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/*
* 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 <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<SdpSemantics> {
protected:
PeerConnectionIntegrationTest()
: PeerConnectionIntegrationBaseTest(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(1000));
}
// 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,
Values(SdpSemantics::kPlanB_DEPRECATED,
SdpSemantics::kUnifiedPlan));
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