Remove enable_rtp_data_channel

This denies the ability to request RTP data channels to callers.
Later CLs will rip out the actual code for creating these channels.

Bug: chromium:928706
Change-Id: Ibb54197f192f567984a348f1539c26be120903f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177901
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33740}

api/peer_connection_interface.h

@@ -404,12 +404,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // from consideration for gathering ICE candidates.
     bool disable_link_local_networks = false;
 
-    // If set to true, use RTP data channels instead of SCTP.
-    // TODO(deadbeef): Remove this. We no longer commit to supporting RTP data
-    // channels, though some applications are still working on moving off of
-    // them.
-    bool enable_rtp_data_channel = false;
-
     // Minimum bitrate at which screencast video tracks will be encoded at.
     // This means adding padding bits up to this bitrate, which can help
     // when switching from a static scene to one with motion.

examples/unityplugin/simple_peer_connection.cc

@@ -190,7 +190,6 @@ bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
   webrtc::PeerConnectionInterface::IceServer stun_server;
   stun_server.uri = GetPeerConnectionString();
   config_.servers.push_back(stun_server);
-  config_.enable_rtp_data_channel = true;
   config_.enable_dtls_srtp = false;
 
   peer_connection_ = g_peer_connection_factory->CreatePeerConnection(

pc/data_channel_integrationtest.cc

@@ -146,204 +146,6 @@ TEST_P(DataChannelIntegrationTest, DataChannelWhileDisconnectedIceRestart) {
                  kDefaultTimeout);
 }
 
-#endif  // WEBRTC_HAVE_SCTP
-
-// This test sets up a call between two parties with audio, video and an RTP
-// data channel.
-TEST_P(DataChannelIntegrationTest, EndToEndCallWithRtpDataChannel) {
-  PeerConnectionInterface::RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  rtc_config.enable_dtls_srtp = false;
-  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
-  ConnectFakeSignaling();
-  // Expect that data channel created on caller side will show up for callee as
-  // well.
-  caller()->CreateDataChannel();
-  caller()->AddAudioVideoTracks();
-  callee()->AddAudioVideoTracks();
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  // Ensure the existence of the RTP data channel didn't impede audio/video.
-  MediaExpectations media_expectations;
-  media_expectations.ExpectBidirectionalAudioAndVideo();
-  ASSERT_TRUE(ExpectNewFrames(media_expectations));
-  ASSERT_NE(nullptr, caller()->data_channel());
-  ASSERT_NE(nullptr, callee()->data_channel());
-  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
-  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
-
-  // Ensure data can be sent in both directions.
-  std::string data = "hello world";
-  SendRtpDataWithRetries(caller()->data_channel(), data, 5);
-  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
-                 kDefaultTimeout);
-  SendRtpDataWithRetries(callee()->data_channel(), data, 5);
-  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
-                 kDefaultTimeout);
-}
-
-TEST_P(DataChannelIntegrationTest, RtpDataChannelWorksAfterRollback) {
-  PeerConnectionInterface::RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  rtc_config.enable_dtls_srtp = false;
-  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
-  ConnectFakeSignaling();
-  auto data_channel = caller()->pc()->CreateDataChannel("label_1", nullptr);
-  ASSERT_TRUE(data_channel.get() != nullptr);
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
-  caller()->CreateDataChannel("label_2", nullptr);
-  rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
-      new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
-  caller()->pc()->SetLocalDescription(observer,
-                                      caller()->CreateOfferAndWait().release());
-  EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
-  caller()->Rollback();
-
-  std::string data = "hello world";
-  SendRtpDataWithRetries(data_channel, data, 5);
-  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
-                 kDefaultTimeout);
-}
-
-// Ensure that an RTP data channel is signaled as closed for the caller when
-// the callee rejects it in a subsequent offer.
-TEST_P(DataChannelIntegrationTest, RtpDataChannelSignaledClosedInCalleeOffer) {
-  // Same procedure as above test.
-  PeerConnectionInterface::RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  rtc_config.enable_dtls_srtp = false;
-  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
-  ConnectFakeSignaling();
-  caller()->CreateDataChannel();
-  caller()->AddAudioVideoTracks();
-  callee()->AddAudioVideoTracks();
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  ASSERT_NE(nullptr, caller()->data_channel());
-  ASSERT_NE(nullptr, callee()->data_channel());
-  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
-  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
-
-  // Close the data channel on the callee, and do an updated offer/answer.
-  callee()->data_channel()->Close();
-  callee()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  EXPECT_FALSE(caller()->data_observer()->IsOpen());
-  EXPECT_FALSE(callee()->data_observer()->IsOpen());
-}
-
-#if !defined(THREAD_SANITIZER)
-// This test provokes TSAN errors. See bugs.webrtc.org/11282
-
-// Tests that data is buffered in an RTP data channel until an observer is
-// registered for it.
-//
-// NOTE: RTP data channels can receive data before the underlying
-// transport has detected that a channel is writable and thus data can be
-// received before the data channel state changes to open. That is hard to test
-// but the same buffering is expected to be used in that case.
-//
-// Use fake clock and simulated network delay so that we predictably can wait
-// until an SCTP message has been delivered without "sleep()"ing.
-TEST_P(DataChannelIntegrationTestWithFakeClock,
-       DataBufferedUntilRtpDataChannelObserverRegistered) {
-  virtual_socket_server()->set_delay_mean(5);  // 5 ms per hop.
-  virtual_socket_server()->UpdateDelayDistribution();
-
-  PeerConnectionInterface::RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  rtc_config.enable_dtls_srtp = false;
-  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
-  ConnectFakeSignaling();
-  caller()->CreateDataChannel();
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE(caller()->data_channel() != nullptr);
-  ASSERT_TRUE_SIMULATED_WAIT(callee()->data_channel() != nullptr,
-                             kDefaultTimeout, FakeClock());
-  ASSERT_TRUE_SIMULATED_WAIT(caller()->data_observer()->IsOpen(),
-                             kDefaultTimeout, FakeClock());
-  ASSERT_EQ_SIMULATED_WAIT(DataChannelInterface::kOpen,
-                           callee()->data_channel()->state(), kDefaultTimeout,
-                           FakeClock());
-
-  // Unregister the observer which is normally automatically registered.
-  callee()->data_channel()->UnregisterObserver();
-  // Send data and advance fake clock until it should have been received.
-  std::string data = "hello world";
-  caller()->data_channel()->Send(DataBuffer(data));
-  SIMULATED_WAIT(false, 50, FakeClock());
-
-  // Attach data channel and expect data to be received immediately. Note that
-  // EXPECT_EQ_WAIT is used, such that the simulated clock is not advanced any
-  // further, but data can be received even if the callback is asynchronous.
-  MockDataChannelObserver new_observer(callee()->data_channel());
-  EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout,
-                           FakeClock());
-}
-
-#endif  // !defined(THREAD_SANITIZER)
-
-// This test sets up a call between two parties with audio, video and but only
-// the caller client supports RTP data channels.
-TEST_P(DataChannelIntegrationTest, RtpDataChannelsRejectedByCallee) {
-  PeerConnectionInterface::RTCConfiguration rtc_config_1;
-  rtc_config_1.enable_rtp_data_channel = true;
-  // Must disable DTLS to make negotiation succeed.
-  rtc_config_1.enable_dtls_srtp = false;
-  PeerConnectionInterface::RTCConfiguration rtc_config_2;
-  rtc_config_2.enable_dtls_srtp = false;
-  rtc_config_2.enable_dtls_srtp = false;
-  ASSERT_TRUE(
-      CreatePeerConnectionWrappersWithConfig(rtc_config_1, rtc_config_2));
-  ConnectFakeSignaling();
-  caller()->CreateDataChannel();
-  ASSERT_TRUE(caller()->data_channel() != nullptr);
-  caller()->AddAudioVideoTracks();
-  callee()->AddAudioVideoTracks();
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  // The caller should still have a data channel, but it should be closed, and
-  // one should ever have been created for the callee.
-  EXPECT_TRUE(caller()->data_channel() != nullptr);
-  EXPECT_FALSE(caller()->data_observer()->IsOpen());
-  EXPECT_EQ(nullptr, callee()->data_channel());
-}
-
-// This test sets up a call between two parties with audio, and video. When
-// audio and video is setup and flowing, an RTP data channel is negotiated.
-TEST_P(DataChannelIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
-  PeerConnectionInterface::RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  rtc_config.enable_dtls_srtp = false;
-  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
-  ConnectFakeSignaling();
-  // Do initial offer/answer with audio/video.
-  caller()->AddAudioVideoTracks();
-  callee()->AddAudioVideoTracks();
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  // Create data channel and do new offer and answer.
-  caller()->CreateDataChannel();
-  caller()->CreateAndSetAndSignalOffer();
-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  ASSERT_NE(nullptr, caller()->data_channel());
-  ASSERT_NE(nullptr, callee()->data_channel());
-  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
-  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
-  // Ensure data can be sent in both directions.
-  std::string data = "hello world";
-  SendRtpDataWithRetries(caller()->data_channel(), data, 5);
-  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
-                 kDefaultTimeout);
-  SendRtpDataWithRetries(callee()->data_channel(), data, 5);
-  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
-                 kDefaultTimeout);
-}
-
-#ifdef WEBRTC_HAVE_SCTP
-
 // This test sets up a call between two parties with audio, video and an SCTP
 // data channel.
 TEST_P(DataChannelIntegrationTest, EndToEndCallWithSctpDataChannel) {

pc/peer_connection.cc

@@ -362,7 +362,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
          disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
          max_ipv6_networks == o.max_ipv6_networks &&
          disable_link_local_networks == o.disable_link_local_networks &&
-         enable_rtp_data_channel == o.enable_rtp_data_channel &&
          screencast_min_bitrate == o.screencast_min_bitrate &&
          combined_audio_video_bwe == o.combined_audio_video_bwe &&
          enable_dtls_srtp == o.enable_dtls_srtp &&
@@ -594,17 +593,10 @@ RTCError PeerConnection::Initialize(
     NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
   }
 
-  if (configuration.enable_rtp_data_channel) {
-    // Enable creation of RTP data channels if the kEnableRtpDataChannels is
-    // set. It takes precendence over the disable_sctp_data_channels
-    // PeerConnectionFactoryInterface::Options.
-    data_channel_controller_.set_data_channel_type(cricket::DCT_RTP);
-  } else {
   // DTLS has to be enabled to use SCTP.
   if (!options_.disable_sctp_data_channels && dtls_enabled_) {
     data_channel_controller_.set_data_channel_type(cricket::DCT_SCTP);
   }
-  }
 
   // Network thread initialization.
   network_thread()->Invoke<void>(RTC_FROM_HERE, [this, &stun_servers,
@@ -684,8 +676,7 @@ void PeerConnection::InitializeTransportController_n(
   config.active_reset_srtp_params = configuration.active_reset_srtp_params;
 
   // DTLS has to be enabled to use SCTP.
-  if (!configuration.enable_rtp_data_channel &&
+  if (!options_.disable_sctp_data_channels && dtls_enabled_) {
-      !options_.disable_sctp_data_channels && dtls_enabled_) {
     config.sctp_factory = context_->sctp_transport_factory();
   }
 

pc/peer_connection_bundle_unittest.cc

@@ -753,11 +753,9 @@ TEST_P(PeerConnectionBundleTest, RejectDescriptionChangingBundleTag) {
 // This tests that removing contents from BUNDLE group and reject the whole
 // BUNDLE group could work. This is a regression test for
 // (https://bugs.chromium.org/p/chromium/issues/detail?id=827917)
+#ifdef HAVE_SCTP
 TEST_P(PeerConnectionBundleTest, RemovingContentAndRejectBundleGroup) {
   RTCConfiguration config;
-#ifndef WEBRTC_HAVE_SCTP
-  config.enable_rtp_data_channel = true;
-#endif
   config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
   auto caller = CreatePeerConnectionWithAudioVideo(config);
   caller->CreateDataChannel("dc");
@@ -782,6 +780,7 @@ TEST_P(PeerConnectionBundleTest, RemovingContentAndRejectBundleGroup) {
 
   EXPECT_TRUE(caller->SetLocalDescription(std::move(re_offer)));
 }
+#endif
 
 // This tests that the BUNDLE group in answer should be a subset of the offered
 // group.

pc/peer_connection_data_channel_unittest.cc

@@ -193,28 +193,6 @@ class PeerConnectionDataChannelUnifiedPlanTest
       : PeerConnectionDataChannelBaseTest(SdpSemantics::kUnifiedPlan) {}
 };
 
-TEST_P(PeerConnectionDataChannelTest,
-       NoSctpTransportCreatedIfRtpDataChannelEnabled) {
-  RTCConfiguration config;
-  config.enable_rtp_data_channel = true;
-  auto caller = CreatePeerConnectionWithDataChannel(config);
-
-  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
-  EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
-}
-
-TEST_P(PeerConnectionDataChannelTest,
-       RtpDataChannelCreatedEvenIfSctpAvailable) {
-  RTCConfiguration config;
-  config.enable_rtp_data_channel = true;
-  PeerConnectionFactoryInterface::Options options;
-  options.disable_sctp_data_channels = false;
-  auto caller = CreatePeerConnectionWithDataChannel(config, options);
-
-  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
-  EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
-}
-
 TEST_P(PeerConnectionDataChannelTest, InternalSctpTransportDeletedOnTeardown) {
   auto caller = CreatePeerConnectionWithDataChannel();
 

pc/peer_connection_interface_unittest.cc

@@ -1901,179 +1901,6 @@ TEST_P(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
   EXPECT_TRUE(DoGetRTCStats());
 }
 
-// This test setup two RTP data channels in loop back.
-TEST_P(PeerConnectionInterfaceTest, TestDataChannel) {
-  RTCConfiguration config;
-  config.enable_rtp_data_channel = true;
-  config.enable_dtls_srtp = false;
-  CreatePeerConnection(config);
-  rtc::scoped_refptr<DataChannelInterface> data1 =
-      pc_->CreateDataChannel("test1", NULL);
-  rtc::scoped_refptr<DataChannelInterface> data2 =
-      pc_->CreateDataChannel("test2", NULL);
-  ASSERT_TRUE(data1 != NULL);
-  std::unique_ptr<MockDataChannelObserver> observer1(
-      new MockDataChannelObserver(data1));
-  std::unique_ptr<MockDataChannelObserver> observer2(
-      new MockDataChannelObserver(data2));
-
-  EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
-  EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
-  std::string data_to_send1 = "testing testing";
-  std::string data_to_send2 = "testing something else";
-  EXPECT_FALSE(data1->Send(DataBuffer(data_to_send1)));
-
-  CreateOfferReceiveAnswer();
-  EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
-  EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
-
-  EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
-  EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
-  EXPECT_TRUE(data1->Send(DataBuffer(data_to_send1)));
-  EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
-
-  EXPECT_EQ_WAIT(data_to_send1, observer1->last_message(), kTimeout);
-  EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
-
-  data1->Close();
-  EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
-  CreateOfferReceiveAnswer();
-  EXPECT_FALSE(observer1->IsOpen());
-  EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
-  EXPECT_TRUE(observer2->IsOpen());
-
-  data_to_send2 = "testing something else again";
-  EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
-
-  EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
-}
-
-// This test verifies that sendnig binary data over RTP data channels should
-// fail.
-TEST_P(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
-  RTCConfiguration config;
-  config.enable_rtp_data_channel = true;
-  config.enable_dtls_srtp = false;
-  CreatePeerConnection(config);
-  rtc::scoped_refptr<DataChannelInterface> data1 =
-      pc_->CreateDataChannel("test1", NULL);
-  rtc::scoped_refptr<DataChannelInterface> data2 =
-      pc_->CreateDataChannel("test2", NULL);
-  ASSERT_TRUE(data1 != NULL);
-  std::unique_ptr<MockDataChannelObserver> observer1(
-      new MockDataChannelObserver(data1));
-  std::unique_ptr<MockDataChannelObserver> observer2(
-      new MockDataChannelObserver(data2));
-
-  EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
-  EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
-
-  CreateOfferReceiveAnswer();
-  EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
-  EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
-
-  EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
-  EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
-
-  rtc::CopyOnWriteBuffer buffer("test", 4);
-  EXPECT_FALSE(data1->Send(DataBuffer(buffer, true)));
-}
-
-// This test setup a RTP data channels in loop back and test that a channel is
-// opened even if the remote end answer with a zero SSRC.
-TEST_P(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
-  RTCConfiguration config;
-  config.enable_rtp_data_channel = true;
-  config.enable_dtls_srtp = false;
-  CreatePeerConnection(config);
-  rtc::scoped_refptr<DataChannelInterface> data1 =
-      pc_->CreateDataChannel("test1", NULL);
-  std::unique_ptr<MockDataChannelObserver> observer1(
-      new MockDataChannelObserver(data1));
-
-  CreateOfferReceiveAnswerWithoutSsrc();
-
-  EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
-
-  data1->Close();
-  EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
-  CreateOfferReceiveAnswerWithoutSsrc();
-  EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
-  EXPECT_FALSE(observer1->IsOpen());
-}
-
-// This test that if a data channel is added in an answer a receive only channel
-// channel is created.
-TEST_P(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
-  RTCConfiguration config;
-  config.enable_rtp_data_channel = true;
-  config.enable_dtls_srtp = false;
-
-  CreatePeerConnection(config);
-
-  std::string offer_label = "offer_channel";
-  rtc::scoped_refptr<DataChannelInterface> offer_channel =
-      pc_->CreateDataChannel(offer_label, NULL);
-
-  CreateOfferAsLocalDescription();
-
-  // Replace the data channel label in the offer and apply it as an answer.
-  std::string receive_label = "answer_channel";
-  std::string sdp;
-  EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
-  absl::StrReplaceAll({{offer_label, receive_label}}, &sdp);
-  CreateAnswerAsRemoteDescription(sdp);
-
-  // Verify that a new incoming data channel has been created and that
-  // it is open but can't we written to.
-  ASSERT_TRUE(observer_.last_datachannel_ != NULL);
-  DataChannelInterface* received_channel = observer_.last_datachannel_;
-  EXPECT_EQ(DataChannelInterface::kConnecting, received_channel->state());
-  EXPECT_EQ(receive_label, received_channel->label());
-  EXPECT_FALSE(received_channel->Send(DataBuffer("something")));
-
-  // Verify that the channel we initially offered has been rejected.
-  EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
-
-  // Do another offer / answer exchange and verify that the data channel is
-  // opened.
-  CreateOfferReceiveAnswer();
-  EXPECT_EQ_WAIT(DataChannelInterface::kOpen, received_channel->state(),
-                 kTimeout);
-}
-
-// This test that no data channel is returned if a reliable channel is
-// requested.
-// TODO(perkj): Remove this test once reliable channels are implemented.
-TEST_P(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
-  RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  CreatePeerConnection(rtc_config);
-
-  std::string label = "test";
-  webrtc::DataChannelInit config;
-  config.reliable = true;
-  rtc::scoped_refptr<DataChannelInterface> channel =
-      pc_->CreateDataChannel(label, &config);
-  EXPECT_TRUE(channel == NULL);
-}
-
-// Verifies that duplicated label is not allowed for RTP data channel.
-TEST_P(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
-  RTCConfiguration config;
-  config.enable_rtp_data_channel = true;
-  CreatePeerConnection(config);
-
-  std::string label = "test";
-  rtc::scoped_refptr<DataChannelInterface> channel =
-      pc_->CreateDataChannel(label, nullptr);
-  EXPECT_NE(channel, nullptr);
-
-  rtc::scoped_refptr<DataChannelInterface> dup_channel =
-      pc_->CreateDataChannel(label, nullptr);
-  EXPECT_EQ(dup_channel, nullptr);
-}
-
 // This tests that a SCTP data channel is returned using different
 // DataChannelInit configurations.
 TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
@@ -2191,78 +2018,6 @@ TEST_P(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
   EXPECT_NE(dup_channel, nullptr);
 }
 
-// This test verifies that OnRenegotiationNeeded is fired for every new RTP
-// DataChannel.
-TEST_P(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
-  RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  rtc_config.enable_dtls_srtp = false;
-  CreatePeerConnection(rtc_config);
-
-  rtc::scoped_refptr<DataChannelInterface> dc1 =
-      pc_->CreateDataChannel("test1", NULL);
-  EXPECT_TRUE(observer_.renegotiation_needed_);
-  observer_.renegotiation_needed_ = false;
-
-  CreateOfferReceiveAnswer();
-
-  rtc::scoped_refptr<DataChannelInterface> dc2 =
-      pc_->CreateDataChannel("test2", NULL);
-  EXPECT_EQ(observer_.renegotiation_needed_,
-            GetParam() == SdpSemantics::kPlanB);
-}
-
-// This test that a data channel closes when a PeerConnection is deleted/closed.
-TEST_P(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
-  RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  rtc_config.enable_dtls_srtp = false;
-  CreatePeerConnection(rtc_config);
-
-  rtc::scoped_refptr<DataChannelInterface> data1 =
-      pc_->CreateDataChannel("test1", NULL);
-  rtc::scoped_refptr<DataChannelInterface> data2 =
-      pc_->CreateDataChannel("test2", NULL);
-  ASSERT_TRUE(data1 != NULL);
-  std::unique_ptr<MockDataChannelObserver> observer1(
-      new MockDataChannelObserver(data1));
-  std::unique_ptr<MockDataChannelObserver> observer2(
-      new MockDataChannelObserver(data2));
-
-  CreateOfferReceiveAnswer();
-  EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
-  EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
-
-  ReleasePeerConnection();
-  EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
-  EXPECT_EQ(DataChannelInterface::kClosed, data2->state());
-}
-
-// This tests that RTP data channels can be rejected in an answer.
-TEST_P(PeerConnectionInterfaceTest, TestRejectRtpDataChannelInAnswer) {
-  RTCConfiguration rtc_config;
-  rtc_config.enable_rtp_data_channel = true;
-  rtc_config.enable_dtls_srtp = false;
-  CreatePeerConnection(rtc_config);
-
-  rtc::scoped_refptr<DataChannelInterface> offer_channel(
-      pc_->CreateDataChannel("offer_channel", NULL));
-
-  CreateOfferAsLocalDescription();
-
-  // Create an answer where the m-line for data channels are rejected.
-  std::string sdp;
-  EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
-  std::unique_ptr<SessionDescriptionInterface> answer(
-      webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
-  ASSERT_TRUE(answer);
-  cricket::ContentInfo* data_info =
-      cricket::GetFirstDataContent(answer->description());
-  data_info->rejected = true;
-
-  DoSetRemoteDescription(std::move(answer));
-  EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
-}
 
 #ifdef WEBRTC_HAVE_SCTP
 // This tests that SCTP data channels can be rejected in an answer.

pc/peer_connection_jsep_unittest.cc

@@ -2266,17 +2266,4 @@ TEST_F(PeerConnectionJsepTest,
   EXPECT_TRUE(callee->CreateOfferAndSetAsLocal());
 }
 
-TEST_F(PeerConnectionJsepTest, RollbackRtpDataChannel) {
-  RTCConfiguration config;
-  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
-  config.enable_rtp_data_channel = true;
-  auto pc = CreatePeerConnection(config);
-  pc->CreateDataChannel("dummy");
-  auto offer = pc->CreateOffer();
-  EXPECT_TRUE(pc->CreateOfferAndSetAsLocal());
-  EXPECT_TRUE(pc->SetRemoteDescription(pc->CreateRollback()));
-  EXPECT_TRUE(pc->SetLocalDescription(std::move(offer)));
-  pc->pc()->Close();
-}
-
 }  // namespace webrtc

sdk/android/api/org/webrtc/PeerConnection.java

@@ -514,7 +514,6 @@ public class PeerConnection {
     public boolean disableIpv6;
     public boolean enableDscp;
     public boolean enableCpuOveruseDetection;
-    public boolean enableRtpDataChannel;
     public boolean suspendBelowMinBitrate;
     @Nullable public Integer screencastMinBitrate;
     @Nullable public Boolean combinedAudioVideoBwe;
@@ -595,7 +594,6 @@ public class PeerConnection {
       disableIpv6 = false;
       enableDscp = false;
       enableCpuOveruseDetection = true;
-      enableRtpDataChannel = false;
       suspendBelowMinBitrate = false;
       screencastMinBitrate = null;
       combinedAudioVideoBwe = null;
@@ -768,11 +766,6 @@ public class PeerConnection {
       return enableCpuOveruseDetection;
     }
 
-    @CalledByNative("RTCConfiguration")
-    boolean getEnableRtpDataChannel() {
-      return enableRtpDataChannel;
-    }
-
     @CalledByNative("RTCConfiguration")
     boolean getSuspendBelowMinBitrate() {
       return suspendBelowMinBitrate;

sdk/android/src/jni/pc/peer_connection.cc

@@ -251,8 +251,6 @@ void JavaToNativeRTCConfiguration(
       Java_RTCConfiguration_getEnableDscp(jni, j_rtc_config);
   rtc_config->media_config.video.enable_cpu_adaptation =
       Java_RTCConfiguration_getEnableCpuOveruseDetection(jni, j_rtc_config);
-  rtc_config->enable_rtp_data_channel =
-      Java_RTCConfiguration_getEnableRtpDataChannel(jni, j_rtc_config);
   rtc_config->media_config.video.suspend_below_min_bitrate =
       Java_RTCConfiguration_getSuspendBelowMinBitrate(jni, j_rtc_config);
   rtc_config->screencast_min_bitrate = JavaToNativeOptionalInt(

sdk/media_constraints.cc

@@ -167,8 +167,6 @@ void CopyConstraintsIntoRtcConfiguration(
   FindConstraint(constraints, MediaConstraints::kCpuOveruseDetection,
                  &configuration->media_config.video.enable_cpu_adaptation,
                  nullptr);
-  FindConstraint(constraints, MediaConstraints::kEnableRtpDataChannels,
-                 &configuration->enable_rtp_data_channel, nullptr);
   // Find Suspend Below Min Bitrate constraint.
   FindConstraint(
       constraints, MediaConstraints::kEnableVideoSuspendBelowMinBitrate,

sdk/media_constraints_unittest.cc

@@ -23,7 +23,6 @@ bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
   return a.disable_ipv6 == b.disable_ipv6 &&
          a.audio_jitter_buffer_max_packets ==
              b.audio_jitter_buffer_max_packets &&
-         a.enable_rtp_data_channel == b.enable_rtp_data_channel &&
          a.screencast_min_bitrate == b.screencast_min_bitrate &&
          a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
          a.enable_dtls_srtp == b.enable_dtls_srtp &&