/* * 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 #include "api/peerconnectionproxy.h" #include "media/base/fakemediaengine.h" #include "pc/mediasession.h" #include "pc/peerconnection.h" #include "pc/peerconnectionfactory.h" #include "pc/peerconnectionwrapper.h" #include "pc/sdputils.h" #ifdef WEBRTC_ANDROID #include "pc/test/androidtestinitializer.h" #endif #include "absl/memory/memory.h" #include "pc/test/fakesctptransport.h" #include "rtc_base/gunit.h" #include "rtc_base/virtualsocketserver.h" namespace webrtc { using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions; using ::testing::Values; class PeerConnectionFactoryForDataChannelTest : public rtc::RefCountedObject { public: PeerConnectionFactoryForDataChannelTest() : rtc::RefCountedObject( rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), absl::make_unique(), CreateCallFactory(), nullptr) {} std::unique_ptr CreateSctpTransportInternalFactory() { auto factory = absl::make_unique(); 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 sctp_content_name() { return GetInternalPeerConnection()->sctp_content_name(); } absl::optional sctp_transport_name() { return GetInternalPeerConnection()->sctp_transport_name(); } PeerConnection* GetInternalPeerConnection() { auto* pci = static_cast*>( pc()); return static_cast(pci->internal()); } private: FakeSctpTransportFactory* sctp_transport_factory_ = nullptr; }; class PeerConnectionDataChannelBaseTest : public ::testing::Test { protected: typedef std::unique_ptr 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 pc_factory( new PeerConnectionFactoryForDataChannelTest()); pc_factory->SetOptions(factory_options); RTC_CHECK(pc_factory->Initialize()); auto observer = absl::make_unique(); 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; } auto wrapper = absl::make_unique( 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 WrapperPtr CreatePeerConnectionWithDataChannel(Args&&... args) { auto wrapper = CreatePeerConnection(std::forward(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 vss_; rtc::AutoSocketServerThread main_; const SdpSemantics sdp_semantics_; }; class PeerConnectionDataChannelTest : public PeerConnectionDataChannelBaseTest, public ::testing::WithParamInterface { 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()); } INSTANTIATE_TEST_CASE_P(PeerConnectionDataChannelTest, PeerConnectionDataChannelTest, Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan)); } // namespace webrtc