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/*
* Copyright 2017 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.
*/
#include <stddef.h>
#include <memory>
#include <ostream>
#include <string>
#include <tuple>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/create_peerconnection_factory.h"
#include "api/crypto/crypto_options.h"
#include "api/jsep.h"
#include "api/peer_connection_interface.h"
#include "api/scoped_refptr.h"
#include "api/video_codecs/video_decoder_factory_template.h"
#include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h"
#include "api/video_codecs/video_encoder_factory_template.h"
#include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/fake_port_allocator.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
#include "pc/media_protocol_names.h"
#include "pc/media_session.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/sdp_utils.h"
#include "pc/session_description.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/checks.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/ssl_fingerprint.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/scoped_key_value_config.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/android_test_initializer.h"
#endif
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/fake_rtc_certificate_generator.h"
#include "rtc_base/gunit.h"
#include "rtc_base/virtual_socket_server.h"
namespace webrtc {
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
using ::testing::Combine;
using ::testing::HasSubstr;
using ::testing::Values;
constexpr int kGenerateCertTimeout = 1000;
class PeerConnectionCryptoBaseTest : public ::testing::Test {
protected:
typedef std::unique_ptr<PeerConnectionWrapper> WrapperPtr;
explicit PeerConnectionCryptoBaseTest(SdpSemantics sdp_semantics)
: vss_(new rtc::VirtualSocketServer()),
main_(vss_.get()),
sdp_semantics_(sdp_semantics) {
#ifdef WEBRTC_ANDROID
InitializeAndroidObjects();
#endif
pc_factory_ = CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
FakeAudioCaptureModule::Create(), CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory(),
std::make_unique<VideoEncoderFactoryTemplate<
LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>(),
std::make_unique<VideoDecoderFactoryTemplate<
LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter,
OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(),
nullptr /* audio_mixer */, nullptr /* audio_processing */);
}
WrapperPtr CreatePeerConnection() {
return CreatePeerConnection(RTCConfiguration());
}
WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
return CreatePeerConnection(config, nullptr);
}
WrapperPtr CreatePeerConnection(
const RTCConfiguration& config,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_gen) {
auto fake_port_allocator = std::make_unique<cricket::FakePortAllocator>(
rtc::Thread::Current(),
std::make_unique<rtc::BasicPacketSocketFactory>(vss_.get()),
&field_trials_);
auto observer = std::make_unique<MockPeerConnectionObserver>();
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
PeerConnectionDependencies pc_dependencies(observer.get());
pc_dependencies.allocator = std::move(fake_port_allocator);
pc_dependencies.cert_generator = std::move(cert_gen);
auto result = pc_factory_->CreatePeerConnectionOrError(
modified_config, std::move(pc_dependencies));
if (!result.ok()) {
return nullptr;
}
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory_, result.MoveValue(), std::move(observer));
}
// Accepts the same arguments as CreatePeerConnection and adds default audio
// and video tracks.
template <typename... Args>
WrapperPtr CreatePeerConnectionWithAudioVideo(Args&&... args) {
auto wrapper = CreatePeerConnection(std::forward<Args>(args)...);
if (!wrapper) {
return nullptr;
}
wrapper->AddAudioTrack("a");
wrapper->AddVideoTrack("v");
return wrapper;
}
cricket::ConnectionRole& AudioConnectionRole(
cricket::SessionDescription* desc) {
return ConnectionRoleFromContent(desc, cricket::GetFirstAudioContent(desc));
}
cricket::ConnectionRole& VideoConnectionRole(
cricket::SessionDescription* desc) {
return ConnectionRoleFromContent(desc, cricket::GetFirstVideoContent(desc));
}
cricket::ConnectionRole& ConnectionRoleFromContent(
cricket::SessionDescription* desc,
cricket::ContentInfo* content) {
RTC_DCHECK(content);
auto* transport_info = desc->GetTransportInfoByName(content->name);
RTC_DCHECK(transport_info);
return transport_info->description.connection_role;
}
test::ScopedKeyValueConfig field_trials_;
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
const SdpSemantics sdp_semantics_;
};
SdpContentPredicate HaveDtlsFingerprint() {
return [](const cricket::ContentInfo* content,
const cricket::TransportInfo* transport) {
return transport->description.identity_fingerprint != nullptr;
};
}
SdpContentPredicate HaveProtocol(const std::string& protocol) {
return [protocol](const cricket::ContentInfo* content,
const cricket::TransportInfo* transport) {
return content->media_description()->protocol() == protocol;
};
}
class PeerConnectionCryptoTest
: public PeerConnectionCryptoBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionCryptoTest() : PeerConnectionCryptoBaseTest(GetParam()) {}
};
SdpContentMutator RemoveDtlsFingerprint() {
return [](cricket::ContentInfo* content, cricket::TransportInfo* transport) {
transport->description.identity_fingerprint.reset();
};
}
// When DTLS is enabled, the SDP offer/answer should have a DTLS fingerprint
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsEnabled) {
RTCConfiguration config;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
ASSERT_TRUE(offer);
ASSERT_FALSE(offer->description()->contents().empty());
EXPECT_TRUE(SdpContentsAll(HaveDtlsFingerprint(), offer->description()));
EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolDtlsSavpf),
offer->description()));
}
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsEnabled) {
RTCConfiguration config;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
callee->SetRemoteDescription(caller->CreateOffer());
auto answer = callee->CreateAnswer();
ASSERT_TRUE(answer);
ASSERT_FALSE(answer->description()->contents().empty());
EXPECT_TRUE(SdpContentsAll(HaveDtlsFingerprint(), answer->description()));
EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolDtlsSavpf),
answer->description()));
}
// The following group tests that two PeerConnections can successfully exchange
// an offer/answer when DTLS is on and that they will refuse any offer/answer
// applied locally/remotely if it does not include a DTLS fingerprint.
TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenDtlsOn) {
RTCConfiguration config;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOfferAndSetAsLocal();
ASSERT_TRUE(offer);
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswerAndSetAsLocal();
ASSERT_TRUE(answer);
ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
}
TEST_P(PeerConnectionCryptoTest,
FailToSetLocalOfferWithNoFingerprintWhenDtlsOn) {
RTCConfiguration config;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
SdpContentsForEach(RemoveDtlsFingerprint(), offer->description());
EXPECT_FALSE(caller->SetLocalDescription(std::move(offer)));
}
TEST_P(PeerConnectionCryptoTest,
FailToSetRemoteOfferWithNoFingerprintWhenDtlsOn) {
RTCConfiguration config;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
SdpContentsForEach(RemoveDtlsFingerprint(), offer->description());
EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer)));
}
TEST_P(PeerConnectionCryptoTest,
FailToSetLocalAnswerWithNoFingerprintWhenDtlsOn) {
RTCConfiguration config;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal());
auto answer = callee->CreateAnswer();
SdpContentsForEach(RemoveDtlsFingerprint(), answer->description());
}
TEST_P(PeerConnectionCryptoTest,
FailToSetRemoteAnswerWithNoFingerprintWhenDtlsOn) {
RTCConfiguration config;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal());
auto answer = callee->CreateAnswerAndSetAsLocal();
SdpContentsForEach(RemoveDtlsFingerprint(), answer->description());
EXPECT_FALSE(caller->SetRemoteDescription(std::move(answer)));
}
// Tests that a DTLS call can be established when the certificate is specified
// in the PeerConnection config and no certificate generator is specified.
TEST_P(PeerConnectionCryptoTest,
ExchangeOfferAnswerWhenDtlsCertificateInConfig) {
RTCConfiguration caller_config;
caller_config.certificates.push_back(
FakeRTCCertificateGenerator::GenerateCertificate());
auto caller = CreatePeerConnectionWithAudioVideo(caller_config);
RTCConfiguration callee_config;
callee_config.certificates.push_back(
FakeRTCCertificateGenerator::GenerateCertificate());
auto callee = CreatePeerConnectionWithAudioVideo(callee_config);
auto offer = caller->CreateOfferAndSetAsLocal();
ASSERT_TRUE(offer);
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswerAndSetAsLocal();
ASSERT_TRUE(answer);
ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
}
// The following parameterized test verifies that CreateOffer/CreateAnswer
// returns successfully (or with failure if the underlying certificate generator
// fails) no matter when the DTLS certificate is generated. If multiple
// CreateOffer/CreateAnswer calls are made while waiting for the certificate,
// they all finish after the certificate is generated.
// Whether the certificate will be generated before calling CreateOffer or
// while CreateOffer is executing.
enum class CertGenTime { kBefore, kDuring };
std::ostream& operator<<(std::ostream& out, CertGenTime value) {
switch (value) {
case CertGenTime::kBefore:
return out << "before";
case CertGenTime::kDuring:
return out << "during";
default:
return out << "unknown";
}
}
// Whether the fake certificate generator will produce a certificate or fail.
enum class CertGenResult { kSucceed, kFail };
std::ostream& operator<<(std::ostream& out, CertGenResult value) {
switch (value) {
case CertGenResult::kSucceed:
return out << "succeed";
case CertGenResult::kFail:
return out << "fail";
default:
return out << "unknown";
}
}
class PeerConnectionCryptoDtlsCertGenTest
: public PeerConnectionCryptoBaseTest,
public ::testing::WithParamInterface<std::tuple<SdpSemantics,
SdpType,
CertGenTime,
CertGenResult,
size_t>> {
protected:
PeerConnectionCryptoDtlsCertGenTest()
: PeerConnectionCryptoBaseTest(std::get<0>(GetParam())) {
sdp_type_ = std::get<1>(GetParam());
cert_gen_time_ = std::get<2>(GetParam());
cert_gen_result_ = std::get<3>(GetParam());
concurrent_calls_ = std::get<4>(GetParam());
}
SdpType sdp_type_;
CertGenTime cert_gen_time_;
CertGenResult cert_gen_result_;
size_t concurrent_calls_;
};
TEST_P(PeerConnectionCryptoDtlsCertGenTest, TestCertificateGeneration) {
RTCConfiguration config;
auto owned_fake_certificate_generator =
std::make_unique<FakeRTCCertificateGenerator>();
auto* fake_certificate_generator = owned_fake_certificate_generator.get();
fake_certificate_generator->set_should_fail(cert_gen_result_ ==
CertGenResult::kFail);
fake_certificate_generator->set_should_wait(cert_gen_time_ ==
CertGenTime::kDuring);
WrapperPtr pc;
if (sdp_type_ == SdpType::kOffer) {
pc = CreatePeerConnectionWithAudioVideo(
config, std::move(owned_fake_certificate_generator));
} else {
auto caller = CreatePeerConnectionWithAudioVideo(config);
pc = CreatePeerConnectionWithAudioVideo(
config, std::move(owned_fake_certificate_generator));
pc->SetRemoteDescription(caller->CreateOfferAndSetAsLocal());
}
if (cert_gen_time_ == CertGenTime::kBefore) {
ASSERT_TRUE_WAIT(fake_certificate_generator->generated_certificates() +
fake_certificate_generator->generated_failures() >
0,
kGenerateCertTimeout);
} else {
ASSERT_EQ(fake_certificate_generator->generated_certificates(), 0);
fake_certificate_generator->set_should_wait(false);
}
std::vector<rtc::scoped_refptr<MockCreateSessionDescriptionObserver>>
observers;
for (size_t i = 0; i < concurrent_calls_; i++) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
observers.push_back(observer);
if (sdp_type_ == SdpType::kOffer) {
pc->pc()->CreateOffer(observer.get(),
PeerConnectionInterface::RTCOfferAnswerOptions());
} else {
pc->pc()->CreateAnswer(observer.get(),
PeerConnectionInterface::RTCOfferAnswerOptions());
}
}
for (auto& observer : observers) {
EXPECT_TRUE_WAIT(observer->called(), 1000);
if (cert_gen_result_ == CertGenResult::kSucceed) {
EXPECT_TRUE(observer->result());
} else {
EXPECT_FALSE(observer->result());
}
}
}
INSTANTIATE_TEST_SUITE_P(
PeerConnectionCryptoTest,
PeerConnectionCryptoDtlsCertGenTest,
Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
Values(SdpType::kOffer, SdpType::kAnswer),
Values(CertGenTime::kBefore, CertGenTime::kDuring),
Values(CertGenResult::kSucceed, CertGenResult::kFail),
Values(1, 3)));
// Test that we can create and set an answer correctly when different
// SSL roles have been negotiated for different transports.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4525
TEST_P(PeerConnectionCryptoTest, CreateAnswerWithDifferentSslRoles) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
RTCOfferAnswerOptions options_no_bundle;
options_no_bundle.use_rtp_mux = false;
// First, negotiate different SSL roles for audio and video.
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
auto answer = callee->CreateAnswer(options_no_bundle);
AudioConnectionRole(answer->description()) = cricket::CONNECTIONROLE_ACTIVE;
VideoConnectionRole(answer->description()) = cricket::CONNECTIONROLE_PASSIVE;
ASSERT_TRUE(
callee->SetLocalDescription(CloneSessionDescription(answer.get())));
ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
// Now create an offer in the reverse direction, and ensure the initial
// offerer responds with an answer with the correct SSL roles.
ASSERT_TRUE(caller->SetRemoteDescription(callee->CreateOfferAndSetAsLocal()));
answer = caller->CreateAnswer(options_no_bundle);
EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE,
AudioConnectionRole(answer->description()));
EXPECT_EQ(cricket::CONNECTIONROLE_ACTIVE,
VideoConnectionRole(answer->description()));
ASSERT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(answer.get())));
ASSERT_TRUE(callee->SetRemoteDescription(std::move(answer)));
// Lastly, start BUNDLE-ing on "audio", expecting that the "passive" role of
// audio is transferred over to video in the answer that completes the BUNDLE
// negotiation.
RTCOfferAnswerOptions options_bundle;
options_bundle.use_rtp_mux = true;
ASSERT_TRUE(caller->SetRemoteDescription(callee->CreateOfferAndSetAsLocal()));
answer = caller->CreateAnswer(options_bundle);
EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE,
AudioConnectionRole(answer->description()));
EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE,
VideoConnectionRole(answer->description()));
ASSERT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(answer.get())));
ASSERT_TRUE(callee->SetRemoteDescription(std::move(answer)));
}
// Tests that if the DTLS fingerprint is invalid then all future calls to
// SetLocalDescription and SetRemoteDescription will fail due to a session
// error.
// This is a regression test for crbug.com/800775
TEST_P(PeerConnectionCryptoTest, SessionErrorIfFingerprintInvalid) {
auto callee_certificate = rtc::RTCCertificate::FromPEM(kRsaPems[0]);
auto other_certificate = rtc::RTCCertificate::FromPEM(kRsaPems[1]);
auto caller = CreatePeerConnectionWithAudioVideo();
RTCConfiguration callee_config;
callee_config.certificates.push_back(callee_certificate);
auto callee = CreatePeerConnectionWithAudioVideo(callee_config);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
// Create an invalid answer with the other certificate's fingerprint.
auto valid_answer = callee->CreateAnswer();
auto invalid_answer = CloneSessionDescription(valid_answer.get());
auto* audio_content =
cricket::GetFirstAudioContent(invalid_answer->description());
ASSERT_TRUE(audio_content);
auto* audio_transport_info =
invalid_answer->description()->GetTransportInfoByName(
audio_content->name);
ASSERT_TRUE(audio_transport_info);
audio_transport_info->description.identity_fingerprint =
rtc::SSLFingerprint::CreateFromCertificate(*other_certificate);
// Set the invalid answer and expect a fingerprint error.
std::string error;
ASSERT_FALSE(callee->SetLocalDescription(std::move(invalid_answer), &error));
EXPECT_THAT(error, HasSubstr("Local fingerprint does not match identity."));
// Make sure that setting a valid remote offer or local answer also fails now.
ASSERT_FALSE(callee->SetRemoteDescription(caller->CreateOffer(), &error));
EXPECT_THAT(error, HasSubstr("Session error code: ERROR_CONTENT."));
ASSERT_FALSE(callee->SetLocalDescription(std::move(valid_answer), &error));
EXPECT_THAT(error, HasSubstr("Session error code: ERROR_CONTENT."));
}
INSTANTIATE_TEST_SUITE_P(PeerConnectionCryptoTest,
PeerConnectionCryptoTest,
Values(SdpSemantics::kPlanB_DEPRECATED,
SdpSemantics::kUnifiedPlan));
} // namespace webrtc