Reland of Moving MediaStreamSignaling logic into PeerConnection.

talk/app/webrtc/webrtcsession_unittest.cc

Index: talk/app/webrtc/webrtcsession_unittest.cc
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index a463ba90bbecfddb81f8addb046b1fb63e9f3713..f0558eee3d45c7c25fe9b755960b7030f9ca9a22 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -31,11 +31,13 @@
 #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"
@@ -79,10 +81,12 @@ 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;
@@ -157,6 +161,14 @@ static const char kSdpWithRtx[] =
     "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|.
@@ -237,10 +249,8 @@ class WebRtcSessionForTest : public webrtc::WebRtcSession {
                        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() {}
@@ -344,7 +354,8 @@ class FakeAudioRenderer : public cricket::AudioRenderer {
 };
 
 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.
@@ -366,7 +377,6 @@ class WebRtcSessionTest
       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_);
@@ -395,10 +405,10 @@ class WebRtcSessionTest
       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_->SignalDataChannelOpenMessage.connect(
+        this, &WebRtcSessionTest::OnDataChannelOpenMessage);
 
     EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
         observer_.ice_connection_state_);
@@ -411,6 +421,12 @@ class WebRtcSessionTest
     session_->set_metrics_observer(metrics_observer_);
   }
 
+  void OnDataChannelOpenMessage(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);
@@ -474,8 +490,97 @@ class WebRtcSessionTest
     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();
@@ -497,7 +602,9 @@ class WebRtcSessionTest
       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);
@@ -508,7 +615,9 @@ class WebRtcSessionTest
       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);
@@ -638,7 +747,7 @@ class WebRtcSessionTest
         value_set);
     session_.reset();
     Init();
-    mediastream_signaling_.SendAudioVideoStream1();
+    SendAudioVideoStream1();
     SessionDescriptionInterface* offer = CreateOffer();
 
     SetLocalDescriptionWithoutError(offer);
@@ -735,7 +844,7 @@ class WebRtcSessionTest
 
   // 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();
@@ -911,11 +1020,11 @@ class WebRtcSessionTest
     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());
   }
 
@@ -943,21 +1052,20 @@ class WebRtcSessionTest
       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;
@@ -1005,7 +1113,7 @@ class WebRtcSessionTest
     } else {
       Init();
     }
-    mediastream_signaling_.SendAudioVideoStream1();
+    SendAudioVideoStream1();
     CreateAndSetRemoteOfferAndLocalAnswer();
     EXPECT_FALSE(session_->CanInsertDtmf(""));
     EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
@@ -1121,7 +1229,7 @@ class WebRtcSessionTest
   void TestLoopbackCall(const LoopbackNetworkConfiguration& config) {
     LoopbackNetworkManager loopback_network_manager(this, config);
     Init();
-    mediastream_signaling_.SendAudioVideoStream1();
+    SendAudioVideoStream1();
     SessionDescriptionInterface* offer = CreateOffer();
 
     EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
@@ -1218,10 +1326,15 @@ class WebRtcSessionTest
     return true;
   }
 
-  void SetLocalDescriptionWithDataChannel() {
+  void CreateDataChannel() {
     webrtc::InternalDataChannelInit dci;
-    dci.reliable = false;
-    session_->CreateDataChannel("datachannel", &dci);
+    dci.reliable = session_->data_channel_type() == cricket::DCT_SCTP;
+    data_channel_ = DataChannel::Create(
+        session_.get(), session_->data_channel_type(), "datachannel", dci);
+  }
+
+  void SetLocalDescriptionWithDataChannel() {
+    CreateDataChannel();
     SessionDescriptionInterface* offer = CreateOffer();
     SetLocalDescriptionWithoutError(offer);
   }
@@ -1252,15 +1365,16 @@ class WebRtcSessionTest
     }
 
     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);
       }
     }
 
@@ -1305,12 +1419,20 @@ class WebRtcSessionTest
   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) {
@@ -1343,7 +1465,7 @@ 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());
@@ -1358,7 +1480,7 @@ TEST_F(WebRtcSessionTest, TestStunError) {
                 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);
@@ -1370,7 +1492,7 @@ TEST_F(WebRtcSessionTest, TestStunError) {
 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());
@@ -1383,7 +1505,7 @@ 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());
@@ -1402,7 +1524,7 @@ TEST_F(WebRtcSessionTest, TestIceTransportsRelay) {
 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.
@@ -1423,13 +1545,13 @@ TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
 // 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);
@@ -1449,7 +1571,7 @@ TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
   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
@@ -1462,7 +1584,7 @@ TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
   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);
 
@@ -1477,12 +1599,12 @@ TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
 // 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);
@@ -1504,12 +1626,12 @@ TEST_F(WebRtcSessionTest, TestReceiveSdesOfferCreateSdesAnswer) {
   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
@@ -1623,7 +1745,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteNonSdesAnswerWhenSdesOn) {
 // 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;
@@ -1652,7 +1774,7 @@ TEST_P(WebRtcSessionTest, TestReceiveDtlsOfferCreateDtlsAnswer) {
 // 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();
 
@@ -1741,7 +1863,7 @@ TEST_P(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) {
 // 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());
 
@@ -1797,7 +1919,7 @@ TEST_P(WebRtcSessionTest, TestCreateAnswerReceiveOfferWithoutEncryption) {
 
 TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   // SetLocalDescription take ownership of offer.
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
@@ -1809,7 +1931,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
 
 TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   // SetLocalDescription take ownership of offer.
   SessionDescriptionInterface* offer = CreateOffer();
   SetRemoteDescriptionWithoutError(offer);
@@ -1820,7 +1942,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
 
 TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
   offer = CreateOffer();
@@ -1830,7 +1952,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
 
 TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   SessionDescriptionInterface* offer = CreateOffer();
   SetRemoteDescriptionWithoutError(offer);
   offer = CreateOffer();
@@ -1840,7 +1962,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
 
 TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   SetRemoteDescriptionExpectState(offer, BaseSession::STATE_RECEIVEDINITIATE);
 
@@ -1849,21 +1971,21 @@ TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
   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);
 
@@ -1874,7 +1996,7 @@ TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
   SetRemoteDescriptionExpectState(pranswer,
                                   BaseSession::STATE_RECEIVEDPRACCEPT);
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   JsepSessionDescription* pranswer2 =
       CreateRemoteAnswer(session_->local_description());
   pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
@@ -1882,7 +2004,7 @@ TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
   SetRemoteDescriptionExpectState(pranswer2,
                                   BaseSession::STATE_RECEIVEDPRACCEPT);
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionExpectState(answer, BaseSession::STATE_RECEIVEDACCEPT);
@@ -1890,7 +2012,7 @@ TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
 
 TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   SessionDescriptionInterface* answer =
@@ -1901,7 +2023,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
 
 TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
   Init();
-  mediastream_signaling_.SendNothing();
+  SendNothing();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   SessionDescriptionInterface* answer =
@@ -1912,7 +2034,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
 
 TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   cricket::Candidate candidate;
   candidate.set_component(1);
@@ -1965,7 +2087,7 @@ TEST_F(WebRtcSessionTest, TestRemoteCandidatesAddedToSessionDescription) {
   candidate1.set_component(1);
   JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
                                   candidate1);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
@@ -2014,7 +2136,7 @@ TEST_F(WebRtcSessionTest, TestRemoteCandidatesAddedToSessionDescription) {
 TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   const SessionDescriptionInterface* local_desc = session_->local_description();
@@ -2034,7 +2156,7 @@ TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
   EXPECT_EQ(0u, candidates->count());
 
   // Update the session descriptions.
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
 
   local_desc = session_->local_description();
@@ -2054,7 +2176,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
   candidate1.set_component(1);
   JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
                                  candidate1);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
@@ -2078,7 +2200,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
 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());
@@ -2105,7 +2227,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
 // 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
@@ -2170,7 +2292,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
 TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
   Init();
   // Test Audio only offer.
-  mediastream_signaling_.UseOptionsAudioOnly();
+  SendAudioOnlyStream2();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   const cricket::ContentInfo* content =
@@ -2180,7 +2302,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
   EXPECT_TRUE(content == NULL);
 
   // Test Audio / Video offer.
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   offer.reset(CreateOffer());
   content = cricket::GetFirstAudioContent(offer->description());
   EXPECT_TRUE(content != NULL);
@@ -2319,7 +2441,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
   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 =
@@ -2369,7 +2491,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
   constraints_no_receive.SetMandatoryReceiveVideo(false);
 
   // Test with a stream with tracks.
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateAnswer(&constraints_no_receive));
 
@@ -2426,7 +2548,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
   EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
   EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   cricket::MediaSessionOptions options;
@@ -2447,7 +2569,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
   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);
@@ -2466,7 +2588,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
   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());
@@ -2482,7 +2604,7 @@ 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;
@@ -2507,7 +2629,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
   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);
@@ -2519,7 +2641,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
   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);
@@ -2533,7 +2655,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
 
 TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
   VerifyCryptoParams(offer->description());
   SetRemoteDescriptionWithoutError(offer.release());
@@ -2544,7 +2666,7 @@ TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
 TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
   options_.disable_encryption = true;
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
   VerifyNoCryptoParams(offer->description(), false);
 }
@@ -2563,7 +2685,7 @@ TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) {
 // 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;
@@ -2589,7 +2711,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
 // 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;
@@ -2695,7 +2817,7 @@ TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) {
   AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
 
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
@@ -2778,7 +2900,7 @@ TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) {
 // 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;
@@ -2789,7 +2911,7 @@ TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) {
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
@@ -2801,7 +2923,7 @@ TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) {
 // 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;
@@ -2812,7 +2934,7 @@ TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) {
   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(
@@ -2831,7 +2953,7 @@ TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) {
 // 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;
@@ -2842,7 +2964,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleBundleInAnswer) {
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
@@ -2855,7 +2977,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleBundleInAnswer) {
 // audio content in the answer.
 TEST_F(WebRtcSessionTest, TestMaxBundleRejectAudio) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
@@ -2866,7 +2988,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleRejectAudio) {
   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;
@@ -2887,7 +3009,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleRejectAudio) {
 // 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;
@@ -2898,7 +3020,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInAnswer) {
   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(
@@ -2917,7 +3039,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInAnswer) {
 // kBundlePolicyMaxBundle policy with BUNDLE in the remote offer.
 TEST_F(WebRtcSessionTest, TestMaxBundleBundleInRemoteOffer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   SetRemoteDescriptionWithoutError(offer);
@@ -2925,7 +3047,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleBundleInRemoteOffer) {
   EXPECT_EQ(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer = CreateAnswer(nullptr);
   SetLocalDescriptionWithoutError(answer);
 
@@ -2936,7 +3058,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleBundleInRemoteOffer) {
 // 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());
@@ -2954,7 +3076,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInRemoteOffer) {
 // 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;
@@ -2965,7 +3087,7 @@ TEST_F(WebRtcSessionTest, TestMaxCompatBundleInAnswer) {
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
@@ -2979,7 +3101,7 @@ TEST_F(WebRtcSessionTest, TestMaxCompatBundleInAnswer) {
 // 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;
 
@@ -2989,7 +3111,7 @@ TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) {
   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(
@@ -3008,7 +3130,7 @@ TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) {
 // 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;
@@ -3022,7 +3144,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleWithSetRemoteDescriptionFirst) {
 
 TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
   InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyRequire);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   SessionDescriptionInterface* offer = CreateOffer(options);
@@ -3031,7 +3153,7 @@ TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
   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);
@@ -3042,7 +3164,7 @@ TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
 
 TEST_F(WebRtcSessionTest, TestNegotiateRtcpMux) {
   InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyNegotiate);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   SessionDescriptionInterface* offer = CreateOffer(options);
@@ -3051,7 +3173,7 @@ TEST_F(WebRtcSessionTest, TestNegotiateRtcpMux) {
   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);
@@ -3064,7 +3186,7 @@ TEST_F(WebRtcSessionTest, TestNegotiateRtcpMux) {
 // 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;
@@ -3092,7 +3214,7 @@ TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
 
 TEST_F(WebRtcSessionTest, SetAudioPlayout) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
@@ -3111,7 +3233,7 @@ TEST_F(WebRtcSessionTest, SetAudioPlayout) {
 
 TEST_F(WebRtcSessionTest, SetAudioSend) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
@@ -3139,7 +3261,7 @@ TEST_F(WebRtcSessionTest, SetAudioSend) {
 
 TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
@@ -3162,7 +3284,7 @@ TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
 
 TEST_F(WebRtcSessionTest, SetVideoPlayout) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
   ASSERT_TRUE(channel != NULL);
@@ -3179,7 +3301,7 @@ TEST_F(WebRtcSessionTest, SetVideoPlayout) {
 
 TEST_F(WebRtcSessionTest, SetVideoSend) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
   ASSERT_TRUE(channel != NULL);
@@ -3204,7 +3326,7 @@ TEST_F(WebRtcSessionTest, CanInsertDtmf) {
 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());
@@ -3255,7 +3377,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) {
 // 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(
@@ -3303,7 +3425,7 @@ TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
 // RFC 3264.
 TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   SetRemoteDescriptionWithoutError(offer);
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
@@ -3324,7 +3446,7 @@ TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
 // before SetLocalDescription is called.
 TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   cricket::Candidate candidate;
   candidate.set_component(1);
@@ -3354,7 +3476,7 @@ TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
 // 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
@@ -3373,7 +3495,7 @@ TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
 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
@@ -3398,7 +3520,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
       CreateRemoteOffer(options));
   SetRemoteDescriptionWithoutError(offer.release());
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateAnswer(NULL));
   SetLocalDescriptionWithoutError(answer.release());
@@ -3429,7 +3551,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
       CreateRemoteOffer(options));
   SetRemoteDescriptionWithoutError(offer.release());
 
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> answer(
       CreateAnswer(NULL));
   SetLocalDescriptionWithoutError(answer.release());
@@ -3452,7 +3574,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
 
 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();
@@ -3461,7 +3583,7 @@ TEST_F(WebRtcSessionTest, TestSessionContentError) {
   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);
@@ -3629,11 +3751,8 @@ TEST_P(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
 
   // 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;
@@ -3655,7 +3774,36 @@ TEST_P(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
   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();
 
@@ -3675,7 +3823,7 @@ TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
   InitWithDtls(GetParam());
 
   EXPECT_TRUE(session_->waiting_for_certificate_for_testing());
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   EXPECT_TRUE(offer != NULL);
@@ -3792,7 +3940,7 @@ TEST_F(WebRtcSessionTest, TestDscpConstraint) {
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kEnableDscp, true);
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
@@ -3816,7 +3964,7 @@ TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) {
       webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
       true);
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
@@ -3844,7 +3992,7 @@ TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) {
       webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe,
       true);
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
@@ -3864,7 +4012,7 @@ TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
   InitWithDtls(GetParam());
   SetFactoryDtlsSrtp();
 
-  mediastream_signaling_.UseOptionsAudioOnly();
+  SendAudioOnlyStream2();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
 
@@ -3894,7 +4042,7 @@ TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) {
   InitWithDtls(GetParam());
   SetFactoryDtlsSrtp();
 
-  mediastream_signaling_.UseOptionsAudioOnly();
+  SendAudioOnlyStream2();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
 
@@ -3920,7 +4068,7 @@ TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) {
 // by local side.
 TEST_F(WebRtcSessionTest, TestRtxRemovedByCreateAnswer) {
   Init();
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
   std::string offer_sdp(kSdpWithRtx);
 
   SessionDescriptionInterface* offer =
@@ -3943,7 +4091,7 @@ TEST_F(WebRtcSessionTest, TestRtxRemovedByCreateAnswer) {
 // and voice channels.
 TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
-  mediastream_signaling_.SendAudioVideoStream1();
+  SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
@@ -3973,7 +4121,7 @@ TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
   EXPECT_NE(session_->voice_rtp_transport_channel(),
             session_->video_rtp_transport_channel());
 
-  mediastream_signaling_.SendAudioVideoStream2();
+  SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
@@ -3997,10 +4145,12 @@ TEST_F(WebRtcSessionTest, CreateOffersAndShutdown) {
   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();