(2u, remote_desc->number_of_mediasections());
   const IceCandidateCollection* candidates =
       remote_desc->candidates(kMediaContentIndex0);
   ASSERT_EQ(1u, candidates->count());
@@ skipping 28 matching lines @@
 
   // Test that the candidate is ignored if we can add the same candidate again.
   EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2));
 }
 
 // Test that local candidates are added to the local session description and
 // that they are retained if the local session description is changed.
 TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   const SessionDescriptionInterface* local_desc = session_->local_description();
   const IceCandidateCollection* candidates =
       local_desc->candidates(kMediaContentIndex0);
   ASSERT_TRUE(candidates != NULL);
   EXPECT_EQ(0u, candidates->count());
 
   EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
 
   local_desc = session_->local_description();
   candidates = local_desc->candidates(kMediaContentIndex0);
   ASSERT_TRUE(candidates != NULL);
   EXPECT_LT(0u, candidates->count());
   candidates = local_desc->candidates(1);
   ASSERT_TRUE(candidates != NULL);
   EXPECT_EQ(0u, candidates->count());
 
   // Update the session descriptions.
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   local_desc = session_->local_description();
   candidates = local_desc->candidates(kMediaContentIndex0);
   ASSERT_TRUE(candidates != NULL);
   EXPECT_LT(0u, candidates->count());
   candidates = local_desc->candidates(1);
   ASSERT_TRUE(candidates != NULL);
   EXPECT_EQ(0u, candidates->count());
 }
 
 // Test that we can set a remote session description with remote candidates.
 TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
   Init();
 
   cricket::Candidate candidate1;
   candidate1.set_component(1);
   JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
                                  candidate1);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
   SetRemoteDescriptionWithoutError(offer);
 
   const SessionDescriptionInterface* remote_desc =
       session_->remote_description();
   ASSERT_TRUE(remote_desc != NULL);
   ASSERT_EQ(2u, remote_desc->number_of_mediasections());
   const IceCandidateCollection* candidates =
       remote_desc->candidates(kMediaContentIndex0);
   ASSERT_EQ(1u, candidates->count());
   EXPECT_EQ(kMediaContentIndex0, candidates->at(0)->sdp_mline_index());
 
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
 }
 
 // Test that offers and answers contains ice candidates when Ice candidates have
 // been gathered.
 TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   // Ice is started but candidates are not provided until SetLocalDescription
   // is called.
   EXPECT_EQ(0u, observer_.mline_0_candidates_.size());
   EXPECT_EQ(0u, observer_.mline_1_candidates_.size());
   CreateAndSetRemoteOfferAndLocalAnswer();
   // Wait until at least one local candidate has been collected.
   EXPECT_TRUE_WAIT(0u < observer_.mline_0_candidates_.size(),
                    kIceCandidatesTimeout);
 
   rtc::scoped_ptr<SessionDescriptionInterface> local_offer(CreateOffer());
 
   ASSERT_TRUE(local_offer->candidates(kMediaContentIndex0) != NULL);
   EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex0)->count());
 
   SessionDescriptionInterface* remote_offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(remote_offer);
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   ASSERT_TRUE(answer->candidates(kMediaContentIndex0) != NULL);
   EXPECT_LT(0u, answer->candidates(kMediaContentIndex0)->count());
   SetLocalDescriptionWithoutError(answer);
 }
 
 // Verifies TransportProxy and media channels are created with content names
 // present in the SessionDescription.
 TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   // CreateOffer creates session description with the content names "audio" and
   // "video". Goal is to modify these content names and verify transport channel
   // proxy in the BaseSession, as proxies are created with the content names
   // present in SDP.
   std::string sdp;
   EXPECT_TRUE(offer->ToString(&sdp));
   const std::string kAudioMid = "a=mid:audio";
   const std::string kAudioMidReplaceStr = "a=mid:audio_content_name";
@@ skipping 44 matching lines @@
   EXPECT_TRUE(content != NULL);
   content = cricket::GetFirstVideoContent(offer->description());
   EXPECT_TRUE(content == NULL);
 }
 
 // Test that an offer contains the correct media content descriptions based on
 // the send streams when no constraints have been set.
 TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
   Init();
   // Test Audio only offer.
-  mediastream_signaling_.UseOptionsAudioOnly();
+  SendAudioOnlyStream2();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(offer->description());
   EXPECT_TRUE(content != NULL);
   content = cricket::GetFirstVideoContent(offer->description());
   EXPECT_TRUE(content == NULL);
 
   // Test Audio / Video offer.
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   offer.reset(CreateOffer());
   content = cricket::GetFirstAudioContent(offer->description());
   EXPECT_TRUE(content != NULL);
   content = cricket::GetFirstVideoContent(offer->description());
   EXPECT_TRUE(content != NULL);
 }
 
 // Test that an offer contains no media content descriptions if
 // kOfferToReceiveVideo and kOfferToReceiveAudio constraints are set to false.
 TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
@@ skipping 118 matching lines @@
 }
 
 // Test that an answer contains the correct media content descriptions when no
 // constraints have been set.
 TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
   Init();
   // Create a remote offer with audio and video content.
   rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(offer.release());
   // Test with a stream with tracks.
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateAnswer(NULL));
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(answer->description());
   ASSERT_TRUE(content != NULL);
   EXPECT_FALSE(content->rejected);
 
   content = cricket::GetFirstVideoContent(answer->description());
   ASSERT_TRUE(content != NULL);
   EXPECT_FALSE(content->rejected);
@@ skipping 29 matching lines @@
   Init();
   // Create a remote offer with audio and video content.
   rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(offer.release());
 
   webrtc::FakeConstraints constraints_no_receive;
   constraints_no_receive.SetMandatoryReceiveAudio(false);
   constraints_no_receive.SetMandatoryReceiveVideo(false);
 
   // Test with a stream with tracks.
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateAnswer(&constraints_no_receive));
 
   // TODO(perkj): Should the direction be set to SEND_ONLY?
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(answer->description());
   ASSERT_TRUE(content != NULL);
   EXPECT_FALSE(content->rejected);
 
   // TODO(perkj): Should the direction be set to SEND_ONLY?
@@ skipping 36 matching lines @@
   EXPECT_TRUE(VerifyNoCNCodecs(content));
 }
 
 // This test verifies the call setup when remote answer with audio only and
 // later updates with video.
 TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
   Init();
   EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
   EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   cricket::MediaSessionOptions options;
   SessionDescriptionInterface* answer = CreateRemoteAnswer(offer, options);
 
   // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
   // and answer;
   SetLocalDescriptionWithoutError(offer);
   SetRemoteDescriptionWithoutError(answer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_TRUE(video_channel_ == NULL);
 
   ASSERT_EQ(0u, voice_channel_->recv_streams().size());
   ASSERT_EQ(1u, voice_channel_->send_streams().size());
   EXPECT_EQ(kAudioTrack1, voice_channel_->send_streams()[0].id);
 
   // Let the remote end update the session descriptions, with Audio and Video.
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_TRUE(video_channel_ != NULL);
   ASSERT_TRUE(voice_channel_ != NULL);
 
   ASSERT_EQ(1u, video_channel_->recv_streams().size());
   ASSERT_EQ(1u, video_channel_->send_streams().size());
   EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
   EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
   ASSERT_EQ(1u, voice_channel_->recv_streams().size());
   ASSERT_EQ(1u, voice_channel_->send_streams().size());
   EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
   EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
 
   // Change session back to audio only.
-  mediastream_signaling_.UseOptionsAudioOnly();
+  SendAudioOnlyStream2();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   EXPECT_EQ(0u, video_channel_->recv_streams().size());
   ASSERT_EQ(1u, voice_channel_->recv_streams().size());
   EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
   ASSERT_EQ(1u, voice_channel_->send_streams().size());
   EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
 }
 
 // This test verifies the call setup when remote answer with video only and
 // later updates with audio.
 TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
   Init();
   EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
   EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   cricket::MediaSessionOptions options;
   options.recv_audio = false;
   options.recv_video = true;
   SessionDescriptionInterface* answer = CreateRemoteAnswer(
       offer, options, cricket::SEC_ENABLED);
 
   // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
   // and answer.
   SetLocalDescriptionWithoutError(offer);
   SetRemoteDescriptionWithoutError(answer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_TRUE(voice_channel_ == NULL);
   ASSERT_TRUE(video_channel_ != NULL);
 
   EXPECT_EQ(0u, video_channel_->recv_streams().size());
   ASSERT_EQ(1u, video_channel_->send_streams().size());
   EXPECT_EQ(kVideoTrack1, video_channel_->send_streams()[0].id);
 
   // Update the session descriptions, with Audio and Video.
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   voice_channel_ = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(voice_channel_ != NULL);
 
   ASSERT_EQ(1u, voice_channel_->recv_streams().size());
   ASSERT_EQ(1u, voice_channel_->send_streams().size());
   EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
   EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
 
   // Change session back to video only.
-  mediastream_signaling_.UseOptionsVideoOnly();
+  SendVideoOnlyStream2();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_EQ(1u, video_channel_->recv_streams().size());
   EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
   ASSERT_EQ(1u, video_channel_->send_streams().size());
   EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
 }
 
 TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
   VerifyCryptoParams(offer->description());
   SetRemoteDescriptionWithoutError(offer.release());
   scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
   VerifyCryptoParams(answer->description());
 }
 
 TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
   options_.disable_encryption = true;
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
   VerifyNoCryptoParams(offer->description(), false);
 }
 
 TEST_F(WebRtcSessionTest, VerifyAnswerFromNonCryptoOffer) {
   Init();
   VerifyAnswerFromNonCryptoOffer();
 }
 
 TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) {
   Init();
   VerifyAnswerFromCryptoOffer();
 }
 
 // This test verifies that setLocalDescription fails if
 // no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
 TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   std::string sdp;
   RemoveIceUfragPwdLines(offer.get(), &sdp);
   SessionDescriptionInterface* modified_offer =
     CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
   SetLocalDescriptionOfferExpectError(kSdpWithoutIceUfragPwd, modified_offer);
 }
 
 // This test verifies that setRemoteDescription fails if
 // no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
 TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
   Init();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
   std::string sdp;
   RemoveIceUfragPwdLines(offer.get(), &sdp);
   SessionDescriptionInterface* modified_offer =
     CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
   SetRemoteDescriptionOfferExpectError(kSdpWithoutIceUfragPwd, modified_offer);
 }
 
 // This test verifies that setLocalDescription fails if local offer has
 // too short ice ufrag and pwd strings.
 TEST_F(WebRtcSessionTest, TestSetLocalDescriptionInvalidIceCredentials) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   std::string sdp;
   // Modifying ice ufrag and pwd in local offer with strings smaller than the
   // recommended values of 4 and 22 bytes respectively.
   ModifyIceUfragPwdLines(offer.get(), "ice", "icepwd", &sdp);
   SessionDescriptionInterface* modified_offer =
       CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
   std::string error;
   EXPECT_FALSE(session_->SetLocalDescription(modified_offer, &error));
@@ skipping 85 matching lines @@
   SetRemoteDescriptionWithoutError(offer4);
   EXPECT_EQ(0, session_->remote_description()->candidates(0)->count());
 }
 
 // Test that candidates sent to the "video" transport do not get pushed down to
 // the "audio" transport channel when bundling using TransportProxy.
 TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
 
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   SetRemoteDescriptionWithoutError(offer);
 
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
 
@@ skipping 62 matching lines @@
 
   // No need here for a _WAIT check since we are checking that state hasn't
   // changed: if this is false we would be doing waits for nothing and if this
   // is true then there will be no messages processed anyways.
   EXPECT_FALSE(connection_with_remote_port(6000));
 }
 
 // kBundlePolicyBalanced BUNDLE policy and answer contains BUNDLE.
 TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyBalanced BUNDLE policy but no BUNDLE in the answer.
 TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
 
   // Remove BUNDLE from the answer.
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateRemoteAnswer(session_->local_description()));
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
   answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
   JsepSessionDescription* modified_answer =
       new JsepSessionDescription(JsepSessionDescription::kAnswer);
   modified_answer->Initialize(answer_copy, "1", "1");
   SetRemoteDescriptionWithoutError(modified_answer);  //
 
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyMaxBundle policy with BUNDLE in the answer.
 TEST_F(WebRtcSessionTest, TestMaxBundleBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyMaxBundle policy with BUNDLE in the answer, but no
 // audio content in the answer.
 TEST_F(WebRtcSessionTest, TestMaxBundleRejectAudio) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   cricket::MediaSessionOptions recv_options;
   recv_options.recv_audio = false;
   recv_options.recv_video = true;
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description(), recv_options);
   SetRemoteDescriptionWithoutError(answer);
 
   EXPECT_TRUE(NULL == session_->voice_channel());
   EXPECT_TRUE(NULL != session_->video_rtp_transport_channel());
 
   session_->Terminate();
   EXPECT_TRUE(NULL == session_->voice_rtp_transport_channel());
   EXPECT_TRUE(NULL == session_->voice_rtcp_transport_channel());
   EXPECT_TRUE(NULL == session_->video_rtp_transport_channel());
   EXPECT_TRUE(NULL == session_->video_rtcp_transport_channel());
 }
 
 // kBundlePolicyMaxBundle policy but no BUNDLE in the answer.
 TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
 
   // Remove BUNDLE from the answer.
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateRemoteAnswer(session_->local_description()));
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
   answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
   JsepSessionDescription* modified_answer =
       new JsepSessionDescription(JsepSessionDescription::kAnswer);
   modified_answer->Initialize(answer_copy, "1", "1");
   SetRemoteDescriptionWithoutError(modified_answer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyMaxBundle policy with BUNDLE in the remote offer.
 TEST_F(WebRtcSessionTest, TestMaxBundleBundleInRemoteOffer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   SetRemoteDescriptionWithoutError(offer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer = CreateAnswer(nullptr);
   SetLocalDescriptionWithoutError(answer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyMaxBundle policy but no BUNDLE in the remote offer.
 TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInRemoteOffer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   // Remove BUNDLE from the offer.
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
   cricket::SessionDescription* offer_copy = offer->description()->Copy();
   offer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
   JsepSessionDescription* modified_offer =
       new JsepSessionDescription(JsepSessionDescription::kOffer);
   modified_offer->Initialize(offer_copy, "1", "1");
 
   // Expect an error when applying the remote description
   SetRemoteDescriptionExpectError(JsepSessionDescription::kOffer,
                                   kCreateChannelFailed, modified_offer);
 }
 
 // kBundlePolicyMaxCompat bundle policy and answer contains BUNDLE.
 TEST_F(WebRtcSessionTest, TestMaxCompatBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxCompat);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   // This should lead to an audio-only call but isn't implemented
   // correctly yet.
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyMaxCompat BUNDLE policy but no BUNDLE in the answer.
 TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxCompat);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
 
   // Remove BUNDLE from the answer.
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateRemoteAnswer(session_->local_description()));
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
   answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
   JsepSessionDescription* modified_answer =
       new JsepSessionDescription(JsepSessionDescription::kAnswer);
   modified_answer->Initialize(answer_copy, "1", "1");
   SetRemoteDescriptionWithoutError(modified_answer);  //
 
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyMaxbundle and then we call SetRemoteDescription first.
 TEST_F(WebRtcSessionTest, TestMaxBundleWithSetRemoteDescriptionFirst) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetRemoteDescriptionWithoutError(offer);
 
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 }
 
 TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
   InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyRequire);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL);
   EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL);
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL);
   EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL);
 }
 
 TEST_F(WebRtcSessionTest, TestNegotiateRtcpMux) {
   InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyNegotiate);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   EXPECT_TRUE(session_->voice_rtcp_transport_channel() != NULL);
   EXPECT_TRUE(session_->video_rtcp_transport_channel() != NULL);
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL);
   EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL);
 }
 
 // This test verifies that SetLocalDescription and SetRemoteDescription fails
 // if BUNDLE is enabled but rtcp-mux is disabled in m-lines.
 TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   std::string offer_str;
   offer->ToString(&offer_str);
   // Disable rtcp-mux
   const std::string rtcp_mux = "rtcp-mux";
   const std::string xrtcp_mux = "xrtcp-mux";
   rtc::replace_substrs(rtcp_mux.c_str(), rtcp_mux.length(),
                              xrtcp_mux.c_str(), xrtcp_mux.length(),
                              &offer_str);
   JsepSessionDescription* local_offer =
       new JsepSessionDescription(JsepSessionDescription::kOffer);
   EXPECT_TRUE((local_offer)->Initialize(offer_str, NULL));
   SetLocalDescriptionOfferExpectError(kBundleWithoutRtcpMux, local_offer);
   JsepSessionDescription* remote_offer =
       new JsepSessionDescription(JsepSessionDescription::kOffer);
   EXPECT_TRUE((remote_offer)->Initialize(offer_str, NULL));
   SetRemoteDescriptionOfferExpectError(kBundleWithoutRtcpMux, remote_offer);
   // Trying unmodified SDP.
   SetLocalDescriptionWithoutError(offer);
 }
 
 TEST_F(WebRtcSessionTest, SetAudioPlayout) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->recv_streams().size());
   uint32 receive_ssrc  = channel->recv_streams()[0].first_ssrc();
   double left_vol, right_vol;
   EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
   EXPECT_EQ(1, left_vol);
   EXPECT_EQ(1, right_vol);
   rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
   session_->SetAudioPlayout(receive_ssrc, false, renderer.get());
   EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
   EXPECT_EQ(0, left_vol);
   EXPECT_EQ(0, right_vol);
   EXPECT_EQ(0, renderer->channel_id());
   session_->SetAudioPlayout(receive_ssrc, true, NULL);
   EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
   EXPECT_EQ(1, left_vol);
   EXPECT_EQ(1, right_vol);
   EXPECT_EQ(-1, renderer->channel_id());
 }
 
 TEST_F(WebRtcSessionTest, SetAudioSend) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
   uint32 send_ssrc  = channel->send_streams()[0].first_ssrc();
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
 
   cricket::AudioOptions options;
   options.echo_cancellation.Set(true);
 
   rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
   session_->SetAudioSend(send_ssrc, false, options, renderer.get());
   EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
   EXPECT_FALSE(channel->options().echo_cancellation.IsSet());
   EXPECT_EQ(0, renderer->channel_id());
   EXPECT_TRUE(renderer->sink() != NULL);
 
   // This will trigger SetSink(NULL) to the |renderer|.
   session_->SetAudioSend(send_ssrc, true, options, NULL);
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
   bool value;
   EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
   EXPECT_TRUE(value);
   EXPECT_EQ(-1, renderer->channel_id());
   EXPECT_TRUE(renderer->sink() == NULL);
 }
 
 TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
   uint32 send_ssrc  = channel->send_streams()[0].first_ssrc();
 
   rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
   cricket::AudioOptions options;
   session_->SetAudioSend(send_ssrc, true, options, renderer.get());
   EXPECT_TRUE(renderer->sink() != NULL);
 
   // Delete the |renderer| and it will trigger OnClose() to the sink, and this
   // will invalidate the |renderer_| pointer in the sink and prevent getting a
   // SetSink(NULL) callback afterwards.
   renderer.reset();
 
   // This will trigger SetSink(NULL) if no OnClose() callback.
   session_->SetAudioSend(send_ssrc, true, options, NULL);
 }
 
 TEST_F(WebRtcSessionTest, SetVideoPlayout) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_LT(0u, channel->renderers().size());
   EXPECT_TRUE(channel->renderers().begin()->second == NULL);
   ASSERT_EQ(1u, channel->recv_streams().size());
   uint32 receive_ssrc  = channel->recv_streams()[0].first_ssrc();
   cricket::FakeVideoRenderer renderer;
   session_->SetVideoPlayout(receive_ssrc, true, &renderer);
   EXPECT_TRUE(channel->renderers().begin()->second == &renderer);
   session_->SetVideoPlayout(receive_ssrc, false, &renderer);
   EXPECT_TRUE(channel->renderers().begin()->second == NULL);
 }
 
 TEST_F(WebRtcSessionTest, SetVideoSend) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
   uint32 send_ssrc  = channel->send_streams()[0].first_ssrc();
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
   cricket::VideoOptions* options = NULL;
   session_->SetVideoSend(send_ssrc, false, options);
   EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
   session_->SetVideoSend(send_ssrc, true, options);
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
 }
 
 TEST_F(WebRtcSessionTest, CanNotInsertDtmf) {
   TestCanInsertDtmf(false);
 }
 
 TEST_F(WebRtcSessionTest, CanInsertDtmf) {
   TestCanInsertDtmf(true);
 }
 
 TEST_F(WebRtcSessionTest, InsertDtmf) {
   // Setup
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   EXPECT_EQ(0U, channel->dtmf_info_queue().size());
 
   // Insert DTMF
   const int expected_flags = DF_SEND;
   const int expected_duration = 90;
   session_->InsertDtmf(kAudioTrack1, 0, expected_duration);
   session_->InsertDtmf(kAudioTrack1, 1, expected_duration);
   session_->InsertDtmf(kAudioTrack1, 2, expected_duration);
@@ skipping 30 matching lines @@
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
 
   EXPECT_FALSE(session_->initiator());
   SetLocalDescriptionWithoutError(answer);
   EXPECT_FALSE(session_->initiator());
 }
 
 // Verifing local offer and remote answer have matching m-lines as per RFC 3264.
 TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateRemoteAnswer(session_->local_description()));
 
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
   answer_copy->RemoveContentByName("video");
   JsepSessionDescription* modified_answer =
       new JsepSessionDescription(JsepSessionDescription::kAnswer);
 
@@ skipping 27 matching lines @@
       CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
   SetRemoteDescriptionAnswerExpectError(kMlineMismatch, modified_answer2);
 
   SetRemoteDescriptionWithoutError(answer.release());
 }
 
 // Verifying remote offer and local answer have matching m-lines as per
 // RFC 3264.
 TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   SetRemoteDescriptionWithoutError(offer);
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
 
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
   answer_copy->RemoveContentByName("video");
   JsepSessionDescription* modified_answer =
       new JsepSessionDescription(JsepSessionDescription::kAnswer);
 
   EXPECT_TRUE(modified_answer->Initialize(answer_copy,
                                           answer->session_id(),
                                           answer->session_version()));
   SetLocalDescriptionAnswerExpectError(kMlineMismatch, modified_answer);
   SetLocalDescriptionWithoutError(answer);
 }
 
 // This test verifies that WebRtcSession does not start candidate allocation
 // before SetLocalDescription is called.
 TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   cricket::Candidate candidate;
   candidate.set_component(1);
   JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
                                  candidate);
   EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
   cricket::Candidate candidate1;
   candidate1.set_component(1);
   JsepIceCandidate ice_candidate1(kMediaContentName1, kMediaContentIndex1,
                                   candidate1);
   EXPECT_TRUE(offer->AddCandidate(&ice_candidate1));
   SetRemoteDescriptionWithoutError(offer);
   ASSERT_TRUE(session_->voice_rtp_transport_channel() != NULL);
   ASSERT_TRUE(session_->video_rtp_transport_channel() != NULL);
 
   // Pump for 1 second and verify that no candidates are generated.
   rtc::Thread::Current()->ProcessMessages(1000);
   EXPECT_TRUE(observer_.mline_0_candidates_.empty());
   EXPECT_TRUE(observer_.mline_1_candidates_.empty());
 
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
   EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
 }
 
 // This test verifies that crypto parameter is updated in local session
 // description as per security policy set in MediaSessionDescriptionFactory.
 TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   // Making sure SetLocalDescription correctly sets crypto value in
   // SessionDescription object after de-serialization of sdp string. The value
   // will be set as per MediaSessionDescriptionFactory.
   std::string offer_str;
   offer->ToString(&offer_str);
   SessionDescriptionInterface* jsep_offer_str =
       CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
   SetLocalDescriptionWithoutError(jsep_offer_str);
   EXPECT_TRUE(session_->voice_channel()->secure_required());
   EXPECT_TRUE(session_->video_channel()->secure_required());
 }
 
 // This test verifies the crypto parameter when security is disabled.
 TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
   options_.disable_encryption = true;
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   // Making sure SetLocalDescription correctly sets crypto value in
   // SessionDescription object after de-serialization of sdp string. The value
   // will be set as per MediaSessionDescriptionFactory.
   std::string offer_str;
   offer->ToString(&offer_str);
   SessionDescriptionInterface* jsep_offer_str =
       CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
   SetLocalDescriptionWithoutError(jsep_offer_str);
   EXPECT_FALSE(session_->voice_channel()->secure_required());
   EXPECT_FALSE(session_->video_channel()->secure_required());
 }
 
 // This test verifies that an answer contains new ufrag and password if an offer
 // with new ufrag and password is received.
 TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
   Init();
   cricket::MediaSessionOptions options;
   options.recv_video = true;
   rtc::scoped_ptr<JsepSessionDescription> offer(
       CreateRemoteOffer(options));
   SetRemoteDescriptionWithoutError(offer.release());
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateAnswer(NULL));
   SetLocalDescriptionWithoutError(answer.release());
 
   // Receive an offer with new ufrag and password.
   options.transport_options.ice_restart = true;
   rtc::scoped_ptr<JsepSessionDescription> updated_offer1(
       CreateRemoteOffer(options, session_->remote_description()));
   SetRemoteDescriptionWithoutError(updated_offer1.release());
 
@@ skipping 10 matching lines @@
 // This test verifies that an answer contains old ufrag and password if an offer
 // with old ufrag and password is received.
 TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
   Init();
   cricket::MediaSessionOptions options;
   options.recv_video = true;
   rtc::scoped_ptr<JsepSessionDescription> offer(
       CreateRemoteOffer(options));
   SetRemoteDescriptionWithoutError(offer.release());
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateAnswer(NULL));
   SetLocalDescriptionWithoutError(answer.release());
 
   // Receive an offer without changed ufrag or password.
   options.transport_options.ice_restart = false;
   rtc::scoped_ptr<JsepSessionDescription> updated_offer2(
       CreateRemoteOffer(options, session_->remote_description()));
   SetRemoteDescriptionWithoutError(updated_offer2.release());
 
   rtc::scoped_ptr<SessionDescriptionInterface> updated_answer2(
       CreateAnswer(NULL));
 
   CompareIceUfragAndPassword(updated_answer2->description(),
                              session_->local_description()->description(),
                              true);
 
   SetLocalDescriptionWithoutError(updated_answer2.release());
 }
 
 TEST_F(WebRtcSessionTest, TestSessionContentError) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
   const std::string session_id_orig = offer->session_id();
   const std::string session_version_orig = offer->session_version();
   SetLocalDescriptionWithoutError(offer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   video_channel_->set_fail_set_send_codecs(true);
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionAnswerExpectError("ERROR_CONTENT", answer);
 }
 
 // Runs the loopback call test with BUNDLE and STUN disabled.
 TEST_F(WebRtcSessionTest, TestIceStatesBasic) {
   // Lets try with only UDP ports.
   allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
                         cricket::PORTALLOCATOR_DISABLE_STUN |
@@ skipping 147 matching lines @@
 
   SessionDescriptionInterface* answer = ChangeSDPSctpPort(
       new_recv_port, CreateAnswer(NULL));
   ASSERT_TRUE(answer != NULL);
 
   // Now set the local description, which'll take ownership of the answer.
   SetLocalDescriptionWithoutError(answer);
 
   // TEST PLAN: Set the port number to something new, set it in the SDP,
   // and pass it all the way down.
-  webrtc::InternalDataChannelInit dci;
-  dci.reliable = true;
   EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
-  rtc::scoped_refptr<webrtc::DataChannel> dc =
-      session_->CreateDataChannel("datachannel", &dci);
+  CreateDataChannel();
 
   cricket::FakeDataMediaChannel* ch = data_engine_->GetChannel(0);
   int portnum = -1;
   ASSERT_TRUE(ch != NULL);
   ASSERT_EQ(1UL, ch->send_codecs().size());
   EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->send_codecs()[0].id);
   EXPECT_EQ(0, strcmp(cricket::kGoogleSctpDataCodecName,
                       ch->send_codecs()[0].name.c_str()));
   EXPECT_TRUE(ch->send_codecs()[0].GetParam(cricket::kCodecParamPort,
                                             &portnum));
   EXPECT_EQ(new_send_port, portnum);
 
   ASSERT_EQ(1UL, ch->recv_codecs().size());
   EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->recv_codecs()[0].id);
   EXPECT_EQ(0, strcmp(cricket::kGoogleSctpDataCodecName,
                       ch->recv_codecs()[0].name.c_str()));
   EXPECT_TRUE(ch->recv_codecs()[0].GetParam(cricket::kCodecParamPort,
                                             &portnum));
   EXPECT_EQ(new_recv_port, portnum);
 }
 
-TEST_F(WebRtcSessionTest, TestUsesProvidedCertificate) {
+// Verifies that when a session's DataChannel receives an OPEN message,
+// WebRtcSession signals the DataChannel creation request with the expected
+// config.
+TEST_P(WebRtcSessionTest, TestSctpDataChannelOpenMessage) {
+  MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
+  InitWithDtls(GetParam());
+
+  SetLocalDescriptionWithDataChannel();
+  EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
+
+  webrtc::DataChannelInit config;
+  config.id = 1;
+  rtc::Buffer payload;
+  webrtc::WriteDataChannelOpenMessage("a", config, &payload);
+  cricket::ReceiveDataParams params;
+  params.ssrc = config.id;
+  params.type = cricket::DMT_CONTROL;
+
+  cricket::DataChannel* data_channel = session_->data_channel();
+  data_channel->SignalDataReceived(data_channel, params, payload);
+
+  EXPECT_EQ("a", last_data_channel_label_);
+  EXPECT_EQ(config.id, last_data_channel_config_.id);
+  EXPECT_FALSE(last_data_channel_config_.negotiated);
+  EXPECT_EQ(webrtc::InternalDataChannelInit::kAcker,
+            last_data_channel_config_.open_handshake_role);
+}
+
+TEST_P(WebRtcSessionTest, TestUsesProvidedCertificate) {
   rtc::scoped_refptr<rtc::RTCCertificate> certificate =
       FakeDtlsIdentityStore::GenerateCertificate();
 
   PeerConnectionInterface::RTCConfiguration configuration;
   configuration.certificates.push_back(certificate);
   Init(nullptr, configuration);
   EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000);
 
   EXPECT_EQ(session_->certificate_for_testing(), certificate);
 }
 
 // Verifies that CreateOffer succeeds when CreateOffer is called before async
 // identity generation is finished (even if a certificate is provided this is
 // an async op).
 TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   InitWithDtls(GetParam());
 
   EXPECT_TRUE(session_->waiting_for_certificate_for_testing());
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   EXPECT_TRUE(offer != NULL);
   VerifyNoCryptoParams(offer->description(), true);
   VerifyFingerprintStatus(offer->description(), true);
 }
 
 // Verifies that CreateAnswer succeeds when CreateOffer is called before async
 // identity generation is finished (even if a certificate is provided this is
 // an async op).
@@ skipping 96 matching lines @@
   SetRemoteDescriptionOfferExpectError(kSdpWithoutSdesCrypto,
                                        offer);
 }
 
 // This test verifies DSCP is properly applied on the media channels.
 TEST_F(WebRtcSessionTest, TestDscpConstraint) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kEnableDscp, true);
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_TRUE(video_channel_ != NULL);
   ASSERT_TRUE(voice_channel_ != NULL);
   const cricket::AudioOptions& audio_options = voice_channel_->options();
   const cricket::VideoOptions& video_options = video_channel_->options();
   EXPECT_TRUE(audio_options.dscp.IsSet());
   EXPECT_TRUE(audio_options.dscp.GetWithDefaultIfUnset(false));
   EXPECT_TRUE(video_options.dscp.IsSet());
   EXPECT_TRUE(video_options.dscp.GetWithDefaultIfUnset(false));
 }
 
 TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
       true);
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
 
   ASSERT_TRUE(video_channel_ != NULL);
   const cricket::VideoOptions& video_options = video_channel_->options();
   EXPECT_TRUE(
       video_options.suspend_below_min_bitrate.GetWithDefaultIfUnset(false));
 }
 
 TEST_F(WebRtcSessionTest, TestNumUnsignalledRecvStreamsConstraint) {
   // Number of unsignalled receiving streams should be between 0 and
   // kMaxUnsignalledRecvStreams.
   SetAndVerifyNumUnsignalledRecvStreams(10, 10);
   SetAndVerifyNumUnsignalledRecvStreams(kMaxUnsignalledRecvStreams + 1,
                                         kMaxUnsignalledRecvStreams);
   SetAndVerifyNumUnsignalledRecvStreams(-1, 0);
 }
 
 TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe,
       true);
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
 
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_TRUE(voice_channel_ != NULL);
   const cricket::AudioOptions& audio_options = voice_channel_->options();
   EXPECT_TRUE(
       audio_options.combined_audio_video_bwe.GetWithDefaultIfUnset(false));
 }
 
 // Tests that we can renegotiate new media content with ICE candidates in the
 // new remote SDP.
 TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   InitWithDtls(GetParam());
   SetFactoryDtlsSrtp();
 
-  mediastream_signaling_.UseOptionsAudioOnly();
+  SendAudioOnlyStream2();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
 
   SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
   SetRemoteDescriptionWithoutError(answer);
 
   cricket::MediaSessionOptions options;
   options.recv_video = true;
   offer = CreateRemoteOffer(options, cricket::SEC_DISABLED);
 
   cricket::Candidate candidate1;
   candidate1.set_address(rtc::SocketAddress("1.1.1.1", 5000));
   candidate1.set_component(1);
   JsepIceCandidate ice_candidate(kMediaContentName1, kMediaContentIndex1,
                                  candidate1);
   EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
   SetRemoteDescriptionWithoutError(offer);
 
   answer = CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
 }
 
 // Tests that we can renegotiate new media content with ICE candidates separated
 // from the remote SDP.
 TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   InitWithDtls(GetParam());
   SetFactoryDtlsSrtp();
 
-  mediastream_signaling_.UseOptionsAudioOnly();
+  SendAudioOnlyStream2();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
 
   SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
   SetRemoteDescriptionWithoutError(answer);
 
   cricket::MediaSessionOptions options;
   options.recv_video = true;
   offer = CreateRemoteOffer(options, cricket::SEC_DISABLED);
   SetRemoteDescriptionWithoutError(offer);
 
   cricket::Candidate candidate1;
   candidate1.set_address(rtc::SocketAddress("1.1.1.1", 5000));
   candidate1.set_component(1);
   JsepIceCandidate ice_candidate(kMediaContentName1, kMediaContentIndex1,
                                  candidate1);
   EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate));
 
   answer = CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
 }
 // Tests that RTX codec is removed from the answer when it isn't supported
 // by local side.
 TEST_F(WebRtcSessionTest, TestRtxRemovedByCreateAnswer) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   std::string offer_sdp(kSdpWithRtx);
 
   SessionDescriptionInterface* offer =
       CreateSessionDescription(JsepSessionDescription::kOffer, offer_sdp, NULL);
   EXPECT_TRUE(offer->ToString(&offer_sdp));
 
   // Offer SDP contains the RTX codec.
   EXPECT_TRUE(offer_sdp.find("rtx") != std::string::npos);
   SetRemoteDescriptionWithoutError(offer);
 
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   std::string answer_sdp;
   answer->ToString(&answer_sdp);
   // Answer SDP removes the unsupported RTX codec.
   EXPECT_TRUE(answer_sdp.find("rtx") == std::string::npos);
   SetLocalDescriptionWithoutError(answer);
 }
 
 // This verifies that the voice channel after bundle has both options from video
 // and voice channels.
 TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
   session_->video_channel()->SetOption(cricket::BaseChannel::ST_RTP,
                                        rtc::Socket::Option::OPT_SNDBUF, 4000);
 
   session_->voice_channel()->SetOption(cricket::BaseChannel::ST_RTP,
                                        rtc::Socket::Option::OPT_RCVBUF, 8000);
 
   int option_val;
   EXPECT_TRUE(session_->video_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
   EXPECT_EQ(4000, option_val);
   EXPECT_FALSE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
 
   EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
   EXPECT_EQ(8000, option_val);
   EXPECT_FALSE(session_->video_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
 
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
   EXPECT_EQ(4000, option_val);
 
   EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
   EXPECT_EQ(8000, option_val);
 }
 
 // Test creating a session, request multiple offers, destroy the session
 // and make sure we got success/failure callbacks for all of the requests.
 // Background: crbug.com/507307
 TEST_F(WebRtcSessionTest, CreateOffersAndShutdown) {
   Init();
 
   rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observers[100];
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.offer_to_receive_audio =
       RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
+  cricket::MediaSessionOptions session_options;
+  session_options.recv_audio = true;
 
   for (auto& o : observers) {
     o = new WebRtcSessionCreateSDPObserverForTest();
-    session_->CreateOffer(o, options);
+    session_->CreateOffer(o, options, session_options);
   }
 
   session_.reset();
 
   for (auto& o : observers) {
     // We expect to have received a notification now even if the session was
     // terminated.  The offer creation may or may not have succeeded, but we
     // must have received a notification which, so the only invalid state
     // is kInit.
     EXPECT_NE(WebRtcSessionCreateSDPObserverForTest::kInit, o->state());
   }
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));