blob: 02bac685b412eb3d568b763bf76c3d755dad4c88 [file] [log] [blame]
/*
* 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 <tuple>
#include "api/peerconnectionproxy.h"
#include "media/base/fakemediaengine.h"
#include "pc/mediasession.h"
#include "pc/peerconnection.h"
#include "pc/peerconnectionfactory.h"
#include "pc/peerconnectionwrapper.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
#include "pc/test/fakesctptransport.h"
#include "rtc_base/gunit.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/virtualsocketserver.h"
namespace webrtc {
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
using ::testing::Values;
class PeerConnectionFactoryForDataChannelTest
: public rtc::RefCountedObject<PeerConnectionFactory> {
public:
PeerConnectionFactoryForDataChannelTest()
: rtc::RefCountedObject<PeerConnectionFactory>(
rtc::Thread::Current(),
rtc::Thread::Current(),
rtc::Thread::Current(),
rtc::MakeUnique<cricket::FakeMediaEngine>(),
CreateCallFactory(),
nullptr) {}
std::unique_ptr<cricket::SctpTransportInternalFactory>
CreateSctpTransportInternalFactory() {
auto factory = rtc::MakeUnique<FakeSctpTransportFactory>();
last_fake_sctp_transport_factory_ = factory.get();
return factory;
}
FakeSctpTransportFactory* last_fake_sctp_transport_factory_ = nullptr;
};
class PeerConnectionWrapperForDataChannelTest : public PeerConnectionWrapper {
public:
using PeerConnectionWrapper::PeerConnectionWrapper;
FakeSctpTransportFactory* sctp_transport_factory() {
return sctp_transport_factory_;
}
void set_sctp_transport_factory(
FakeSctpTransportFactory* sctp_transport_factory) {
sctp_transport_factory_ = sctp_transport_factory;
}
rtc::Optional<std::string> sctp_content_name() {
return GetInternalPeerConnection()->sctp_content_name();
}
rtc::Optional<std::string> sctp_transport_name() {
return GetInternalPeerConnection()->sctp_transport_name();
}
PeerConnection* GetInternalPeerConnection() {
auto* pci =
static_cast<PeerConnectionProxyWithInternal<PeerConnectionInterface>*>(
pc());
return static_cast<PeerConnection*>(pci->internal());
}
private:
FakeSctpTransportFactory* sctp_transport_factory_ = nullptr;
};
class PeerConnectionDataChannelTest : public ::testing::Test {
protected:
typedef std::unique_ptr<PeerConnectionWrapperForDataChannelTest> WrapperPtr;
PeerConnectionDataChannelTest()
: vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
#ifdef WEBRTC_ANDROID
InitializeAndroidObjects();
#endif
}
WrapperPtr CreatePeerConnection() {
return CreatePeerConnection(RTCConfiguration());
}
WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
return CreatePeerConnection(config,
PeerConnectionFactoryInterface::Options());
}
WrapperPtr CreatePeerConnection(
const RTCConfiguration& config,
const PeerConnectionFactoryInterface::Options factory_options) {
rtc::scoped_refptr<PeerConnectionFactoryForDataChannelTest> pc_factory(
new PeerConnectionFactoryForDataChannelTest());
pc_factory->SetOptions(factory_options);
RTC_CHECK(pc_factory->Initialize());
auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
auto pc = pc_factory->CreatePeerConnection(config, nullptr, nullptr,
observer.get());
if (!pc) {
return nullptr;
}
auto wrapper = rtc::MakeUnique<PeerConnectionWrapperForDataChannelTest>(
pc_factory, pc, std::move(observer));
RTC_DCHECK(pc_factory->last_fake_sctp_transport_factory_);
wrapper->set_sctp_transport_factory(
pc_factory->last_fake_sctp_transport_factory_);
return wrapper;
}
// Accepts the same arguments as CreatePeerConnection and adds a default data
// channel.
template <typename... Args>
WrapperPtr CreatePeerConnectionWithDataChannel(Args&&... args) {
auto wrapper = CreatePeerConnection(std::forward<Args>(args)...);
if (!wrapper) {
return nullptr;
}
EXPECT_TRUE(wrapper->pc()->CreateDataChannel("dc", nullptr));
return wrapper;
}
// Changes the SCTP data channel port on the given session description.
void ChangeSctpPortOnDescription(cricket::SessionDescription* desc,
int port) {
cricket::DataCodec sctp_codec(cricket::kGoogleSctpDataCodecPlType,
cricket::kGoogleSctpDataCodecName);
sctp_codec.SetParam(cricket::kCodecParamPort, port);
auto* data_content = cricket::GetFirstDataContent(desc);
RTC_DCHECK(data_content);
auto* data_desc = static_cast<cricket::DataContentDescription*>(
data_content->description);
data_desc->set_codecs({sctp_codec});
}
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
};
TEST_F(PeerConnectionDataChannelTest,
NoSctpTransportCreatedIfRtpDataChannelEnabled) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
auto caller = CreatePeerConnectionWithDataChannel(config);
ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
}
TEST_F(PeerConnectionDataChannelTest,
RtpDataChannelCreatedEvenIfSctpAvailable) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
PeerConnectionFactoryInterface::Options options;
options.disable_sctp_data_channels = false;
auto caller = CreatePeerConnectionWithDataChannel(config, options);
ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
}
// Test that sctp_content_name/sctp_transport_name (used for stats) are correct
// before and after BUNDLE is negotiated.
TEST_F(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
// Initially these fields should be empty.
EXPECT_FALSE(caller->sctp_content_name());
EXPECT_FALSE(caller->sctp_transport_name());
// Create offer with audio/video/data.
// Default bundle policy is "balanced", so data should be using its own
// transport.
caller->AddAudioTrack("a");
caller->AddVideoTrack("v");
caller->pc()->CreateDataChannel("dc", nullptr);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
ASSERT_TRUE(caller->sctp_content_name());
EXPECT_EQ(cricket::CN_DATA, *caller->sctp_content_name());
ASSERT_TRUE(caller->sctp_transport_name());
EXPECT_EQ(cricket::CN_DATA, *caller->sctp_transport_name());
// Create answer that finishes BUNDLE negotiation, which means everything
// should be bundled on the first transport (audio).
RTCOfferAnswerOptions options;
options.use_rtp_mux = true;
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
ASSERT_TRUE(caller->sctp_content_name());
EXPECT_EQ(cricket::CN_DATA, *caller->sctp_content_name());
ASSERT_TRUE(caller->sctp_transport_name());
EXPECT_EQ(cricket::CN_AUDIO, *caller->sctp_transport_name());
}
TEST_F(PeerConnectionDataChannelTest,
CreateOfferWithNoDataChannelsGivesNoDataSection) {
auto caller = CreatePeerConnection();
auto offer = caller->CreateOffer();
EXPECT_FALSE(offer->description()->GetContentByName(cricket::CN_DATA));
EXPECT_FALSE(offer->description()->GetTransportInfoByName(cricket::CN_DATA));
}
TEST_F(PeerConnectionDataChannelTest,
CreateAnswerWithRemoteSctpDataChannelIncludesDataSection) {
auto caller = CreatePeerConnectionWithDataChannel();
auto callee = CreatePeerConnection();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
auto answer = callee->CreateAnswer();
ASSERT_TRUE(answer);
auto* data_content =
answer->description()->GetContentByName(cricket::CN_DATA);
ASSERT_TRUE(data_content);
EXPECT_FALSE(data_content->rejected);
EXPECT_TRUE(answer->description()->GetTransportInfoByName(cricket::CN_DATA));
}
TEST_F(PeerConnectionDataChannelTest,
CreateDataChannelWithDtlsDisabledSucceeds) {
RTCConfiguration config;
config.enable_dtls_srtp.emplace(false);
auto caller = CreatePeerConnection();
EXPECT_TRUE(caller->pc()->CreateDataChannel("dc", nullptr));
}
TEST_F(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) {
PeerConnectionFactoryInterface::Options options;
options.disable_sctp_data_channels = true;
auto caller = CreatePeerConnection(RTCConfiguration(), options);
EXPECT_FALSE(caller->pc()->CreateDataChannel("dc", nullptr));
}
// Test that if a callee has SCTP disabled and receives an offer with an SCTP
// data channel, the data section is rejected and no SCTP transport is created
// on the callee.
TEST_F(PeerConnectionDataChannelTest,
DataSectionRejectedIfCalleeHasSctpDisabled) {
auto caller = CreatePeerConnectionWithDataChannel();
PeerConnectionFactoryInterface::Options options;
options.disable_sctp_data_channels = true;
auto callee = CreatePeerConnection(RTCConfiguration(), options);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
EXPECT_FALSE(callee->sctp_transport_factory()->last_fake_sctp_transport());
auto answer = callee->CreateAnswer();
auto* data_content =
answer->description()->GetContentByName(cricket::CN_DATA);
ASSERT_TRUE(data_content);
EXPECT_TRUE(data_content->rejected);
}
TEST_F(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
constexpr int kNewSendPort = 9998;
constexpr int kNewRecvPort = 7775;
auto caller = CreatePeerConnectionWithDataChannel();
auto callee = CreatePeerConnectionWithDataChannel();
auto offer = caller->CreateOffer();
ChangeSctpPortOnDescription(offer->description(), kNewSendPort);
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswer();
ChangeSctpPortOnDescription(answer->description(), kNewRecvPort);
ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
auto* callee_transport =
callee->sctp_transport_factory()->last_fake_sctp_transport();
ASSERT_TRUE(callee_transport);
EXPECT_EQ(kNewSendPort, callee_transport->remote_port());
EXPECT_EQ(kNewRecvPort, callee_transport->local_port());
}
} // namespace webrtc