| /* | 
 |  *  Copyright 2013 The WebRTC project authors. All Rights Reserved. | 
 |  * | 
 |  *  Use of this source code is governed by a BSD-style license | 
 |  *  that can be found in the LICENSE file in the root of the source | 
 |  *  tree. An additional intellectual property rights grant can be found | 
 |  *  in the file PATENTS.  All contributing project authors may | 
 |  *  be found in the AUTHORS file in the root of the source tree. | 
 |  */ | 
 |  | 
 | #include <memory> | 
 |  | 
 | #include "webrtc/api/test/peerconnectiontestwrapper.h" | 
 | // Notice that mockpeerconnectionobservers.h must be included after the above! | 
 | #include "webrtc/api/test/mockpeerconnectionobservers.h" | 
 | #ifdef WEBRTC_ANDROID | 
 | #include "webrtc/api/test/androidtestinitializer.h" | 
 | #endif | 
 | #include "webrtc/base/gunit.h" | 
 | #include "webrtc/base/logging.h" | 
 | #include "webrtc/base/ssladapter.h" | 
 | #include "webrtc/base/thread.h" | 
 | #include "webrtc/base/sslstreamadapter.h" | 
 | #include "webrtc/base/stringencode.h" | 
 | #include "webrtc/base/stringutils.h" | 
 |  | 
 | #define MAYBE_SKIP_TEST(feature)                    \ | 
 |   if (!(feature())) {                               \ | 
 |     LOG(LS_INFO) << "Feature disabled... skipping"; \ | 
 |     return;                                         \ | 
 |   } | 
 |  | 
 | using webrtc::DataChannelInterface; | 
 | using webrtc::FakeConstraints; | 
 | using webrtc::MediaConstraintsInterface; | 
 | using webrtc::MediaStreamInterface; | 
 | using webrtc::PeerConnectionInterface; | 
 |  | 
 | namespace { | 
 |  | 
 | const int kMaxWait = 10000; | 
 |  | 
 | }  // namespace | 
 |  | 
 | class PeerConnectionEndToEndTest | 
 |     : public sigslot::has_slots<>, | 
 |       public testing::Test { | 
 |  public: | 
 |   typedef std::vector<rtc::scoped_refptr<DataChannelInterface> > | 
 |       DataChannelList; | 
 |  | 
 |   PeerConnectionEndToEndTest() { | 
 |     RTC_CHECK(network_thread_.Start()); | 
 |     RTC_CHECK(worker_thread_.Start()); | 
 |     caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>( | 
 |         "caller", &network_thread_, &worker_thread_); | 
 |     callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>( | 
 |         "callee", &network_thread_, &worker_thread_); | 
 |     webrtc::PeerConnectionInterface::IceServer ice_server; | 
 |     ice_server.uri = "stun:stun.l.google.com:19302"; | 
 |     config_.servers.push_back(ice_server); | 
 |  | 
 | #ifdef WEBRTC_ANDROID | 
 |     webrtc::InitializeAndroidObjects(); | 
 | #endif | 
 |   } | 
 |  | 
 |   void CreatePcs() { | 
 |     CreatePcs(NULL); | 
 |   } | 
 |  | 
 |   void CreatePcs(const MediaConstraintsInterface* pc_constraints) { | 
 |     EXPECT_TRUE(caller_->CreatePc(pc_constraints, config_)); | 
 |     EXPECT_TRUE(callee_->CreatePc(pc_constraints, config_)); | 
 |     PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get()); | 
 |  | 
 |     caller_->SignalOnDataChannel.connect( | 
 |         this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel); | 
 |     callee_->SignalOnDataChannel.connect( | 
 |         this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel); | 
 |   } | 
 |  | 
 |   void GetAndAddUserMedia() { | 
 |     FakeConstraints audio_constraints; | 
 |     FakeConstraints video_constraints; | 
 |     GetAndAddUserMedia(true, audio_constraints, true, video_constraints); | 
 |   } | 
 |  | 
 |   void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints, | 
 |                           bool video, FakeConstraints video_constraints) { | 
 |     caller_->GetAndAddUserMedia(audio, audio_constraints, | 
 |                                 video, video_constraints); | 
 |     callee_->GetAndAddUserMedia(audio, audio_constraints, | 
 |                                 video, video_constraints); | 
 |   } | 
 |  | 
 |   void Negotiate() { | 
 |     caller_->CreateOffer(NULL); | 
 |   } | 
 |  | 
 |   void WaitForCallEstablished() { | 
 |     caller_->WaitForCallEstablished(); | 
 |     callee_->WaitForCallEstablished(); | 
 |   } | 
 |  | 
 |   void WaitForConnection() { | 
 |     caller_->WaitForConnection(); | 
 |     callee_->WaitForConnection(); | 
 |   } | 
 |  | 
 |   void OnCallerAddedDataChanel(DataChannelInterface* dc) { | 
 |     caller_signaled_data_channels_.push_back(dc); | 
 |   } | 
 |  | 
 |   void OnCalleeAddedDataChannel(DataChannelInterface* dc) { | 
 |     callee_signaled_data_channels_.push_back(dc); | 
 |   } | 
 |  | 
 |   // Tests that |dc1| and |dc2| can send to and receive from each other. | 
 |   void TestDataChannelSendAndReceive( | 
 |       DataChannelInterface* dc1, DataChannelInterface* dc2) { | 
 |     std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer( | 
 |         new webrtc::MockDataChannelObserver(dc1)); | 
 |  | 
 |     std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer( | 
 |         new webrtc::MockDataChannelObserver(dc2)); | 
 |  | 
 |     static const std::string kDummyData = "abcdefg"; | 
 |     webrtc::DataBuffer buffer(kDummyData); | 
 |     EXPECT_TRUE(dc1->Send(buffer)); | 
 |     EXPECT_EQ_WAIT(kDummyData, dc2_observer->last_message(), kMaxWait); | 
 |  | 
 |     EXPECT_TRUE(dc2->Send(buffer)); | 
 |     EXPECT_EQ_WAIT(kDummyData, dc1_observer->last_message(), kMaxWait); | 
 |  | 
 |     EXPECT_EQ(1U, dc1_observer->received_message_count()); | 
 |     EXPECT_EQ(1U, dc2_observer->received_message_count()); | 
 |   } | 
 |  | 
 |   void WaitForDataChannelsToOpen(DataChannelInterface* local_dc, | 
 |                                  const DataChannelList& remote_dc_list, | 
 |                                  size_t remote_dc_index) { | 
 |     EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait); | 
 |  | 
 |     EXPECT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait); | 
 |     EXPECT_EQ_WAIT(DataChannelInterface::kOpen, | 
 |                    remote_dc_list[remote_dc_index]->state(), | 
 |                    kMaxWait); | 
 |     EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id()); | 
 |   } | 
 |  | 
 |   void CloseDataChannels(DataChannelInterface* local_dc, | 
 |                          const DataChannelList& remote_dc_list, | 
 |                          size_t remote_dc_index) { | 
 |     local_dc->Close(); | 
 |     EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait); | 
 |     EXPECT_EQ_WAIT(DataChannelInterface::kClosed, | 
 |                    remote_dc_list[remote_dc_index]->state(), | 
 |                    kMaxWait); | 
 |   } | 
 |  | 
 |  protected: | 
 |   rtc::Thread network_thread_; | 
 |   rtc::Thread worker_thread_; | 
 |   rtc::scoped_refptr<PeerConnectionTestWrapper> caller_; | 
 |   rtc::scoped_refptr<PeerConnectionTestWrapper> callee_; | 
 |   DataChannelList caller_signaled_data_channels_; | 
 |   DataChannelList callee_signaled_data_channels_; | 
 |   webrtc::PeerConnectionInterface::RTCConfiguration config_; | 
 | }; | 
 |  | 
 | // Disabled for TSan v2, see | 
 | // https://bugs.chromium.org/p/webrtc/issues/detail?id=4719 for details. | 
 | // Disabled for Mac, see | 
 | // https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details. | 
 | #if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC) | 
 | TEST_F(PeerConnectionEndToEndTest, Call) { | 
 |   CreatePcs(); | 
 |   GetAndAddUserMedia(); | 
 |   Negotiate(); | 
 |   WaitForCallEstablished(); | 
 | } | 
 | #endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC) | 
 |  | 
 | TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) { | 
 |   FakeConstraints pc_constraints; | 
 |   pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | 
 |                               false); | 
 |   CreatePcs(&pc_constraints); | 
 |   GetAndAddUserMedia(); | 
 |   Negotiate(); | 
 |   WaitForCallEstablished(); | 
 | } | 
 |  | 
 | // Verifies that a DataChannel created before the negotiation can transition to | 
 | // "OPEN" and transfer data. | 
 | TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { | 
 |   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); | 
 |  | 
 |   CreatePcs(); | 
 |  | 
 |   webrtc::DataChannelInit init; | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc( | 
 |       caller_->CreateDataChannel("data", init)); | 
 |   rtc::scoped_refptr<DataChannelInterface> callee_dc( | 
 |       callee_->CreateDataChannel("data", init)); | 
 |  | 
 |   Negotiate(); | 
 |   WaitForConnection(); | 
 |  | 
 |   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0); | 
 |   WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0); | 
 |  | 
 |   TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[0]); | 
 |   TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]); | 
 |  | 
 |   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0); | 
 |   CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0); | 
 | } | 
 |  | 
 | // Verifies that a DataChannel created after the negotiation can transition to | 
 | // "OPEN" and transfer data. | 
 | TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) { | 
 |   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); | 
 |  | 
 |   CreatePcs(); | 
 |  | 
 |   webrtc::DataChannelInit init; | 
 |  | 
 |   // This DataChannel is for creating the data content in the negotiation. | 
 |   rtc::scoped_refptr<DataChannelInterface> dummy( | 
 |       caller_->CreateDataChannel("data", init)); | 
 |   Negotiate(); | 
 |   WaitForConnection(); | 
 |  | 
 |   // Wait for the data channel created pre-negotiation to be opened. | 
 |   WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0); | 
 |  | 
 |   // Create new DataChannels after the negotiation and verify their states. | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc( | 
 |       caller_->CreateDataChannel("hello", init)); | 
 |   rtc::scoped_refptr<DataChannelInterface> callee_dc( | 
 |       callee_->CreateDataChannel("hello", init)); | 
 |  | 
 |   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1); | 
 |   WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0); | 
 |  | 
 |   TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]); | 
 |   TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]); | 
 |  | 
 |   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1); | 
 |   CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0); | 
 | } | 
 |  | 
 | // Verifies that DataChannel IDs are even/odd based on the DTLS roles. | 
 | TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) { | 
 |   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); | 
 |  | 
 |   CreatePcs(); | 
 |  | 
 |   webrtc::DataChannelInit init; | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc_1( | 
 |       caller_->CreateDataChannel("data", init)); | 
 |   rtc::scoped_refptr<DataChannelInterface> callee_dc_1( | 
 |       callee_->CreateDataChannel("data", init)); | 
 |  | 
 |   Negotiate(); | 
 |   WaitForConnection(); | 
 |  | 
 |   EXPECT_EQ(1U, caller_dc_1->id() % 2); | 
 |   EXPECT_EQ(0U, callee_dc_1->id() % 2); | 
 |  | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc_2( | 
 |       caller_->CreateDataChannel("data", init)); | 
 |   rtc::scoped_refptr<DataChannelInterface> callee_dc_2( | 
 |       callee_->CreateDataChannel("data", init)); | 
 |  | 
 |   EXPECT_EQ(1U, caller_dc_2->id() % 2); | 
 |   EXPECT_EQ(0U, callee_dc_2->id() % 2); | 
 | } | 
 |  | 
 | // Verifies that the message is received by the right remote DataChannel when | 
 | // there are multiple DataChannels. | 
 | TEST_F(PeerConnectionEndToEndTest, | 
 |        MessageTransferBetweenTwoPairsOfDataChannels) { | 
 |   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); | 
 |  | 
 |   CreatePcs(); | 
 |  | 
 |   webrtc::DataChannelInit init; | 
 |  | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc_1( | 
 |       caller_->CreateDataChannel("data", init)); | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc_2( | 
 |       caller_->CreateDataChannel("data", init)); | 
 |  | 
 |   Negotiate(); | 
 |   WaitForConnection(); | 
 |   WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0); | 
 |   WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1); | 
 |  | 
 |   std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer( | 
 |       new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0])); | 
 |  | 
 |   std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer( | 
 |       new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1])); | 
 |  | 
 |   const std::string message_1 = "hello 1"; | 
 |   const std::string message_2 = "hello 2"; | 
 |  | 
 |   caller_dc_1->Send(webrtc::DataBuffer(message_1)); | 
 |   EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait); | 
 |  | 
 |   caller_dc_2->Send(webrtc::DataBuffer(message_2)); | 
 |   EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait); | 
 |  | 
 |   EXPECT_EQ(1U, dc_1_observer->received_message_count()); | 
 |   EXPECT_EQ(1U, dc_2_observer->received_message_count()); | 
 | } | 
 |  | 
 | #ifdef HAVE_QUIC | 
 | // Test that QUIC data channels can be used and that messages go to the correct | 
 | // remote data channel when both peers want to use QUIC. It is assumed that the | 
 | // application has externally negotiated the data channel parameters. | 
 | TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) { | 
 |   config_.enable_quic = true; | 
 |   CreatePcs(); | 
 |  | 
 |   webrtc::DataChannelInit init_1; | 
 |   init_1.id = 0; | 
 |   init_1.ordered = false; | 
 |   init_1.reliable = true; | 
 |  | 
 |   webrtc::DataChannelInit init_2; | 
 |   init_2.id = 1; | 
 |   init_2.ordered = false; | 
 |   init_2.reliable = true; | 
 |  | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc_1( | 
 |       caller_->CreateDataChannel("data", init_1)); | 
 |   ASSERT_NE(nullptr, caller_dc_1); | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc_2( | 
 |       caller_->CreateDataChannel("data", init_2)); | 
 |   ASSERT_NE(nullptr, caller_dc_2); | 
 |   rtc::scoped_refptr<DataChannelInterface> callee_dc_1( | 
 |       callee_->CreateDataChannel("data", init_1)); | 
 |   ASSERT_NE(nullptr, callee_dc_1); | 
 |   rtc::scoped_refptr<DataChannelInterface> callee_dc_2( | 
 |       callee_->CreateDataChannel("data", init_2)); | 
 |   ASSERT_NE(nullptr, callee_dc_2); | 
 |  | 
 |   Negotiate(); | 
 |   WaitForConnection(); | 
 |   EXPECT_TRUE_WAIT(caller_dc_1->state() == webrtc::DataChannelInterface::kOpen, | 
 |                    kMaxWait); | 
 |   EXPECT_TRUE_WAIT(callee_dc_1->state() == webrtc::DataChannelInterface::kOpen, | 
 |                    kMaxWait); | 
 |   EXPECT_TRUE_WAIT(caller_dc_2->state() == webrtc::DataChannelInterface::kOpen, | 
 |                    kMaxWait); | 
 |   EXPECT_TRUE_WAIT(callee_dc_2->state() == webrtc::DataChannelInterface::kOpen, | 
 |                    kMaxWait); | 
 |  | 
 |   std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer( | 
 |       new webrtc::MockDataChannelObserver(callee_dc_1.get())); | 
 |  | 
 |   std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer( | 
 |       new webrtc::MockDataChannelObserver(callee_dc_2.get())); | 
 |  | 
 |   const std::string message_1 = "hello 1"; | 
 |   const std::string message_2 = "hello 2"; | 
 |  | 
 |   // Send data from caller to callee. | 
 |   caller_dc_1->Send(webrtc::DataBuffer(message_1)); | 
 |   EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait); | 
 |  | 
 |   caller_dc_2->Send(webrtc::DataBuffer(message_2)); | 
 |   EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait); | 
 |  | 
 |   EXPECT_EQ(1U, dc_1_observer->received_message_count()); | 
 |   EXPECT_EQ(1U, dc_2_observer->received_message_count()); | 
 |  | 
 |   // Send data from callee to caller. | 
 |   dc_1_observer.reset(new webrtc::MockDataChannelObserver(caller_dc_1.get())); | 
 |   dc_2_observer.reset(new webrtc::MockDataChannelObserver(caller_dc_2.get())); | 
 |  | 
 |   callee_dc_1->Send(webrtc::DataBuffer(message_1)); | 
 |   EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait); | 
 |  | 
 |   callee_dc_2->Send(webrtc::DataBuffer(message_2)); | 
 |   EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait); | 
 |  | 
 |   EXPECT_EQ(1U, dc_1_observer->received_message_count()); | 
 |   EXPECT_EQ(1U, dc_2_observer->received_message_count()); | 
 | } | 
 | #endif  // HAVE_QUIC | 
 |  | 
 | // Verifies that a DataChannel added from an OPEN message functions after | 
 | // a channel has been previously closed (webrtc issue 3778). | 
 | // This previously failed because the new channel re-uses the ID of the closed | 
 | // channel, and the closed channel was incorrectly still assigned to the id. | 
 | // TODO(deadbeef): This is disabled because there's currently a race condition | 
 | // caused by the fact that a data channel signals that it's closed before it | 
 | // really is. Re-enable this test once that's fixed. | 
 | // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453 | 
 | TEST_F(PeerConnectionEndToEndTest, | 
 |        DISABLED_DataChannelFromOpenWorksAfterClose) { | 
 |   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); | 
 |  | 
 |   CreatePcs(); | 
 |  | 
 |   webrtc::DataChannelInit init; | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc( | 
 |       caller_->CreateDataChannel("data", init)); | 
 |  | 
 |   Negotiate(); | 
 |   WaitForConnection(); | 
 |  | 
 |   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0); | 
 |   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0); | 
 |  | 
 |   // Create a new channel and ensure it works after closing the previous one. | 
 |   caller_dc = caller_->CreateDataChannel("data2", init); | 
 |  | 
 |   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1); | 
 |   TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]); | 
 |  | 
 |   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1); | 
 | } | 
 |  | 
 | // This tests that if a data channel is closed remotely while not referenced | 
 | // by the application (meaning only the PeerConnection contributes to its | 
 | // reference count), no memory access violation will occur. | 
 | // See: https://code.google.com/p/chromium/issues/detail?id=565048 | 
 | TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) { | 
 |   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); | 
 |  | 
 |   CreatePcs(); | 
 |  | 
 |   webrtc::DataChannelInit init; | 
 |   rtc::scoped_refptr<DataChannelInterface> caller_dc( | 
 |       caller_->CreateDataChannel("data", init)); | 
 |  | 
 |   Negotiate(); | 
 |   WaitForConnection(); | 
 |  | 
 |   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0); | 
 |   // This removes the reference to the remote data channel that we hold. | 
 |   callee_signaled_data_channels_.clear(); | 
 |   caller_dc->Close(); | 
 |   EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait); | 
 |  | 
 |   // Wait for a bit longer so the remote data channel will receive the | 
 |   // close message and be destroyed. | 
 |   rtc::Thread::Current()->ProcessMessages(100); | 
 | } |