webrtc_m130/pc/peerconnection_crypto_unittest.cc

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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 "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "p2p/base/fakeportallocator.h"
#include "pc/mediasession.h"
#include "pc/peerconnectionwrapper.h"
#include "pc/sdputils.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
#include "absl/memory/memory.h"
#include "pc/test/fakeaudiocapturemodule.h"
#include "pc/test/fakertccertificategenerator.h"
#include "rtc_base/gunit.h"
#include "rtc_base/virtualsocketserver.h"
namespace webrtc {
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
using ::testing::Values;
using ::testing::Combine;
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(), CreateBuiltinVideoEncoderFactory(),
CreateBuiltinVideoDecoderFactory(), 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 = absl::make_unique<cricket::FakePortAllocator>(
rtc::Thread::Current(), nullptr);
auto observer = absl::make_unique<MockPeerConnectionObserver>();
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
auto pc = pc_factory_->CreatePeerConnection(
modified_config, std::move(fake_port_allocator), std::move(cert_gen),
observer.get());
if (!pc) {
return nullptr;
}
observer->SetPeerConnectionInterface(pc.get());
return absl::make_unique<PeerConnectionWrapper>(pc_factory_, pc,
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;
}
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 HaveSdesCryptos() {
return [](const cricket::ContentInfo* content,
const cricket::TransportInfo* transport) {
return !content->media_description()->cryptos().empty();
};
}
SdpContentPredicate HaveProtocol(const std::string& protocol) {
return [protocol](const cricket::ContentInfo* content,
const cricket::TransportInfo* transport) {
return content->media_description()->protocol() == protocol;
};
}
SdpContentPredicate HaveSdesGcmCryptos(size_t num_crypto_suites) {
return [num_crypto_suites](const cricket::ContentInfo* content,
const cricket::TransportInfo* transport) {
const auto& cryptos = content->media_description()->cryptos();
if (cryptos.size() != num_crypto_suites) {
return false;
}
const cricket::CryptoParams first_params = cryptos[0];
return first_params.key_params.size() == 67U &&
first_params.cipher_suite == "AEAD_AES_256_GCM";
};
}
class PeerConnectionCryptoTest
: public PeerConnectionCryptoBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionCryptoTest() : PeerConnectionCryptoBaseTest(GetParam()) {}
};
SdpContentMutator RemoveSdesCryptos() {
return [](cricket::ContentInfo* content, cricket::TransportInfo* transport) {
content->media_description()->set_cryptos({});
};
}
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 and
// no SDES cryptos.
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsEnabled) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
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(SdpContentsNone(HaveSdesCryptos(), offer->description()));
EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolDtlsSavpf),
offer->description()));
}
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsEnabled) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
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(SdpContentsNone(HaveSdesCryptos(), answer->description()));
EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolDtlsSavpf),
answer->description()));
}
// When DTLS is disabled, the SDP offer/answer should include SDES cryptos and
// should not have a DTLS fingerprint.
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsDisabled) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
ASSERT_TRUE(offer);
ASSERT_FALSE(offer->description()->contents().empty());
EXPECT_TRUE(SdpContentsAll(HaveSdesCryptos(), offer->description()));
EXPECT_TRUE(SdpContentsNone(HaveDtlsFingerprint(), offer->description()));
EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolSavpf),
offer->description()));
}
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsDisabled) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
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(HaveSdesCryptos(), answer->description()));
EXPECT_TRUE(SdpContentsNone(HaveDtlsFingerprint(), answer->description()));
EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolSavpf),
answer->description()));
}
// When encryption is disabled, the SDP offer/answer should have neither a DTLS
// fingerprint nor any SDES crypto options.
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenEncryptionDisabled) {
PeerConnectionFactoryInterface::Options options;
options.disable_encryption = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
ASSERT_TRUE(offer);
ASSERT_FALSE(offer->description()->contents().empty());
EXPECT_TRUE(SdpContentsNone(HaveSdesCryptos(), offer->description()));
EXPECT_TRUE(SdpContentsNone(HaveDtlsFingerprint(), offer->description()));
EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolAvpf),
offer->description()));
}
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenEncryptionDisabled) {
PeerConnectionFactoryInterface::Options options;
options.disable_encryption = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
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(SdpContentsNone(HaveSdesCryptos(), answer->description()));
EXPECT_TRUE(SdpContentsNone(HaveDtlsFingerprint(), answer->description()));
EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolAvpf),
answer->description()));
}
// CryptoOptions has been promoted to RTCConfiguration. As such if it is ever
// set in the configuration it should overrite the settings set in the factory.
TEST_P(PeerConnectionCryptoTest, RTCConfigurationCryptoOptionOverridesFactory) {
PeerConnectionFactoryInterface::Options options;
options.crypto_options.srtp.enable_gcm_crypto_suites = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
CryptoOptions crypto_options;
crypto_options.srtp.enable_gcm_crypto_suites = false;
config.crypto_options = crypto_options;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
ASSERT_TRUE(offer);
ASSERT_FALSE(offer->description()->contents().empty());
// This should exist if GCM is enabled see CorrectCryptoInOfferWithSdesAndGcm
EXPECT_FALSE(SdpContentsAll(HaveSdesGcmCryptos(3), offer->description()));
}
// When DTLS is disabled and GCM cipher suites are enabled, the SDP offer/answer
// should have the correct ciphers in the SDES crypto options.
// With GCM cipher suites enabled, there will be 3 cryptos in the offer and 1
// in the answer.
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWithSdesAndGcm) {
PeerConnectionFactoryInterface::Options options;
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
options.crypto_options.srtp.enable_gcm_crypto_suites = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
ASSERT_TRUE(offer);
ASSERT_FALSE(offer->description()->contents().empty());
EXPECT_TRUE(SdpContentsAll(HaveSdesGcmCryptos(3), offer->description()));
}
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWithSdesAndGcm) {
PeerConnectionFactoryInterface::Options options;
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
options.crypto_options.srtp.enable_gcm_crypto_suites = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
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(HaveSdesGcmCryptos(1), answer->description()));
}
TEST_P(PeerConnectionCryptoTest, CanSetSdesGcmRemoteOfferAndLocalAnswer) {
PeerConnectionFactoryInterface::Options options;
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
options.crypto_options.srtp.enable_gcm_crypto_suites = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
ASSERT_TRUE(offer);
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswer();
ASSERT_TRUE(answer);
ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
}
// The following group tests that two PeerConnections can successfully exchange
// an offer/answer when DTLS is off and that they will refuse any offer/answer
// applied locally/remotely if it does not include SDES cryptos.
TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenSdesOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
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, FailToSetLocalOfferWithNoCryptosWhenSdesOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
SdpContentsForEach(RemoveSdesCryptos(), offer->description());
EXPECT_FALSE(caller->SetLocalDescription(std::move(offer)));
}
TEST_P(PeerConnectionCryptoTest, FailToSetRemoteOfferWithNoCryptosWhenSdesOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
SdpContentsForEach(RemoveSdesCryptos(), offer->description());
EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer)));
}
TEST_P(PeerConnectionCryptoTest, FailToSetLocalAnswerWithNoCryptosWhenSdesOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal());
auto answer = callee->CreateAnswer();
SdpContentsForEach(RemoveSdesCryptos(), answer->description());
EXPECT_FALSE(callee->SetLocalDescription(std::move(answer)));
}
TEST_P(PeerConnectionCryptoTest, FailToSetRemoteAnswerWithNoCryptosWhenSdesOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal());
auto answer = callee->CreateAnswerAndSetAsLocal();
SdpContentsForEach(RemoveSdesCryptos(), answer->description());
EXPECT_FALSE(caller->SetRemoteDescription(std::move(answer)));
}
// 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;
config.enable_dtls_srtp.emplace(true);
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;
config.enable_dtls_srtp.emplace(true);
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;
config.enable_dtls_srtp.emplace(true);
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;
config.enable_dtls_srtp.emplace(true);
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;
config.enable_dtls_srtp.emplace(true);
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)));
}
// Test that an offer/answer can be exchanged when encryption is disabled.
TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenNoEncryption) {
PeerConnectionFactoryInterface::Options options;
options.disable_encryption = true;
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
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)));
}
// 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.enable_dtls_srtp.emplace(true);
caller_config.certificates.push_back(
FakeRTCCertificateGenerator::GenerateCertificate());
auto caller = CreatePeerConnectionWithAudioVideo(caller_config);
RTCConfiguration callee_config;
callee_config.enable_dtls_srtp.emplace(true);
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;
config.enable_dtls_srtp.emplace(true);
auto owned_fake_certificate_generator =
absl::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 =
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>();
observers.push_back(observer);
if (sdp_type_ == SdpType::kOffer) {
pc->pc()->CreateOffer(observer,
PeerConnectionInterface::RTCOfferAnswerOptions());
} else {
pc->pc()->CreateAnswer(observer,
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_CASE_P(
PeerConnectionCryptoTest,
PeerConnectionCryptoDtlsCertGenTest,
Combine(Values(SdpSemantics::kPlanB, 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.enable_dtls_srtp.emplace(true);
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 invalid_answer = callee->CreateAnswer();
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_PRED_FORMAT2(AssertStringContains, error,
"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_PRED_FORMAT2(AssertStringContains, error,
"Session error code: ERROR_CONTENT.");
ASSERT_FALSE(callee->SetLocalDescription(callee->CreateAnswer(), &error));
EXPECT_PRED_FORMAT2(AssertStringContains, error,
"Session error code: ERROR_CONTENT.");
}
INSTANTIATE_TEST_CASE_P(PeerConnectionCryptoTest,
PeerConnectionCryptoTest,
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
} // namespace webrtc