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/*
* Copyright 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/create_peerconnection_factory.h"
#include "api/peer_connection_proxy.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "p2p/base/fake_port_allocator.h"
#include "p2p/base/test_stun_server.h"
#include "p2p/client/basic_port_allocator.h"
#include "pc/media_session.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/sdp_utils.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/android_test_initializer.h"
#endif
#include "pc/test/fake_audio_capture_module.h"
#include "rtc_base/fake_network.h"
#include "rtc_base/gunit.h"
#include "rtc_base/virtual_socket_server.h"
#include "test/gmock.h"
namespace webrtc {
using BundlePolicy = PeerConnectionInterface::BundlePolicy;
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
using RtcpMuxPolicy = PeerConnectionInterface::RtcpMuxPolicy;
using rtc::SocketAddress;
using ::testing::Combine;
using ::testing::ElementsAre;
using ::testing::UnorderedElementsAre;
using ::testing::Values;
constexpr int kDefaultTimeout = 10000;
// TODO(steveanton): These tests should be rewritten to use the standard
// RtpSenderInterface/DtlsTransportInterface objects once they're available in
// the API. The RtpSender can be used to determine which transport a given media
// will use: https://www.w3.org/TR/webrtc/#dom-rtcrtpsender-transport
// Should also be able to remove GetTransceiversForTesting at that point.
class FakeNetworkManagerWithNoAnyNetwork : public rtc::FakeNetworkManager {
public:
void GetAnyAddressNetworks(NetworkList* networks) override {
// This function allocates networks that are owned by the
// NetworkManager. But some tests assume that they can release
// all networks independent of the network manager.
// In order to prevent use-after-free issues, don't allow this
// function to have any effect when run in tests.
RTC_LOG(LS_INFO) << "FakeNetworkManager::GetAnyAddressNetworks ignored";
}
};
class PeerConnectionWrapperForBundleTest : public PeerConnectionWrapper {
public:
using PeerConnectionWrapper::PeerConnectionWrapper;
bool AddIceCandidateToMedia(cricket::Candidate* candidate,
cricket::MediaType media_type) {
auto* desc = pc()->remote_description()->description();
for (size_t i = 0; i < desc->contents().size(); i++) {
const auto& content = desc->contents()[i];
if (content.media_description()->type() == media_type) {
candidate->set_transport_name(content.name);
std::unique_ptr<IceCandidateInterface> jsep_candidate =
CreateIceCandidate(content.name, i, *candidate);
return pc()->AddIceCandidate(jsep_candidate.get());
}
}
RTC_NOTREACHED();
return false;
}
RtpTransportInternal* voice_rtp_transport() {
return (voice_channel() ? voice_channel()->rtp_transport() : nullptr);
}
cricket::VoiceChannel* voice_channel() {
auto transceivers = GetInternalPeerConnection()->GetTransceiversInternal();
for (const auto& transceiver : transceivers) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
return static_cast<cricket::VoiceChannel*>(
transceiver->internal()->channel());
}
}
return nullptr;
}
RtpTransportInternal* video_rtp_transport() {
return (video_channel() ? video_channel()->rtp_transport() : nullptr);
}
cricket::VideoChannel* video_channel() {
auto transceivers = GetInternalPeerConnection()->GetTransceiversInternal();
for (const auto& transceiver : transceivers) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
return static_cast<cricket::VideoChannel*>(
transceiver->internal()->channel());
}
}
return nullptr;
}
PeerConnection* GetInternalPeerConnection() {
auto* pci =
static_cast<PeerConnectionProxyWithInternal<PeerConnectionInterface>*>(
pc());
return static_cast<PeerConnection*>(pci->internal());
}
// Returns true if the stats indicate that an ICE connection is either in
// progress or established with the given remote address.
bool HasConnectionWithRemoteAddress(const SocketAddress& address) {
auto report = GetStats();
if (!report) {
return false;
}
std::string matching_candidate_id;
for (auto* ice_candidate_stats :
report->GetStatsOfType<RTCRemoteIceCandidateStats>()) {
if (*ice_candidate_stats->ip == address.HostAsURIString() &&
*ice_candidate_stats->port == address.port()) {
matching_candidate_id = ice_candidate_stats->id();
break;
}
}
if (matching_candidate_id.empty()) {
return false;
}
for (auto* pair_stats :
report->GetStatsOfType<RTCIceCandidatePairStats>()) {
if (*pair_stats->remote_candidate_id == matching_candidate_id) {
if (*pair_stats->state == RTCStatsIceCandidatePairState::kInProgress ||
*pair_stats->state == RTCStatsIceCandidatePairState::kSucceeded) {
return true;
}
}
}
return false;
}
rtc::FakeNetworkManager* network() { return network_; }
void set_network(rtc::FakeNetworkManager* network) { network_ = network; }
private:
rtc::FakeNetworkManager* network_;
};
class PeerConnectionBundleBaseTest : public ::testing::Test {
protected:
typedef std::unique_ptr<PeerConnectionWrapperForBundleTest> WrapperPtr;
explicit PeerConnectionBundleBaseTest(SdpSemantics sdp_semantics)
: vss_(new rtc::VirtualSocketServer()),
main_(vss_.get()),
sdp_semantics_(sdp_semantics) {
#ifdef WEBRTC_ANDROID
InitializeAndroidObjects();
#endif
pc_factory_ = CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
CreateBuiltinVideoEncoderFactory(), CreateBuiltinVideoDecoderFactory(),
nullptr /* audio_mixer */, nullptr /* audio_processing */);
}
WrapperPtr CreatePeerConnection() {
return CreatePeerConnection(RTCConfiguration());
}
WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
auto* fake_network = NewFakeNetwork();
auto port_allocator =
std::make_unique<cricket::BasicPortAllocator>(fake_network);
port_allocator->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
cricket::PORTALLOCATOR_DISABLE_RELAY);
port_allocator->set_step_delay(cricket::kMinimumStepDelay);
auto observer = std::make_unique<MockPeerConnectionObserver>();
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
auto pc = pc_factory_->CreatePeerConnection(
modified_config, std::move(port_allocator), nullptr, observer.get());
if (!pc) {
return nullptr;
}
auto wrapper = std::make_unique<PeerConnectionWrapperForBundleTest>(
pc_factory_, pc, std::move(observer));
wrapper->set_network(fake_network);
return wrapper;
}
// Accepts the same arguments as CreatePeerConnection and adds default audio
// and video tracks.
template <typename... Args>
WrapperPtr CreatePeerConnectionWithAudioVideo(Args&&... args) {
auto wrapper = CreatePeerConnection(std::forward<Args>(args)...);
if (!wrapper) {
return nullptr;
}
wrapper->AddAudioTrack("a");
wrapper->AddVideoTrack("v");
return wrapper;
}
cricket::Candidate CreateLocalUdpCandidate(
const rtc::SocketAddress& address) {
cricket::Candidate candidate;
candidate.set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
candidate.set_protocol(cricket::UDP_PROTOCOL_NAME);
candidate.set_address(address);
candidate.set_type(cricket::LOCAL_PORT_TYPE);
return candidate;
}
rtc::FakeNetworkManager* NewFakeNetwork() {
// The PeerConnection's port allocator is tied to the PeerConnection's
// lifetime and expects the underlying NetworkManager to outlive it. If
// PeerConnectionWrapper owned the NetworkManager, it would be destroyed
// before the PeerConnection (since subclass members are destroyed before
// base class members). Therefore, the test fixture will own all the fake
// networks even though tests should access the fake network through the
// PeerConnectionWrapper.
auto* fake_network = new FakeNetworkManagerWithNoAnyNetwork();
fake_networks_.emplace_back(fake_network);
return fake_network;
}
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
std::vector<std::unique_ptr<rtc::FakeNetworkManager>> fake_networks_;
const SdpSemantics sdp_semantics_;
};
class PeerConnectionBundleTest
: public PeerConnectionBundleBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionBundleTest() : PeerConnectionBundleBaseTest(GetParam()) {}
};
class PeerConnectionBundleTestUnifiedPlan
: public PeerConnectionBundleBaseTest {
protected:
PeerConnectionBundleTestUnifiedPlan()
: PeerConnectionBundleBaseTest(SdpSemantics::kUnifiedPlan) {}
};
SdpContentMutator RemoveRtcpMux() {
return [](cricket::ContentInfo* content, cricket::TransportInfo* transport) {
content->media_description()->set_rtcp_mux(false);
};
}
std::vector<int> GetCandidateComponents(
const std::vector<IceCandidateInterface*> candidates) {
std::vector<int> components;
components.reserve(candidates.size());
for (auto* candidate : candidates) {
components.push_back(candidate->candidate().component());
}
return components;
}
// Test that there are 2 local UDP candidates (1 RTP and 1 RTCP candidate) for
// each media section when disabling bundling and disabling RTCP multiplexing.
TEST_P(PeerConnectionBundleTest,
TwoCandidatesForEachTransportWhenNoBundleNoRtcpMux) {
const SocketAddress kCallerAddress("1.1.1.1", 0);
const SocketAddress kCalleeAddress("2.2.2.2", 0);
RTCConfiguration config;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
auto caller = CreatePeerConnectionWithAudioVideo(config);
caller->network()->AddInterface(kCallerAddress);
auto callee = CreatePeerConnectionWithAudioVideo(config);
callee->network()->AddInterface(kCalleeAddress);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
RTCOfferAnswerOptions options_no_bundle;
options_no_bundle.use_rtp_mux = false;
auto answer = callee->CreateAnswer(options_no_bundle);
SdpContentsForEach(RemoveRtcpMux(), answer->description());
ASSERT_TRUE(
callee->SetLocalDescription(CloneSessionDescription(answer.get())));
ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
// Check that caller has separate RTP and RTCP candidates for each media.
EXPECT_TRUE_WAIT(caller->IsIceGatheringDone(), kDefaultTimeout);
EXPECT_THAT(
GetCandidateComponents(caller->observer()->GetCandidatesByMline(0)),
UnorderedElementsAre(cricket::ICE_CANDIDATE_COMPONENT_RTP,
cricket::ICE_CANDIDATE_COMPONENT_RTCP));
EXPECT_THAT(
GetCandidateComponents(caller->observer()->GetCandidatesByMline(1)),
UnorderedElementsAre(cricket::ICE_CANDIDATE_COMPONENT_RTP,
cricket::ICE_CANDIDATE_COMPONENT_RTCP));
// Check that callee has separate RTP and RTCP candidates for each media.
EXPECT_TRUE_WAIT(callee->IsIceGatheringDone(), kDefaultTimeout);
EXPECT_THAT(
GetCandidateComponents(callee->observer()->GetCandidatesByMline(0)),
UnorderedElementsAre(cricket::ICE_CANDIDATE_COMPONENT_RTP,
cricket::ICE_CANDIDATE_COMPONENT_RTCP));
EXPECT_THAT(
GetCandidateComponents(callee->observer()->GetCandidatesByMline(1)),
UnorderedElementsAre(cricket::ICE_CANDIDATE_COMPONENT_RTP,
cricket::ICE_CANDIDATE_COMPONENT_RTCP));
}
// Test that there is 1 local UDP candidate for both RTP and RTCP for each media
// section when disabling bundle but enabling RTCP multiplexing.
TEST_P(PeerConnectionBundleTest,
OneCandidateForEachTransportWhenNoBundleButRtcpMux) {
const SocketAddress kCallerAddress("1.1.1.1", 0);
auto caller = CreatePeerConnectionWithAudioVideo();
caller->network()->AddInterface(kCallerAddress);
auto callee = CreatePeerConnectionWithAudioVideo();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
RTCOfferAnswerOptions options_no_bundle;
options_no_bundle.use_rtp_mux = false;
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswer(options_no_bundle)));
EXPECT_TRUE_WAIT(caller->IsIceGatheringDone(), kDefaultTimeout);
EXPECT_EQ(1u, caller->observer()->GetCandidatesByMline(0).size());
EXPECT_EQ(1u, caller->observer()->GetCandidatesByMline(1).size());
}
// Test that there is 1 local UDP candidate in only the first media section when
// bundling and enabling RTCP multiplexing.
TEST_P(PeerConnectionBundleTest,
OneCandidateOnlyOnFirstTransportWhenBundleAndRtcpMux) {
const SocketAddress kCallerAddress("1.1.1.1", 0);
RTCConfiguration config;
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
auto caller = CreatePeerConnectionWithAudioVideo(config);
caller->network()->AddInterface(kCallerAddress);
auto callee = CreatePeerConnectionWithAudioVideo(config);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
ASSERT_TRUE(caller->SetRemoteDescription(callee->CreateAnswer()));
EXPECT_TRUE_WAIT(caller->IsIceGatheringDone(), kDefaultTimeout);
EXPECT_EQ(1u, caller->observer()->GetCandidatesByMline(0).size());
EXPECT_EQ(0u, caller->observer()->GetCandidatesByMline(1).size());
}
// It will fail if the offerer uses the mux-BUNDLE policy but the answerer
// doesn't support BUNDLE.
TEST_P(PeerConnectionBundleTest, MaxBundleNotSupportedInAnswer) {
RTCConfiguration config;
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
bool equal_before =
(caller->voice_rtp_transport() == caller->video_rtp_transport());
EXPECT_EQ(true, equal_before);
RTCOfferAnswerOptions options;
options.use_rtp_mux = false;
EXPECT_FALSE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options)));
}
// The following parameterized test verifies that an offer/answer with varying
// bundle policies and either bundle in the answer or not will produce the
// expected RTP transports for audio and video. In particular, for bundling we
// care about whether they are separate transports or the same.
enum class BundleIncluded { kBundleInAnswer, kBundleNotInAnswer };
std::ostream& operator<<(std::ostream& out, BundleIncluded value) {
switch (value) {
case BundleIncluded::kBundleInAnswer:
return out << "bundle in answer";
case BundleIncluded::kBundleNotInAnswer:
return out << "bundle not in answer";
}
return out << "unknown";
}
class PeerConnectionBundleMatrixTest
: public PeerConnectionBundleBaseTest,
public ::testing::WithParamInterface<
std::tuple<SdpSemantics,
std::tuple<BundlePolicy, BundleIncluded, bool, bool>>> {
protected:
PeerConnectionBundleMatrixTest()
: PeerConnectionBundleBaseTest(std::get<0>(GetParam())) {
auto param = std::get<1>(GetParam());
bundle_policy_ = std::get<0>(param);
bundle_included_ = std::get<1>(param);
expected_same_before_ = std::get<2>(param);
expected_same_after_ = std::get<3>(param);
}
PeerConnectionInterface::BundlePolicy bundle_policy_;
BundleIncluded bundle_included_;
bool expected_same_before_;
bool expected_same_after_;
};
TEST_P(PeerConnectionBundleMatrixTest,
VerifyTransportsBeforeAndAfterSettingRemoteAnswer) {
RTCConfiguration config;
config.bundle_policy = bundle_policy_;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
bool equal_before =
(caller->voice_rtp_transport() == caller->video_rtp_transport());
EXPECT_EQ(expected_same_before_, equal_before);
RTCOfferAnswerOptions options;
options.use_rtp_mux = (bundle_included_ == BundleIncluded::kBundleInAnswer);
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options)));
bool equal_after =
(caller->voice_rtp_transport() == caller->video_rtp_transport());
EXPECT_EQ(expected_same_after_, equal_after);
}
// The max-bundle policy means we should anticipate bundling being negotiated,
// and multiplex audio/video from the start.
// For all other policies, bundling should only be enabled if negotiated by the
// answer.
INSTANTIATE_TEST_SUITE_P(
PeerConnectionBundleTest,
PeerConnectionBundleMatrixTest,
Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
Values(std::make_tuple(BundlePolicy::kBundlePolicyBalanced,
BundleIncluded::kBundleInAnswer,
false,
true),
std::make_tuple(BundlePolicy::kBundlePolicyBalanced,
BundleIncluded::kBundleNotInAnswer,
false,
false),
std::make_tuple(BundlePolicy::kBundlePolicyMaxBundle,
BundleIncluded::kBundleInAnswer,
true,
true),
std::make_tuple(BundlePolicy::kBundlePolicyMaxCompat,
BundleIncluded::kBundleInAnswer,
false,
true),
std::make_tuple(BundlePolicy::kBundlePolicyMaxCompat,
BundleIncluded::kBundleNotInAnswer,
false,
false))));
// Test that the audio/video transports on the callee side are the same before
// and after setting a local answer when max BUNDLE is enabled and an offer with
// BUNDLE is received.
TEST_P(PeerConnectionBundleTest,
TransportsSameForMaxBundleWithBundleInRemoteOffer) {
auto caller = CreatePeerConnectionWithAudioVideo();
RTCConfiguration config;
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
auto callee = CreatePeerConnectionWithAudioVideo(config);
RTCOfferAnswerOptions options_with_bundle;
options_with_bundle.use_rtp_mux = true;
ASSERT_TRUE(callee->SetRemoteDescription(
caller->CreateOfferAndSetAsLocal(options_with_bundle)));
EXPECT_EQ(callee->voice_rtp_transport(), callee->video_rtp_transport());
ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
EXPECT_EQ(callee->voice_rtp_transport(), callee->video_rtp_transport());
}
TEST_P(PeerConnectionBundleTest,
FailToSetRemoteOfferWithNoBundleWhenBundlePolicyMaxBundle) {
auto caller = CreatePeerConnectionWithAudioVideo();
RTCConfiguration config;
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
auto callee = CreatePeerConnectionWithAudioVideo(config);
RTCOfferAnswerOptions options_no_bundle;
options_no_bundle.use_rtp_mux = false;
EXPECT_FALSE(callee->SetRemoteDescription(
caller->CreateOfferAndSetAsLocal(options_no_bundle)));
}
// Test that if the media section which has the bundled transport is rejected,
// then the peers still connect and the bundled transport switches to the other
// media section.
// Note: This is currently failing because of the following bug:
// https://bugs.chromium.org/p/webrtc/issues/detail?id=6280
TEST_P(PeerConnectionBundleTest,
DISABLED_SuccessfullyNegotiateMaxBundleIfBundleTransportMediaRejected) {
RTCConfiguration config;
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnection();
callee->AddVideoTrack("v");
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 0;
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options)));
EXPECT_FALSE(caller->voice_rtp_transport());
EXPECT_TRUE(caller->video_rtp_transport());
}
// When requiring RTCP multiplexing, the PeerConnection never makes RTCP
// transport channels.
TEST_P(PeerConnectionBundleTest, NeverCreateRtcpTransportWithRtcpMuxRequired) {
RTCConfiguration config;
config.rtcp_mux_policy = RtcpMuxPolicy::kRtcpMuxPolicyRequire;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
EXPECT_FALSE(caller->voice_rtp_transport()->rtcp_mux_enabled());
EXPECT_FALSE(caller->video_rtp_transport()->rtcp_mux_enabled());
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
EXPECT_TRUE(caller->voice_rtp_transport()->rtcp_mux_enabled());
EXPECT_TRUE(caller->video_rtp_transport()->rtcp_mux_enabled());
}
// When negotiating RTCP multiplexing, the PeerConnection makes RTCP transports
// when the offer is sent, but will destroy them once the remote answer is set.
TEST_P(PeerConnectionBundleTest,
CreateRtcpTransportOnlyBeforeAnswerWithRtcpMuxNegotiate) {
RTCConfiguration config;
config.rtcp_mux_policy = RtcpMuxPolicy::kRtcpMuxPolicyNegotiate;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
EXPECT_FALSE(caller->voice_rtp_transport()->rtcp_mux_enabled());
EXPECT_FALSE(caller->video_rtp_transport()->rtcp_mux_enabled());
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
EXPECT_TRUE(caller->voice_rtp_transport()->rtcp_mux_enabled());
EXPECT_TRUE(caller->video_rtp_transport()->rtcp_mux_enabled());
}
TEST_P(PeerConnectionBundleTest, FailToSetDescriptionWithBundleAndNoRtcpMux) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
RTCOfferAnswerOptions options;
options.use_rtp_mux = true;
auto offer = caller->CreateOffer(options);
SdpContentsForEach(RemoveRtcpMux(), offer->description());
std::string error;
EXPECT_FALSE(caller->SetLocalDescription(CloneSessionDescription(offer.get()),
&error));
EXPECT_EQ(
"Failed to set local offer sdp: rtcp-mux must be enabled when BUNDLE is "
"enabled.",
error);
EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer), &error));
EXPECT_EQ(
"Failed to set remote offer sdp: rtcp-mux must be enabled when BUNDLE is "
"enabled.",
error);
}
// Test that candidates sent to the "video" transport do not get pushed down to
// the "audio" transport channel when bundling.
TEST_P(PeerConnectionBundleTest,
IgnoreCandidatesForUnusedTransportWhenBundling) {
const SocketAddress kAudioAddress1("1.1.1.1", 1111);
const SocketAddress kAudioAddress2("2.2.2.2", 2222);
const SocketAddress kVideoAddress("3.3.3.3", 3333);
const SocketAddress kCallerAddress("4.4.4.4", 0);
const SocketAddress kCalleeAddress("5.5.5.5", 0);
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
caller->network()->AddInterface(kCallerAddress);
callee->network()->AddInterface(kCalleeAddress);
RTCOfferAnswerOptions options;
options.use_rtp_mux = true;
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options)));
// The way the *_WAIT checks work is they only wait if the condition fails,
// which does not help in the case where state is not changing. This is
// problematic in this test since we want to verify that adding a video
// candidate does _not_ change state. So we interleave candidates and assume
// that messages are executed in the order they were posted.
cricket::Candidate audio_candidate1 = CreateLocalUdpCandidate(kAudioAddress1);
ASSERT_TRUE(caller->AddIceCandidateToMedia(&audio_candidate1,
cricket::MEDIA_TYPE_AUDIO));
cricket::Candidate video_candidate = CreateLocalUdpCandidate(kVideoAddress);
ASSERT_TRUE(caller->AddIceCandidateToMedia(&video_candidate,
cricket::MEDIA_TYPE_VIDEO));
cricket::Candidate audio_candidate2 = CreateLocalUdpCandidate(kAudioAddress2);
ASSERT_TRUE(caller->AddIceCandidateToMedia(&audio_candidate2,
cricket::MEDIA_TYPE_AUDIO));
EXPECT_TRUE_WAIT(caller->HasConnectionWithRemoteAddress(kAudioAddress1),
kDefaultTimeout);
EXPECT_TRUE_WAIT(caller->HasConnectionWithRemoteAddress(kAudioAddress2),
kDefaultTimeout);
EXPECT_FALSE(caller->HasConnectionWithRemoteAddress(kVideoAddress));
}
// Test that the transport used by both audio and video is the transport
// associated with the first MID in the answer BUNDLE group, even if it's in a
// different order from the offer.
TEST_P(PeerConnectionBundleTest, BundleOnFirstMidInAnswer) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
auto* old_video_transport = caller->video_rtp_transport();
auto answer = callee->CreateAnswer();
auto* old_bundle_group =
answer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
std::string first_mid = old_bundle_group->content_names()[0];
std::string second_mid = old_bundle_group->content_names()[1];
answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
new_bundle_group.AddContentName(second_mid);
new_bundle_group.AddContentName(first_mid);
answer->description()->AddGroup(new_bundle_group);
ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
EXPECT_EQ(old_video_transport, caller->video_rtp_transport());
EXPECT_EQ(caller->voice_rtp_transport(), caller->video_rtp_transport());
}
// This tests that applying description with conflicted RTP demuxing criteria
// will fail.
TEST_P(PeerConnectionBundleTest,
ApplyDescriptionWithConflictedDemuxCriteriaFail) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
RTCOfferAnswerOptions options;
options.use_rtp_mux = false;
auto offer = caller->CreateOffer(options);
// Modified the SDP to make two m= sections have the same SSRC.
ASSERT_GE(offer->description()->contents().size(), 2U);
offer->description()
->contents()[0]
.media_description()
->mutable_streams()[0]
.ssrcs[0] = 1111222;
offer->description()
->contents()[1]
.media_description()
->mutable_streams()[0]
.ssrcs[0] = 1111222;
EXPECT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
EXPECT_TRUE(callee->CreateAnswerAndSetAsLocal(options));
// Enable BUNDLE in subsequent offer/answer exchange and two m= sections are
// expectd to use one RtpTransport underneath.
options.use_rtp_mux = true;
EXPECT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(options)));
auto answer = callee->CreateAnswer(options);
// When BUNDLE is enabled, applying the description is expected to fail
// because the demuxing criteria is conflicted.
EXPECT_FALSE(callee->SetLocalDescription(std::move(answer)));
}
// This tests that changing the pre-negotiated BUNDLE tag is not supported.
TEST_P(PeerConnectionBundleTest, RejectDescriptionChangingBundleTag) {
RTCConfiguration config;
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
auto caller = CreatePeerConnectionWithAudioVideo(config);
auto callee = CreatePeerConnectionWithAudioVideo(config);
RTCOfferAnswerOptions options;
options.use_rtp_mux = true;
auto offer = caller->CreateOfferAndSetAsLocal(options);
// Create a new bundle-group with different bundled_mid.
auto* old_bundle_group =
offer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
std::string first_mid = old_bundle_group->content_names()[0];
std::string second_mid = old_bundle_group->content_names()[1];
cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
new_bundle_group.AddContentName(second_mid);
auto re_offer = CloneSessionDescription(offer.get());
callee->SetRemoteDescription(std::move(offer));
auto answer = callee->CreateAnswer(options);
// Reject the first MID.
answer->description()->contents()[0].rejected = true;
// Remove the first MID from the bundle group.
answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
answer->description()->AddGroup(new_bundle_group);
// The answer is expected to be rejected.
EXPECT_FALSE(caller->SetRemoteDescription(std::move(answer)));
// Do the same thing for re-offer.
re_offer->description()->contents()[0].rejected = true;
re_offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
re_offer->description()->AddGroup(new_bundle_group);
// The re-offer is expected to be rejected.
EXPECT_FALSE(caller->SetLocalDescription(std::move(re_offer)));
}
// This tests that removing contents from BUNDLE group and reject the whole
// BUNDLE group could work. This is a regression test for
// (https://bugs.chromium.org/p/chromium/issues/detail?id=827917)
TEST_P(PeerConnectionBundleTest, RemovingContentAndRejectBundleGroup) {
RTCConfiguration config;
#ifndef HAVE_SCTP
config.enable_rtp_data_channel = true;
#endif
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
auto caller = CreatePeerConnectionWithAudioVideo(config);
caller->CreateDataChannel("dc");
auto offer = caller->CreateOfferAndSetAsLocal();
auto re_offer = CloneSessionDescription(offer.get());
// Removing the second MID from the BUNDLE group.
auto* old_bundle_group =
offer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
std::string first_mid = old_bundle_group->content_names()[0];
std::string third_mid = old_bundle_group->content_names()[2];
cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
new_bundle_group.AddContentName(first_mid);
new_bundle_group.AddContentName(third_mid);
// Reject the entire new bundle group.
re_offer->description()->contents()[0].rejected = true;
re_offer->description()->contents()[2].rejected = true;
re_offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
re_offer->description()->AddGroup(new_bundle_group);
EXPECT_TRUE(caller->SetLocalDescription(std::move(re_offer)));
}
// This tests that the BUNDLE group in answer should be a subset of the offered
// group.
TEST_P(PeerConnectionBundleTest, AddContentToBundleGroupInAnswerNotSupported) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
auto offer = caller->CreateOffer();
std::string first_mid = offer->description()->contents()[0].name;
std::string second_mid = offer->description()->contents()[1].name;
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName(first_mid);
offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
offer->description()->AddGroup(bundle_group);
EXPECT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswer();
bundle_group.AddContentName(second_mid);
answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
answer->description()->AddGroup(bundle_group);
// The answer is expected to be rejected because second mid is not in the
// offered BUNDLE group.
EXPECT_FALSE(callee->SetLocalDescription(std::move(answer)));
}
// This tests that the BUNDLE group with non-existing MID should be rejectd.
TEST_P(PeerConnectionBundleTest, RejectBundleGroupWithNonExistingMid) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
auto offer = caller->CreateOffer();
auto invalid_bundle_group =
*offer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
invalid_bundle_group.AddContentName("non-existing-MID");
offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
offer->description()->AddGroup(invalid_bundle_group);
EXPECT_FALSE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer)));
}
// This tests that an answer shouldn't be able to remove an m= section from an
// established group without rejecting it.
TEST_P(PeerConnectionBundleTest, RemoveContentFromBundleGroup) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
EXPECT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
auto answer = callee->CreateAnswer();
std::string second_mid = answer->description()->contents()[1].name;
auto invalid_bundle_group =
*answer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
invalid_bundle_group.RemoveContentName(second_mid);
answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
answer->description()->AddGroup(invalid_bundle_group);
EXPECT_FALSE(
callee->SetLocalDescription(CloneSessionDescription(answer.get())));
}
INSTANTIATE_TEST_SUITE_P(PeerConnectionBundleTest,
PeerConnectionBundleTest,
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
// According to RFC5888, if an endpoint understands the semantics of an
// "a=group", it MUST return an answer with that group. So, an empty BUNDLE
// group is valid when the answerer rejects all m= sections (by stopping all
// transceivers), meaning there's nothing to bundle.
//
// Only writing this test for Unified Plan mode, since there's no way to reject
// m= sections in answers for Plan B without SDP munging.
TEST_F(PeerConnectionBundleTestUnifiedPlan,
EmptyBundleGroupCreatedInAnswerWhenAppropriate) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnection();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
// Stop all transceivers, causing all m= sections to be rejected.
for (const auto& transceiver : callee->pc()->GetTransceivers()) {
transceiver->Stop();
}
EXPECT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
// Verify that the answer actually contained an empty bundle group.
const SessionDescriptionInterface* desc = callee->pc()->local_description();
ASSERT_NE(nullptr, desc);
const cricket::ContentGroup* bundle_group =
desc->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
ASSERT_NE(nullptr, bundle_group);
EXPECT_TRUE(bundle_group->content_names().empty());
}
} // namespace webrtc
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