admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Remove enable_dtls_srtp option

Browse Source 
This is part of the removal of support for SDES.

Bug: webrtc:11066
Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35262}
This commit is contained in:
Harald Alvestrand 2021-10-13 15:26:26 +00:00 committed by WebRTC LUCI CQ
parent aaa848e078
commit f9e502d935
19 changed files with 36 additions and 299 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
api/peer_connection_interface.h
View File

@ -427,12 +427,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// Use new combined audio/video bandwidth estimation?
absl::optional<bool> combined_audio_video_bwe;
// TODO(bugs.webrtc.org/9891) - Move to crypto_options
// Can be used to disable DTLS-SRTP. This should never be done, but can be
// useful for testing purposes, for example in setting up a loopback call
// with a single PeerConnection.
absl::optional<bool> enable_dtls_srtp;
/////////////////////////////////////////////////
// The below fields are not part of the standard.
/////////////////////////////////////////////////

6
examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
View File

@ -442,6 +442,10 @@ public class PeerConnectionClient {
decoderFactory = new SoftwareVideoDecoderFactory();
}
// Disable encryption for loopback calls.
if (peerConnectionParameters.loopback) {
options.disableEncryption = true;
}
factory = PeerConnectionFactory.builder()
.setOptions(options)
.setAudioDeviceModule(adm)
@ -600,8 +604,6 @@ public class PeerConnectionClient {
rtcConfig.continualGatheringPolicy = PeerConnection.ContinualGatheringPolicy.GATHER_CONTINUALLY;
// Use ECDSA encryption.
rtcConfig.keyType = PeerConnection.KeyType.ECDSA;
// Enable DTLS for normal calls and disable for loopback calls.
rtcConfig.enableDtlsSrtp = !peerConnectionParameters.loopback;
rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN;
peerConnection = factory.createPeerConnection(rtcConfig, pcObserver);

6
examples/androidnativeapi/jni/android_call_client.cc
View File

@ -177,8 +177,10 @@ void AndroidCallClient::CreatePeerConnection() {
webrtc::MutexLock lock(&pc_mutex_);
webrtc::PeerConnectionInterface::RTCConfiguration config;
config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
// DTLS SRTP has to be disabled for loopback to work.
config.enable_dtls_srtp = false;
// Encryption has to be disabled for loopback to work.
webrtc::PeerConnectionFactoryInterface::Options options;
options.disable_encryption = true;
pcf_->SetOptions(options);
webrtc::PeerConnectionDependencies deps(pc_observer_.get());
pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(deps)).MoveValue();

6
examples/objcnativeapi/objc/objc_call_client.mm
View File

@ -141,8 +141,10 @@ void ObjCCallClient::CreatePeerConnection() {
webrtc::MutexLock lock(&pc_mutex_);
webrtc::PeerConnectionInterface::RTCConfiguration config;
config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
// DTLS SRTP has to be disabled for loopback to work.
config.enable_dtls_srtp = false;
// Encryption has to be disabled for loopback to work.
webrtc::PeerConnectionFactoryInterface::Options options;
options.disable_encryption = true;
pcf_->SetOptions(options);
webrtc::PeerConnectionDependencies pc_dependencies(pc_observer_.get());
pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)).MoveValue();
RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_;

13
examples/peerconnection/client/conductor.cc
View File

@ -150,7 +150,7 @@ bool Conductor::InitializePeerConnection() {
return false;
}
if (!CreatePeerConnection(/*dtls=*/true)) {
if (!CreatePeerConnection()) {
main_wnd_->MessageBox("Error", "CreatePeerConnection failed", true);
DeletePeerConnection();
}
@ -165,23 +165,28 @@ bool Conductor::ReinitializePeerConnectionForLoopback() {
std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> senders =
peer_connection_->GetSenders();
peer_connection_ = nullptr;
if (CreatePeerConnection(/*dtls=*/false)) {
// Loopback is only possible if encryption is disabled.
webrtc::PeerConnectionFactoryInterface::Options options;
options.disable_encryption = true;
peer_connection_factory_->SetOptions(options);
if (CreatePeerConnection()) {
for (const auto& sender : senders) {
peer_connection_->AddTrack(sender->track(), sender->stream_ids());
}
peer_connection_->CreateOffer(
this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
}
options.disable_encryption = false;
peer_connection_factory_->SetOptions(options);
return peer_connection_ != nullptr;
}
bool Conductor::CreatePeerConnection(bool dtls) {
bool Conductor::CreatePeerConnection() {
RTC_DCHECK(peer_connection_factory_);
RTC_DCHECK(!peer_connection_);
webrtc::PeerConnectionInterface::RTCConfiguration config;
config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
config.enable_dtls_srtp = dtls;
webrtc::PeerConnectionInterface::IceServer server;
server.uri = GetPeerConnectionString();
config.servers.push_back(server);

2
examples/peerconnection/client/conductor.h
View File

@ -54,7 +54,7 @@ class Conductor : public webrtc::PeerConnectionObserver,
~Conductor();
bool InitializePeerConnection();
bool ReinitializePeerConnectionForLoopback();
bool CreatePeerConnection(bool dtls);
bool CreatePeerConnection();
void DeletePeerConnection();
void EnsureStreamingUI();
void AddTracks();

1
examples/unityplugin/simple_peer_connection.cc
View File

@ -190,7 +190,6 @@ bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
webrtc::PeerConnectionInterface::IceServer stun_server;
stun_server.uri = GetPeerConnectionString();
config_.servers.push_back(stun_server);
config_.enable_dtls_srtp = false;
auto result = g_peer_connection_factory->CreatePeerConnectionOrError(
config_, webrtc::PeerConnectionDependencies(this));

6
pc/peer_connection.cc
View File

@ -276,8 +276,8 @@ bool DtlsEnabled(const PeerConnectionInterface::RTCConfiguration& configuration,
bool default_enabled =
(dependencies.cert_generator || !configuration.certificates.empty());
// The `configuration` can override the default value.
return configuration.enable_dtls_srtp.value_or(default_enabled);
RTC_DCHECK(default_enabled) << "Configuration error: No certs for DTLS";
return default_enabled;
}
} // namespace
@ -300,7 +300,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
bool enable_rtp_data_channel;
absl::optional<int> screencast_min_bitrate;
absl::optional<bool> combined_audio_video_bwe;
absl::optional<bool> enable_dtls_srtp;
TcpCandidatePolicy tcp_candidate_policy;
CandidateNetworkPolicy candidate_network_policy;
int audio_jitter_buffer_max_packets;
@ -368,7 +367,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
disable_link_local_networks == o.disable_link_local_networks &&
screencast_min_bitrate == o.screencast_min_bitrate &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
enable_dtls_srtp == o.enable_dtls_srtp &&
ice_candidate_pool_size == o.ice_candidate_pool_size &&
prune_turn_ports == o.prune_turn_ports &&
turn_port_prune_policy == o.turn_port_prune_policy &&

180
pc/peer_connection_crypto_unittest.cc
View File

@ -181,7 +181,6 @@ SdpContentMutator RemoveDtlsFingerprint() {
// no SDES cryptos.
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsEnabled) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
@ -195,7 +194,6 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsEnabled) {
}
TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsEnabled) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
@ -210,39 +208,6 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsEnabled) {
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) {
@ -251,7 +216,6 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenEncryptionDisabled) {
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
@ -269,7 +233,6 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenEncryptionDisabled) {
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
@ -284,80 +247,12 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenEncryptionDisabled) {
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;
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;
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();
for (cricket::ContentInfo& content : offer->description()->contents()) {
auto cryptos = content.media_description()->cryptos();
cryptos.erase(cryptos.begin()); // Assumes that non-GCM is the default.
content.media_description()->set_cryptos(cryptos);
}
callee->SetRemoteDescription(std::move(offer));
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;
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);
@ -368,69 +263,8 @@ TEST_P(PeerConnectionCryptoTest, CanSetSdesGcmRemoteOfferAndLocalAnswer) {
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)));
// Note - this test doesn't verify that Gcm is present, just that it
// does not caue a failure.
}
// The following group tests that two PeerConnections can successfully exchange
@ -438,7 +272,6 @@ TEST_P(PeerConnectionCryptoTest, FailToSetRemoteAnswerWithNoCryptosWhenSdesOn) {
// 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);
@ -453,7 +286,6 @@ TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenDtlsOn) {
TEST_P(PeerConnectionCryptoTest,
FailToSetLocalOfferWithNoFingerprintWhenDtlsOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto offer = caller->CreateOffer();
@ -464,7 +296,6 @@ TEST_P(PeerConnectionCryptoTest,
TEST_P(PeerConnectionCryptoTest,
FailToSetRemoteOfferWithNoFingerprintWhenDtlsOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
@ -476,7 +307,6 @@ TEST_P(PeerConnectionCryptoTest,
TEST_P(PeerConnectionCryptoTest,
FailToSetLocalAnswerWithNoFingerprintWhenDtlsOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
@ -487,7 +317,6 @@ TEST_P(PeerConnectionCryptoTest,
TEST_P(PeerConnectionCryptoTest,
FailToSetRemoteAnswerWithNoFingerprintWhenDtlsOn) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
@ -505,7 +334,6 @@ TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenNoEncryption) {
pc_factory_->SetOptions(options);
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
@ -523,13 +351,11 @@ TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenNoEncryption) {
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);
@ -600,7 +426,6 @@ class PeerConnectionCryptoDtlsCertGenTest
TEST_P(PeerConnectionCryptoDtlsCertGenTest, TestCertificateGeneration) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(true);
auto owned_fake_certificate_generator =
std::make_unique<FakeRTCCertificateGenerator>();
auto* fake_certificate_generator = owned_fake_certificate_generator.get();
@ -724,7 +549,6 @@ TEST_P(PeerConnectionCryptoTest, SessionErrorIfFingerprintInvalid) {
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);

9
pc/peer_connection_data_channel_unittest.cc
View File

@ -280,15 +280,6 @@ TEST_P(PeerConnectionDataChannelTest,
answer->description()->GetTransportInfoByName(data_content->name));
}
TEST_P(PeerConnectionDataChannelTest,
CreateDataChannelWithDtlsDisabledSucceeds) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnection();
EXPECT_TRUE(caller->pc()->CreateDataChannel("dc", nullptr));
}
TEST_P(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
constexpr int kNewSendPort = 9998;
constexpr int kNewRecvPort = 7775;

9
pc/peer_connection_end_to_end_unittest.cc
View File

@ -366,15 +366,6 @@ TEST_P(PeerConnectionEndToEndTest, Call) {
WaitForCallEstablished();
}
TEST_P(PeerConnectionEndToEndTest, CallWithSdesKeyNegotiation) {
config_.enable_dtls_srtp = false;
CreatePcs(webrtc::CreateOpusAudioEncoderFactory(),
webrtc::CreateOpusAudioDecoderFactory());
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
class IdLoggingAudioEncoderFactory : public webrtc::AudioEncoderFactory {
public:

24
pc/peer_connection_integrationtest.cc
View File

@ -264,30 +264,6 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
webrtc::kEnumCounterKeyProtocolSdes));
}
// Uses SDES instead of DTLS for key agreement.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
PeerConnectionInterface::RTCConfiguration sdes_config;
sdes_config.enable_dtls_srtp.emplace(false);
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config));
ConnectFakeSignaling();
// Do normal offer/answer and wait for some frames to be received in each
// direction.
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
EXPECT_METRIC_LE(
2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_METRIC_EQ(
0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
}
// Basic end-to-end test specifying the `enable_encrypted_rtp_header_extensions`
// option to offer encrypted versions of all header extensions alongside the
// unencrypted versions.

45
pc/peer_connection_interface_unittest.cc
View File

@ -710,13 +710,16 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration());
}
// DTLS does not work in a loopback call, so is disabled for most of the
// DTLS does not work in a loopback call, so is disabled for many
// tests in this file.
void CreatePeerConnectionWithoutDtls() {
RTCConfiguration config;
config.enable_dtls_srtp = false;
PeerConnectionFactoryInterface::Options options;
options.disable_encryption = true;
pc_factory_->SetOptions(options);
CreatePeerConnection(config);
options.disable_encryption = false;
pc_factory_->SetOptions(options);
}
void CreatePeerConnectionWithIceTransportsType(
@ -751,10 +754,10 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
// false.
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
if (config.enable_dtls_srtp.value_or(true)) {
fake_certificate_generator_ = new FakeRTCCertificateGenerator();
cert_generator.reset(fake_certificate_generator_);
}
// These won't be used if encryption is turned off, but that's harmless.
fake_certificate_generator_ = new FakeRTCCertificateGenerator();
cert_generator.reset(fake_certificate_generator_);
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
pc_ = pc_factory_->CreatePeerConnection(
@ -1903,7 +1906,6 @@ TEST_P(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
// DataChannelInit configurations.
TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
webrtc::DataChannelInit config;
@ -1939,7 +1941,6 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
// and maxRetransmitTime by setting them to -1 to get what they want.
TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWithMinusOne) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
webrtc::DataChannelInit config;
@ -1955,7 +1956,6 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWithMinusOne) {
TEST_P(PeerConnectionInterfaceTest,
CreateSctpDataChannelShouldFailForInvalidConfig) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
std::string label = "test";
@ -1973,7 +1973,6 @@ TEST_P(PeerConnectionInterfaceTest,
TEST_P(PeerConnectionInterfaceTest,
CreateSctpDataChannelWithInvalidIdShouldFail) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
webrtc::DataChannelInit config;
@ -2003,7 +2002,6 @@ TEST_P(PeerConnectionInterfaceTest,
// Verifies that duplicated label is allowed for SCTP data channel.
TEST_P(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
std::string label = "test";
@ -2051,7 +2049,6 @@ TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer)
// the answer as a local description.
TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
@ -2085,7 +2082,6 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
// and because it's non-standard.
TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
// Wait for fake certificate to be generated. Previously, this is what caused
// the "a=crypto" lines to be rejected.
@ -2129,7 +2125,6 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
// will have m-lines with a=recvonly.
TEST_P(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
CreateAnswerAsLocalDescription();
@ -2155,7 +2150,6 @@ TEST_P(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
// false, the generated m-lines will be a=inactive.
TEST_P(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
CreateAnswerAsLocalDescription();
@ -2508,7 +2502,6 @@ TEST_P(PeerConnectionInterfaceTest, CloseAndGetStats) {
// signaled.
TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
@ -2534,7 +2527,6 @@ TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
TEST_F(PeerConnectionInterfaceTestPlanB,
AddRemoveTrackFromExistingRemoteMediaStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
std::unique_ptr<SessionDescriptionInterface> desc_ms1 =
CreateSessionDescriptionAndReference(1, 1);
@ -2577,7 +2569,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// that rejects the media content type.
TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
// First create and set a remote offer, then reject its video content in our
// answer.
@ -2626,7 +2617,6 @@ TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) {
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
@ -2652,7 +2642,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
// See: https://code.google.com/p/webrtc/issues/detail?id=5054
TEST_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
std::string recvonly_offer = GetSdpStringWithStream1();
@ -2669,7 +2658,6 @@ TEST_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithoutMsidCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
@ -2698,7 +2686,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithoutMsidCreatesDefaultStream) {
TEST_F(PeerConnectionInterfaceTestPlanB,
SendOnlySdpWithoutMsidCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringSendOnlyWithoutStreams);
@ -2715,7 +2702,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveAlreadyGoneRemoteStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
@ -2734,7 +2720,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveAlreadyGoneRemoteStream) {
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithoutMsidAndStreamsCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
@ -2749,7 +2734,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithMsidDontCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithMsidWithoutStreams);
EXPECT_EQ(0u, observer_.remote_streams()->count());
@ -2762,7 +2746,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithMsidDontCreatesDefaultStream) {
TEST_F(PeerConnectionInterfaceTestPlanB,
DefaultTracksNotDestroyedAndRecreated) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
@ -2783,7 +2766,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, VerifyDefaultStreamIsNotCreated) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
@ -2799,7 +2781,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB, VerifyDefaultStreamIsNotCreated) {
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithMsidWithoutSsrcCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
std::string sdp_string = kSdpStringWithoutStreamsAudioOnly;
// Add a=msid lines to simulate a Unified Plan endpoint that only
@ -2822,7 +2803,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithEmptyMsidAndSsrcCreatesDefaultStreamId) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
// Add a a=msid line to the SDP. This is prioritized when parsing the SDP, so
// the sender's stream ID will be interpreted as no stream IDs.
@ -2860,7 +2840,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, LocalDescriptionChanged) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
// Create an offer with 1 stream with 2 tracks of each type.
@ -2899,7 +2878,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB, LocalDescriptionChanged) {
TEST_F(PeerConnectionInterfaceTestPlanB,
AddLocalStreamAfterLocalDescriptionChanged) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
rtc::scoped_refptr<StreamCollection> stream_collection =
@ -2928,7 +2906,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
TEST_P(PeerConnectionInterfaceTest,
ChangeSsrcOnTrackInLocalSessionDescription) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
AddAudioTrack(kAudioTracks[0]);
@ -2981,7 +2958,6 @@ TEST_P(PeerConnectionInterfaceTest,
TEST_F(PeerConnectionInterfaceTestPlanB,
SignalSameTracksInSeparateMediaStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
rtc::scoped_refptr<StreamCollection> stream_collection =
@ -3018,7 +2994,6 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// This tests that PeerConnectionObserver::OnAddTrack is correctly called.
TEST_P(PeerConnectionInterfaceTest, OnAddTrackCallback) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithStream1AudioTrackOnly);
EXPECT_EQ(observer_.num_added_tracks_, 1);

8
sdk/android/api/org/webrtc/PeerConnection.java
View File

@ -527,7 +527,6 @@ public class PeerConnection {
public boolean suspendBelowMinBitrate;
@Nullable public Integer screencastMinBitrate;
@Nullable public Boolean combinedAudioVideoBwe;
@Nullable public Boolean enableDtlsSrtp;
// Use "Unknown" to represent no preference of adapter types, not the
// preference of adapters of unknown types.
public AdapterType networkPreference;
@ -608,7 +607,6 @@ public class PeerConnection {
suspendBelowMinBitrate = false;
screencastMinBitrate = null;
combinedAudioVideoBwe = null;
enableDtlsSrtp = null;
networkPreference = AdapterType.UNKNOWN;
sdpSemantics = SdpSemantics.PLAN_B;
activeResetSrtpParams = false;
@ -800,12 +798,6 @@ public class PeerConnection {
return combinedAudioVideoBwe;
}
@Nullable
@CalledByNative("RTCConfiguration")
Boolean getEnableDtlsSrtp() {
return enableDtlsSrtp;
}
@CalledByNative("RTCConfiguration")
AdapterType getNetworkPreference() {
return networkPreference;

3
sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java
View File

@ -677,7 +677,6 @@ public class PeerConnectionEndToEndTest {
.createIceServer());
PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers);
rtcConfig.enableDtlsSrtp = true;
ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
PeerConnection offeringPC = factory.createPeerConnection(rtcConfig, offeringExpectations);
@ -944,7 +943,6 @@ public class PeerConnectionEndToEndTest {
.createIceServer());
PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers);
rtcConfig.enableDtlsSrtp = true;
ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
PeerConnection offeringPC = factory.createPeerConnection(rtcConfig, offeringExpectations);
@ -1172,7 +1170,6 @@ public class PeerConnectionEndToEndTest {
PeerConnection.IceServer.builder("stun:stun.l.google.com:19302").createIceServer());
PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers);
rtcConfig.enableDtlsSrtp = true;
ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
PeerConnection offeringPC = factory.createPeerConnection(rtcConfig, offeringExpectations);

2
sdk/android/src/jni/pc/peer_connection.cc
View File

@ -263,8 +263,6 @@ void JavaToNativeRTCConfiguration(
jni, Java_RTCConfiguration_getScreencastMinBitrate(jni, j_rtc_config));
rtc_config->combined_audio_video_bwe = JavaToNativeOptionalBool(
jni, Java_RTCConfiguration_getCombinedAudioVideoBwe(jni, j_rtc_config));
rtc_config->enable_dtls_srtp = JavaToNativeOptionalBool(
jni, Java_RTCConfiguration_getEnableDtlsSrtp(jni, j_rtc_config));
rtc_config->network_preference =
JavaToNativeNetworkPreference(jni, j_network_preference);
rtc_config->sdp_semantics = JavaToNativeSdpSemantics(jni, j_sdp_semantics);

3
sdk/media_constraints.cc
View File

@ -117,7 +117,6 @@ const char MediaConstraints::kIceRestart[] = "IceRestart";
const char MediaConstraints::kUseRtpMux[] = "googUseRtpMUX";
// Below constraints should be used during PeerConnection construction.
const char MediaConstraints::kEnableDtlsSrtp[] = "DtlsSrtpKeyAgreement";
// Google-specific constraint keys.
const char MediaConstraints::kEnableDscp[] = "googDscp";
const char MediaConstraints::kEnableIPv6[] = "googIPv6";
@ -176,8 +175,6 @@ void CopyConstraintsIntoRtcConfiguration(
ConstraintToOptional<bool>(constraints,
MediaConstraints::kCombinedAudioVideoBwe,
&configuration->combined_audio_video_bwe);
ConstraintToOptional<bool>(constraints, MediaConstraints::kEnableDtlsSrtp,
&configuration->enable_dtls_srtp);
}
void CopyConstraintsIntoAudioOptions(const MediaConstraints* constraints,

2
sdk/media_constraints.h
View File

@ -83,8 +83,6 @@ class MediaConstraints {
static const char kValueFalse[]; // false
// PeerConnection constraint keys.
// Temporary pseudo-constraints used to enable DTLS-SRTP
static const char kEnableDtlsSrtp[]; // Enable DTLS-SRTP
// Google-specific constraint keys.
// Temporary pseudo-constraint for enabling DSCP through JS.
static const char kEnableDscp[]; // googDscp

4
sdk/media_constraints_unittest.cc
View File

@ -25,7 +25,6 @@ bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
b.audio_jitter_buffer_max_packets &&
a.screencast_min_bitrate == b.screencast_min_bitrate &&
a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
a.enable_dtls_srtp == b.enable_dtls_srtp &&
a.media_config == b.media_config;
}
@ -60,12 +59,9 @@ TEST(MediaConstraints, CopyConstraintsIntoRtcConfiguration) {
// An empty set of constraints will not overwrite
// values that are already present.
configuration = old_configuration;
configuration.enable_dtls_srtp = true;
configuration.audio_jitter_buffer_max_packets = 34;
CopyConstraintsIntoRtcConfiguration(&constraints_empty, &configuration);
EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets);
ASSERT_TRUE(configuration.enable_dtls_srtp);
EXPECT_TRUE(*(configuration.enable_dtls_srtp));
}
} // namespace