TransportController refactoring

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,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * 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/mediastreamsignaling.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"
@@ skipping 118 matching lines @@
     "a=rtpmap:96 rtx/90000\r\n"
     "a=fmtp:96 apt=0\r\n";
 
 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);
+  rtc::replace_substrs(line.c_str(), line.length(), tmp.c_str(), tmp.length(),
+                       message);
 }
 
 class MockIceObserver : public webrtc::IceObserver {
  public:
   MockIceObserver()
       : oncandidatesready_(false),
         ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
         ice_gathering_state_(PeerConnectionInterface::kIceGatheringNew) {
   }
 
@@ skipping 59 matching lines @@
                        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) {
     RegisterIceObserver(ice_observer);
   }
   virtual ~WebRtcSessionForTest() {}
 
-  using cricket::BaseSession::GetTransportProxy;
+  // 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());
+  }
+
+  cricket::TransportChannel* voice_rtcp_transport_channel() {
+    return rtcp_transport_channel(voice_channel());
+  }
+
+  cricket::TransportChannel* video_rtp_transport_channel() {
+    return rtp_transport_channel(video_channel());
+  }
+
+  cricket::TransportChannel* video_rtcp_transport_channel() {
+    return rtcp_transport_channel(video_channel());
+  }
+
+  cricket::TransportChannel* data_rtp_transport_channel() {
+    return rtp_transport_channel(data_channel());
+  }
+
+  cricket::TransportChannel* data_rtcp_transport_channel() {
+    return rtcp_transport_channel(data_channel());
+  }
+
   using webrtc::WebRtcSession::SetAudioPlayout;
   using webrtc::WebRtcSession::SetAudioSend;
   using webrtc::WebRtcSession::SetCaptureDevice;
   using webrtc::WebRtcSession::SetVideoPlayout;
   using webrtc::WebRtcSession::SetVideoSend;
+
+ private:
+  cricket::TransportChannel* rtp_transport_channel(cricket::BaseChannel* ch) {
+    if (!ch) {
+      return nullptr;
+    }
+    return ch->transport_channel();
+  }
+
+  cricket::TransportChannel* rtcp_transport_channel(cricket::BaseChannel* ch) {
+    if (!ch) {
+      return nullptr;
+    }
+    return ch->rtcp_transport_channel();
+  }
 };
 
 class WebRtcSessionCreateSDPObserverForTest
     : public rtc::RefCountedObject<CreateSessionDescriptionObserver> {
  public:
   enum State {
     kInit,
     kFailed,
     kSucceeded,
   };
@@ skipping 105 matching lines @@
         channel_manager_.get(), rtc::Thread::Current(),
         rtc::Thread::Current(), allocator_.get(),
         &observer_,
         &mediastream_signaling_));
 
     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));
+    EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
+                                     dtls_identity_store.Pass(),
+                                     rtc_configuration));
     session_->set_metrics_observer(metrics_observer_);
   }
 
   void Init() {
     PeerConnectionInterface::RTCConfiguration configuration;
     Init(nullptr, configuration);
   }
 
   void InitWithIceTransport(
       PeerConnectionInterface::IceTransportsType ice_transport_type) {
@@ skipping 92 matching lines @@
         observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
         2000);
     return observer->ReleaseDescription();
   }
 
   bool ChannelsExist() const {
     return (session_->voice_channel() != NULL &&
             session_->video_channel() != NULL);
   }
 
-  void CheckTransportChannels() const {
-    EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 1) != NULL);
-    EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 2) != NULL);
-    EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 1) != NULL);
-    EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 2) != NULL);
-  }
-
   void VerifyCryptoParams(const cricket::SessionDescription* sdp) {
     ASSERT_TRUE(session_.get() != NULL);
     const cricket::ContentInfo* content = cricket::GetFirstAudioContent(sdp);
     ASSERT_TRUE(content != NULL);
     const cricket::AudioContentDescription* audio_content =
         static_cast<const cricket::AudioContentDescription*>(
             content->description);
     ASSERT_TRUE(audio_content != NULL);
     ASSERT_EQ(1U, audio_content->cryptos().size());
     ASSERT_EQ(47U, audio_content->cryptos()[0].key_params.size());
@@ skipping 452 matching lines @@
                                  &sdp);
     }
 
     SessionDescriptionInterface* new_answer = CreateSessionDescription(
         JsepSessionDescription::kAnswer, sdp, NULL);
 
     // SetRemoteDescription to enable rtcp mux.
     SetRemoteDescriptionWithoutError(new_answer);
     EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
     EXPECT_EQ(expected_candidate_num, observer_.mline_0_candidates_.size());
-    EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
-    for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
-      cricket::Candidate c0 = observer_.mline_0_candidates_[i];
-      cricket::Candidate c1 = observer_.mline_1_candidates_[i];
-      if (bundle) {
-        EXPECT_TRUE(c0.IsEquivalent(c1));
-      } else {
-        EXPECT_FALSE(c0.IsEquivalent(c1));
-      }
+    if (bundle) {
+      EXPECT_EQ(0, observer_.mline_1_candidates_.size());
+    } else {
+      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();
     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
   // verifies the best connection is using the right IP protocol after
   // initial ICE convergences.
 
   class LoopbackNetworkConfiguration {
-  public:
+   public:
     LoopbackNetworkConfiguration()
         : test_ipv6_network_(false),
           test_extra_ipv4_network_(false),
           best_connection_after_initial_ice_converged_(1, 0) {}
 
     // Used to track the expected best connection count in each IP protocol.
     struct ExpectedBestConnection {
       ExpectedBestConnection(int ipv4_count, int ipv6_count)
           : ipv4_count_(ipv4_count),
             ipv6_count_(ipv6_count) {}
@@ skipping 128 matching lines @@
 
     LOG(LS_INFO) << "Firewall Rules applied";
     EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
                    observer_.ice_connection_state_,
                    kIceCandidatesTimeout);
 
     metrics_observer_->Reset();
 
     // Clearing the rules, session should move back to completed state.
     loopback_network_manager.ClearRules(fss_.get());
-    // Session is automatically calling OnSignalingReady after creation of
-    // new portallocator session which will allocate new set of candidates.
 
     LOG(LS_INFO) << "Firewall Rules cleared";
-
     EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                    observer_.ice_connection_state_,
                    kIceCandidatesTimeout);
 
     // Now we block ping requests and wait until the ICE connection transitions
     // to the Failed state.  This will take at least 30 seconds because it must
     // wait for the Port to timeout.
     int port_timeout = 30000;
 
     loopback_network_manager.ApplyFirewallRules(fss_.get());
@@ skipping 49 matching lines @@
     VerifyMultipleAsyncCreateDescriptionAfterInit(true, type);
   }
 
   void VerifyMultipleAsyncCreateDescriptionIdentityGenFailure(
       CreateSessionDescriptionRequest::Type type) {
     InitWithDtlsIdentityGenFail();
     VerifyMultipleAsyncCreateDescriptionAfterInit(false, type);
   }
 
   void VerifyMultipleAsyncCreateDescriptionAfterInit(
-      bool success, CreateSessionDescriptionRequest::Type type) {
+      bool success,
+      CreateSessionDescriptionRequest::Type type) {
     CHECK(session_);
     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());
     }
 
@@ skipping 462 matching lines @@
   // the offer and answer.
   SetRemoteDescriptionWithoutError(offer);
   SetLocalDescriptionAnswerExpectError(
       kSdpWithoutDtlsFingerprint, answer);
 }
 
 // Test that we return a failure when applying a remote answer that doesn't have
 // a DTLS fingerprint when DTLS is required.
 TEST_P(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  // Enable both SDES and DTLS, so that offer won't be outright rejected as a
-  // result of using the "UDP/TLS/RTP/SAVPF" profile.
   InitWithDtls(GetParam());
-  session_->SetSdesPolicy(cricket::SEC_ENABLED);
   SessionDescriptionInterface* offer = CreateOffer();
   cricket::MediaSessionOptions options;
   options.recv_video = true;
+  rtc::scoped_ptr<SessionDescriptionInterface> temp_offer(
+      CreateRemoteOffer(options, cricket::SEC_ENABLED));
   JsepSessionDescription* answer =
-      CreateRemoteAnswer(offer, options, cricket::SEC_ENABLED);
+      CreateRemoteAnswer(temp_offer.get(), options, cricket::SEC_ENABLED);
 
   // 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.
@@ skipping 281 matching lines @@
   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_LT(0u, candidates->count());
+  EXPECT_EQ(0u, candidates->count());
 
   // Update the session descriptions.
   mediastream_signaling_.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_LT(0u, candidates->count());
+  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);
@@ skipping 23 matching lines @@
   Init();
   mediastream_signaling_.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);
-  EXPECT_TRUE_WAIT(0u < observer_.mline_1_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());
-  ASSERT_TRUE(local_offer->candidates(kMediaContentIndex1) != NULL);
-  EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex1)->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());
-  ASSERT_TRUE(answer->candidates(kMediaContentIndex1) != NULL);
-  EXPECT_LT(0u, answer->candidates(kMediaContentIndex1)->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();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
@@ skipping 21 matching lines @@
 
   SessionDescriptionInterface* modified_offer =
       CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
 
   SetRemoteDescriptionWithoutError(modified_offer);
 
   SessionDescriptionInterface* answer =
       CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
 
-  EXPECT_TRUE(session_->GetTransportProxy("audio_content_name") != NULL);
-  EXPECT_TRUE(session_->GetTransportProxy("video_content_name") != NULL);
+  cricket::TransportChannel* voice_transport_channel =
+      session_->voice_rtp_transport_channel();
+  EXPECT_TRUE(voice_transport_channel != NULL);
+  EXPECT_EQ(voice_transport_channel->transport_name(), "audio_content_name");
+  cricket::TransportChannel* video_transport_channel =
+      session_->video_rtp_transport_channel();
+  EXPECT_TRUE(video_transport_channel != NULL);
+  EXPECT_EQ(video_transport_channel->transport_name(), "video_content_name");
   EXPECT_TRUE((video_channel_ = media_engine_->GetVideoChannel(0)) != NULL);
   EXPECT_TRUE((voice_channel_ = media_engine_->GetVoiceChannel(0)) != 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, CreateOfferWithoutConstraintsOrStreams) {
   Init();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
@@ skipping 539 matching lines @@
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateRemoteOffer();
   SetRemoteDescriptionWithoutError(offer);
 
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
-  cricket::Transport* t = session_->GetTransport("audio");
+  cricket::BaseChannel* voice_channel = session_->voice_channel();
+  ASSERT(voice_channel != NULL);
 
   // Checks if one of the transport channels contains a connection using a given
   // port.
-  auto connection_with_remote_port = [t](int port) {
-    cricket::TransportStats stats;
-    t->GetStats(&stats);
-    for (auto& chan_stat : stats.channel_stats) {
-      for (auto& conn_info : chan_stat.connection_infos) {
-        if (conn_info.remote_candidate.address().port() == port) {
-          return true;
+  auto connection_with_remote_port = [this, voice_channel](int port) {
+    cricket::SessionStats stats;
+    session_->GetChannelTransportStats(voice_channel, &stats);
+    for (auto& kv : stats.transport_stats) {
+      for (auto& chan_stat : kv.second.channel_stats) {
+        for (auto& conn_info : chan_stat.connection_infos) {
+          if (conn_info.remote_candidate.address().port() == port) {
+            return true;
+          }
         }
       }
     }
     return false;
   };
 
   EXPECT_FALSE(connection_with_remote_port(5000));
   EXPECT_FALSE(connection_with_remote_port(5001));
   EXPECT_FALSE(connection_with_remote_port(6000));
 
@@ skipping 32 matching lines @@
 
   EXPECT_TRUE_WAIT(connection_with_remote_port(5000), 1000);
   EXPECT_TRUE_WAIT(connection_with_remote_port(5001), 1000);
 
   // 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.
+// kBundlePolicyBalanced BUNDLE policy and answer contains BUNDLE.
 TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
   mediastream_signaling_.SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 }
 
-// kBundlePolicyBalanced bundle policy but no BUNDLE in the answer.
+// kBundlePolicyBalanced BUNDLE policy but no BUNDLE in the answer.
 TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
   mediastream_signaling_.SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.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_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  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();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  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();
+
+  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();
+  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();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.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_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  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();
+
+  SessionDescriptionInterface* offer = CreateRemoteOffer();
+  SetRemoteDescriptionWithoutError(offer);
+
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
+
+  mediastream_signaling_.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();
+
+  // 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();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.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_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 }
 
-// kBundlePolicyMaxCompat bundle policy but no BUNDLE in the answer.
+// kBundlePolicyMaxCompat BUNDLE policy but no BUNDLE in the answer.
 TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxCompat);
   mediastream_signaling_.SendAudioVideoStream1();
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.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_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  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();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
 
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetRemoteDescriptionWithoutError(offer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 }
 
 TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
   InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyRequire);
   mediastream_signaling_.SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
-  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+  EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL);
+  EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL);
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
-  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+  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();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_TRUE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
-  EXPECT_TRUE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+  EXPECT_TRUE(session_->voice_rtcp_transport_channel() != NULL);
+  EXPECT_TRUE(session_->video_rtcp_transport_channel() != NULL);
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
-  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+  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();
 
   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 =
+  JsepSessionDescription* local_offer =
       new JsepSessionDescription(JsepSessionDescription::kOffer);
   EXPECT_TRUE((local_offer)->Initialize(offer_str, NULL));
   SetLocalDescriptionOfferExpectError(kBundleWithoutRtcpMux, local_offer);
-  JsepSessionDescription *remote_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();
@@ skipping 242 matching lines @@
   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_->GetTransportProxy("audio") != NULL);
-  ASSERT_TRUE(session_->GetTransportProxy("video") != NULL);
+  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(session_->GetTransportProxy("audio")->negotiated());
-  EXPECT_TRUE(session_->GetTransportProxy("video")->negotiated());
   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();
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
@@ skipping 14 matching lines @@
   options_.disable_encryption = true;
   Init();
   mediastream_signaling_.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 =
+  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();
@@ skipping 332 matching lines @@
 
   rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
   EXPECT_TRUE(offer == NULL);
 }
 
 // Verifies that CreateOffer succeeds when Multiple CreateOffer calls are made
 // before async identity generation is finished.
 TEST_P(WebRtcSessionTest,
        TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  VerifyMultipleAsyncCreateDescription(
-      GetParam(), CreateSessionDescriptionRequest::kOffer);
+  VerifyMultipleAsyncCreateDescription(GetParam(),
+                                       CreateSessionDescriptionRequest::kOffer);
 }
 
 // Verifies that CreateOffer fails when Multiple CreateOffer calls are made
 // before async identity generation fails.
 TEST_F(WebRtcSessionTest,
        TestMultipleCreateOfferBeforeIdentityRequestReturnFailure) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   VerifyMultipleAsyncCreateDescriptionIdentityGenFailure(
       CreateSessionDescriptionRequest::kOffer);
 }
@@ skipping 206 matching lines @@
   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_channel()->transport_channel()->GetOption(
+  EXPECT_TRUE(session_->video_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
   EXPECT_EQ(4000, option_val);
-  EXPECT_FALSE(session_->voice_channel()->transport_channel()->GetOption(
+  EXPECT_FALSE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
 
-  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+  EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
   EXPECT_EQ(8000, option_val);
-  EXPECT_FALSE(session_->video_channel()->transport_channel()->GetOption(
+  EXPECT_FALSE(session_->video_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
 
-  EXPECT_NE(session_->voice_channel()->transport_channel(),
-            session_->video_channel()->transport_channel());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+  EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
   EXPECT_EQ(4000, option_val);
 
-  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+  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();
 
@@ skipping 15 matching lines @@
     // 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));
+INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
+                        WebRtcSessionTest,
+                        testing::Values(ALREADY_GENERATED,
+                                        DTLS_IDENTITY_STORE));