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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 <memory>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/scoped_refptr.h"
#include "api/sctp_transport_interface.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/sctp_transport_factory_interface.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port_allocator.h"
#include "pc/media_session.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_proxy.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/sctp_transport.h"
#include "pc/sdp_utils.h"
#include "pc/session_description.h"
#include "pc/test/enable_fake_media.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/android_test_initializer.h"
#endif
#include "rtc_base/virtual_socket_server.h"
#include "test/pc/sctp/fake_sctp_transport.h"
namespace webrtc {
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
using ::testing::HasSubstr;
using ::testing::Not;
using ::testing::Values;
namespace {
PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies() {
PeerConnectionFactoryDependencies deps;
deps.network_thread = rtc::Thread::Current();
deps.worker_thread = rtc::Thread::Current();
deps.signaling_thread = rtc::Thread::Current();
deps.task_queue_factory = CreateDefaultTaskQueueFactory();
EnableFakeMedia(deps);
deps.sctp_factory = std::make_unique<FakeSctpTransportFactory>();
return deps;
}
} // namespace
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;
}
absl::optional<std::string> sctp_mid() {
return GetInternalPeerConnection()->sctp_mid();
}
absl::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 PeerConnectionDataChannelBaseTest : public ::testing::Test {
protected:
typedef std::unique_ptr<PeerConnectionWrapperForDataChannelTest> WrapperPtr;
explicit PeerConnectionDataChannelBaseTest(SdpSemantics sdp_semantics)
: vss_(new rtc::VirtualSocketServer()),
main_(vss_.get()),
sdp_semantics_(sdp_semantics) {
#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) {
auto factory_deps = CreatePeerConnectionFactoryDependencies();
FakeSctpTransportFactory* fake_sctp_transport_factory =
static_cast<FakeSctpTransportFactory*>(factory_deps.sctp_factory.get());
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory =
CreateModularPeerConnectionFactory(std::move(factory_deps));
pc_factory->SetOptions(factory_options);
auto observer = std::make_unique<MockPeerConnectionObserver>();
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
auto result = pc_factory->CreatePeerConnectionOrError(
modified_config, PeerConnectionDependencies(observer.get()));
if (!result.ok()) {
return nullptr;
}
observer->SetPeerConnectionInterface(result.value().get());
auto wrapper = std::make_unique<PeerConnectionWrapperForDataChannelTest>(
pc_factory, result.MoveValue(), std::move(observer));
wrapper->set_sctp_transport_factory(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()->CreateDataChannelOrError("dc", nullptr).ok());
return wrapper;
}
// Changes the SCTP data channel port on the given session description.
void ChangeSctpPortOnDescription(cricket::SessionDescription* desc,
int port) {
auto* data_content = cricket::GetFirstDataContent(desc);
RTC_DCHECK(data_content);
auto* data_desc = data_content->media_description()->as_sctp();
RTC_DCHECK(data_desc);
data_desc->set_port(port);
}
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
const SdpSemantics sdp_semantics_;
};
class PeerConnectionDataChannelTest
: public PeerConnectionDataChannelBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionDataChannelTest()
: PeerConnectionDataChannelBaseTest(GetParam()) {}
};
class PeerConnectionDataChannelUnifiedPlanTest
: public PeerConnectionDataChannelBaseTest {
protected:
PeerConnectionDataChannelUnifiedPlanTest()
: PeerConnectionDataChannelBaseTest(SdpSemantics::kUnifiedPlan) {}
};
TEST_P(PeerConnectionDataChannelTest, InternalSctpTransportDeletedOnTeardown) {
auto caller = CreatePeerConnectionWithDataChannel();
ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
EXPECT_TRUE(caller->sctp_transport_factory()->last_fake_sctp_transport());
rtc::scoped_refptr<SctpTransportInterface> sctp_transport =
caller->GetInternalPeerConnection()->GetSctpTransport();
caller.reset();
EXPECT_EQ(static_cast<SctpTransport*>(sctp_transport.get())->internal(),
nullptr);
}
// Test that sctp_mid/sctp_transport_name (used for stats) are correct
// before and after BUNDLE is negotiated.
TEST_P(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
// Initially these fields should be empty.
EXPECT_FALSE(caller->sctp_mid());
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()->CreateDataChannelOrError("dc", nullptr);
auto offer = caller->CreateOffer();
const auto& offer_contents = offer->description()->contents();
ASSERT_EQ(cricket::MEDIA_TYPE_AUDIO,
offer_contents[0].media_description()->type());
std::string audio_mid = offer_contents[0].name;
ASSERT_EQ(cricket::MEDIA_TYPE_DATA,
offer_contents[2].media_description()->type());
std::string data_mid = offer_contents[2].name;
ASSERT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
ASSERT_TRUE(caller->sctp_mid());
EXPECT_EQ(data_mid, *caller->sctp_mid());
ASSERT_TRUE(caller->sctp_transport_name());
EXPECT_EQ(data_mid, *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_mid());
EXPECT_EQ(data_mid, *caller->sctp_mid());
ASSERT_TRUE(caller->sctp_transport_name());
EXPECT_EQ(audio_mid, *caller->sctp_transport_name());
}
TEST_P(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_P(PeerConnectionDataChannelTest,
CreateAnswerWithRemoteSctpDataChannelIncludesDataSection) {
auto caller = CreatePeerConnectionWithDataChannel();
auto callee = CreatePeerConnection();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
auto answer = callee->CreateAnswer();
ASSERT_TRUE(answer);
auto* data_content = cricket::GetFirstDataContent(answer->description());
ASSERT_TRUE(data_content);
EXPECT_FALSE(data_content->rejected);
EXPECT_TRUE(
answer->description()->GetTransportInfoByName(data_content->name));
}
TEST_P(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);
std::string sdp;
answer->ToString(&sdp);
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());
}
TEST_P(PeerConnectionDataChannelTest, ModernSdpSyntaxByDefault) {
PeerConnectionInterface::RTCOfferAnswerOptions options;
auto caller = CreatePeerConnectionWithDataChannel();
auto offer = caller->CreateOffer(options);
EXPECT_FALSE(cricket::GetFirstSctpDataContentDescription(offer->description())
->use_sctpmap());
std::string sdp;
offer->ToString(&sdp);
RTC_LOG(LS_ERROR) << sdp;
EXPECT_THAT(sdp, HasSubstr(" UDP/DTLS/SCTP webrtc-datachannel"));
EXPECT_THAT(sdp, Not(HasSubstr("a=sctpmap:")));
}
TEST_P(PeerConnectionDataChannelTest, ObsoleteSdpSyntaxIfSet) {
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.use_obsolete_sctp_sdp = true;
auto caller = CreatePeerConnectionWithDataChannel();
auto offer = caller->CreateOffer(options);
EXPECT_TRUE(cricket::GetFirstSctpDataContentDescription(offer->description())
->use_sctpmap());
std::string sdp;
offer->ToString(&sdp);
EXPECT_THAT(sdp, Not(HasSubstr(" UDP/DTLS/SCTP webrtc-datachannel")));
EXPECT_THAT(sdp, HasSubstr("a=sctpmap:"));
}
INSTANTIATE_TEST_SUITE_P(PeerConnectionDataChannelTest,
PeerConnectionDataChannelTest,
Values(SdpSemantics::kPlanB_DEPRECATED,
SdpSemantics::kUnifiedPlan));
} // namespace webrtc