webrtc_m130/webrtc/api/peerconnectionendtoend_unittest.cc
zhihuang 9763d56464 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
To allow end-to-end QuicDataChannel usage with a
PeerConnection, RTCConfiguration has been modified to
include a boolean for whether to do QUIC, since negotiation of
QUIC is not implemented. If one peer does QUIC, then it will be
assumed that the other peer must do QUIC or the connection
will fail.

PeerConnection has been modified to create data channels of type
QuicDataChannel when the peer wants to do QUIC.

WebRtcSession has ben modified to use a QuicDataTransport
instead of a DtlsTransportChannelWrapper/DataChannel
when QUIC should be used

QuicDataTransport implements the generic functions of
BaseChannel to manage the QuicTransportChannel.

Committed: https://crrev.com/34b54c36a533dadb6ceb70795119194e6f530ef5
Review-Url: https://codereview.webrtc.org/2166873002
Cr-Original-Commit-Position: refs/heads/master@{#13645}
Cr-Commit-Position: refs/heads/master@{#13657}
2016-08-05 18:14:54 +00:00

455 lines
16 KiB
C++

/*
* Copyright 2013 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 "webrtc/api/test/peerconnectiontestwrapper.h"
// Notice that mockpeerconnectionobservers.h must be included after the above!
#include "webrtc/api/test/mockpeerconnectionobservers.h"
#ifdef WEBRTC_ANDROID
#include "webrtc/api/test/androidtestinitializer.h"
#endif
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
namespace {
const int kMaxWait = 10000;
} // namespace
class PeerConnectionEndToEndTest
: public sigslot::has_slots<>,
public testing::Test {
public:
typedef std::vector<rtc::scoped_refptr<DataChannelInterface> >
DataChannelList;
PeerConnectionEndToEndTest() {
RTC_CHECK(network_thread_.Start());
RTC_CHECK(worker_thread_.Start());
caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
"caller", &network_thread_, &worker_thread_);
callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
"callee", &network_thread_, &worker_thread_);
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:stun.l.google.com:19302";
config_.servers.push_back(ice_server);
#ifdef WEBRTC_ANDROID
webrtc::InitializeAndroidObjects();
#endif
}
void CreatePcs() {
CreatePcs(NULL);
}
void CreatePcs(const MediaConstraintsInterface* pc_constraints) {
EXPECT_TRUE(caller_->CreatePc(pc_constraints, config_));
EXPECT_TRUE(callee_->CreatePc(pc_constraints, config_));
PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
caller_->SignalOnDataChannel.connect(
this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel);
callee_->SignalOnDataChannel.connect(
this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel);
}
void GetAndAddUserMedia() {
FakeConstraints audio_constraints;
FakeConstraints video_constraints;
GetAndAddUserMedia(true, audio_constraints, true, video_constraints);
}
void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints,
bool video, FakeConstraints video_constraints) {
caller_->GetAndAddUserMedia(audio, audio_constraints,
video, video_constraints);
callee_->GetAndAddUserMedia(audio, audio_constraints,
video, video_constraints);
}
void Negotiate() {
caller_->CreateOffer(NULL);
}
void WaitForCallEstablished() {
caller_->WaitForCallEstablished();
callee_->WaitForCallEstablished();
}
void WaitForConnection() {
caller_->WaitForConnection();
callee_->WaitForConnection();
}
void OnCallerAddedDataChanel(DataChannelInterface* dc) {
caller_signaled_data_channels_.push_back(dc);
}
void OnCalleeAddedDataChannel(DataChannelInterface* dc) {
callee_signaled_data_channels_.push_back(dc);
}
// Tests that |dc1| and |dc2| can send to and receive from each other.
void TestDataChannelSendAndReceive(
DataChannelInterface* dc1, DataChannelInterface* dc2) {
std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer(
new webrtc::MockDataChannelObserver(dc1));
std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer(
new webrtc::MockDataChannelObserver(dc2));
static const std::string kDummyData = "abcdefg";
webrtc::DataBuffer buffer(kDummyData);
EXPECT_TRUE(dc1->Send(buffer));
EXPECT_EQ_WAIT(kDummyData, dc2_observer->last_message(), kMaxWait);
EXPECT_TRUE(dc2->Send(buffer));
EXPECT_EQ_WAIT(kDummyData, dc1_observer->last_message(), kMaxWait);
EXPECT_EQ(1U, dc1_observer->received_message_count());
EXPECT_EQ(1U, dc2_observer->received_message_count());
}
void WaitForDataChannelsToOpen(DataChannelInterface* local_dc,
const DataChannelList& remote_dc_list,
size_t remote_dc_index) {
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait);
EXPECT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait);
EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
remote_dc_list[remote_dc_index]->state(),
kMaxWait);
EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id());
}
void CloseDataChannels(DataChannelInterface* local_dc,
const DataChannelList& remote_dc_list,
size_t remote_dc_index) {
local_dc->Close();
EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait);
EXPECT_EQ_WAIT(DataChannelInterface::kClosed,
remote_dc_list[remote_dc_index]->state(),
kMaxWait);
}
protected:
rtc::Thread network_thread_;
rtc::Thread worker_thread_;
rtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
rtc::scoped_refptr<PeerConnectionTestWrapper> callee_;
DataChannelList caller_signaled_data_channels_;
DataChannelList callee_signaled_data_channels_;
webrtc::PeerConnectionInterface::RTCConfiguration config_;
};
// Disabled for TSan v2, see
// https://bugs.chromium.org/p/webrtc/issues/detail?id=4719 for details.
// Disabled for Mac, see
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details.
#if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
TEST_F(PeerConnectionEndToEndTest, Call) {
CreatePcs();
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
#endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
FakeConstraints pc_constraints;
pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
false);
CreatePcs(&pc_constraints);
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
// Verifies that a DataChannel created before the negotiation can transition to
// "OPEN" and transfer data.
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
rtc::scoped_refptr<DataChannelInterface> caller_dc(
caller_->CreateDataChannel("data", init));
rtc::scoped_refptr<DataChannelInterface> callee_dc(
callee_->CreateDataChannel("data", init));
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[0]);
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
}
// Verifies that a DataChannel created after the negotiation can transition to
// "OPEN" and transfer data.
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
// This DataChannel is for creating the data content in the negotiation.
rtc::scoped_refptr<DataChannelInterface> dummy(
caller_->CreateDataChannel("data", init));
Negotiate();
WaitForConnection();
// Wait for the data channel created pre-negotiation to be opened.
WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0);
// Create new DataChannels after the negotiation and verify their states.
rtc::scoped_refptr<DataChannelInterface> caller_dc(
caller_->CreateDataChannel("hello", init));
rtc::scoped_refptr<DataChannelInterface> callee_dc(
callee_->CreateDataChannel("hello", init));
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
}
// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
caller_->CreateDataChannel("data", init));
rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
callee_->CreateDataChannel("data", init));
Negotiate();
WaitForConnection();
EXPECT_EQ(1U, caller_dc_1->id() % 2);
EXPECT_EQ(0U, callee_dc_1->id() % 2);
rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
caller_->CreateDataChannel("data", init));
rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
callee_->CreateDataChannel("data", init));
EXPECT_EQ(1U, caller_dc_2->id() % 2);
EXPECT_EQ(0U, callee_dc_2->id() % 2);
}
// Verifies that the message is received by the right remote DataChannel when
// there are multiple DataChannels.
TEST_F(PeerConnectionEndToEndTest,
MessageTransferBetweenTwoPairsOfDataChannels) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
caller_->CreateDataChannel("data", init));
rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
caller_->CreateDataChannel("data", init));
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1);
std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0]));
std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1]));
const std::string message_1 = "hello 1";
const std::string message_2 = "hello 2";
caller_dc_1->Send(webrtc::DataBuffer(message_1));
EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
caller_dc_2->Send(webrtc::DataBuffer(message_2));
EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
EXPECT_EQ(1U, dc_1_observer->received_message_count());
EXPECT_EQ(1U, dc_2_observer->received_message_count());
}
#ifdef HAVE_QUIC
// Test that QUIC data channels can be used and that messages go to the correct
// remote data channel when both peers want to use QUIC. It is assumed that the
// application has externally negotiated the data channel parameters.
TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) {
config_.enable_quic = true;
CreatePcs();
webrtc::DataChannelInit init_1;
init_1.id = 0;
init_1.ordered = false;
init_1.reliable = true;
webrtc::DataChannelInit init_2;
init_2.id = 1;
init_2.ordered = false;
init_2.reliable = true;
rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
caller_->CreateDataChannel("data", init_1));
ASSERT_NE(nullptr, caller_dc_1);
rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
caller_->CreateDataChannel("data", init_2));
ASSERT_NE(nullptr, caller_dc_2);
rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
callee_->CreateDataChannel("data", init_1));
ASSERT_NE(nullptr, callee_dc_1);
rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
callee_->CreateDataChannel("data", init_2));
ASSERT_NE(nullptr, callee_dc_2);
Negotiate();
WaitForConnection();
EXPECT_TRUE_WAIT(caller_dc_1->state() == webrtc::DataChannelInterface::kOpen,
kMaxWait);
EXPECT_TRUE_WAIT(callee_dc_1->state() == webrtc::DataChannelInterface::kOpen,
kMaxWait);
EXPECT_TRUE_WAIT(caller_dc_2->state() == webrtc::DataChannelInterface::kOpen,
kMaxWait);
EXPECT_TRUE_WAIT(callee_dc_2->state() == webrtc::DataChannelInterface::kOpen,
kMaxWait);
std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
new webrtc::MockDataChannelObserver(callee_dc_1.get()));
std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
new webrtc::MockDataChannelObserver(callee_dc_2.get()));
const std::string message_1 = "hello 1";
const std::string message_2 = "hello 2";
// Send data from caller to callee.
caller_dc_1->Send(webrtc::DataBuffer(message_1));
EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
caller_dc_2->Send(webrtc::DataBuffer(message_2));
EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
EXPECT_EQ(1U, dc_1_observer->received_message_count());
EXPECT_EQ(1U, dc_2_observer->received_message_count());
// Send data from callee to caller.
dc_1_observer.reset(new webrtc::MockDataChannelObserver(caller_dc_1.get()));
dc_2_observer.reset(new webrtc::MockDataChannelObserver(caller_dc_2.get()));
callee_dc_1->Send(webrtc::DataBuffer(message_1));
EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
callee_dc_2->Send(webrtc::DataBuffer(message_2));
EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
EXPECT_EQ(1U, dc_1_observer->received_message_count());
EXPECT_EQ(1U, dc_2_observer->received_message_count());
}
#endif // HAVE_QUIC
// Verifies that a DataChannel added from an OPEN message functions after
// a channel has been previously closed (webrtc issue 3778).
// This previously failed because the new channel re-uses the ID of the closed
// channel, and the closed channel was incorrectly still assigned to the id.
// TODO(deadbeef): This is disabled because there's currently a race condition
// caused by the fact that a data channel signals that it's closed before it
// really is. Re-enable this test once that's fixed.
TEST_F(PeerConnectionEndToEndTest,
DISABLED_DataChannelFromOpenWorksAfterClose) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
rtc::scoped_refptr<DataChannelInterface> caller_dc(
caller_->CreateDataChannel("data", init));
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
// Create a new channel and ensure it works after closing the previous one.
caller_dc = caller_->CreateDataChannel("data2", init);
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
}
// This tests that if a data channel is closed remotely while not referenced
// by the application (meaning only the PeerConnection contributes to its
// reference count), no memory access violation will occur.
// See: https://code.google.com/p/chromium/issues/detail?id=565048
TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
webrtc::DataChannelInit init;
rtc::scoped_refptr<DataChannelInterface> caller_dc(
caller_->CreateDataChannel("data", init));
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
// This removes the reference to the remote data channel that we hold.
callee_signaled_data_channels_.clear();
caller_dc->Close();
EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
// Wait for a bit longer so the remote data channel will receive the
// close message and be destroyed.
rtc::Thread::Current()->ProcessMessages(100);
}