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webrtc_m130/pc/peer_connection_bundle_unittest.cc
Danil Chapovalov b1ec813339 Expose direct access to PeerConnection in PeerConnectionWrapper helper 
Multiple derived classes duplcated that code, and one more fixture
can benefit from the same direct access to avoid saving reference to port allocator

Cleaned includes and build dependencies on the way, in particular left single build target that contains peer_connection_wrapper

Bug: webrtc:42232556
Change-Id: Ieb3d5449f3a0285230847716e33fb3b2d1b47882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376300
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43847}
2025-02-04 10:46:11 -08:00

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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 <stddef.h>
#include <memory>
#include <ostream>
#include <string>
#include <tuple>
#include <utility>
#include <vector>
#include "api/candidate.h"
#include "api/enable_media_with_defaults.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/stats/rtc_stats_report.h"
#include "api/stats/rtcstats_objects.h"
#include "api/test/rtc_error_matchers.h"
#include "api/video_codecs/video_decoder_factory_template.h"
#include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h"
#include "api/video_codecs/video_encoder_factory_template.h"
#include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
#include "media/base/stream_params.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_info.h"
#include "pc/channel.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/rtp_transceiver.h"
#include "pc/rtp_transport_internal.h"
#include "pc/sdp_utils.h"
#include "pc/session_description.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/integration_test_helpers.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/checks.h"
#include "rtc_base/fake_network.h"
#include "rtc_base/logging.h"
#include "rtc_base/net_helper.h"
#include "rtc_base/network.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/thread.h"
#include "rtc_base/virtual_socket_server.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/wait_until.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/android_test_initializer.h"
#endif
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::UnorderedElementsAre;
using ::testing::Values;
// 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:
std::vector<const rtc::Network*> GetAnyAddressNetworks() 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";
return {};
}
};
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.mid());
std::unique_ptr<IceCandidateInterface> jsep_candidate =
CreateIceCandidate(content.mid(), i, *candidate);
return pc()->AddIceCandidate(jsep_candidate.get());
}
}
RTC_DCHECK_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;
}
// 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 == "in-progress" ||
*pair_stats->state == "succeeded") {
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)
: main_(&vss_), sdp_semantics_(sdp_semantics) {
#ifdef WEBRTC_ANDROID
InitializeAndroidObjects();
#endif
}
WrapperPtr CreatePeerConnection() {
return CreatePeerConnection(RTCConfiguration());
}
WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
// Each PeerConnection has its own `NetworkManager` which is injected into
// `PeerConnectionFactoryDependencies`, thus each PeerConnection in these
// tests is created with own PeerConnectionFactory.
PeerConnectionFactoryDependencies pcf_deps;
pcf_deps.network_thread = rtc::Thread::Current();
pcf_deps.worker_thread = rtc::Thread::Current();
pcf_deps.signaling_thread = rtc::Thread::Current();
pcf_deps.socket_factory = &vss_;
auto network_manager =
std::make_unique<FakeNetworkManagerWithNoAnyNetwork>();
auto* fake_network = network_manager.get();
pcf_deps.network_manager = std::move(network_manager);
pcf_deps.adm = FakeAudioCaptureModule::Create();
pcf_deps.video_encoder_factory =
std::make_unique<VideoEncoderFactoryTemplate<
LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>();
pcf_deps.video_decoder_factory =
std::make_unique<VideoDecoderFactoryTemplate<
LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter,
OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>();
EnableMediaWithDefaults(pcf_deps);
scoped_refptr<PeerConnectionFactoryInterface> pc_factory =
CreateModularPeerConnectionFactory(std::move(pcf_deps));
auto observer = std::make_unique<MockPeerConnectionObserver>();
RTCConfiguration modified_config = config;
modified_config.set_port_allocator_flags(
cricket::PORTALLOCATOR_DISABLE_TCP |
cricket::PORTALLOCATOR_DISABLE_RELAY);
modified_config.sdp_semantics = sdp_semantics_;
auto result = pc_factory->CreatePeerConnectionOrError(
modified_config, PeerConnectionDependencies(observer.get()));
if (!result.ok()) {
return nullptr;
}
auto wrapper = std::make_unique<PeerConnectionWrapperForBundleTest>(
std::move(pc_factory), result.MoveValue(), 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);
return candidate;
}
rtc::VirtualSocketServer vss_;
rtc::AutoSocketServerThread main_;
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_THAT(WaitUntil([&] { return caller->IsIceGatheringDone(); },
::testing::IsTrue()),
IsRtcOk());
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_THAT(WaitUntil([&] { return callee->IsIceGatheringDone(); },
::testing::IsTrue()),
IsRtcOk());
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_THAT(WaitUntil([&] { return caller->IsIceGatheringDone(); },
::testing::IsTrue()),
IsRtcOk());
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_THAT(WaitUntil([&] { return caller->IsIceGatheringDone(); },
::testing::IsTrue()),
IsRtcOk());
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_DEPRECATED, 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_THAT(
WaitUntil(
[&] {
return caller->HasConnectionWithRemoteAddress(kAudioAddress1);
},
::testing::IsTrue()),
IsRtcOk());
EXPECT_THAT(
WaitUntil(
[&] {
return caller->HasConnectionWithRemoteAddress(kAudioAddress2);
},
::testing::IsTrue()),
IsRtcOk());
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 when using BUNDLE.
TEST_P(PeerConnectionBundleTest, ApplyDescriptionWithSameSsrcsBundledFails) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
RTCOfferAnswerOptions options;
options.use_rtp_mux = true;
auto offer = caller->CreateOffer(options);
EXPECT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
// Modify the remote SDP to make two m= sections have the same SSRC.
ASSERT_GE(offer->description()->contents().size(), 2U);
ReplaceFirstSsrc(offer->description()
->contents()[0]
.media_description()
->mutable_streams()[0],
1111222);
ReplaceFirstSsrc(offer->description()
->contents()[1]
.media_description()
->mutable_streams()[0],
1111222);
EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
// When BUNDLE is enabled, applying the description is expected to fail
// because the demuxing criteria can not be satisfied.
auto answer = callee->CreateAnswer(options);
EXPECT_FALSE(callee->SetLocalDescription(std::move(answer)));
}
// A variant of the above, without BUNDLE duplicate SSRCs are allowed.
TEST_P(PeerConnectionBundleTest,
ApplyDescriptionWithSameSsrcsUnbundledSucceeds) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
RTCOfferAnswerOptions options;
options.use_rtp_mux = false;
auto offer = caller->CreateOffer(options);
EXPECT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
// Modify the remote SDP to make two m= sections have the same SSRC.
ASSERT_GE(offer->description()->contents().size(), 2U);
ReplaceFirstSsrc(offer->description()
->contents()[0]
.media_description()
->mutable_streams()[0],
1111222);
ReplaceFirstSsrc(offer->description()
->contents()[1]
.media_description()
->mutable_streams()[0],
1111222);
EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
// Without BUNDLE, demuxing is done per-transport.
auto answer = callee->CreateAnswer(options);
EXPECT_TRUE(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)
#ifdef HAVE_SCTP
TEST_P(PeerConnectionBundleTest, RemovingContentAndRejectBundleGroup) {
RTCConfiguration config;
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)));
}
#endif
// 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();
const auto first_mid = offer->description()->contents()[0].mid();
const auto second_mid = offer->description()->contents()[1].mid();
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();
const auto second_mid = answer->description()->contents()[1].mid();
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_DEPRECATED,
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->StopInternal();
}
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());
}
TEST_F(PeerConnectionBundleTestUnifiedPlan, MultipleBundleGroups) {
auto caller = CreatePeerConnection();
caller->AddAudioTrack("0_audio");
caller->AddAudioTrack("1_audio");
caller->AddVideoTrack("2_audio");
caller->AddVideoTrack("3_audio");
auto callee = CreatePeerConnection();
auto offer = caller->CreateOffer(RTCOfferAnswerOptions());
// Modify the GROUP to have two BUNDLEs. We know that the MIDs will be 0,1,2,4
// because our implementation has predictable MIDs.
offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
cricket::ContentGroup bundle_group1(cricket::GROUP_TYPE_BUNDLE);
bundle_group1.AddContentName("0");
bundle_group1.AddContentName("1");
cricket::ContentGroup bundle_group2(cricket::GROUP_TYPE_BUNDLE);
bundle_group2.AddContentName("2");
bundle_group2.AddContentName("3");
offer->description()->AddGroup(bundle_group1);
offer->description()->AddGroup(bundle_group2);
EXPECT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswer();
EXPECT_TRUE(
callee->SetLocalDescription(CloneSessionDescription(answer.get())));
EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer)));
// Verify bundling on sender side.
auto senders = caller->pc()->GetSenders();
ASSERT_EQ(senders.size(), 4u);
auto sender0_transport = senders[0]->dtls_transport();
auto sender1_transport = senders[1]->dtls_transport();
auto sender2_transport = senders[2]->dtls_transport();
auto sender3_transport = senders[3]->dtls_transport();
EXPECT_EQ(sender0_transport, sender1_transport);
EXPECT_EQ(sender2_transport, sender3_transport);
EXPECT_NE(sender0_transport, sender2_transport);
// Verify bundling on receiver side.
auto receivers = callee->pc()->GetReceivers();
ASSERT_EQ(receivers.size(), 4u);
auto receiver0_transport = receivers[0]->dtls_transport();
auto receiver1_transport = receivers[1]->dtls_transport();
auto receiver2_transport = receivers[2]->dtls_transport();
auto receiver3_transport = receivers[3]->dtls_transport();
EXPECT_EQ(receiver0_transport, receiver1_transport);
EXPECT_EQ(receiver2_transport, receiver3_transport);
EXPECT_NE(receiver0_transport, receiver2_transport);
}
// Test that, with the "max-compat" bundle policy, it's possible to add an m=
// section that's not part of an existing bundle group.
TEST_F(PeerConnectionBundleTestUnifiedPlan, AddNonBundledSection) {
RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxCompat;
auto caller = CreatePeerConnection(config);
caller->AddAudioTrack("0_audio");
caller->AddAudioTrack("1_audio");
auto callee = CreatePeerConnection(config);
// Establish an existing BUNDLE group.
auto offer = caller->CreateOffer(RTCOfferAnswerOptions());
EXPECT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswer();
EXPECT_TRUE(
callee->SetLocalDescription(CloneSessionDescription(answer.get())));
EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer)));
// Add a track but munge SDP so it's not part of the bundle group.
caller->AddAudioTrack("3_audio");
offer = caller->CreateOffer(RTCOfferAnswerOptions());
offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
bundle_group.AddContentName("0");
bundle_group.AddContentName("1");
offer->description()->AddGroup(bundle_group);
EXPECT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
answer = callee->CreateAnswer();
EXPECT_TRUE(
callee->SetLocalDescription(CloneSessionDescription(answer.get())));
EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer)));
// Verify bundling on the sender side.
auto senders = caller->pc()->GetSenders();
ASSERT_EQ(senders.size(), 3u);
auto sender0_transport = senders[0]->dtls_transport();
auto sender1_transport = senders[1]->dtls_transport();
auto sender2_transport = senders[2]->dtls_transport();
EXPECT_EQ(sender0_transport, sender1_transport);
EXPECT_NE(sender0_transport, sender2_transport);
// Verify bundling on receiver side.
auto receivers = callee->pc()->GetReceivers();
ASSERT_EQ(receivers.size(), 3u);
auto receiver0_transport = receivers[0]->dtls_transport();
auto receiver1_transport = receivers[1]->dtls_transport();
auto receiver2_transport = receivers[2]->dtls_transport();
EXPECT_EQ(receiver0_transport, receiver1_transport);
EXPECT_NE(receiver0_transport, receiver2_transport);
}
} // namespace webrtc
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