Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -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 | |
Karl Wiberg | 32df86e | 2017-11-03 10:24:27 +0100 | [diff] [blame] | 11 | #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| 12 | #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -0700 | [diff] [blame] | 13 | #include "api/peerconnectionproxy.h" |
| 14 | #include "p2p/base/fakeportallocator.h" |
| 15 | #include "p2p/base/teststunserver.h" |
| 16 | #include "p2p/client/basicportallocator.h" |
| 17 | #include "pc/mediasession.h" |
| 18 | #include "pc/peerconnection.h" |
| 19 | #include "pc/peerconnectionwrapper.h" |
| 20 | #include "pc/sdputils.h" |
| 21 | #ifdef WEBRTC_ANDROID |
| 22 | #include "pc/test/androidtestinitializer.h" |
| 23 | #endif |
| 24 | #include "pc/test/fakeaudiocapturemodule.h" |
| 25 | #include "rtc_base/fakenetwork.h" |
| 26 | #include "rtc_base/gunit.h" |
| 27 | #include "rtc_base/ptr_util.h" |
| 28 | #include "rtc_base/virtualsocketserver.h" |
| 29 | #include "test/gmock.h" |
| 30 | |
| 31 | namespace webrtc { |
| 32 | |
| 33 | using BundlePolicy = PeerConnectionInterface::BundlePolicy; |
| 34 | using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; |
| 35 | using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions; |
| 36 | using RtcpMuxPolicy = PeerConnectionInterface::RtcpMuxPolicy; |
| 37 | using rtc::SocketAddress; |
| 38 | using ::testing::ElementsAre; |
| 39 | using ::testing::UnorderedElementsAre; |
| 40 | using ::testing::Values; |
| 41 | |
| 42 | constexpr int kDefaultTimeout = 10000; |
| 43 | |
| 44 | // TODO(steveanton): These tests should be rewritten to use the standard |
| 45 | // RtpSenderInterface/DtlsTransportInterface objects once they're available in |
| 46 | // the API. The RtpSender can be used to determine which transport a given media |
| 47 | // will use: https://www.w3.org/TR/webrtc/#dom-rtcrtpsender-transport |
| 48 | |
| 49 | class PeerConnectionWrapperForBundleTest : public PeerConnectionWrapper { |
| 50 | public: |
| 51 | using PeerConnectionWrapper::PeerConnectionWrapper; |
| 52 | |
| 53 | bool AddIceCandidateToMedia(cricket::Candidate* candidate, |
| 54 | cricket::MediaType media_type) { |
| 55 | auto* desc = pc()->remote_description()->description(); |
| 56 | for (size_t i = 0; i < desc->contents().size(); i++) { |
| 57 | const auto& content = desc->contents()[i]; |
| 58 | auto* media_desc = |
| 59 | static_cast<cricket::MediaContentDescription*>(content.description); |
| 60 | if (media_desc->type() == media_type) { |
| 61 | candidate->set_transport_name(content.name); |
| 62 | JsepIceCandidate jsep_candidate(content.name, i, *candidate); |
| 63 | return pc()->AddIceCandidate(&jsep_candidate); |
| 64 | } |
| 65 | } |
| 66 | RTC_NOTREACHED(); |
| 67 | return false; |
| 68 | } |
| 69 | |
| 70 | rtc::PacketTransportInternal* voice_rtp_transport_channel() { |
| 71 | return (voice_channel() ? voice_channel()->rtp_dtls_transport() : nullptr); |
| 72 | } |
| 73 | |
| 74 | rtc::PacketTransportInternal* voice_rtcp_transport_channel() { |
| 75 | return (voice_channel() ? voice_channel()->rtcp_dtls_transport() : nullptr); |
| 76 | } |
| 77 | |
| 78 | cricket::VoiceChannel* voice_channel() { |
| 79 | return GetInternalPeerConnection()->voice_channel(); |
| 80 | } |
| 81 | |
| 82 | rtc::PacketTransportInternal* video_rtp_transport_channel() { |
| 83 | return (video_channel() ? video_channel()->rtp_dtls_transport() : nullptr); |
| 84 | } |
| 85 | |
| 86 | rtc::PacketTransportInternal* video_rtcp_transport_channel() { |
| 87 | return (video_channel() ? video_channel()->rtcp_dtls_transport() : nullptr); |
| 88 | } |
| 89 | |
| 90 | cricket::VideoChannel* video_channel() { |
| 91 | return GetInternalPeerConnection()->video_channel(); |
| 92 | } |
| 93 | |
| 94 | PeerConnection* GetInternalPeerConnection() { |
| 95 | auto* pci = reinterpret_cast< |
| 96 | PeerConnectionProxyWithInternal<PeerConnectionInterface>*>(pc()); |
| 97 | return reinterpret_cast<PeerConnection*>(pci->internal()); |
| 98 | } |
| 99 | |
| 100 | // Returns true if the stats indicate that an ICE connection is either in |
| 101 | // progress or established with the given remote address. |
| 102 | bool HasConnectionWithRemoteAddress(const SocketAddress& address) { |
| 103 | auto report = GetStats(); |
| 104 | if (!report) { |
| 105 | return false; |
| 106 | } |
| 107 | std::string matching_candidate_id; |
| 108 | for (auto* ice_candidate_stats : |
| 109 | report->GetStatsOfType<RTCRemoteIceCandidateStats>()) { |
| 110 | if (*ice_candidate_stats->ip == address.HostAsURIString() && |
| 111 | *ice_candidate_stats->port == address.port()) { |
| 112 | matching_candidate_id = ice_candidate_stats->id(); |
| 113 | break; |
| 114 | } |
| 115 | } |
| 116 | if (matching_candidate_id.empty()) { |
| 117 | return false; |
| 118 | } |
| 119 | for (auto* pair_stats : |
| 120 | report->GetStatsOfType<RTCIceCandidatePairStats>()) { |
| 121 | if (*pair_stats->remote_candidate_id == matching_candidate_id) { |
| 122 | if (*pair_stats->state == RTCStatsIceCandidatePairState::kInProgress || |
| 123 | *pair_stats->state == RTCStatsIceCandidatePairState::kSucceeded) { |
| 124 | return true; |
| 125 | } |
| 126 | } |
| 127 | } |
| 128 | return false; |
| 129 | } |
| 130 | |
| 131 | rtc::FakeNetworkManager* network() { return network_; } |
| 132 | |
| 133 | void set_network(rtc::FakeNetworkManager* network) { network_ = network; } |
| 134 | |
| 135 | private: |
| 136 | rtc::FakeNetworkManager* network_; |
| 137 | }; |
| 138 | |
| 139 | class PeerConnectionBundleTest : public ::testing::Test { |
| 140 | protected: |
| 141 | typedef std::unique_ptr<PeerConnectionWrapperForBundleTest> WrapperPtr; |
| 142 | |
| 143 | PeerConnectionBundleTest() |
| 144 | : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) { |
| 145 | #ifdef WEBRTC_ANDROID |
| 146 | InitializeAndroidObjects(); |
| 147 | #endif |
| 148 | pc_factory_ = CreatePeerConnectionFactory( |
| 149 | rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), |
Karl Wiberg | 32df86e | 2017-11-03 10:24:27 +0100 | [diff] [blame] | 150 | FakeAudioCaptureModule::Create(), CreateBuiltinAudioEncoderFactory(), |
| 151 | CreateBuiltinAudioDecoderFactory(), nullptr, nullptr); |
Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -0700 | [diff] [blame] | 152 | } |
| 153 | |
| 154 | WrapperPtr CreatePeerConnection() { |
| 155 | return CreatePeerConnection(RTCConfiguration()); |
| 156 | } |
| 157 | |
| 158 | WrapperPtr CreatePeerConnection(const RTCConfiguration& config) { |
| 159 | auto* fake_network = NewFakeNetwork(); |
| 160 | auto port_allocator = |
| 161 | rtc::MakeUnique<cricket::BasicPortAllocator>(fake_network); |
| 162 | port_allocator->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP | |
| 163 | cricket::PORTALLOCATOR_DISABLE_RELAY); |
| 164 | port_allocator->set_step_delay(cricket::kMinimumStepDelay); |
| 165 | auto observer = rtc::MakeUnique<MockPeerConnectionObserver>(); |
| 166 | auto pc = pc_factory_->CreatePeerConnection( |
| 167 | config, std::move(port_allocator), nullptr, observer.get()); |
| 168 | if (!pc) { |
| 169 | return nullptr; |
| 170 | } |
| 171 | |
| 172 | auto wrapper = rtc::MakeUnique<PeerConnectionWrapperForBundleTest>( |
| 173 | pc_factory_, pc, std::move(observer)); |
| 174 | wrapper->set_network(fake_network); |
| 175 | return wrapper; |
| 176 | } |
| 177 | |
| 178 | // Accepts the same arguments as CreatePeerConnection and adds default audio |
| 179 | // and video tracks. |
| 180 | template <typename... Args> |
| 181 | WrapperPtr CreatePeerConnectionWithAudioVideo(Args&&... args) { |
| 182 | auto wrapper = CreatePeerConnection(std::forward<Args>(args)...); |
| 183 | if (!wrapper) { |
| 184 | return nullptr; |
| 185 | } |
| 186 | wrapper->AddAudioTrack("a"); |
| 187 | wrapper->AddVideoTrack("v"); |
| 188 | return wrapper; |
| 189 | } |
| 190 | |
| 191 | cricket::Candidate CreateLocalUdpCandidate( |
| 192 | const rtc::SocketAddress& address) { |
| 193 | cricket::Candidate candidate; |
| 194 | candidate.set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT); |
| 195 | candidate.set_protocol(cricket::UDP_PROTOCOL_NAME); |
| 196 | candidate.set_address(address); |
| 197 | candidate.set_type(cricket::LOCAL_PORT_TYPE); |
| 198 | return candidate; |
| 199 | } |
| 200 | |
| 201 | rtc::FakeNetworkManager* NewFakeNetwork() { |
| 202 | // The PeerConnection's port allocator is tied to the PeerConnection's |
| 203 | // lifetime and expects the underlying NetworkManager to outlive it. If |
| 204 | // PeerConnectionWrapper owned the NetworkManager, it would be destroyed |
| 205 | // before the PeerConnection (since subclass members are destroyed before |
| 206 | // base class members). Therefore, the test fixture will own all the fake |
| 207 | // networks even though tests should access the fake network through the |
| 208 | // PeerConnectionWrapper. |
| 209 | auto* fake_network = new rtc::FakeNetworkManager(); |
| 210 | fake_networks_.emplace_back(fake_network); |
| 211 | return fake_network; |
| 212 | } |
| 213 | |
| 214 | std::unique_ptr<rtc::VirtualSocketServer> vss_; |
| 215 | rtc::AutoSocketServerThread main_; |
| 216 | rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_; |
| 217 | std::vector<std::unique_ptr<rtc::FakeNetworkManager>> fake_networks_; |
| 218 | }; |
| 219 | |
| 220 | SdpContentMutator RemoveRtcpMux() { |
| 221 | return [](cricket::ContentInfo* content, cricket::TransportInfo* transport) { |
| 222 | auto* media_desc = |
| 223 | static_cast<cricket::MediaContentDescription*>(content->description); |
| 224 | media_desc->set_rtcp_mux(false); |
| 225 | }; |
| 226 | } |
| 227 | |
| 228 | std::vector<int> GetCandidateComponents( |
| 229 | const std::vector<IceCandidateInterface*> candidates) { |
| 230 | std::vector<int> components; |
| 231 | for (auto* candidate : candidates) { |
| 232 | components.push_back(candidate->candidate().component()); |
| 233 | } |
| 234 | return components; |
| 235 | } |
| 236 | |
| 237 | // Test that there are 2 local UDP candidates (1 RTP and 1 RTCP candidate) for |
| 238 | // each media section when disabling bundling and disabling RTCP multiplexing. |
| 239 | TEST_F(PeerConnectionBundleTest, |
| 240 | TwoCandidatesForEachTransportWhenNoBundleNoRtcpMux) { |
| 241 | const SocketAddress kCallerAddress("1.1.1.1", 0); |
| 242 | const SocketAddress kCalleeAddress("2.2.2.2", 0); |
| 243 | |
| 244 | RTCConfiguration config; |
| 245 | config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate; |
| 246 | auto caller = CreatePeerConnectionWithAudioVideo(config); |
| 247 | caller->network()->AddInterface(kCallerAddress); |
| 248 | auto callee = CreatePeerConnectionWithAudioVideo(config); |
| 249 | callee->network()->AddInterface(kCalleeAddress); |
| 250 | |
| 251 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 252 | RTCOfferAnswerOptions options_no_bundle; |
| 253 | options_no_bundle.use_rtp_mux = false; |
| 254 | auto answer = callee->CreateAnswer(options_no_bundle); |
| 255 | SdpContentsForEach(RemoveRtcpMux(), answer->description()); |
| 256 | ASSERT_TRUE( |
| 257 | callee->SetLocalDescription(CloneSessionDescription(answer.get()))); |
| 258 | ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer))); |
| 259 | |
| 260 | // Check that caller has separate RTP and RTCP candidates for each media. |
| 261 | EXPECT_TRUE_WAIT(caller->IsIceGatheringDone(), kDefaultTimeout); |
| 262 | EXPECT_THAT( |
| 263 | GetCandidateComponents(caller->observer()->GetCandidatesByMline(0)), |
| 264 | UnorderedElementsAre(cricket::ICE_CANDIDATE_COMPONENT_RTP, |
| 265 | cricket::ICE_CANDIDATE_COMPONENT_RTCP)); |
| 266 | EXPECT_THAT( |
| 267 | GetCandidateComponents(caller->observer()->GetCandidatesByMline(1)), |
| 268 | UnorderedElementsAre(cricket::ICE_CANDIDATE_COMPONENT_RTP, |
| 269 | cricket::ICE_CANDIDATE_COMPONENT_RTCP)); |
| 270 | |
| 271 | // Check that callee has separate RTP and RTCP candidates for each media. |
| 272 | EXPECT_TRUE_WAIT(callee->IsIceGatheringDone(), kDefaultTimeout); |
| 273 | EXPECT_THAT( |
| 274 | GetCandidateComponents(callee->observer()->GetCandidatesByMline(0)), |
| 275 | UnorderedElementsAre(cricket::ICE_CANDIDATE_COMPONENT_RTP, |
| 276 | cricket::ICE_CANDIDATE_COMPONENT_RTCP)); |
| 277 | EXPECT_THAT( |
| 278 | GetCandidateComponents(callee->observer()->GetCandidatesByMline(1)), |
| 279 | UnorderedElementsAre(cricket::ICE_CANDIDATE_COMPONENT_RTP, |
| 280 | cricket::ICE_CANDIDATE_COMPONENT_RTCP)); |
| 281 | } |
| 282 | |
| 283 | // Test that there is 1 local UDP candidate for both RTP and RTCP for each media |
| 284 | // section when disabling bundle but enabling RTCP multiplexing. |
| 285 | TEST_F(PeerConnectionBundleTest, |
| 286 | OneCandidateForEachTransportWhenNoBundleButRtcpMux) { |
| 287 | const SocketAddress kCallerAddress("1.1.1.1", 0); |
| 288 | |
| 289 | auto caller = CreatePeerConnectionWithAudioVideo(); |
| 290 | caller->network()->AddInterface(kCallerAddress); |
| 291 | auto callee = CreatePeerConnectionWithAudioVideo(); |
| 292 | |
| 293 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 294 | RTCOfferAnswerOptions options_no_bundle; |
| 295 | options_no_bundle.use_rtp_mux = false; |
| 296 | ASSERT_TRUE( |
| 297 | caller->SetRemoteDescription(callee->CreateAnswer(options_no_bundle))); |
| 298 | |
| 299 | EXPECT_TRUE_WAIT(caller->IsIceGatheringDone(), kDefaultTimeout); |
| 300 | |
| 301 | EXPECT_EQ(1u, caller->observer()->GetCandidatesByMline(0).size()); |
| 302 | EXPECT_EQ(1u, caller->observer()->GetCandidatesByMline(1).size()); |
| 303 | } |
| 304 | |
| 305 | // Test that there is 1 local UDP candidate in only the first media section when |
| 306 | // bundling and enabling RTCP multiplexing. |
| 307 | TEST_F(PeerConnectionBundleTest, |
| 308 | OneCandidateOnlyOnFirstTransportWhenBundleAndRtcpMux) { |
| 309 | const SocketAddress kCallerAddress("1.1.1.1", 0); |
| 310 | |
| 311 | RTCConfiguration config; |
| 312 | config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle; |
| 313 | auto caller = CreatePeerConnectionWithAudioVideo(config); |
| 314 | caller->network()->AddInterface(kCallerAddress); |
| 315 | auto callee = CreatePeerConnectionWithAudioVideo(config); |
| 316 | |
| 317 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 318 | ASSERT_TRUE(caller->SetRemoteDescription(callee->CreateAnswer())); |
| 319 | |
| 320 | EXPECT_TRUE_WAIT(caller->IsIceGatheringDone(), kDefaultTimeout); |
| 321 | |
| 322 | EXPECT_EQ(1u, caller->observer()->GetCandidatesByMline(0).size()); |
| 323 | EXPECT_EQ(0u, caller->observer()->GetCandidatesByMline(1).size()); |
| 324 | } |
| 325 | |
| 326 | // The following parameterized test verifies that an offer/answer with varying |
| 327 | // bundle policies and either bundle in the answer or not will produce the |
| 328 | // expected RTP transports for audio and video. In particular, for bundling we |
| 329 | // care about whether they are separate transports or the same. |
| 330 | |
| 331 | enum class BundleIncluded { kBundleInAnswer, kBundleNotInAnswer }; |
| 332 | std::ostream& operator<<(std::ostream& out, BundleIncluded value) { |
| 333 | switch (value) { |
| 334 | case BundleIncluded::kBundleInAnswer: |
| 335 | return out << "bundle in answer"; |
| 336 | case BundleIncluded::kBundleNotInAnswer: |
| 337 | return out << "bundle not in answer"; |
| 338 | } |
| 339 | return out << "unknown"; |
| 340 | } |
| 341 | |
| 342 | class PeerConnectionBundleMatrixTest |
| 343 | : public PeerConnectionBundleTest, |
| 344 | public ::testing::WithParamInterface< |
| 345 | std::tuple<BundlePolicy, BundleIncluded, bool, bool>> { |
| 346 | protected: |
| 347 | PeerConnectionBundleMatrixTest() { |
| 348 | bundle_policy_ = std::get<0>(GetParam()); |
| 349 | bundle_included_ = std::get<1>(GetParam()); |
| 350 | expected_same_before_ = std::get<2>(GetParam()); |
| 351 | expected_same_after_ = std::get<3>(GetParam()); |
| 352 | } |
| 353 | |
| 354 | PeerConnectionInterface::BundlePolicy bundle_policy_; |
| 355 | BundleIncluded bundle_included_; |
| 356 | bool expected_same_before_; |
| 357 | bool expected_same_after_; |
| 358 | }; |
| 359 | |
| 360 | TEST_P(PeerConnectionBundleMatrixTest, |
| 361 | VerifyTransportsBeforeAndAfterSettingRemoteAnswer) { |
| 362 | RTCConfiguration config; |
| 363 | config.bundle_policy = bundle_policy_; |
| 364 | auto caller = CreatePeerConnectionWithAudioVideo(config); |
| 365 | auto callee = CreatePeerConnectionWithAudioVideo(); |
| 366 | |
| 367 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 368 | bool equal_before = (caller->voice_rtp_transport_channel() == |
| 369 | caller->video_rtp_transport_channel()); |
| 370 | EXPECT_EQ(expected_same_before_, equal_before); |
| 371 | |
| 372 | RTCOfferAnswerOptions options; |
| 373 | options.use_rtp_mux = (bundle_included_ == BundleIncluded::kBundleInAnswer); |
| 374 | ASSERT_TRUE( |
| 375 | caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options))); |
| 376 | bool equal_after = (caller->voice_rtp_transport_channel() == |
| 377 | caller->video_rtp_transport_channel()); |
| 378 | EXPECT_EQ(expected_same_after_, equal_after); |
| 379 | } |
| 380 | |
| 381 | // The max-bundle policy means we should anticipate bundling being negotiated, |
| 382 | // and multiplex audio/video from the start. |
| 383 | // For all other policies, bundling should only be enabled if negotiated by the |
| 384 | // answer. |
| 385 | INSTANTIATE_TEST_CASE_P( |
| 386 | PeerConnectionBundleTest, |
| 387 | PeerConnectionBundleMatrixTest, |
| 388 | Values(std::make_tuple(BundlePolicy::kBundlePolicyBalanced, |
| 389 | BundleIncluded::kBundleInAnswer, |
| 390 | false, |
| 391 | true), |
| 392 | std::make_tuple(BundlePolicy::kBundlePolicyBalanced, |
| 393 | BundleIncluded::kBundleNotInAnswer, |
| 394 | false, |
| 395 | false), |
| 396 | std::make_tuple(BundlePolicy::kBundlePolicyMaxBundle, |
| 397 | BundleIncluded::kBundleInAnswer, |
| 398 | true, |
| 399 | true), |
| 400 | std::make_tuple(BundlePolicy::kBundlePolicyMaxBundle, |
| 401 | BundleIncluded::kBundleNotInAnswer, |
| 402 | true, |
| 403 | true), |
| 404 | std::make_tuple(BundlePolicy::kBundlePolicyMaxCompat, |
| 405 | BundleIncluded::kBundleInAnswer, |
| 406 | false, |
| 407 | true), |
| 408 | std::make_tuple(BundlePolicy::kBundlePolicyMaxCompat, |
| 409 | BundleIncluded::kBundleNotInAnswer, |
| 410 | false, |
| 411 | false))); |
| 412 | |
| 413 | // Test that the audio/video transports on the callee side are the same before |
| 414 | // and after setting a local answer when max BUNDLE is enabled and an offer with |
| 415 | // BUNDLE is received. |
| 416 | TEST_F(PeerConnectionBundleTest, |
| 417 | TransportsSameForMaxBundleWithBundleInRemoteOffer) { |
| 418 | auto caller = CreatePeerConnectionWithAudioVideo(); |
| 419 | RTCConfiguration config; |
| 420 | config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle; |
| 421 | auto callee = CreatePeerConnectionWithAudioVideo(config); |
| 422 | |
| 423 | RTCOfferAnswerOptions options_with_bundle; |
| 424 | options_with_bundle.use_rtp_mux = true; |
| 425 | ASSERT_TRUE(callee->SetRemoteDescription( |
| 426 | caller->CreateOfferAndSetAsLocal(options_with_bundle))); |
| 427 | |
| 428 | EXPECT_EQ(callee->voice_rtp_transport_channel(), |
| 429 | callee->video_rtp_transport_channel()); |
| 430 | |
| 431 | ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer())); |
| 432 | |
| 433 | EXPECT_EQ(callee->voice_rtp_transport_channel(), |
| 434 | callee->video_rtp_transport_channel()); |
| 435 | } |
| 436 | |
| 437 | TEST_F(PeerConnectionBundleTest, |
| 438 | FailToSetRemoteOfferWithNoBundleWhenBundlePolicyMaxBundle) { |
| 439 | auto caller = CreatePeerConnectionWithAudioVideo(); |
| 440 | RTCConfiguration config; |
| 441 | config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle; |
| 442 | auto callee = CreatePeerConnectionWithAudioVideo(config); |
| 443 | |
| 444 | RTCOfferAnswerOptions options_no_bundle; |
| 445 | options_no_bundle.use_rtp_mux = false; |
| 446 | EXPECT_FALSE(callee->SetRemoteDescription( |
| 447 | caller->CreateOfferAndSetAsLocal(options_no_bundle))); |
| 448 | } |
| 449 | |
| 450 | // Test that if the media section which has the bundled transport is rejected, |
| 451 | // then the peers still connect and the bundled transport switches to the other |
| 452 | // media section. |
| 453 | // Note: This is currently failing because of the following bug: |
| 454 | // https://bugs.chromium.org/p/webrtc/issues/detail?id=6280 |
| 455 | TEST_F(PeerConnectionBundleTest, |
| 456 | DISABLED_SuccessfullyNegotiateMaxBundleIfBundleTransportMediaRejected) { |
| 457 | RTCConfiguration config; |
| 458 | config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle; |
| 459 | auto caller = CreatePeerConnectionWithAudioVideo(config); |
| 460 | auto callee = CreatePeerConnection(); |
| 461 | callee->AddVideoTrack("v"); |
| 462 | |
| 463 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 464 | |
| 465 | RTCOfferAnswerOptions options; |
| 466 | options.offer_to_receive_audio = 0; |
| 467 | ASSERT_TRUE( |
| 468 | caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options))); |
| 469 | |
| 470 | EXPECT_FALSE(caller->voice_rtp_transport_channel()); |
| 471 | EXPECT_TRUE(caller->video_rtp_transport_channel()); |
| 472 | } |
| 473 | |
| 474 | // When requiring RTCP multiplexing, the PeerConnection never makes RTCP |
| 475 | // transport channels. |
| 476 | TEST_F(PeerConnectionBundleTest, NeverCreateRtcpTransportWithRtcpMuxRequired) { |
| 477 | RTCConfiguration config; |
| 478 | config.rtcp_mux_policy = RtcpMuxPolicy::kRtcpMuxPolicyRequire; |
| 479 | auto caller = CreatePeerConnectionWithAudioVideo(config); |
| 480 | auto callee = CreatePeerConnectionWithAudioVideo(); |
| 481 | |
| 482 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 483 | |
| 484 | EXPECT_FALSE(caller->voice_rtcp_transport_channel()); |
| 485 | EXPECT_FALSE(caller->video_rtcp_transport_channel()); |
| 486 | |
| 487 | ASSERT_TRUE( |
| 488 | caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal())); |
| 489 | |
| 490 | EXPECT_FALSE(caller->voice_rtcp_transport_channel()); |
| 491 | EXPECT_FALSE(caller->video_rtcp_transport_channel()); |
| 492 | } |
| 493 | |
| 494 | // When negotiating RTCP multiplexing, the PeerConnection makes RTCP transport |
| 495 | // channels when the offer is sent, but will destroy them once the remote answer |
| 496 | // is set. |
| 497 | TEST_F(PeerConnectionBundleTest, |
| 498 | CreateRtcpTransportOnlyBeforeAnswerWithRtcpMuxNegotiate) { |
| 499 | RTCConfiguration config; |
| 500 | config.rtcp_mux_policy = RtcpMuxPolicy::kRtcpMuxPolicyNegotiate; |
| 501 | auto caller = CreatePeerConnectionWithAudioVideo(config); |
| 502 | auto callee = CreatePeerConnectionWithAudioVideo(); |
| 503 | |
| 504 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 505 | |
| 506 | EXPECT_TRUE(caller->voice_rtcp_transport_channel()); |
| 507 | EXPECT_TRUE(caller->video_rtcp_transport_channel()); |
| 508 | |
| 509 | ASSERT_TRUE( |
| 510 | caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal())); |
| 511 | |
| 512 | EXPECT_FALSE(caller->voice_rtcp_transport_channel()); |
| 513 | EXPECT_FALSE(caller->video_rtcp_transport_channel()); |
| 514 | } |
| 515 | |
| 516 | TEST_F(PeerConnectionBundleTest, FailToSetDescriptionWithBundleAndNoRtcpMux) { |
| 517 | auto caller = CreatePeerConnectionWithAudioVideo(); |
| 518 | auto callee = CreatePeerConnectionWithAudioVideo(); |
| 519 | |
| 520 | RTCOfferAnswerOptions options; |
| 521 | options.use_rtp_mux = true; |
| 522 | |
| 523 | auto offer = caller->CreateOffer(options); |
| 524 | SdpContentsForEach(RemoveRtcpMux(), offer->description()); |
| 525 | |
| 526 | std::string error; |
| 527 | EXPECT_FALSE(caller->SetLocalDescription(CloneSessionDescription(offer.get()), |
| 528 | &error)); |
| 529 | EXPECT_EQ( |
| 530 | "Failed to set local offer sdp: rtcp-mux must be enabled when BUNDLE is " |
| 531 | "enabled.", |
| 532 | error); |
| 533 | |
| 534 | EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer), &error)); |
| 535 | EXPECT_EQ( |
| 536 | "Failed to set remote offer sdp: rtcp-mux must be enabled when BUNDLE is " |
| 537 | "enabled.", |
| 538 | error); |
| 539 | } |
| 540 | |
| 541 | // Test that candidates sent to the "video" transport do not get pushed down to |
| 542 | // the "audio" transport channel when bundling. |
| 543 | TEST_F(PeerConnectionBundleTest, |
| 544 | IgnoreCandidatesForUnusedTransportWhenBundling) { |
| 545 | const SocketAddress kAudioAddress1("1.1.1.1", 1111); |
| 546 | const SocketAddress kAudioAddress2("2.2.2.2", 2222); |
| 547 | const SocketAddress kVideoAddress("3.3.3.3", 3333); |
| 548 | const SocketAddress kCallerAddress("4.4.4.4", 0); |
| 549 | const SocketAddress kCalleeAddress("5.5.5.5", 0); |
| 550 | |
| 551 | auto caller = CreatePeerConnectionWithAudioVideo(); |
| 552 | auto callee = CreatePeerConnectionWithAudioVideo(); |
| 553 | |
| 554 | caller->network()->AddInterface(kCallerAddress); |
| 555 | callee->network()->AddInterface(kCalleeAddress); |
| 556 | |
| 557 | RTCOfferAnswerOptions options; |
| 558 | options.use_rtp_mux = true; |
| 559 | |
| 560 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 561 | ASSERT_TRUE( |
| 562 | caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal())); |
| 563 | |
| 564 | // The way the *_WAIT checks work is they only wait if the condition fails, |
| 565 | // which does not help in the case where state is not changing. This is |
| 566 | // problematic in this test since we want to verify that adding a video |
| 567 | // candidate does _not_ change state. So we interleave candidates and assume |
| 568 | // that messages are executed in the order they were posted. |
| 569 | |
| 570 | cricket::Candidate audio_candidate1 = CreateLocalUdpCandidate(kAudioAddress1); |
| 571 | ASSERT_TRUE(caller->AddIceCandidateToMedia(&audio_candidate1, |
| 572 | cricket::MEDIA_TYPE_AUDIO)); |
| 573 | |
| 574 | cricket::Candidate video_candidate = CreateLocalUdpCandidate(kVideoAddress); |
| 575 | ASSERT_TRUE(caller->AddIceCandidateToMedia(&video_candidate, |
| 576 | cricket::MEDIA_TYPE_VIDEO)); |
| 577 | |
| 578 | cricket::Candidate audio_candidate2 = CreateLocalUdpCandidate(kAudioAddress2); |
| 579 | ASSERT_TRUE(caller->AddIceCandidateToMedia(&audio_candidate2, |
| 580 | cricket::MEDIA_TYPE_AUDIO)); |
| 581 | |
| 582 | EXPECT_TRUE_WAIT(caller->HasConnectionWithRemoteAddress(kAudioAddress1), |
| 583 | kDefaultTimeout); |
| 584 | EXPECT_TRUE_WAIT(caller->HasConnectionWithRemoteAddress(kAudioAddress2), |
| 585 | kDefaultTimeout); |
| 586 | EXPECT_FALSE(caller->HasConnectionWithRemoteAddress(kVideoAddress)); |
| 587 | } |
| 588 | |
| 589 | // Test that the transport used by both audio and video is the transport |
| 590 | // associated with the first MID in the answer BUNDLE group, even if it's in a |
| 591 | // different order from the offer. |
| 592 | TEST_F(PeerConnectionBundleTest, BundleOnFirstMidInAnswer) { |
| 593 | auto caller = CreatePeerConnectionWithAudioVideo(); |
| 594 | auto callee = CreatePeerConnectionWithAudioVideo(); |
| 595 | |
| 596 | ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); |
| 597 | |
| 598 | auto* old_video_transport = caller->video_rtp_transport_channel(); |
| 599 | |
| 600 | auto answer = callee->CreateAnswer(); |
| 601 | auto* old_bundle_group = |
| 602 | answer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE); |
| 603 | ASSERT_THAT(old_bundle_group->content_names(), |
| 604 | ElementsAre(cricket::CN_AUDIO, cricket::CN_VIDEO)); |
| 605 | answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE); |
| 606 | |
| 607 | cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE); |
| 608 | new_bundle_group.AddContentName(cricket::CN_VIDEO); |
| 609 | new_bundle_group.AddContentName(cricket::CN_AUDIO); |
| 610 | answer->description()->AddGroup(new_bundle_group); |
| 611 | |
| 612 | ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer))); |
| 613 | |
| 614 | EXPECT_EQ(old_video_transport, caller->video_rtp_transport_channel()); |
| 615 | EXPECT_EQ(caller->voice_rtp_transport_channel(), |
| 616 | caller->video_rtp_transport_channel()); |
| 617 | } |
| 618 | |
| 619 | } // namespace webrtc |