zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
171df9326200d1e01bce530e2ff01ac5890e6cb7
platform-external-webrtc/pc/peer_connection_integrationtest.cc
Jonas Olsson df919fb425 Don't pretend we've received an end-of-candidates indication. 
Since end-of-candidates signalling isn't implemented yet, the ice transport shouldn't reach completed. We also shouldn't assume that the transport has failed because gathering is complete without candidates, as we might still get remote candidates.

Bug: chromium:922588
Change-Id: I332f57be494efc775819d80908e9f39610311f82
Reviewed-on: https://webrtc-review.googlesource.com/c/118741
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26365}
2019-01-23 10:16:43 +00:00

5096 lines
214 KiB
C++
Raw Blame History

/*
* Copyright 2012 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.
*/
// Disable for TSan v2, see
// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
#if !defined(THREAD_SANITIZER)
#include <stdio.h>
#include <algorithm>
#include <functional>
#include <list>
#include <map>
#include <memory>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/media_stream_interface.h"
#include "api/peer_connection_interface.h"
#include "api/peer_connection_proxy.h"
#include "api/rtp_receiver_interface.h"
#include "api/test/loopback_media_transport.h"
#include "api/uma_metrics.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/call.h"
#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
#include "logging/rtc_event_log/rtc_event_log_factory.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/engine/fake_webrtc_video_engine.h"
#include "media/engine/webrtc_media_engine.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/mock_async_resolver.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port_interface.h"
#include "p2p/base/test_stun_server.h"
#include "p2p/base/test_turn_customizer.h"
#include "p2p/base/test_turn_server.h"
#include "p2p/client/basic_port_allocator.h"
#include "pc/dtmf_sender.h"
#include "pc/local_audio_source.h"
#include "pc/media_session.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_factory.h"
#include "pc/rtp_media_utils.h"
#include "pc/session_description.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/fake_periodic_video_track_source.h"
#include "pc/test/fake_rtc_certificate_generator.h"
#include "pc/test/fake_video_track_renderer.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/fake_network.h"
#include "rtc_base/firewall_socket_server.h"
#include "rtc_base/gunit.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/test_certificate_verifier.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
#include "system_wrappers/include/metrics.h"
#include "test/gmock.h"
namespace webrtc {
namespace {
using ::cricket::ContentInfo;
using ::cricket::StreamParams;
using ::rtc::SocketAddress;
using ::testing::_;
using ::testing::Combine;
using ::testing::ElementsAre;
using ::testing::Return;
using ::testing::NiceMock;
using ::testing::SetArgPointee;
using ::testing::UnorderedElementsAreArray;
using ::testing::Values;
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
static const int kDefaultTimeout = 10000;
static const int kMaxWaitForStatsMs = 3000;
static const int kMaxWaitForActivationMs = 5000;
static const int kMaxWaitForFramesMs = 10000;
// Default number of audio/video frames to wait for before considering a test
// successful.
static const int kDefaultExpectedAudioFrameCount = 3;
static const int kDefaultExpectedVideoFrameCount = 3;
static const char kDataChannelLabel[] = "data_channel";
// SRTP cipher name negotiated by the tests. This must be updated if the
// default changes.
static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_80;
static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;
static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0);
// Helper function for constructing offer/answer options to initiate an ICE
// restart.
PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() {
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.ice_restart = true;
return options;
}
// Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic"
// attribute from received SDP, simulating a legacy endpoint.
void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) {
for (ContentInfo& content : desc->contents()) {
content.media_description()->mutable_streams().clear();
}
desc->set_msid_supported(false);
desc->set_msid_signaling(0);
}
// Removes all stream information besides the stream ids, simulating an
// endpoint that only signals a=msid lines to convey stream_ids.
void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
for (ContentInfo& content : desc->contents()) {
std::string track_id;
std::vector<std::string> stream_ids;
if (!content.media_description()->streams().empty()) {
const StreamParams& first_stream =
content.media_description()->streams()[0];
track_id = first_stream.id;
stream_ids = first_stream.stream_ids();
}
content.media_description()->mutable_streams().clear();
StreamParams new_stream;
new_stream.id = track_id;
new_stream.set_stream_ids(stream_ids);
content.media_description()->AddStream(new_stream);
}
}
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
media_stats_vec) {
for (size_t i = 0; i < media_stats_vec.size(); i++) {
if (media_stats_vec[i]->kind.ValueToString() == kind) {
return i;
}
}
return -1;
}
class SignalingMessageReceiver {
public:
virtual void ReceiveSdpMessage(SdpType type, const std::string& msg) = 0;
virtual void ReceiveIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) = 0;
protected:
SignalingMessageReceiver() {}
virtual ~SignalingMessageReceiver() {}
};
class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
public:
explicit MockRtpReceiverObserver(cricket::MediaType media_type)
: expected_media_type_(media_type) {}
void OnFirstPacketReceived(cricket::MediaType media_type) override {
ASSERT_EQ(expected_media_type_, media_type);
first_packet_received_ = true;
}
bool first_packet_received() const { return first_packet_received_; }
virtual ~MockRtpReceiverObserver() {}
private:
bool first_packet_received_ = false;
cricket::MediaType expected_media_type_;
};
// Helper class that wraps a peer connection, observes it, and can accept
// signaling messages from another wrapper.
//
// Uses a fake network, fake A/V capture, and optionally fake
// encoders/decoders, though they aren't used by default since they don't
// advertise support of any codecs.
// TODO(steveanton): See how this could become a subclass of
// PeerConnectionWrapper defined in peerconnectionwrapper.h.
class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
public SignalingMessageReceiver {
public:
// Different factory methods for convenience.
// TODO(deadbeef): Could use the pattern of:
//
// PeerConnectionWrapper =
// WrapperBuilder.WithConfig(...).WithOptions(...).build();
//
// To reduce some code duplication.
static PeerConnectionWrapper* CreateWithDtlsIdentityStore(
const std::string& debug_name,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
webrtc::PeerConnectionDependencies dependencies(nullptr);
dependencies.cert_generator = std::move(cert_generator);
if (!client->Init(nullptr, nullptr, std::move(dependencies), network_thread,
worker_thread, nullptr,
/*media_transport_factory=*/nullptr)) {
delete client;
return nullptr;
}
return client;
}
webrtc::PeerConnectionFactoryInterface* pc_factory() const {
return peer_connection_factory_.get();
}
webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
// If a signaling message receiver is set (via ConnectFakeSignaling), this
// will set the whole offer/answer exchange in motion. Just need to wait for
// the signaling state to reach "stable".
void CreateAndSetAndSignalOffer() {
auto offer = CreateOffer();
ASSERT_NE(nullptr, offer);
EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer)));
}
// Sets the options to be used when CreateAndSetAndSignalOffer is called, or
// when a remote offer is received (via fake signaling) and an answer is
// generated. By default, uses default options.
void SetOfferAnswerOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
offer_answer_options_ = options;
}
// Set a callback to be invoked when SDP is received via the fake signaling
// channel, which provides an opportunity to munge (modify) the SDP. This is
// used to test SDP being applied that a PeerConnection would normally not
// generate, but a non-JSEP endpoint might.
void SetReceivedSdpMunger(
std::function<void(cricket::SessionDescription*)> munger) {
received_sdp_munger_ = std::move(munger);
}
// Similar to the above, but this is run on SDP immediately after it's
// generated.
void SetGeneratedSdpMunger(
std::function<void(cricket::SessionDescription*)> munger) {
generated_sdp_munger_ = std::move(munger);
}
// Set a callback to be invoked when a remote offer is received via the fake
// signaling channel. This provides an opportunity to change the
// PeerConnection state before an answer is created and sent to the caller.
void SetRemoteOfferHandler(std::function<void()> handler) {
remote_offer_handler_ = std::move(handler);
}
void SetRemoteAsyncResolver(rtc::MockAsyncResolver* resolver) {
remote_async_resolver_ = resolver;
}
// Every ICE connection state in order that has been seen by the observer.
std::vector<PeerConnectionInterface::IceConnectionState>
ice_connection_state_history() const {
return ice_connection_state_history_;
}
void clear_ice_connection_state_history() {
ice_connection_state_history_.clear();
}
// Every PeerConnection state in order that has been seen by the observer.
std::vector<PeerConnectionInterface::PeerConnectionState>
peer_connection_state_history() const {
return peer_connection_state_history_;
}
// Every ICE gathering state in order that has been seen by the observer.
std::vector<PeerConnectionInterface::IceGatheringState>
ice_gathering_state_history() const {
return ice_gathering_state_history_;
}
void AddAudioVideoTracks() {
AddAudioTrack();
AddVideoTrack();
}
rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack() {
return AddTrack(CreateLocalAudioTrack());
}
rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack() {
return AddTrack(CreateLocalVideoTrack());
}
rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
cricket::AudioOptions options;
// Disable highpass filter so that we can get all the test audio frames.
options.highpass_filter = false;
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
peer_connection_factory_->CreateAudioSource(options);
// TODO(perkj): Test audio source when it is implemented. Currently audio
// always use the default input.
return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(),
source);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
webrtc::FakePeriodicVideoSource::Config config;
config.timestamp_offset_ms = rtc::TimeMillis();
return CreateLocalVideoTrackInternal(config);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface>
CreateLocalVideoTrackWithConfig(
webrtc::FakePeriodicVideoSource::Config config) {
return CreateLocalVideoTrackInternal(config);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface>
CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
webrtc::FakePeriodicVideoSource::Config config;
config.rotation = rotation;
config.timestamp_offset_ms = rtc::TimeMillis();
return CreateLocalVideoTrackInternal(config);
}
rtc::scoped_refptr<RtpSenderInterface> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids = {}) {
auto result = pc()->AddTrack(track, stream_ids);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
return result.MoveValue();
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType(
cricket::MediaType media_type) {
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
for (auto receiver : pc()->GetReceivers()) {
if (receiver->media_type() == media_type) {
receivers.push_back(receiver);
}
}
return receivers;
}
rtc::scoped_refptr<RtpTransceiverInterface> GetFirstTransceiverOfType(
cricket::MediaType media_type) {
for (auto transceiver : pc()->GetTransceivers()) {
if (transceiver->receiver()->media_type() == media_type) {
return transceiver;
}
}
return nullptr;
}
bool SignalingStateStable() {
return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
}
void CreateDataChannel() { CreateDataChannel(nullptr); }
void CreateDataChannel(const webrtc::DataChannelInit* init) {
CreateDataChannel(kDataChannelLabel, init);
}
void CreateDataChannel(const std::string& label,
const webrtc::DataChannelInit* init) {
data_channel_ = pc()->CreateDataChannel(label, init);
ASSERT_TRUE(data_channel_.get() != nullptr);
data_observer_.reset(new MockDataChannelObserver(data_channel_));
}
DataChannelInterface* data_channel() { return data_channel_; }
const MockDataChannelObserver* data_observer() const {
return data_observer_.get();
}
int audio_frames_received() const {
return fake_audio_capture_module_->frames_received();
}
// Takes minimum of video frames received for each track.
//
// Can be used like:
// EXPECT_GE(expected_frames, min_video_frames_received_per_track());
//
// To ensure that all video tracks received at least a certain number of
// frames.
int min_video_frames_received_per_track() const {
int min_frames = INT_MAX;
if (fake_video_renderers_.empty()) {
return 0;
}
for (const auto& pair : fake_video_renderers_) {
min_frames = std::min(min_frames, pair.second->num_rendered_frames());
}
return min_frames;
}
// Returns a MockStatsObserver in a state after stats gathering finished,
// which can be used to access the gathered stats.
rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack(
webrtc::MediaStreamTrackInterface* track) {
rtc::scoped_refptr<MockStatsObserver> observer(
new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
return observer;
}
// Version that doesn't take a track "filter", and gathers all stats.
rtc::scoped_refptr<MockStatsObserver> OldGetStats() {
return OldGetStatsForTrack(nullptr);
}
// Synchronously gets stats and returns them. If it times out, fails the test
// and returns null.
rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
peer_connection_->GetStats(callback);
EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
return callback->report();
}
int rendered_width() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty()
? 0
: fake_video_renderers_.begin()->second->width();
}
int rendered_height() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty()
? 0
: fake_video_renderers_.begin()->second->height();
}
double rendered_aspect_ratio() {
if (rendered_height() == 0) {
return 0.0;
}
return static_cast<double>(rendered_width()) / rendered_height();
}
webrtc::VideoRotation rendered_rotation() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty()
? webrtc::kVideoRotation_0
: fake_video_renderers_.begin()->second->rotation();
}
int local_rendered_width() {
return local_video_renderer_ ? local_video_renderer_->width() : 0;
}
int local_rendered_height() {
return local_video_renderer_ ? local_video_renderer_->height() : 0;
}
double local_rendered_aspect_ratio() {
if (local_rendered_height() == 0) {
return 0.0;
}
return static_cast<double>(local_rendered_width()) /
local_rendered_height();
}
size_t number_of_remote_streams() {
if (!pc()) {
return 0;
}
return pc()->remote_streams()->count();
}
StreamCollectionInterface* remote_streams() const {
if (!pc()) {
ADD_FAILURE();
return nullptr;
}
return pc()->remote_streams();
}
StreamCollectionInterface* local_streams() {
if (!pc()) {
ADD_FAILURE();
return nullptr;
}
return pc()->local_streams();
}
webrtc::PeerConnectionInterface::SignalingState signaling_state() {
return pc()->signaling_state();
}
webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
return pc()->ice_connection_state();
}
webrtc::PeerConnectionInterface::IceConnectionState
standardized_ice_connection_state() {
return pc()->standardized_ice_connection_state();
}
webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
return pc()->ice_gathering_state();
}
// Returns a MockRtpReceiverObserver for each RtpReceiver returned by
// GetReceivers. They're updated automatically when a remote offer/answer
// from the fake signaling channel is applied, or when
// ResetRtpReceiverObservers below is called.
const std::vector<std::unique_ptr<MockRtpReceiverObserver>>&
rtp_receiver_observers() {
return rtp_receiver_observers_;
}
void ResetRtpReceiverObservers() {
rtp_receiver_observers_.clear();
for (const rtc::scoped_refptr<RtpReceiverInterface>& receiver :
pc()->GetReceivers()) {
std::unique_ptr<MockRtpReceiverObserver> observer(
new MockRtpReceiverObserver(receiver->media_type()));
receiver->SetObserver(observer.get());
rtp_receiver_observers_.push_back(std::move(observer));
}
}
rtc::FakeNetworkManager* network() const {
return fake_network_manager_.get();
}
cricket::PortAllocator* port_allocator() const { return port_allocator_; }
webrtc::FakeRtcEventLogFactory* event_log_factory() const {
return event_log_factory_;
}
private:
explicit PeerConnectionWrapper(const std::string& debug_name)
: debug_name_(debug_name) {}
bool Init(
const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies,
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) {
// There's an error in this test code if Init ends up being called twice.
RTC_DCHECK(!peer_connection_);
RTC_DCHECK(!peer_connection_factory_);
fake_network_manager_.reset(new rtc::FakeNetworkManager());
fake_network_manager_->AddInterface(kDefaultLocalAddress);
std::unique_ptr<cricket::PortAllocator> port_allocator(
new cricket::BasicPortAllocator(fake_network_manager_.get()));
port_allocator_ = port_allocator.get();
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
if (!fake_audio_capture_module_) {
return false;
}
rtc::Thread* const signaling_thread = rtc::Thread::Current();
webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies;
pc_factory_dependencies.network_thread = network_thread;
pc_factory_dependencies.worker_thread = worker_thread;
pc_factory_dependencies.signaling_thread = signaling_thread;
pc_factory_dependencies.media_engine =
cricket::WebRtcMediaEngineFactory::Create(
rtc::scoped_refptr<webrtc::AudioDeviceModule>(
fake_audio_capture_module_),
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(), nullptr,
webrtc::AudioProcessingBuilder().Create());
pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();
if (event_log_factory) {
event_log_factory_ = event_log_factory.get();
pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
} else {
pc_factory_dependencies.event_log_factory =
webrtc::CreateRtcEventLogFactory();
}
if (media_transport_factory) {
pc_factory_dependencies.media_transport_factory =
std::move(media_transport_factory);
}
peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
std::move(pc_factory_dependencies));
if (!peer_connection_factory_) {
return false;
}
if (options) {
peer_connection_factory_->SetOptions(*options);
}
if (config) {
sdp_semantics_ = config->sdp_semantics;
}
dependencies.allocator = std::move(port_allocator);
peer_connection_ = CreatePeerConnection(config, std::move(dependencies));
return peer_connection_.get() != nullptr;
}
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies) {
PeerConnectionInterface::RTCConfiguration modified_config;
// If |config| is null, this will result in a default configuration being
// used.
if (config) {
modified_config = *config;
}
// Disable resolution adaptation; we don't want it interfering with the
// test results.
// TODO(deadbeef): Do something more robust. Since we're testing for aspect
// ratios and not specific resolutions, is this even necessary?
modified_config.set_cpu_adaptation(false);
dependencies.observer = this;
return peer_connection_factory_->CreatePeerConnection(
modified_config, std::move(dependencies));
}
void set_signaling_message_receiver(
SignalingMessageReceiver* signaling_message_receiver) {
signaling_message_receiver_ = signaling_message_receiver;
}
void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; }
void set_signal_ice_candidates(bool signal) {
signal_ice_candidates_ = signal;
}
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
webrtc::FakePeriodicVideoSource::Config config) {
// Set max frame rate to 10fps to reduce the risk of test flakiness.
// TODO(deadbeef): Do something more robust.
config.frame_interval_ms = 100;
video_track_sources_.emplace_back(
new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>(
config, false /* remote */));
rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
peer_connection_factory_->CreateVideoTrack(
rtc::CreateRandomUuid(), video_track_sources_.back()));
if (!local_video_renderer_) {
local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
}
return track;
}
void HandleIncomingOffer(const std::string& msg) {
RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer";
std::unique_ptr<SessionDescriptionInterface> desc =
webrtc::CreateSessionDescription(SdpType::kOffer, msg);
if (received_sdp_munger_) {
received_sdp_munger_(desc->description());
}
EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
// Setting a remote description may have changed the number of receivers,
// so reset the receiver observers.
ResetRtpReceiverObservers();
if (remote_offer_handler_) {
remote_offer_handler_();
}
auto answer = CreateAnswer();
ASSERT_NE(nullptr, answer);
EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer)));
}
void HandleIncomingAnswer(const std::string& msg) {
RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer";
std::unique_ptr<SessionDescriptionInterface> desc =
webrtc::CreateSessionDescription(SdpType::kAnswer, msg);
if (received_sdp_munger_) {
received_sdp_munger_(desc->description());
}
EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
// Set the RtpReceiverObserver after receivers are created.
ResetRtpReceiverObservers();
}
// Returns null on failure.
std::unique_ptr<SessionDescriptionInterface> CreateOffer() {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
pc()->CreateOffer(observer, offer_answer_options_);
return WaitForDescriptionFromObserver(observer);
}
// Returns null on failure.
std::unique_ptr<SessionDescriptionInterface> CreateAnswer() {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
pc()->CreateAnswer(observer, offer_answer_options_);
return WaitForDescriptionFromObserver(observer);
}
std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver(
MockCreateSessionDescriptionObserver* observer) {
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
if (!observer->result()) {
return nullptr;
}
auto description = observer->MoveDescription();
if (generated_sdp_munger_) {
generated_sdp_munger_(description->description());
}
return description;
}
// Setting the local description and sending the SDP message over the fake
// signaling channel are combined into the same method because the SDP
// message needs to be sent as soon as SetLocalDescription finishes, without
// waiting for the observer to be called. This ensures that ICE candidates
// don't outrace the description.
bool SetLocalDescriptionAndSendSdpMessage(
std::unique_ptr<SessionDescriptionInterface> desc) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage";
SdpType type = desc->GetType();
std::string sdp;
EXPECT_TRUE(desc->ToString(&sdp));
pc()->SetLocalDescription(observer, desc.release());
if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
RemoveUnusedVideoRenderers();
}
// As mentioned above, we need to send the message immediately after
// SetLocalDescription.
SendSdpMessage(type, sdp);
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
return true;
}
bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription";
pc()->SetRemoteDescription(observer, desc.release());
if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
RemoveUnusedVideoRenderers();
}
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
return observer->result();
}
// This is a work around to remove unused fake_video_renderers from
// transceivers that have either stopped or are no longer receiving.
void RemoveUnusedVideoRenderers() {
auto transceivers = pc()->GetTransceivers();
for (auto& transceiver : transceivers) {
if (transceiver->receiver()->media_type() != cricket::MEDIA_TYPE_VIDEO) {
continue;
}
// Remove fake video renderers from any stopped transceivers.
if (transceiver->stopped()) {
auto it =
fake_video_renderers_.find(transceiver->receiver()->track()->id());
if (it != fake_video_renderers_.end()) {
fake_video_renderers_.erase(it);
}
}
// Remove fake video renderers from any transceivers that are no longer
// receiving.
if ((transceiver->current_direction() &&
!webrtc::RtpTransceiverDirectionHasRecv(
*transceiver->current_direction()))) {
auto it =
fake_video_renderers_.find(transceiver->receiver()->track()->id());
if (it != fake_video_renderers_.end()) {
fake_video_renderers_.erase(it);
}
}
}
}
// Simulate sending a blob of SDP with delay |signaling_delay_ms_| (0 by
// default).
void SendSdpMessage(SdpType type, const std::string& msg) {
if (signaling_delay_ms_ == 0) {
RelaySdpMessageIfReceiverExists(type, msg);
} else {
invoker_.AsyncInvokeDelayed<void>(
RTC_FROM_HERE, rtc::Thread::Current(),
rtc::Bind(&PeerConnectionWrapper::RelaySdpMessageIfReceiverExists,
this, type, msg),
signaling_delay_ms_);
}
}
void RelaySdpMessageIfReceiverExists(SdpType type, const std::string& msg) {
if (signaling_message_receiver_) {
signaling_message_receiver_->ReceiveSdpMessage(type, msg);
}
}
// Simulate trickling an ICE candidate with delay |signaling_delay_ms_| (0 by
// default).
void SendIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) {
if (signaling_delay_ms_ == 0) {
RelayIceMessageIfReceiverExists(sdp_mid, sdp_mline_index, msg);
} else {
invoker_.AsyncInvokeDelayed<void>(
RTC_FROM_HERE, rtc::Thread::Current(),
rtc::Bind(&PeerConnectionWrapper::RelayIceMessageIfReceiverExists,
this, sdp_mid, sdp_mline_index, msg),
signaling_delay_ms_);
}
}
void RelayIceMessageIfReceiverExists(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) {
if (signaling_message_receiver_) {
signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index,
msg);
}
}
// SignalingMessageReceiver callbacks.
void ReceiveSdpMessage(SdpType type, const std::string& msg) override {
if (type == SdpType::kOffer) {
HandleIncomingOffer(msg);
} else {
HandleIncomingAnswer(msg);
}
}
void ReceiveIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) override {
RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage";
std::unique_ptr<webrtc::IceCandidateInterface> candidate(
webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
}
// PeerConnectionObserver callbacks.
void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) override {
EXPECT_EQ(pc()->signaling_state(), new_state);
}
void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override {
if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
rtc::scoped_refptr<VideoTrackInterface> video_track(
static_cast<VideoTrackInterface*>(receiver->track().get()));
ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
fake_video_renderers_.end());
fake_video_renderers_[video_track->id()] =
absl::make_unique<FakeVideoTrackRenderer>(video_track);
}
}
void OnRemoveTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
auto it = fake_video_renderers_.find(receiver->track()->id());
RTC_DCHECK(it != fake_video_renderers_.end());
fake_video_renderers_.erase(it);
}
}
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
EXPECT_EQ(pc()->ice_connection_state(), new_state);
ice_connection_state_history_.push_back(new_state);
}
void OnConnectionChange(
webrtc::PeerConnectionInterface::PeerConnectionState new_state) override {
peer_connection_state_history_.push_back(new_state);
}
void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
EXPECT_EQ(pc()->ice_gathering_state(), new_state);
ice_gathering_state_history_.push_back(new_state);
}
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate";
if (remote_async_resolver_) {
const auto& local_candidate = candidate->candidate();
const auto& mdns_responder = network()->GetMdnsResponderForTesting();
if (local_candidate.address().IsUnresolvedIP()) {
RTC_DCHECK(local_candidate.type() == cricket::LOCAL_PORT_TYPE);
rtc::SocketAddress resolved_addr(local_candidate.address());
const auto resolved_ip = mdns_responder->GetMappedAddressForName(
local_candidate.address().hostname());
RTC_DCHECK(!resolved_ip.IsNil());
resolved_addr.SetResolvedIP(resolved_ip);
EXPECT_CALL(*remote_async_resolver_, GetResolvedAddress(_, _))
.WillOnce(DoAll(SetArgPointee<1>(resolved_addr), Return(true)));
EXPECT_CALL(*remote_async_resolver_, Destroy(_));
}
}
std::string ice_sdp;
EXPECT_TRUE(candidate->ToString(&ice_sdp));
if (signaling_message_receiver_ == nullptr || !signal_ice_candidates_) {
// Remote party may be deleted.
return;
}
SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
}
void OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) override {
RTC_LOG(LS_INFO) << debug_name_ << ": OnDataChannel";
data_channel_ = data_channel;
data_observer_.reset(new MockDataChannelObserver(data_channel));
}
std::string debug_name_;
std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
cricket::PortAllocator* port_allocator_;
// Needed to keep track of number of frames sent.
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
// Needed to keep track of number of frames received.
std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
fake_video_renderers_;
// Needed to ensure frames aren't received for removed tracks.
std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
removed_fake_video_renderers_;
// For remote peer communication.
SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
int signaling_delay_ms_ = 0;
bool signal_ice_candidates_ = true;
// Store references to the video sources we've created, so that we can stop
// them, if required.
std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>>
video_track_sources_;
// |local_video_renderer_| attached to the first created local video track.
std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
SdpSemantics sdp_semantics_;
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
std::function<void(cricket::SessionDescription*)> received_sdp_munger_;
std::function<void(cricket::SessionDescription*)> generated_sdp_munger_;
std::function<void()> remote_offer_handler_;
rtc::MockAsyncResolver* remote_async_resolver_ = nullptr;
rtc::scoped_refptr<DataChannelInterface> data_channel_;
std::unique_ptr<MockDataChannelObserver> data_observer_;
std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;
std::vector<PeerConnectionInterface::IceConnectionState>
ice_connection_state_history_;
std::vector<PeerConnectionInterface::PeerConnectionState>
peer_connection_state_history_;
std::vector<PeerConnectionInterface::IceGatheringState>
ice_gathering_state_history_;
webrtc::FakeRtcEventLogFactory* event_log_factory_;
rtc::AsyncInvoker invoker_;
friend class PeerConnectionIntegrationBaseTest;
};
class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
public:
virtual ~MockRtcEventLogOutput() = default;
MOCK_CONST_METHOD0(IsActive, bool());
MOCK_METHOD1(Write, bool(const std::string&));
};
// This helper object is used for both specifying how many audio/video frames
// are expected to be received for a caller/callee. It provides helper functions
// to specify these expectations. The object initially starts in a state of no
// expectations.
class MediaExpectations {
public:
enum ExpectFrames {
kExpectSomeFrames,
kExpectNoFrames,
kNoExpectation,
};
void ExpectBidirectionalAudioAndVideo() {
ExpectBidirectionalAudio();
ExpectBidirectionalVideo();
}
void ExpectBidirectionalAudio() {
CallerExpectsSomeAudio();
CalleeExpectsSomeAudio();
}
void ExpectNoAudio() {
CallerExpectsNoAudio();
CalleeExpectsNoAudio();
}
void ExpectBidirectionalVideo() {
CallerExpectsSomeVideo();
CalleeExpectsSomeVideo();
}
void ExpectNoVideo() {
CallerExpectsNoVideo();
CalleeExpectsNoVideo();
}
void CallerExpectsSomeAudioAndVideo() {
CallerExpectsSomeAudio();
CallerExpectsSomeVideo();
}
void CalleeExpectsSomeAudioAndVideo() {
CalleeExpectsSomeAudio();
CalleeExpectsSomeVideo();
}
// Caller's audio functions.
void CallerExpectsSomeAudio(
int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
caller_audio_expectation_ = kExpectSomeFrames;
caller_audio_frames_expected_ = expected_audio_frames;
}
void CallerExpectsNoAudio() {
caller_audio_expectation_ = kExpectNoFrames;
caller_audio_frames_expected_ = 0;
}
// Caller's video functions.
void CallerExpectsSomeVideo(
int expected_video_frames = kDefaultExpectedVideoFrameCount) {
caller_video_expectation_ = kExpectSomeFrames;
caller_video_frames_expected_ = expected_video_frames;
}
void CallerExpectsNoVideo() {
caller_video_expectation_ = kExpectNoFrames;
caller_video_frames_expected_ = 0;
}
// Callee's audio functions.
void CalleeExpectsSomeAudio(
int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
callee_audio_expectation_ = kExpectSomeFrames;
callee_audio_frames_expected_ = expected_audio_frames;
}
void CalleeExpectsNoAudio() {
callee_audio_expectation_ = kExpectNoFrames;
callee_audio_frames_expected_ = 0;
}
// Callee's video functions.
void CalleeExpectsSomeVideo(
int expected_video_frames = kDefaultExpectedVideoFrameCount) {
callee_video_expectation_ = kExpectSomeFrames;
callee_video_frames_expected_ = expected_video_frames;
}
void CalleeExpectsNoVideo() {
callee_video_expectation_ = kExpectNoFrames;
callee_video_frames_expected_ = 0;
}
ExpectFrames caller_audio_expectation_ = kNoExpectation;
ExpectFrames caller_video_expectation_ = kNoExpectation;
ExpectFrames callee_audio_expectation_ = kNoExpectation;
ExpectFrames callee_video_expectation_ = kNoExpectation;
int caller_audio_frames_expected_ = 0;
int caller_video_frames_expected_ = 0;
int callee_audio_frames_expected_ = 0;
int callee_video_frames_expected_ = 0;
};
// Tests two PeerConnections connecting to each other end-to-end, using a
// virtual network, fake A/V capture and fake encoder/decoders. The
// PeerConnections share the threads/socket servers, but use separate versions
// of everything else (including "PeerConnectionFactory"s).
class PeerConnectionIntegrationBaseTest : public testing::Test {
public:
explicit PeerConnectionIntegrationBaseTest(SdpSemantics sdp_semantics)
: sdp_semantics_(sdp_semantics),
ss_(new rtc::VirtualSocketServer()),
fss_(new rtc::FirewallSocketServer(ss_.get())),
network_thread_(new rtc::Thread(fss_.get())),
worker_thread_(rtc::Thread::Create()),
loopback_media_transports_(network_thread_.get()) {
network_thread_->SetName("PCNetworkThread", this);
worker_thread_->SetName("PCWorkerThread", this);
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
webrtc::metrics::Reset();
}
~PeerConnectionIntegrationBaseTest() {
// The PeerConnections should deleted before the TurnCustomizers.
// A TurnPort is created with a raw pointer to a TurnCustomizer. The
// TurnPort has the same lifetime as the PeerConnection, so it's expected
// that the TurnCustomizer outlives the life of the PeerConnection or else
// when Send() is called it will hit a seg fault.
if (caller_) {
caller_->set_signaling_message_receiver(nullptr);
delete SetCallerPcWrapperAndReturnCurrent(nullptr);
}
if (callee_) {
callee_->set_signaling_message_receiver(nullptr);
delete SetCalleePcWrapperAndReturnCurrent(nullptr);
}
// If turn servers were created for the test they need to be destroyed on
// the network thread.
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
turn_servers_.clear();
turn_customizers_.clear();
});
}
bool SignalingStateStable() {
return caller_->SignalingStateStable() && callee_->SignalingStateStable();
}
bool DtlsConnected() {
// TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
// are connected. This is an important distinction. Once we have separate
// ICE and DTLS state, this check needs to use the DTLS state.
return (callee()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionConnected ||
callee()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
(caller()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionConnected ||
caller()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionCompleted);
}
// When |event_log_factory| is null, the default implementation of the event
// log factory will be used.
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWrapper(
const std::string& debug_name,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies,
std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) {
RTCConfiguration modified_config;
if (config) {
modified_config = *config;
}
modified_config.sdp_semantics = sdp_semantics_;
if (!dependencies.cert_generator) {
dependencies.cert_generator =
absl::make_unique<FakeRTCCertificateGenerator>();
}
std::unique_ptr<PeerConnectionWrapper> client(
new PeerConnectionWrapper(debug_name));
if (!client->Init(options, &modified_config, std::move(dependencies),
network_thread_.get(), worker_thread_.get(),
std::move(event_log_factory),
std::move(media_transport_factory))) {
return nullptr;
}
return client;
}
std::unique_ptr<PeerConnectionWrapper>
CreatePeerConnectionWrapperWithFakeRtcEventLog(
const std::string& debug_name,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
webrtc::PeerConnectionDependencies dependencies) {
std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory(
new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current()));
return CreatePeerConnectionWrapper(debug_name, options, config,
std::move(dependencies),
std::move(event_log_factory),
/*media_transport_factory=*/nullptr);
}
bool CreatePeerConnectionWrappers() {
return CreatePeerConnectionWrappersWithConfig(
PeerConnectionInterface::RTCConfiguration(),
PeerConnectionInterface::RTCConfiguration());
}
bool CreatePeerConnectionWrappersWithSdpSemantics(
SdpSemantics caller_semantics,
SdpSemantics callee_semantics) {
// Can't specify the sdp_semantics in the passed-in configuration since it
// will be overwritten by CreatePeerConnectionWrapper with whatever is
// stored in sdp_semantics_. So get around this by modifying the instance
// variable before calling CreatePeerConnectionWrapper for the caller and
// callee PeerConnections.
SdpSemantics original_semantics = sdp_semantics_;
sdp_semantics_ = caller_semantics;
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
nullptr, /*media_transport_factory=*/nullptr);
sdp_semantics_ = callee_semantics;
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
nullptr, /*media_transport_factory=*/nullptr);
sdp_semantics_ = original_semantics;
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithConfig(
const PeerConnectionInterface::RTCConfiguration& caller_config,
const PeerConnectionInterface::RTCConfiguration& callee_config) {
caller_ = CreatePeerConnectionWrapper(
"Caller", nullptr, &caller_config,
webrtc::PeerConnectionDependencies(nullptr), nullptr,
/*media_transport_factory=*/nullptr);
callee_ = CreatePeerConnectionWrapper(
"Callee", nullptr, &callee_config,
webrtc::PeerConnectionDependencies(nullptr), nullptr,
/*media_transport_factory=*/nullptr);
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
const PeerConnectionInterface::RTCConfiguration& caller_config,
const PeerConnectionInterface::RTCConfiguration& callee_config,
std::unique_ptr<webrtc::MediaTransportFactory> caller_factory,
std::unique_ptr<webrtc::MediaTransportFactory> callee_factory) {
caller_ =
CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
webrtc::PeerConnectionDependencies(nullptr),
nullptr, std::move(caller_factory));
callee_ =
CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
webrtc::PeerConnectionDependencies(nullptr),
nullptr, std::move(callee_factory));
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithConfigAndDeps(
const PeerConnectionInterface::RTCConfiguration& caller_config,
webrtc::PeerConnectionDependencies caller_dependencies,
const PeerConnectionInterface::RTCConfiguration& callee_config,
webrtc::PeerConnectionDependencies callee_dependencies) {
caller_ =
CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
std::move(caller_dependencies), nullptr,
/*media_transport_factory=*/nullptr);
callee_ =
CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
std::move(callee_dependencies), nullptr,
/*media_transport_factory=*/nullptr);
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithOptions(
const PeerConnectionFactory::Options& caller_options,
const PeerConnectionFactory::Options& callee_options) {
caller_ = CreatePeerConnectionWrapper(
"Caller", &caller_options, nullptr,
webrtc::PeerConnectionDependencies(nullptr), nullptr,
/*media_transport_factory=*/nullptr);
callee_ = CreatePeerConnectionWrapper(
"Callee", &callee_options, nullptr,
webrtc::PeerConnectionDependencies(nullptr), nullptr,
/*media_transport_factory=*/nullptr);
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithFakeRtcEventLog() {
PeerConnectionInterface::RTCConfiguration default_config;
caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
"Caller", nullptr, &default_config,
webrtc::PeerConnectionDependencies(nullptr));
callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
"Callee", nullptr, &default_config,
webrtc::PeerConnectionDependencies(nullptr));
return caller_ && callee_;
}
std::unique_ptr<PeerConnectionWrapper>
CreatePeerConnectionWrapperWithAlternateKey() {
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
cert_generator->use_alternate_key();
webrtc::PeerConnectionDependencies dependencies(nullptr);
dependencies.cert_generator = std::move(cert_generator);
return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
std::move(dependencies), nullptr,
/*media_transport_factory=*/nullptr);
}
cricket::TestTurnServer* CreateTurnServer(
rtc::SocketAddress internal_address,
rtc::SocketAddress external_address,
cricket::ProtocolType type = cricket::ProtocolType::PROTO_UDP,
const std::string& common_name = "test turn server") {
rtc::Thread* thread = network_thread();
std::unique_ptr<cricket::TestTurnServer> turn_server =
network_thread()->Invoke<std::unique_ptr<cricket::TestTurnServer>>(
RTC_FROM_HERE,
[thread, internal_address, external_address, type, common_name] {
return absl::make_unique<cricket::TestTurnServer>(
thread, internal_address, external_address, type,
/*ignore_bad_certs=*/true, common_name);
});
turn_servers_.push_back(std::move(turn_server));
// Interactions with the turn server should be done on the network thread.
return turn_servers_.back().get();
}
cricket::TestTurnCustomizer* CreateTurnCustomizer() {
std::unique_ptr<cricket::TestTurnCustomizer> turn_customizer =
network_thread()->Invoke<std::unique_ptr<cricket::TestTurnCustomizer>>(
RTC_FROM_HERE,
[] { return absl::make_unique<cricket::TestTurnCustomizer>(); });
turn_customizers_.push_back(std::move(turn_customizer));
// Interactions with the turn customizer should be done on the network
// thread.
return turn_customizers_.back().get();
}
// Checks that the function counters for a TestTurnCustomizer are greater than
// 0.
void ExpectTurnCustomizerCountersIncremented(
cricket::TestTurnCustomizer* turn_customizer) {
unsigned int allow_channel_data_counter =
network_thread()->Invoke<unsigned int>(
RTC_FROM_HERE, [turn_customizer] {
return turn_customizer->allow_channel_data_cnt_;
});
EXPECT_GT(allow_channel_data_counter, 0u);
unsigned int modify_counter = network_thread()->Invoke<unsigned int>(
RTC_FROM_HERE,
[turn_customizer] { return turn_customizer->modify_cnt_; });
EXPECT_GT(modify_counter, 0u);
}
// Once called, SDP blobs and ICE candidates will be automatically signaled
// between PeerConnections.
void ConnectFakeSignaling() {
caller_->set_signaling_message_receiver(callee_.get());
callee_->set_signaling_message_receiver(caller_.get());
}
// Once called, SDP blobs will be automatically signaled between
// PeerConnections. Note that ICE candidates will not be signaled unless they
// are in the exchanged SDP blobs.
void ConnectFakeSignalingForSdpOnly() {
ConnectFakeSignaling();
SetSignalIceCandidates(false);
}
void SetSignalingDelayMs(int delay_ms) {
caller_->set_signaling_delay_ms(delay_ms);
callee_->set_signaling_delay_ms(delay_ms);
}
void SetSignalIceCandidates(bool signal) {
caller_->set_signal_ice_candidates(signal);
callee_->set_signal_ice_candidates(signal);
}
// Messages may get lost on the unreliable DataChannel, so we send multiple
// times to avoid test flakiness.
void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc,
const std::string& data,
int retries) {
for (int i = 0; i < retries; ++i) {
dc->Send(DataBuffer(data));
}
}
rtc::Thread* network_thread() { return network_thread_.get(); }
rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
webrtc::MediaTransportPair* loopback_media_transports() {
return &loopback_media_transports_;
}
PeerConnectionWrapper* caller() { return caller_.get(); }
// Set the |caller_| to the |wrapper| passed in and return the
// original |caller_|.
PeerConnectionWrapper* SetCallerPcWrapperAndReturnCurrent(
PeerConnectionWrapper* wrapper) {
PeerConnectionWrapper* old = caller_.release();
caller_.reset(wrapper);
return old;
}
PeerConnectionWrapper* callee() { return callee_.get(); }
// Set the |callee_| to the |wrapper| passed in and return the
// original |callee_|.
PeerConnectionWrapper* SetCalleePcWrapperAndReturnCurrent(
PeerConnectionWrapper* wrapper) {
PeerConnectionWrapper* old = callee_.release();
callee_.reset(wrapper);
return old;
}
void SetPortAllocatorFlags(uint32_t caller_flags, uint32_t callee_flags) {
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::set_flags,
caller()->port_allocator(), caller_flags));
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::set_flags,
callee()->port_allocator(), callee_flags));
}
rtc::FirewallSocketServer* firewall() const { return fss_.get(); }
// Expects the provided number of new frames to be received within
// kMaxWaitForFramesMs. The new expected frames are specified in
// |media_expectations|. Returns false if any of the expectations were
// not met.
bool ExpectNewFrames(const MediaExpectations& media_expectations) {
// First initialize the expected frame counts based upon the current
// frame count.
int total_caller_audio_frames_expected = caller()->audio_frames_received();
if (media_expectations.caller_audio_expectation_ ==
MediaExpectations::kExpectSomeFrames) {
total_caller_audio_frames_expected +=
media_expectations.caller_audio_frames_expected_;
}
int total_caller_video_frames_expected =
caller()->min_video_frames_received_per_track();
if (media_expectations.caller_video_expectation_ ==
MediaExpectations::kExpectSomeFrames) {
total_caller_video_frames_expected +=
media_expectations.caller_video_frames_expected_;
}
int total_callee_audio_frames_expected = callee()->audio_frames_received();
if (media_expectations.callee_audio_expectation_ ==
MediaExpectations::kExpectSomeFrames) {
total_callee_audio_frames_expected +=
media_expectations.callee_audio_frames_expected_;
}
int total_callee_video_frames_expected =
callee()->min_video_frames_received_per_track();
if (media_expectations.callee_video_expectation_ ==
MediaExpectations::kExpectSomeFrames) {
total_callee_video_frames_expected +=
media_expectations.callee_video_frames_expected_;
}
// Wait for the expected frames.
EXPECT_TRUE_WAIT(caller()->audio_frames_received() >=
total_caller_audio_frames_expected &&
caller()->min_video_frames_received_per_track() >=
total_caller_video_frames_expected &&
callee()->audio_frames_received() >=
total_callee_audio_frames_expected &&
callee()->min_video_frames_received_per_track() >=
total_callee_video_frames_expected,
kMaxWaitForFramesMs);
bool expectations_correct =
caller()->audio_frames_received() >=
total_caller_audio_frames_expected &&
caller()->min_video_frames_received_per_track() >=
total_caller_video_frames_expected &&
callee()->audio_frames_received() >=
total_callee_audio_frames_expected &&
callee()->min_video_frames_received_per_track() >=
total_callee_video_frames_expected;
// After the combined wait, print out a more detailed message upon
// failure.
EXPECT_GE(caller()->audio_frames_received(),
total_caller_audio_frames_expected);
EXPECT_GE(caller()->min_video_frames_received_per_track(),
total_caller_video_frames_expected);
EXPECT_GE(callee()->audio_frames_received(),
total_callee_audio_frames_expected);
EXPECT_GE(callee()->min_video_frames_received_per_track(),
total_callee_video_frames_expected);
// We want to make sure nothing unexpected was received.
if (media_expectations.caller_audio_expectation_ ==
MediaExpectations::kExpectNoFrames) {
EXPECT_EQ(caller()->audio_frames_received(),
total_caller_audio_frames_expected);
if (caller()->audio_frames_received() !=
total_caller_audio_frames_expected) {
expectations_correct = false;
}
}
if (media_expectations.caller_video_expectation_ ==
MediaExpectations::kExpectNoFrames) {
EXPECT_EQ(caller()->min_video_frames_received_per_track(),
total_caller_video_frames_expected);
if (caller()->min_video_frames_received_per_track() !=
total_caller_video_frames_expected) {
expectations_correct = false;
}
}
if (media_expectations.callee_audio_expectation_ ==
MediaExpectations::kExpectNoFrames) {
EXPECT_EQ(callee()->audio_frames_received(),
total_callee_audio_frames_expected);
if (callee()->audio_frames_received() !=
total_callee_audio_frames_expected) {
expectations_correct = false;
}
}
if (media_expectations.callee_video_expectation_ ==
MediaExpectations::kExpectNoFrames) {
EXPECT_EQ(callee()->min_video_frames_received_per_track(),
total_callee_video_frames_expected);
if (callee()->min_video_frames_received_per_track() !=
total_callee_video_frames_expected) {
expectations_correct = false;
}
}
return expectations_correct;
}
void TestNegotiatedCipherSuite(
const PeerConnectionFactory::Options& caller_options,
const PeerConnectionFactory::Options& callee_options,
int expected_cipher_suite) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
callee_options));
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
expected_cipher_suite));
}
void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
bool remote_gcm_enabled,
int expected_cipher_suite) {
PeerConnectionFactory::Options caller_options;
caller_options.crypto_options.srtp.enable_gcm_crypto_suites =
local_gcm_enabled;
PeerConnectionFactory::Options callee_options;
callee_options.crypto_options.srtp.enable_gcm_crypto_suites =
remote_gcm_enabled;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
}
protected:
SdpSemantics sdp_semantics_;
private:
// |ss_| is used by |network_thread_| so it must be destroyed later.
std::unique_ptr<rtc::VirtualSocketServer> ss_;
std::unique_ptr<rtc::FirewallSocketServer> fss_;
// |network_thread_| and |worker_thread_| are used by both
// |caller_| and |callee_| so they must be destroyed
// later.
std::unique_ptr<rtc::Thread> network_thread_;
std::unique_ptr<rtc::Thread> worker_thread_;
// The turn servers and turn customizers should be accessed & deleted on the
// network thread to avoid a race with the socket read/write that occurs
// on the network thread.
std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_;
webrtc::MediaTransportPair loopback_media_transports_;
std::unique_ptr<PeerConnectionWrapper> caller_;
std::unique_ptr<PeerConnectionWrapper> callee_;
};
class PeerConnectionIntegrationTest
: public PeerConnectionIntegrationBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionIntegrationTest()
: PeerConnectionIntegrationBaseTest(GetParam()) {}
};
class PeerConnectionIntegrationTestPlanB
: public PeerConnectionIntegrationBaseTest {
protected:
PeerConnectionIntegrationTestPlanB()
: PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB) {}
};
class PeerConnectionIntegrationTestUnifiedPlan
: public PeerConnectionIntegrationBaseTest {
protected:
PeerConnectionIntegrationTestUnifiedPlan()
: PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
};
// Test the OnFirstPacketReceived callback from audio/video RtpReceivers. This
// includes testing that the callback is invoked if an observer is connected
// after the first packet has already been received.
TEST_P(PeerConnectionIntegrationTest,
RtpReceiverObserverOnFirstPacketReceived) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Start offer/answer exchange and wait for it to complete.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Should be one receiver each for audio/video.
EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
// Wait for all "first packet received" callbacks to be fired.
EXPECT_TRUE_WAIT(
std::all_of(caller()->rtp_receiver_observers().begin(),
caller()->rtp_receiver_observers().end(),
[](const std::unique_ptr<MockRtpReceiverObserver>& o) {
return o->first_packet_received();
}),
kMaxWaitForFramesMs);
EXPECT_TRUE_WAIT(
std::all_of(callee()->rtp_receiver_observers().begin(),
callee()->rtp_receiver_observers().end(),
[](const std::unique_ptr<MockRtpReceiverObserver>& o) {
return o->first_packet_received();
}),
kMaxWaitForFramesMs);
// If new observers are set after the first packet was already received, the
// callback should still be invoked.
caller()->ResetRtpReceiverObservers();
callee()->ResetRtpReceiverObservers();
EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
EXPECT_TRUE(
std::all_of(caller()->rtp_receiver_observers().begin(),
caller()->rtp_receiver_observers().end(),
[](const std::unique_ptr<MockRtpReceiverObserver>& o) {
return o->first_packet_received();
}));
EXPECT_TRUE(
std::all_of(callee()->rtp_receiver_observers().begin(),
callee()->rtp_receiver_observers().end(),
[](const std::unique_ptr<MockRtpReceiverObserver>& o) {
return o->first_packet_received();
}));
}
class DummyDtmfObserver : public DtmfSenderObserverInterface {
public:
DummyDtmfObserver() : completed_(false) {}
// Implements DtmfSenderObserverInterface.
void OnToneChange(const std::string& tone) override {
tones_.push_back(tone);
if (tone.empty()) {
completed_ = true;
}
}
const std::vector<std::string>& tones() const { return tones_; }
bool completed() const { return completed_; }
private:
bool completed_;
std::vector<std::string> tones_;
};
// Assumes |sender| already has an audio track added and the offer/answer
// exchange is done.
void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender,
PeerConnectionWrapper* receiver) {
// We should be able to get a DTMF sender from the local sender.
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender =
sender->pc()->GetSenders().at(0)->GetDtmfSender();
ASSERT_TRUE(dtmf_sender);
DummyDtmfObserver observer;
dtmf_sender->RegisterObserver(&observer);
// Test the DtmfSender object just created.
EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));
EXPECT_TRUE_WAIT(observer.completed(), kDefaultTimeout);
std::vector<std::string> tones = {"1", "a", ""};
EXPECT_EQ(tones, observer.tones());
dtmf_sender->UnregisterObserver();
// TODO(deadbeef): Verify the tones were actually received end-to-end.
}
// Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each
// direction).
TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Only need audio for DTMF.
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// DTLS must finish before the DTMF sender can be used reliably.
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
TestDtmfFromSenderToReceiver(caller(), callee());
TestDtmfFromSenderToReceiver(callee(), caller());
}
// Basic end-to-end test, verifying media can be encoded/transmitted/decoded
// between two connections, using DTLS-SRTP.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
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_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
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_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
}
// Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
// certificate once the DTLS handshake has finished.
TEST_P(PeerConnectionIntegrationTest,
GetRemoteAudioSSLCertificateReturnsExchangedCertificate) {
auto GetRemoteAudioSSLCertificate = [](PeerConnectionWrapper* wrapper) {
auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
return pc->GetRemoteAudioSSLCertificate();
};
auto GetRemoteAudioSSLCertChain = [](PeerConnectionWrapper* wrapper) {
auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
return pc->GetRemoteAudioSSLCertChain();
};
auto caller_cert = rtc::RTCCertificate::FromPEM(kRsaPems[0]);
auto callee_cert = rtc::RTCCertificate::FromPEM(kRsaPems[1]);
// Configure each side with a known certificate so they can be compared later.
PeerConnectionInterface::RTCConfiguration caller_config;
caller_config.enable_dtls_srtp.emplace(true);
caller_config.certificates.push_back(caller_cert);
PeerConnectionInterface::RTCConfiguration callee_config;
callee_config.enable_dtls_srtp.emplace(true);
callee_config.certificates.push_back(callee_cert);
ASSERT_TRUE(
CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));
ConnectFakeSignaling();
// When first initialized, there should not be a remote SSL certificate (and
// calling this method should not crash).
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller()));
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee()));
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(caller()));
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(callee()));
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
// Once DTLS has been connected, each side should return the other's SSL
// certificate when calling GetRemoteAudioSSLCertificate.
auto caller_remote_cert = GetRemoteAudioSSLCertificate(caller());
ASSERT_TRUE(caller_remote_cert);
EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
caller_remote_cert->ToPEMString());
auto callee_remote_cert = GetRemoteAudioSSLCertificate(callee());
ASSERT_TRUE(callee_remote_cert);
EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
callee_remote_cert->ToPEMString());
auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller());
ASSERT_TRUE(caller_remote_cert_chain);
ASSERT_EQ(1U, caller_remote_cert_chain->GetSize());
auto remote_cert = &caller_remote_cert_chain->Get(0);
EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
remote_cert->ToPEMString());
auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee());
ASSERT_TRUE(callee_remote_cert_chain);
ASSERT_EQ(1U, callee_remote_cert_chain->GetSize());
remote_cert = &callee_remote_cert_chain->Get(0);
EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
remote_cert->ToPEMString());
}
// This test sets up a call between two parties with a source resolution of
// 1280x720 and verifies that a 16:9 aspect ratio is received.
TEST_P(PeerConnectionIntegrationTest,
Send1280By720ResolutionAndReceive16To9AspectRatio) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add video tracks with 16:9 aspect ratio, size 1280 x 720.
webrtc::FakePeriodicVideoSource::Config config;
config.width = 1280;
config.height = 720;
config.timestamp_offset_ms = rtc::TimeMillis();
caller()->AddTrack(caller()->CreateLocalVideoTrackWithConfig(config));
callee()->AddTrack(callee()->CreateLocalVideoTrackWithConfig(config));
// Do normal offer/answer and wait for at least one frame to be received in
// each direction.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
callee()->min_video_frames_received_per_track() > 0,
kMaxWaitForFramesMs);
// Check rendered aspect ratio.
EXPECT_EQ(16.0 / 9, caller()->local_rendered_aspect_ratio());
EXPECT_EQ(16.0 / 9, caller()->rendered_aspect_ratio());
EXPECT_EQ(16.0 / 9, callee()->local_rendered_aspect_ratio());
EXPECT_EQ(16.0 / 9, callee()->rendered_aspect_ratio());
}
// This test sets up an one-way call, with media only from caller to
// callee.
TEST_P(PeerConnectionIntegrationTest, OneWayMediaCall) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudioAndVideo();
media_expectations.CallerExpectsNoAudio();
media_expectations.CallerExpectsNoVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This test sets up a audio call initially, with the callee rejecting video
// initially. Then later the callee decides to upgrade to audio/video, and
// initiates a new offer/answer exchange.
TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Initially, offer an audio/video stream from the caller, but refuse to
// send/receive video on the callee side.
caller()->AddAudioVideoTracks();
callee()->AddAudioTrack();
if (sdp_semantics_ == SdpSemantics::kPlanB) {
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_video = 0;
callee()->SetOfferAnswerOptions(options);
} else {
callee()->SetRemoteOfferHandler([this] {
callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
});
}
// Do offer/answer and make sure audio is still received end-to-end.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
{
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudio();
media_expectations.ExpectNoVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Sanity check that the callee's description has a rejected video section.
ASSERT_NE(nullptr, callee()->pc()->local_description());
const ContentInfo* callee_video_content =
GetFirstVideoContent(callee()->pc()->local_description()->description());
ASSERT_NE(nullptr, callee_video_content);
EXPECT_TRUE(callee_video_content->rejected);
// Now negotiate with video and ensure negotiation succeeds, with video
// frames and additional audio frames being received.
callee()->AddVideoTrack();
if (sdp_semantics_ == SdpSemantics::kPlanB) {
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_video = 1;
callee()->SetOfferAnswerOptions(options);
} else {
callee()->SetRemoteOfferHandler(nullptr);
caller()->SetRemoteOfferHandler([this] {
// The caller creates a new transceiver to receive video on when receiving
// the offer, but by default it is send only.
auto transceivers = caller()->pc()->GetTransceivers();
ASSERT_EQ(3U, transceivers.size());
ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
transceivers[2]->receiver()->media_type());
transceivers[2]->sender()->SetTrack(caller()->CreateLocalVideoTrack());
transceivers[2]->SetDirection(RtpTransceiverDirection::kSendRecv);
});
}
callee()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
{
// Expect additional audio frames to be received after the upgrade.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
}
// Simpler than the above test; just add an audio track to an established
// video-only connection.
TEST_P(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do initial offer/answer with just a video track.
caller()->AddVideoTrack();
callee()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Now add an audio track and do another offer/answer.
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure both audio and video frames are received end-to-end.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This test sets up a call that's transferred to a new caller with a different
// DTLS fingerprint.
TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Keep the original peer around which will still send packets to the
// receiving client. These SRTP packets will be dropped.
std::unique_ptr<PeerConnectionWrapper> original_peer(
SetCallerPcWrapperAndReturnCurrent(
CreatePeerConnectionWrapperWithAlternateKey().release()));
// TODO(deadbeef): Why do we call Close here? That goes against the comment
// directly above.
original_peer->pc()->Close();
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for some additional frames to be transmitted end-to-end.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This test sets up a call that's transferred to a new callee with a different
// DTLS fingerprint.
TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Keep the original peer around which will still send packets to the
// receiving client. These SRTP packets will be dropped.
std::unique_ptr<PeerConnectionWrapper> original_peer(
SetCalleePcWrapperAndReturnCurrent(
CreatePeerConnectionWrapperWithAlternateKey().release()));
// TODO(deadbeef): Why do we call Close here? That goes against the comment
// directly above.
original_peer->pc()->Close();
ConnectFakeSignaling();
callee()->AddAudioVideoTracks();
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for some additional frames to be transmitted end-to-end.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This test sets up a non-bundled call and negotiates bundling at the same
// time as starting an ICE restart. When bundling is in effect in the restart,
// the DTLS-SRTP context should be successfully reset.
TEST_P(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Remove the bundle group from the SDP received by the callee.
callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
desc->RemoveGroupByName("BUNDLE");
});
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
{
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Now stop removing the BUNDLE group, and trigger an ICE restart.
callee()->SetReceivedSdpMunger(nullptr);
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Expect additional frames to be received after the ICE restart.
{
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
}
// Test CVO (Coordination of Video Orientation). If a video source is rotated
// and both peers support the CVO RTP header extension, the actual video frames
// don't need to be encoded in different resolutions, since the rotation is
// communicated through the RTP header extension.
TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add rotated video tracks.
caller()->AddTrack(
caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
callee()->AddTrack(
callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
// Wait for video frames to be received by both sides.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
callee()->min_video_frames_received_per_track() > 0,
kMaxWaitForFramesMs);
// Ensure that the aspect ratio is unmodified.
// TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
// not just assumed.
EXPECT_EQ(4.0 / 3, caller()->local_rendered_aspect_ratio());
EXPECT_EQ(4.0 / 3, caller()->rendered_aspect_ratio());
EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio());
EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio());
// Ensure that the CVO bits were surfaced to the renderer.
EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation());
EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation());
}
// Test that when the CVO extension isn't supported, video is rotated the
// old-fashioned way, by encoding rotated frames.
TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add rotated video tracks.
caller()->AddTrack(
caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
callee()->AddTrack(
callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
// Remove the CVO extension from the offered SDP.
callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
cricket::VideoContentDescription* video =
GetFirstVideoContentDescription(desc);
video->ClearRtpHeaderExtensions();
});
// Wait for video frames to be received by both sides.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
callee()->min_video_frames_received_per_track() > 0,
kMaxWaitForFramesMs);
// Expect that the aspect ratio is inversed to account for the 90/270 degree
// rotation.
// TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
// not just assumed.
EXPECT_EQ(3.0 / 4, caller()->local_rendered_aspect_ratio());
EXPECT_EQ(3.0 / 4, caller()->rendered_aspect_ratio());
EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio());
EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio());
// Expect that each endpoint is unaware of the rotation of the other endpoint.
EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation());
EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation());
}
// Test that if the answerer rejects the audio m= section, no audio is sent or
// received, but video still can be.
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
if (sdp_semantics_ == SdpSemantics::kPlanB) {
// Only add video track for callee, and set offer_to_receive_audio to 0, so
// it will reject the audio m= section completely.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 0;
callee()->SetOfferAnswerOptions(options);
} else {
// Stopping the audio RtpTransceiver will cause the media section to be
// rejected in the answer.
callee()->SetRemoteOfferHandler([this] {
callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)->Stop();
});
}
callee()->AddTrack(callee()->CreateLocalVideoTrack());
// Do offer/answer and wait for successful end-to-end video frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalVideo();
media_expectations.ExpectNoAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// Sanity check that the callee's description has a rejected audio section.
ASSERT_NE(nullptr, callee()->pc()->local_description());
const ContentInfo* callee_audio_content =
GetFirstAudioContent(callee()->pc()->local_description()->description());
ASSERT_NE(nullptr, callee_audio_content);
EXPECT_TRUE(callee_audio_content->rejected);
if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
// The caller's transceiver should have stopped after receiving the answer.
EXPECT_TRUE(caller()
->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
->stopped());
}
}
// Test that if the answerer rejects the video m= section, no video is sent or
// received, but audio still can be.
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
if (sdp_semantics_ == SdpSemantics::kPlanB) {
// Only add audio track for callee, and set offer_to_receive_video to 0, so
// it will reject the video m= section completely.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_video = 0;
callee()->SetOfferAnswerOptions(options);
} else {
// Stopping the video RtpTransceiver will cause the media section to be
// rejected in the answer.
callee()->SetRemoteOfferHandler([this] {
callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
});
}
callee()->AddTrack(callee()->CreateLocalAudioTrack());
// Do offer/answer and wait for successful end-to-end audio frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudio();
media_expectations.ExpectNoVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// Sanity check that the callee's description has a rejected video section.
ASSERT_NE(nullptr, callee()->pc()->local_description());
const ContentInfo* callee_video_content =
GetFirstVideoContent(callee()->pc()->local_description()->description());
ASSERT_NE(nullptr, callee_video_content);
EXPECT_TRUE(callee_video_content->rejected);
if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
// The caller's transceiver should have stopped after receiving the answer.
EXPECT_TRUE(caller()
->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
->stopped());
}
}
// Test that if the answerer rejects both audio and video m= sections, nothing
// bad happens.
// TODO(deadbeef): Test that a data channel still works. Currently this doesn't
// test anything but the fact that negotiation succeeds, which doesn't mean
// much.
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
if (sdp_semantics_ == SdpSemantics::kPlanB) {
// Don't give the callee any tracks, and set offer_to_receive_X to 0, so it
// will reject both audio and video m= sections.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 0;
options.offer_to_receive_video = 0;
callee()->SetOfferAnswerOptions(options);
} else {
callee()->SetRemoteOfferHandler([this] {
// Stopping all transceivers will cause all media sections to be rejected.
for (auto transceiver : callee()->pc()->GetTransceivers()) {
transceiver->Stop();
}
});
}
// Do offer/answer and wait for stable signaling state.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Sanity check that the callee's description has rejected m= sections.
ASSERT_NE(nullptr, callee()->pc()->local_description());
const ContentInfo* callee_audio_content =
GetFirstAudioContent(callee()->pc()->local_description()->description());
ASSERT_NE(nullptr, callee_audio_content);
EXPECT_TRUE(callee_audio_content->rejected);
const ContentInfo* callee_video_content =
GetFirstVideoContent(callee()->pc()->local_description()->description());
ASSERT_NE(nullptr, callee_video_content);
EXPECT_TRUE(callee_video_content->rejected);
}
// This test sets up an audio and video call between two parties. After the
// call runs for a while, the caller sends an updated offer with video being
// rejected. Once the re-negotiation is done, the video flow should stop and
// the audio flow should continue.
TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
{
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Renegotiate, rejecting the video m= section.
if (sdp_semantics_ == SdpSemantics::kPlanB) {
caller()->SetGeneratedSdpMunger(
[](cricket::SessionDescription* description) {
for (cricket::ContentInfo& content : description->contents()) {
if (cricket::IsVideoContent(&content)) {
content.rejected = true;
}
}
});
} else {
caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
}
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
// Sanity check that the caller's description has a rejected video section.
ASSERT_NE(nullptr, caller()->pc()->local_description());
const ContentInfo* caller_video_content =
GetFirstVideoContent(caller()->pc()->local_description()->description());
ASSERT_NE(nullptr, caller_video_content);
EXPECT_TRUE(caller_video_content->rejected);
// Wait for some additional audio frames to be received.
{
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudio();
media_expectations.ExpectNoVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
}
// Do one offer/answer with audio, another that disables it (rejecting the m=
// section), and another that re-enables it. Regression test for:
// bugs.webrtc.org/6023
TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add audio track, do normal offer/answer.
rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
caller()->CreateLocalAudioTrack();
rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Remove audio track, and set offer_to_receive_audio to false to cause the
// m= section to be completely disabled, not just "recvonly".
caller()->pc()->RemoveTrack(sender);
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 0;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Add the audio track again, expecting negotiation to succeed and frames to
// flow.
sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
options.offer_to_receive_audio = 1;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio();
EXPECT_TRUE(ExpectNewFrames(media_expectations));
}
// Basic end-to-end test, but without SSRC/MSID signaling. This functionality
// is needed to support legacy endpoints.
// TODO(deadbeef): When we support the MID extension and demuxing on MID, also
// add a test for an end-to-end test without MID signaling either (basically,
// the minimum acceptable SDP).
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add audio and video, testing that packets can be demuxed on payload type.
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Remove SSRCs and MSIDs from the received offer SDP.
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Basic end-to-end test, without SSRC signaling. This means that the track
// was created properly and frames are delivered when the MSIDs are communicated
// with a=msid lines and no a=ssrc lines.
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
EndToEndCallWithoutSsrcSignaling) {
const char kStreamId[] = "streamId";
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add just audio tracks.
caller()->AddTrack(caller()->CreateLocalAudioTrack(), {kStreamId});
callee()->AddAudioTrack();
// Remove SSRCs from the received offer SDP.
callee()->SetReceivedSdpMunger(RemoveSsrcsAndKeepMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Tests that video flows between multiple video tracks when SSRCs are not
// signaled. This exercises the MID RTP header extension which is needed to
// demux the incoming video tracks.
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
EndToEndCallWithTwoVideoTracksAndNoSignaledSsrc) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddVideoTrack();
caller()->AddVideoTrack();
callee()->AddVideoTrack();
callee()->AddVideoTrack();
caller()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
callee()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_EQ(2u, caller()->pc()->GetReceivers().size());
ASSERT_EQ(2u, callee()->pc()->GetReceivers().size());
// Expect video to be received in both directions on both tracks.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalVideo();
EXPECT_TRUE(ExpectNewFrames(media_expectations));
}
TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLinePresent) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
auto callee_receivers = callee()->pc()->GetReceivers();
ASSERT_EQ(2u, callee_receivers.size());
EXPECT_TRUE(callee_receivers[0]->stream_ids().empty());
EXPECT_TRUE(callee_receivers[1]->stream_ids().empty());
}
TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLineMissing) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
caller()->AddVideoTrack();
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
auto callee_receivers = callee()->pc()->GetReceivers();
ASSERT_EQ(2u, callee_receivers.size());
ASSERT_EQ(1u, callee_receivers[0]->stream_ids().size());
ASSERT_EQ(1u, callee_receivers[1]->stream_ids().size());
EXPECT_EQ(callee_receivers[0]->stream_ids()[0],
callee_receivers[1]->stream_ids()[0]);
EXPECT_EQ(callee_receivers[0]->streams()[0],
callee_receivers[1]->streams()[0]);
}
// Test that if two video tracks are sent (from caller to callee, in this test),
// they're transmitted correctly end-to-end.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add one audio/video stream, and one video-only stream.
caller()->AddAudioVideoTracks();
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_EQ(3u, callee()->pc()->GetReceivers().size());
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) {
bool first = true;
for (cricket::ContentInfo& content : desc->contents()) {
if (first) {
first = false;
continue;
}
content.bundle_only = true;
}
first = true;
for (cricket::TransportInfo& transport : desc->transport_infos()) {
if (first) {
first = false;
continue;
}
transport.description.ice_ufrag.clear();
transport.description.ice_pwd.clear();
transport.description.connection_role = cricket::CONNECTIONROLE_NONE;
transport.description.identity_fingerprint.reset(nullptr);
}
}
// Test that if applying a true "max bundle" offer, which uses ports of 0,
// "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and
// "a=ice-pwd" for all but the audio "m=" section, negotiation still completes
// successfully and media flows.
// TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works.
// TODO(deadbeef): Won't need this test once we start generating actual
// standards-compliant SDP.
TEST_P(PeerConnectionIntegrationTest,
EndToEndCallWithSpecCompliantMaxBundleOffer) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Do the equivalent of setting the port to 0, adding a=bundle-only, and
// removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all
// but the first m= section.
callee()->SetReceivedSdpMunger(MakeSpecCompliantMaxBundleOffer);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Test that we can receive the audio output level from a remote audio track.
// TODO(deadbeef): Use a fake audio source and verify that the output level is
// exactly what the source on the other side was configured with.
TEST_P(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Just add an audio track.
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Get the audio output level stats. Note that the level is not available
// until an RTCP packet has been received.
EXPECT_TRUE_WAIT(callee()->OldGetStats()->AudioOutputLevel() > 0,
kMaxWaitForFramesMs);
}
// Test that an audio input level is reported.
// TODO(deadbeef): Use a fake audio source and verify that the input level is
// exactly what the source was configured with.
TEST_P(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Just add an audio track.
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Get the audio input level stats. The level should be available very
// soon after the test starts.
EXPECT_TRUE_WAIT(caller()->OldGetStats()->AudioInputLevel() > 0,
kMaxWaitForStatsMs);
}
// Test that we can get incoming byte counts from both audio and video tracks.
TEST_P(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
// Do offer/answer, wait for the callee to receive some frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// Get a handle to the remote tracks created, so they can be used as GetStats
// filters.
for (auto receiver : callee()->pc()->GetReceivers()) {
// We received frames, so we definitely should have nonzero "received bytes"
// stats at this point.
EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(),
0);
}
}
// Test that we can get outgoing byte counts from both audio and video tracks.
TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
auto audio_track = caller()->CreateLocalAudioTrack();
auto video_track = caller()->CreateLocalVideoTrack();
caller()->AddTrack(audio_track);
caller()->AddTrack(video_track);
// Do offer/answer, wait for the callee to receive some frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// The callee received frames, so we definitely should have nonzero "sent
// bytes" stats at this point.
EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0);
EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
}
// Test that we can get capture start ntp time.
TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
// Do offer/answer, wait for the callee to receive some frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Get the remote audio track created on the receiver, so they can be used as
// GetStats filters.
auto receivers = callee()->pc()->GetReceivers();
ASSERT_EQ(1u, receivers.size());
auto remote_audio_track = receivers[0]->track();
// Get the audio output level stats. Note that the level is not available
// until an RTCP packet has been received.
EXPECT_TRUE_WAIT(
callee()->OldGetStatsForTrack(remote_audio_track)->CaptureStartNtpTime() >
0,
2 * kMaxWaitForFramesMs);
}
// Test that the track ID is associated with all local and remote SSRC stats
// using the old GetStats() and more than 1 audio and more than 1 video track.
// This is a regression test for crbug.com/906988
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
OldGetStatsAssociatesTrackIdForManyMediaSections) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
auto audio_sender_1 = caller()->AddAudioTrack();
auto video_sender_1 = caller()->AddVideoTrack();
auto audio_sender_2 = caller()->AddAudioTrack();
auto video_sender_2 = caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudioAndVideo();
ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);
std::vector<std::string> track_ids = {
audio_sender_1->track()->id(), video_sender_1->track()->id(),
audio_sender_2->track()->id(), video_sender_2->track()->id()};
auto caller_stats = caller()->OldGetStats();
EXPECT_THAT(caller_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
auto callee_stats = callee()->OldGetStats();
EXPECT_THAT(callee_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
}
// Test that the new GetStats() returns stats for all outgoing/incoming streams
// with the correct track IDs if there are more than one audio and more than one
// video senders/receivers.
TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
auto audio_sender_1 = caller()->AddAudioTrack();
auto video_sender_1 = caller()->AddVideoTrack();
auto audio_sender_2 = caller()->AddAudioTrack();
auto video_sender_2 = caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudioAndVideo();
ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);
std::vector<std::string> track_ids = {
audio_sender_1->track()->id(), video_sender_1->track()->id(),
audio_sender_2->track()->id(), video_sender_2->track()->id()};
rtc::scoped_refptr<const webrtc::RTCStatsReport> caller_report =
caller()->NewGetStats();
ASSERT_TRUE(caller_report);
auto outbound_stream_stats =
caller_report->GetStatsOfType<webrtc::RTCOutboundRTPStreamStats>();
ASSERT_EQ(4u, outbound_stream_stats.size());
std::vector<std::string> outbound_track_ids;
for (const auto& stat : outbound_stream_stats) {
ASSERT_TRUE(stat->bytes_sent.is_defined());
EXPECT_LT(0u, *stat->bytes_sent);
ASSERT_TRUE(stat->track_id.is_defined());
const auto* track_stat =
caller_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
ASSERT_TRUE(track_stat);
outbound_track_ids.push_back(*track_stat->track_identifier);
}
EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids));
rtc::scoped_refptr<const webrtc::RTCStatsReport> callee_report =
callee()->NewGetStats();
ASSERT_TRUE(callee_report);
auto inbound_stream_stats =
callee_report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
ASSERT_EQ(4u, inbound_stream_stats.size());
std::vector<std::string> inbound_track_ids;
for (const auto& stat : inbound_stream_stats) {
ASSERT_TRUE(stat->bytes_received.is_defined());
EXPECT_LT(0u, *stat->bytes_received);
ASSERT_TRUE(stat->track_id.is_defined());
const auto* track_stat =
callee_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
ASSERT_TRUE(track_stat);
inbound_track_ids.push_back(*track_stat->track_identifier);
}
EXPECT_THAT(inbound_track_ids, UnorderedElementsAreArray(track_ids));
}
// Test that we can get stats (using the new stats implementation) for
// unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
// SDP.
TEST_P(PeerConnectionIntegrationTest,
GetStatsForUnsignaledStreamWithNewStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
// Remove SSRCs and MSIDs from the received offer SDP.
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio(1);
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// We received a frame, so we should have nonzero "bytes received" stats for
// the unsignaled stream, if stats are working for it.
rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
callee()->NewGetStats();
ASSERT_NE(nullptr, report);
auto inbound_stream_stats =
report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
ASSERT_EQ(1U, inbound_stream_stats.size());
ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined());
}
// Same as above but for the legacy stats implementation.
TEST_P(PeerConnectionIntegrationTest,
GetStatsForUnsignaledStreamWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
// Remove SSRCs and MSIDs from the received offer SDP.
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Note that, since the old stats implementation associates SSRCs with tracks
// using SDP, when SSRCs aren't signaled in SDP these stats won't have an
// associated track ID. So we can't use the track "selector" argument.
//
// Also, we use "EXPECT_TRUE_WAIT" because the stats collector may decide to
// return cached stats if not enough time has passed since the last update.
EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0,
kDefaultTimeout);
}
// Test that we can successfully get the media related stats (audio level
// etc.) for the unsignaled stream.
TEST_P(PeerConnectionIntegrationTest,
GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
// Remove SSRCs and MSIDs from the received offer SDP.
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio(1);
media_expectations.CalleeExpectsSomeVideo(1);
ASSERT_TRUE(ExpectNewFrames(media_expectations));
rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
callee()->NewGetStats();
ASSERT_NE(nullptr, report);
auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats);
ASSERT_GE(audio_index, 0);
EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined());
}
// Helper for test below.
void ModifySsrcs(cricket::SessionDescription* desc) {
for (ContentInfo& content : desc->contents()) {
for (StreamParams& stream :
content.media_description()->mutable_streams()) {
for (uint32_t& ssrc : stream.ssrcs) {
ssrc = rtc::CreateRandomId();
}
}
}
}
// Test that the "RTCMediaSteamTrackStats" object is updated correctly when
// SSRCs are unsignaled, and the SSRC of the received (audio) stream changes.
// This should result in two "RTCInboundRTPStreamStats", but only one
// "RTCMediaStreamTrackStats", whose counters go up continuously rather than
// being reset to 0 once the SSRC change occurs.
//
// Regression test for this bug:
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
//
// The bug causes the track stats to only represent one of the two streams:
// whichever one has the higher SSRC. So with this bug, there was a 50% chance
// that the track stat counters would reset to 0 when the new stream is
// received, and a 50% chance that they'll stop updating (while
// "concealed_samples" continues increasing, due to silence being generated for
// the inactive stream).
TEST_P(PeerConnectionIntegrationTest,
TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
// Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint
// that doesn't signal SSRCs (from the callee's perspective).
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for 50 audio frames (500ms of audio) to be received by the callee.
{
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio(50);
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Some audio frames were received, so we should have nonzero "samples
// received" for the track.
rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
callee()->NewGetStats();
ASSERT_NE(nullptr, report);
auto track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
ASSERT_EQ(1U, track_stats.size());
ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
ASSERT_GT(*track_stats[0]->total_samples_received, 0U);
// uint64_t prev_samples_received = *track_stats[0]->total_samples_received;
// Create a new offer and munge it to cause the caller to use a new SSRC.
caller()->SetGeneratedSdpMunger(ModifySsrcs);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for 25 more audio frames (250ms of audio) to be received, from the new
// SSRC.
{
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio(25);
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
report = callee()->NewGetStats();
ASSERT_NE(nullptr, report);
track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
ASSERT_EQ(1U, track_stats.size());
ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
// The "total samples received" stat should only be greater than it was
// before.
// TODO(deadbeef): Uncomment this assertion once the bug is completely fixed.
// Right now, the new SSRC will cause the counters to reset to 0.
// EXPECT_GT(*track_stats[0]->total_samples_received, prev_samples_received);
// Additionally, the percentage of concealed samples (samples generated to
// conceal packet loss) should be less than 50%. If it's greater, that's a
// good sign that we're seeing stats from the old stream that's no longer
// receiving packets, and is generating concealed samples of silence.
constexpr double kAcceptableConcealedSamplesPercentage = 0.50;
ASSERT_TRUE(track_stats[0]->concealed_samples.is_defined());
EXPECT_LT(*track_stats[0]->concealed_samples,
*track_stats[0]->total_samples_received *
kAcceptableConcealedSamplesPercentage);
// Also ensure that we have two "RTCInboundRTPStreamStats" as expected, as a
// sanity check that the SSRC really changed.
// TODO(deadbeef): This isn't working right now, because we're not returning
// *any* stats for the inactive stream. Uncomment when the bug is completely
// fixed.
// auto inbound_stream_stats =
// report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
// ASSERT_EQ(2U, inbound_stream_stats.size());
}
// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
PeerConnectionFactory::Options dtls_10_options;
dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
dtls_10_options));
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));
}
// Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated.
TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
PeerConnectionFactory::Options dtls_10_options;
dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
dtls_10_options));
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
}
// Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
PeerConnectionFactory::Options dtls_12_options;
dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options,
dtls_12_options));
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
// callee only supports 1.0.
TEST_P(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
PeerConnectionFactory::Options caller_options;
caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
PeerConnectionFactory::Options callee_options;
callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
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));
}
// Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the
// callee supports 1.2.
TEST_P(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
PeerConnectionFactory::Options caller_options;
caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
PeerConnectionFactory::Options callee_options;
callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
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));
}
// The three tests below verify that "enable_aes128_sha1_32_crypto_cipher"
// works as expected; the cipher should only be used if enabled by both sides.
TEST_P(PeerConnectionIntegrationTest,
Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
PeerConnectionFactory::Options caller_options;
caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
PeerConnectionFactory::Options callee_options;
callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
false;
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
}
TEST_P(PeerConnectionIntegrationTest,
Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
PeerConnectionFactory::Options caller_options;
caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
false;
PeerConnectionFactory::Options callee_options;
callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
}
TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
PeerConnectionFactory::Options caller_options;
caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
PeerConnectionFactory::Options callee_options;
callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32;
TestNegotiatedCipherSuite(caller_options, callee_options,
expected_cipher_suite);
}
// Test that a non-GCM cipher is used if both sides only support non-GCM.
TEST_P(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
bool local_gcm_enabled = false;
bool remote_gcm_enabled = false;
int expected_cipher_suite = kDefaultSrtpCryptoSuite;
TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
expected_cipher_suite);
}
// Test that a GCM cipher is used if both ends support it.
TEST_P(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) {
bool local_gcm_enabled = true;
bool remote_gcm_enabled = true;
int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm;
TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
expected_cipher_suite);
}
// Test that GCM isn't used if only the offerer supports it.
TEST_P(PeerConnectionIntegrationTest,
NonGcmCipherUsedWhenOnlyCallerSupportsGcm) {
bool local_gcm_enabled = true;
bool remote_gcm_enabled = false;
int expected_cipher_suite = kDefaultSrtpCryptoSuite;
TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
expected_cipher_suite);
}
// Test that GCM isn't used if only the answerer supports it.
TEST_P(PeerConnectionIntegrationTest,
NonGcmCipherUsedWhenOnlyCalleeSupportsGcm) {
bool local_gcm_enabled = false;
bool remote_gcm_enabled = true;
int expected_cipher_suite = kDefaultSrtpCryptoSuite;
TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
expected_cipher_suite);
}
// Verify that media can be transmitted end-to-end when GCM crypto suites are
// enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported,
// only verify that a GCM cipher is negotiated, and not necessarily that SRTP
// works with it.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
PeerConnectionFactory::Options gcm_options;
gcm_options.crypto_options.srtp.enable_gcm_crypto_suites = true;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options));
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));
}
// This test sets up a call between two parties with audio, video and an RTP
// data channel.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.enable_rtp_data_channel = true;
rtc_config.enable_dtls_srtp = false;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
ConnectFakeSignaling();
// Expect that data channel created on caller side will show up for callee as
// well.
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure the existence of the RTP data channel didn't impede audio/video.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_NE(nullptr, callee()->data_channel());
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Ensure data can be sent in both directions.
std::string data = "hello world";
SendRtpDataWithRetries(caller()->data_channel(), data, 5);
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
kDefaultTimeout);
SendRtpDataWithRetries(callee()->data_channel(), data, 5);
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
kDefaultTimeout);
}
// Ensure that an RTP data channel is signaled as closed for the caller when
// the callee rejects it in a subsequent offer.
TEST_P(PeerConnectionIntegrationTest,
RtpDataChannelSignaledClosedInCalleeOffer) {
// Same procedure as above test.
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.enable_rtp_data_channel = true;
rtc_config.enable_dtls_srtp = false;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_NE(nullptr, callee()->data_channel());
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Close the data channel on the callee, and do an updated offer/answer.
callee()->data_channel()->Close();
callee()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_FALSE(caller()->data_observer()->IsOpen());
EXPECT_FALSE(callee()->data_observer()->IsOpen());
}
// Tests that data is buffered in an RTP data channel until an observer is
// registered for it.
//
// NOTE: RTP data channels can receive data before the underlying
// transport has detected that a channel is writable and thus data can be
// received before the data channel state changes to open. That is hard to test
// but the same buffering is expected to be used in that case.
TEST_P(PeerConnectionIntegrationTest,
DataBufferedUntilRtpDataChannelObserverRegistered) {
// Use fake clock and simulated network delay so that we predictably can wait
// until an SCTP message has been delivered without "sleep()"ing.
rtc::ScopedFakeClock fake_clock;
// Some things use a time of "0" as a special value, so we need to start out
// the fake clock at a nonzero time.
// TODO(deadbeef): Fix this.
fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
virtual_socket_server()->set_delay_mean(5); // 5 ms per hop.
virtual_socket_server()->UpdateDelayDistribution();
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.enable_rtp_data_channel = true;
rtc_config.enable_dtls_srtp = false;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE(caller()->data_channel() != nullptr);
ASSERT_TRUE_SIMULATED_WAIT(callee()->data_channel() != nullptr,
kDefaultTimeout, fake_clock);
ASSERT_TRUE_SIMULATED_WAIT(caller()->data_observer()->IsOpen(),
kDefaultTimeout, fake_clock);
ASSERT_EQ_SIMULATED_WAIT(DataChannelInterface::kOpen,
callee()->data_channel()->state(), kDefaultTimeout,
fake_clock);
// Unregister the observer which is normally automatically registered.
callee()->data_channel()->UnregisterObserver();
// Send data and advance fake clock until it should have been received.
std::string data = "hello world";
caller()->data_channel()->Send(DataBuffer(data));
SIMULATED_WAIT(false, 50, fake_clock);
// Attach data channel and expect data to be received immediately. Note that
// EXPECT_EQ_WAIT is used, such that the simulated clock is not advanced any
// further, but data can be received even if the callback is asynchronous.
MockDataChannelObserver new_observer(callee()->data_channel());
EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout,
fake_clock);
// Closing the PeerConnections destroys the ports before the ScopedFakeClock.
// If this is not done a DCHECK can be hit in ports.cc, because a large
// negative number is calculated for the rtt due to the global clock changing.
caller()->pc()->Close();
callee()->pc()->Close();
}
// This test sets up a call between two parties with audio, video and but only
// the caller client supports RTP data channels.
TEST_P(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
PeerConnectionInterface::RTCConfiguration rtc_config_1;
rtc_config_1.enable_rtp_data_channel = true;
// Must disable DTLS to make negotiation succeed.
rtc_config_1.enable_dtls_srtp = false;
PeerConnectionInterface::RTCConfiguration rtc_config_2;
rtc_config_2.enable_dtls_srtp = false;
rtc_config_2.enable_dtls_srtp = false;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithConfig(rtc_config_1, rtc_config_2));
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// The caller should still have a data channel, but it should be closed, and
// one should ever have been created for the callee.
EXPECT_TRUE(caller()->data_channel() != nullptr);
EXPECT_FALSE(caller()->data_observer()->IsOpen());
EXPECT_EQ(nullptr, callee()->data_channel());
}
// This test sets up a call between two parties with audio, and video. When
// audio and video is setup and flowing, an RTP data channel is negotiated.
TEST_P(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.enable_rtp_data_channel = true;
rtc_config.enable_dtls_srtp = false;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
ConnectFakeSignaling();
// Do initial offer/answer with audio/video.
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Create data channel and do new offer and answer.
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_NE(nullptr, callee()->data_channel());
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Ensure data can be sent in both directions.
std::string data = "hello world";
SendRtpDataWithRetries(caller()->data_channel(), data, 5);
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
kDefaultTimeout);
SendRtpDataWithRetries(callee()->data_channel(), data, 5);
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
kDefaultTimeout);
}
#ifdef HAVE_SCTP
// This test sets up a call between two parties with audio, video and an SCTP
// data channel.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Expect that data channel created on caller side will show up for callee as
// well.
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure the existence of the SCTP data channel didn't impede audio/video.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// Caller data channel should already exist (it created one). Callee data
// channel may not exist yet, since negotiation happens in-band, not in SDP.
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Ensure data can be sent in both directions.
std::string data = "hello world";
caller()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
kDefaultTimeout);
callee()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
kDefaultTimeout);
}
// Ensure that when the callee closes an SCTP data channel, the closing
// procedure results in the data channel being closed for the caller as well.
TEST_P(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) {
// Same procedure as above test.
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Close the data channel on the callee side, and wait for it to reach the
// "closed" state on both sides.
callee()->data_channel()->Close();
EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
}
TEST_P(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
webrtc::DataChannelInit init;
init.id = 53;
init.maxRetransmits = 52;
caller()->CreateDataChannel("data-channel", &init);
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_EQ(init.id, callee()->data_channel()->id());
EXPECT_EQ("data-channel", callee()->data_channel()->label());
EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
EXPECT_FALSE(callee()->data_channel()->negotiated());
}
// Test usrsctp's ability to process unordered data stream, where data actually
// arrives out of order using simulated delays. Previously there have been some
// bugs in this area.
TEST_P(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
// Introduce random network delays.
// Otherwise it's not a true "unordered" test.
virtual_socket_server()->set_delay_mean(20);
virtual_socket_server()->set_delay_stddev(5);
virtual_socket_server()->UpdateDelayDistribution();
// Normal procedure, but with unordered data channel config.
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
webrtc::DataChannelInit init;
init.ordered = false;
caller()->CreateDataChannel(&init);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
static constexpr int kNumMessages = 100;
// Deliberately chosen to be larger than the MTU so messages get fragmented.
static constexpr size_t kMaxMessageSize = 4096;
// Create and send random messages.
std::vector<std::string> sent_messages;
for (int i = 0; i < kNumMessages; ++i) {
size_t length =
(rand() % kMaxMessageSize) + 1; // NOLINT (rand_r instead of rand)
std::string message;
ASSERT_TRUE(rtc::CreateRandomString(length, &message));
caller()->data_channel()->Send(DataBuffer(message));
callee()->data_channel()->Send(DataBuffer(message));
sent_messages.push_back(message);
}
// Wait for all messages to be received.
EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
caller()->data_observer()->received_message_count(),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
callee()->data_observer()->received_message_count(),
kDefaultTimeout);
// Sort and compare to make sure none of the messages were corrupted.
std::vector<std::string> caller_received_messages =
caller()->data_observer()->messages();
std::vector<std::string> callee_received_messages =
callee()->data_observer()->messages();
std::sort(sent_messages.begin(), sent_messages.end());
std::sort(caller_received_messages.begin(), caller_received_messages.end());
std::sort(callee_received_messages.begin(), callee_received_messages.end());
EXPECT_EQ(sent_messages, caller_received_messages);
EXPECT_EQ(sent_messages, callee_received_messages);
}
// This test sets up a call between two parties with audio, and video. When
// audio and video are setup and flowing, an SCTP data channel is negotiated.
TEST_P(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do initial offer/answer with audio/video.
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Create data channel and do new offer and answer.
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Caller data channel should already exist (it created one). Callee data
// channel may not exist yet, since negotiation happens in-band, not in SDP.
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Ensure data can be sent in both directions.
std::string data = "hello world";
caller()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
kDefaultTimeout);
callee()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
kDefaultTimeout);
}
// Set up a connection initially just using SCTP data channels, later upgrading
// to audio/video, ensuring frames are received end-to-end. Effectively the
// inverse of the test above.
// This was broken in M57; see https://crbug.com/711243
TEST_P(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do initial offer/answer with just data channel.
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait until data can be sent over the data channel.
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Do subsequent offer/answer with two-way audio and video. Audio and video
// should end up bundled on the DTLS/ICE transport already used for data.
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) {
cricket::DataContentDescription* dcd_offer =
GetFirstDataContentDescription(desc);
ASSERT_TRUE(dcd_offer);
dcd_offer->set_use_sctpmap(false);
dcd_offer->set_protocol("UDP/DTLS/SCTP");
}
// Test that the data channel works when a spec-compliant SCTP m= section is
// offered (using "a=sctp-port" instead of "a=sctpmap", and using
// "UDP/DTLS/SCTP" as the protocol).
TEST_P(PeerConnectionIntegrationTest,
DataChannelWorksWhenSpecCompliantSctpOfferReceived) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->SetGeneratedSdpMunger(MakeSpecCompliantSctpOffer);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Ensure data can be sent in both directions.
std::string data = "hello world";
caller()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
kDefaultTimeout);
callee()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
kDefaultTimeout);
}
#endif // HAVE_SCTP
// This test sets up a call between two parties with a media transport data
// channel.
TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelEndToEnd) {
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.use_media_transport_for_data_channels = true;
rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
rtc_config, rtc_config, loopback_media_transports()->first_factory(),
loopback_media_transports()->second_factory()));
ConnectFakeSignaling();
// Expect that data channel created on caller side will show up for callee as
// well.
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure that the media transport is ready.
loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
loopback_media_transports()->FlushAsyncInvokes();
// Caller data channel should already exist (it created one). Callee data
// channel may not exist yet, since negotiation happens in-band, not in SDP.
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Ensure data can be sent in both directions.
std::string data = "hello world";
caller()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
kDefaultTimeout);
callee()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
kDefaultTimeout);
}
// Ensure that when the callee closes a media transport data channel, the
// closing procedure results in the data channel being closed for the caller
// as well.
TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelCalleeCloses) {
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.use_media_transport_for_data_channels = true;
rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
rtc_config, rtc_config, loopback_media_transports()->first_factory(),
loopback_media_transports()->second_factory()));
ConnectFakeSignaling();
// Create a data channel on the caller and signal it to the callee.
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure that the media transport is ready.
loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
loopback_media_transports()->FlushAsyncInvokes();
// Data channels exist and open on both ends of the connection.
ASSERT_NE(nullptr, caller()->data_channel());
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
// Close the data channel on the callee side, and wait for it to reach the
// "closed" state on both sides.
callee()->data_channel()->Close();
EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
}
TEST_P(PeerConnectionIntegrationTest,
MediaTransportDataChannelConfigSentToOtherSide) {
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.use_media_transport_for_data_channels = true;
rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
rtc_config, rtc_config, loopback_media_transports()->first_factory(),
loopback_media_transports()->second_factory()));
ConnectFakeSignaling();
// Create a data channel with a non-default configuration and signal it to the
// callee.
webrtc::DataChannelInit init;
init.id = 53;
init.maxRetransmits = 52;
caller()->CreateDataChannel("data-channel", &init);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure that the media transport is ready.
loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
loopback_media_transports()->FlushAsyncInvokes();
// Ensure that the data channel exists on the callee with the correct
// configuration.
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
EXPECT_EQ(init.id, callee()->data_channel()->id());
EXPECT_EQ("data-channel", callee()->data_channel()->label());
EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
EXPECT_FALSE(callee()->data_channel()->negotiated());
}
TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalAudio) {
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.use_media_transport = true;
rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
rtc_config, rtc_config, loopback_media_transports()->first_factory(),
loopback_media_transports()->second_factory()));
ConnectFakeSignaling();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
// Start offer/answer exchange and wait for it to complete.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure that the media transport is ready.
loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
loopback_media_transports()->FlushAsyncInvokes();
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
webrtc::MediaTransportPair::Stats first_stats =
loopback_media_transports()->FirstStats();
webrtc::MediaTransportPair::Stats second_stats =
loopback_media_transports()->SecondStats();
EXPECT_GT(first_stats.received_audio_frames, 0);
EXPECT_GE(second_stats.sent_audio_frames, first_stats.received_audio_frames);
EXPECT_GT(second_stats.received_audio_frames, 0);
EXPECT_GE(first_stats.sent_audio_frames, second_stats.received_audio_frames);
}
TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalVideo) {
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.use_media_transport = true;
rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
rtc_config, rtc_config, loopback_media_transports()->first_factory(),
loopback_media_transports()->second_factory()));
ConnectFakeSignaling();
caller()->AddVideoTrack();
callee()->AddVideoTrack();
// Start offer/answer exchange and wait for it to complete.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure that the media transport is ready.
loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
loopback_media_transports()->FlushAsyncInvokes();
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
webrtc::MediaTransportPair::Stats first_stats =
loopback_media_transports()->FirstStats();
webrtc::MediaTransportPair::Stats second_stats =
loopback_media_transports()->SecondStats();
EXPECT_GT(first_stats.received_video_frames, 0);
EXPECT_GE(second_stats.sent_video_frames, first_stats.received_video_frames);
EXPECT_GT(second_stats.received_video_frames, 0);
EXPECT_GE(first_stats.sent_video_frames, second_stats.received_video_frames);
}
// Test that the ICE connection and gathering states eventually reach
// "complete".
TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do normal offer/answer.
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
caller()->ice_gathering_state(), kMaxWaitForFramesMs);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
callee()->ice_gathering_state(), kMaxWaitForFramesMs);
// After the best candidate pair is selected and all candidates are signaled,
// the ICE connection state should reach "complete".
// TODO(deadbeef): Currently, the ICE "controlled" agent (the
// answerer/"callee" by default) only reaches "connected". When this is
// fixed, this test should be updated.
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kDefaultTimeout);
}
constexpr int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
cricket::PORTALLOCATOR_DISABLE_RELAY |
cricket::PORTALLOCATOR_DISABLE_TCP;
// Use a mock resolver to resolve the hostname back to the original IP on both
// sides and check that the ICE connection connects.
TEST_P(PeerConnectionIntegrationTest,
IceStatesReachCompletionWithRemoteHostname) {
auto caller_resolver_factory =
absl::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
auto callee_resolver_factory =
absl::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
NiceMock<rtc::MockAsyncResolver> callee_async_resolver;
NiceMock<rtc::MockAsyncResolver> caller_async_resolver;
// This also verifies that the injected AsyncResolverFactory is used by
// P2PTransportChannel.
EXPECT_CALL(*caller_resolver_factory, Create())
.WillOnce(Return(&caller_async_resolver));
webrtc::PeerConnectionDependencies caller_deps(nullptr);
caller_deps.async_resolver_factory = std::move(caller_resolver_factory);
EXPECT_CALL(*callee_resolver_factory, Create())
.WillOnce(Return(&callee_async_resolver));
webrtc::PeerConnectionDependencies callee_deps(nullptr);
callee_deps.async_resolver_factory = std::move(callee_resolver_factory);
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
config, std::move(caller_deps), config, std::move(callee_deps)));
caller()->SetRemoteAsyncResolver(&callee_async_resolver);
callee()->SetRemoteAsyncResolver(&caller_async_resolver);
// Enable hostname candidates with mDNS names.
caller()->network()->CreateMdnsResponder(network_thread());
callee()->network()->CreateMdnsResponder(network_thread());
SetPortAllocatorFlags(kOnlyLocalPorts, kOnlyLocalPorts);
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kDefaultTimeout);
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostNameHostName));
}
// Test that firewalling the ICE connection causes the clients to identify the
// disconnected state and then removing the firewall causes them to reconnect.
class PeerConnectionIntegrationIceStatesTest
: public PeerConnectionIntegrationBaseTest,
public ::testing::WithParamInterface<
std::tuple<SdpSemantics, std::tuple<std::string, uint32_t>>> {
protected:
PeerConnectionIntegrationIceStatesTest()
: PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())) {
port_allocator_flags_ = std::get<1>(std::get<1>(GetParam()));
}
void StartStunServer(const SocketAddress& server_address) {
stun_server_.reset(
cricket::TestStunServer::Create(network_thread(), server_address));
}
bool TestIPv6() {
return (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6);
}
void SetPortAllocatorFlags() {
PeerConnectionIntegrationBaseTest::SetPortAllocatorFlags(
port_allocator_flags_, port_allocator_flags_);
}
std::vector<SocketAddress> CallerAddresses() {
std::vector<SocketAddress> addresses;
addresses.push_back(SocketAddress("1.1.1.1", 0));
if (TestIPv6()) {
addresses.push_back(SocketAddress("1111:0:a:b:c:d:e:f", 0));
}
return addresses;
}
std::vector<SocketAddress> CalleeAddresses() {
std::vector<SocketAddress> addresses;
addresses.push_back(SocketAddress("2.2.2.2", 0));
if (TestIPv6()) {
addresses.push_back(SocketAddress("2222:0:a:b:c:d:e:f", 0));
}
return addresses;
}
void SetUpNetworkInterfaces() {
// Remove the default interfaces added by the test infrastructure.
caller()->network()->RemoveInterface(kDefaultLocalAddress);
callee()->network()->RemoveInterface(kDefaultLocalAddress);
// Add network addresses for test.
for (const auto& caller_address : CallerAddresses()) {
caller()->network()->AddInterface(caller_address);
}
for (const auto& callee_address : CalleeAddresses()) {
callee()->network()->AddInterface(callee_address);
}
}
private:
uint32_t port_allocator_flags_;
std::unique_ptr<cricket::TestStunServer> stun_server_;
};
// Tests that the PeerConnection goes through all the ICE gathering/connection
// states over the duration of the call. This includes Disconnected and Failed
// states, induced by putting a firewall between the peers and waiting for them
// to time out.
TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyIceStates) {
// TODO(bugs.webrtc.org/8295): When using a ScopedFakeClock, this test will
// sometimes hit a DCHECK in platform_thread.cc about the PacerThread being
// too busy. For now, revert to running without a fake clock.
const SocketAddress kStunServerAddress =
SocketAddress("99.99.99.1", cricket::STUN_SERVER_PORT);
StartStunServer(kStunServerAddress);
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer ice_stun_server;
ice_stun_server.urls.push_back(
"stun:" + kStunServerAddress.HostAsURIString() + ":" +
kStunServerAddress.PortAsString());
config.servers.push_back(ice_stun_server);
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
SetPortAllocatorFlags();
SetUpNetworkInterfaces();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Initial state before anything happens.
ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew,
caller()->ice_gathering_state());
ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
caller()->ice_connection_state());
ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
caller()->standardized_ice_connection_state());
// Start the call by creating the offer, setting it as the local description,
// then sending it to the peer who will respond with an answer. This happens
// asynchronously so that we can watch the states as it runs in the
// background.
caller()->CreateAndSetAndSignalOffer();
ASSERT_EQ(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state());
ASSERT_EQ(PeerConnectionInterface::kIceConnectionConnected,
caller()->standardized_ice_connection_state());
// Verify that the observer was notified of the intermediate transitions.
EXPECT_THAT(caller()->ice_connection_state_history(),
ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
PeerConnectionInterface::kIceConnectionConnected,
PeerConnectionInterface::kIceConnectionCompleted));
EXPECT_THAT(
caller()->peer_connection_state_history(),
ElementsAre(PeerConnectionInterface::PeerConnectionState::kConnecting,
PeerConnectionInterface::PeerConnectionState::kConnected));
EXPECT_THAT(caller()->ice_gathering_state_history(),
ElementsAre(PeerConnectionInterface::kIceGatheringGathering,
PeerConnectionInterface::kIceGatheringComplete));
// Block connections to/from the caller and wait for ICE to become
// disconnected.
for (const auto& caller_address : CallerAddresses()) {
firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
}
RTC_LOG(LS_INFO) << "Firewall rules applied";
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
caller()->ice_connection_state(), kDefaultTimeout);
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
caller()->standardized_ice_connection_state(),
kDefaultTimeout);
// Let ICE re-establish by removing the firewall rules.
firewall()->ClearRules();
RTC_LOG(LS_INFO) << "Firewall rules cleared";
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout);
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
caller()->standardized_ice_connection_state(),
kDefaultTimeout);
// According to RFC7675, if there is no response within 30 seconds then the
// peer should consider the other side to have rejected the connection. This
// is signaled by the state transitioning to "failed".
constexpr int kConsentTimeout = 30000;
for (const auto& caller_address : CallerAddresses()) {
firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
}
RTC_LOG(LS_INFO) << "Firewall rules applied again";
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
caller()->ice_connection_state(), kConsentTimeout);
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
caller()->standardized_ice_connection_state(),
kConsentTimeout);
}
// Tests that the best connection is set to the appropriate IPv4/IPv6 connection
// and that the statistics in the metric observers are updated correctly.
TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
SetPortAllocatorFlags();
SetUpNetworkInterfaces();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
const int num_best_ipv4 = webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
const int num_best_ipv6 = webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
if (TestIPv6()) {
// When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
// connection.
EXPECT_EQ(0, num_best_ipv4);
EXPECT_EQ(1, num_best_ipv6);
} else {
EXPECT_EQ(1, num_best_ipv4);
EXPECT_EQ(0, num_best_ipv6);
}
EXPECT_EQ(0, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostHost));
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostPublicHostPublic));
}
constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
cricket::PORTALLOCATOR_DISABLE_STUN |
cricket::PORTALLOCATOR_DISABLE_RELAY;
constexpr uint32_t kFlagsIPv6NoStun =
cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_STUN |
cricket::PORTALLOCATOR_ENABLE_IPV6 | cricket::PORTALLOCATOR_DISABLE_RELAY;
constexpr uint32_t kFlagsIPv4Stun =
cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_RELAY;
INSTANTIATE_TEST_CASE_P(
PeerConnectionIntegrationTest,
PeerConnectionIntegrationIceStatesTest,
Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
std::make_pair("IPv4 with STUN", kFlagsIPv4Stun))));
// This test sets up a call between two parties with audio and video.
// During the call, the caller restarts ICE and the test verifies that
// new ICE candidates are generated and audio and video still can flow, and the
// ICE state reaches completed again.
TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do normal offer/answer and wait for ICE to complete.
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kMaxWaitForFramesMs);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// To verify that the ICE restart actually occurs, get
// ufrag/password/candidates before and after restart.
// Create an SDP string of the first audio candidate for both clients.
const webrtc::IceCandidateCollection* audio_candidates_caller =
caller()->pc()->local_description()->candidates(0);
const webrtc::IceCandidateCollection* audio_candidates_callee =
callee()->pc()->local_description()->candidates(0);
ASSERT_GT(audio_candidates_caller->count(), 0u);
ASSERT_GT(audio_candidates_callee->count(), 0u);
std::string caller_candidate_pre_restart;
ASSERT_TRUE(
audio_candidates_caller->at(0)->ToString(&caller_candidate_pre_restart));
std::string callee_candidate_pre_restart;
ASSERT_TRUE(
audio_candidates_callee->at(0)->ToString(&callee_candidate_pre_restart));
const cricket::SessionDescription* desc =
caller()->pc()->local_description()->description();
std::string caller_ufrag_pre_restart =
desc->transport_infos()[0].description.ice_ufrag;
desc = callee()->pc()->local_description()->description();
std::string callee_ufrag_pre_restart =
desc->transport_infos()[0].description.ice_ufrag;
// Have the caller initiate an ICE restart.
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kMaxWaitForFramesMs);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// Grab the ufrags/candidates again.
audio_candidates_caller = caller()->pc()->local_description()->candidates(0);
audio_candidates_callee = callee()->pc()->local_description()->candidates(0);
ASSERT_GT(audio_candidates_caller->count(), 0u);
ASSERT_GT(audio_candidates_callee->count(), 0u);
std::string caller_candidate_post_restart;
ASSERT_TRUE(
audio_candidates_caller->at(0)->ToString(&caller_candidate_post_restart));
std::string callee_candidate_post_restart;
ASSERT_TRUE(
audio_candidates_callee->at(0)->ToString(&callee_candidate_post_restart));
desc = caller()->pc()->local_description()->description();
std::string caller_ufrag_post_restart =
desc->transport_infos()[0].description.ice_ufrag;
desc = callee()->pc()->local_description()->description();
std::string callee_ufrag_post_restart =
desc->transport_infos()[0].description.ice_ufrag;
// Sanity check that an ICE restart was actually negotiated in SDP.
ASSERT_NE(caller_candidate_pre_restart, caller_candidate_post_restart);
ASSERT_NE(callee_candidate_pre_restart, callee_candidate_post_restart);
ASSERT_NE(caller_ufrag_pre_restart, caller_ufrag_post_restart);
ASSERT_NE(callee_ufrag_pre_restart, callee_ufrag_post_restart);
// Ensure that additional frames are received after the ICE restart.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Verify that audio/video can be received end-to-end when ICE renomination is
// enabled.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = true;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, 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);
// Sanity check that ICE renomination was actually negotiated.
const cricket::SessionDescription* desc =
caller()->pc()->local_description()->description();
for (const cricket::TransportInfo& info : desc->transport_infos()) {
ASSERT_NE(
info.description.transport_options.end(),
std::find(info.description.transport_options.begin(),
info.description.transport_options.end(), "renomination"));
}
desc = callee()->pc()->local_description()->description();
for (const cricket::TransportInfo& info : desc->transport_infos()) {
ASSERT_NE(
info.description.transport_options.end(),
std::find(info.description.transport_options.begin(),
info.description.transport_options.end(), "renomination"));
}
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// With a max bundle policy and RTCP muxing, adding a new media description to
// the connection should not affect ICE at all because the new media will use
// the existing connection.
TEST_P(PeerConnectionIntegrationTest,
AddMediaToConnectedBundleDoesNotRestartIce) {
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(
config, PeerConnectionInterface::RTCConfiguration()));
ConnectFakeSignaling();
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout);
caller()->clear_ice_connection_state_history();
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ(0u, caller()->ice_connection_state_history().size());
}
// This test sets up a call between two parties with audio and video. It then
// renegotiates setting the video m-line to "port 0", then later renegotiates
// again, enabling video.
TEST_P(PeerConnectionIntegrationTest,
VideoFlowsAfterMediaSectionIsRejectedAndRecycled) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do initial negotiation, only sending media from the caller. Will result in
// video and audio recvonly "m=" sections.
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Negotiate again, disabling the video "m=" section (the callee will set the
// port to 0 due to offer_to_receive_video = 0).
if (sdp_semantics_ == SdpSemantics::kPlanB) {
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_video = 0;
callee()->SetOfferAnswerOptions(options);
} else {
callee()->SetRemoteOfferHandler([this] {
callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
});
}
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Sanity check that video "m=" section was actually rejected.
const ContentInfo* answer_video_content = cricket::GetFirstVideoContent(
callee()->pc()->local_description()->description());
ASSERT_NE(nullptr, answer_video_content);
ASSERT_TRUE(answer_video_content->rejected);
// Enable video and do negotiation again, making sure video is received
// end-to-end, also adding media stream to callee.
if (sdp_semantics_ == SdpSemantics::kPlanB) {
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_video = 1;
callee()->SetOfferAnswerOptions(options);
} else {
// The caller's transceiver is stopped, so we need to add another track.
auto caller_transceiver =
caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
EXPECT_TRUE(caller_transceiver->stopped());
caller()->AddVideoTrack();
}
callee()->AddVideoTrack();
callee()->SetRemoteOfferHandler(nullptr);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Verify the caller receives frames from the newly added stream, and the
// callee receives additional frames from the re-enabled video m= section.
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio();
media_expectations.ExpectBidirectionalVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This tests that if we negotiate after calling CreateSender but before we
// have a track, then set a track later, frames from the newly-set track are
// received end-to-end.
TEST_F(PeerConnectionIntegrationTestPlanB,
MediaFlowsAfterEarlyWarmupWithCreateSender) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
auto caller_audio_sender =
caller()->pc()->CreateSender("audio", "caller_stream");
auto caller_video_sender =
caller()->pc()->CreateSender("video", "caller_stream");
auto callee_audio_sender =
callee()->pc()->CreateSender("audio", "callee_stream");
auto callee_video_sender =
callee()->pc()->CreateSender("video", "callee_stream");
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
// Wait for ICE to complete, without any tracks being set.
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kMaxWaitForFramesMs);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// Now set the tracks, and expect frames to immediately start flowing.
EXPECT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
EXPECT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
EXPECT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
EXPECT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This tests that if we negotiate after calling AddTransceiver but before we
// have a track, then set a track later, frames from the newly-set tracks are
// received end-to-end.
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
MediaFlowsAfterEarlyWarmupWithAddTransceiver) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
auto audio_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
ASSERT_EQ(RTCErrorType::NONE, audio_result.error().type());
auto caller_audio_sender = audio_result.MoveValue()->sender();
auto video_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
ASSERT_EQ(RTCErrorType::NONE, video_result.error().type());
auto caller_video_sender = video_result.MoveValue()->sender();
callee()->SetRemoteOfferHandler([this] {
ASSERT_EQ(2u, callee()->pc()->GetTransceivers().size());
callee()->pc()->GetTransceivers()[0]->SetDirection(
RtpTransceiverDirection::kSendRecv);
callee()->pc()->GetTransceivers()[1]->SetDirection(
RtpTransceiverDirection::kSendRecv);
});
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
// Wait for ICE to complete, without any tracks being set.
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kMaxWaitForFramesMs);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// Now set the tracks, and expect frames to immediately start flowing.
auto callee_audio_sender = callee()->pc()->GetSenders()[0];
auto callee_video_sender = callee()->pc()->GetSenders()[1];
ASSERT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
ASSERT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
ASSERT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
ASSERT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This test verifies that a remote video track can be added via AddStream,
// and sent end-to-end. For this particular test, it's simply echoed back
// from the caller to the callee, rather than being forwarded to a third
// PeerConnection.
TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Just send a video track from the caller.
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
ASSERT_EQ(1U, callee()->remote_streams()->count());
// Echo the stream back, and do a new offer/anwer (initiated by callee this
// time).
callee()->pc()->AddStream(callee()->remote_streams()->at(0));
callee()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Test that we achieve the expected end-to-end connection time, using a
// fake clock and simulated latency on the media and signaling paths.
// We use a TURN<->TURN connection because this is usually the quickest to
// set up initially, especially when we're confident the connection will work
// and can start sending media before we get a STUN response.
//
// With various optimizations enabled, here are the network delays we expect to
// be on the critical path:
// 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then
// signaling answer (with DTLS fingerprint).
// 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when
// using TURN<->TURN pair, and DTLS exchange is 4 packets,
// the first of which should have arrived before the answer.
TEST_P(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
rtc::ScopedFakeClock fake_clock;
// Some things use a time of "0" as a special value, so we need to start out
// the fake clock at a nonzero time.
// TODO(deadbeef): Fix this.
fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
static constexpr int media_hop_delay_ms = 50;
static constexpr int signaling_trip_delay_ms = 500;
// For explanation of these values, see comment above.
static constexpr int required_media_hops = 9;
static constexpr int required_signaling_trips = 2;
// For internal delays (such as posting an event asychronously).
static constexpr int allowed_internal_delay_ms = 20;
static constexpr int total_connection_time_ms =
media_hop_delay_ms * required_media_hops +
signaling_trip_delay_ms * required_signaling_trips +
allowed_internal_delay_ms;
static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
3478};
static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
0};
static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
3478};
static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
0};
cricket::TestTurnServer* turn_server_1 = CreateTurnServer(
turn_server_1_internal_address, turn_server_1_external_address);
cricket::TestTurnServer* turn_server_2 = CreateTurnServer(
turn_server_2_internal_address, turn_server_2_external_address);
// Bypass permission check on received packets so media can be sent before
// the candidate is signaled.
network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_1] {
turn_server_1->set_enable_permission_checks(false);
});
network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_2] {
turn_server_2->set_enable_permission_checks(false);
});
PeerConnectionInterface::RTCConfiguration client_1_config;
webrtc::PeerConnectionInterface::IceServer ice_server_1;
ice_server_1.urls.push_back("turn:88.88.88.0:3478");
ice_server_1.username = "test";
ice_server_1.password = "test";
client_1_config.servers.push_back(ice_server_1);
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
client_1_config.presume_writable_when_fully_relayed = true;
PeerConnectionInterface::RTCConfiguration client_2_config;
webrtc::PeerConnectionInterface::IceServer ice_server_2;
ice_server_2.urls.push_back("turn:99.99.99.0:3478");
ice_server_2.username = "test";
ice_server_2.password = "test";
client_2_config.servers.push_back(ice_server_2);
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
client_2_config.presume_writable_when_fully_relayed = true;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
// Set up the simulated delays.
SetSignalingDelayMs(signaling_trip_delay_ms);
ConnectFakeSignaling();
virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
virtual_socket_server()->UpdateDelayDistribution();
// Set "offer to receive audio/video" without adding any tracks, so we just
// set up ICE/DTLS with no media.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
EXPECT_TRUE_SIMULATED_WAIT(DtlsConnected(), total_connection_time_ms,
fake_clock);
// Closing the PeerConnections destroys the ports before the ScopedFakeClock.
// If this is not done a DCHECK can be hit in ports.cc, because a large
// negative number is calculated for the rtt due to the global clock changing.
caller()->pc()->Close();
callee()->pc()->Close();
}
// Verify that a TurnCustomizer passed in through RTCConfiguration
// is actually used by the underlying TURN candidate pair.
// Note that turnport_unittest.cc contains more detailed, lower-level tests.
TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
3478};
static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
0};
static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
3478};
static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
0};
CreateTurnServer(turn_server_1_internal_address,
turn_server_1_external_address);
CreateTurnServer(turn_server_2_internal_address,
turn_server_2_external_address);
PeerConnectionInterface::RTCConfiguration client_1_config;
webrtc::PeerConnectionInterface::IceServer ice_server_1;
ice_server_1.urls.push_back("turn:88.88.88.0:3478");
ice_server_1.username = "test";
ice_server_1.password = "test";
client_1_config.servers.push_back(ice_server_1);
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
auto* customizer1 = CreateTurnCustomizer();
client_1_config.turn_customizer = customizer1;
PeerConnectionInterface::RTCConfiguration client_2_config;
webrtc::PeerConnectionInterface::IceServer ice_server_2;
ice_server_2.urls.push_back("turn:99.99.99.0:3478");
ice_server_2.username = "test";
ice_server_2.password = "test";
client_2_config.servers.push_back(ice_server_2);
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
auto* customizer2 = CreateTurnCustomizer();
client_2_config.turn_customizer = customizer2;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
ConnectFakeSignaling();
// Set "offer to receive audio/video" without adding any tracks, so we just
// set up ICE/DTLS with no media.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ExpectTurnCustomizerCountersIncremented(customizer1);
ExpectTurnCustomizerCountersIncremented(customizer2);
}
// Verifies that you can use TCP instead of UDP to connect to a TURN server and
// send media between the caller and the callee.
TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
3478};
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
// Enable TCP for the fake turn server.
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
cricket::PROTO_TCP);
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
// Do normal offer/answer and wait for ICE to complete.
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
EXPECT_TRUE(ExpectNewFrames(media_expectations));
}
// Verify that a SSLCertificateVerifier passed in through
// PeerConnectionDependencies is actually used by the underlying SSL
// implementation to determine whether a certificate presented by the TURN
// server is accepted by the client. Note that openssladapter_unittest.cc
// contains more detailed, lower-level tests.
TEST_P(PeerConnectionIntegrationTest,
SSLCertificateVerifierUsedForTurnConnections) {
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
3478};
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
// Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
// that host name verification passes on the fake certificate.
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
cricket::PROTO_TLS, "88.88.88.0");
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
// Setting the type to kRelay forces the connection to go through a TURN
// server.
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
// Get a copy to the pointer so we can verify calls later.
rtc::TestCertificateVerifier* client_1_cert_verifier =
new rtc::TestCertificateVerifier();
client_1_cert_verifier->verify_certificate_ = true;
rtc::TestCertificateVerifier* client_2_cert_verifier =
new rtc::TestCertificateVerifier();
client_2_cert_verifier->verify_certificate_ = true;
// Create the dependencies with the test certificate verifier.
webrtc::PeerConnectionDependencies client_1_deps(nullptr);
client_1_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
webrtc::PeerConnectionDependencies client_2_deps(nullptr);
client_2_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
client_1_config, std::move(client_1_deps), client_2_config,
std::move(client_2_deps)));
ConnectFakeSignaling();
// Set "offer to receive audio/video" without adding any tracks, so we just
// set up ICE/DTLS with no media.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
}
TEST_P(PeerConnectionIntegrationTest,
SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection) {
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
3478};
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
// Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
// that host name verification passes on the fake certificate.
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
cricket::PROTO_TLS, "88.88.88.0");
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
// Setting the type to kRelay forces the connection to go through a TURN
// server.
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
// Get a copy to the pointer so we can verify calls later.
rtc::TestCertificateVerifier* client_1_cert_verifier =
new rtc::TestCertificateVerifier();
client_1_cert_verifier->verify_certificate_ = false;
rtc::TestCertificateVerifier* client_2_cert_verifier =
new rtc::TestCertificateVerifier();
client_2_cert_verifier->verify_certificate_ = false;
// Create the dependencies with the test certificate verifier.
webrtc::PeerConnectionDependencies client_1_deps(nullptr);
client_1_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
webrtc::PeerConnectionDependencies client_2_deps(nullptr);
client_2_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
client_1_config, std::move(client_1_deps), client_2_config,
std::move(client_2_deps)));
ConnectFakeSignaling();
// Set "offer to receive audio/video" without adding any tracks, so we just
// set up ICE/DTLS with no media.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
bool wait_res = true;
// TODO(bugs.webrtc.org/9219): When IceConnectionState is implemented
// properly, should be able to just wait for a state of "failed" instead of
// waiting a fixed 10 seconds.
WAIT_(DtlsConnected(), kDefaultTimeout, wait_res);
ASSERT_FALSE(wait_res);
EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
}
// Test that audio and video flow end-to-end when codec names don't use the
// expected casing, given that they're supposed to be case insensitive. To test
// this, all but one codec is removed from each media description, and its
// casing is changed.
//
// In the past, this has regressed and caused crashes/black video, due to the
// fact that code at some layers was doing case-insensitive comparisons and
// code at other layers was not.
TEST_P(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Remove all but one audio/video codec (opus and VP8), and change the
// casing of the caller's generated offer.
caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) {
cricket::AudioContentDescription* audio =
GetFirstAudioContentDescription(description);
ASSERT_NE(nullptr, audio);
auto audio_codecs = audio->codecs();
audio_codecs.erase(std::remove_if(audio_codecs.begin(), audio_codecs.end(),
[](const cricket::AudioCodec& codec) {
return codec.name != "opus";
}),
audio_codecs.end());
ASSERT_EQ(1u, audio_codecs.size());
audio_codecs[0].name = "OpUs";
audio->set_codecs(audio_codecs);
cricket::VideoContentDescription* video =
GetFirstVideoContentDescription(description);
ASSERT_NE(nullptr, video);
auto video_codecs = video->codecs();
video_codecs.erase(std::remove_if(video_codecs.begin(), video_codecs.end(),
[](const cricket::VideoCodec& codec) {
return codec.name != "VP8";
}),
video_codecs.end());
ASSERT_EQ(1u, video_codecs.size());
video_codecs[0].name = "vP8";
video->set_codecs(video_codecs);
});
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Verify frames are still received end-to-end.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for one audio frame to be received by the callee.
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio(1);
ASSERT_TRUE(ExpectNewFrames(media_expectations));
ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
auto receiver = callee()->pc()->GetReceivers()[0];
ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
auto sources = receiver->GetSources();
ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
sources[0].source_id());
EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
}
TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for one video frame to be received by the callee.
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeVideo(1);
ASSERT_TRUE(ExpectNewFrames(media_expectations));
ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
auto receiver = callee()->pc()->GetReceivers()[0];
ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_VIDEO);
auto sources = receiver->GetSources();
ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
sources[0].source_id());
EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
}
// Test that if a track is removed and added again with a different stream ID,
// the new stream ID is successfully communicated in SDP and media continues to
// flow end-to-end.
// TODO(webrtc.bugs.org/8734): This test does not work for Unified Plan because
// it will not reuse a transceiver that has already been sending. After creating
// a new transceiver it tries to create an offer with two senders of the same
// track ids and it fails.
TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add track using stream 1, do offer/answer.
rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
caller()->CreateLocalAudioTrack();
rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
caller()->AddTrack(track, {"stream_1"});
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
{
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio(1);
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Remove the sender, and create a new one with the new stream.
caller()->pc()->RemoveTrack(sender);
sender = caller()->AddTrack(track, {"stream_2"});
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for additional audio frames to be received by the callee.
{
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
}
TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
auto output = absl::make_unique<testing::NiceMock<MockRtcEventLogOutput>>();
ON_CALL(*output, IsActive()).WillByDefault(testing::Return(true));
ON_CALL(*output, Write(::testing::_)).WillByDefault(testing::Return(true));
EXPECT_CALL(*output, Write(::testing::_)).Times(::testing::AtLeast(1));
EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
std::move(output), webrtc::RtcEventLog::kImmediateOutput));
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
}
// Test that if candidates are only signaled by applying full session
// descriptions (instead of using AddIceCandidate), the peers can connect to
// each other and exchange media.
TEST_P(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
// Each side will signal the session descriptions but not candidates.
ConnectFakeSignalingForSdpOnly();
// Add audio video track and exchange the initial offer/answer with media
// information only. This will start ICE gathering on each side.
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
// Wait for all candidates to be gathered on both the caller and callee.
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
caller()->ice_gathering_state(), kDefaultTimeout);
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
callee()->ice_gathering_state(), kDefaultTimeout);
// The candidates will now be included in the session description, so
// signaling them will start the ICE connection.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure that media flows in both directions.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Test that SetAudioPlayout can be used to disable audio playout from the
// start, then later enable it. This may be useful, for example, if the caller
// needs to play a local ringtone until some event occurs, after which it
// switches to playing the received audio.
TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Set up audio-only call where audio playout is disabled on caller's side.
caller()->pc()->SetAudioPlayout(false);
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Pump messages for a second.
WAIT(false, 1000);
// Since audio playout is disabled, the caller shouldn't have received
// anything (at the playout level, at least).
EXPECT_EQ(0, caller()->audio_frames_received());
// As a sanity check, make sure the callee (for which playout isn't disabled)
// did still see frames on its audio level.
ASSERT_GT(callee()->audio_frames_received(), 0);
// Enable playout again, and ensure audio starts flowing.
caller()->pc()->SetAudioPlayout(true);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
double GetAudioEnergyStat(PeerConnectionWrapper* pc) {
auto report = pc->NewGetStats();
auto track_stats_list =
report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
const webrtc::RTCMediaStreamTrackStats* remote_track_stats = nullptr;
for (const auto* track_stats : track_stats_list) {
if (track_stats->remote_source.is_defined() &&
*track_stats->remote_source) {
remote_track_stats = track_stats;
break;
}
}
if (!remote_track_stats->total_audio_energy.is_defined()) {
return 0.0;
}
return *remote_track_stats->total_audio_energy;
}
// Test that if audio playout is disabled via the SetAudioPlayout() method, then
// incoming audio is still processed and statistics are generated.
TEST_P(PeerConnectionIntegrationTest,
DisableAudioPlayoutStillGeneratesAudioStats) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Set up audio-only call where playout is disabled but audio-processing is
// still active.
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->pc()->SetAudioPlayout(false);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for the callee to receive audio stats.
EXPECT_TRUE_WAIT(GetAudioEnergyStat(caller()) > 0, kMaxWaitForFramesMs);
}
// Test that SetAudioRecording can be used to disable audio recording from the
// start, then later enable it. This may be useful, for example, if the caller
// wants to ensure that no audio resources are active before a certain state
// is reached.
TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Set up audio-only call where audio recording is disabled on caller's side.
caller()->pc()->SetAudioRecording(false);
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Pump messages for a second.
WAIT(false, 1000);
// Since caller has disabled audio recording, the callee shouldn't have
// received anything.
EXPECT_EQ(0, callee()->audio_frames_received());
// As a sanity check, make sure the caller did still see frames on its
// audio level since audio recording is enabled on the calle side.
ASSERT_GT(caller()->audio_frames_received(), 0);
// Enable audio recording again, and ensure audio starts flowing.
caller()->pc()->SetAudioRecording(true);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Test that after closing PeerConnections, they stop sending any packets (ICE,
// DTLS, RTP...).
TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
// Set up audio/video/data, wait for some frames to be received.
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
#ifdef HAVE_SCTP
caller()->CreateDataChannel();
#endif
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
// Close PeerConnections.
caller()->pc()->Close();
callee()->pc()->Close();
// Pump messages for a second, and ensure no new packets end up sent.
uint32_t sent_packets_a = virtual_socket_server()->sent_packets();
WAIT(false, 1000);
uint32_t sent_packets_b = virtual_socket_server()->sent_packets();
EXPECT_EQ(sent_packets_a, sent_packets_b);
}
// Test that transport stats are generated by the RTCStatsCollector for a
// connection that only involves data channels. This is a regression test for
// crbug.com/826972.
#ifdef HAVE_SCTP
TEST_P(PeerConnectionIntegrationTest,
TransportStatsReportedForDataChannelOnlyConnection) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
auto caller_report = caller()->NewGetStats();
EXPECT_EQ(1u, caller_report->GetStatsOfType<RTCTransportStats>().size());
auto callee_report = callee()->NewGetStats();
EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
}
#endif // HAVE_SCTP
TEST_P(PeerConnectionIntegrationTest,
IceEventsGeneratedAndLoggedInRtcEventLog) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithFakeRtcEventLog());
ConnectFakeSignaling();
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->event_log_factory());
ASSERT_NE(nullptr, callee()->event_log_factory());
webrtc::FakeRtcEventLog* caller_event_log =
static_cast<webrtc::FakeRtcEventLog*>(
caller()->event_log_factory()->last_log_created());
webrtc::FakeRtcEventLog* callee_event_log =
static_cast<webrtc::FakeRtcEventLog*>(
callee()->event_log_factory()->last_log_created());
ASSERT_NE(nullptr, caller_event_log);
ASSERT_NE(nullptr, callee_event_log);
int caller_ice_config_count = caller_event_log->GetEventCount(
webrtc::RtcEvent::Type::IceCandidatePairConfig);
int caller_ice_event_count = caller_event_log->GetEventCount(
webrtc::RtcEvent::Type::IceCandidatePairEvent);
int callee_ice_config_count = callee_event_log->GetEventCount(
webrtc::RtcEvent::Type::IceCandidatePairConfig);
int callee_ice_event_count = callee_event_log->GetEventCount(
webrtc::RtcEvent::Type::IceCandidatePairEvent);
EXPECT_LT(0, caller_ice_config_count);
EXPECT_LT(0, caller_ice_event_count);
EXPECT_LT(0, callee_ice_config_count);
EXPECT_LT(0, callee_ice_event_count);
}
INSTANTIATE_TEST_CASE_P(PeerConnectionIntegrationTest,
PeerConnectionIntegrationTest,
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
// Tests that verify interoperability between Plan B and Unified Plan
// PeerConnections.
class PeerConnectionIntegrationInteropTest
: public PeerConnectionIntegrationBaseTest,
public ::testing::WithParamInterface<
std::tuple<SdpSemantics, SdpSemantics>> {
protected:
// Setting the SdpSemantics for the base test to kDefault does not matter
// because we specify not to use the test semantics when creating
// PeerConnectionWrappers.
PeerConnectionIntegrationInteropTest()
: PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB),
caller_semantics_(std::get<0>(GetParam())),
callee_semantics_(std::get<1>(GetParam())) {}
bool CreatePeerConnectionWrappersWithSemantics() {
return CreatePeerConnectionWrappersWithSdpSemantics(caller_semantics_,
callee_semantics_);
}
const SdpSemantics caller_semantics_;
const SdpSemantics callee_semantics_;
};
TEST_P(PeerConnectionIntegrationInteropTest, NoMediaLocalToNoMediaRemote) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
ConnectFakeSignaling();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
}
TEST_P(PeerConnectionIntegrationInteropTest, OneAudioLocalToNoMediaRemote) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
ConnectFakeSignaling();
auto audio_sender = caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Verify that one audio receiver has been created on the remote and that it
// has the same track ID as the sending track.
auto receivers = callee()->pc()->GetReceivers();
ASSERT_EQ(1u, receivers.size());
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, receivers[0]->media_type());
EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id());
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
TEST_P(PeerConnectionIntegrationInteropTest, OneAudioOneVideoToNoMediaRemote) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
ConnectFakeSignaling();
auto video_sender = caller()->AddVideoTrack();
auto audio_sender = caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Verify that one audio and one video receiver have been created on the
// remote and that they have the same track IDs as the sending tracks.
auto audio_receivers =
callee()->GetReceiversOfType(cricket::MEDIA_TYPE_AUDIO);
ASSERT_EQ(1u, audio_receivers.size());
EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id());
auto video_receivers =
callee()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO);
ASSERT_EQ(1u, video_receivers.size());
EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id());
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
TEST_P(PeerConnectionIntegrationInteropTest,
OneAudioOneVideoLocalToOneAudioOneVideoRemote) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
TEST_P(PeerConnectionIntegrationInteropTest,
ReverseRolesOneAudioLocalToOneVideoRemote) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
ConnectFakeSignaling();
caller()->AddAudioTrack();
callee()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Verify that only the audio track has been negotiated.
EXPECT_EQ(0u, caller()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO).size());
// Might also check that the callee's NegotiationNeeded flag is set.
// Reverse roles.
callee()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
MediaExpectations media_expectations;
media_expectations.CallerExpectsSomeVideo();
media_expectations.CalleeExpectsSomeAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
INSTANTIATE_TEST_CASE_P(
PeerConnectionIntegrationTest,
PeerConnectionIntegrationInteropTest,
Values(std::make_tuple(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
std::make_tuple(SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB)));
// Test that if the Unified Plan side offers two video tracks then the Plan B
// side will only see the first one and ignore the second.
TEST_F(PeerConnectionIntegrationTestPlanB, TwoVideoUnifiedPlanToNoMediaPlanB) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithSdpSemantics(
SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB));
ConnectFakeSignaling();
auto first_sender = caller()->AddVideoTrack();
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Verify that there is only one receiver and it corresponds to the first
// added track.
auto receivers = callee()->pc()->GetReceivers();
ASSERT_EQ(1u, receivers.size());
EXPECT_TRUE(receivers[0]->track()->enabled());
EXPECT_EQ(first_sender->track()->id(), receivers[0]->track()->id());
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// Test that if the initial offer tagged BUNDLE section is rejected due to its
// associated RtpTransceiver being stopped and another transceiver is added,
// then renegotiation causes the callee to receive the new video track without
// error.
// This is a regression test for bugs.webrtc.org/9954
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
ReOfferWithStoppedBundleTaggedTransceiver) {
RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
auto audio_transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
ASSERT_TRUE(audio_transceiver_or_error.ok());
auto audio_transceiver = audio_transceiver_or_error.MoveValue();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
{
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeAudio();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
audio_transceiver->Stop();
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
{
MediaExpectations media_expectations;
media_expectations.CalleeExpectsSomeVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
}
} // namespace
} // namespace webrtc
#endif // if !defined(THREAD_SANITIZER)
Reference in New Issue View Git Blame Copy Permalink