1 /*
2 * Copyright 2018 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 <memory>
12 #include <string>
13 #include <utility>
14 #include <vector>
15
16 #include "absl/types/optional.h"
17 #include "api/audio/audio_mixer.h"
18 #include "api/audio_codecs/audio_decoder_factory.h"
19 #include "api/audio_codecs/audio_encoder_factory.h"
20 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
21 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
22 #include "api/audio_options.h"
23 #include "api/create_peerconnection_factory.h"
24 #include "api/jsep.h"
25 #include "api/media_stream_interface.h"
26 #include "api/peer_connection_interface.h"
27 #include "api/scoped_refptr.h"
28 #include "api/stats/rtc_stats.h"
29 #include "api/stats/rtc_stats_report.h"
30 #include "api/stats/rtcstats_objects.h"
31 #include "api/video_codecs/builtin_video_decoder_factory.h"
32 #include "api/video_codecs/builtin_video_encoder_factory.h"
33 #include "api/video_codecs/video_decoder_factory.h"
34 #include "api/video_codecs/video_encoder_factory.h"
35 #include "modules/audio_device/include/audio_device.h"
36 #include "modules/audio_processing/include/audio_processing.h"
37 #include "p2p/base/port_allocator.h"
38 #include "p2p/base/port_interface.h"
39 #include "p2p/base/test_turn_server.h"
40 #include "p2p/client/basic_port_allocator.h"
41 #include "pc/peer_connection.h"
42 #include "pc/peer_connection_wrapper.h"
43 #include "pc/test/fake_audio_capture_module.h"
44 #include "pc/test/frame_generator_capturer_video_track_source.h"
45 #include "pc/test/mock_peer_connection_observers.h"
46 #include "rtc_base/checks.h"
47 #include "rtc_base/fake_network.h"
48 #include "rtc_base/firewall_socket_server.h"
49 #include "rtc_base/gunit.h"
50 #include "rtc_base/helpers.h"
51 #include "rtc_base/location.h"
52 #include "rtc_base/ref_counted_object.h"
53 #include "rtc_base/socket_address.h"
54 #include "rtc_base/ssl_certificate.h"
55 #include "rtc_base/test_certificate_verifier.h"
56 #include "rtc_base/thread.h"
57 #include "rtc_base/virtual_socket_server.h"
58 #include "system_wrappers/include/clock.h"
59 #include "test/gtest.h"
60 #include "test/testsupport/perf_test.h"
61
62 namespace webrtc {
63
64 namespace {
65 static const int kDefaultTestTimeMs = 15000;
66 static const int kRampUpTimeMs = 5000;
67 static const int kPollIntervalTimeMs = 50;
68 static const int kDefaultTimeoutMs = 10000;
69 static const rtc::SocketAddress kDefaultLocalAddress("1.1.1.1", 0);
70 static const char kTurnInternalAddress[] = "88.88.88.0";
71 static const char kTurnExternalAddress[] = "88.88.88.1";
72 static const int kTurnInternalPort = 3478;
73 static const int kTurnExternalPort = 0;
74 // The video's configured max bitrate in webrtcvideoengine.cc is 1.7 Mbps.
75 // Setting the network bandwidth to 1 Mbps allows the video's bitrate to push
76 // the network's limitations.
77 static const int kNetworkBandwidth = 1000000;
78 } // namespace
79
80 using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
81
82 // This is an end to end test to verify that BWE is functioning when setting
83 // up a one to one call at the PeerConnection level. The intention of the test
84 // is to catch potential regressions for different ICE path configurations. The
85 // test uses a VirtualSocketServer for it's underlying simulated network and
86 // fake audio and video sources. The test is based upon rampup_tests.cc, but
87 // instead is at the PeerConnection level and uses a different fake network
88 // (rampup_tests.cc uses SimulatedNetwork). In the future, this test could
89 // potentially test different network conditions and test video quality as well
90 // (video_quality_test.cc does this, but at the call level).
91 //
92 // The perf test results are printed using the perf test support. If the
93 // isolated_script_test_perf_output flag is specified in test_main.cc, then
94 // the results are written to a JSON formatted file for the Chrome perf
95 // dashboard. Since this test is a webrtc_perf_test, it will be run in the perf
96 // console every webrtc commit.
97 class PeerConnectionWrapperForRampUpTest : public PeerConnectionWrapper {
98 public:
99 using PeerConnectionWrapper::PeerConnectionWrapper;
100
PeerConnectionWrapperForRampUpTest(rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,rtc::scoped_refptr<PeerConnectionInterface> pc,std::unique_ptr<MockPeerConnectionObserver> observer)101 PeerConnectionWrapperForRampUpTest(
102 rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
103 rtc::scoped_refptr<PeerConnectionInterface> pc,
104 std::unique_ptr<MockPeerConnectionObserver> observer)
105 : PeerConnectionWrapper::PeerConnectionWrapper(pc_factory,
106 pc,
107 std::move(observer)) {}
108
AddIceCandidates(std::vector<const IceCandidateInterface * > candidates)109 bool AddIceCandidates(std::vector<const IceCandidateInterface*> candidates) {
110 bool success = true;
111 for (const auto candidate : candidates) {
112 if (!pc()->AddIceCandidate(candidate)) {
113 success = false;
114 }
115 }
116 return success;
117 }
118
CreateLocalVideoTrack(FrameGeneratorCapturerVideoTrackSource::Config config,Clock * clock)119 rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrack(
120 FrameGeneratorCapturerVideoTrackSource::Config config,
121 Clock* clock) {
122 video_track_sources_.emplace_back(
123 new rtc::RefCountedObject<FrameGeneratorCapturerVideoTrackSource>(
124 config, clock, /*is_screencast=*/false));
125 video_track_sources_.back()->Start();
126 return rtc::scoped_refptr<VideoTrackInterface>(
127 pc_factory()->CreateVideoTrack(rtc::CreateRandomUuid(),
128 video_track_sources_.back()));
129 }
130
CreateLocalAudioTrack(const cricket::AudioOptions options)131 rtc::scoped_refptr<AudioTrackInterface> CreateLocalAudioTrack(
132 const cricket::AudioOptions options) {
133 rtc::scoped_refptr<AudioSourceInterface> source =
134 pc_factory()->CreateAudioSource(options);
135 return pc_factory()->CreateAudioTrack(rtc::CreateRandomUuid(), source);
136 }
137
138 private:
139 std::vector<rtc::scoped_refptr<FrameGeneratorCapturerVideoTrackSource>>
140 video_track_sources_;
141 };
142
143 // TODO(shampson): Paramaterize the test to run for both Plan B & Unified Plan.
144 class PeerConnectionRampUpTest : public ::testing::Test {
145 public:
PeerConnectionRampUpTest()146 PeerConnectionRampUpTest()
147 : clock_(Clock::GetRealTimeClock()),
148 virtual_socket_server_(new rtc::VirtualSocketServer()),
149 firewall_socket_server_(
150 new rtc::FirewallSocketServer(virtual_socket_server_.get())),
151 network_thread_(new rtc::Thread(firewall_socket_server_.get())),
152 worker_thread_(rtc::Thread::Create()) {
153 network_thread_->SetName("PCNetworkThread", this);
154 worker_thread_->SetName("PCWorkerThread", this);
155 RTC_CHECK(network_thread_->Start());
156 RTC_CHECK(worker_thread_->Start());
157
158 virtual_socket_server_->set_bandwidth(kNetworkBandwidth / 8);
159 pc_factory_ = CreatePeerConnectionFactory(
160 network_thread_.get(), worker_thread_.get(), rtc::Thread::Current(),
161 rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
162 CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
163 CreateBuiltinVideoEncoderFactory(), CreateBuiltinVideoDecoderFactory(),
164 nullptr /* audio_mixer */, nullptr /* audio_processing */);
165 }
166
~PeerConnectionRampUpTest()167 virtual ~PeerConnectionRampUpTest() {
168 network_thread()->Invoke<void>(RTC_FROM_HERE,
169 [this] { turn_servers_.clear(); });
170 }
171
CreatePeerConnectionWrappers(const RTCConfiguration & caller_config,const RTCConfiguration & callee_config)172 bool CreatePeerConnectionWrappers(const RTCConfiguration& caller_config,
173 const RTCConfiguration& callee_config) {
174 caller_ = CreatePeerConnectionWrapper(caller_config);
175 callee_ = CreatePeerConnectionWrapper(callee_config);
176 return caller_ && callee_;
177 }
178
179 std::unique_ptr<PeerConnectionWrapperForRampUpTest>
CreatePeerConnectionWrapper(const RTCConfiguration & config)180 CreatePeerConnectionWrapper(const RTCConfiguration& config) {
181 auto* fake_network_manager = new rtc::FakeNetworkManager();
182 fake_network_manager->AddInterface(kDefaultLocalAddress);
183 fake_network_managers_.emplace_back(fake_network_manager);
184
185 auto observer = std::make_unique<MockPeerConnectionObserver>();
186 webrtc::PeerConnectionDependencies dependencies(observer.get());
187 cricket::BasicPortAllocator* port_allocator =
188 new cricket::BasicPortAllocator(fake_network_manager);
189 port_allocator->set_step_delay(cricket::kDefaultStepDelay);
190 dependencies.allocator =
191 std::unique_ptr<cricket::BasicPortAllocator>(port_allocator);
192 dependencies.tls_cert_verifier =
193 std::make_unique<rtc::TestCertificateVerifier>();
194
195 auto pc =
196 pc_factory_->CreatePeerConnection(config, std::move(dependencies));
197 if (!pc) {
198 return nullptr;
199 }
200
201 return std::make_unique<PeerConnectionWrapperForRampUpTest>(
202 pc_factory_, pc, std::move(observer));
203 }
204
SetupOneWayCall()205 void SetupOneWayCall() {
206 ASSERT_TRUE(caller_);
207 ASSERT_TRUE(callee_);
208 FrameGeneratorCapturerVideoTrackSource::Config config;
209 caller_->AddTrack(caller_->CreateLocalVideoTrack(config, clock_));
210 // Disable highpass filter so that we can get all the test audio frames.
211 cricket::AudioOptions options;
212 options.highpass_filter = false;
213 caller_->AddTrack(caller_->CreateLocalAudioTrack(options));
214
215 // Do the SDP negotiation, and also exchange ice candidates.
216 ASSERT_TRUE(caller_->ExchangeOfferAnswerWith(callee_.get()));
217 ASSERT_TRUE_WAIT(
218 caller_->signaling_state() == PeerConnectionInterface::kStable,
219 kDefaultTimeoutMs);
220 ASSERT_TRUE_WAIT(caller_->IsIceGatheringDone(), kDefaultTimeoutMs);
221 ASSERT_TRUE_WAIT(callee_->IsIceGatheringDone(), kDefaultTimeoutMs);
222
223 // Connect an ICE candidate pairs.
224 ASSERT_TRUE(
225 callee_->AddIceCandidates(caller_->observer()->GetAllCandidates()));
226 ASSERT_TRUE(
227 caller_->AddIceCandidates(callee_->observer()->GetAllCandidates()));
228 // This means that ICE and DTLS are connected.
229 ASSERT_TRUE_WAIT(callee_->IsIceConnected(), kDefaultTimeoutMs);
230 ASSERT_TRUE_WAIT(caller_->IsIceConnected(), kDefaultTimeoutMs);
231 }
232
CreateTurnServer(cricket::ProtocolType type,const std::string & common_name="test turn server")233 void CreateTurnServer(cricket::ProtocolType type,
234 const std::string& common_name = "test turn server") {
235 rtc::Thread* thread = network_thread();
236 std::unique_ptr<cricket::TestTurnServer> turn_server =
237 network_thread_->Invoke<std::unique_ptr<cricket::TestTurnServer>>(
238 RTC_FROM_HERE, [thread, type, common_name] {
239 static const rtc::SocketAddress turn_server_internal_address{
240 kTurnInternalAddress, kTurnInternalPort};
241 static const rtc::SocketAddress turn_server_external_address{
242 kTurnExternalAddress, kTurnExternalPort};
243 return std::make_unique<cricket::TestTurnServer>(
244 thread, turn_server_internal_address,
245 turn_server_external_address, type,
246 true /*ignore_bad_certs=*/, common_name);
247 });
248 turn_servers_.push_back(std::move(turn_server));
249 }
250
251 // First runs the call for kRampUpTimeMs to ramp up the bandwidth estimate.
252 // Then runs the test for the remaining test time, grabbing the bandwidth
253 // estimation stat, every kPollIntervalTimeMs. When finished, averages the
254 // bandwidth estimations and prints the bandwidth estimation result as a perf
255 // metric.
RunTest(const std::string & test_string)256 void RunTest(const std::string& test_string) {
257 rtc::Thread::Current()->ProcessMessages(kRampUpTimeMs);
258 int number_of_polls =
259 (kDefaultTestTimeMs - kRampUpTimeMs) / kPollIntervalTimeMs;
260 int total_bwe = 0;
261 for (int i = 0; i < number_of_polls; ++i) {
262 rtc::Thread::Current()->ProcessMessages(kPollIntervalTimeMs);
263 total_bwe += static_cast<int>(GetCallerAvailableBitrateEstimate());
264 }
265 double average_bandwidth_estimate = total_bwe / number_of_polls;
266 std::string value_description =
267 "bwe_after_" + std::to_string(kDefaultTestTimeMs / 1000) + "_seconds";
268 test::PrintResult("peerconnection_ramp_up_", test_string, value_description,
269 average_bandwidth_estimate, "bwe", false);
270 }
271
network_thread()272 rtc::Thread* network_thread() { return network_thread_.get(); }
273
firewall_socket_server()274 rtc::FirewallSocketServer* firewall_socket_server() {
275 return firewall_socket_server_.get();
276 }
277
caller()278 PeerConnectionWrapperForRampUpTest* caller() { return caller_.get(); }
279
callee()280 PeerConnectionWrapperForRampUpTest* callee() { return callee_.get(); }
281
282 private:
283 // Gets the caller's outgoing available bitrate from the stats. Returns 0 if
284 // something went wrong. It takes the outgoing bitrate from the current
285 // selected ICE candidate pair's stats.
GetCallerAvailableBitrateEstimate()286 double GetCallerAvailableBitrateEstimate() {
287 auto stats = caller_->GetStats();
288 auto transport_stats = stats->GetStatsOfType<RTCTransportStats>();
289 if (transport_stats.size() == 0u ||
290 !transport_stats[0]->selected_candidate_pair_id.is_defined()) {
291 return 0;
292 }
293 std::string selected_ice_id =
294 transport_stats[0]->selected_candidate_pair_id.ValueToString();
295 // Use the selected ICE candidate pair ID to get the appropriate ICE stats.
296 const RTCIceCandidatePairStats ice_candidate_pair_stats =
297 stats->Get(selected_ice_id)->cast_to<const RTCIceCandidatePairStats>();
298 if (ice_candidate_pair_stats.available_outgoing_bitrate.is_defined()) {
299 return *ice_candidate_pair_stats.available_outgoing_bitrate;
300 }
301 // We couldn't get the |available_outgoing_bitrate| for the active candidate
302 // pair.
303 return 0;
304 }
305
306 Clock* const clock_;
307 // The turn servers should be accessed & deleted on the network thread to
308 // avoid a race with the socket read/write which occurs on the network thread.
309 std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
310 // |virtual_socket_server_| is used by |network_thread_| so it must be
311 // destroyed later.
312 // TODO(bugs.webrtc.org/7668): We would like to update the virtual network we
313 // use for this test. VirtualSocketServer isn't ideal because:
314 // 1) It uses the same queue & network capacity for both directions.
315 // 2) VirtualSocketServer implements how the network bandwidth affects the
316 // send delay differently than the SimulatedNetwork, used by the
317 // FakeNetworkPipe. It would be ideal if all of levels of virtual
318 // networks used in testing were consistent.
319 // We would also like to update this test to record the time to ramp up,
320 // down, and back up (similar to in rampup_tests.cc). This is problematic with
321 // the VirtualSocketServer. The first ramp down time is very noisy and the
322 // second ramp up time can take up to 300 seconds, most likely due to a built
323 // up queue.
324 std::unique_ptr<rtc::VirtualSocketServer> virtual_socket_server_;
325 std::unique_ptr<rtc::FirewallSocketServer> firewall_socket_server_;
326 std::unique_ptr<rtc::Thread> network_thread_;
327 std::unique_ptr<rtc::Thread> worker_thread_;
328 // The |pc_factory| uses |network_thread_| & |worker_thread_|, so it must be
329 // destroyed first.
330 std::vector<std::unique_ptr<rtc::FakeNetworkManager>> fake_network_managers_;
331 rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
332 std::unique_ptr<PeerConnectionWrapperForRampUpTest> caller_;
333 std::unique_ptr<PeerConnectionWrapperForRampUpTest> callee_;
334 };
335
TEST_F(PeerConnectionRampUpTest,TurnOverTCP)336 TEST_F(PeerConnectionRampUpTest, TurnOverTCP) {
337 CreateTurnServer(cricket::ProtocolType::PROTO_TCP);
338 PeerConnectionInterface::IceServer ice_server;
339 std::string ice_server_url = "turn:" + std::string(kTurnInternalAddress) +
340 ":" + std::to_string(kTurnInternalPort) +
341 "?transport=tcp";
342 ice_server.urls.push_back(ice_server_url);
343 ice_server.username = "test";
344 ice_server.password = "test";
345 PeerConnectionInterface::RTCConfiguration client_1_config;
346 client_1_config.servers.push_back(ice_server);
347 client_1_config.type = PeerConnectionInterface::kRelay;
348 PeerConnectionInterface::RTCConfiguration client_2_config;
349 client_2_config.servers.push_back(ice_server);
350 client_2_config.type = PeerConnectionInterface::kRelay;
351 ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
352
353 SetupOneWayCall();
354 RunTest("turn_over_tcp");
355 }
356
TEST_F(PeerConnectionRampUpTest,TurnOverUDP)357 TEST_F(PeerConnectionRampUpTest, TurnOverUDP) {
358 CreateTurnServer(cricket::ProtocolType::PROTO_UDP);
359 PeerConnectionInterface::IceServer ice_server;
360 std::string ice_server_url = "turn:" + std::string(kTurnInternalAddress) +
361 ":" + std::to_string(kTurnInternalPort);
362
363 ice_server.urls.push_back(ice_server_url);
364 ice_server.username = "test";
365 ice_server.password = "test";
366 PeerConnectionInterface::RTCConfiguration client_1_config;
367 client_1_config.servers.push_back(ice_server);
368 client_1_config.type = PeerConnectionInterface::kRelay;
369 PeerConnectionInterface::RTCConfiguration client_2_config;
370 client_2_config.servers.push_back(ice_server);
371 client_2_config.type = PeerConnectionInterface::kRelay;
372 ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
373
374 SetupOneWayCall();
375 RunTest("turn_over_udp");
376 }
377
TEST_F(PeerConnectionRampUpTest,TurnOverTLS)378 TEST_F(PeerConnectionRampUpTest, TurnOverTLS) {
379 CreateTurnServer(cricket::ProtocolType::PROTO_TLS, kTurnInternalAddress);
380 PeerConnectionInterface::IceServer ice_server;
381 std::string ice_server_url = "turns:" + std::string(kTurnInternalAddress) +
382 ":" + std::to_string(kTurnInternalPort) +
383 "?transport=tcp";
384 ice_server.urls.push_back(ice_server_url);
385 ice_server.username = "test";
386 ice_server.password = "test";
387 PeerConnectionInterface::RTCConfiguration client_1_config;
388 client_1_config.servers.push_back(ice_server);
389 client_1_config.type = PeerConnectionInterface::kRelay;
390 PeerConnectionInterface::RTCConfiguration client_2_config;
391 client_2_config.servers.push_back(ice_server);
392 client_2_config.type = PeerConnectionInterface::kRelay;
393
394 ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
395
396 SetupOneWayCall();
397 RunTest("turn_over_tls");
398 }
399
TEST_F(PeerConnectionRampUpTest,UDPPeerToPeer)400 TEST_F(PeerConnectionRampUpTest, UDPPeerToPeer) {
401 PeerConnectionInterface::RTCConfiguration client_1_config;
402 client_1_config.tcp_candidate_policy =
403 PeerConnection::kTcpCandidatePolicyDisabled;
404 PeerConnectionInterface::RTCConfiguration client_2_config;
405 client_2_config.tcp_candidate_policy =
406 PeerConnection::kTcpCandidatePolicyDisabled;
407 ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
408
409 SetupOneWayCall();
410 RunTest("udp_peer_to_peer");
411 }
412
TEST_F(PeerConnectionRampUpTest,TCPPeerToPeer)413 TEST_F(PeerConnectionRampUpTest, TCPPeerToPeer) {
414 firewall_socket_server()->set_udp_sockets_enabled(false);
415 ASSERT_TRUE(CreatePeerConnectionWrappers(
416 PeerConnectionInterface::RTCConfiguration(),
417 PeerConnectionInterface::RTCConfiguration()));
418
419 SetupOneWayCall();
420 RunTest("tcp_peer_to_peer");
421 }
422
423 } // namespace webrtc
424