blob: 981cec778a690108de0b21850e4c3328a035cabe [file] [log] [blame]
/*
* 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.
*/
#include <stdio.h>
#include <algorithm>
#include <list>
#include <map>
#include <memory>
#include <utility>
#include <vector>
#include "webrtc/api/dtmfsender.h"
#include "webrtc/api/fakemetricsobserver.h"
#include "webrtc/api/localaudiosource.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnection.h"
#include "webrtc/api/peerconnectionfactory.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/test/fakeaudiocapturemodule.h"
#include "webrtc/api/test/fakeconstraints.h"
#include "webrtc/api/test/fakeperiodicvideocapturer.h"
#include "webrtc/api/test/fakertccertificategenerator.h"
#include "webrtc/api/test/fakevideotrackrenderer.h"
#include "webrtc/api/test/mockpeerconnectionobservers.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/virtualsocketserver.h"
#include "webrtc/media/engine/fakewebrtcvideoengine.h"
#include "webrtc/p2p/base/p2pconstants.h"
#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/p2p/base/testturnserver.h"
#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/pc/mediasession.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
LOG(LS_INFO) << "Feature disabled... skipping"; \
return; \
}
using cricket::ContentInfo;
using cricket::FakeWebRtcVideoDecoder;
using cricket::FakeWebRtcVideoDecoderFactory;
using cricket::FakeWebRtcVideoEncoder;
using cricket::FakeWebRtcVideoEncoderFactory;
using cricket::MediaContentDescription;
using webrtc::DataBuffer;
using webrtc::DataChannelInterface;
using webrtc::DtmfSender;
using webrtc::DtmfSenderInterface;
using webrtc::DtmfSenderObserverInterface;
using webrtc::FakeConstraints;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
using webrtc::MockCreateSessionDescriptionObserver;
using webrtc::MockDataChannelObserver;
using webrtc::MockSetSessionDescriptionObserver;
using webrtc::MockStatsObserver;
using webrtc::ObserverInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionFactory;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollectionInterface;
namespace {
static const int kMaxWaitMs = 10000;
// Disable for TSan v2, see
// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
// This declaration is also #ifdef'd as it causes uninitialized-variable
// warnings.
#if !defined(THREAD_SANITIZER)
static const int kMaxWaitForStatsMs = 3000;
#endif
static const int kMaxWaitForActivationMs = 5000;
static const int kMaxWaitForFramesMs = 10000;
static const int kEndAudioFrameCount = 3;
static const int kEndVideoFrameCount = 3;
static const char kStreamLabelBase[] = "stream_label";
static const char kVideoTrackLabelBase[] = "video_track";
static const char kAudioTrackLabelBase[] = "audio_track";
static const char kDataChannelLabel[] = "data_channel";
// Disable for TSan v2, see
// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
// This declaration is also #ifdef'd as it causes unused-variable errors.
#if !defined(THREAD_SANITIZER)
// SRTP cipher name negotiated by the tests. This must be updated if the
// default changes.
static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32;
static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;
#endif
// Used to simulate signaling ICE/SDP between two PeerConnections.
enum Message { MSG_SDP_MESSAGE, MSG_ICE_MESSAGE };
struct SdpMessage {
std::string type;
std::string msg;
};
struct IceMessage {
std::string sdp_mid;
int sdp_mline_index;
std::string msg;
};
static void RemoveLinesFromSdp(const std::string& line_start,
std::string* sdp) {
const char kSdpLineEnd[] = "\r\n";
size_t ssrc_pos = 0;
while ((ssrc_pos = sdp->find(line_start, ssrc_pos)) !=
std::string::npos) {
size_t end_ssrc = sdp->find(kSdpLineEnd, ssrc_pos);
sdp->erase(ssrc_pos, end_ssrc - ssrc_pos + strlen(kSdpLineEnd));
}
}
bool StreamsHaveAudioTrack(StreamCollectionInterface* streams) {
for (size_t idx = 0; idx < streams->count(); idx++) {
auto stream = streams->at(idx);
if (stream->GetAudioTracks().size() > 0) {
return true;
}
}
return false;
}
bool StreamsHaveVideoTrack(StreamCollectionInterface* streams) {
for (size_t idx = 0; idx < streams->count(); idx++) {
auto stream = streams->at(idx);
if (stream->GetVideoTracks().size() > 0) {
return true;
}
}
return false;
}
class SignalingMessageReceiver {
public:
virtual void ReceiveSdpMessage(const std::string& type,
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:
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() { return first_packet_received_; }
virtual ~MockRtpReceiverObserver() {}
private:
bool first_packet_received_ = false;
cricket::MediaType expected_media_type_;
};
class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
public SignalingMessageReceiver,
public ObserverInterface,
public rtc::MessageHandler {
public:
// If |config| is not provided, uses a default constructed RTCConfiguration.
static PeerConnectionTestClient* CreateClientWithDtlsIdentityStore(
const std::string& id,
const MediaConstraintsInterface* constraints,
const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
bool prefer_constraint_apis,
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
if (!client->Init(constraints, options, config, std::move(cert_generator),
prefer_constraint_apis, network_thread, worker_thread)) {
delete client;
return nullptr;
}
return client;
}
static PeerConnectionTestClient* CreateClient(
const std::string& id,
const MediaConstraintsInterface* constraints,
const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
new FakeRTCCertificateGenerator() : nullptr);
return CreateClientWithDtlsIdentityStore(id, constraints, options, config,
std::move(cert_generator), true,
network_thread, worker_thread);
}
static PeerConnectionTestClient* CreateClientPreferNoConstraints(
const std::string& id,
const PeerConnectionFactory::Options* options,
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
new FakeRTCCertificateGenerator() : nullptr);
return CreateClientWithDtlsIdentityStore(id, nullptr, options, nullptr,
std::move(cert_generator), false,
network_thread, worker_thread);
}
~PeerConnectionTestClient() {
}
void Negotiate() { Negotiate(true, true); }
void Negotiate(bool audio, bool video) {
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer));
if (offer->description()->GetContentByName("audio")) {
offer->description()->GetContentByName("audio")->rejected = !audio;
}
if (offer->description()->GetContentByName("video")) {
offer->description()->GetContentByName("video")->rejected = !video;
}
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
EXPECT_TRUE(DoSetLocalDescription(offer.release()));
SendSdpMessage(webrtc::SessionDescriptionInterface::kOffer, sdp);
}
void SendSdpMessage(const std::string& type, std::string& msg) {
if (signaling_delay_ms_ == 0) {
if (signaling_message_receiver_) {
signaling_message_receiver_->ReceiveSdpMessage(type, msg);
}
} else {
rtc::Thread::Current()->PostDelayed(
RTC_FROM_HERE, signaling_delay_ms_, this, MSG_SDP_MESSAGE,
new rtc::TypedMessageData<SdpMessage>({type, msg}));
}
}
void SendIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) {
if (signaling_delay_ms_ == 0) {
if (signaling_message_receiver_) {
signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index,
msg);
}
} else {
rtc::Thread::Current()->PostDelayed(RTC_FROM_HERE, signaling_delay_ms_,
this, MSG_ICE_MESSAGE,
new rtc::TypedMessageData<IceMessage>(
{sdp_mid, sdp_mline_index, msg}));
}
}
// MessageHandler callback.
void OnMessage(rtc::Message* msg) override {
switch (msg->message_id) {
case MSG_SDP_MESSAGE: {
auto sdp_message =
static_cast<rtc::TypedMessageData<SdpMessage>*>(msg->pdata);
if (signaling_message_receiver_) {
signaling_message_receiver_->ReceiveSdpMessage(
sdp_message->data().type, sdp_message->data().msg);
}
delete sdp_message;
break;
}
case MSG_ICE_MESSAGE: {
auto ice_message =
static_cast<rtc::TypedMessageData<IceMessage>*>(msg->pdata);
if (signaling_message_receiver_) {
signaling_message_receiver_->ReceiveIceMessage(
ice_message->data().sdp_mid, ice_message->data().sdp_mline_index,
ice_message->data().msg);
}
delete ice_message;
break;
}
default:
RTC_CHECK(false);
}
}
// SignalingMessageReceiver callback.
void ReceiveSdpMessage(const std::string& type, std::string& msg) override {
FilterIncomingSdpMessage(&msg);
if (type == webrtc::SessionDescriptionInterface::kOffer) {
HandleIncomingOffer(msg);
} else {
HandleIncomingAnswer(msg);
}
}
// SignalingMessageReceiver callback.
void ReceiveIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) override {
LOG(INFO) << id_ << "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 OnAddStream(
rtc::scoped_refptr<MediaStreamInterface> media_stream) override {
media_stream->RegisterObserver(this);
for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) {
const std::string id = media_stream->GetVideoTracks()[i]->id();
ASSERT_TRUE(fake_video_renderers_.find(id) ==
fake_video_renderers_.end());
fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
media_stream->GetVideoTracks()[i]));
}
}
void OnRemoveStream(
rtc::scoped_refptr<MediaStreamInterface> media_stream) override {}
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
EXPECT_EQ(pc()->ice_connection_state(), new_state);
}
void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
EXPECT_EQ(pc()->ice_gathering_state(), new_state);
}
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
LOG(INFO) << id_ << "OnIceCandidate";
std::string ice_sdp;
EXPECT_TRUE(candidate->ToString(&ice_sdp));
if (signaling_message_receiver_ == nullptr) {
// Remote party may be deleted.
return;
}
SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
}
// MediaStreamInterface callback
void OnChanged() override {
// Track added or removed from MediaStream, so update our renderers.
rtc::scoped_refptr<StreamCollectionInterface> remote_streams =
pc()->remote_streams();
// Remove renderers for tracks that were removed.
for (auto it = fake_video_renderers_.begin();
it != fake_video_renderers_.end();) {
if (remote_streams->FindVideoTrack(it->first) == nullptr) {
auto to_remove = it++;
removed_fake_video_renderers_.push_back(std::move(to_remove->second));
fake_video_renderers_.erase(to_remove);
} else {
++it;
}
}
// Create renderers for new video tracks.
for (size_t stream_index = 0; stream_index < remote_streams->count();
++stream_index) {
MediaStreamInterface* remote_stream = remote_streams->at(stream_index);
for (size_t track_index = 0;
track_index < remote_stream->GetVideoTracks().size();
++track_index) {
const std::string id =
remote_stream->GetVideoTracks()[track_index]->id();
if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
continue;
}
fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
remote_stream->GetVideoTracks()[track_index]));
}
}
}
void SetVideoConstraints(const webrtc::FakeConstraints& video_constraint) {
video_constraints_ = video_constraint;
}
void AddMediaStream(bool audio, bool video) {
std::string stream_label =
kStreamLabelBase +
rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count()));
rtc::scoped_refptr<MediaStreamInterface> stream =
peer_connection_factory_->CreateLocalMediaStream(stream_label);
if (audio && can_receive_audio()) {
stream->AddTrack(CreateLocalAudioTrack(stream_label));
}
if (video && can_receive_video()) {
stream->AddTrack(CreateLocalVideoTrack(stream_label));
}
EXPECT_TRUE(pc()->AddStream(stream));
}
size_t NumberOfLocalMediaStreams() { return pc()->local_streams()->count(); }
bool SessionActive() {
return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
}
// Automatically add a stream when receiving an offer, if we don't have one.
// Defaults to true.
void set_auto_add_stream(bool auto_add_stream) {
auto_add_stream_ = auto_add_stream;
}
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 EnableVideoDecoderFactory() {
video_decoder_factory_enabled_ = true;
fake_video_decoder_factory_->AddSupportedVideoCodecType(
webrtc::kVideoCodecVP8);
}
void IceRestart() {
offer_answer_constraints_.SetMandatoryIceRestart(true);
offer_answer_options_.ice_restart = true;
SetExpectIceRestart(true);
}
void SetExpectIceRestart(bool expect_restart) {
expect_ice_restart_ = expect_restart;
}
bool ExpectIceRestart() const { return expect_ice_restart_; }
void SetExpectIceRenomination(bool expect_renomination) {
expect_ice_renomination_ = expect_renomination;
}
void SetExpectRemoteIceRenomination(bool expect_renomination) {
expect_remote_ice_renomination_ = expect_renomination;
}
bool ExpectIceRenomination() { return expect_ice_renomination_; }
bool ExpectRemoteIceRenomination() { return expect_remote_ice_renomination_; }
// The below 3 methods assume streams will be offered.
// Thus they'll only set the "offer to receive" flag to true if it's
// currently false, not if it's just unset.
void SetReceiveAudioVideo(bool audio, bool video) {
SetReceiveAudio(audio);
SetReceiveVideo(video);
ASSERT_EQ(audio, can_receive_audio());
ASSERT_EQ(video, can_receive_video());
}
void SetReceiveAudio(bool audio) {
if (audio && can_receive_audio()) {
return;
}
offer_answer_constraints_.SetMandatoryReceiveAudio(audio);
offer_answer_options_.offer_to_receive_audio = audio ? 1 : 0;
}
void SetReceiveVideo(bool video) {
if (video && can_receive_video()) {
return;
}
offer_answer_constraints_.SetMandatoryReceiveVideo(video);
offer_answer_options_.offer_to_receive_video = video ? 1 : 0;
}
void SetOfferToReceiveAudioVideo(bool audio, bool video) {
offer_answer_constraints_.SetMandatoryReceiveAudio(audio);
offer_answer_options_.offer_to_receive_audio = audio ? 1 : 0;
offer_answer_constraints_.SetMandatoryReceiveVideo(video);
offer_answer_options_.offer_to_receive_video = video ? 1 : 0;
}
void RemoveMsidFromReceivedSdp(bool remove) { remove_msid_ = remove; }
void RemoveSdesCryptoFromReceivedSdp(bool remove) { remove_sdes_ = remove; }
void RemoveBundleFromReceivedSdp(bool remove) { remove_bundle_ = remove; }
void RemoveCvoFromReceivedSdp(bool remove) { remove_cvo_ = remove; }
bool can_receive_audio() {
bool value;
if (prefer_constraint_apis_) {
if (webrtc::FindConstraint(
&offer_answer_constraints_,
MediaConstraintsInterface::kOfferToReceiveAudio, &value,
nullptr)) {
return value;
}
return true;
}
return offer_answer_options_.offer_to_receive_audio > 0 ||
offer_answer_options_.offer_to_receive_audio ==
PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined;
}
bool can_receive_video() {
bool value;
if (prefer_constraint_apis_) {
if (webrtc::FindConstraint(
&offer_answer_constraints_,
MediaConstraintsInterface::kOfferToReceiveVideo, &value,
nullptr)) {
return value;
}
return true;
}
return offer_answer_options_.offer_to_receive_video > 0 ||
offer_answer_options_.offer_to_receive_video ==
PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined;
}
void OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) override {
LOG(INFO) << id_ << "OnDataChannel";
data_channel_ = data_channel;
data_observer_.reset(new MockDataChannelObserver(data_channel));
}
void CreateDataChannel() { CreateDataChannel(nullptr); }
void CreateDataChannel(const webrtc::DataChannelInit* init) {
data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, init);
ASSERT_TRUE(data_channel_.get() != nullptr);
data_observer_.reset(new MockDataChannelObserver(data_channel_));
}
rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
const std::string& stream_label) {
FakeConstraints constraints;
// Disable highpass filter so that we can get all the test audio frames.
constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
peer_connection_factory_->CreateAudioSource(&constraints);
// TODO(perkj): Test audio source when it is implemented. Currently audio
// always use the default input.
std::string label = stream_label + kAudioTrackLabelBase;
return peer_connection_factory_->CreateAudioTrack(label, source);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
const std::string& stream_label) {
// Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
FakeConstraints source_constraints = video_constraints_;
source_constraints.SetMandatoryMaxFrameRate(10);
cricket::FakeVideoCapturer* fake_capturer =
new webrtc::FakePeriodicVideoCapturer();
fake_capturer->SetRotation(capture_rotation_);
video_capturers_.push_back(fake_capturer);
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
peer_connection_factory_->CreateVideoSource(fake_capturer,
&source_constraints);
std::string label = stream_label + kVideoTrackLabelBase;
rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
peer_connection_factory_->CreateVideoTrack(label, source));
if (!local_video_renderer_) {
local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
}
return track;
}
DataChannelInterface* data_channel() { return data_channel_; }
const MockDataChannelObserver* data_observer() const {
return data_observer_.get();
}
webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
void StopVideoCapturers() {
for (auto* capturer : video_capturers_) {
capturer->Stop();
}
}
void SetCaptureRotation(webrtc::VideoRotation rotation) {
ASSERT_TRUE(video_capturers_.empty());
capture_rotation_ = rotation;
}
bool AudioFramesReceivedCheck(int number_of_frames) const {
return number_of_frames <= fake_audio_capture_module_->frames_received();
}
int audio_frames_received() const {
return fake_audio_capture_module_->frames_received();
}
bool VideoFramesReceivedCheck(int number_of_frames) {
if (video_decoder_factory_enabled_) {
const std::vector<FakeWebRtcVideoDecoder*>& decoders
= fake_video_decoder_factory_->decoders();
if (decoders.empty()) {
return number_of_frames <= 0;
}
// Note - this checks that EACH decoder has the requisite number
// of frames. The video_frames_received() function sums them.
for (FakeWebRtcVideoDecoder* decoder : decoders) {
if (number_of_frames > decoder->GetNumFramesReceived()) {
return false;
}
}
return true;
} else {
if (fake_video_renderers_.empty()) {
return number_of_frames <= 0;
}
for (const auto& pair : fake_video_renderers_) {
if (number_of_frames > pair.second->num_rendered_frames()) {
return false;
}
}
return true;
}
}
int video_frames_received() const {
int total = 0;
if (video_decoder_factory_enabled_) {
const std::vector<FakeWebRtcVideoDecoder*>& decoders =
fake_video_decoder_factory_->decoders();
for (const FakeWebRtcVideoDecoder* decoder : decoders) {
total += decoder->GetNumFramesReceived();
}
} else {
for (const auto& pair : fake_video_renderers_) {
total += pair.second->num_rendered_frames();
}
for (const auto& renderer : removed_fake_video_renderers_) {
total += renderer->num_rendered_frames();
}
}
return total;
}
// Verify the CreateDtmfSender interface
void VerifyDtmf() {
std::unique_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender;
// We can't create a DTMF sender with an invalid audio track or a non local
// track.
EXPECT_TRUE(peer_connection_->CreateDtmfSender(nullptr) == nullptr);
rtc::scoped_refptr<webrtc::AudioTrackInterface> non_localtrack(
peer_connection_factory_->CreateAudioTrack("dummy_track", nullptr));
EXPECT_TRUE(peer_connection_->CreateDtmfSender(non_localtrack) == nullptr);
// We should be able to create a DTMF sender from a local track.
webrtc::AudioTrackInterface* localtrack =
peer_connection_->local_streams()->at(0)->GetAudioTracks()[0];
dtmf_sender = peer_connection_->CreateDtmfSender(localtrack);
EXPECT_TRUE(dtmf_sender.get() != nullptr);
dtmf_sender->RegisterObserver(observer.get());
// Test the DtmfSender object just created.
EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));
// We don't need to verify that the DTMF tones are actually sent out because
// that is already covered by the tests of the lower level components.
EXPECT_TRUE_WAIT(observer->completed(), kMaxWaitMs);
std::vector<std::string> tones;
tones.push_back("1");
tones.push_back("a");
tones.push_back("");
observer->Verify(tones);
dtmf_sender->UnregisterObserver();
}
// Verifies that the SessionDescription have rejected the appropriate media
// content.
void VerifyRejectedMediaInSessionDescription() {
ASSERT_TRUE(peer_connection_->remote_description() != nullptr);
ASSERT_TRUE(peer_connection_->local_description() != nullptr);
const cricket::SessionDescription* remote_desc =
peer_connection_->remote_description()->description();
const cricket::SessionDescription* local_desc =
peer_connection_->local_description()->description();
const ContentInfo* remote_audio_content = GetFirstAudioContent(remote_desc);
if (remote_audio_content) {
const ContentInfo* audio_content =
GetFirstAudioContent(local_desc);
EXPECT_EQ(can_receive_audio(), !audio_content->rejected);
}
const ContentInfo* remote_video_content = GetFirstVideoContent(remote_desc);
if (remote_video_content) {
const ContentInfo* video_content =
GetFirstVideoContent(local_desc);
EXPECT_EQ(can_receive_video(), !video_content->rejected);
}
}
void VerifyLocalIceUfragAndPassword() {
ASSERT_TRUE(peer_connection_->local_description() != nullptr);
const cricket::SessionDescription* desc =
peer_connection_->local_description()->description();
const cricket::ContentInfos& contents = desc->contents();
for (size_t index = 0; index < contents.size(); ++index) {
if (contents[index].rejected)
continue;
const cricket::TransportDescription* transport_desc =
desc->GetTransportDescriptionByName(contents[index].name);
std::map<int, IceUfragPwdPair>::const_iterator ufragpair_it =
ice_ufrag_pwd_.find(static_cast<int>(index));
if (ufragpair_it == ice_ufrag_pwd_.end()) {
ASSERT_FALSE(ExpectIceRestart());
ice_ufrag_pwd_[static_cast<int>(index)] =
IceUfragPwdPair(transport_desc->ice_ufrag, transport_desc->ice_pwd);
} else if (ExpectIceRestart()) {
const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second;
EXPECT_NE(ufrag_pwd.first, transport_desc->ice_ufrag);
EXPECT_NE(ufrag_pwd.second, transport_desc->ice_pwd);
} else {
const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second;
EXPECT_EQ(ufrag_pwd.first, transport_desc->ice_ufrag);
EXPECT_EQ(ufrag_pwd.second, transport_desc->ice_pwd);
}
}
}
void VerifyLocalIceRenomination() {
ASSERT_TRUE(peer_connection_->local_description() != nullptr);
const cricket::SessionDescription* desc =
peer_connection_->local_description()->description();
const cricket::ContentInfos& contents = desc->contents();
for (auto content : contents) {
if (content.rejected)
continue;
const cricket::TransportDescription* transport_desc =
desc->GetTransportDescriptionByName(content.name);
const auto& options = transport_desc->transport_options;
auto iter = std::find(options.begin(), options.end(),
cricket::ICE_RENOMINATION_STR);
EXPECT_EQ(ExpectIceRenomination(), iter != options.end());
}
}
void VerifyRemoteIceRenomination() {
ASSERT_TRUE(peer_connection_->remote_description() != nullptr);
const cricket::SessionDescription* desc =
peer_connection_->remote_description()->description();
const cricket::ContentInfos& contents = desc->contents();
for (auto content : contents) {
if (content.rejected)
continue;
const cricket::TransportDescription* transport_desc =
desc->GetTransportDescriptionByName(content.name);
const auto& options = transport_desc->transport_options;
auto iter = std::find(options.begin(), options.end(),
cricket::ICE_RENOMINATION_STR);
EXPECT_EQ(ExpectRemoteIceRenomination(), iter != options.end());
}
}
int GetAudioOutputLevelStats(webrtc::MediaStreamTrackInterface* track) {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
EXPECT_NE(0, observer->timestamp());
return observer->AudioOutputLevel();
}
int GetAudioInputLevelStats() {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
EXPECT_NE(0, observer->timestamp());
return observer->AudioInputLevel();
}
int GetBytesReceivedStats(webrtc::MediaStreamTrackInterface* track) {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
EXPECT_NE(0, observer->timestamp());
return observer->BytesReceived();
}
int GetBytesSentStats(webrtc::MediaStreamTrackInterface* track) {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
EXPECT_NE(0, observer->timestamp());
return observer->BytesSent();
}
int GetAvailableReceivedBandwidthStats() {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
EXPECT_NE(0, observer->timestamp());
int bw = observer->AvailableReceiveBandwidth();
return bw;
}
std::string GetDtlsCipherStats() {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
EXPECT_NE(0, observer->timestamp());
return observer->DtlsCipher();
}
std::string GetSrtpCipherStats() {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
EXPECT_NE(0, observer->timestamp());
return observer->SrtpCipher();
}
int rendered_width() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty() ? 1 :
fake_video_renderers_.begin()->second->width();
}
int rendered_height() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty() ? 1 :
fake_video_renderers_.begin()->second->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() : 1;
}
int local_rendered_height() {
return local_video_renderer_ ? local_video_renderer_->height() : 1;
}
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();
}
bool HasLocalAudioTrack() { return StreamsHaveAudioTrack(local_streams()); }
bool HasLocalVideoTrack() { return StreamsHaveVideoTrack(local_streams()); }
webrtc::PeerConnectionInterface::SignalingState signaling_state() {
return pc()->signaling_state();
}
webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
return pc()->ice_connection_state();
}
webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
return pc()->ice_gathering_state();
}
std::vector<std::unique_ptr<MockRtpReceiverObserver>> const&
rtp_receiver_observers() {
return rtp_receiver_observers_;
}
void SetRtpReceiverObservers() {
rtp_receiver_observers_.clear();
for (auto 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));
}
}
private:
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;
}
}
void Verify(const std::vector<std::string>& tones) const {
ASSERT_TRUE(tones_.size() == tones.size());
EXPECT_TRUE(std::equal(tones.begin(), tones.end(), tones_.begin()));
}
bool completed() const { return completed_; }
private:
bool completed_;
std::vector<std::string> tones_;
};
explicit PeerConnectionTestClient(const std::string& id) : id_(id) {}
bool Init(
const MediaConstraintsInterface* constraints,
const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
bool prefer_constraint_apis,
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
EXPECT_TRUE(!peer_connection_);
EXPECT_TRUE(!peer_connection_factory_);
if (!prefer_constraint_apis) {
EXPECT_TRUE(!constraints);
}
prefer_constraint_apis_ = prefer_constraint_apis;
fake_network_manager_.reset(new rtc::FakeNetworkManager());
fake_network_manager_->AddInterface(rtc::SocketAddress("192.168.1.1", 0));
std::unique_ptr<cricket::PortAllocator> port_allocator(
new cricket::BasicPortAllocator(fake_network_manager_.get()));
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
if (fake_audio_capture_module_ == nullptr) {
return false;
}
fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory();
fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory();
rtc::Thread* const signaling_thread = rtc::Thread::Current();
peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
network_thread, worker_thread, signaling_thread,
fake_audio_capture_module_, fake_video_encoder_factory_,
fake_video_decoder_factory_);
if (!peer_connection_factory_) {
return false;
}
if (options) {
peer_connection_factory_->SetOptions(*options);
}
peer_connection_ =
CreatePeerConnection(std::move(port_allocator), constraints, config,
std::move(cert_generator));
return peer_connection_.get() != nullptr;
}
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
std::unique_ptr<cricket::PortAllocator> port_allocator,
const MediaConstraintsInterface* constraints,
const PeerConnectionInterface::RTCConfiguration* config,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) {
// CreatePeerConnection with RTCConfiguration.
PeerConnectionInterface::RTCConfiguration default_config;
if (!config) {
config = &default_config;
}
return peer_connection_factory_->CreatePeerConnection(
*config, constraints, std::move(port_allocator),
std::move(cert_generator), this);
}
void HandleIncomingOffer(const std::string& msg) {
LOG(INFO) << id_ << "HandleIncomingOffer ";
if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) {
// If we are not sending any streams ourselves it is time to add some.
AddMediaStream(true, true);
}
std::unique_ptr<SessionDescriptionInterface> desc(
webrtc::CreateSessionDescription("offer", msg, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
// Set the RtpReceiverObserver after receivers are created.
SetRtpReceiverObservers();
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer));
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
EXPECT_TRUE(DoSetLocalDescription(answer.release()));
SendSdpMessage(webrtc::SessionDescriptionInterface::kAnswer, sdp);
}
void HandleIncomingAnswer(const std::string& msg) {
LOG(INFO) << id_ << "HandleIncomingAnswer";
std::unique_ptr<SessionDescriptionInterface> desc(
webrtc::CreateSessionDescription("answer", msg, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
// Set the RtpReceiverObserver after receivers are created.
SetRtpReceiverObservers();
}
bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
bool offer) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockCreateSessionDescriptionObserver>());
if (prefer_constraint_apis_) {
if (offer) {
pc()->CreateOffer(observer, &offer_answer_constraints_);
} else {
pc()->CreateAnswer(observer, &offer_answer_constraints_);
}
} else {
if (offer) {
pc()->CreateOffer(observer, offer_answer_options_);
} else {
pc()->CreateAnswer(observer, offer_answer_options_);
}
}
EXPECT_EQ_WAIT(true, observer->called(), kMaxWaitMs);
desc->reset(observer->release_desc());
if (observer->result() && ExpectIceRestart()) {
EXPECT_EQ(0u, (*desc)->candidates(0)->count());
}
return observer->result();
}
bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc) {
return DoCreateOfferAnswer(desc, true);
}
bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc) {
return DoCreateOfferAnswer(desc, false);
}
bool DoSetLocalDescription(SessionDescriptionInterface* desc) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockSetSessionDescriptionObserver>());
LOG(INFO) << id_ << "SetLocalDescription ";
pc()->SetLocalDescription(observer, desc);
// Ignore the observer result. If we wait for the result with
// EXPECT_TRUE_WAIT, local ice candidates might be sent to the remote peer
// before the offer which is an error.
// The reason is that EXPECT_TRUE_WAIT uses
// rtc::Thread::Current()->ProcessMessages(1);
// ProcessMessages waits at least 1ms but processes all messages before
// returning. Since this test is synchronous and send messages to the remote
// peer whenever a callback is invoked, this can lead to messages being
// sent to the remote peer in the wrong order.
// TODO(perkj): Find a way to check the result without risking that the
// order of sent messages are changed. Ex- by posting all messages that are
// sent to the remote peer.
return true;
}
bool DoSetRemoteDescription(SessionDescriptionInterface* desc) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockSetSessionDescriptionObserver>());
LOG(INFO) << id_ << "SetRemoteDescription ";
pc()->SetRemoteDescription(observer, desc);
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
return observer->result();
}
// This modifies all received SDP messages before they are processed.
void FilterIncomingSdpMessage(std::string* sdp) {
if (remove_msid_) {
const char kSdpSsrcAttribute[] = "a=ssrc:";
RemoveLinesFromSdp(kSdpSsrcAttribute, sdp);
const char kSdpMsidSupportedAttribute[] = "a=msid-semantic:";
RemoveLinesFromSdp(kSdpMsidSupportedAttribute, sdp);
}
if (remove_bundle_) {
const char kSdpBundleAttribute[] = "a=group:BUNDLE";
RemoveLinesFromSdp(kSdpBundleAttribute, sdp);
}
if (remove_sdes_) {
const char kSdpSdesCryptoAttribute[] = "a=crypto";
RemoveLinesFromSdp(kSdpSdesCryptoAttribute, sdp);
}
if (remove_cvo_) {
const char kSdpCvoExtenstion[] = "urn:3gpp:video-orientation";
RemoveLinesFromSdp(kSdpCvoExtenstion, sdp);
}
}
std::string id_;
std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
bool prefer_constraint_apis_ = true;
bool auto_add_stream_ = true;
typedef std::pair<std::string, std::string> IceUfragPwdPair;
std::map<int, IceUfragPwdPair> ice_ufrag_pwd_;
bool expect_ice_restart_ = false;
bool expect_ice_renomination_ = false;
bool expect_remote_ice_renomination_ = false;
// 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_;
// Needed to keep track of number of frames received when external decoder
// used.
FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr;
FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_ = nullptr;
bool video_decoder_factory_enabled_ = false;
webrtc::FakeConstraints video_constraints_;
// For remote peer communication.
SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
int signaling_delay_ms_ = 0;
// Store references to the video capturers we've created, so that we can stop
// them, if required.
std::vector<cricket::FakeVideoCapturer*> video_capturers_;
webrtc::VideoRotation capture_rotation_ = webrtc::kVideoRotation_0;
// |local_video_renderer_| attached to the first created local video track.
std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
webrtc::FakeConstraints offer_answer_constraints_;
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
bool remove_msid_ = false; // True if MSID should be removed in received SDP.
bool remove_bundle_ =
false; // True if bundle should be removed in received SDP.
bool remove_sdes_ =
false; // True if a=crypto should be removed in received SDP.
// |remove_cvo_| is true if extension urn:3gpp:video-orientation should be
// removed in the received SDP.
bool remove_cvo_ = false;
rtc::scoped_refptr<DataChannelInterface> data_channel_;
std::unique_ptr<MockDataChannelObserver> data_observer_;
std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;
};
class P2PTestConductor : public testing::Test {
public:
P2PTestConductor()
: pss_(new rtc::PhysicalSocketServer),
ss_(new rtc::VirtualSocketServer(pss_.get())),
network_thread_(new rtc::Thread(ss_.get())),
worker_thread_(rtc::Thread::Create()) {
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
}
bool SessionActive() {
return initiating_client_->SessionActive() &&
receiving_client_->SessionActive();
}
// Return true if the number of frames provided have been received
// on the video and audio tracks provided.
bool FramesHaveArrived(int audio_frames_to_receive,
int video_frames_to_receive) {
bool all_good = true;
if (initiating_client_->HasLocalAudioTrack() &&
receiving_client_->can_receive_audio()) {
all_good &=
receiving_client_->AudioFramesReceivedCheck(audio_frames_to_receive);
}
if (initiating_client_->HasLocalVideoTrack() &&
receiving_client_->can_receive_video()) {
all_good &=
receiving_client_->VideoFramesReceivedCheck(video_frames_to_receive);
}
if (receiving_client_->HasLocalAudioTrack() &&
initiating_client_->can_receive_audio()) {
all_good &=
initiating_client_->AudioFramesReceivedCheck(audio_frames_to_receive);
}
if (receiving_client_->HasLocalVideoTrack() &&
initiating_client_->can_receive_video()) {
all_good &=
initiating_client_->VideoFramesReceivedCheck(video_frames_to_receive);
}
return all_good;
}
void VerifyDtmf() {
initiating_client_->VerifyDtmf();
receiving_client_->VerifyDtmf();
}
void TestUpdateOfferWithRejectedContent() {
// Renegotiate, rejecting the video m-line.
initiating_client_->Negotiate(true, false);
ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
int pc1_audio_received = initiating_client_->audio_frames_received();
int pc1_video_received = initiating_client_->video_frames_received();
int pc2_audio_received = receiving_client_->audio_frames_received();
int pc2_video_received = receiving_client_->video_frames_received();
// Wait for some additional audio frames to be received.
EXPECT_TRUE_WAIT(initiating_client_->AudioFramesReceivedCheck(
pc1_audio_received + kEndAudioFrameCount) &&
receiving_client_->AudioFramesReceivedCheck(
pc2_audio_received + kEndAudioFrameCount),
kMaxWaitForFramesMs);
// During this time, we shouldn't have received any additional video frames
// for the rejected video tracks.
EXPECT_EQ(pc1_video_received, initiating_client_->video_frames_received());
EXPECT_EQ(pc2_video_received, receiving_client_->video_frames_received());
}
void VerifyRenderedSize(int width, int height) {
VerifyRenderedSize(width, height, webrtc::kVideoRotation_0);
}
void VerifyRenderedSize(int width,
int height,
webrtc::VideoRotation rotation) {
EXPECT_EQ(width, receiving_client()->rendered_width());
EXPECT_EQ(height, receiving_client()->rendered_height());
EXPECT_EQ(rotation, receiving_client()->rendered_rotation());
EXPECT_EQ(width, initializing_client()->rendered_width());
EXPECT_EQ(height, initializing_client()->rendered_height());
EXPECT_EQ(rotation, initializing_client()->rendered_rotation());
// Verify size of the local preview.
EXPECT_EQ(width, initializing_client()->local_rendered_width());
EXPECT_EQ(height, initializing_client()->local_rendered_height());
}
void VerifySessionDescriptions() {
initiating_client_->VerifyRejectedMediaInSessionDescription();
receiving_client_->VerifyRejectedMediaInSessionDescription();
initiating_client_->VerifyLocalIceUfragAndPassword();
receiving_client_->VerifyLocalIceUfragAndPassword();
}
~P2PTestConductor() {
if (initiating_client_) {
initiating_client_->set_signaling_message_receiver(nullptr);
}
if (receiving_client_) {
receiving_client_->set_signaling_message_receiver(nullptr);
}
}
bool CreateTestClients() { return CreateTestClients(nullptr, nullptr); }
bool CreateTestClients(MediaConstraintsInterface* init_constraints,
MediaConstraintsInterface* recv_constraints) {
return CreateTestClients(init_constraints, nullptr, nullptr,
recv_constraints, nullptr, nullptr);
}
bool CreateTestClients(
const PeerConnectionInterface::RTCConfiguration& init_config,
const PeerConnectionInterface::RTCConfiguration& recv_config) {
return CreateTestClients(nullptr, nullptr, &init_config, nullptr, nullptr,
&recv_config);
}
bool CreateTestClientsThatPreferNoConstraints() {
initiating_client_.reset(
PeerConnectionTestClient::CreateClientPreferNoConstraints(
"Caller: ", nullptr, network_thread_.get(), worker_thread_.get()));
receiving_client_.reset(
PeerConnectionTestClient::CreateClientPreferNoConstraints(
"Callee: ", nullptr, network_thread_.get(), worker_thread_.get()));
if (!initiating_client_ || !receiving_client_) {
return false;
}
// Remember the choice for possible later resets of the clients.
prefer_constraint_apis_ = false;
SetSignalingReceivers();
return true;
}
bool CreateTestClients(
MediaConstraintsInterface* init_constraints,
PeerConnectionFactory::Options* init_options,
const PeerConnectionInterface::RTCConfiguration* init_config,
MediaConstraintsInterface* recv_constraints,
PeerConnectionFactory::Options* recv_options,
const PeerConnectionInterface::RTCConfiguration* recv_config) {
initiating_client_.reset(PeerConnectionTestClient::CreateClient(
"Caller: ", init_constraints, init_options, init_config,
network_thread_.get(), worker_thread_.get()));
receiving_client_.reset(PeerConnectionTestClient::CreateClient(
"Callee: ", recv_constraints, recv_options, recv_config,
network_thread_.get(), worker_thread_.get()));
if (!initiating_client_ || !receiving_client_) {
return false;
}
SetSignalingReceivers();
return true;
}
void SetSignalingReceivers() {
initiating_client_->set_signaling_message_receiver(receiving_client_.get());
receiving_client_->set_signaling_message_receiver(initiating_client_.get());
}
void SetSignalingDelayMs(int delay_ms) {
initiating_client_->set_signaling_delay_ms(delay_ms);
receiving_client_->set_signaling_delay_ms(delay_ms);
}
void SetVideoConstraints(const webrtc::FakeConstraints& init_constraints,
const webrtc::FakeConstraints& recv_constraints) {
initiating_client_->SetVideoConstraints(init_constraints);
receiving_client_->SetVideoConstraints(recv_constraints);
}
void SetCaptureRotation(webrtc::VideoRotation rotation) {
initiating_client_->SetCaptureRotation(rotation);
receiving_client_->SetCaptureRotation(rotation);
}
void EnableVideoDecoderFactory() {
initiating_client_->EnableVideoDecoderFactory();
receiving_client_->EnableVideoDecoderFactory();
}
// This test sets up a call between two parties. Both parties send static
// frames to each other. Once the test is finished the number of sent frames
// is compared to the number of received frames.
void LocalP2PTest() {
if (initiating_client_->NumberOfLocalMediaStreams() == 0) {
initiating_client_->AddMediaStream(true, true);
}
initiating_client_->Negotiate();
// Assert true is used here since next tests are guaranteed to fail and
// would eat up 5 seconds.
ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
VerifySessionDescriptions();
int audio_frame_count = kEndAudioFrameCount;
int video_frame_count = kEndVideoFrameCount;
// TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
if ((!initiating_client_->can_receive_audio() &&
!initiating_client_->can_receive_video()) ||
(!receiving_client_->can_receive_audio() &&
!receiving_client_->can_receive_video())) {
// Neither audio nor video will flow, so connections won't be
// established. There's nothing more to check.
// TODO(hta): Check connection if there's a data channel.
return;
}
// Audio or video is expected to flow, so both clients should reach the
// Connected state, and the offerer (ICE controller) should proceed to
// Completed.
// Note: These tests have been observed to fail under heavy load at
// shorter timeouts, so they may be flaky.
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
initiating_client_->ice_connection_state(),
kMaxWaitForFramesMs);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
receiving_client_->ice_connection_state(),
kMaxWaitForFramesMs);
// The ICE gathering state should end up in kIceGatheringComplete,
// but there's a bug that prevents this at the moment, and the state
// machine is being updated by the WEBRTC WG.
// TODO(hta): Update this check when spec revisions finish.
EXPECT_NE(webrtc::PeerConnectionInterface::kIceGatheringNew,
initiating_client_->ice_gathering_state());
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
receiving_client_->ice_gathering_state(),
kMaxWaitForFramesMs);
// Check that the expected number of frames have arrived.
EXPECT_TRUE_WAIT(FramesHaveArrived(audio_frame_count, video_frame_count),
kMaxWaitForFramesMs);
}
void SetupAndVerifyDtlsCall() {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
LocalP2PTest();
VerifyRenderedSize(640, 480);
}
PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() {
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
new FakeRTCCertificateGenerator() : nullptr);
cert_generator->use_alternate_key();
// Make sure the new client is using a different certificate.
return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
"New Peer: ", &setup_constraints, nullptr, nullptr,
std::move(cert_generator), prefer_constraint_apis_,
network_thread_.get(), worker_thread_.get());
}
void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
// Messages may get lost on the unreliable DataChannel, so we send multiple
// times to avoid test flakiness.
static const size_t kSendAttempts = 5;
for (size_t i = 0; i < kSendAttempts; ++i) {
dc->Send(DataBuffer(data));
}
}
rtc::Thread* network_thread() { return network_thread_.get(); }
rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
PeerConnectionTestClient* initializing_client() {
return initiating_client_.get();
}
// Set the |initiating_client_| to the |client| passed in and return the
// original |initiating_client_|.
PeerConnectionTestClient* set_initializing_client(
PeerConnectionTestClient* client) {
PeerConnectionTestClient* old = initiating_client_.release();
initiating_client_.reset(client);
return old;
}
PeerConnectionTestClient* receiving_client() {
return receiving_client_.get();
}
// Set the |receiving_client_| to the |client| passed in and return the
// original |receiving_client_|.
PeerConnectionTestClient* set_receiving_client(
PeerConnectionTestClient* client) {
PeerConnectionTestClient* old = receiving_client_.release();
receiving_client_.reset(client);
return old;
}
bool AllObserversReceived(
const std::vector<std::unique_ptr<MockRtpReceiverObserver>>& observers) {
for (auto& observer : observers) {
if (!observer->first_packet_received()) {
return false;
}
}
return true;
}
void TestGcmNegotiation(bool local_gcm_enabled, bool remote_gcm_enabled,
int expected_cipher_suite) {
PeerConnectionFactory::Options init_options;
init_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled;
PeerConnectionFactory::Options recv_options;
recv_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled;
ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
&recv_options, nullptr));
rtc::scoped_refptr<webrtc::FakeMetricsObserver>
init_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitMs);
EXPECT_EQ(1,
init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
expected_cipher_suite));
}
private:
// |ss_| is used by |network_thread_| so it must be destroyed later.
std::unique_ptr<rtc::PhysicalSocketServer> pss_;
std::unique_ptr<rtc::VirtualSocketServer> ss_;
// |network_thread_| and |worker_thread_| are used by both
// |initiating_client_| and |receiving_client_| so they must be destroyed
// later.
std::unique_ptr<rtc::Thread> network_thread_;
std::unique_ptr<rtc::Thread> worker_thread_;
std::unique_ptr<PeerConnectionTestClient> initiating_client_;
std::unique_ptr<PeerConnectionTestClient> receiving_client_;
bool prefer_constraint_apis_ = true;
};
// Disable for TSan v2, see
// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
#if !defined(THREAD_SANITIZER)
TEST_F(P2PTestConductor, TestRtpReceiverObserverCallbackFunction) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
EXPECT_TRUE_WAIT(
AllObserversReceived(initializing_client()->rtp_receiver_observers()),
kMaxWaitForFramesMs);
EXPECT_TRUE_WAIT(
AllObserversReceived(receiving_client()->rtp_receiver_observers()),
kMaxWaitForFramesMs);
}
// The observers are expected to fire the signal even if they are set after the
// first packet is received.
TEST_F(P2PTestConductor, TestSetRtpReceiverObserverAfterFirstPacketIsReceived) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
// Reset the RtpReceiverObservers.
initializing_client()->SetRtpReceiverObservers();
receiving_client()->SetRtpReceiverObservers();
EXPECT_TRUE_WAIT(
AllObserversReceived(initializing_client()->rtp_receiver_observers()),
kMaxWaitForFramesMs);
EXPECT_TRUE_WAIT(
AllObserversReceived(receiving_client()->rtp_receiver_observers()),
kMaxWaitForFramesMs);
}
// This test sets up a Jsep call between two parties and test Dtmf.
// TODO(holmer): Disabled due to sometimes crashing on buildbots.
// See issue webrtc/2378.
TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
VerifyDtmf();
}
// This test sets up a Jsep call between two parties and test that we can get a
// video aspect ratio of 16:9.
TEST_F(P2PTestConductor, LocalP2PTest16To9) {
ASSERT_TRUE(CreateTestClients());
FakeConstraints constraint;
double requested_ratio = 640.0/360;
constraint.SetMandatoryMinAspectRatio(requested_ratio);
SetVideoConstraints(constraint, constraint);
LocalP2PTest();
ASSERT_LE(0, initializing_client()->rendered_height());
double initiating_video_ratio =
static_cast<double>(initializing_client()->rendered_width()) /
initializing_client()->rendered_height();
EXPECT_LE(requested_ratio, initiating_video_ratio);
ASSERT_LE(0, receiving_client()->rendered_height());
double receiving_video_ratio =
static_cast<double>(receiving_client()->rendered_width()) /
receiving_client()->rendered_height();
EXPECT_LE(requested_ratio, receiving_video_ratio);
}
// This test sets up a Jsep call between two parties and test that the
// received video has a resolution of 1280*720.
// TODO(mallinath): Enable when
// http://code.google.com/p/webrtc/issues/detail?id=981 is fixed.
TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) {
ASSERT_TRUE(CreateTestClients());
FakeConstraints constraint;
constraint.SetMandatoryMinWidth(1280);
constraint.SetMandatoryMinHeight(720);
SetVideoConstraints(constraint, constraint);
LocalP2PTest();
VerifyRenderedSize(1280, 720);
}
// This test sets up a call between two endpoints that are configured to use
// DTLS key agreement. As a result, DTLS is negotiated and used for transport.
TEST_F(P2PTestConductor, LocalP2PTestDtls) {
SetupAndVerifyDtlsCall();
}
// This test sets up an one-way call, with media only from initiator to
// responder.
TEST_F(P2PTestConductor, OneWayMediaCall) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->set_auto_add_stream(false);
LocalP2PTest();
}
TEST_F(P2PTestConductor, OneWayMediaCallWithoutConstraints) {
ASSERT_TRUE(CreateTestClientsThatPreferNoConstraints());
receiving_client()->set_auto_add_stream(false);
LocalP2PTest();
}
// This test sets up a audio call initially and then upgrades to audio/video,
// using DTLS.
TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
receiving_client()->SetReceiveAudioVideo(true, false);
LocalP2PTest();
receiving_client()->SetReceiveAudioVideo(true, true);
receiving_client()->Negotiate();
}
// This test sets up a call transfer to a new caller with a different DTLS
// fingerprint.
TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetupAndVerifyDtlsCall();
// Keeping the original peer around which will still send packets to the
// receiving client. These SRTP packets will be dropped.
std::unique_ptr<PeerConnectionTestClient> original_peer(
set_initializing_client(CreateDtlsClientWithAlternateKey()));
original_peer->pc()->Close();
SetSignalingReceivers();
receiving_client()->SetExpectIceRestart(true);
LocalP2PTest();
VerifyRenderedSize(640, 480);
}
// This test sets up a non-bundle call and apply bundle during ICE restart. When
// bundle is in effect in the restart, the channel can successfully reset its
// DTLS-SRTP context.
TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
receiving_client()->RemoveBundleFromReceivedSdp(true);
LocalP2PTest();
VerifyRenderedSize(640, 480);
initializing_client()->IceRestart();
receiving_client()->SetExpectIceRestart(true);
receiving_client()->RemoveBundleFromReceivedSdp(false);
LocalP2PTest();
VerifyRenderedSize(640, 480);
}
// This test sets up a call transfer to a new callee with a different DTLS
// fingerprint.
TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetupAndVerifyDtlsCall();
// Keeping the original peer around which will still send packets to the
// receiving client. These SRTP packets will be dropped.
std::unique_ptr<PeerConnectionTestClient> original_peer(
set_receiving_client(CreateDtlsClientWithAlternateKey()));
original_peer->pc()->Close();
SetSignalingReceivers();
initializing_client()->IceRestart();
LocalP2PTest();
VerifyRenderedSize(640, 480);
}
TEST_F(P2PTestConductor, LocalP2PTestCVO) {
ASSERT_TRUE(CreateTestClients());
SetCaptureRotation(webrtc::kVideoRotation_90);
LocalP2PTest();
VerifyRenderedSize(640, 480, webrtc::kVideoRotation_90);
}
TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportCVO) {
ASSERT_TRUE(CreateTestClients());
SetCaptureRotation(webrtc::kVideoRotation_90);
receiving_client()->RemoveCvoFromReceivedSdp(true);
LocalP2PTest();
VerifyRenderedSize(480, 640, webrtc::kVideoRotation_0);
}
// This test sets up a call between two endpoints that are configured to use
// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
// negotiated and used for transport.
TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
receiving_client()->RemoveSdesCryptoFromReceivedSdp(true);
LocalP2PTest();
VerifyRenderedSize(640, 480);
}
// This test sets up a Jsep call between two parties, and the callee only
// accept to receive video.
TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->SetReceiveAudioVideo(false, true);
LocalP2PTest();
}
// This test sets up a Jsep call between two parties, and the callee only
// accept to receive audio.
TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->SetReceiveAudioVideo(true, false);
LocalP2PTest();
}
// This test sets up a Jsep call between two parties, and the callee reject both
// audio and video.
TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->SetReceiveAudioVideo(false, false);
LocalP2PTest();
}
// This test sets up an audio and video call between two parties. After the call
// runs for a while (10 frames), the caller sends an update offer with video
// being rejected. Once the re-negotiation is done, the video flow should stop
// and the audio flow should continue.
TEST_F(P2PTestConductor, UpdateOfferWithRejectedContent) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
TestUpdateOfferWithRejectedContent();
}
// This test sets up a Jsep call between two parties. The MSID is removed from
// the SDP strings from the caller.
TEST_F(P2PTestConductor, LocalP2PTestWithoutMsid) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->RemoveMsidFromReceivedSdp(true);
// TODO(perkj): Currently there is a bug that cause audio to stop playing if
// audio and video is muxed when MSID is disabled. Remove
// SetRemoveBundleFromSdp once
// https://code.google.com/p/webrtc/issues/detail?id=1193 is fixed.
receiving_client()->RemoveBundleFromReceivedSdp(true);
LocalP2PTest();
}
// This test sets up a Jsep call between two parties and the initiating peer
// sends two steams.
// TODO(perkj): Disabled due to
// https://code.google.com/p/webrtc/issues/detail?id=1454
TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) {
ASSERT_TRUE(CreateTestClients());
// Set optional video constraint to max 320pixels to decrease CPU usage.
FakeConstraints constraint;
constraint.SetOptionalMaxWidth(320);
SetVideoConstraints(constraint, constraint);
initializing_client()->AddMediaStream(true, true);
initializing_client()->AddMediaStream(false, true);
ASSERT_EQ(2u, initializing_client()->NumberOfLocalMediaStreams());
LocalP2PTest();
EXPECT_EQ(2u, receiving_client()->number_of_remote_streams());
}
// Test that we can receive the audio output level from a remote audio track.
TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
StreamCollectionInterface* remote_streams =
initializing_client()->remote_streams();
ASSERT_GT(remote_streams->count(), 0u);
ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u);
MediaStreamTrackInterface* remote_audio_track =
remote_streams->at(0)->GetAudioTracks()[0];
// Get the audio output level stats. Note that the level is not available
// until a RTCP packet has been received.
EXPECT_TRUE_WAIT(
initializing_client()->GetAudioOutputLevelStats(remote_audio_track) > 0,
kMaxWaitForStatsMs);
}
// Test that an audio input level is reported.
TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
// Get the audio input level stats. The level should be available very
// soon after the test starts.
EXPECT_TRUE_WAIT(initializing_client()->GetAudioInputLevelStats() > 0,
kMaxWaitForStatsMs);
}
// Test that we can get incoming byte counts from both audio and video tracks.
TEST_F(P2PTestConductor, GetBytesReceivedStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
StreamCollectionInterface* remote_streams =
initializing_client()->remote_streams();
ASSERT_GT(remote_streams->count(), 0u);
ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u);
MediaStreamTrackInterface* remote_audio_track =
remote_streams->at(0)->GetAudioTracks()[0];
EXPECT_TRUE_WAIT(
initializing_client()->GetBytesReceivedStats(remote_audio_track) > 0,
kMaxWaitForStatsMs);
MediaStreamTrackInterface* remote_video_track =
remote_streams->at(0)->GetVideoTracks()[0];
EXPECT_TRUE_WAIT(
initializing_client()->GetBytesReceivedStats(remote_video_track) > 0,
kMaxWaitForStatsMs);
}
// Test that we can get outgoing byte counts from both audio and video tracks.
TEST_F(P2PTestConductor, GetBytesSentStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
StreamCollectionInterface* local_streams =
initializing_client()->local_streams();
ASSERT_GT(local_streams->count(), 0u);
ASSERT_GT(local_streams->at(0)->GetAudioTracks().size(), 0u);
MediaStreamTrackInterface* local_audio_track =
local_streams->at(0)->GetAudioTracks()[0];
EXPECT_TRUE_WAIT(
initializing_client()->GetBytesSentStats(local_audio_track) > 0,
kMaxWaitForStatsMs);
MediaStreamTrackInterface* local_video_track =
local_streams->at(0)->GetVideoTracks()[0];
EXPECT_TRUE_WAIT(
initializing_client()->GetBytesSentStats(local_video_track) > 0,
kMaxWaitForStatsMs);
}
// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
TEST_F(P2PTestConductor, GetDtls12None) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
PeerConnectionFactory::Options recv_options;
recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
&recv_options, nullptr));
rtc::scoped_refptr<webrtc::FakeMetricsObserver>
init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
EXPECT_TRUE_WAIT(
rtc::SSLStreamAdapter::IsAcceptableCipher(
initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
kMaxWaitForStatsMs);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
EXPECT_EQ(1,
init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.2 is used if both ends support it.
TEST_F(P2PTestConductor, GetDtls12Both) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
PeerConnectionFactory::Options recv_options;
recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
&recv_options, nullptr));
rtc::scoped_refptr<webrtc::FakeMetricsObserver>
init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
EXPECT_TRUE_WAIT(
rtc::SSLStreamAdapter::IsAcceptableCipher(
initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
kMaxWaitForStatsMs);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
EXPECT_EQ(1,
init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
// received supports 1.0.
TEST_F(P2PTestConductor, GetDtls12Init) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
PeerConnectionFactory::Options recv_options;
recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
&recv_options, nullptr));
rtc::scoped_refptr<webrtc::FakeMetricsObserver>
init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
EXPECT_TRUE_WAIT(
rtc::SSLStreamAdapter::IsAcceptableCipher(
initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
kMaxWaitForStatsMs);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
EXPECT_EQ(1,
init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
// received supports 1.2.
TEST_F(P2PTestConductor, GetDtls12Recv) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
PeerConnectionFactory::Options recv_options;
recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
&recv_options, nullptr));
rtc::scoped_refptr<webrtc::FakeMetricsObserver>
init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
EXPECT_TRUE_WAIT(
rtc::SSLStreamAdapter::IsAcceptableCipher(
initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
kMaxWaitForStatsMs);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
EXPECT_EQ(1,
init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
}
// Test that a non-GCM cipher is used if both sides only support non-GCM.
TEST_F(P2PTestConductor, GetGcmNone) {
TestGcmNegotiation(false, false, kDefaultSrtpCryptoSuite);
}
// Test that a GCM cipher is used if both ends support it.
TEST_F(P2PTestConductor, GetGcmBoth) {
TestGcmNegotiation(true, true, kDefaultSrtpCryptoSuiteGcm);
}
// Test that GCM isn't used if only the initiator supports it.
TEST_F(P2PTestConductor, GetGcmInit) {
TestGcmNegotiation(true, false, kDefaultSrtpCryptoSuite);
}
// Test that GCM isn't used if only the receiver supports it.
TEST_F(P2PTestConductor, GetGcmRecv) {
TestGcmNegotiation(false, true, kDefaultSrtpCryptoSuite);
}
// This test sets up a call between two parties with audio, video and an RTP
// data channel.
TEST_F(P2PTestConductor, LocalP2PTestRtpDataChannel) {
FakeConstraints setup_constraints;
setup_constraints.SetAllowRtpDataChannels();
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
initializing_client()->CreateDataChannel();
LocalP2PTest();
ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
kMaxWaitMs);
EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(),
kMaxWaitMs);
std::string data = "hello world";
SendRtpData(initializing_client()->data_channel(), data);
EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
kMaxWaitMs);
SendRtpData(receiving_client()->data_channel(), data);
EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
kMaxWaitMs);
receiving_client()->data_channel()->Close();
// Send new offer and answer.
receiving_client()->Negotiate();
EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
EXPECT_FALSE(receiving_client()->data_observer()->IsOpen());
}
// This test sets up a call between two parties with audio, video and an SCTP
// data channel.
TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) {
ASSERT_TRUE(CreateTestClients());
initializing_client()->CreateDataChannel();
LocalP2PTest();
ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
kMaxWaitMs);
EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
std::string data = "hello world";
initializing_client()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
kMaxWaitMs);
receiving_client()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
kMaxWaitMs);
receiving_client()->data_channel()->Close();
EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
kMaxWaitMs);
EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
}
TEST_F(P2PTestConductor, UnorderedSctpDataChannel) {
ASSERT_TRUE(CreateTestClients());
webrtc::DataChannelInit init;
init.ordered = false;
initializing_client()->CreateDataChannel(&init);
// 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();
initializing_client()->Negotiate();
ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
kMaxWaitMs);
EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
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;
std::string message;
ASSERT_TRUE(rtc::CreateRandomString(length, &message));
initializing_client()->data_channel()->Send(DataBuffer(message));
receiving_client()->data_channel()->Send(DataBuffer(message));
sent_messages.push_back(message);
}
EXPECT_EQ_WAIT(
kNumMessages,
initializing_client()->data_observer()->received_message_count(),
kMaxWaitMs);
EXPECT_EQ_WAIT(kNumMessages,
receiving_client()->data_observer()->received_message_count(),
kMaxWaitMs);
// Sort and compare to make sure none of the messages were corrupted.
std::vector<std::string> initializing_client_received_messages =
initializing_client()->data_observer()->messages();
std::vector<std::string> receiving_client_received_messages =
receiving_client()->data_observer()->messages();
std::sort(sent_messages.begin(), sent_messages.end());
std::sort(initializing_client_received_messages.begin(),
initializing_client_received_messages.end());
std::sort(receiving_client_received_messages.begin(),
receiving_client_received_messages.end());
EXPECT_EQ(sent_messages, initializing_client_received_messages);
EXPECT_EQ(sent_messages, receiving_client_received_messages);
receiving_client()->data_channel()->Close();
EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
kMaxWaitMs);
EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
}
// This test sets up a call between two parties and creates a data channel.
// The test tests that received data is buffered unless an observer has been
// registered.
// 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 used in that case.
TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
FakeConstraints setup_constraints;
setup_constraints.SetAllowRtpDataChannels();
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
initializing_client()->CreateDataChannel();
initializing_client()->Negotiate();
ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
kMaxWaitMs);
EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
receiving_client()->data_channel()->state(), kMaxWaitMs);
// Unregister the existing observer.
receiving_client()->data_channel()->UnregisterObserver();
std::string data = "hello world";
SendRtpData(initializing_client()->data_channel(), data);
// Wait a while to allow the sent data to arrive before an observer is
// registered..
rtc::Thread::Current()->ProcessMessages(100);
MockDataChannelObserver new_observer(receiving_client()->data_channel());
EXPECT_EQ_WAIT(data, new_observer.last_message(), kMaxWaitMs);
}
// This test sets up a call between two parties with audio, video and but only
// the initiating client support data.
TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) {
FakeConstraints setup_constraints_1;
setup_constraints_1.SetAllowRtpDataChannels();
// Must disable DTLS to make negotiation succeed.
setup_constraints_1.SetMandatory(
MediaConstraintsInterface::kEnableDtlsSrtp, false);
FakeConstraints setup_constraints_2;
setup_constraints_2.SetMandatory(
MediaConstraintsInterface::kEnableDtlsSrtp, false);
ASSERT_TRUE(CreateTestClients(&setup_constraints_1, &setup_constraints_2));
initializing_client()->CreateDataChannel();
LocalP2PTest();
EXPECT_TRUE(initializing_client()->data_channel() != nullptr);
EXPECT_FALSE(receiving_client()->data_channel());
EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
}
// This test sets up a call between two parties with audio, video. When audio
// and video is setup and flowing and data channel is negotiated.
TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
FakeConstraints setup_constraints;
setup_constraints.SetAllowRtpDataChannels();
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
LocalP2PTest();
initializing_client()->CreateDataChannel();
// Send new offer and answer.
initializing_client()->Negotiate();
ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
kMaxWaitMs);
EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(),
kMaxWaitMs);
}
// This test sets up a Jsep call with SCTP DataChannel and verifies the
// negotiation is completed without error.
#ifdef HAVE_SCTP
TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints constraints;
constraints.SetMandatory(
MediaConstraintsInterface::kEnableDtlsSrtp, true);
ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
initializing_client()->CreateDataChannel();
initializing_client()->Negotiate(false, false);
}
#endif
// This test sets up a call between two parties with audio, and video.
// During the call, the initializing side restart ice and the test verifies that
// new ice candidates are generated and audio and video still can flow.
TEST_F(P2PTestConductor, IceRestart) {
ASSERT_TRUE(CreateTestClients());
// Negotiate and wait for ice completion and make sure audio and video plays.
LocalP2PTest();
// Create a SDP string of the first audio candidate for both clients.
const webrtc::IceCandidateCollection* audio_candidates_initiator =
initializing_client()->pc()->local_description()->candidates(0);
const webrtc::IceCandidateCollection* audio_candidates_receiver =
receiving_client()->pc()->local_description()->candidates(0);
ASSERT_GT(audio_candidates_initiator->count(), 0u);
ASSERT_GT(audio_candidates_receiver->count(), 0u);
std::string initiator_candidate;
EXPECT_TRUE(
audio_candidates_initiator->at(0)->ToString(&initiator_candidate));
std::string receiver_candidate;
EXPECT_TRUE(audio_candidates_receiver->at(0)->ToString(&receiver_candidate));
// Restart ice on the initializing client.
receiving_client()->SetExpectIceRestart(true);
initializing_client()->IceRestart();
// Negotiate and wait for ice completion again and make sure audio and video
// plays.
LocalP2PTest();
// Create a SDP string of the first audio candidate for both clients again.
const webrtc::IceCandidateCollection* audio_candidates_initiator_restart =
initializing_client()->pc()->local_description()->candidates(0);
const webrtc::IceCandidateCollection* audio_candidates_reciever_restart =
receiving_client()->pc()->local_description()->candidates(0);
ASSERT_GT(audio_candidates_initiator_restart->count(), 0u);
ASSERT_GT(audio_candidates_reciever_restart->count(), 0u);
std::string initiator_candidate_restart;
EXPECT_TRUE(audio_candidates_initiator_restart->at(0)->ToString(
&initiator_candidate_restart));
std::string receiver_candidate_restart;
EXPECT_TRUE(audio_candidates_reciever_restart->at(0)->ToString(
&receiver_candidate_restart));
// Verify that the first candidates in the local session descriptions has
// changed.
EXPECT_NE(initiator_candidate, initiator_candidate_restart);
EXPECT_NE(receiver_candidate, receiver_candidate_restart);
}
TEST_F(P2PTestConductor, IceRenominationDisabled) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = false;
ASSERT_TRUE(CreateTestClients(config, config));
LocalP2PTest();
initializing_client()->VerifyLocalIceRenomination();
receiving_client()->VerifyLocalIceRenomination();
initializing_client()->VerifyRemoteIceRenomination();
receiving_client()->VerifyRemoteIceRenomination();
}
TEST_F(P2PTestConductor, IceRenominationEnabled) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = true;
ASSERT_TRUE(CreateTestClients(config, config));
initializing_client()->SetExpectIceRenomination(true);
initializing_client()->SetExpectRemoteIceRenomination(true);
receiving_client()->SetExpectIceRenomination(true);
receiving_client()->SetExpectRemoteIceRenomination(true);
LocalP2PTest();
initializing_client()->VerifyLocalIceRenomination();
receiving_client()->VerifyLocalIceRenomination();
initializing_client()->VerifyRemoteIceRenomination();
receiving_client()->VerifyRemoteIceRenomination();
}
// 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_F(P2PTestConductor, LocalP2PTestVideoDisableEnable) {
ASSERT_TRUE(CreateTestClients());
// Do initial negotiation. Will result in video and audio sendonly m-lines.
receiving_client()->set_auto_add_stream(false);
initializing_client()->AddMediaStream(true, true);
initializing_client()->Negotiate();
// Negotiate again, disabling the video m-line (receiving client will
// set port to 0 due to mandatory "OfferToReceiveVideo: false" constraint).
receiving_client()->SetReceiveVideo(false);
initializing_client()->Negotiate();
// Enable video and do negotiation again, making sure video is received
// end-to-end.
receiving_client()->SetReceiveVideo(true);
receiving_client()->AddMediaStream(true, true);
LocalP2PTest();
}
// This test sets up a Jsep call between two parties with external
// VideoDecoderFactory.
// TODO(holmer): Disabled due to sometimes crashing on buildbots.
// See issue webrtc/2378.
TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) {
ASSERT_TRUE(CreateTestClients());
EnableVideoDecoderFactory();
LocalP2PTest();
}
// 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(P2PTestConductor, EarlyWarmupTest) {
ASSERT_TRUE(CreateTestClients());
auto audio_sender =
initializing_client()->pc()->CreateSender("audio", "stream_id");
auto video_sender =
initializing_client()->pc()->CreateSender("video", "stream_id");
initializing_client()->Negotiate();
// Wait for ICE connection to complete, without any tracks.
// Note that the receiving client WILL (in HandleIncomingOffer) create
// tracks, so it's only the initiator here that's doing early warmup.
ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
VerifySessionDescriptions();
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
initializing_client()->ice_connection_state(),
kMaxWaitForFramesMs);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
receiving_client()->ice_connection_state(),
kMaxWaitForFramesMs);
// Now set the tracks, and expect frames to immediately start flowing.
EXPECT_TRUE(
audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack("")));
EXPECT_TRUE(
video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack("")));
EXPECT_TRUE_WAIT(FramesHaveArrived(kEndAudioFrameCount, kEndVideoFrameCount),
kMaxWaitForFramesMs);
}
#ifdef HAVE_QUIC
// This test sets up a call between two parties using QUIC instead of DTLS for
// audio and video, and a QUIC data channel.
TEST_F(P2PTestConductor, LocalP2PTestQuicDataChannel) {
PeerConnectionInterface::RTCConfiguration quic_config;
quic_config.enable_quic = true;
ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
webrtc::DataChannelInit init;
init.ordered = false;
init.reliable = true;
init.id = 1;
initializing_client()->CreateDataChannel(&init);
receiving_client()->CreateDataChannel(&init);
LocalP2PTest();
ASSERT_NE(nullptr, initializing_client()->data_channel());
ASSERT_NE(nullptr, receiving_client()->data_channel());
EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
kMaxWaitMs);
EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
std::string data = "hello world";
initializing_client()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
kMaxWaitMs);
receiving_client()->data_channel()->Send(DataBuffer(data));
EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
kMaxWaitMs);
}
// Tests that negotiation of QUIC data channels is completed without error.
TEST_F(P2PTestConductor, NegotiateQuicDataChannel) {
PeerConnectionInterface::RTCConfiguration quic_config;
quic_config.enable_quic = true;
ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
FakeConstraints constraints;
constraints.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true);
ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
webrtc::DataChannelInit init;
init.ordered = false;
init.reliable = true;
init.id = 1;
initializing_client()->CreateDataChannel(&init);
initializing_client()->Negotiate(false, false);
}
// This test sets up a JSEP call using QUIC. The callee only receives video.
TEST_F(P2PTestConductor, LocalP2PTestVideoOnlyWithQuic) {
PeerConnectionInterface::RTCConfiguration quic_config;
quic_config.enable_quic = true;
ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
receiving_client()->SetReceiveAudioVideo(false, true);
LocalP2PTest();
}
// This test sets up a JSEP call using QUIC. The callee only receives audio.
TEST_F(P2PTestConductor, LocalP2PTestAudioOnlyWithQuic) {
PeerConnectionInterface::RTCConfiguration quic_config;
quic_config.enable_quic = true;
ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
receiving_client()->SetReceiveAudioVideo(true, false);
LocalP2PTest();
}
// This test sets up a JSEP call using QUIC. The callee rejects both audio and
// video.
TEST_F(P2PTestConductor, LocalP2PTestNoVideoAudioWithQuic) {
PeerConnectionInterface::RTCConfiguration quic_config;
quic_config.enable_quic = true;
ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
receiving_client()->SetReceiveAudioVideo(false, false);
LocalP2PTest();
}
#endif // HAVE_QUIC
TEST_F(P2PTestConductor, ForwardVideoOnlyStream) {
ASSERT_TRUE(CreateTestClients());
// One-way stream
receiving_client()->set_auto_add_stream(false);
// Video only, audio forwarding not expected to work.
initializing_client()->AddMediaStream(false, true);
initializing_client()->Negotiate();
ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
VerifySessionDescriptions();
ASSERT_TRUE(initializing_client()->can_receive_video());
ASSERT_TRUE(receiving_client()->can_receive_video());
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
initializing_client()->ice_connection_state(),
kMaxWaitForFramesMs);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
receiving_client()->ice_connection_state(),
kMaxWaitForFramesMs);
ASSERT_TRUE(receiving_client()->remote_streams()->count() == 1);
// Echo the stream back.
receiving_client()->pc()->AddStream(
receiving_client()->remote_streams()->at(0));
receiving_client()->Negotiate();
EXPECT_TRUE_WAIT(
initializing_client()->VideoFramesReceivedCheck(kEndVideoFrameCount),
kMaxWaitForFramesMs);
}
// 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_F(P2PTestConductor, 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(rtc::TimeDelta::FromSeconds(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(network_thread(),
turn_server_1_internal_address,
turn_server_1_external_address);
cricket::TestTurnServer turn_server_2(network_thread(),
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.
turn_server_1.set_enable_permission_checks(false);
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(CreateTestClients(client_1_config, client_2_config));
// Set up the simulated delays.
SetSignalingDelayMs(signaling_trip_delay_ms);
virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
virtual_socket_server()->UpdateDelayDistribution();
initializing_client()->SetOfferToReceiveAudioVideo(true, true);
initializing_client()->Negotiate();
// 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.
EXPECT_TRUE_SIMULATED_WAIT(
(receiving_client()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionConnected ||
receiving_client()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
(initializing_client()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionConnected ||
initializing_client()->ice_connection_state() ==
webrtc::PeerConnectionInterface::kIceConnectionCompleted),
total_connection_time_ms, fake_clock);
// Need to free the clients here since they're using things we created on
// the stack.
delete set_initializing_client(nullptr);
delete set_receiving_client(nullptr);
}
class IceServerParsingTest : public testing::Test {
public:
// Convenience for parsing a single URL.
bool ParseUrl(const std::string& url) {
return ParseUrl(url, std::string(), std::string());
}
bool ParseUrl(const std::string& url,
const std::string& username,
const std::string& password) {
PeerConnectionInterface::IceServers servers;
PeerConnectionInterface::IceServer server;
server.urls.push_back(url);
server.username = username;
server.password = password;
servers.push_back(server);
return webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_);
}
protected:
cricket::ServerAddresses stun_servers_;
std::vector<cricket::RelayServerConfig> turn_servers_;
};
// Make sure all STUN/TURN prefixes are parsed correctly.
TEST_F(IceServerParsingTest, ParseStunPrefixes) {
EXPECT_TRUE(ParseUrl("stun:hostname"));
EXPECT_EQ(1U, stun_servers_.size());
EXPECT_EQ(0U, turn_servers_.size());
stun_servers_.clear();
EXPECT_TRUE(ParseUrl("stuns:hostname"));
EXPECT_EQ(1U, stun_servers_.size());
EXPECT_EQ(0U, turn_servers_.size());
stun_servers_.clear();
EXPECT_TRUE(ParseUrl("turn:hostname"));
EXPECT_EQ(0U, stun_servers_.size());
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_FALSE(turn_servers_[0].ports[0].secure);
turn_servers_.clear();
EXPECT_TRUE(ParseUrl("turns:hostname"));
EXPECT_EQ(0U, stun_servers_.size());
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_TRUE(turn_servers_[0].ports[0].secure);
turn_servers_.clear();
// invalid prefixes
EXPECT_FALSE(ParseUrl("stunn:hostname"));
EXPECT_FALSE(ParseUrl(":hostname"));
EXPECT_FALSE(ParseUrl(":"));
EXPECT_FALSE(ParseUrl(""));
}