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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc')
-rw-r--r-- | third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc | 512 |
1 files changed, 512 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc b/third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc new file mode 100644 index 0000000000..b45571ec22 --- /dev/null +++ b/third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc @@ -0,0 +1,512 @@ +/* + * Copyright 2022 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// This file is intended for PeerConnection integration tests that are +// slow to execute (currently defined as more than 5 seconds per test). + +#include <stdint.h> + +#include <memory> +#include <string> +#include <tuple> +#include <utility> +#include <vector> + +#include "absl/algorithm/container.h" +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/dtmf_sender_interface.h" +#include "api/peer_connection_interface.h" +#include "api/rtp_receiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/units/time_delta.h" +#include "p2p/base/port_allocator.h" +#include "p2p/base/port_interface.h" +#include "p2p/base/stun_server.h" +#include "p2p/base/test_stun_server.h" +#include "pc/test/integration_test_helpers.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/fake_clock.h" +#include "rtc_base/fake_network.h" +#include "rtc_base/firewall_socket_server.h" +#include "rtc_base/gunit.h" +#include "rtc_base/logging.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/ssl_certificate.h" +#include "rtc_base/test_certificate_verifier.h" +#include "test/gmock.h" +#include "test/gtest.h" + +namespace webrtc { + +namespace { + +class PeerConnectionIntegrationTest + : public PeerConnectionIntegrationBaseTest, + public ::testing::WithParamInterface< + std::tuple<SdpSemantics, std::string>> { + protected: + PeerConnectionIntegrationTest() + : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam()), + std::get<1>(GetParam())) {} +}; + +// Fake clock must be set before threads are started to prevent race on +// Set/GetClockForTesting(). +// To achieve that, multiple inheritance is used as a mixin pattern +// where order of construction is finely controlled. +// This also ensures peerconnection is closed before switching back to non-fake +// clock, avoiding other races and DCHECK failures such as in rtp_sender.cc. +class FakeClockForTest : public rtc::ScopedFakeClock { + protected: + FakeClockForTest() { + // Some things use a time of "0" as a special value, so we need to start out + // the fake clock at a nonzero time. + // TODO(deadbeef): Fix this. + AdvanceTime(webrtc::TimeDelta::Seconds(1)); + } + + // Explicit handle. + ScopedFakeClock& FakeClock() { return *this; } +}; + +// Ensure FakeClockForTest is constructed first (see class for rationale). +class PeerConnectionIntegrationTestWithFakeClock + : public FakeClockForTest, + public PeerConnectionIntegrationTest {}; + +class PeerConnectionIntegrationTestPlanB + : public PeerConnectionIntegrationBaseTest { + protected: + PeerConnectionIntegrationTestPlanB() + : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB_DEPRECATED) {} +}; + +class PeerConnectionIntegrationTestUnifiedPlan + : public PeerConnectionIntegrationBaseTest { + protected: + PeerConnectionIntegrationTestUnifiedPlan() + : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {} +}; + +// Test the OnFirstPacketReceived callback from audio/video RtpReceivers. This +// includes testing that the callback is invoked if an observer is connected +// after the first packet has already been received. +TEST_P(PeerConnectionIntegrationTest, + RtpReceiverObserverOnFirstPacketReceived) { + ASSERT_TRUE(CreatePeerConnectionWrappers()); + ConnectFakeSignaling(); + caller()->AddAudioVideoTracks(); + callee()->AddAudioVideoTracks(); + // Start offer/answer exchange and wait for it to complete. + caller()->CreateAndSetAndSignalOffer(); + ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); + // Should be one receiver each for audio/video. + EXPECT_EQ(2U, caller()->rtp_receiver_observers().size()); + EXPECT_EQ(2U, callee()->rtp_receiver_observers().size()); + // Wait for all "first packet received" callbacks to be fired. + EXPECT_TRUE_WAIT( + absl::c_all_of(caller()->rtp_receiver_observers(), + [](const std::unique_ptr<MockRtpReceiverObserver>& o) { + return o->first_packet_received(); + }), + kMaxWaitForFramesMs); + EXPECT_TRUE_WAIT( + absl::c_all_of(callee()->rtp_receiver_observers(), + [](const std::unique_ptr<MockRtpReceiverObserver>& o) { + return o->first_packet_received(); + }), + kMaxWaitForFramesMs); + // If new observers are set after the first packet was already received, the + // callback should still be invoked. + caller()->ResetRtpReceiverObservers(); + callee()->ResetRtpReceiverObservers(); + EXPECT_EQ(2U, caller()->rtp_receiver_observers().size()); + EXPECT_EQ(2U, callee()->rtp_receiver_observers().size()); + EXPECT_TRUE( + absl::c_all_of(caller()->rtp_receiver_observers(), + [](const std::unique_ptr<MockRtpReceiverObserver>& o) { + return o->first_packet_received(); + })); + EXPECT_TRUE( + absl::c_all_of(callee()->rtp_receiver_observers(), + [](const std::unique_ptr<MockRtpReceiverObserver>& o) { + return o->first_packet_received(); + })); +} + +class DummyDtmfObserver : public DtmfSenderObserverInterface { + public: + DummyDtmfObserver() : completed_(false) {} + + // Implements DtmfSenderObserverInterface. + void OnToneChange(const std::string& tone) override { + tones_.push_back(tone); + if (tone.empty()) { + completed_ = true; + } + } + + const std::vector<std::string>& tones() const { return tones_; } + bool completed() const { return completed_; } + + private: + bool completed_; + std::vector<std::string> tones_; +}; + +TEST_P(PeerConnectionIntegrationTest, + SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection) { + static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0", + 3478}; + static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0}; + + // Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so + // that host name verification passes on the fake certificate. + CreateTurnServer(turn_server_internal_address, turn_server_external_address, + cricket::PROTO_TLS, "88.88.88.0"); + + webrtc::PeerConnectionInterface::IceServer ice_server; + ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp"); + ice_server.username = "test"; + ice_server.password = "test"; + + PeerConnectionInterface::RTCConfiguration client_1_config; + client_1_config.servers.push_back(ice_server); + client_1_config.type = webrtc::PeerConnectionInterface::kRelay; + + PeerConnectionInterface::RTCConfiguration client_2_config; + client_2_config.servers.push_back(ice_server); + // Setting the type to kRelay forces the connection to go through a TURN + // server. + client_2_config.type = webrtc::PeerConnectionInterface::kRelay; + + // Get a copy to the pointer so we can verify calls later. + rtc::TestCertificateVerifier* client_1_cert_verifier = + new rtc::TestCertificateVerifier(); + client_1_cert_verifier->verify_certificate_ = false; + rtc::TestCertificateVerifier* client_2_cert_verifier = + new rtc::TestCertificateVerifier(); + client_2_cert_verifier->verify_certificate_ = false; + + // Create the dependencies with the test certificate verifier. + webrtc::PeerConnectionDependencies client_1_deps(nullptr); + client_1_deps.tls_cert_verifier = + std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier); + webrtc::PeerConnectionDependencies client_2_deps(nullptr); + client_2_deps.tls_cert_verifier = + std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier); + + ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps( + client_1_config, std::move(client_1_deps), client_2_config, + std::move(client_2_deps))); + ConnectFakeSignaling(); + + // Set "offer to receive audio/video" without adding any tracks, so we just + // set up ICE/DTLS with no media. + PeerConnectionInterface::RTCOfferAnswerOptions options; + options.offer_to_receive_audio = 1; + options.offer_to_receive_video = 1; + caller()->SetOfferAnswerOptions(options); + caller()->CreateAndSetAndSignalOffer(); + bool wait_res = true; + // TODO(bugs.webrtc.org/9219): When IceConnectionState is implemented + // properly, should be able to just wait for a state of "failed" instead of + // waiting a fixed 10 seconds. + WAIT_(DtlsConnected(), kDefaultTimeout, wait_res); + ASSERT_FALSE(wait_res); + + EXPECT_GT(client_1_cert_verifier->call_count_, 0u); + EXPECT_GT(client_2_cert_verifier->call_count_, 0u); +} + +// Test that we can get capture start ntp time. +TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) { + ASSERT_TRUE(CreatePeerConnectionWrappers()); + ConnectFakeSignaling(); + caller()->AddAudioTrack(); + + callee()->AddAudioTrack(); + + // Do offer/answer, wait for the callee to receive some frames. + caller()->CreateAndSetAndSignalOffer(); + ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); + + // Get the remote audio track created on the receiver, so they can be used as + // GetStats filters. + auto receivers = callee()->pc()->GetReceivers(); + ASSERT_EQ(1u, receivers.size()); + auto remote_audio_track = receivers[0]->track(); + + // Get the audio output level stats. Note that the level is not available + // until an RTCP packet has been received. + EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track.get()) + ->CaptureStartNtpTime() > 0, + 2 * kMaxWaitForFramesMs); +} + +// Test that firewalling the ICE connection causes the clients to identify the +// disconnected state and then removing the firewall causes them to reconnect. +class PeerConnectionIntegrationIceStatesTest + : public PeerConnectionIntegrationBaseTest, + public ::testing::WithParamInterface< + std::tuple<SdpSemantics, std::tuple<std::string, uint32_t>>> { + protected: + PeerConnectionIntegrationIceStatesTest() + : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())) { + port_allocator_flags_ = std::get<1>(std::get<1>(GetParam())); + } + + void StartStunServer(const SocketAddress& server_address) { + stun_server_.reset( + cricket::TestStunServer::Create(firewall(), server_address)); + } + + bool TestIPv6() { + return (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6); + } + + void SetPortAllocatorFlags() { + PeerConnectionIntegrationBaseTest::SetPortAllocatorFlags( + port_allocator_flags_, port_allocator_flags_); + } + + std::vector<SocketAddress> CallerAddresses() { + std::vector<SocketAddress> addresses; + addresses.push_back(SocketAddress("1.1.1.1", 0)); + if (TestIPv6()) { + addresses.push_back(SocketAddress("1111:0:a:b:c:d:e:f", 0)); + } + return addresses; + } + + std::vector<SocketAddress> CalleeAddresses() { + std::vector<SocketAddress> addresses; + addresses.push_back(SocketAddress("2.2.2.2", 0)); + if (TestIPv6()) { + addresses.push_back(SocketAddress("2222:0:a:b:c:d:e:f", 0)); + } + return addresses; + } + + void SetUpNetworkInterfaces() { + // Remove the default interfaces added by the test infrastructure. + caller()->network_manager()->RemoveInterface(kDefaultLocalAddress); + callee()->network_manager()->RemoveInterface(kDefaultLocalAddress); + + // Add network addresses for test. + for (const auto& caller_address : CallerAddresses()) { + caller()->network_manager()->AddInterface(caller_address); + } + for (const auto& callee_address : CalleeAddresses()) { + callee()->network_manager()->AddInterface(callee_address); + } + } + + private: + uint32_t port_allocator_flags_; + std::unique_ptr<cricket::TestStunServer> stun_server_; +}; + +// Ensure FakeClockForTest is constructed first (see class for rationale). +class PeerConnectionIntegrationIceStatesTestWithFakeClock + : public FakeClockForTest, + public PeerConnectionIntegrationIceStatesTest {}; + +#if !defined(THREAD_SANITIZER) +// This test provokes TSAN errors. bugs.webrtc.org/11282 + +// Tests that the PeerConnection goes through all the ICE gathering/connection +// states over the duration of the call. This includes Disconnected and Failed +// states, induced by putting a firewall between the peers and waiting for them +// to time out. +TEST_P(PeerConnectionIntegrationIceStatesTestWithFakeClock, VerifyIceStates) { + const SocketAddress kStunServerAddress = + SocketAddress("99.99.99.1", cricket::STUN_SERVER_PORT); + StartStunServer(kStunServerAddress); + + PeerConnectionInterface::RTCConfiguration config; + PeerConnectionInterface::IceServer ice_stun_server; + ice_stun_server.urls.push_back( + "stun:" + kStunServerAddress.HostAsURIString() + ":" + + kStunServerAddress.PortAsString()); + config.servers.push_back(ice_stun_server); + + ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config)); + ConnectFakeSignaling(); + SetPortAllocatorFlags(); + SetUpNetworkInterfaces(); + caller()->AddAudioVideoTracks(); + callee()->AddAudioVideoTracks(); + + // Initial state before anything happens. + ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew, + caller()->ice_gathering_state()); + ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew, + caller()->ice_connection_state()); + ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew, + caller()->standardized_ice_connection_state()); + + // Start the call by creating the offer, setting it as the local description, + // then sending it to the peer who will respond with an answer. This happens + // asynchronously so that we can watch the states as it runs in the + // background. + caller()->CreateAndSetAndSignalOffer(); + + ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted, + caller()->ice_connection_state(), kDefaultTimeout, + FakeClock()); + ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted, + caller()->standardized_ice_connection_state(), + kDefaultTimeout, FakeClock()); + + // Verify that the observer was notified of the intermediate transitions. + EXPECT_THAT(caller()->ice_connection_state_history(), + ElementsAre(PeerConnectionInterface::kIceConnectionChecking, + PeerConnectionInterface::kIceConnectionConnected, + PeerConnectionInterface::kIceConnectionCompleted)); + EXPECT_THAT(caller()->standardized_ice_connection_state_history(), + ElementsAre(PeerConnectionInterface::kIceConnectionChecking, + PeerConnectionInterface::kIceConnectionConnected, + PeerConnectionInterface::kIceConnectionCompleted)); + EXPECT_THAT( + caller()->peer_connection_state_history(), + ElementsAre(PeerConnectionInterface::PeerConnectionState::kConnecting, + PeerConnectionInterface::PeerConnectionState::kConnected)); + EXPECT_THAT(caller()->ice_gathering_state_history(), + ElementsAre(PeerConnectionInterface::kIceGatheringGathering, + PeerConnectionInterface::kIceGatheringComplete)); + + // Block connections to/from the caller and wait for ICE to become + // disconnected. + for (const auto& caller_address : CallerAddresses()) { + firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address); + } + RTC_LOG(LS_INFO) << "Firewall rules applied"; + ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionDisconnected, + caller()->ice_connection_state(), kDefaultTimeout, + FakeClock()); + ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionDisconnected, + caller()->standardized_ice_connection_state(), + kDefaultTimeout, FakeClock()); + + // Let ICE re-establish by removing the firewall rules. + firewall()->ClearRules(); + RTC_LOG(LS_INFO) << "Firewall rules cleared"; + ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted, + caller()->ice_connection_state(), kDefaultTimeout, + FakeClock()); + ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted, + caller()->standardized_ice_connection_state(), + kDefaultTimeout, FakeClock()); + + // According to RFC7675, if there is no response within 30 seconds then the + // peer should consider the other side to have rejected the connection. This + // is signaled by the state transitioning to "failed". + constexpr int kConsentTimeout = 30000; + for (const auto& caller_address : CallerAddresses()) { + firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address); + } + RTC_LOG(LS_INFO) << "Firewall rules applied again"; + ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed, + caller()->ice_connection_state(), kConsentTimeout, + FakeClock()); + ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed, + caller()->standardized_ice_connection_state(), + kConsentTimeout, FakeClock()); +} +#endif + +// This test sets up a call that's transferred to a new caller with a different +// DTLS fingerprint. +TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) { + ASSERT_TRUE(CreatePeerConnectionWrappers()); + ConnectFakeSignaling(); + caller()->AddAudioVideoTracks(); + callee()->AddAudioVideoTracks(); + caller()->CreateAndSetAndSignalOffer(); + ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); + + // Keep the original peer around which will still send packets to the + // receiving client. These SRTP packets will be dropped. + std::unique_ptr<PeerConnectionIntegrationWrapper> original_peer( + SetCallerPcWrapperAndReturnCurrent( + CreatePeerConnectionWrapperWithAlternateKey().release())); + // TODO(deadbeef): Why do we call Close here? That goes against the comment + // directly above. + original_peer->pc()->Close(); + + ConnectFakeSignaling(); + caller()->AddAudioVideoTracks(); + caller()->CreateAndSetAndSignalOffer(); + ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); + // Wait for some additional frames to be transmitted end-to-end. + MediaExpectations media_expectations; + media_expectations.ExpectBidirectionalAudioAndVideo(); + ASSERT_TRUE(ExpectNewFrames(media_expectations)); +} + +// This test sets up a call that's transferred to a new callee with a different +// DTLS fingerprint. +TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) { + ASSERT_TRUE(CreatePeerConnectionWrappers()); + ConnectFakeSignaling(); + caller()->AddAudioVideoTracks(); + callee()->AddAudioVideoTracks(); + caller()->CreateAndSetAndSignalOffer(); + ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); + + // Keep the original peer around which will still send packets to the + // receiving client. These SRTP packets will be dropped. + std::unique_ptr<PeerConnectionIntegrationWrapper> original_peer( + SetCalleePcWrapperAndReturnCurrent( + CreatePeerConnectionWrapperWithAlternateKey().release())); + // TODO(deadbeef): Why do we call Close here? That goes against the comment + // directly above. + original_peer->pc()->Close(); + + ConnectFakeSignaling(); + callee()->AddAudioVideoTracks(); + caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); + caller()->CreateAndSetAndSignalOffer(); + ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); + // Wait for some additional frames to be transmitted end-to-end. + MediaExpectations media_expectations; + media_expectations.ExpectBidirectionalAudioAndVideo(); + ASSERT_TRUE(ExpectNewFrames(media_expectations)); +} + +INSTANTIATE_TEST_SUITE_P( + PeerConnectionIntegrationTest, + PeerConnectionIntegrationTest, + Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan), + Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/", + "WebRTC-FrameBuffer3/arm:FrameBuffer3/", + "WebRTC-FrameBuffer3/arm:SyncDecoding/"))); + +constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP | + cricket::PORTALLOCATOR_DISABLE_STUN | + cricket::PORTALLOCATOR_DISABLE_RELAY; +constexpr uint32_t kFlagsIPv6NoStun = + cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_STUN | + cricket::PORTALLOCATOR_ENABLE_IPV6 | cricket::PORTALLOCATOR_DISABLE_RELAY; +constexpr uint32_t kFlagsIPv4Stun = + cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_RELAY; + +INSTANTIATE_TEST_SUITE_P( + PeerConnectionIntegrationTest, + PeerConnectionIntegrationIceStatesTestWithFakeClock, + Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan), + Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun), + std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun), + std::make_pair("IPv4 with STUN", kFlagsIPv4Stun)))); + +} // namespace +} // namespace webrtc |