| /* |
| * 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. |
| 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); |
| } |