Steve Anton | da6c095 | 2017-10-23 11:41:54 -0700 | [diff] [blame] | 1 | /* |
| 2 | * Copyright 2017 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #include <tuple> |
| 12 | |
| 13 | #include "api/peerconnectionproxy.h" |
| 14 | #include "media/base/fakemediaengine.h" |
| 15 | #include "pc/mediasession.h" |
| 16 | #include "pc/peerconnection.h" |
| 17 | #include "pc/peerconnectionfactory.h" |
| 18 | #include "pc/peerconnectionwrapper.h" |
| 19 | #ifdef WEBRTC_ANDROID |
| 20 | #include "pc/test/androidtestinitializer.h" |
| 21 | #endif |
| 22 | #include "pc/test/fakesctptransport.h" |
| 23 | #include "rtc_base/gunit.h" |
| 24 | #include "rtc_base/ptr_util.h" |
| 25 | #include "rtc_base/virtualsocketserver.h" |
| 26 | |
| 27 | namespace webrtc { |
| 28 | |
| 29 | using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; |
| 30 | using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions; |
| 31 | using ::testing::Values; |
| 32 | |
| 33 | class PeerConnectionFactoryForDataChannelTest |
| 34 | : public rtc::RefCountedObject<PeerConnectionFactory> { |
| 35 | public: |
| 36 | PeerConnectionFactoryForDataChannelTest() |
| 37 | : rtc::RefCountedObject<PeerConnectionFactory>( |
| 38 | rtc::Thread::Current(), |
| 39 | rtc::Thread::Current(), |
| 40 | rtc::Thread::Current(), |
| 41 | rtc::MakeUnique<cricket::FakeMediaEngine>(), |
| 42 | CreateCallFactory(), |
| 43 | nullptr) {} |
| 44 | |
| 45 | std::unique_ptr<cricket::SctpTransportInternalFactory> |
| 46 | CreateSctpTransportInternalFactory() { |
| 47 | auto factory = rtc::MakeUnique<FakeSctpTransportFactory>(); |
| 48 | last_fake_sctp_transport_factory_ = factory.get(); |
| 49 | return factory; |
| 50 | } |
| 51 | |
| 52 | FakeSctpTransportFactory* last_fake_sctp_transport_factory_ = nullptr; |
| 53 | }; |
| 54 | |
| 55 | class PeerConnectionWrapperForDataChannelTest : public PeerConnectionWrapper { |
| 56 | public: |
| 57 | using PeerConnectionWrapper::PeerConnectionWrapper; |
| 58 | |
| 59 | FakeSctpTransportFactory* sctp_transport_factory() { |
| 60 | return sctp_transport_factory_; |
| 61 | } |
| 62 | |
| 63 | void set_sctp_transport_factory( |
| 64 | FakeSctpTransportFactory* sctp_transport_factory) { |
| 65 | sctp_transport_factory_ = sctp_transport_factory; |
| 66 | } |
| 67 | |
| 68 | rtc::Optional<std::string> sctp_content_name() { |
| 69 | return GetInternalPeerConnection()->sctp_content_name(); |
| 70 | } |
| 71 | |
| 72 | rtc::Optional<std::string> sctp_transport_name() { |
| 73 | return GetInternalPeerConnection()->sctp_transport_name(); |
| 74 | } |
| 75 | |
| 76 | PeerConnection* GetInternalPeerConnection() { |
| 77 | auto* pci = reinterpret_cast< |
| 78 | PeerConnectionProxyWithInternal<PeerConnectionInterface>*>(pc()); |
| 79 | return reinterpret_cast<PeerConnection*>(pci->internal()); |
| 80 | } |
| 81 | |
| 82 | private: |
| 83 | FakeSctpTransportFactory* sctp_transport_factory_ = nullptr; |
| 84 | }; |
| 85 | |
| 86 | class PeerConnectionDataChannelTest : public ::testing::Test { |
| 87 | protected: |
| 88 | typedef std::unique_ptr<PeerConnectionWrapperForDataChannelTest> WrapperPtr; |
| 89 | |
| 90 | PeerConnectionDataChannelTest() |
| 91 | : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) { |
| 92 | #ifdef WEBRTC_ANDROID |
| 93 | InitializeAndroidObjects(); |
| 94 | #endif |
| 95 | } |
| 96 | |
| 97 | WrapperPtr CreatePeerConnection() { |
| 98 | return CreatePeerConnection(RTCConfiguration()); |
| 99 | } |
| 100 | |
| 101 | WrapperPtr CreatePeerConnection(const RTCConfiguration& config) { |
| 102 | return CreatePeerConnection(config, |
| 103 | PeerConnectionFactoryInterface::Options()); |
| 104 | } |
| 105 | |
| 106 | WrapperPtr CreatePeerConnection( |
| 107 | const RTCConfiguration& config, |
| 108 | const PeerConnectionFactoryInterface::Options factory_options) { |
| 109 | rtc::scoped_refptr<PeerConnectionFactoryForDataChannelTest> pc_factory( |
| 110 | new PeerConnectionFactoryForDataChannelTest()); |
| 111 | pc_factory->SetOptions(factory_options); |
| 112 | RTC_CHECK(pc_factory->Initialize()); |
| 113 | auto observer = rtc::MakeUnique<MockPeerConnectionObserver>(); |
| 114 | auto pc = pc_factory->CreatePeerConnection(config, nullptr, nullptr, |
| 115 | observer.get()); |
| 116 | if (!pc) { |
| 117 | return nullptr; |
| 118 | } |
| 119 | |
| 120 | auto wrapper = rtc::MakeUnique<PeerConnectionWrapperForDataChannelTest>( |
| 121 | pc_factory, pc, std::move(observer)); |
| 122 | RTC_DCHECK(pc_factory->last_fake_sctp_transport_factory_); |
| 123 | wrapper->set_sctp_transport_factory( |
| 124 | pc_factory->last_fake_sctp_transport_factory_); |
| 125 | return wrapper; |
| 126 | } |
| 127 | |
| 128 | // Accepts the same arguments as CreatePeerConnection and adds a default data |
| 129 | // channel. |
| 130 | template <typename... Args> |
| 131 | WrapperPtr CreatePeerConnectionWithDataChannel(Args&&... args) { |
| 132 | auto wrapper = CreatePeerConnection(std::forward<Args>(args)...); |
| 133 | if (!wrapper) { |
| 134 | return nullptr; |
| 135 | } |
| 136 | EXPECT_TRUE(wrapper->pc()->CreateDataChannel("dc", nullptr)); |
| 137 | return wrapper; |
| 138 | } |
| 139 | |
| 140 | // Changes the SCTP data channel port on the given session description. |
| 141 | void ChangeSctpPortOnDescription(cricket::SessionDescription* desc, |
| 142 | int port) { |
| 143 | cricket::DataCodec sctp_codec(cricket::kGoogleSctpDataCodecPlType, |
| 144 | cricket::kGoogleSctpDataCodecName); |
| 145 | sctp_codec.SetParam(cricket::kCodecParamPort, port); |
| 146 | |
| 147 | auto* data_content = cricket::GetFirstDataContent(desc); |
| 148 | RTC_DCHECK(data_content); |
| 149 | auto* data_desc = static_cast<cricket::DataContentDescription*>( |
| 150 | data_content->description); |
| 151 | data_desc->set_codecs({sctp_codec}); |
| 152 | } |
| 153 | |
| 154 | std::unique_ptr<rtc::VirtualSocketServer> vss_; |
| 155 | rtc::AutoSocketServerThread main_; |
| 156 | }; |
| 157 | |
| 158 | TEST_F(PeerConnectionDataChannelTest, |
| 159 | NoSctpTransportCreatedIfRtpDataChannelEnabled) { |
| 160 | RTCConfiguration config; |
| 161 | config.enable_rtp_data_channel = true; |
| 162 | auto caller = CreatePeerConnectionWithDataChannel(config); |
| 163 | |
| 164 | ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer())); |
| 165 | EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport()); |
| 166 | } |
| 167 | |
| 168 | TEST_F(PeerConnectionDataChannelTest, |
| 169 | RtpDataChannelCreatedEvenIfSctpAvailable) { |
| 170 | RTCConfiguration config; |
| 171 | config.enable_rtp_data_channel = true; |
| 172 | PeerConnectionFactoryInterface::Options options; |
| 173 | options.disable_sctp_data_channels = false; |
| 174 | auto caller = CreatePeerConnectionWithDataChannel(config, options); |
| 175 | |
| 176 | ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer())); |
| 177 | EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport()); |
| 178 | } |
| 179 | |
| 180 | // Test that sctp_content_name/sctp_transport_name (used for stats) are correct |
| 181 | // before and after BUNDLE is negotiated. |
| 182 | TEST_F(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) { |
| 183 | auto caller = CreatePeerConnection(); |
| 184 | auto callee = CreatePeerConnection(); |
| 185 | |
| 186 | // Initially these fields should be empty. |
| 187 | EXPECT_FALSE(caller->sctp_content_name()); |
| 188 | EXPECT_FALSE(caller->sctp_transport_name()); |
| 189 | |
| 190 | // Create offer with audio/video/data. |
| 191 | // Default bundle policy is "balanced", so data should be using its own |
| 192 | // transport. |
| 193 | caller->AddAudioTrack("a"); |
| 194 | caller->AddVideoTrack("v"); |
| 195 | caller->pc()->CreateDataChannel("dc", nullptr); |
| 196 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 197 | |
| 198 | ASSERT_TRUE(caller->sctp_content_name()); |
| 199 | EXPECT_EQ(cricket::CN_DATA, *caller->sctp_content_name()); |
| 200 | ASSERT_TRUE(caller->sctp_transport_name()); |
| 201 | EXPECT_EQ(cricket::CN_DATA, *caller->sctp_transport_name()); |
| 202 | |
| 203 | // Create answer that finishes BUNDLE negotiation, which means everything |
| 204 | // should be bundled on the first transport (audio). |
| 205 | RTCOfferAnswerOptions options; |
| 206 | options.use_rtp_mux = true; |
| 207 | ASSERT_TRUE( |
| 208 | caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal())); |
| 209 | |
| 210 | ASSERT_TRUE(caller->sctp_content_name()); |
| 211 | EXPECT_EQ(cricket::CN_DATA, *caller->sctp_content_name()); |
| 212 | ASSERT_TRUE(caller->sctp_transport_name()); |
| 213 | EXPECT_EQ(cricket::CN_AUDIO, *caller->sctp_transport_name()); |
| 214 | } |
| 215 | |
| 216 | TEST_F(PeerConnectionDataChannelTest, |
| 217 | CreateOfferWithNoDataChannelsGivesNoDataSection) { |
| 218 | auto caller = CreatePeerConnection(); |
| 219 | auto offer = caller->CreateOffer(); |
| 220 | |
| 221 | EXPECT_FALSE(offer->description()->GetContentByName(cricket::CN_DATA)); |
| 222 | EXPECT_FALSE(offer->description()->GetTransportInfoByName(cricket::CN_DATA)); |
| 223 | } |
| 224 | |
| 225 | TEST_F(PeerConnectionDataChannelTest, |
| 226 | CreateAnswerWithRemoteSctpDataChannelIncludesDataSection) { |
| 227 | auto caller = CreatePeerConnectionWithDataChannel(); |
| 228 | auto callee = CreatePeerConnection(); |
| 229 | |
| 230 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 231 | |
| 232 | auto answer = callee->CreateAnswer(); |
| 233 | ASSERT_TRUE(answer); |
| 234 | auto* data_content = |
| 235 | answer->description()->GetContentByName(cricket::CN_DATA); |
| 236 | ASSERT_TRUE(data_content); |
| 237 | EXPECT_FALSE(data_content->rejected); |
| 238 | EXPECT_TRUE(answer->description()->GetTransportInfoByName(cricket::CN_DATA)); |
| 239 | } |
| 240 | |
| 241 | TEST_F(PeerConnectionDataChannelTest, |
| 242 | CreateDataChannelWithDtlsDisabledSucceeds) { |
| 243 | RTCConfiguration config; |
| 244 | config.enable_dtls_srtp.emplace(false); |
| 245 | auto caller = CreatePeerConnection(); |
| 246 | |
| 247 | EXPECT_TRUE(caller->pc()->CreateDataChannel("dc", nullptr)); |
| 248 | } |
| 249 | |
| 250 | TEST_F(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) { |
| 251 | PeerConnectionFactoryInterface::Options options; |
| 252 | options.disable_sctp_data_channels = true; |
| 253 | auto caller = CreatePeerConnection(RTCConfiguration(), options); |
| 254 | |
| 255 | EXPECT_FALSE(caller->pc()->CreateDataChannel("dc", nullptr)); |
| 256 | } |
| 257 | |
| 258 | // Test that if a callee has SCTP disabled and receives an offer with an SCTP |
| 259 | // data channel, the data section is rejected and no SCTP transport is created |
| 260 | // on the callee. |
| 261 | TEST_F(PeerConnectionDataChannelTest, |
| 262 | DataSectionRejectedIfCalleeHasSctpDisabled) { |
| 263 | auto caller = CreatePeerConnectionWithDataChannel(); |
| 264 | PeerConnectionFactoryInterface::Options options; |
| 265 | options.disable_sctp_data_channels = true; |
| 266 | auto callee = CreatePeerConnection(RTCConfiguration(), options); |
| 267 | |
| 268 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 269 | |
| 270 | EXPECT_FALSE(callee->sctp_transport_factory()->last_fake_sctp_transport()); |
| 271 | |
| 272 | auto answer = callee->CreateAnswer(); |
| 273 | auto* data_content = |
| 274 | answer->description()->GetContentByName(cricket::CN_DATA); |
| 275 | ASSERT_TRUE(data_content); |
| 276 | EXPECT_TRUE(data_content->rejected); |
| 277 | } |
| 278 | |
| 279 | TEST_F(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) { |
| 280 | constexpr int kNewSendPort = 9998; |
| 281 | constexpr int kNewRecvPort = 7775; |
| 282 | |
| 283 | auto caller = CreatePeerConnectionWithDataChannel(); |
| 284 | auto callee = CreatePeerConnectionWithDataChannel(); |
| 285 | |
| 286 | auto offer = caller->CreateOffer(); |
| 287 | ChangeSctpPortOnDescription(offer->description(), kNewSendPort); |
| 288 | ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); |
| 289 | |
| 290 | auto answer = callee->CreateAnswer(); |
| 291 | ChangeSctpPortOnDescription(answer->description(), kNewRecvPort); |
| 292 | ASSERT_TRUE(callee->SetLocalDescription(std::move(answer))); |
| 293 | |
| 294 | auto* callee_transport = |
| 295 | callee->sctp_transport_factory()->last_fake_sctp_transport(); |
| 296 | ASSERT_TRUE(callee_transport); |
| 297 | EXPECT_EQ(kNewSendPort, callee_transport->remote_port()); |
| 298 | EXPECT_EQ(kNewRecvPort, callee_transport->local_port()); |
| 299 | } |
| 300 | |
| 301 | } // namespace webrtc |