| /* |
| * Copyright 2017 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 <string> |
| #include <type_traits> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/call/call_factory_interface.h" |
| #include "api/jsep.h" |
| #include "api/media_transport_interface.h" |
| #include "api/media_types.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/peer_connection_proxy.h" |
| #include "api/scoped_refptr.h" |
| #include "api/test/fake_media_transport.h" |
| #include "media/base/codec.h" |
| #include "media/base/fake_media_engine.h" |
| #include "media/base/media_constants.h" |
| #include "media/base/media_engine.h" |
| #include "media/sctp/sctp_transport_internal.h" |
| #include "p2p/base/p2p_constants.h" |
| #include "p2p/base/port_allocator.h" |
| #include "pc/media_session.h" |
| #include "pc/peer_connection.h" |
| #include "pc/peer_connection_factory.h" |
| #include "pc/peer_connection_wrapper.h" |
| #include "pc/sdp_utils.h" |
| #include "pc/session_description.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/rtc_certificate_generator.h" |
| #include "rtc_base/thread.h" |
| #include "test/gtest.h" |
| #ifdef WEBRTC_ANDROID |
| #include "pc/test/android_test_initializer.h" |
| #endif |
| #include "absl/memory/memory.h" |
| #include "pc/test/fake_sctp_transport.h" |
| #include "rtc_base/virtual_socket_server.h" |
| |
| namespace webrtc { |
| |
| using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; |
| using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions; |
| using ::testing::Values; |
| |
| namespace { |
| |
| PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies( |
| rtc::Thread* network_thread, |
| rtc::Thread* worker_thread, |
| rtc::Thread* signaling_thread, |
| std::unique_ptr<cricket::MediaEngineInterface> media_engine, |
| std::unique_ptr<CallFactoryInterface> call_factory, |
| std::unique_ptr<MediaTransportFactory> media_transport_factory) { |
| PeerConnectionFactoryDependencies deps; |
| deps.network_thread = network_thread; |
| deps.worker_thread = worker_thread; |
| deps.signaling_thread = signaling_thread; |
| deps.media_engine = std::move(media_engine); |
| deps.call_factory = std::move(call_factory); |
| deps.media_transport_factory = std::move(media_transport_factory); |
| return deps; |
| } |
| |
| } // namespace |
| |
| class PeerConnectionFactoryForDataChannelTest |
| : public rtc::RefCountedObject<PeerConnectionFactory> { |
| public: |
| PeerConnectionFactoryForDataChannelTest() |
| : rtc::RefCountedObject<PeerConnectionFactory>( |
| CreatePeerConnectionFactoryDependencies( |
| rtc::Thread::Current(), |
| rtc::Thread::Current(), |
| rtc::Thread::Current(), |
| absl::make_unique<cricket::FakeMediaEngine>(), |
| CreateCallFactory(), |
| absl::make_unique<FakeMediaTransportFactory>())) {} |
| |
| std::unique_ptr<cricket::SctpTransportInternalFactory> |
| CreateSctpTransportInternalFactory() { |
| auto factory = absl::make_unique<FakeSctpTransportFactory>(); |
| last_fake_sctp_transport_factory_ = factory.get(); |
| return factory; |
| } |
| |
| FakeSctpTransportFactory* last_fake_sctp_transport_factory_ = nullptr; |
| }; |
| |
| class PeerConnectionWrapperForDataChannelTest : public PeerConnectionWrapper { |
| public: |
| using PeerConnectionWrapper::PeerConnectionWrapper; |
| |
| FakeSctpTransportFactory* sctp_transport_factory() { |
| return sctp_transport_factory_; |
| } |
| |
| void set_sctp_transport_factory( |
| FakeSctpTransportFactory* sctp_transport_factory) { |
| sctp_transport_factory_ = sctp_transport_factory; |
| } |
| |
| absl::optional<std::string> sctp_content_name() { |
| return GetInternalPeerConnection()->sctp_content_name(); |
| } |
| |
| absl::optional<std::string> sctp_transport_name() { |
| return GetInternalPeerConnection()->sctp_transport_name(); |
| } |
| |
| PeerConnection* GetInternalPeerConnection() { |
| auto* pci = |
| static_cast<PeerConnectionProxyWithInternal<PeerConnectionInterface>*>( |
| pc()); |
| return static_cast<PeerConnection*>(pci->internal()); |
| } |
| |
| private: |
| FakeSctpTransportFactory* sctp_transport_factory_ = nullptr; |
| }; |
| |
| class PeerConnectionDataChannelBaseTest : public ::testing::Test { |
| protected: |
| typedef std::unique_ptr<PeerConnectionWrapperForDataChannelTest> WrapperPtr; |
| |
| explicit PeerConnectionDataChannelBaseTest(SdpSemantics sdp_semantics) |
| : vss_(new rtc::VirtualSocketServer()), |
| main_(vss_.get()), |
| sdp_semantics_(sdp_semantics) { |
| #ifdef WEBRTC_ANDROID |
| InitializeAndroidObjects(); |
| #endif |
| } |
| |
| WrapperPtr CreatePeerConnection() { |
| return CreatePeerConnection(RTCConfiguration()); |
| } |
| |
| WrapperPtr CreatePeerConnection(const RTCConfiguration& config) { |
| return CreatePeerConnection(config, |
| PeerConnectionFactoryInterface::Options()); |
| } |
| |
| WrapperPtr CreatePeerConnection( |
| const RTCConfiguration& config, |
| const PeerConnectionFactoryInterface::Options factory_options) { |
| rtc::scoped_refptr<PeerConnectionFactoryForDataChannelTest> pc_factory( |
| new PeerConnectionFactoryForDataChannelTest()); |
| pc_factory->SetOptions(factory_options); |
| RTC_CHECK(pc_factory->Initialize()); |
| auto observer = absl::make_unique<MockPeerConnectionObserver>(); |
| RTCConfiguration modified_config = config; |
| modified_config.sdp_semantics = sdp_semantics_; |
| auto pc = pc_factory->CreatePeerConnection(modified_config, nullptr, |
| nullptr, observer.get()); |
| if (!pc) { |
| return nullptr; |
| } |
| |
| observer->SetPeerConnectionInterface(pc.get()); |
| auto wrapper = absl::make_unique<PeerConnectionWrapperForDataChannelTest>( |
| pc_factory, pc, std::move(observer)); |
| RTC_DCHECK(pc_factory->last_fake_sctp_transport_factory_); |
| wrapper->set_sctp_transport_factory( |
| pc_factory->last_fake_sctp_transport_factory_); |
| return wrapper; |
| } |
| |
| // Accepts the same arguments as CreatePeerConnection and adds a default data |
| // channel. |
| template <typename... Args> |
| WrapperPtr CreatePeerConnectionWithDataChannel(Args&&... args) { |
| auto wrapper = CreatePeerConnection(std::forward<Args>(args)...); |
| if (!wrapper) { |
| return nullptr; |
| } |
| EXPECT_TRUE(wrapper->pc()->CreateDataChannel("dc", nullptr)); |
| return wrapper; |
| } |
| |
| // Changes the SCTP data channel port on the given session description. |
| void ChangeSctpPortOnDescription(cricket::SessionDescription* desc, |
| int port) { |
| cricket::DataCodec sctp_codec(cricket::kGoogleSctpDataCodecPlType, |
| cricket::kGoogleSctpDataCodecName); |
| sctp_codec.SetParam(cricket::kCodecParamPort, port); |
| |
| auto* data_content = cricket::GetFirstDataContent(desc); |
| RTC_DCHECK(data_content); |
| auto* data_desc = data_content->media_description()->as_data(); |
| data_desc->set_codecs({sctp_codec}); |
| } |
| |
| std::unique_ptr<rtc::VirtualSocketServer> vss_; |
| rtc::AutoSocketServerThread main_; |
| const SdpSemantics sdp_semantics_; |
| }; |
| |
| class PeerConnectionDataChannelTest |
| : public PeerConnectionDataChannelBaseTest, |
| public ::testing::WithParamInterface<SdpSemantics> { |
| protected: |
| PeerConnectionDataChannelTest() |
| : PeerConnectionDataChannelBaseTest(GetParam()) {} |
| }; |
| |
| 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 that sctp_content_name/sctp_transport_name (used for stats) are correct |
| // before and after BUNDLE is negotiated. |
| TEST_P(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) { |
| auto caller = CreatePeerConnection(); |
| auto callee = CreatePeerConnection(); |
| |
| // Initially these fields should be empty. |
| EXPECT_FALSE(caller->sctp_content_name()); |
| EXPECT_FALSE(caller->sctp_transport_name()); |
| |
| // Create offer with audio/video/data. |
| // Default bundle policy is "balanced", so data should be using its own |
| // transport. |
| caller->AddAudioTrack("a"); |
| caller->AddVideoTrack("v"); |
| caller->pc()->CreateDataChannel("dc", nullptr); |
| |
| auto offer = caller->CreateOffer(); |
| const auto& offer_contents = offer->description()->contents(); |
| ASSERT_EQ(cricket::MEDIA_TYPE_AUDIO, |
| offer_contents[0].media_description()->type()); |
| std::string audio_mid = offer_contents[0].name; |
| ASSERT_EQ(cricket::MEDIA_TYPE_DATA, |
| offer_contents[2].media_description()->type()); |
| std::string data_mid = offer_contents[2].name; |
| |
| ASSERT_TRUE( |
| caller->SetLocalDescription(CloneSessionDescription(offer.get()))); |
| ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); |
| |
| ASSERT_TRUE(caller->sctp_content_name()); |
| EXPECT_EQ(data_mid, *caller->sctp_content_name()); |
| ASSERT_TRUE(caller->sctp_transport_name()); |
| EXPECT_EQ(data_mid, *caller->sctp_transport_name()); |
| |
| // Create answer that finishes BUNDLE negotiation, which means everything |
| // should be bundled on the first transport (audio). |
| RTCOfferAnswerOptions options; |
| options.use_rtp_mux = true; |
| ASSERT_TRUE( |
| caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal())); |
| |
| ASSERT_TRUE(caller->sctp_content_name()); |
| EXPECT_EQ(data_mid, *caller->sctp_content_name()); |
| ASSERT_TRUE(caller->sctp_transport_name()); |
| EXPECT_EQ(audio_mid, *caller->sctp_transport_name()); |
| } |
| |
| TEST_P(PeerConnectionDataChannelTest, |
| CreateOfferWithNoDataChannelsGivesNoDataSection) { |
| auto caller = CreatePeerConnection(); |
| auto offer = caller->CreateOffer(); |
| |
| EXPECT_FALSE(offer->description()->GetContentByName(cricket::CN_DATA)); |
| EXPECT_FALSE(offer->description()->GetTransportInfoByName(cricket::CN_DATA)); |
| } |
| |
| TEST_P(PeerConnectionDataChannelTest, |
| CreateAnswerWithRemoteSctpDataChannelIncludesDataSection) { |
| auto caller = CreatePeerConnectionWithDataChannel(); |
| auto callee = CreatePeerConnection(); |
| |
| ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| |
| auto answer = callee->CreateAnswer(); |
| ASSERT_TRUE(answer); |
| auto* data_content = cricket::GetFirstDataContent(answer->description()); |
| ASSERT_TRUE(data_content); |
| EXPECT_FALSE(data_content->rejected); |
| EXPECT_TRUE( |
| answer->description()->GetTransportInfoByName(data_content->name)); |
| } |
| |
| TEST_P(PeerConnectionDataChannelTest, |
| CreateDataChannelWithDtlsDisabledSucceeds) { |
| RTCConfiguration config; |
| config.enable_dtls_srtp.emplace(false); |
| auto caller = CreatePeerConnection(); |
| |
| EXPECT_TRUE(caller->pc()->CreateDataChannel("dc", nullptr)); |
| } |
| |
| TEST_P(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) { |
| PeerConnectionFactoryInterface::Options options; |
| options.disable_sctp_data_channels = true; |
| auto caller = CreatePeerConnection(RTCConfiguration(), options); |
| |
| EXPECT_FALSE(caller->pc()->CreateDataChannel("dc", nullptr)); |
| } |
| |
| // Test that if a callee has SCTP disabled and receives an offer with an SCTP |
| // data channel, the data section is rejected and no SCTP transport is created |
| // on the callee. |
| TEST_P(PeerConnectionDataChannelTest, |
| DataSectionRejectedIfCalleeHasSctpDisabled) { |
| auto caller = CreatePeerConnectionWithDataChannel(); |
| PeerConnectionFactoryInterface::Options options; |
| options.disable_sctp_data_channels = true; |
| auto callee = CreatePeerConnection(RTCConfiguration(), options); |
| |
| ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| |
| EXPECT_FALSE(callee->sctp_transport_factory()->last_fake_sctp_transport()); |
| |
| auto answer = callee->CreateAnswer(); |
| auto* data_content = cricket::GetFirstDataContent(answer->description()); |
| ASSERT_TRUE(data_content); |
| EXPECT_TRUE(data_content->rejected); |
| } |
| |
| TEST_P(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) { |
| constexpr int kNewSendPort = 9998; |
| constexpr int kNewRecvPort = 7775; |
| |
| auto caller = CreatePeerConnectionWithDataChannel(); |
| auto callee = CreatePeerConnectionWithDataChannel(); |
| |
| auto offer = caller->CreateOffer(); |
| ChangeSctpPortOnDescription(offer->description(), kNewSendPort); |
| ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); |
| |
| auto answer = callee->CreateAnswer(); |
| ChangeSctpPortOnDescription(answer->description(), kNewRecvPort); |
| ASSERT_TRUE(callee->SetLocalDescription(std::move(answer))); |
| |
| auto* callee_transport = |
| callee->sctp_transport_factory()->last_fake_sctp_transport(); |
| ASSERT_TRUE(callee_transport); |
| EXPECT_EQ(kNewSendPort, callee_transport->remote_port()); |
| EXPECT_EQ(kNewRecvPort, callee_transport->local_port()); |
| } |
| |
| TEST_P(PeerConnectionDataChannelTest, |
| NoSctpTransportCreatedIfMediaTransportDataChannelsEnabled) { |
| RTCConfiguration config; |
| config.use_media_transport_for_data_channels = true; |
| config.enable_dtls_srtp = false; // SDES is required to use media transport. |
| auto caller = CreatePeerConnectionWithDataChannel(config); |
| |
| ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer())); |
| EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport()); |
| } |
| |
| TEST_P(PeerConnectionDataChannelTest, |
| MediaTransportDataChannelCreatedEvenIfSctpAvailable) { |
| RTCConfiguration config; |
| config.use_media_transport_for_data_channels = true; |
| config.enable_dtls_srtp = false; // SDES is required to use media transport. |
| 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, |
| CannotEnableBothMediaTransportAndRtpDataChannels) { |
| RTCConfiguration config; |
| config.enable_rtp_data_channel = true; |
| config.use_media_transport_for_data_channels = true; |
| config.enable_dtls_srtp = false; // SDES is required to use media transport. |
| EXPECT_EQ(CreatePeerConnection(config), nullptr); |
| } |
| |
| // This test now DCHECKs, instead of failing to SetLocalDescription. |
| TEST_P(PeerConnectionDataChannelTest, MediaTransportWithoutSdesFails) { |
| RTCConfiguration config; |
| config.use_media_transport_for_data_channels = true; |
| config.enable_dtls_srtp = true; // Disables SDES for data sections. |
| |
| auto caller = CreatePeerConnectionWithDataChannel(config); |
| |
| EXPECT_EQ(nullptr, caller); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P(PeerConnectionDataChannelTest, |
| PeerConnectionDataChannelTest, |
| Values(SdpSemantics::kPlanB, |
| SdpSemantics::kUnifiedPlan)); |
| |
| } // namespace webrtc |