• Home
  • Line#
  • Scopes#
  • Navigate#
  • Raw
  • Download
1 /*
2  *  Copyright 2012 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 "pc/peer_connection_factory.h"
12 
13 #include <utility>
14 #include <vector>
15 
16 #include "api/audio/audio_mixer.h"
17 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
18 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
19 #include "api/create_peerconnection_factory.h"
20 #include "api/data_channel_interface.h"
21 #include "api/jsep.h"
22 #include "api/media_stream_interface.h"
23 #include "api/test/mock_packet_socket_factory.h"
24 #include "api/video_codecs/builtin_video_decoder_factory.h"
25 #include "api/video_codecs/builtin_video_encoder_factory.h"
26 #include "media/base/fake_frame_source.h"
27 #include "modules/audio_device/include/audio_device.h"
28 #include "modules/audio_processing/include/audio_processing.h"
29 #include "p2p/base/fake_port_allocator.h"
30 #include "p2p/base/port.h"
31 #include "p2p/base/port_allocator.h"
32 #include "p2p/base/port_interface.h"
33 #include "pc/test/fake_audio_capture_module.h"
34 #include "pc/test/fake_video_track_source.h"
35 #include "pc/test/mock_peer_connection_observers.h"
36 #include "rtc_base/gunit.h"
37 #include "rtc_base/internal/default_socket_server.h"
38 #include "rtc_base/rtc_certificate_generator.h"
39 #include "rtc_base/socket_address.h"
40 #include "rtc_base/time_utils.h"
41 #include "test/gmock.h"
42 #include "test/gtest.h"
43 
44 #ifdef WEBRTC_ANDROID
45 #include "pc/test/android_test_initializer.h"
46 #endif
47 #include "pc/test/fake_rtc_certificate_generator.h"
48 #include "pc/test/fake_video_track_renderer.h"
49 
50 using webrtc::DataChannelInterface;
51 using webrtc::FakeVideoTrackRenderer;
52 using webrtc::MediaStreamInterface;
53 using webrtc::PeerConnectionFactoryInterface;
54 using webrtc::PeerConnectionInterface;
55 using webrtc::PeerConnectionObserver;
56 using webrtc::VideoTrackInterface;
57 using webrtc::VideoTrackSourceInterface;
58 
59 using ::testing::_;
60 using ::testing::AtLeast;
61 using ::testing::InvokeWithoutArgs;
62 using ::testing::NiceMock;
63 using ::testing::Return;
64 using ::testing::UnorderedElementsAre;
65 
66 namespace {
67 
68 static const char kStunIceServer[] = "stun:stun.l.google.com:19302";
69 static const char kTurnIceServer[] = "turn:test.com:1234";
70 static const char kTurnIceServerWithTransport[] =
71     "turn:hello.com?transport=tcp";
72 static const char kSecureTurnIceServer[] = "turns:hello.com?transport=tcp";
73 static const char kSecureTurnIceServerWithoutTransportParam[] =
74     "turns:hello.com:443";
75 static const char kSecureTurnIceServerWithoutTransportAndPortParam[] =
76     "turns:hello.com";
77 static const char kTurnIceServerWithNoUsernameInUri[] = "turn:test.com:1234";
78 static const char kTurnPassword[] = "turnpassword";
79 static const int kDefaultStunPort = 3478;
80 static const int kDefaultStunTlsPort = 5349;
81 static const char kTurnUsername[] = "test";
82 static const char kStunIceServerWithIPv4Address[] = "stun:1.2.3.4:1234";
83 static const char kStunIceServerWithIPv4AddressWithoutPort[] = "stun:1.2.3.4";
84 static const char kStunIceServerWithIPv6Address[] = "stun:[2401:fa00:4::]:1234";
85 static const char kStunIceServerWithIPv6AddressWithoutPort[] =
86     "stun:[2401:fa00:4::]";
87 static const char kTurnIceServerWithIPv6Address[] = "turn:[2401:fa00:4::]:1234";
88 
89 class NullPeerConnectionObserver : public PeerConnectionObserver {
90  public:
91   virtual ~NullPeerConnectionObserver() = default;
OnSignalingChange(PeerConnectionInterface::SignalingState new_state)92   void OnSignalingChange(
93       PeerConnectionInterface::SignalingState new_state) override {}
OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream)94   void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override {}
OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream)95   void OnRemoveStream(
96       rtc::scoped_refptr<MediaStreamInterface> stream) override {}
OnDataChannel(rtc::scoped_refptr<DataChannelInterface> data_channel)97   void OnDataChannel(
98       rtc::scoped_refptr<DataChannelInterface> data_channel) override {}
OnRenegotiationNeeded()99   void OnRenegotiationNeeded() override {}
OnIceConnectionChange(PeerConnectionInterface::IceConnectionState new_state)100   void OnIceConnectionChange(
101       PeerConnectionInterface::IceConnectionState new_state) override {}
OnIceGatheringChange(PeerConnectionInterface::IceGatheringState new_state)102   void OnIceGatheringChange(
103       PeerConnectionInterface::IceGatheringState new_state) override {}
OnIceCandidate(const webrtc::IceCandidateInterface * candidate)104   void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
105   }
106 };
107 
108 class MockNetworkManager : public rtc::NetworkManager {
109  public:
110   MOCK_METHOD(void, StartUpdating, (), (override));
111   MOCK_METHOD(void, StopUpdating, (), (override));
112   MOCK_METHOD(std::vector<const rtc::Network*>,
113               GetNetworks,
114               (),
115               (const override));
116   MOCK_METHOD(std::vector<const rtc::Network*>,
117               GetAnyAddressNetworks,
118               (),
119               (override));
120 };
121 
122 }  // namespace
123 
124 class PeerConnectionFactoryTest : public ::testing::Test {
125  public:
PeerConnectionFactoryTest()126   PeerConnectionFactoryTest()
127       : socket_server_(rtc::CreateDefaultSocketServer()),
128         main_thread_(socket_server_.get()) {}
129 
130  private:
SetUp()131   void SetUp() {
132 #ifdef WEBRTC_ANDROID
133     webrtc::InitializeAndroidObjects();
134 #endif
135     // Use fake audio device module since we're only testing the interface
136     // level, and using a real one could make tests flaky e.g. when run in
137     // parallel.
138     factory_ = webrtc::CreatePeerConnectionFactory(
139         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
140         rtc::scoped_refptr<webrtc::AudioDeviceModule>(
141             FakeAudioCaptureModule::Create()),
142         webrtc::CreateBuiltinAudioEncoderFactory(),
143         webrtc::CreateBuiltinAudioDecoderFactory(),
144         webrtc::CreateBuiltinVideoEncoderFactory(),
145         webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
146         nullptr /* audio_processing */);
147 
148     ASSERT_TRUE(factory_.get() != NULL);
149     packet_socket_factory_.reset(
150         new rtc::BasicPacketSocketFactory(socket_server_.get()));
151     port_allocator_.reset(new cricket::FakePortAllocator(
152         rtc::Thread::Current(), packet_socket_factory_.get()));
153     raw_port_allocator_ = port_allocator_.get();
154   }
155 
156  protected:
VerifyStunServers(cricket::ServerAddresses stun_servers)157   void VerifyStunServers(cricket::ServerAddresses stun_servers) {
158     EXPECT_EQ(stun_servers, raw_port_allocator_->stun_servers());
159   }
160 
VerifyTurnServers(std::vector<cricket::RelayServerConfig> turn_servers)161   void VerifyTurnServers(std::vector<cricket::RelayServerConfig> turn_servers) {
162     EXPECT_EQ(turn_servers.size(), raw_port_allocator_->turn_servers().size());
163     for (size_t i = 0; i < turn_servers.size(); ++i) {
164       ASSERT_EQ(1u, turn_servers[i].ports.size());
165       EXPECT_EQ(1u, raw_port_allocator_->turn_servers()[i].ports.size());
166       EXPECT_EQ(
167           turn_servers[i].ports[0].address.ToString(),
168           raw_port_allocator_->turn_servers()[i].ports[0].address.ToString());
169       EXPECT_EQ(turn_servers[i].ports[0].proto,
170                 raw_port_allocator_->turn_servers()[i].ports[0].proto);
171       EXPECT_EQ(turn_servers[i].credentials.username,
172                 raw_port_allocator_->turn_servers()[i].credentials.username);
173       EXPECT_EQ(turn_servers[i].credentials.password,
174                 raw_port_allocator_->turn_servers()[i].credentials.password);
175     }
176   }
177 
VerifyAudioCodecCapability(const webrtc::RtpCodecCapability & codec)178   void VerifyAudioCodecCapability(const webrtc::RtpCodecCapability& codec) {
179     EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_AUDIO);
180     EXPECT_FALSE(codec.name.empty());
181     EXPECT_GT(codec.clock_rate, 0);
182     EXPECT_GT(codec.num_channels, 0);
183   }
184 
VerifyVideoCodecCapability(const webrtc::RtpCodecCapability & codec,bool sender)185   void VerifyVideoCodecCapability(const webrtc::RtpCodecCapability& codec,
186                                   bool sender) {
187     EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO);
188     EXPECT_FALSE(codec.name.empty());
189     EXPECT_GT(codec.clock_rate, 0);
190     if (sender) {
191       if (codec.name == "VP8" || codec.name == "H264") {
192         EXPECT_THAT(codec.scalability_modes,
193                     UnorderedElementsAre(webrtc::ScalabilityMode::kL1T1,
194                                          webrtc::ScalabilityMode::kL1T2,
195                                          webrtc::ScalabilityMode::kL1T3))
196             << "Codec: " << codec.name;
197       } else if (codec.name == "VP9" || codec.name == "AV1") {
198         EXPECT_THAT(
199             codec.scalability_modes,
200             UnorderedElementsAre(
201                 // clang-format off
202                 webrtc::ScalabilityMode::kL1T1,
203                 webrtc::ScalabilityMode::kL1T2,
204                 webrtc::ScalabilityMode::kL1T3,
205                 webrtc::ScalabilityMode::kL2T1,
206                 webrtc::ScalabilityMode::kL2T1h,
207                 webrtc::ScalabilityMode::kL2T1_KEY,
208                 webrtc::ScalabilityMode::kL2T2,
209                 webrtc::ScalabilityMode::kL2T2h,
210                 webrtc::ScalabilityMode::kL2T2_KEY,
211                 webrtc::ScalabilityMode::kL2T2_KEY_SHIFT,
212                 webrtc::ScalabilityMode::kL2T3,
213                 webrtc::ScalabilityMode::kL2T3h,
214                 webrtc::ScalabilityMode::kL2T3_KEY,
215                 webrtc::ScalabilityMode::kL3T1,
216                 webrtc::ScalabilityMode::kL3T1h,
217                 webrtc::ScalabilityMode::kL3T1_KEY,
218                 webrtc::ScalabilityMode::kL3T2,
219                 webrtc::ScalabilityMode::kL3T2h,
220                 webrtc::ScalabilityMode::kL3T2_KEY,
221                 webrtc::ScalabilityMode::kL3T3,
222                 webrtc::ScalabilityMode::kL3T3h,
223                 webrtc::ScalabilityMode::kL3T3_KEY,
224                 webrtc::ScalabilityMode::kS2T1,
225                 webrtc::ScalabilityMode::kS2T1h,
226                 webrtc::ScalabilityMode::kS2T2,
227                 webrtc::ScalabilityMode::kS2T2h,
228                 webrtc::ScalabilityMode::kS2T3,
229                 webrtc::ScalabilityMode::kS2T3h,
230                 webrtc::ScalabilityMode::kS3T1,
231                 webrtc::ScalabilityMode::kS3T1h,
232                 webrtc::ScalabilityMode::kS3T2,
233                 webrtc::ScalabilityMode::kS3T2h,
234                 webrtc::ScalabilityMode::kS3T3,
235                 webrtc::ScalabilityMode::kS3T3h)
236             // clang-format on
237             )
238             << "Codec: " << codec.name;
239       } else {
240         EXPECT_TRUE(codec.scalability_modes.empty());
241       }
242     } else {
243       EXPECT_TRUE(codec.scalability_modes.empty());
244     }
245   }
246 
247   std::unique_ptr<rtc::SocketServer> socket_server_;
248   rtc::AutoSocketServerThread main_thread_;
249   rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_;
250   NullPeerConnectionObserver observer_;
251   std::unique_ptr<rtc::PacketSocketFactory> packet_socket_factory_;
252   std::unique_ptr<cricket::FakePortAllocator> port_allocator_;
253   // Since the PC owns the port allocator after it's been initialized,
254   // this should only be used when known to be safe.
255   cricket::FakePortAllocator* raw_port_allocator_;
256 };
257 
258 // Verify creation of PeerConnection using internal ADM, video factory and
259 // internal libjingle threads.
260 // TODO(henrika): disabling this test since relying on real audio can result in
261 // flaky tests and focus on details that are out of scope for you might expect
262 // for a PeerConnectionFactory unit test.
263 // See https://bugs.chromium.org/p/webrtc/issues/detail?id=7806 for details.
TEST(PeerConnectionFactoryTestInternal,DISABLED_CreatePCUsingInternalModules)264 TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) {
265 #ifdef WEBRTC_ANDROID
266   webrtc::InitializeAndroidObjects();
267 #endif
268 
269   rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
270       webrtc::CreatePeerConnectionFactory(
271           nullptr /* network_thread */, nullptr /* worker_thread */,
272           nullptr /* signaling_thread */, nullptr /* default_adm */,
273           webrtc::CreateBuiltinAudioEncoderFactory(),
274           webrtc::CreateBuiltinAudioDecoderFactory(),
275           webrtc::CreateBuiltinVideoEncoderFactory(),
276           webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
277           nullptr /* audio_processing */));
278 
279   NullPeerConnectionObserver observer;
280   webrtc::PeerConnectionInterface::RTCConfiguration config;
281   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
282 
283   std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
284       new FakeRTCCertificateGenerator());
285   webrtc::PeerConnectionDependencies pc_dependencies(&observer);
286   pc_dependencies.cert_generator = std::move(cert_generator);
287   auto result =
288       factory->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
289 
290   EXPECT_TRUE(result.ok());
291 }
292 
TEST_F(PeerConnectionFactoryTest,CheckRtpSenderAudioCapabilities)293 TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) {
294   webrtc::RtpCapabilities audio_capabilities =
295       factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO);
296   EXPECT_FALSE(audio_capabilities.codecs.empty());
297   for (const auto& codec : audio_capabilities.codecs) {
298     VerifyAudioCodecCapability(codec);
299   }
300   EXPECT_FALSE(audio_capabilities.header_extensions.empty());
301   for (const auto& header_extension : audio_capabilities.header_extensions) {
302     EXPECT_FALSE(header_extension.uri.empty());
303   }
304 }
305 
TEST_F(PeerConnectionFactoryTest,CheckRtpSenderVideoCapabilities)306 TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) {
307   webrtc::RtpCapabilities video_capabilities =
308       factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
309   EXPECT_FALSE(video_capabilities.codecs.empty());
310   for (const auto& codec : video_capabilities.codecs) {
311     VerifyVideoCodecCapability(codec, true);
312   }
313   EXPECT_FALSE(video_capabilities.header_extensions.empty());
314   for (const auto& header_extension : video_capabilities.header_extensions) {
315     EXPECT_FALSE(header_extension.uri.empty());
316   }
317 }
318 
TEST_F(PeerConnectionFactoryTest,CheckRtpSenderDataCapabilities)319 TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) {
320   webrtc::RtpCapabilities data_capabilities =
321       factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_DATA);
322   EXPECT_TRUE(data_capabilities.codecs.empty());
323   EXPECT_TRUE(data_capabilities.header_extensions.empty());
324 }
325 
TEST_F(PeerConnectionFactoryTest,CheckRtpReceiverAudioCapabilities)326 TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) {
327   webrtc::RtpCapabilities audio_capabilities =
328       factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO);
329   EXPECT_FALSE(audio_capabilities.codecs.empty());
330   for (const auto& codec : audio_capabilities.codecs) {
331     VerifyAudioCodecCapability(codec);
332   }
333   EXPECT_FALSE(audio_capabilities.header_extensions.empty());
334   for (const auto& header_extension : audio_capabilities.header_extensions) {
335     EXPECT_FALSE(header_extension.uri.empty());
336   }
337 }
338 
TEST_F(PeerConnectionFactoryTest,CheckRtpReceiverVideoCapabilities)339 TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) {
340   webrtc::RtpCapabilities video_capabilities =
341       factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
342   EXPECT_FALSE(video_capabilities.codecs.empty());
343   for (const auto& codec : video_capabilities.codecs) {
344     VerifyVideoCodecCapability(codec, false);
345   }
346   EXPECT_FALSE(video_capabilities.header_extensions.empty());
347   for (const auto& header_extension : video_capabilities.header_extensions) {
348     EXPECT_FALSE(header_extension.uri.empty());
349   }
350 }
351 
TEST_F(PeerConnectionFactoryTest,CheckRtpReceiverDataCapabilities)352 TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) {
353   webrtc::RtpCapabilities data_capabilities =
354       factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_DATA);
355   EXPECT_TRUE(data_capabilities.codecs.empty());
356   EXPECT_TRUE(data_capabilities.header_extensions.empty());
357 }
358 
359 // This test verifies creation of PeerConnection with valid STUN and TURN
360 // configuration. Also verifies the URL's parsed correctly as expected.
TEST_F(PeerConnectionFactoryTest,CreatePCUsingIceServers)361 TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
362   PeerConnectionInterface::RTCConfiguration config;
363   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
364   webrtc::PeerConnectionInterface::IceServer ice_server;
365   ice_server.uri = kStunIceServer;
366   config.servers.push_back(ice_server);
367   ice_server.uri = kTurnIceServer;
368   ice_server.username = kTurnUsername;
369   ice_server.password = kTurnPassword;
370   config.servers.push_back(ice_server);
371   ice_server.uri = kTurnIceServerWithTransport;
372   ice_server.username = kTurnUsername;
373   ice_server.password = kTurnPassword;
374   config.servers.push_back(ice_server);
375   webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
376   pc_dependencies.cert_generator =
377       std::make_unique<FakeRTCCertificateGenerator>();
378   pc_dependencies.allocator = std::move(port_allocator_);
379   auto result =
380       factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
381   ASSERT_TRUE(result.ok());
382   cricket::ServerAddresses stun_servers;
383   rtc::SocketAddress stun1("stun.l.google.com", 19302);
384   stun_servers.insert(stun1);
385   VerifyStunServers(stun_servers);
386   std::vector<cricket::RelayServerConfig> turn_servers;
387   cricket::RelayServerConfig turn1("test.com", 1234, kTurnUsername,
388                                    kTurnPassword, cricket::PROTO_UDP);
389   turn_servers.push_back(turn1);
390   cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, kTurnUsername,
391                                    kTurnPassword, cricket::PROTO_TCP);
392   turn_servers.push_back(turn2);
393   VerifyTurnServers(turn_servers);
394 }
395 
396 // This test verifies creation of PeerConnection with valid STUN and TURN
397 // configuration. Also verifies the list of URL's parsed correctly as expected.
TEST_F(PeerConnectionFactoryTest,CreatePCUsingIceServersUrls)398 TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) {
399   PeerConnectionInterface::RTCConfiguration config;
400   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
401   webrtc::PeerConnectionInterface::IceServer ice_server;
402   ice_server.urls.push_back(kStunIceServer);
403   ice_server.urls.push_back(kTurnIceServer);
404   ice_server.urls.push_back(kTurnIceServerWithTransport);
405   ice_server.username = kTurnUsername;
406   ice_server.password = kTurnPassword;
407   config.servers.push_back(ice_server);
408   webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
409   pc_dependencies.cert_generator =
410       std::make_unique<FakeRTCCertificateGenerator>();
411   pc_dependencies.allocator = std::move(port_allocator_);
412   auto result =
413       factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
414   ASSERT_TRUE(result.ok());
415   cricket::ServerAddresses stun_servers;
416   rtc::SocketAddress stun1("stun.l.google.com", 19302);
417   stun_servers.insert(stun1);
418   VerifyStunServers(stun_servers);
419   std::vector<cricket::RelayServerConfig> turn_servers;
420   cricket::RelayServerConfig turn1("test.com", 1234, kTurnUsername,
421                                    kTurnPassword, cricket::PROTO_UDP);
422   turn_servers.push_back(turn1);
423   cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, kTurnUsername,
424                                    kTurnPassword, cricket::PROTO_TCP);
425   turn_servers.push_back(turn2);
426   VerifyTurnServers(turn_servers);
427 }
428 
TEST_F(PeerConnectionFactoryTest,CreatePCUsingNoUsernameInUri)429 TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) {
430   PeerConnectionInterface::RTCConfiguration config;
431   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
432   webrtc::PeerConnectionInterface::IceServer ice_server;
433   ice_server.uri = kStunIceServer;
434   config.servers.push_back(ice_server);
435   ice_server.uri = kTurnIceServerWithNoUsernameInUri;
436   ice_server.username = kTurnUsername;
437   ice_server.password = kTurnPassword;
438   config.servers.push_back(ice_server);
439   webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
440   pc_dependencies.cert_generator =
441       std::make_unique<FakeRTCCertificateGenerator>();
442   pc_dependencies.allocator = std::move(port_allocator_);
443   auto result =
444       factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
445   ASSERT_TRUE(result.ok());
446   std::vector<cricket::RelayServerConfig> turn_servers;
447   cricket::RelayServerConfig turn("test.com", 1234, kTurnUsername,
448                                   kTurnPassword, cricket::PROTO_UDP);
449   turn_servers.push_back(turn);
450   VerifyTurnServers(turn_servers);
451 }
452 
453 // This test verifies the PeerConnection created properly with TURN url which
454 // has transport parameter in it.
TEST_F(PeerConnectionFactoryTest,CreatePCUsingTurnUrlWithTransportParam)455 TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) {
456   PeerConnectionInterface::RTCConfiguration config;
457   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
458   webrtc::PeerConnectionInterface::IceServer ice_server;
459   ice_server.uri = kTurnIceServerWithTransport;
460   ice_server.username = kTurnUsername;
461   ice_server.password = kTurnPassword;
462   config.servers.push_back(ice_server);
463   webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
464   pc_dependencies.cert_generator =
465       std::make_unique<FakeRTCCertificateGenerator>();
466   pc_dependencies.allocator = std::move(port_allocator_);
467   auto result =
468       factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
469   ASSERT_TRUE(result.ok());
470   std::vector<cricket::RelayServerConfig> turn_servers;
471   cricket::RelayServerConfig turn("hello.com", kDefaultStunPort, kTurnUsername,
472                                   kTurnPassword, cricket::PROTO_TCP);
473   turn_servers.push_back(turn);
474   VerifyTurnServers(turn_servers);
475 }
476 
TEST_F(PeerConnectionFactoryTest,CreatePCUsingSecureTurnUrl)477 TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) {
478   PeerConnectionInterface::RTCConfiguration config;
479   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
480   webrtc::PeerConnectionInterface::IceServer ice_server;
481   ice_server.uri = kSecureTurnIceServer;
482   ice_server.username = kTurnUsername;
483   ice_server.password = kTurnPassword;
484   config.servers.push_back(ice_server);
485   ice_server.uri = kSecureTurnIceServerWithoutTransportParam;
486   ice_server.username = kTurnUsername;
487   ice_server.password = kTurnPassword;
488   config.servers.push_back(ice_server);
489   ice_server.uri = kSecureTurnIceServerWithoutTransportAndPortParam;
490   ice_server.username = kTurnUsername;
491   ice_server.password = kTurnPassword;
492   config.servers.push_back(ice_server);
493   webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
494   pc_dependencies.cert_generator =
495       std::make_unique<FakeRTCCertificateGenerator>();
496   pc_dependencies.allocator = std::move(port_allocator_);
497   auto result =
498       factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
499   ASSERT_TRUE(result.ok());
500   std::vector<cricket::RelayServerConfig> turn_servers;
501   cricket::RelayServerConfig turn1("hello.com", kDefaultStunTlsPort,
502                                    kTurnUsername, kTurnPassword,
503                                    cricket::PROTO_TLS);
504   turn_servers.push_back(turn1);
505   // TURNS with transport param should be default to tcp.
506   cricket::RelayServerConfig turn2("hello.com", 443, kTurnUsername,
507                                    kTurnPassword, cricket::PROTO_TLS);
508   turn_servers.push_back(turn2);
509   cricket::RelayServerConfig turn3("hello.com", kDefaultStunTlsPort,
510                                    kTurnUsername, kTurnPassword,
511                                    cricket::PROTO_TLS);
512   turn_servers.push_back(turn3);
513   VerifyTurnServers(turn_servers);
514 }
515 
TEST_F(PeerConnectionFactoryTest,CreatePCUsingIPLiteralAddress)516 TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
517   PeerConnectionInterface::RTCConfiguration config;
518   config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
519   webrtc::PeerConnectionInterface::IceServer ice_server;
520   ice_server.uri = kStunIceServerWithIPv4Address;
521   config.servers.push_back(ice_server);
522   ice_server.uri = kStunIceServerWithIPv4AddressWithoutPort;
523   config.servers.push_back(ice_server);
524   ice_server.uri = kStunIceServerWithIPv6Address;
525   config.servers.push_back(ice_server);
526   ice_server.uri = kStunIceServerWithIPv6AddressWithoutPort;
527   config.servers.push_back(ice_server);
528   ice_server.uri = kTurnIceServerWithIPv6Address;
529   ice_server.username = kTurnUsername;
530   ice_server.password = kTurnPassword;
531   config.servers.push_back(ice_server);
532   webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
533   pc_dependencies.cert_generator =
534       std::make_unique<FakeRTCCertificateGenerator>();
535   pc_dependencies.allocator = std::move(port_allocator_);
536   auto result =
537       factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
538   ASSERT_TRUE(result.ok());
539   cricket::ServerAddresses stun_servers;
540   rtc::SocketAddress stun1("1.2.3.4", 1234);
541   stun_servers.insert(stun1);
542   rtc::SocketAddress stun2("1.2.3.4", 3478);
543   stun_servers.insert(stun2);  // Default port
544   rtc::SocketAddress stun3("2401:fa00:4::", 1234);
545   stun_servers.insert(stun3);
546   rtc::SocketAddress stun4("2401:fa00:4::", 3478);
547   stun_servers.insert(stun4);  // Default port
548   VerifyStunServers(stun_servers);
549 
550   std::vector<cricket::RelayServerConfig> turn_servers;
551   cricket::RelayServerConfig turn1("2401:fa00:4::", 1234, kTurnUsername,
552                                    kTurnPassword, cricket::PROTO_UDP);
553   turn_servers.push_back(turn1);
554   VerifyTurnServers(turn_servers);
555 }
556 
557 // This test verifies the captured stream is rendered locally using a
558 // local video track.
TEST_F(PeerConnectionFactoryTest,LocalRendering)559 TEST_F(PeerConnectionFactoryTest, LocalRendering) {
560   rtc::scoped_refptr<webrtc::FakeVideoTrackSource> source =
561       webrtc::FakeVideoTrackSource::Create(/*is_screencast=*/false);
562 
563   cricket::FakeFrameSource frame_source(1280, 720,
564                                         rtc::kNumMicrosecsPerSec / 30);
565 
566   ASSERT_TRUE(source.get() != NULL);
567   rtc::scoped_refptr<VideoTrackInterface> track(
568       factory_->CreateVideoTrack("testlabel", source.get()));
569   ASSERT_TRUE(track.get() != NULL);
570   FakeVideoTrackRenderer local_renderer(track.get());
571 
572   EXPECT_EQ(0, local_renderer.num_rendered_frames());
573   source->InjectFrame(frame_source.GetFrame());
574   EXPECT_EQ(1, local_renderer.num_rendered_frames());
575   EXPECT_FALSE(local_renderer.black_frame());
576 
577   track->set_enabled(false);
578   source->InjectFrame(frame_source.GetFrame());
579   EXPECT_EQ(2, local_renderer.num_rendered_frames());
580   EXPECT_TRUE(local_renderer.black_frame());
581 
582   track->set_enabled(true);
583   source->InjectFrame(frame_source.GetFrame());
584   EXPECT_EQ(3, local_renderer.num_rendered_frames());
585   EXPECT_FALSE(local_renderer.black_frame());
586 }
587 
TEST(PeerConnectionFactoryDependenciesTest,UsesNetworkManager)588 TEST(PeerConnectionFactoryDependenciesTest, UsesNetworkManager) {
589   constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10);
590   auto mock_network_manager = std::make_unique<NiceMock<MockNetworkManager>>();
591 
592   rtc::Event called;
593   EXPECT_CALL(*mock_network_manager, StartUpdating())
594       .Times(AtLeast(1))
595       .WillRepeatedly(InvokeWithoutArgs([&] { called.Set(); }));
596 
597   webrtc::PeerConnectionFactoryDependencies pcf_dependencies;
598   pcf_dependencies.network_manager = std::move(mock_network_manager);
599 
600   rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf =
601       CreateModularPeerConnectionFactory(std::move(pcf_dependencies));
602 
603   PeerConnectionInterface::RTCConfiguration config;
604   config.ice_candidate_pool_size = 2;
605   NullPeerConnectionObserver observer;
606   auto pc = pcf->CreatePeerConnectionOrError(
607       config, webrtc::PeerConnectionDependencies(&observer));
608   ASSERT_TRUE(pc.ok());
609 
610   called.Wait(kWaitTimeout);
611 }
612 
TEST(PeerConnectionFactoryDependenciesTest,UsesPacketSocketFactory)613 TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) {
614   constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10);
615   auto mock_socket_factory =
616       std::make_unique<NiceMock<rtc::MockPacketSocketFactory>>();
617 
618   rtc::Event called;
619   EXPECT_CALL(*mock_socket_factory, CreateUdpSocket(_, _, _))
620       .WillOnce(InvokeWithoutArgs([&] {
621         called.Set();
622         return nullptr;
623       }))
624       .WillRepeatedly(Return(nullptr));
625 
626   webrtc::PeerConnectionFactoryDependencies pcf_dependencies;
627   pcf_dependencies.packet_socket_factory = std::move(mock_socket_factory);
628 
629   rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf =
630       CreateModularPeerConnectionFactory(std::move(pcf_dependencies));
631 
632   // By default, localhost addresses are ignored, which makes tests fail if test
633   // machine is offline.
634   PeerConnectionFactoryInterface::Options options;
635   options.network_ignore_mask = 0;
636   pcf->SetOptions(options);
637 
638   PeerConnectionInterface::RTCConfiguration config;
639   config.ice_candidate_pool_size = 2;
640   NullPeerConnectionObserver observer;
641   auto pc = pcf->CreatePeerConnectionOrError(
642       config, webrtc::PeerConnectionDependencies(&observer));
643   ASSERT_TRUE(pc.ok());
644 
645   called.Wait(kWaitTimeout);
646 }
647