5651 lines
237 KiB
C++
5651 lines
237 KiB
C++
/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Disable for TSan v2, see
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// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
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#if !defined(THREAD_SANITIZER)
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#include <stdio.h>
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#include <functional>
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#include <list>
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#include <map>
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#include <memory>
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#include <utility>
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#include <vector>
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#include "absl/algorithm/container.h"
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#include "api/media_stream_interface.h"
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#include "api/peer_connection_interface.h"
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#include "api/peer_connection_proxy.h"
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#include "api/rtc_event_log/rtc_event_log_factory.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "api/uma_metrics.h"
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#include "api/video_codecs/sdp_video_format.h"
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#include "call/call.h"
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#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
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#include "media/engine/fake_webrtc_video_engine.h"
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#include "media/engine/webrtc_media_engine.h"
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#include "media/engine/webrtc_media_engine_defaults.h"
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#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
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#include "p2p/base/fake_ice_transport.h"
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#include "p2p/base/mock_async_resolver.h"
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#include "p2p/base/p2p_constants.h"
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#include "p2p/base/port_interface.h"
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#include "p2p/base/test_stun_server.h"
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#include "p2p/base/test_turn_customizer.h"
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#include "p2p/base/test_turn_server.h"
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#include "p2p/client/basic_port_allocator.h"
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#include "pc/dtmf_sender.h"
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#include "pc/local_audio_source.h"
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#include "pc/media_session.h"
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#include "pc/peer_connection.h"
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#include "pc/peer_connection_factory.h"
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#include "pc/rtp_media_utils.h"
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#include "pc/session_description.h"
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#include "pc/test/fake_audio_capture_module.h"
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#include "pc/test/fake_periodic_video_track_source.h"
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#include "pc/test/fake_rtc_certificate_generator.h"
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#include "pc/test/fake_video_track_renderer.h"
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#include "pc/test/mock_peer_connection_observers.h"
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#include "rtc_base/fake_clock.h"
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#include "rtc_base/fake_mdns_responder.h"
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#include "rtc_base/fake_network.h"
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#include "rtc_base/firewall_socket_server.h"
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#include "rtc_base/gunit.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/test_certificate_verifier.h"
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#include "rtc_base/time_utils.h"
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#include "rtc_base/virtual_socket_server.h"
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#include "system_wrappers/include/metrics.h"
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#include "test/field_trial.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace {
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using ::cricket::ContentInfo;
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using ::cricket::StreamParams;
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using ::rtc::SocketAddress;
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using ::testing::_;
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using ::testing::Combine;
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using ::testing::Contains;
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using ::testing::DoAll;
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using ::testing::ElementsAre;
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using ::testing::NiceMock;
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using ::testing::Return;
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using ::testing::SetArgPointee;
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using ::testing::UnorderedElementsAreArray;
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using ::testing::Values;
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using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
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static const int kDefaultTimeout = 10000;
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static const int kMaxWaitForStatsMs = 3000;
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static const int kMaxWaitForActivationMs = 5000;
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static const int kMaxWaitForFramesMs = 10000;
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// Default number of audio/video frames to wait for before considering a test
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// successful.
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static const int kDefaultExpectedAudioFrameCount = 3;
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static const int kDefaultExpectedVideoFrameCount = 3;
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static const char kDataChannelLabel[] = "data_channel";
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// SRTP cipher name negotiated by the tests. This must be updated if the
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// default changes.
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static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_80;
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static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;
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static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0);
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// Helper function for constructing offer/answer options to initiate an ICE
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// restart.
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PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() {
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PeerConnectionInterface::RTCOfferAnswerOptions options;
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options.ice_restart = true;
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return options;
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}
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// Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic"
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// attribute from received SDP, simulating a legacy endpoint.
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void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) {
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for (ContentInfo& content : desc->contents()) {
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content.media_description()->mutable_streams().clear();
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}
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desc->set_msid_supported(false);
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desc->set_msid_signaling(0);
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}
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// Removes all stream information besides the stream ids, simulating an
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// endpoint that only signals a=msid lines to convey stream_ids.
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void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
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for (ContentInfo& content : desc->contents()) {
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std::string track_id;
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std::vector<std::string> stream_ids;
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if (!content.media_description()->streams().empty()) {
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const StreamParams& first_stream =
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content.media_description()->streams()[0];
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track_id = first_stream.id;
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stream_ids = first_stream.stream_ids();
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}
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content.media_description()->mutable_streams().clear();
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StreamParams new_stream;
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new_stream.id = track_id;
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new_stream.set_stream_ids(stream_ids);
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content.media_description()->AddStream(new_stream);
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}
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}
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int FindFirstMediaStatsIndexByKind(
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const std::string& kind,
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const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
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media_stats_vec) {
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for (size_t i = 0; i < media_stats_vec.size(); i++) {
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if (media_stats_vec[i]->kind.ValueToString() == kind) {
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return i;
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}
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}
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return -1;
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}
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class SignalingMessageReceiver {
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public:
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virtual void ReceiveSdpMessage(SdpType type, const std::string& msg) = 0;
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virtual void ReceiveIceMessage(const std::string& sdp_mid,
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int sdp_mline_index,
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const std::string& msg) = 0;
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protected:
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SignalingMessageReceiver() {}
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virtual ~SignalingMessageReceiver() {}
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};
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class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
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public:
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explicit MockRtpReceiverObserver(cricket::MediaType media_type)
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: expected_media_type_(media_type) {}
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void OnFirstPacketReceived(cricket::MediaType media_type) override {
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ASSERT_EQ(expected_media_type_, media_type);
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first_packet_received_ = true;
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}
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bool first_packet_received() const { return first_packet_received_; }
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virtual ~MockRtpReceiverObserver() {}
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private:
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bool first_packet_received_ = false;
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cricket::MediaType expected_media_type_;
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};
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// Helper class that wraps a peer connection, observes it, and can accept
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// signaling messages from another wrapper.
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//
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// Uses a fake network, fake A/V capture, and optionally fake
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// encoders/decoders, though they aren't used by default since they don't
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// advertise support of any codecs.
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// TODO(steveanton): See how this could become a subclass of
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// PeerConnectionWrapper defined in peerconnectionwrapper.h.
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class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
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public SignalingMessageReceiver {
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public:
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// Different factory methods for convenience.
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// TODO(deadbeef): Could use the pattern of:
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//
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// PeerConnectionWrapper =
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// WrapperBuilder.WithConfig(...).WithOptions(...).build();
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//
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// To reduce some code duplication.
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static PeerConnectionWrapper* CreateWithDtlsIdentityStore(
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const std::string& debug_name,
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std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
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rtc::Thread* network_thread,
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rtc::Thread* worker_thread) {
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PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
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webrtc::PeerConnectionDependencies dependencies(nullptr);
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dependencies.cert_generator = std::move(cert_generator);
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if (!client->Init(nullptr, nullptr, std::move(dependencies), network_thread,
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worker_thread, nullptr,
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/*reset_encoder_factory=*/false,
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/*reset_decoder_factory=*/false)) {
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delete client;
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return nullptr;
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}
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return client;
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}
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webrtc::PeerConnectionFactoryInterface* pc_factory() const {
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return peer_connection_factory_.get();
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}
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webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
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// If a signaling message receiver is set (via ConnectFakeSignaling), this
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// will set the whole offer/answer exchange in motion. Just need to wait for
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// the signaling state to reach "stable".
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void CreateAndSetAndSignalOffer() {
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auto offer = CreateOfferAndWait();
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ASSERT_NE(nullptr, offer);
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EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer)));
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}
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// Sets the options to be used when CreateAndSetAndSignalOffer is called, or
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// when a remote offer is received (via fake signaling) and an answer is
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// generated. By default, uses default options.
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void SetOfferAnswerOptions(
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const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
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offer_answer_options_ = options;
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}
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// Set a callback to be invoked when SDP is received via the fake signaling
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// channel, which provides an opportunity to munge (modify) the SDP. This is
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// used to test SDP being applied that a PeerConnection would normally not
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// generate, but a non-JSEP endpoint might.
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void SetReceivedSdpMunger(
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std::function<void(cricket::SessionDescription*)> munger) {
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received_sdp_munger_ = std::move(munger);
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}
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// Similar to the above, but this is run on SDP immediately after it's
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// generated.
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void SetGeneratedSdpMunger(
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std::function<void(cricket::SessionDescription*)> munger) {
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generated_sdp_munger_ = std::move(munger);
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}
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// Set a callback to be invoked when a remote offer is received via the fake
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// signaling channel. This provides an opportunity to change the
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// PeerConnection state before an answer is created and sent to the caller.
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void SetRemoteOfferHandler(std::function<void()> handler) {
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remote_offer_handler_ = std::move(handler);
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}
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void SetRemoteAsyncResolver(rtc::MockAsyncResolver* resolver) {
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remote_async_resolver_ = resolver;
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}
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// Every ICE connection state in order that has been seen by the observer.
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std::vector<PeerConnectionInterface::IceConnectionState>
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ice_connection_state_history() const {
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return ice_connection_state_history_;
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}
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void clear_ice_connection_state_history() {
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ice_connection_state_history_.clear();
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}
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// Every standardized ICE connection state in order that has been seen by the
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// observer.
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std::vector<PeerConnectionInterface::IceConnectionState>
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standardized_ice_connection_state_history() const {
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return standardized_ice_connection_state_history_;
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}
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// Every PeerConnection state in order that has been seen by the observer.
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std::vector<PeerConnectionInterface::PeerConnectionState>
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peer_connection_state_history() const {
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return peer_connection_state_history_;
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}
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// Every ICE gathering state in order that has been seen by the observer.
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std::vector<PeerConnectionInterface::IceGatheringState>
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ice_gathering_state_history() const {
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return ice_gathering_state_history_;
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}
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std::vector<cricket::CandidatePairChangeEvent>
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ice_candidate_pair_change_history() const {
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return ice_candidate_pair_change_history_;
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}
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// Every PeerConnection signaling state in order that has been seen by the
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// observer.
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std::vector<PeerConnectionInterface::SignalingState>
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peer_connection_signaling_state_history() const {
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return peer_connection_signaling_state_history_;
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}
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void AddAudioVideoTracks() {
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AddAudioTrack();
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AddVideoTrack();
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}
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rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack() {
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return AddTrack(CreateLocalAudioTrack());
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}
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rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack() {
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return AddTrack(CreateLocalVideoTrack());
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}
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rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
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cricket::AudioOptions options;
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// Disable highpass filter so that we can get all the test audio frames.
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options.highpass_filter = false;
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rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
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peer_connection_factory_->CreateAudioSource(options);
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// TODO(perkj): Test audio source when it is implemented. Currently audio
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// always use the default input.
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return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(),
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source);
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}
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rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
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webrtc::FakePeriodicVideoSource::Config config;
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config.timestamp_offset_ms = rtc::TimeMillis();
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return CreateLocalVideoTrackInternal(config);
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}
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rtc::scoped_refptr<webrtc::VideoTrackInterface>
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CreateLocalVideoTrackWithConfig(
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webrtc::FakePeriodicVideoSource::Config config) {
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return CreateLocalVideoTrackInternal(config);
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}
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rtc::scoped_refptr<webrtc::VideoTrackInterface>
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CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
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webrtc::FakePeriodicVideoSource::Config config;
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config.rotation = rotation;
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config.timestamp_offset_ms = rtc::TimeMillis();
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return CreateLocalVideoTrackInternal(config);
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}
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rtc::scoped_refptr<RtpSenderInterface> AddTrack(
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rtc::scoped_refptr<MediaStreamTrackInterface> track,
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const std::vector<std::string>& stream_ids = {}) {
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auto result = pc()->AddTrack(track, stream_ids);
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EXPECT_EQ(RTCErrorType::NONE, result.error().type());
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return result.MoveValue();
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}
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std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType(
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cricket::MediaType media_type) {
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std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
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for (const auto& receiver : pc()->GetReceivers()) {
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if (receiver->media_type() == media_type) {
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receivers.push_back(receiver);
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}
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}
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return receivers;
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}
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rtc::scoped_refptr<RtpTransceiverInterface> GetFirstTransceiverOfType(
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cricket::MediaType media_type) {
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for (auto transceiver : pc()->GetTransceivers()) {
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if (transceiver->receiver()->media_type() == media_type) {
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return transceiver;
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}
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}
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return nullptr;
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}
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bool SignalingStateStable() {
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return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
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}
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void CreateDataChannel() { CreateDataChannel(nullptr); }
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void CreateDataChannel(const webrtc::DataChannelInit* init) {
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CreateDataChannel(kDataChannelLabel, init);
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}
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void CreateDataChannel(const std::string& label,
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const webrtc::DataChannelInit* init) {
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data_channel_ = pc()->CreateDataChannel(label, init);
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ASSERT_TRUE(data_channel_.get() != nullptr);
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data_observer_.reset(new MockDataChannelObserver(data_channel_));
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}
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DataChannelInterface* data_channel() { return data_channel_; }
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const MockDataChannelObserver* data_observer() const {
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return data_observer_.get();
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}
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int audio_frames_received() const {
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return fake_audio_capture_module_->frames_received();
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}
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// Takes minimum of video frames received for each track.
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//
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// Can be used like:
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// EXPECT_GE(expected_frames, min_video_frames_received_per_track());
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//
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// To ensure that all video tracks received at least a certain number of
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// frames.
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int min_video_frames_received_per_track() const {
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int min_frames = INT_MAX;
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if (fake_video_renderers_.empty()) {
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return 0;
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}
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for (const auto& pair : fake_video_renderers_) {
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min_frames = std::min(min_frames, pair.second->num_rendered_frames());
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}
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return min_frames;
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}
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// Returns a MockStatsObserver in a state after stats gathering finished,
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// which can be used to access the gathered stats.
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rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack(
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webrtc::MediaStreamTrackInterface* track) {
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rtc::scoped_refptr<MockStatsObserver> observer(
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new rtc::RefCountedObject<MockStatsObserver>());
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EXPECT_TRUE(peer_connection_->GetStats(
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observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
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EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
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return observer;
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}
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// Version that doesn't take a track "filter", and gathers all stats.
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rtc::scoped_refptr<MockStatsObserver> OldGetStats() {
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return OldGetStatsForTrack(nullptr);
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}
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// Synchronously gets stats and returns them. If it times out, fails the test
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// and returns null.
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rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
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rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
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new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
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peer_connection_->GetStats(callback);
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EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
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return callback->report();
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}
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int rendered_width() {
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EXPECT_FALSE(fake_video_renderers_.empty());
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return fake_video_renderers_.empty()
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? 0
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: fake_video_renderers_.begin()->second->width();
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}
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int rendered_height() {
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EXPECT_FALSE(fake_video_renderers_.empty());
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return fake_video_renderers_.empty()
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? 0
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: fake_video_renderers_.begin()->second->height();
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}
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double rendered_aspect_ratio() {
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if (rendered_height() == 0) {
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return 0.0;
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}
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return static_cast<double>(rendered_width()) / rendered_height();
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}
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|
|
webrtc::VideoRotation rendered_rotation() {
|
|
EXPECT_FALSE(fake_video_renderers_.empty());
|
|
return fake_video_renderers_.empty()
|
|
? webrtc::kVideoRotation_0
|
|
: fake_video_renderers_.begin()->second->rotation();
|
|
}
|
|
|
|
int local_rendered_width() {
|
|
return local_video_renderer_ ? local_video_renderer_->width() : 0;
|
|
}
|
|
|
|
int local_rendered_height() {
|
|
return local_video_renderer_ ? local_video_renderer_->height() : 0;
|
|
}
|
|
|
|
double local_rendered_aspect_ratio() {
|
|
if (local_rendered_height() == 0) {
|
|
return 0.0;
|
|
}
|
|
return static_cast<double>(local_rendered_width()) /
|
|
local_rendered_height();
|
|
}
|
|
|
|
size_t number_of_remote_streams() {
|
|
if (!pc()) {
|
|
return 0;
|
|
}
|
|
return pc()->remote_streams()->count();
|
|
}
|
|
|
|
StreamCollectionInterface* remote_streams() const {
|
|
if (!pc()) {
|
|
ADD_FAILURE();
|
|
return nullptr;
|
|
}
|
|
return pc()->remote_streams();
|
|
}
|
|
|
|
StreamCollectionInterface* local_streams() {
|
|
if (!pc()) {
|
|
ADD_FAILURE();
|
|
return nullptr;
|
|
}
|
|
return pc()->local_streams();
|
|
}
|
|
|
|
webrtc::PeerConnectionInterface::SignalingState signaling_state() {
|
|
return pc()->signaling_state();
|
|
}
|
|
|
|
webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
|
|
return pc()->ice_connection_state();
|
|
}
|
|
|
|
webrtc::PeerConnectionInterface::IceConnectionState
|
|
standardized_ice_connection_state() {
|
|
return pc()->standardized_ice_connection_state();
|
|
}
|
|
|
|
webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
|
|
return pc()->ice_gathering_state();
|
|
}
|
|
|
|
// Returns a MockRtpReceiverObserver for each RtpReceiver returned by
|
|
// GetReceivers. They're updated automatically when a remote offer/answer
|
|
// from the fake signaling channel is applied, or when
|
|
// ResetRtpReceiverObservers below is called.
|
|
const std::vector<std::unique_ptr<MockRtpReceiverObserver>>&
|
|
rtp_receiver_observers() {
|
|
return rtp_receiver_observers_;
|
|
}
|
|
|
|
void ResetRtpReceiverObservers() {
|
|
rtp_receiver_observers_.clear();
|
|
for (const rtc::scoped_refptr<RtpReceiverInterface>& receiver :
|
|
pc()->GetReceivers()) {
|
|
std::unique_ptr<MockRtpReceiverObserver> observer(
|
|
new MockRtpReceiverObserver(receiver->media_type()));
|
|
receiver->SetObserver(observer.get());
|
|
rtp_receiver_observers_.push_back(std::move(observer));
|
|
}
|
|
}
|
|
|
|
rtc::FakeNetworkManager* network_manager() const {
|
|
return fake_network_manager_.get();
|
|
}
|
|
cricket::PortAllocator* port_allocator() const { return port_allocator_; }
|
|
|
|
webrtc::FakeRtcEventLogFactory* event_log_factory() const {
|
|
return event_log_factory_;
|
|
}
|
|
|
|
const cricket::Candidate& last_candidate_gathered() const {
|
|
return last_candidate_gathered_;
|
|
}
|
|
const cricket::IceCandidateErrorEvent& error_event() const {
|
|
return error_event_;
|
|
}
|
|
|
|
// Sets the mDNS responder for the owned fake network manager and keeps a
|
|
// reference to the responder.
|
|
void SetMdnsResponder(
|
|
std::unique_ptr<webrtc::FakeMdnsResponder> mdns_responder) {
|
|
RTC_DCHECK(mdns_responder != nullptr);
|
|
mdns_responder_ = mdns_responder.get();
|
|
network_manager()->set_mdns_responder(std::move(mdns_responder));
|
|
}
|
|
|
|
// Returns null on failure.
|
|
std::unique_ptr<SessionDescriptionInterface> CreateOfferAndWait() {
|
|
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
|
|
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
|
|
pc()->CreateOffer(observer, offer_answer_options_);
|
|
return WaitForDescriptionFromObserver(observer);
|
|
}
|
|
bool Rollback() {
|
|
return SetRemoteDescription(
|
|
webrtc::CreateSessionDescription(SdpType::kRollback, ""));
|
|
}
|
|
|
|
private:
|
|
explicit PeerConnectionWrapper(const std::string& debug_name)
|
|
: debug_name_(debug_name) {}
|
|
|
|
bool Init(
|
|
const PeerConnectionFactory::Options* options,
|
|
const PeerConnectionInterface::RTCConfiguration* config,
|
|
webrtc::PeerConnectionDependencies dependencies,
|
|
rtc::Thread* network_thread,
|
|
rtc::Thread* worker_thread,
|
|
std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
|
|
bool reset_encoder_factory,
|
|
bool reset_decoder_factory) {
|
|
// There's an error in this test code if Init ends up being called twice.
|
|
RTC_DCHECK(!peer_connection_);
|
|
RTC_DCHECK(!peer_connection_factory_);
|
|
|
|
fake_network_manager_.reset(new rtc::FakeNetworkManager());
|
|
fake_network_manager_->AddInterface(kDefaultLocalAddress);
|
|
|
|
std::unique_ptr<cricket::PortAllocator> port_allocator(
|
|
new cricket::BasicPortAllocator(fake_network_manager_.get()));
|
|
port_allocator_ = port_allocator.get();
|
|
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
|
|
if (!fake_audio_capture_module_) {
|
|
return false;
|
|
}
|
|
rtc::Thread* const signaling_thread = rtc::Thread::Current();
|
|
|
|
webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies;
|
|
pc_factory_dependencies.network_thread = network_thread;
|
|
pc_factory_dependencies.worker_thread = worker_thread;
|
|
pc_factory_dependencies.signaling_thread = signaling_thread;
|
|
pc_factory_dependencies.task_queue_factory =
|
|
webrtc::CreateDefaultTaskQueueFactory();
|
|
cricket::MediaEngineDependencies media_deps;
|
|
media_deps.task_queue_factory =
|
|
pc_factory_dependencies.task_queue_factory.get();
|
|
media_deps.adm = fake_audio_capture_module_;
|
|
webrtc::SetMediaEngineDefaults(&media_deps);
|
|
|
|
if (reset_encoder_factory) {
|
|
media_deps.video_encoder_factory.reset();
|
|
}
|
|
if (reset_decoder_factory) {
|
|
media_deps.video_decoder_factory.reset();
|
|
}
|
|
|
|
if (!media_deps.audio_processing) {
|
|
// If the standard Creation method for APM returns a null pointer, instead
|
|
// use the builder for testing to create an APM object.
|
|
media_deps.audio_processing = AudioProcessingBuilderForTesting().Create();
|
|
}
|
|
|
|
pc_factory_dependencies.media_engine =
|
|
cricket::CreateMediaEngine(std::move(media_deps));
|
|
pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();
|
|
if (event_log_factory) {
|
|
event_log_factory_ = event_log_factory.get();
|
|
pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
|
|
} else {
|
|
pc_factory_dependencies.event_log_factory =
|
|
std::make_unique<webrtc::RtcEventLogFactory>(
|
|
pc_factory_dependencies.task_queue_factory.get());
|
|
}
|
|
peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
|
|
std::move(pc_factory_dependencies));
|
|
|
|
if (!peer_connection_factory_) {
|
|
return false;
|
|
}
|
|
if (options) {
|
|
peer_connection_factory_->SetOptions(*options);
|
|
}
|
|
if (config) {
|
|
sdp_semantics_ = config->sdp_semantics;
|
|
}
|
|
|
|
dependencies.allocator = std::move(port_allocator);
|
|
peer_connection_ = CreatePeerConnection(config, std::move(dependencies));
|
|
return peer_connection_.get() != nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
|
|
const PeerConnectionInterface::RTCConfiguration* config,
|
|
webrtc::PeerConnectionDependencies dependencies) {
|
|
PeerConnectionInterface::RTCConfiguration modified_config;
|
|
// If |config| is null, this will result in a default configuration being
|
|
// used.
|
|
if (config) {
|
|
modified_config = *config;
|
|
}
|
|
// Disable resolution adaptation; we don't want it interfering with the
|
|
// test results.
|
|
// TODO(deadbeef): Do something more robust. Since we're testing for aspect
|
|
// ratios and not specific resolutions, is this even necessary?
|
|
modified_config.set_cpu_adaptation(false);
|
|
|
|
dependencies.observer = this;
|
|
return peer_connection_factory_->CreatePeerConnection(
|
|
modified_config, std::move(dependencies));
|
|
}
|
|
|
|
void set_signaling_message_receiver(
|
|
SignalingMessageReceiver* signaling_message_receiver) {
|
|
signaling_message_receiver_ = signaling_message_receiver;
|
|
}
|
|
|
|
void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; }
|
|
|
|
void set_signal_ice_candidates(bool signal) {
|
|
signal_ice_candidates_ = signal;
|
|
}
|
|
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
|
|
webrtc::FakePeriodicVideoSource::Config config) {
|
|
// Set max frame rate to 10fps to reduce the risk of test flakiness.
|
|
// TODO(deadbeef): Do something more robust.
|
|
config.frame_interval_ms = 100;
|
|
|
|
video_track_sources_.emplace_back(
|
|
new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>(
|
|
config, false /* remote */));
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
|
|
peer_connection_factory_->CreateVideoTrack(
|
|
rtc::CreateRandomUuid(), video_track_sources_.back()));
|
|
if (!local_video_renderer_) {
|
|
local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
|
|
}
|
|
return track;
|
|
}
|
|
|
|
void HandleIncomingOffer(const std::string& msg) {
|
|
RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer";
|
|
std::unique_ptr<SessionDescriptionInterface> desc =
|
|
webrtc::CreateSessionDescription(SdpType::kOffer, msg);
|
|
if (received_sdp_munger_) {
|
|
received_sdp_munger_(desc->description());
|
|
}
|
|
|
|
EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
|
|
// Setting a remote description may have changed the number of receivers,
|
|
// so reset the receiver observers.
|
|
ResetRtpReceiverObservers();
|
|
if (remote_offer_handler_) {
|
|
remote_offer_handler_();
|
|
}
|
|
auto answer = CreateAnswer();
|
|
ASSERT_NE(nullptr, answer);
|
|
EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer)));
|
|
}
|
|
|
|
void HandleIncomingAnswer(const std::string& msg) {
|
|
RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer";
|
|
std::unique_ptr<SessionDescriptionInterface> desc =
|
|
webrtc::CreateSessionDescription(SdpType::kAnswer, msg);
|
|
if (received_sdp_munger_) {
|
|
received_sdp_munger_(desc->description());
|
|
}
|
|
|
|
EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
|
|
// Set the RtpReceiverObserver after receivers are created.
|
|
ResetRtpReceiverObservers();
|
|
}
|
|
|
|
// Returns null on failure.
|
|
std::unique_ptr<SessionDescriptionInterface> CreateAnswer() {
|
|
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
|
|
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
|
|
pc()->CreateAnswer(observer, offer_answer_options_);
|
|
return WaitForDescriptionFromObserver(observer);
|
|
}
|
|
|
|
std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver(
|
|
MockCreateSessionDescriptionObserver* observer) {
|
|
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
|
|
if (!observer->result()) {
|
|
return nullptr;
|
|
}
|
|
auto description = observer->MoveDescription();
|
|
if (generated_sdp_munger_) {
|
|
generated_sdp_munger_(description->description());
|
|
}
|
|
return description;
|
|
}
|
|
|
|
// Setting the local description and sending the SDP message over the fake
|
|
// signaling channel are combined into the same method because the SDP
|
|
// message needs to be sent as soon as SetLocalDescription finishes, without
|
|
// waiting for the observer to be called. This ensures that ICE candidates
|
|
// don't outrace the description.
|
|
bool SetLocalDescriptionAndSendSdpMessage(
|
|
std::unique_ptr<SessionDescriptionInterface> desc) {
|
|
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
|
|
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
|
|
RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage";
|
|
SdpType type = desc->GetType();
|
|
std::string sdp;
|
|
EXPECT_TRUE(desc->ToString(&sdp));
|
|
RTC_LOG(LS_INFO) << debug_name_ << ": local SDP contents=\n" << sdp;
|
|
pc()->SetLocalDescription(observer, desc.release());
|
|
RemoveUnusedVideoRenderers();
|
|
// As mentioned above, we need to send the message immediately after
|
|
// SetLocalDescription.
|
|
SendSdpMessage(type, sdp);
|
|
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
|
|
return true;
|
|
}
|
|
|
|
bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) {
|
|
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
|
|
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
|
|
RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription";
|
|
pc()->SetRemoteDescription(observer, desc.release());
|
|
RemoveUnusedVideoRenderers();
|
|
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
|
|
return observer->result();
|
|
}
|
|
|
|
// This is a work around to remove unused fake_video_renderers from
|
|
// transceivers that have either stopped or are no longer receiving.
|
|
void RemoveUnusedVideoRenderers() {
|
|
if (sdp_semantics_ != SdpSemantics::kUnifiedPlan) {
|
|
return;
|
|
}
|
|
auto transceivers = pc()->GetTransceivers();
|
|
std::set<std::string> active_renderers;
|
|
for (auto& transceiver : transceivers) {
|
|
// Note - we don't check for direction here. This function is called
|
|
// before direction is set, and in that case, we should not remove
|
|
// the renderer.
|
|
if (transceiver->receiver()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
active_renderers.insert(transceiver->receiver()->track()->id());
|
|
}
|
|
}
|
|
for (auto it = fake_video_renderers_.begin();
|
|
it != fake_video_renderers_.end();) {
|
|
// Remove fake video renderers belonging to any non-active transceivers.
|
|
if (!active_renderers.count(it->first)) {
|
|
it = fake_video_renderers_.erase(it);
|
|
} else {
|
|
it++;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Simulate sending a blob of SDP with delay |signaling_delay_ms_| (0 by
|
|
// default).
|
|
void SendSdpMessage(SdpType type, const std::string& msg) {
|
|
if (signaling_delay_ms_ == 0) {
|
|
RelaySdpMessageIfReceiverExists(type, msg);
|
|
} else {
|
|
invoker_.AsyncInvokeDelayed<void>(
|
|
RTC_FROM_HERE, rtc::Thread::Current(),
|
|
rtc::Bind(&PeerConnectionWrapper::RelaySdpMessageIfReceiverExists,
|
|
this, type, msg),
|
|
signaling_delay_ms_);
|
|
}
|
|
}
|
|
|
|
void RelaySdpMessageIfReceiverExists(SdpType type, const std::string& msg) {
|
|
if (signaling_message_receiver_) {
|
|
signaling_message_receiver_->ReceiveSdpMessage(type, msg);
|
|
}
|
|
}
|
|
|
|
// Simulate trickling an ICE candidate with delay |signaling_delay_ms_| (0 by
|
|
// default).
|
|
void SendIceMessage(const std::string& sdp_mid,
|
|
int sdp_mline_index,
|
|
const std::string& msg) {
|
|
if (signaling_delay_ms_ == 0) {
|
|
RelayIceMessageIfReceiverExists(sdp_mid, sdp_mline_index, msg);
|
|
} else {
|
|
invoker_.AsyncInvokeDelayed<void>(
|
|
RTC_FROM_HERE, rtc::Thread::Current(),
|
|
rtc::Bind(&PeerConnectionWrapper::RelayIceMessageIfReceiverExists,
|
|
this, sdp_mid, sdp_mline_index, msg),
|
|
signaling_delay_ms_);
|
|
}
|
|
}
|
|
|
|
void RelayIceMessageIfReceiverExists(const std::string& sdp_mid,
|
|
int sdp_mline_index,
|
|
const std::string& msg) {
|
|
if (signaling_message_receiver_) {
|
|
signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index,
|
|
msg);
|
|
}
|
|
}
|
|
|
|
// SignalingMessageReceiver callbacks.
|
|
void ReceiveSdpMessage(SdpType type, const std::string& msg) override {
|
|
if (type == SdpType::kOffer) {
|
|
HandleIncomingOffer(msg);
|
|
} else {
|
|
HandleIncomingAnswer(msg);
|
|
}
|
|
}
|
|
|
|
void ReceiveIceMessage(const std::string& sdp_mid,
|
|
int sdp_mline_index,
|
|
const std::string& msg) override {
|
|
RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage";
|
|
std::unique_ptr<webrtc::IceCandidateInterface> candidate(
|
|
webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
|
|
EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
|
|
}
|
|
|
|
// PeerConnectionObserver callbacks.
|
|
void OnSignalingChange(
|
|
webrtc::PeerConnectionInterface::SignalingState new_state) override {
|
|
EXPECT_EQ(pc()->signaling_state(), new_state);
|
|
peer_connection_signaling_state_history_.push_back(new_state);
|
|
}
|
|
void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
|
|
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
|
|
streams) override {
|
|
if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
rtc::scoped_refptr<VideoTrackInterface> video_track(
|
|
static_cast<VideoTrackInterface*>(receiver->track().get()));
|
|
ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
|
|
fake_video_renderers_.end());
|
|
fake_video_renderers_[video_track->id()] =
|
|
std::make_unique<FakeVideoTrackRenderer>(video_track);
|
|
}
|
|
}
|
|
void OnRemoveTrack(
|
|
rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
|
|
if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
auto it = fake_video_renderers_.find(receiver->track()->id());
|
|
if (it != fake_video_renderers_.end()) {
|
|
fake_video_renderers_.erase(it);
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "OnRemoveTrack called for non-active renderer";
|
|
}
|
|
}
|
|
}
|
|
void OnRenegotiationNeeded() override {}
|
|
void OnIceConnectionChange(
|
|
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
|
|
EXPECT_EQ(pc()->ice_connection_state(), new_state);
|
|
ice_connection_state_history_.push_back(new_state);
|
|
}
|
|
void OnStandardizedIceConnectionChange(
|
|
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
|
|
standardized_ice_connection_state_history_.push_back(new_state);
|
|
}
|
|
void OnConnectionChange(
|
|
webrtc::PeerConnectionInterface::PeerConnectionState new_state) override {
|
|
peer_connection_state_history_.push_back(new_state);
|
|
}
|
|
|
|
void OnIceGatheringChange(
|
|
webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
|
|
EXPECT_EQ(pc()->ice_gathering_state(), new_state);
|
|
ice_gathering_state_history_.push_back(new_state);
|
|
}
|
|
|
|
void OnIceSelectedCandidatePairChanged(
|
|
const cricket::CandidatePairChangeEvent& event) {
|
|
ice_candidate_pair_change_history_.push_back(event);
|
|
}
|
|
|
|
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
|
|
RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate";
|
|
|
|
if (remote_async_resolver_) {
|
|
const auto& local_candidate = candidate->candidate();
|
|
if (local_candidate.address().IsUnresolvedIP()) {
|
|
RTC_DCHECK(local_candidate.type() == cricket::LOCAL_PORT_TYPE);
|
|
rtc::SocketAddress resolved_addr(local_candidate.address());
|
|
const auto resolved_ip = mdns_responder_->GetMappedAddressForName(
|
|
local_candidate.address().hostname());
|
|
RTC_DCHECK(!resolved_ip.IsNil());
|
|
resolved_addr.SetResolvedIP(resolved_ip);
|
|
EXPECT_CALL(*remote_async_resolver_, GetResolvedAddress(_, _))
|
|
.WillOnce(DoAll(SetArgPointee<1>(resolved_addr), Return(true)));
|
|
EXPECT_CALL(*remote_async_resolver_, Destroy(_));
|
|
}
|
|
}
|
|
|
|
std::string ice_sdp;
|
|
EXPECT_TRUE(candidate->ToString(&ice_sdp));
|
|
if (signaling_message_receiver_ == nullptr || !signal_ice_candidates_) {
|
|
// Remote party may be deleted.
|
|
return;
|
|
}
|
|
SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
|
|
last_candidate_gathered_ = candidate->candidate();
|
|
}
|
|
void OnIceCandidateError(const std::string& address,
|
|
int port,
|
|
const std::string& url,
|
|
int error_code,
|
|
const std::string& error_text) override {
|
|
error_event_ = cricket::IceCandidateErrorEvent(address, port, url,
|
|
error_code, error_text);
|
|
}
|
|
void OnDataChannel(
|
|
rtc::scoped_refptr<DataChannelInterface> data_channel) override {
|
|
RTC_LOG(LS_INFO) << debug_name_ << ": OnDataChannel";
|
|
data_channel_ = data_channel;
|
|
data_observer_.reset(new MockDataChannelObserver(data_channel));
|
|
}
|
|
|
|
std::string debug_name_;
|
|
|
|
std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
|
|
// Reference to the mDNS responder owned by |fake_network_manager_| after set.
|
|
webrtc::FakeMdnsResponder* mdns_responder_ = nullptr;
|
|
|
|
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
|
|
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
|
|
peer_connection_factory_;
|
|
|
|
cricket::PortAllocator* port_allocator_;
|
|
// Needed to keep track of number of frames sent.
|
|
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
|
|
// Needed to keep track of number of frames received.
|
|
std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
|
|
fake_video_renderers_;
|
|
// Needed to ensure frames aren't received for removed tracks.
|
|
std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
|
|
removed_fake_video_renderers_;
|
|
|
|
// For remote peer communication.
|
|
SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
|
|
int signaling_delay_ms_ = 0;
|
|
bool signal_ice_candidates_ = true;
|
|
cricket::Candidate last_candidate_gathered_;
|
|
cricket::IceCandidateErrorEvent error_event_;
|
|
|
|
// Store references to the video sources we've created, so that we can stop
|
|
// them, if required.
|
|
std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>>
|
|
video_track_sources_;
|
|
// |local_video_renderer_| attached to the first created local video track.
|
|
std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
|
|
|
|
SdpSemantics sdp_semantics_;
|
|
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
|
|
std::function<void(cricket::SessionDescription*)> received_sdp_munger_;
|
|
std::function<void(cricket::SessionDescription*)> generated_sdp_munger_;
|
|
std::function<void()> remote_offer_handler_;
|
|
rtc::MockAsyncResolver* remote_async_resolver_ = nullptr;
|
|
rtc::scoped_refptr<DataChannelInterface> data_channel_;
|
|
std::unique_ptr<MockDataChannelObserver> data_observer_;
|
|
|
|
std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;
|
|
|
|
std::vector<PeerConnectionInterface::IceConnectionState>
|
|
ice_connection_state_history_;
|
|
std::vector<PeerConnectionInterface::IceConnectionState>
|
|
standardized_ice_connection_state_history_;
|
|
std::vector<PeerConnectionInterface::PeerConnectionState>
|
|
peer_connection_state_history_;
|
|
std::vector<PeerConnectionInterface::IceGatheringState>
|
|
ice_gathering_state_history_;
|
|
std::vector<cricket::CandidatePairChangeEvent>
|
|
ice_candidate_pair_change_history_;
|
|
std::vector<PeerConnectionInterface::SignalingState>
|
|
peer_connection_signaling_state_history_;
|
|
webrtc::FakeRtcEventLogFactory* event_log_factory_;
|
|
|
|
rtc::AsyncInvoker invoker_;
|
|
|
|
friend class PeerConnectionIntegrationBaseTest;
|
|
};
|
|
|
|
class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
|
|
public:
|
|
virtual ~MockRtcEventLogOutput() = default;
|
|
MOCK_METHOD(bool, IsActive, (), (const, override));
|
|
MOCK_METHOD(bool, Write, (const std::string&), (override));
|
|
};
|
|
|
|
// This helper object is used for both specifying how many audio/video frames
|
|
// are expected to be received for a caller/callee. It provides helper functions
|
|
// to specify these expectations. The object initially starts in a state of no
|
|
// expectations.
|
|
class MediaExpectations {
|
|
public:
|
|
enum ExpectFrames {
|
|
kExpectSomeFrames,
|
|
kExpectNoFrames,
|
|
kNoExpectation,
|
|
};
|
|
|
|
void ExpectBidirectionalAudioAndVideo() {
|
|
ExpectBidirectionalAudio();
|
|
ExpectBidirectionalVideo();
|
|
}
|
|
|
|
void ExpectBidirectionalAudio() {
|
|
CallerExpectsSomeAudio();
|
|
CalleeExpectsSomeAudio();
|
|
}
|
|
|
|
void ExpectNoAudio() {
|
|
CallerExpectsNoAudio();
|
|
CalleeExpectsNoAudio();
|
|
}
|
|
|
|
void ExpectBidirectionalVideo() {
|
|
CallerExpectsSomeVideo();
|
|
CalleeExpectsSomeVideo();
|
|
}
|
|
|
|
void ExpectNoVideo() {
|
|
CallerExpectsNoVideo();
|
|
CalleeExpectsNoVideo();
|
|
}
|
|
|
|
void CallerExpectsSomeAudioAndVideo() {
|
|
CallerExpectsSomeAudio();
|
|
CallerExpectsSomeVideo();
|
|
}
|
|
|
|
void CalleeExpectsSomeAudioAndVideo() {
|
|
CalleeExpectsSomeAudio();
|
|
CalleeExpectsSomeVideo();
|
|
}
|
|
|
|
// Caller's audio functions.
|
|
void CallerExpectsSomeAudio(
|
|
int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
|
|
caller_audio_expectation_ = kExpectSomeFrames;
|
|
caller_audio_frames_expected_ = expected_audio_frames;
|
|
}
|
|
|
|
void CallerExpectsNoAudio() {
|
|
caller_audio_expectation_ = kExpectNoFrames;
|
|
caller_audio_frames_expected_ = 0;
|
|
}
|
|
|
|
// Caller's video functions.
|
|
void CallerExpectsSomeVideo(
|
|
int expected_video_frames = kDefaultExpectedVideoFrameCount) {
|
|
caller_video_expectation_ = kExpectSomeFrames;
|
|
caller_video_frames_expected_ = expected_video_frames;
|
|
}
|
|
|
|
void CallerExpectsNoVideo() {
|
|
caller_video_expectation_ = kExpectNoFrames;
|
|
caller_video_frames_expected_ = 0;
|
|
}
|
|
|
|
// Callee's audio functions.
|
|
void CalleeExpectsSomeAudio(
|
|
int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
|
|
callee_audio_expectation_ = kExpectSomeFrames;
|
|
callee_audio_frames_expected_ = expected_audio_frames;
|
|
}
|
|
|
|
void CalleeExpectsNoAudio() {
|
|
callee_audio_expectation_ = kExpectNoFrames;
|
|
callee_audio_frames_expected_ = 0;
|
|
}
|
|
|
|
// Callee's video functions.
|
|
void CalleeExpectsSomeVideo(
|
|
int expected_video_frames = kDefaultExpectedVideoFrameCount) {
|
|
callee_video_expectation_ = kExpectSomeFrames;
|
|
callee_video_frames_expected_ = expected_video_frames;
|
|
}
|
|
|
|
void CalleeExpectsNoVideo() {
|
|
callee_video_expectation_ = kExpectNoFrames;
|
|
callee_video_frames_expected_ = 0;
|
|
}
|
|
|
|
ExpectFrames caller_audio_expectation_ = kNoExpectation;
|
|
ExpectFrames caller_video_expectation_ = kNoExpectation;
|
|
ExpectFrames callee_audio_expectation_ = kNoExpectation;
|
|
ExpectFrames callee_video_expectation_ = kNoExpectation;
|
|
int caller_audio_frames_expected_ = 0;
|
|
int caller_video_frames_expected_ = 0;
|
|
int callee_audio_frames_expected_ = 0;
|
|
int callee_video_frames_expected_ = 0;
|
|
};
|
|
|
|
class MockIceTransport : public webrtc::IceTransportInterface {
|
|
public:
|
|
MockIceTransport(const std::string& name, int component)
|
|
: internal_(std::make_unique<cricket::FakeIceTransport>(
|
|
name,
|
|
component,
|
|
nullptr /* network_thread */)) {}
|
|
~MockIceTransport() = default;
|
|
cricket::IceTransportInternal* internal() { return internal_.get(); }
|
|
|
|
private:
|
|
std::unique_ptr<cricket::FakeIceTransport> internal_;
|
|
};
|
|
|
|
class MockIceTransportFactory : public IceTransportFactory {
|
|
public:
|
|
~MockIceTransportFactory() override = default;
|
|
rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
|
|
const std::string& transport_name,
|
|
int component,
|
|
IceTransportInit init) {
|
|
RecordIceTransportCreated();
|
|
return new rtc::RefCountedObject<MockIceTransport>(transport_name,
|
|
component);
|
|
}
|
|
MOCK_METHOD(void, RecordIceTransportCreated, ());
|
|
};
|
|
|
|
// Tests two PeerConnections connecting to each other end-to-end, using a
|
|
// virtual network, fake A/V capture and fake encoder/decoders. The
|
|
// PeerConnections share the threads/socket servers, but use separate versions
|
|
// of everything else (including "PeerConnectionFactory"s).
|
|
class PeerConnectionIntegrationBaseTest : public ::testing::Test {
|
|
public:
|
|
explicit PeerConnectionIntegrationBaseTest(SdpSemantics sdp_semantics)
|
|
: sdp_semantics_(sdp_semantics),
|
|
ss_(new rtc::VirtualSocketServer()),
|
|
fss_(new rtc::FirewallSocketServer(ss_.get())),
|
|
network_thread_(new rtc::Thread(fss_.get())),
|
|
worker_thread_(rtc::Thread::Create()) {
|
|
network_thread_->SetName("PCNetworkThread", this);
|
|
worker_thread_->SetName("PCWorkerThread", this);
|
|
RTC_CHECK(network_thread_->Start());
|
|
RTC_CHECK(worker_thread_->Start());
|
|
webrtc::metrics::Reset();
|
|
}
|
|
|
|
~PeerConnectionIntegrationBaseTest() {
|
|
// The PeerConnections should deleted before the TurnCustomizers.
|
|
// A TurnPort is created with a raw pointer to a TurnCustomizer. The
|
|
// TurnPort has the same lifetime as the PeerConnection, so it's expected
|
|
// that the TurnCustomizer outlives the life of the PeerConnection or else
|
|
// when Send() is called it will hit a seg fault.
|
|
if (caller_) {
|
|
caller_->set_signaling_message_receiver(nullptr);
|
|
delete SetCallerPcWrapperAndReturnCurrent(nullptr);
|
|
}
|
|
if (callee_) {
|
|
callee_->set_signaling_message_receiver(nullptr);
|
|
delete SetCalleePcWrapperAndReturnCurrent(nullptr);
|
|
}
|
|
|
|
// If turn servers were created for the test they need to be destroyed on
|
|
// the network thread.
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
turn_servers_.clear();
|
|
turn_customizers_.clear();
|
|
});
|
|
}
|
|
|
|
bool SignalingStateStable() {
|
|
return caller_->SignalingStateStable() && callee_->SignalingStateStable();
|
|
}
|
|
|
|
bool DtlsConnected() {
|
|
// TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
|
|
// are connected. This is an important distinction. Once we have separate
|
|
// ICE and DTLS state, this check needs to use the DTLS state.
|
|
return (callee()->ice_connection_state() ==
|
|
webrtc::PeerConnectionInterface::kIceConnectionConnected ||
|
|
callee()->ice_connection_state() ==
|
|
webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
|
|
(caller()->ice_connection_state() ==
|
|
webrtc::PeerConnectionInterface::kIceConnectionConnected ||
|
|
caller()->ice_connection_state() ==
|
|
webrtc::PeerConnectionInterface::kIceConnectionCompleted);
|
|
}
|
|
|
|
// When |event_log_factory| is null, the default implementation of the event
|
|
// log factory will be used.
|
|
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWrapper(
|
|
const std::string& debug_name,
|
|
const PeerConnectionFactory::Options* options,
|
|
const RTCConfiguration* config,
|
|
webrtc::PeerConnectionDependencies dependencies,
|
|
std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
|
|
bool reset_encoder_factory,
|
|
bool reset_decoder_factory) {
|
|
RTCConfiguration modified_config;
|
|
if (config) {
|
|
modified_config = *config;
|
|
}
|
|
modified_config.sdp_semantics = sdp_semantics_;
|
|
if (!dependencies.cert_generator) {
|
|
dependencies.cert_generator =
|
|
std::make_unique<FakeRTCCertificateGenerator>();
|
|
}
|
|
std::unique_ptr<PeerConnectionWrapper> client(
|
|
new PeerConnectionWrapper(debug_name));
|
|
|
|
if (!client->Init(options, &modified_config, std::move(dependencies),
|
|
network_thread_.get(), worker_thread_.get(),
|
|
std::move(event_log_factory), reset_encoder_factory,
|
|
reset_decoder_factory)) {
|
|
return nullptr;
|
|
}
|
|
return client;
|
|
}
|
|
|
|
std::unique_ptr<PeerConnectionWrapper>
|
|
CreatePeerConnectionWrapperWithFakeRtcEventLog(
|
|
const std::string& debug_name,
|
|
const PeerConnectionFactory::Options* options,
|
|
const RTCConfiguration* config,
|
|
webrtc::PeerConnectionDependencies dependencies) {
|
|
std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory(
|
|
new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current()));
|
|
return CreatePeerConnectionWrapper(debug_name, options, config,
|
|
std::move(dependencies),
|
|
std::move(event_log_factory),
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
}
|
|
|
|
bool CreatePeerConnectionWrappers() {
|
|
return CreatePeerConnectionWrappersWithConfig(
|
|
PeerConnectionInterface::RTCConfiguration(),
|
|
PeerConnectionInterface::RTCConfiguration());
|
|
}
|
|
|
|
bool CreatePeerConnectionWrappersWithSdpSemantics(
|
|
SdpSemantics caller_semantics,
|
|
SdpSemantics callee_semantics) {
|
|
// Can't specify the sdp_semantics in the passed-in configuration since it
|
|
// will be overwritten by CreatePeerConnectionWrapper with whatever is
|
|
// stored in sdp_semantics_. So get around this by modifying the instance
|
|
// variable before calling CreatePeerConnectionWrapper for the caller and
|
|
// callee PeerConnections.
|
|
SdpSemantics original_semantics = sdp_semantics_;
|
|
sdp_semantics_ = caller_semantics;
|
|
caller_ = CreatePeerConnectionWrapper(
|
|
"Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
|
|
nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
sdp_semantics_ = callee_semantics;
|
|
callee_ = CreatePeerConnectionWrapper(
|
|
"Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
|
|
nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
sdp_semantics_ = original_semantics;
|
|
return caller_ && callee_;
|
|
}
|
|
|
|
bool CreatePeerConnectionWrappersWithConfig(
|
|
const PeerConnectionInterface::RTCConfiguration& caller_config,
|
|
const PeerConnectionInterface::RTCConfiguration& callee_config) {
|
|
caller_ = CreatePeerConnectionWrapper(
|
|
"Caller", nullptr, &caller_config,
|
|
webrtc::PeerConnectionDependencies(nullptr), nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
callee_ = CreatePeerConnectionWrapper(
|
|
"Callee", nullptr, &callee_config,
|
|
webrtc::PeerConnectionDependencies(nullptr), nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
return caller_ && callee_;
|
|
}
|
|
|
|
bool CreatePeerConnectionWrappersWithConfigAndDeps(
|
|
const PeerConnectionInterface::RTCConfiguration& caller_config,
|
|
webrtc::PeerConnectionDependencies caller_dependencies,
|
|
const PeerConnectionInterface::RTCConfiguration& callee_config,
|
|
webrtc::PeerConnectionDependencies callee_dependencies) {
|
|
caller_ =
|
|
CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
|
|
std::move(caller_dependencies), nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
callee_ =
|
|
CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
|
|
std::move(callee_dependencies), nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
return caller_ && callee_;
|
|
}
|
|
|
|
bool CreatePeerConnectionWrappersWithOptions(
|
|
const PeerConnectionFactory::Options& caller_options,
|
|
const PeerConnectionFactory::Options& callee_options) {
|
|
caller_ = CreatePeerConnectionWrapper(
|
|
"Caller", &caller_options, nullptr,
|
|
webrtc::PeerConnectionDependencies(nullptr), nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
callee_ = CreatePeerConnectionWrapper(
|
|
"Callee", &callee_options, nullptr,
|
|
webrtc::PeerConnectionDependencies(nullptr), nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
return caller_ && callee_;
|
|
}
|
|
|
|
bool CreatePeerConnectionWrappersWithFakeRtcEventLog() {
|
|
PeerConnectionInterface::RTCConfiguration default_config;
|
|
caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
|
|
"Caller", nullptr, &default_config,
|
|
webrtc::PeerConnectionDependencies(nullptr));
|
|
callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
|
|
"Callee", nullptr, &default_config,
|
|
webrtc::PeerConnectionDependencies(nullptr));
|
|
return caller_ && callee_;
|
|
}
|
|
|
|
std::unique_ptr<PeerConnectionWrapper>
|
|
CreatePeerConnectionWrapperWithAlternateKey() {
|
|
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
|
|
new FakeRTCCertificateGenerator());
|
|
cert_generator->use_alternate_key();
|
|
|
|
webrtc::PeerConnectionDependencies dependencies(nullptr);
|
|
dependencies.cert_generator = std::move(cert_generator);
|
|
return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
|
|
std::move(dependencies), nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
}
|
|
|
|
bool CreateOneDirectionalPeerConnectionWrappers(bool caller_to_callee) {
|
|
caller_ = CreatePeerConnectionWrapper(
|
|
"Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
|
|
nullptr,
|
|
/*reset_encoder_factory=*/!caller_to_callee,
|
|
/*reset_decoder_factory=*/caller_to_callee);
|
|
callee_ = CreatePeerConnectionWrapper(
|
|
"Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
|
|
nullptr,
|
|
/*reset_encoder_factory=*/caller_to_callee,
|
|
/*reset_decoder_factory=*/!caller_to_callee);
|
|
return caller_ && callee_;
|
|
}
|
|
|
|
cricket::TestTurnServer* CreateTurnServer(
|
|
rtc::SocketAddress internal_address,
|
|
rtc::SocketAddress external_address,
|
|
cricket::ProtocolType type = cricket::ProtocolType::PROTO_UDP,
|
|
const std::string& common_name = "test turn server") {
|
|
rtc::Thread* thread = network_thread();
|
|
std::unique_ptr<cricket::TestTurnServer> turn_server =
|
|
network_thread()->Invoke<std::unique_ptr<cricket::TestTurnServer>>(
|
|
RTC_FROM_HERE,
|
|
[thread, internal_address, external_address, type, common_name] {
|
|
return std::make_unique<cricket::TestTurnServer>(
|
|
thread, internal_address, external_address, type,
|
|
/*ignore_bad_certs=*/true, common_name);
|
|
});
|
|
turn_servers_.push_back(std::move(turn_server));
|
|
// Interactions with the turn server should be done on the network thread.
|
|
return turn_servers_.back().get();
|
|
}
|
|
|
|
cricket::TestTurnCustomizer* CreateTurnCustomizer() {
|
|
std::unique_ptr<cricket::TestTurnCustomizer> turn_customizer =
|
|
network_thread()->Invoke<std::unique_ptr<cricket::TestTurnCustomizer>>(
|
|
RTC_FROM_HERE,
|
|
[] { return std::make_unique<cricket::TestTurnCustomizer>(); });
|
|
turn_customizers_.push_back(std::move(turn_customizer));
|
|
// Interactions with the turn customizer should be done on the network
|
|
// thread.
|
|
return turn_customizers_.back().get();
|
|
}
|
|
|
|
// Checks that the function counters for a TestTurnCustomizer are greater than
|
|
// 0.
|
|
void ExpectTurnCustomizerCountersIncremented(
|
|
cricket::TestTurnCustomizer* turn_customizer) {
|
|
unsigned int allow_channel_data_counter =
|
|
network_thread()->Invoke<unsigned int>(
|
|
RTC_FROM_HERE, [turn_customizer] {
|
|
return turn_customizer->allow_channel_data_cnt_;
|
|
});
|
|
EXPECT_GT(allow_channel_data_counter, 0u);
|
|
unsigned int modify_counter = network_thread()->Invoke<unsigned int>(
|
|
RTC_FROM_HERE,
|
|
[turn_customizer] { return turn_customizer->modify_cnt_; });
|
|
EXPECT_GT(modify_counter, 0u);
|
|
}
|
|
|
|
// Once called, SDP blobs and ICE candidates will be automatically signaled
|
|
// between PeerConnections.
|
|
void ConnectFakeSignaling() {
|
|
caller_->set_signaling_message_receiver(callee_.get());
|
|
callee_->set_signaling_message_receiver(caller_.get());
|
|
}
|
|
|
|
// Once called, SDP blobs will be automatically signaled between
|
|
// PeerConnections. Note that ICE candidates will not be signaled unless they
|
|
// are in the exchanged SDP blobs.
|
|
void ConnectFakeSignalingForSdpOnly() {
|
|
ConnectFakeSignaling();
|
|
SetSignalIceCandidates(false);
|
|
}
|
|
|
|
void SetSignalingDelayMs(int delay_ms) {
|
|
caller_->set_signaling_delay_ms(delay_ms);
|
|
callee_->set_signaling_delay_ms(delay_ms);
|
|
}
|
|
|
|
void SetSignalIceCandidates(bool signal) {
|
|
caller_->set_signal_ice_candidates(signal);
|
|
callee_->set_signal_ice_candidates(signal);
|
|
}
|
|
|
|
// Messages may get lost on the unreliable DataChannel, so we send multiple
|
|
// times to avoid test flakiness.
|
|
void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc,
|
|
const std::string& data,
|
|
int retries) {
|
|
for (int i = 0; i < retries; ++i) {
|
|
dc->Send(DataBuffer(data));
|
|
}
|
|
}
|
|
|
|
rtc::Thread* network_thread() { return network_thread_.get(); }
|
|
|
|
rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
|
|
|
|
PeerConnectionWrapper* caller() { return caller_.get(); }
|
|
|
|
// Set the |caller_| to the |wrapper| passed in and return the
|
|
// original |caller_|.
|
|
PeerConnectionWrapper* SetCallerPcWrapperAndReturnCurrent(
|
|
PeerConnectionWrapper* wrapper) {
|
|
PeerConnectionWrapper* old = caller_.release();
|
|
caller_.reset(wrapper);
|
|
return old;
|
|
}
|
|
|
|
PeerConnectionWrapper* callee() { return callee_.get(); }
|
|
|
|
// Set the |callee_| to the |wrapper| passed in and return the
|
|
// original |callee_|.
|
|
PeerConnectionWrapper* SetCalleePcWrapperAndReturnCurrent(
|
|
PeerConnectionWrapper* wrapper) {
|
|
PeerConnectionWrapper* old = callee_.release();
|
|
callee_.reset(wrapper);
|
|
return old;
|
|
}
|
|
|
|
void SetPortAllocatorFlags(uint32_t caller_flags, uint32_t callee_flags) {
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::set_flags,
|
|
caller()->port_allocator(), caller_flags));
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::set_flags,
|
|
callee()->port_allocator(), callee_flags));
|
|
}
|
|
|
|
rtc::FirewallSocketServer* firewall() const { return fss_.get(); }
|
|
|
|
// Expects the provided number of new frames to be received within
|
|
// kMaxWaitForFramesMs. The new expected frames are specified in
|
|
// |media_expectations|. Returns false if any of the expectations were
|
|
// not met.
|
|
bool ExpectNewFrames(const MediaExpectations& media_expectations) {
|
|
// Make sure there are no bogus tracks confusing the issue.
|
|
caller()->RemoveUnusedVideoRenderers();
|
|
callee()->RemoveUnusedVideoRenderers();
|
|
// First initialize the expected frame counts based upon the current
|
|
// frame count.
|
|
int total_caller_audio_frames_expected = caller()->audio_frames_received();
|
|
if (media_expectations.caller_audio_expectation_ ==
|
|
MediaExpectations::kExpectSomeFrames) {
|
|
total_caller_audio_frames_expected +=
|
|
media_expectations.caller_audio_frames_expected_;
|
|
}
|
|
int total_caller_video_frames_expected =
|
|
caller()->min_video_frames_received_per_track();
|
|
if (media_expectations.caller_video_expectation_ ==
|
|
MediaExpectations::kExpectSomeFrames) {
|
|
total_caller_video_frames_expected +=
|
|
media_expectations.caller_video_frames_expected_;
|
|
}
|
|
int total_callee_audio_frames_expected = callee()->audio_frames_received();
|
|
if (media_expectations.callee_audio_expectation_ ==
|
|
MediaExpectations::kExpectSomeFrames) {
|
|
total_callee_audio_frames_expected +=
|
|
media_expectations.callee_audio_frames_expected_;
|
|
}
|
|
int total_callee_video_frames_expected =
|
|
callee()->min_video_frames_received_per_track();
|
|
if (media_expectations.callee_video_expectation_ ==
|
|
MediaExpectations::kExpectSomeFrames) {
|
|
total_callee_video_frames_expected +=
|
|
media_expectations.callee_video_frames_expected_;
|
|
}
|
|
|
|
// Wait for the expected frames.
|
|
EXPECT_TRUE_WAIT(caller()->audio_frames_received() >=
|
|
total_caller_audio_frames_expected &&
|
|
caller()->min_video_frames_received_per_track() >=
|
|
total_caller_video_frames_expected &&
|
|
callee()->audio_frames_received() >=
|
|
total_callee_audio_frames_expected &&
|
|
callee()->min_video_frames_received_per_track() >=
|
|
total_callee_video_frames_expected,
|
|
kMaxWaitForFramesMs);
|
|
bool expectations_correct =
|
|
caller()->audio_frames_received() >=
|
|
total_caller_audio_frames_expected &&
|
|
caller()->min_video_frames_received_per_track() >=
|
|
total_caller_video_frames_expected &&
|
|
callee()->audio_frames_received() >=
|
|
total_callee_audio_frames_expected &&
|
|
callee()->min_video_frames_received_per_track() >=
|
|
total_callee_video_frames_expected;
|
|
|
|
// After the combined wait, print out a more detailed message upon
|
|
// failure.
|
|
EXPECT_GE(caller()->audio_frames_received(),
|
|
total_caller_audio_frames_expected);
|
|
EXPECT_GE(caller()->min_video_frames_received_per_track(),
|
|
total_caller_video_frames_expected);
|
|
EXPECT_GE(callee()->audio_frames_received(),
|
|
total_callee_audio_frames_expected);
|
|
EXPECT_GE(callee()->min_video_frames_received_per_track(),
|
|
total_callee_video_frames_expected);
|
|
|
|
// We want to make sure nothing unexpected was received.
|
|
if (media_expectations.caller_audio_expectation_ ==
|
|
MediaExpectations::kExpectNoFrames) {
|
|
EXPECT_EQ(caller()->audio_frames_received(),
|
|
total_caller_audio_frames_expected);
|
|
if (caller()->audio_frames_received() !=
|
|
total_caller_audio_frames_expected) {
|
|
expectations_correct = false;
|
|
}
|
|
}
|
|
if (media_expectations.caller_video_expectation_ ==
|
|
MediaExpectations::kExpectNoFrames) {
|
|
EXPECT_EQ(caller()->min_video_frames_received_per_track(),
|
|
total_caller_video_frames_expected);
|
|
if (caller()->min_video_frames_received_per_track() !=
|
|
total_caller_video_frames_expected) {
|
|
expectations_correct = false;
|
|
}
|
|
}
|
|
if (media_expectations.callee_audio_expectation_ ==
|
|
MediaExpectations::kExpectNoFrames) {
|
|
EXPECT_EQ(callee()->audio_frames_received(),
|
|
total_callee_audio_frames_expected);
|
|
if (callee()->audio_frames_received() !=
|
|
total_callee_audio_frames_expected) {
|
|
expectations_correct = false;
|
|
}
|
|
}
|
|
if (media_expectations.callee_video_expectation_ ==
|
|
MediaExpectations::kExpectNoFrames) {
|
|
EXPECT_EQ(callee()->min_video_frames_received_per_track(),
|
|
total_callee_video_frames_expected);
|
|
if (callee()->min_video_frames_received_per_track() !=
|
|
total_callee_video_frames_expected) {
|
|
expectations_correct = false;
|
|
}
|
|
}
|
|
return expectations_correct;
|
|
}
|
|
|
|
void ClosePeerConnections() {
|
|
caller()->pc()->Close();
|
|
callee()->pc()->Close();
|
|
}
|
|
|
|
void TestNegotiatedCipherSuite(
|
|
const PeerConnectionFactory::Options& caller_options,
|
|
const PeerConnectionFactory::Options& callee_options,
|
|
int expected_cipher_suite) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
|
|
callee_options));
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
|
|
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
|
|
// TODO(bugs.webrtc.org/9456): Fix it.
|
|
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
|
|
expected_cipher_suite));
|
|
}
|
|
|
|
void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
|
|
bool remote_gcm_enabled,
|
|
bool aes_ctr_enabled,
|
|
int expected_cipher_suite) {
|
|
PeerConnectionFactory::Options caller_options;
|
|
caller_options.crypto_options.srtp.enable_gcm_crypto_suites =
|
|
local_gcm_enabled;
|
|
caller_options.crypto_options.srtp.enable_aes128_sha1_80_crypto_cipher =
|
|
aes_ctr_enabled;
|
|
PeerConnectionFactory::Options callee_options;
|
|
callee_options.crypto_options.srtp.enable_gcm_crypto_suites =
|
|
remote_gcm_enabled;
|
|
callee_options.crypto_options.srtp.enable_aes128_sha1_80_crypto_cipher =
|
|
aes_ctr_enabled;
|
|
TestNegotiatedCipherSuite(caller_options, callee_options,
|
|
expected_cipher_suite);
|
|
}
|
|
|
|
protected:
|
|
SdpSemantics sdp_semantics_;
|
|
|
|
private:
|
|
// |ss_| is used by |network_thread_| so it must be destroyed later.
|
|
std::unique_ptr<rtc::VirtualSocketServer> ss_;
|
|
std::unique_ptr<rtc::FirewallSocketServer> fss_;
|
|
// |network_thread_| and |worker_thread_| are used by both
|
|
// |caller_| and |callee_| so they must be destroyed
|
|
// later.
|
|
std::unique_ptr<rtc::Thread> network_thread_;
|
|
std::unique_ptr<rtc::Thread> worker_thread_;
|
|
// The turn servers and turn customizers should be accessed & deleted on the
|
|
// network thread to avoid a race with the socket read/write that occurs
|
|
// on the network thread.
|
|
std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
|
|
std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_;
|
|
std::unique_ptr<PeerConnectionWrapper> caller_;
|
|
std::unique_ptr<PeerConnectionWrapper> callee_;
|
|
};
|
|
|
|
class PeerConnectionIntegrationTest
|
|
: public PeerConnectionIntegrationBaseTest,
|
|
public ::testing::WithParamInterface<SdpSemantics> {
|
|
protected:
|
|
PeerConnectionIntegrationTest()
|
|
: PeerConnectionIntegrationBaseTest(GetParam()) {}
|
|
};
|
|
|
|
// Fake clock must be set before threads are started to prevent race on
|
|
// Set/GetClockForTesting().
|
|
// To achieve that, multiple inheritance is used as a mixin pattern
|
|
// where order of construction is finely controlled.
|
|
// This also ensures peerconnection is closed before switching back to non-fake
|
|
// clock, avoiding other races and DCHECK failures such as in rtp_sender.cc.
|
|
class FakeClockForTest : public rtc::ScopedFakeClock {
|
|
protected:
|
|
FakeClockForTest() {
|
|
// 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.
|
|
AdvanceTime(webrtc::TimeDelta::Seconds(1));
|
|
}
|
|
|
|
// Explicit handle.
|
|
ScopedFakeClock& FakeClock() { return *this; }
|
|
};
|
|
|
|
// Ensure FakeClockForTest is constructed first (see class for rationale).
|
|
class PeerConnectionIntegrationTestWithFakeClock
|
|
: public FakeClockForTest,
|
|
public PeerConnectionIntegrationTest {};
|
|
|
|
class PeerConnectionIntegrationTestPlanB
|
|
: public PeerConnectionIntegrationBaseTest {
|
|
protected:
|
|
PeerConnectionIntegrationTestPlanB()
|
|
: PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB) {}
|
|
};
|
|
|
|
class PeerConnectionIntegrationTestUnifiedPlan
|
|
: public PeerConnectionIntegrationBaseTest {
|
|
protected:
|
|
PeerConnectionIntegrationTestUnifiedPlan()
|
|
: PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
|
|
};
|
|
|
|
// Test the OnFirstPacketReceived callback from audio/video RtpReceivers. This
|
|
// includes testing that the callback is invoked if an observer is connected
|
|
// after the first packet has already been received.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
RtpReceiverObserverOnFirstPacketReceived) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
// Start offer/answer exchange and wait for it to complete.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Should be one receiver each for audio/video.
|
|
EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
|
|
EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
|
|
// Wait for all "first packet received" callbacks to be fired.
|
|
EXPECT_TRUE_WAIT(
|
|
absl::c_all_of(caller()->rtp_receiver_observers(),
|
|
[](const std::unique_ptr<MockRtpReceiverObserver>& o) {
|
|
return o->first_packet_received();
|
|
}),
|
|
kMaxWaitForFramesMs);
|
|
EXPECT_TRUE_WAIT(
|
|
absl::c_all_of(callee()->rtp_receiver_observers(),
|
|
[](const std::unique_ptr<MockRtpReceiverObserver>& o) {
|
|
return o->first_packet_received();
|
|
}),
|
|
kMaxWaitForFramesMs);
|
|
// If new observers are set after the first packet was already received, the
|
|
// callback should still be invoked.
|
|
caller()->ResetRtpReceiverObservers();
|
|
callee()->ResetRtpReceiverObservers();
|
|
EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
|
|
EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
|
|
EXPECT_TRUE(
|
|
absl::c_all_of(caller()->rtp_receiver_observers(),
|
|
[](const std::unique_ptr<MockRtpReceiverObserver>& o) {
|
|
return o->first_packet_received();
|
|
}));
|
|
EXPECT_TRUE(
|
|
absl::c_all_of(callee()->rtp_receiver_observers(),
|
|
[](const std::unique_ptr<MockRtpReceiverObserver>& o) {
|
|
return o->first_packet_received();
|
|
}));
|
|
}
|
|
|
|
class DummyDtmfObserver : public DtmfSenderObserverInterface {
|
|
public:
|
|
DummyDtmfObserver() : completed_(false) {}
|
|
|
|
// Implements DtmfSenderObserverInterface.
|
|
void OnToneChange(const std::string& tone) override {
|
|
tones_.push_back(tone);
|
|
if (tone.empty()) {
|
|
completed_ = true;
|
|
}
|
|
}
|
|
|
|
const std::vector<std::string>& tones() const { return tones_; }
|
|
bool completed() const { return completed_; }
|
|
|
|
private:
|
|
bool completed_;
|
|
std::vector<std::string> tones_;
|
|
};
|
|
|
|
// Assumes |sender| already has an audio track added and the offer/answer
|
|
// exchange is done.
|
|
void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender,
|
|
PeerConnectionWrapper* receiver) {
|
|
// We should be able to get a DTMF sender from the local sender.
|
|
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender =
|
|
sender->pc()->GetSenders().at(0)->GetDtmfSender();
|
|
ASSERT_TRUE(dtmf_sender);
|
|
DummyDtmfObserver observer;
|
|
dtmf_sender->RegisterObserver(&observer);
|
|
|
|
// Test the DtmfSender object just created.
|
|
EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
|
|
EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));
|
|
|
|
EXPECT_TRUE_WAIT(observer.completed(), kDefaultTimeout);
|
|
std::vector<std::string> tones = {"1", "a", ""};
|
|
EXPECT_EQ(tones, observer.tones());
|
|
dtmf_sender->UnregisterObserver();
|
|
// TODO(deadbeef): Verify the tones were actually received end-to-end.
|
|
}
|
|
|
|
// Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each
|
|
// direction).
|
|
TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Only need audio for DTMF.
|
|
caller()->AddAudioTrack();
|
|
callee()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// DTLS must finish before the DTMF sender can be used reliably.
|
|
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
|
|
TestDtmfFromSenderToReceiver(caller(), callee());
|
|
TestDtmfFromSenderToReceiver(callee(), caller());
|
|
}
|
|
|
|
// Basic end-to-end test, verifying media can be encoded/transmitted/decoded
|
|
// between two connections, using DTLS-SRTP.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
// Do normal offer/answer and wait for some frames to be received in each
|
|
// direction.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
EXPECT_METRIC_LE(
|
|
2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
|
|
webrtc::kEnumCounterKeyProtocolDtls));
|
|
EXPECT_METRIC_EQ(
|
|
0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
|
|
webrtc::kEnumCounterKeyProtocolSdes));
|
|
}
|
|
|
|
// Uses SDES instead of DTLS for key agreement.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
|
|
PeerConnectionInterface::RTCConfiguration sdes_config;
|
|
sdes_config.enable_dtls_srtp.emplace(false);
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config));
|
|
ConnectFakeSignaling();
|
|
|
|
// Do normal offer/answer and wait for some frames to be received in each
|
|
// direction.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
EXPECT_METRIC_LE(
|
|
2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
|
|
webrtc::kEnumCounterKeyProtocolSdes));
|
|
EXPECT_METRIC_EQ(
|
|
0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
|
|
webrtc::kEnumCounterKeyProtocolDtls));
|
|
}
|
|
|
|
// Basic end-to-end test specifying the |enable_encrypted_rtp_header_extensions|
|
|
// option to offer encrypted versions of all header extensions alongside the
|
|
// unencrypted versions.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
EndToEndCallWithEncryptedRtpHeaderExtensions) {
|
|
CryptoOptions crypto_options;
|
|
crypto_options.srtp.enable_encrypted_rtp_header_extensions = true;
|
|
PeerConnectionInterface::RTCConfiguration config;
|
|
config.crypto_options = crypto_options;
|
|
// Note: This allows offering >14 RTP header extensions.
|
|
config.offer_extmap_allow_mixed = true;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
|
|
ConnectFakeSignaling();
|
|
|
|
// Do normal offer/answer and wait for some frames to be received in each
|
|
// direction.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
|
|
// certificate once the DTLS handshake has finished.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
GetRemoteAudioSSLCertificateReturnsExchangedCertificate) {
|
|
auto GetRemoteAudioSSLCertificate = [](PeerConnectionWrapper* wrapper) {
|
|
auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
|
|
auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
|
|
return pc->GetRemoteAudioSSLCertificate();
|
|
};
|
|
auto GetRemoteAudioSSLCertChain = [](PeerConnectionWrapper* wrapper) {
|
|
auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
|
|
auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
|
|
return pc->GetRemoteAudioSSLCertChain();
|
|
};
|
|
|
|
auto caller_cert = rtc::RTCCertificate::FromPEM(kRsaPems[0]);
|
|
auto callee_cert = rtc::RTCCertificate::FromPEM(kRsaPems[1]);
|
|
|
|
// Configure each side with a known certificate so they can be compared later.
|
|
PeerConnectionInterface::RTCConfiguration caller_config;
|
|
caller_config.enable_dtls_srtp.emplace(true);
|
|
caller_config.certificates.push_back(caller_cert);
|
|
PeerConnectionInterface::RTCConfiguration callee_config;
|
|
callee_config.enable_dtls_srtp.emplace(true);
|
|
callee_config.certificates.push_back(callee_cert);
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));
|
|
ConnectFakeSignaling();
|
|
|
|
// When first initialized, there should not be a remote SSL certificate (and
|
|
// calling this method should not crash).
|
|
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller()));
|
|
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee()));
|
|
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(caller()));
|
|
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(callee()));
|
|
|
|
caller()->AddAudioTrack();
|
|
callee()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
|
|
|
|
// Once DTLS has been connected, each side should return the other's SSL
|
|
// certificate when calling GetRemoteAudioSSLCertificate.
|
|
|
|
auto caller_remote_cert = GetRemoteAudioSSLCertificate(caller());
|
|
ASSERT_TRUE(caller_remote_cert);
|
|
EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
|
|
caller_remote_cert->ToPEMString());
|
|
|
|
auto callee_remote_cert = GetRemoteAudioSSLCertificate(callee());
|
|
ASSERT_TRUE(callee_remote_cert);
|
|
EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
|
|
callee_remote_cert->ToPEMString());
|
|
|
|
auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller());
|
|
ASSERT_TRUE(caller_remote_cert_chain);
|
|
ASSERT_EQ(1U, caller_remote_cert_chain->GetSize());
|
|
auto remote_cert = &caller_remote_cert_chain->Get(0);
|
|
EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
|
|
remote_cert->ToPEMString());
|
|
|
|
auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee());
|
|
ASSERT_TRUE(callee_remote_cert_chain);
|
|
ASSERT_EQ(1U, callee_remote_cert_chain->GetSize());
|
|
remote_cert = &callee_remote_cert_chain->Get(0);
|
|
EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
|
|
remote_cert->ToPEMString());
|
|
}
|
|
|
|
// This test sets up a call between two parties with a source resolution of
|
|
// 1280x720 and verifies that a 16:9 aspect ratio is received.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
Send1280By720ResolutionAndReceive16To9AspectRatio) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
// Add video tracks with 16:9 aspect ratio, size 1280 x 720.
|
|
webrtc::FakePeriodicVideoSource::Config config;
|
|
config.width = 1280;
|
|
config.height = 720;
|
|
config.timestamp_offset_ms = rtc::TimeMillis();
|
|
caller()->AddTrack(caller()->CreateLocalVideoTrackWithConfig(config));
|
|
callee()->AddTrack(callee()->CreateLocalVideoTrackWithConfig(config));
|
|
|
|
// Do normal offer/answer and wait for at least one frame to be received in
|
|
// each direction.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
|
|
callee()->min_video_frames_received_per_track() > 0,
|
|
kMaxWaitForFramesMs);
|
|
|
|
// Check rendered aspect ratio.
|
|
EXPECT_EQ(16.0 / 9, caller()->local_rendered_aspect_ratio());
|
|
EXPECT_EQ(16.0 / 9, caller()->rendered_aspect_ratio());
|
|
EXPECT_EQ(16.0 / 9, callee()->local_rendered_aspect_ratio());
|
|
EXPECT_EQ(16.0 / 9, callee()->rendered_aspect_ratio());
|
|
}
|
|
|
|
// This test sets up an one-way call, with media only from caller to
|
|
// callee.
|
|
TEST_P(PeerConnectionIntegrationTest, OneWayMediaCall) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudioAndVideo();
|
|
media_expectations.CallerExpectsNoAudio();
|
|
media_expectations.CallerExpectsNoVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Tests that send only works without the caller having a decoder factory and
|
|
// the callee having an encoder factory.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSendOnlyVideo) {
|
|
ASSERT_TRUE(
|
|
CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true));
|
|
ConnectFakeSignaling();
|
|
// Add one-directional video, from caller to callee.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
|
|
caller()->CreateLocalVideoTrack();
|
|
caller()->AddTrack(caller_track);
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_video = 0;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
|
|
|
|
// Expect video to be received in one direction.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CallerExpectsNoVideo();
|
|
media_expectations.CalleeExpectsSomeVideo();
|
|
|
|
EXPECT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Tests that receive only works without the caller having an encoder factory
|
|
// and the callee having a decoder factory.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithReceiveOnlyVideo) {
|
|
ASSERT_TRUE(
|
|
CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/false));
|
|
ConnectFakeSignaling();
|
|
// Add one-directional video, from callee to caller.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
|
|
callee()->CreateLocalVideoTrack();
|
|
callee()->AddTrack(callee_track);
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_video = 1;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_EQ(caller()->pc()->GetReceivers().size(), 1u);
|
|
|
|
// Expect video to be received in one direction.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CallerExpectsSomeVideo();
|
|
media_expectations.CalleeExpectsNoVideo();
|
|
|
|
EXPECT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
EndToEndCallAddReceiveVideoToSendOnlyCall) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add one-directional video, from caller to callee.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
|
|
caller()->CreateLocalVideoTrack();
|
|
caller()->AddTrack(caller_track);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Add receive video.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
|
|
callee()->CreateLocalVideoTrack();
|
|
callee()->AddTrack(callee_track);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Ensure that video frames are received end-to-end.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
EndToEndCallAddSendVideoToReceiveOnlyCall) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add one-directional video, from callee to caller.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
|
|
callee()->CreateLocalVideoTrack();
|
|
callee()->AddTrack(callee_track);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Add send video.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
|
|
caller()->CreateLocalVideoTrack();
|
|
caller()->AddTrack(caller_track);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Expect video to be received in one direction.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
EndToEndCallRemoveReceiveVideoFromSendReceiveCall) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add send video, from caller to callee.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
|
|
caller()->CreateLocalVideoTrack();
|
|
rtc::scoped_refptr<webrtc::RtpSenderInterface> caller_sender =
|
|
caller()->AddTrack(caller_track);
|
|
// Add receive video, from callee to caller.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
|
|
callee()->CreateLocalVideoTrack();
|
|
|
|
rtc::scoped_refptr<webrtc::RtpSenderInterface> callee_sender =
|
|
callee()->AddTrack(callee_track);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Remove receive video (i.e., callee sender track).
|
|
callee()->pc()->RemoveTrack(callee_sender);
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Expect one-directional video.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CallerExpectsNoVideo();
|
|
media_expectations.CalleeExpectsSomeVideo();
|
|
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
EndToEndCallRemoveSendVideoFromSendReceiveCall) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add send video, from caller to callee.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
|
|
caller()->CreateLocalVideoTrack();
|
|
rtc::scoped_refptr<webrtc::RtpSenderInterface> caller_sender =
|
|
caller()->AddTrack(caller_track);
|
|
// Add receive video, from callee to caller.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
|
|
callee()->CreateLocalVideoTrack();
|
|
|
|
rtc::scoped_refptr<webrtc::RtpSenderInterface> callee_sender =
|
|
callee()->AddTrack(callee_track);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Remove send video (i.e., caller sender track).
|
|
caller()->pc()->RemoveTrack(caller_sender);
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Expect one-directional video.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsNoVideo();
|
|
media_expectations.CallerExpectsSomeVideo();
|
|
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// This test sets up a audio call initially, with the callee rejecting video
|
|
// initially. Then later the callee decides to upgrade to audio/video, and
|
|
// initiates a new offer/answer exchange.
|
|
TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Initially, offer an audio/video stream from the caller, but refuse to
|
|
// send/receive video on the callee side.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioTrack();
|
|
if (sdp_semantics_ == SdpSemantics::kPlanB) {
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_video = 0;
|
|
callee()->SetOfferAnswerOptions(options);
|
|
} else {
|
|
callee()->SetRemoteOfferHandler([this] {
|
|
callee()
|
|
->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
|
|
->StopInternal();
|
|
});
|
|
}
|
|
// Do offer/answer and make sure audio is still received end-to-end.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudio();
|
|
media_expectations.ExpectNoVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
// Sanity check that the callee's description has a rejected video section.
|
|
ASSERT_NE(nullptr, callee()->pc()->local_description());
|
|
const ContentInfo* callee_video_content =
|
|
GetFirstVideoContent(callee()->pc()->local_description()->description());
|
|
ASSERT_NE(nullptr, callee_video_content);
|
|
EXPECT_TRUE(callee_video_content->rejected);
|
|
|
|
// Now negotiate with video and ensure negotiation succeeds, with video
|
|
// frames and additional audio frames being received.
|
|
callee()->AddVideoTrack();
|
|
if (sdp_semantics_ == SdpSemantics::kPlanB) {
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_video = 1;
|
|
callee()->SetOfferAnswerOptions(options);
|
|
} else {
|
|
callee()->SetRemoteOfferHandler(nullptr);
|
|
caller()->SetRemoteOfferHandler([this] {
|
|
// The caller creates a new transceiver to receive video on when receiving
|
|
// the offer, but by default it is send only.
|
|
auto transceivers = caller()->pc()->GetTransceivers();
|
|
ASSERT_EQ(2U, transceivers.size());
|
|
ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
|
|
transceivers[1]->receiver()->media_type());
|
|
transceivers[1]->sender()->SetTrack(caller()->CreateLocalVideoTrack());
|
|
transceivers[1]->SetDirectionWithError(
|
|
RtpTransceiverDirection::kSendRecv);
|
|
});
|
|
}
|
|
callee()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
{
|
|
// Expect additional audio frames to be received after the upgrade.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
}
|
|
|
|
// Simpler than the above test; just add an audio track to an established
|
|
// video-only connection.
|
|
TEST_P(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Do initial offer/answer with just a video track.
|
|
caller()->AddVideoTrack();
|
|
callee()->AddVideoTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Now add an audio track and do another offer/answer.
|
|
caller()->AddAudioTrack();
|
|
callee()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Ensure both audio and video frames are received end-to-end.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// This test sets up a call that's transferred to a new caller with a different
|
|
// DTLS fingerprint.
|
|
TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Keep the original peer around which will still send packets to the
|
|
// receiving client. These SRTP packets will be dropped.
|
|
std::unique_ptr<PeerConnectionWrapper> original_peer(
|
|
SetCallerPcWrapperAndReturnCurrent(
|
|
CreatePeerConnectionWrapperWithAlternateKey().release()));
|
|
// TODO(deadbeef): Why do we call Close here? That goes against the comment
|
|
// directly above.
|
|
original_peer->pc()->Close();
|
|
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Wait for some additional frames to be transmitted end-to-end.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// This test sets up a call that's transferred to a new callee with a different
|
|
// DTLS fingerprint.
|
|
TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Keep the original peer around which will still send packets to the
|
|
// receiving client. These SRTP packets will be dropped.
|
|
std::unique_ptr<PeerConnectionWrapper> original_peer(
|
|
SetCalleePcWrapperAndReturnCurrent(
|
|
CreatePeerConnectionWrapperWithAlternateKey().release()));
|
|
// TODO(deadbeef): Why do we call Close here? That goes against the comment
|
|
// directly above.
|
|
original_peer->pc()->Close();
|
|
|
|
ConnectFakeSignaling();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Wait for some additional frames to be transmitted end-to-end.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// This test sets up a non-bundled call and negotiates bundling at the same
|
|
// time as starting an ICE restart. When bundling is in effect in the restart,
|
|
// the DTLS-SRTP context should be successfully reset.
|
|
TEST_P(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
// Remove the bundle group from the SDP received by the callee.
|
|
callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
|
|
desc->RemoveGroupByName("BUNDLE");
|
|
});
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
// Now stop removing the BUNDLE group, and trigger an ICE restart.
|
|
callee()->SetReceivedSdpMunger(nullptr);
|
|
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Expect additional frames to be received after the ICE restart.
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
}
|
|
|
|
// Test CVO (Coordination of Video Orientation). If a video source is rotated
|
|
// and both peers support the CVO RTP header extension, the actual video frames
|
|
// don't need to be encoded in different resolutions, since the rotation is
|
|
// communicated through the RTP header extension.
|
|
TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add rotated video tracks.
|
|
caller()->AddTrack(
|
|
caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
|
|
callee()->AddTrack(
|
|
callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
|
|
|
|
// Wait for video frames to be received by both sides.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
|
|
callee()->min_video_frames_received_per_track() > 0,
|
|
kMaxWaitForFramesMs);
|
|
|
|
// Ensure that the aspect ratio is unmodified.
|
|
// TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
|
|
// not just assumed.
|
|
EXPECT_EQ(4.0 / 3, caller()->local_rendered_aspect_ratio());
|
|
EXPECT_EQ(4.0 / 3, caller()->rendered_aspect_ratio());
|
|
EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio());
|
|
EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio());
|
|
// Ensure that the CVO bits were surfaced to the renderer.
|
|
EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation());
|
|
EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation());
|
|
}
|
|
|
|
// Test that when the CVO extension isn't supported, video is rotated the
|
|
// old-fashioned way, by encoding rotated frames.
|
|
TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add rotated video tracks.
|
|
caller()->AddTrack(
|
|
caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
|
|
callee()->AddTrack(
|
|
callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
|
|
|
|
// Remove the CVO extension from the offered SDP.
|
|
callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
|
|
cricket::VideoContentDescription* video =
|
|
GetFirstVideoContentDescription(desc);
|
|
video->ClearRtpHeaderExtensions();
|
|
});
|
|
// Wait for video frames to be received by both sides.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
|
|
callee()->min_video_frames_received_per_track() > 0,
|
|
kMaxWaitForFramesMs);
|
|
|
|
// Expect that the aspect ratio is inversed to account for the 90/270 degree
|
|
// rotation.
|
|
// TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
|
|
// not just assumed.
|
|
EXPECT_EQ(3.0 / 4, caller()->local_rendered_aspect_ratio());
|
|
EXPECT_EQ(3.0 / 4, caller()->rendered_aspect_ratio());
|
|
EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio());
|
|
EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio());
|
|
// Expect that each endpoint is unaware of the rotation of the other endpoint.
|
|
EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation());
|
|
EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation());
|
|
}
|
|
|
|
// Test that if the answerer rejects the audio m= section, no audio is sent or
|
|
// received, but video still can be.
|
|
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
if (sdp_semantics_ == SdpSemantics::kPlanB) {
|
|
// Only add video track for callee, and set offer_to_receive_audio to 0, so
|
|
// it will reject the audio m= section completely.
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_audio = 0;
|
|
callee()->SetOfferAnswerOptions(options);
|
|
} else {
|
|
// Stopping the audio RtpTransceiver will cause the media section to be
|
|
// rejected in the answer.
|
|
callee()->SetRemoteOfferHandler([this] {
|
|
callee()
|
|
->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
|
|
->StopInternal();
|
|
});
|
|
}
|
|
callee()->AddTrack(callee()->CreateLocalVideoTrack());
|
|
// Do offer/answer and wait for successful end-to-end video frames.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalVideo();
|
|
media_expectations.ExpectNoAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
|
|
// Sanity check that the callee's description has a rejected audio section.
|
|
ASSERT_NE(nullptr, callee()->pc()->local_description());
|
|
const ContentInfo* callee_audio_content =
|
|
GetFirstAudioContent(callee()->pc()->local_description()->description());
|
|
ASSERT_NE(nullptr, callee_audio_content);
|
|
EXPECT_TRUE(callee_audio_content->rejected);
|
|
if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
|
|
// The caller's transceiver should have stopped after receiving the answer,
|
|
// and thus no longer listed in transceivers.
|
|
EXPECT_EQ(nullptr,
|
|
caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO));
|
|
}
|
|
}
|
|
|
|
// Test that if the answerer rejects the video m= section, no video is sent or
|
|
// received, but audio still can be.
|
|
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
if (sdp_semantics_ == SdpSemantics::kPlanB) {
|
|
// Only add audio track for callee, and set offer_to_receive_video to 0, so
|
|
// it will reject the video m= section completely.
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_video = 0;
|
|
callee()->SetOfferAnswerOptions(options);
|
|
} else {
|
|
// Stopping the video RtpTransceiver will cause the media section to be
|
|
// rejected in the answer.
|
|
callee()->SetRemoteOfferHandler([this] {
|
|
callee()
|
|
->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
|
|
->StopInternal();
|
|
});
|
|
}
|
|
callee()->AddTrack(callee()->CreateLocalAudioTrack());
|
|
// Do offer/answer and wait for successful end-to-end audio frames.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudio();
|
|
media_expectations.ExpectNoVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
|
|
// Sanity check that the callee's description has a rejected video section.
|
|
ASSERT_NE(nullptr, callee()->pc()->local_description());
|
|
const ContentInfo* callee_video_content =
|
|
GetFirstVideoContent(callee()->pc()->local_description()->description());
|
|
ASSERT_NE(nullptr, callee_video_content);
|
|
EXPECT_TRUE(callee_video_content->rejected);
|
|
if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
|
|
// The caller's transceiver should have stopped after receiving the answer,
|
|
// and thus is no longer present.
|
|
EXPECT_EQ(nullptr,
|
|
caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO));
|
|
}
|
|
}
|
|
|
|
// Test that if the answerer rejects both audio and video m= sections, nothing
|
|
// bad happens.
|
|
// TODO(deadbeef): Test that a data channel still works. Currently this doesn't
|
|
// test anything but the fact that negotiation succeeds, which doesn't mean
|
|
// much.
|
|
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
if (sdp_semantics_ == SdpSemantics::kPlanB) {
|
|
// Don't give the callee any tracks, and set offer_to_receive_X to 0, so it
|
|
// will reject both audio and video m= sections.
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_audio = 0;
|
|
options.offer_to_receive_video = 0;
|
|
callee()->SetOfferAnswerOptions(options);
|
|
} else {
|
|
callee()->SetRemoteOfferHandler([this] {
|
|
// Stopping all transceivers will cause all media sections to be rejected.
|
|
for (const auto& transceiver : callee()->pc()->GetTransceivers()) {
|
|
transceiver->StopInternal();
|
|
}
|
|
});
|
|
}
|
|
// Do offer/answer and wait for stable signaling state.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Sanity check that the callee's description has rejected m= sections.
|
|
ASSERT_NE(nullptr, callee()->pc()->local_description());
|
|
const ContentInfo* callee_audio_content =
|
|
GetFirstAudioContent(callee()->pc()->local_description()->description());
|
|
ASSERT_NE(nullptr, callee_audio_content);
|
|
EXPECT_TRUE(callee_audio_content->rejected);
|
|
const ContentInfo* callee_video_content =
|
|
GetFirstVideoContent(callee()->pc()->local_description()->description());
|
|
ASSERT_NE(nullptr, callee_video_content);
|
|
EXPECT_TRUE(callee_video_content->rejected);
|
|
}
|
|
|
|
// This test sets up an audio and video call between two parties. After the
|
|
// call runs for a while, the caller sends an updated offer with video being
|
|
// rejected. Once the re-negotiation is done, the video flow should stop and
|
|
// the audio flow should continue.
|
|
TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
// Renegotiate, rejecting the video m= section.
|
|
if (sdp_semantics_ == SdpSemantics::kPlanB) {
|
|
caller()->SetGeneratedSdpMunger(
|
|
[](cricket::SessionDescription* description) {
|
|
for (cricket::ContentInfo& content : description->contents()) {
|
|
if (cricket::IsVideoContent(&content)) {
|
|
content.rejected = true;
|
|
}
|
|
}
|
|
});
|
|
} else {
|
|
caller()
|
|
->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
|
|
->StopInternal();
|
|
}
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
|
|
|
|
// Sanity check that the caller's description has a rejected video section.
|
|
ASSERT_NE(nullptr, caller()->pc()->local_description());
|
|
const ContentInfo* caller_video_content =
|
|
GetFirstVideoContent(caller()->pc()->local_description()->description());
|
|
ASSERT_NE(nullptr, caller_video_content);
|
|
EXPECT_TRUE(caller_video_content->rejected);
|
|
// Wait for some additional audio frames to be received.
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudio();
|
|
media_expectations.ExpectNoVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
}
|
|
|
|
// Do one offer/answer with audio, another that disables it (rejecting the m=
|
|
// section), and another that re-enables it. Regression test for:
|
|
// bugs.webrtc.org/6023
|
|
TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
// Add audio track, do normal offer/answer.
|
|
rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
|
|
caller()->CreateLocalAudioTrack();
|
|
rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
|
|
caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Remove audio track, and set offer_to_receive_audio to false to cause the
|
|
// m= section to be completely disabled, not just "recvonly".
|
|
caller()->pc()->RemoveTrack(sender);
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_audio = 0;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Add the audio track again, expecting negotiation to succeed and frames to
|
|
// flow.
|
|
sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
|
|
options.offer_to_receive_audio = 1;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio();
|
|
EXPECT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Basic end-to-end test, but without SSRC/MSID signaling. This functionality
|
|
// is needed to support legacy endpoints.
|
|
// TODO(deadbeef): When we support the MID extension and demuxing on MID, also
|
|
// add a test for an end-to-end test without MID signaling either (basically,
|
|
// the minimum acceptable SDP).
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add audio and video, testing that packets can be demuxed on payload type.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
// Remove SSRCs and MSIDs from the received offer SDP.
|
|
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Basic end-to-end test, without SSRC signaling. This means that the track
|
|
// was created properly and frames are delivered when the MSIDs are communicated
|
|
// with a=msid lines and no a=ssrc lines.
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
EndToEndCallWithoutSsrcSignaling) {
|
|
const char kStreamId[] = "streamId";
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add just audio tracks.
|
|
caller()->AddTrack(caller()->CreateLocalAudioTrack(), {kStreamId});
|
|
callee()->AddAudioTrack();
|
|
|
|
// Remove SSRCs from the received offer SDP.
|
|
callee()->SetReceivedSdpMunger(RemoveSsrcsAndKeepMsids);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
EndToEndCallAddReceiveVideoToSendOnlyCall) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add one-directional video, from caller to callee.
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
|
|
caller()->CreateLocalVideoTrack();
|
|
|
|
RtpTransceiverInit video_transceiver_init;
|
|
video_transceiver_init.stream_ids = {"video1"};
|
|
video_transceiver_init.direction = RtpTransceiverDirection::kSendOnly;
|
|
auto video_sender =
|
|
caller()->pc()->AddTransceiver(track, video_transceiver_init).MoveValue();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Add receive direction.
|
|
video_sender->SetDirectionWithError(RtpTransceiverDirection::kSendRecv);
|
|
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
|
|
callee()->CreateLocalVideoTrack();
|
|
|
|
callee()->AddTrack(callee_track);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Ensure that video frames are received end-to-end.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Tests that video flows between multiple video tracks when SSRCs are not
|
|
// signaled. This exercises the MID RTP header extension which is needed to
|
|
// demux the incoming video tracks.
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
EndToEndCallWithTwoVideoTracksAndNoSignaledSsrc) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddVideoTrack();
|
|
caller()->AddVideoTrack();
|
|
callee()->AddVideoTrack();
|
|
callee()->AddVideoTrack();
|
|
|
|
caller()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
|
|
callee()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_EQ(2u, caller()->pc()->GetReceivers().size());
|
|
ASSERT_EQ(2u, callee()->pc()->GetReceivers().size());
|
|
|
|
// Expect video to be received in both directions on both tracks.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalVideo();
|
|
EXPECT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLinePresent) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
caller()->AddVideoTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
auto callee_receivers = callee()->pc()->GetReceivers();
|
|
ASSERT_EQ(2u, callee_receivers.size());
|
|
EXPECT_TRUE(callee_receivers[0]->stream_ids().empty());
|
|
EXPECT_TRUE(callee_receivers[1]->stream_ids().empty());
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLineMissing) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
caller()->AddVideoTrack();
|
|
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
auto callee_receivers = callee()->pc()->GetReceivers();
|
|
ASSERT_EQ(2u, callee_receivers.size());
|
|
ASSERT_EQ(1u, callee_receivers[0]->stream_ids().size());
|
|
ASSERT_EQ(1u, callee_receivers[1]->stream_ids().size());
|
|
EXPECT_EQ(callee_receivers[0]->stream_ids()[0],
|
|
callee_receivers[1]->stream_ids()[0]);
|
|
EXPECT_EQ(callee_receivers[0]->streams()[0],
|
|
callee_receivers[1]->streams()[0]);
|
|
}
|
|
|
|
// Test that if two video tracks are sent (from caller to callee, in this test),
|
|
// they're transmitted correctly end-to-end.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Add one audio/video stream, and one video-only stream.
|
|
caller()->AddAudioVideoTracks();
|
|
caller()->AddVideoTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_EQ(3u, callee()->pc()->GetReceivers().size());
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) {
|
|
bool first = true;
|
|
for (cricket::ContentInfo& content : desc->contents()) {
|
|
if (first) {
|
|
first = false;
|
|
continue;
|
|
}
|
|
content.bundle_only = true;
|
|
}
|
|
first = true;
|
|
for (cricket::TransportInfo& transport : desc->transport_infos()) {
|
|
if (first) {
|
|
first = false;
|
|
continue;
|
|
}
|
|
transport.description.ice_ufrag.clear();
|
|
transport.description.ice_pwd.clear();
|
|
transport.description.connection_role = cricket::CONNECTIONROLE_NONE;
|
|
transport.description.identity_fingerprint.reset(nullptr);
|
|
}
|
|
}
|
|
|
|
// Test that if applying a true "max bundle" offer, which uses ports of 0,
|
|
// "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and
|
|
// "a=ice-pwd" for all but the audio "m=" section, negotiation still completes
|
|
// successfully and media flows.
|
|
// TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works.
|
|
// TODO(deadbeef): Won't need this test once we start generating actual
|
|
// standards-compliant SDP.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
EndToEndCallWithSpecCompliantMaxBundleOffer) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
// Do the equivalent of setting the port to 0, adding a=bundle-only, and
|
|
// removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all
|
|
// but the first m= section.
|
|
callee()->SetReceivedSdpMunger(MakeSpecCompliantMaxBundleOffer);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Test that we can receive the audio output level from a remote audio track.
|
|
// TODO(deadbeef): Use a fake audio source and verify that the output level is
|
|
// exactly what the source on the other side was configured with.
|
|
TEST_P(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Just add an audio track.
|
|
caller()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Get the audio output level stats. Note that the level is not available
|
|
// until an RTCP packet has been received.
|
|
EXPECT_TRUE_WAIT(callee()->OldGetStats()->AudioOutputLevel() > 0,
|
|
kMaxWaitForFramesMs);
|
|
}
|
|
|
|
// Test that an audio input level is reported.
|
|
// TODO(deadbeef): Use a fake audio source and verify that the input level is
|
|
// exactly what the source was configured with.
|
|
TEST_P(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Just add an audio track.
|
|
caller()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Get the audio input level stats. The level should be available very
|
|
// soon after the test starts.
|
|
EXPECT_TRUE_WAIT(caller()->OldGetStats()->AudioInputLevel() > 0,
|
|
kMaxWaitForStatsMs);
|
|
}
|
|
|
|
// Test that we can get incoming byte counts from both audio and video tracks.
|
|
TEST_P(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
// Do offer/answer, wait for the callee to receive some frames.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
|
|
// Get a handle to the remote tracks created, so they can be used as GetStats
|
|
// filters.
|
|
for (const auto& receiver : callee()->pc()->GetReceivers()) {
|
|
// We received frames, so we definitely should have nonzero "received bytes"
|
|
// stats at this point.
|
|
EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(),
|
|
0);
|
|
}
|
|
}
|
|
|
|
// Test that we can get outgoing byte counts from both audio and video tracks.
|
|
TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
auto audio_track = caller()->CreateLocalAudioTrack();
|
|
auto video_track = caller()->CreateLocalVideoTrack();
|
|
caller()->AddTrack(audio_track);
|
|
caller()->AddTrack(video_track);
|
|
// Do offer/answer, wait for the callee to receive some frames.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
|
|
// The callee received frames, so we definitely should have nonzero "sent
|
|
// bytes" stats at this point.
|
|
EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0);
|
|
EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
|
|
}
|
|
|
|
// Test that we can get capture start ntp time.
|
|
TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
|
|
callee()->AddAudioTrack();
|
|
|
|
// Do offer/answer, wait for the callee to receive some frames.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Get the remote audio track created on the receiver, so they can be used as
|
|
// GetStats filters.
|
|
auto receivers = callee()->pc()->GetReceivers();
|
|
ASSERT_EQ(1u, receivers.size());
|
|
auto remote_audio_track = receivers[0]->track();
|
|
|
|
// Get the audio output level stats. Note that the level is not available
|
|
// until an RTCP packet has been received.
|
|
EXPECT_TRUE_WAIT(
|
|
callee()->OldGetStatsForTrack(remote_audio_track)->CaptureStartNtpTime() >
|
|
0,
|
|
2 * kMaxWaitForFramesMs);
|
|
}
|
|
|
|
// Test that the track ID is associated with all local and remote SSRC stats
|
|
// using the old GetStats() and more than 1 audio and more than 1 video track.
|
|
// This is a regression test for crbug.com/906988
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
OldGetStatsAssociatesTrackIdForManyMediaSections) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
auto audio_sender_1 = caller()->AddAudioTrack();
|
|
auto video_sender_1 = caller()->AddVideoTrack();
|
|
auto audio_sender_2 = caller()->AddAudioTrack();
|
|
auto video_sender_2 = caller()->AddVideoTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudioAndVideo();
|
|
ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);
|
|
|
|
std::vector<std::string> track_ids = {
|
|
audio_sender_1->track()->id(), video_sender_1->track()->id(),
|
|
audio_sender_2->track()->id(), video_sender_2->track()->id()};
|
|
|
|
auto caller_stats = caller()->OldGetStats();
|
|
EXPECT_THAT(caller_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
|
|
auto callee_stats = callee()->OldGetStats();
|
|
EXPECT_THAT(callee_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
|
|
}
|
|
|
|
// Test that the new GetStats() returns stats for all outgoing/incoming streams
|
|
// with the correct track IDs if there are more than one audio and more than one
|
|
// video senders/receivers.
|
|
TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
auto audio_sender_1 = caller()->AddAudioTrack();
|
|
auto video_sender_1 = caller()->AddVideoTrack();
|
|
auto audio_sender_2 = caller()->AddAudioTrack();
|
|
auto video_sender_2 = caller()->AddVideoTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudioAndVideo();
|
|
ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);
|
|
|
|
std::vector<std::string> track_ids = {
|
|
audio_sender_1->track()->id(), video_sender_1->track()->id(),
|
|
audio_sender_2->track()->id(), video_sender_2->track()->id()};
|
|
|
|
rtc::scoped_refptr<const webrtc::RTCStatsReport> caller_report =
|
|
caller()->NewGetStats();
|
|
ASSERT_TRUE(caller_report);
|
|
auto outbound_stream_stats =
|
|
caller_report->GetStatsOfType<webrtc::RTCOutboundRTPStreamStats>();
|
|
ASSERT_EQ(outbound_stream_stats.size(), 4u);
|
|
std::vector<std::string> outbound_track_ids;
|
|
for (const auto& stat : outbound_stream_stats) {
|
|
ASSERT_TRUE(stat->bytes_sent.is_defined());
|
|
EXPECT_LT(0u, *stat->bytes_sent);
|
|
if (*stat->kind == "video") {
|
|
ASSERT_TRUE(stat->key_frames_encoded.is_defined());
|
|
EXPECT_GT(*stat->key_frames_encoded, 0u);
|
|
ASSERT_TRUE(stat->frames_encoded.is_defined());
|
|
EXPECT_GE(*stat->frames_encoded, *stat->key_frames_encoded);
|
|
}
|
|
ASSERT_TRUE(stat->track_id.is_defined());
|
|
const auto* track_stat =
|
|
caller_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
|
|
ASSERT_TRUE(track_stat);
|
|
outbound_track_ids.push_back(*track_stat->track_identifier);
|
|
}
|
|
EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids));
|
|
|
|
rtc::scoped_refptr<const webrtc::RTCStatsReport> callee_report =
|
|
callee()->NewGetStats();
|
|
ASSERT_TRUE(callee_report);
|
|
auto inbound_stream_stats =
|
|
callee_report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
|
|
ASSERT_EQ(4u, inbound_stream_stats.size());
|
|
std::vector<std::string> inbound_track_ids;
|
|
for (const auto& stat : inbound_stream_stats) {
|
|
ASSERT_TRUE(stat->bytes_received.is_defined());
|
|
EXPECT_LT(0u, *stat->bytes_received);
|
|
if (*stat->kind == "video") {
|
|
ASSERT_TRUE(stat->key_frames_decoded.is_defined());
|
|
EXPECT_GT(*stat->key_frames_decoded, 0u);
|
|
ASSERT_TRUE(stat->frames_decoded.is_defined());
|
|
EXPECT_GE(*stat->frames_decoded, *stat->key_frames_decoded);
|
|
}
|
|
ASSERT_TRUE(stat->track_id.is_defined());
|
|
const auto* track_stat =
|
|
callee_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
|
|
ASSERT_TRUE(track_stat);
|
|
inbound_track_ids.push_back(*track_stat->track_identifier);
|
|
}
|
|
EXPECT_THAT(inbound_track_ids, UnorderedElementsAreArray(track_ids));
|
|
}
|
|
|
|
// Test that we can get stats (using the new stats implementation) for
|
|
// unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
|
|
// SDP.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
GetStatsForUnsignaledStreamWithNewStatsApi) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
// Remove SSRCs and MSIDs from the received offer SDP.
|
|
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio(1);
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
|
|
// We received a frame, so we should have nonzero "bytes received" stats for
|
|
// the unsignaled stream, if stats are working for it.
|
|
rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
|
|
callee()->NewGetStats();
|
|
ASSERT_NE(nullptr, report);
|
|
auto inbound_stream_stats =
|
|
report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
|
|
ASSERT_EQ(1U, inbound_stream_stats.size());
|
|
ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
|
|
ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
|
|
ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined());
|
|
}
|
|
|
|
// Same as above but for the legacy stats implementation.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
GetStatsForUnsignaledStreamWithOldStatsApi) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
// Remove SSRCs and MSIDs from the received offer SDP.
|
|
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Note that, since the old stats implementation associates SSRCs with tracks
|
|
// using SDP, when SSRCs aren't signaled in SDP these stats won't have an
|
|
// associated track ID. So we can't use the track "selector" argument.
|
|
//
|
|
// Also, we use "EXPECT_TRUE_WAIT" because the stats collector may decide to
|
|
// return cached stats if not enough time has passed since the last update.
|
|
EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0,
|
|
kDefaultTimeout);
|
|
}
|
|
|
|
// Test that we can successfully get the media related stats (audio level
|
|
// etc.) for the unsignaled stream.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
// Remove SSRCs and MSIDs from the received offer SDP.
|
|
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio(1);
|
|
media_expectations.CalleeExpectsSomeVideo(1);
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
|
|
rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
|
|
callee()->NewGetStats();
|
|
ASSERT_NE(nullptr, report);
|
|
|
|
auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
|
|
auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats);
|
|
ASSERT_GE(audio_index, 0);
|
|
EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined());
|
|
}
|
|
|
|
// Helper for test below.
|
|
void ModifySsrcs(cricket::SessionDescription* desc) {
|
|
for (ContentInfo& content : desc->contents()) {
|
|
for (StreamParams& stream :
|
|
content.media_description()->mutable_streams()) {
|
|
for (uint32_t& ssrc : stream.ssrcs) {
|
|
ssrc = rtc::CreateRandomId();
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Test that the "RTCMediaSteamTrackStats" object is updated correctly when
|
|
// SSRCs are unsignaled, and the SSRC of the received (audio) stream changes.
|
|
// This should result in two "RTCInboundRTPStreamStats", but only one
|
|
// "RTCMediaStreamTrackStats", whose counters go up continuously rather than
|
|
// being reset to 0 once the SSRC change occurs.
|
|
//
|
|
// Regression test for this bug:
|
|
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
|
|
//
|
|
// The bug causes the track stats to only represent one of the two streams:
|
|
// whichever one has the higher SSRC. So with this bug, there was a 50% chance
|
|
// that the track stat counters would reset to 0 when the new stream is
|
|
// received, and a 50% chance that they'll stop updating (while
|
|
// "concealed_samples" continues increasing, due to silence being generated for
|
|
// the inactive stream).
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
// Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint
|
|
// that doesn't signal SSRCs (from the callee's perspective).
|
|
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Wait for 50 audio frames (500ms of audio) to be received by the callee.
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio(50);
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
// Some audio frames were received, so we should have nonzero "samples
|
|
// received" for the track.
|
|
rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
|
|
callee()->NewGetStats();
|
|
ASSERT_NE(nullptr, report);
|
|
auto track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
|
|
ASSERT_EQ(1U, track_stats.size());
|
|
ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
|
|
ASSERT_GT(*track_stats[0]->total_samples_received, 0U);
|
|
// uint64_t prev_samples_received = *track_stats[0]->total_samples_received;
|
|
|
|
// Create a new offer and munge it to cause the caller to use a new SSRC.
|
|
caller()->SetGeneratedSdpMunger(ModifySsrcs);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Wait for 25 more audio frames (250ms of audio) to be received, from the new
|
|
// SSRC.
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio(25);
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
report = callee()->NewGetStats();
|
|
ASSERT_NE(nullptr, report);
|
|
track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
|
|
ASSERT_EQ(1U, track_stats.size());
|
|
ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
|
|
// The "total samples received" stat should only be greater than it was
|
|
// before.
|
|
// TODO(deadbeef): Uncomment this assertion once the bug is completely fixed.
|
|
// Right now, the new SSRC will cause the counters to reset to 0.
|
|
// EXPECT_GT(*track_stats[0]->total_samples_received, prev_samples_received);
|
|
|
|
// Additionally, the percentage of concealed samples (samples generated to
|
|
// conceal packet loss) should be less than 50%. If it's greater, that's a
|
|
// good sign that we're seeing stats from the old stream that's no longer
|
|
// receiving packets, and is generating concealed samples of silence.
|
|
constexpr double kAcceptableConcealedSamplesPercentage = 0.50;
|
|
ASSERT_TRUE(track_stats[0]->concealed_samples.is_defined());
|
|
EXPECT_LT(*track_stats[0]->concealed_samples,
|
|
*track_stats[0]->total_samples_received *
|
|
kAcceptableConcealedSamplesPercentage);
|
|
|
|
// Also ensure that we have two "RTCInboundRTPStreamStats" as expected, as a
|
|
// sanity check that the SSRC really changed.
|
|
// TODO(deadbeef): This isn't working right now, because we're not returning
|
|
// *any* stats for the inactive stream. Uncomment when the bug is completely
|
|
// fixed.
|
|
// auto inbound_stream_stats =
|
|
// report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
|
|
// ASSERT_EQ(2U, inbound_stream_stats.size());
|
|
}
|
|
|
|
// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
|
|
PeerConnectionFactory::Options dtls_10_options;
|
|
dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
|
|
dtls_10_options));
|
|
ConnectFakeSignaling();
|
|
// Do normal offer/answer and wait for some frames to be received in each
|
|
// direction.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated.
|
|
TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
|
|
PeerConnectionFactory::Options dtls_10_options;
|
|
dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
|
|
dtls_10_options));
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
|
|
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
|
|
kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
|
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
|
|
// TODO(bugs.webrtc.org/9456): Fix it.
|
|
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
|
|
kDefaultSrtpCryptoSuite));
|
|
}
|
|
|
|
// Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
|
|
TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
|
|
PeerConnectionFactory::Options dtls_12_options;
|
|
dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options,
|
|
dtls_12_options));
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
|
|
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
|
|
kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
|
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
|
|
// TODO(bugs.webrtc.org/9456): Fix it.
|
|
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
|
|
kDefaultSrtpCryptoSuite));
|
|
}
|
|
|
|
// Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
|
|
// callee only supports 1.0.
|
|
TEST_P(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
|
|
PeerConnectionFactory::Options caller_options;
|
|
caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
|
PeerConnectionFactory::Options callee_options;
|
|
callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
|
|
ConnectFakeSignaling();
|
|
// Do normal offer/answer and wait for some frames to be received in each
|
|
// direction.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the
|
|
// callee supports 1.2.
|
|
TEST_P(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
|
|
PeerConnectionFactory::Options caller_options;
|
|
caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
|
PeerConnectionFactory::Options callee_options;
|
|
callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
|
|
ConnectFakeSignaling();
|
|
// Do normal offer/answer and wait for some frames to be received in each
|
|
// direction.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// The three tests below verify that "enable_aes128_sha1_32_crypto_cipher"
|
|
// works as expected; the cipher should only be used if enabled by both sides.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
|
|
PeerConnectionFactory::Options caller_options;
|
|
caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
|
|
PeerConnectionFactory::Options callee_options;
|
|
callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
|
|
false;
|
|
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
|
|
TestNegotiatedCipherSuite(caller_options, callee_options,
|
|
expected_cipher_suite);
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
|
|
PeerConnectionFactory::Options caller_options;
|
|
caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
|
|
false;
|
|
PeerConnectionFactory::Options callee_options;
|
|
callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
|
|
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
|
|
TestNegotiatedCipherSuite(caller_options, callee_options,
|
|
expected_cipher_suite);
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
|
|
PeerConnectionFactory::Options caller_options;
|
|
caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
|
|
PeerConnectionFactory::Options callee_options;
|
|
callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
|
|
int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32;
|
|
TestNegotiatedCipherSuite(caller_options, callee_options,
|
|
expected_cipher_suite);
|
|
}
|
|
|
|
// Test that a non-GCM cipher is used if both sides only support non-GCM.
|
|
TEST_P(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
|
|
bool local_gcm_enabled = false;
|
|
bool remote_gcm_enabled = false;
|
|
bool aes_ctr_enabled = true;
|
|
int expected_cipher_suite = kDefaultSrtpCryptoSuite;
|
|
TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
|
|
aes_ctr_enabled, expected_cipher_suite);
|
|
}
|
|
|
|
// Test that a GCM cipher is used if both ends support it and non-GCM is
|
|
// disabled.
|
|
TEST_P(PeerConnectionIntegrationTest, GcmCipherUsedWhenOnlyGcmSupported) {
|
|
bool local_gcm_enabled = true;
|
|
bool remote_gcm_enabled = true;
|
|
bool aes_ctr_enabled = false;
|
|
int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm;
|
|
TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
|
|
aes_ctr_enabled, expected_cipher_suite);
|
|
}
|
|
|
|
// Verify that media can be transmitted end-to-end when GCM crypto suites are
|
|
// enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported,
|
|
// only verify that a GCM cipher is negotiated, and not necessarily that SRTP
|
|
// works with it.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
|
|
PeerConnectionFactory::Options gcm_options;
|
|
gcm_options.crypto_options.srtp.enable_gcm_crypto_suites = true;
|
|
gcm_options.crypto_options.srtp.enable_aes128_sha1_80_crypto_cipher = false;
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options));
|
|
ConnectFakeSignaling();
|
|
// Do normal offer/answer and wait for some frames to be received in each
|
|
// direction.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// This test sets up a call between two parties with audio, video and an RTP
|
|
// data channel.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
|
|
PeerConnectionInterface::RTCConfiguration rtc_config;
|
|
rtc_config.enable_rtp_data_channel = true;
|
|
rtc_config.enable_dtls_srtp = false;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
|
|
ConnectFakeSignaling();
|
|
// Expect that data channel created on caller side will show up for callee as
|
|
// well.
|
|
caller()->CreateDataChannel();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Ensure the existence of the RTP data channel didn't impede audio/video.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
ASSERT_NE(nullptr, caller()->data_channel());
|
|
ASSERT_NE(nullptr, callee()->data_channel());
|
|
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
|
|
// Ensure data can be sent in both directions.
|
|
std::string data = "hello world";
|
|
SendRtpDataWithRetries(caller()->data_channel(), data, 5);
|
|
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
SendRtpDataWithRetries(callee()->data_channel(), data, 5);
|
|
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, RtpDataChannelWorksAfterRollback) {
|
|
PeerConnectionInterface::RTCConfiguration rtc_config;
|
|
rtc_config.enable_rtp_data_channel = true;
|
|
rtc_config.enable_dtls_srtp = false;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
|
|
ConnectFakeSignaling();
|
|
auto data_channel = caller()->pc()->CreateDataChannel("label_1", nullptr);
|
|
ASSERT_TRUE(data_channel.get() != nullptr);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
caller()->CreateDataChannel("label_2", nullptr);
|
|
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
|
|
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
|
|
caller()->pc()->SetLocalDescription(observer,
|
|
caller()->CreateOfferAndWait().release());
|
|
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
|
|
caller()->Rollback();
|
|
|
|
std::string data = "hello world";
|
|
SendRtpDataWithRetries(data_channel, data, 5);
|
|
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
}
|
|
|
|
// Ensure that an RTP data channel is signaled as closed for the caller when
|
|
// the callee rejects it in a subsequent offer.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
RtpDataChannelSignaledClosedInCalleeOffer) {
|
|
// Same procedure as above test.
|
|
PeerConnectionInterface::RTCConfiguration rtc_config;
|
|
rtc_config.enable_rtp_data_channel = true;
|
|
rtc_config.enable_dtls_srtp = false;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_NE(nullptr, caller()->data_channel());
|
|
ASSERT_NE(nullptr, callee()->data_channel());
|
|
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
|
|
// Close the data channel on the callee, and do an updated offer/answer.
|
|
callee()->data_channel()->Close();
|
|
callee()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_FALSE(caller()->data_observer()->IsOpen());
|
|
EXPECT_FALSE(callee()->data_observer()->IsOpen());
|
|
}
|
|
|
|
// Tests that data is buffered in an RTP data channel until an observer is
|
|
// registered for it.
|
|
//
|
|
// NOTE: RTP data channels can receive data before the underlying
|
|
// transport has detected that a channel is writable and thus data can be
|
|
// received before the data channel state changes to open. That is hard to test
|
|
// but the same buffering is expected to be used in that case.
|
|
//
|
|
// Use fake clock and simulated network delay so that we predictably can wait
|
|
// until an SCTP message has been delivered without "sleep()"ing.
|
|
TEST_P(PeerConnectionIntegrationTestWithFakeClock,
|
|
DataBufferedUntilRtpDataChannelObserverRegistered) {
|
|
virtual_socket_server()->set_delay_mean(5); // 5 ms per hop.
|
|
virtual_socket_server()->UpdateDelayDistribution();
|
|
|
|
PeerConnectionInterface::RTCConfiguration rtc_config;
|
|
rtc_config.enable_rtp_data_channel = true;
|
|
rtc_config.enable_dtls_srtp = false;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE(caller()->data_channel() != nullptr);
|
|
ASSERT_TRUE_SIMULATED_WAIT(callee()->data_channel() != nullptr,
|
|
kDefaultTimeout, FakeClock());
|
|
ASSERT_TRUE_SIMULATED_WAIT(caller()->data_observer()->IsOpen(),
|
|
kDefaultTimeout, FakeClock());
|
|
ASSERT_EQ_SIMULATED_WAIT(DataChannelInterface::kOpen,
|
|
callee()->data_channel()->state(), kDefaultTimeout,
|
|
FakeClock());
|
|
|
|
// Unregister the observer which is normally automatically registered.
|
|
callee()->data_channel()->UnregisterObserver();
|
|
// Send data and advance fake clock until it should have been received.
|
|
std::string data = "hello world";
|
|
caller()->data_channel()->Send(DataBuffer(data));
|
|
SIMULATED_WAIT(false, 50, FakeClock());
|
|
|
|
// Attach data channel and expect data to be received immediately. Note that
|
|
// EXPECT_EQ_WAIT is used, such that the simulated clock is not advanced any
|
|
// further, but data can be received even if the callback is asynchronous.
|
|
MockDataChannelObserver new_observer(callee()->data_channel());
|
|
EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout,
|
|
FakeClock());
|
|
}
|
|
|
|
// This test sets up a call between two parties with audio, video and but only
|
|
// the caller client supports RTP data channels.
|
|
TEST_P(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
|
|
PeerConnectionInterface::RTCConfiguration rtc_config_1;
|
|
rtc_config_1.enable_rtp_data_channel = true;
|
|
// Must disable DTLS to make negotiation succeed.
|
|
rtc_config_1.enable_dtls_srtp = false;
|
|
PeerConnectionInterface::RTCConfiguration rtc_config_2;
|
|
rtc_config_2.enable_dtls_srtp = false;
|
|
rtc_config_2.enable_dtls_srtp = false;
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithConfig(rtc_config_1, rtc_config_2));
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
ASSERT_TRUE(caller()->data_channel() != nullptr);
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// The caller should still have a data channel, but it should be closed, and
|
|
// one should ever have been created for the callee.
|
|
EXPECT_TRUE(caller()->data_channel() != nullptr);
|
|
EXPECT_FALSE(caller()->data_observer()->IsOpen());
|
|
EXPECT_EQ(nullptr, callee()->data_channel());
|
|
}
|
|
|
|
// This test sets up a call between two parties with audio, and video. When
|
|
// audio and video is setup and flowing, an RTP data channel is negotiated.
|
|
TEST_P(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
|
|
PeerConnectionInterface::RTCConfiguration rtc_config;
|
|
rtc_config.enable_rtp_data_channel = true;
|
|
rtc_config.enable_dtls_srtp = false;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
|
|
ConnectFakeSignaling();
|
|
// Do initial offer/answer with audio/video.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Create data channel and do new offer and answer.
|
|
caller()->CreateDataChannel();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_NE(nullptr, caller()->data_channel());
|
|
ASSERT_NE(nullptr, callee()->data_channel());
|
|
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
// Ensure data can be sent in both directions.
|
|
std::string data = "hello world";
|
|
SendRtpDataWithRetries(caller()->data_channel(), data, 5);
|
|
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
SendRtpDataWithRetries(callee()->data_channel(), data, 5);
|
|
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
}
|
|
|
|
#ifdef HAVE_SCTP
|
|
|
|
// This test sets up a call between two parties with audio, video and an SCTP
|
|
// data channel.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Expect that data channel created on caller side will show up for callee as
|
|
// well.
|
|
caller()->CreateDataChannel();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Ensure the existence of the SCTP data channel didn't impede audio/video.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
// Caller data channel should already exist (it created one). Callee data
|
|
// channel may not exist yet, since negotiation happens in-band, not in SDP.
|
|
ASSERT_NE(nullptr, caller()->data_channel());
|
|
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
|
|
// Ensure data can be sent in both directions.
|
|
std::string data = "hello world";
|
|
caller()->data_channel()->Send(DataBuffer(data));
|
|
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
callee()->data_channel()->Send(DataBuffer(data));
|
|
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
}
|
|
|
|
// Ensure that when the callee closes an SCTP data channel, the closing
|
|
// procedure results in the data channel being closed for the caller as well.
|
|
TEST_P(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) {
|
|
// Same procedure as above test.
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_NE(nullptr, caller()->data_channel());
|
|
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
|
|
// Close the data channel on the callee side, and wait for it to reach the
|
|
// "closed" state on both sides.
|
|
callee()->data_channel()->Close();
|
|
EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
webrtc::DataChannelInit init;
|
|
init.id = 53;
|
|
init.maxRetransmits = 52;
|
|
caller()->CreateDataChannel("data-channel", &init);
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
// Since "negotiated" is false, the "id" parameter should be ignored.
|
|
EXPECT_NE(init.id, callee()->data_channel()->id());
|
|
EXPECT_EQ("data-channel", callee()->data_channel()->label());
|
|
EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
|
|
EXPECT_FALSE(callee()->data_channel()->negotiated());
|
|
}
|
|
|
|
// Test usrsctp's ability to process unordered data stream, where data actually
|
|
// arrives out of order using simulated delays. Previously there have been some
|
|
// bugs in this area.
|
|
TEST_P(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
|
|
// Introduce random network delays.
|
|
// Otherwise it's not a true "unordered" test.
|
|
virtual_socket_server()->set_delay_mean(20);
|
|
virtual_socket_server()->set_delay_stddev(5);
|
|
virtual_socket_server()->UpdateDelayDistribution();
|
|
// Normal procedure, but with unordered data channel config.
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
webrtc::DataChannelInit init;
|
|
init.ordered = false;
|
|
caller()->CreateDataChannel(&init);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_NE(nullptr, caller()->data_channel());
|
|
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
|
|
static constexpr int kNumMessages = 100;
|
|
// Deliberately chosen to be larger than the MTU so messages get fragmented.
|
|
static constexpr size_t kMaxMessageSize = 4096;
|
|
// Create and send random messages.
|
|
std::vector<std::string> sent_messages;
|
|
for (int i = 0; i < kNumMessages; ++i) {
|
|
size_t length =
|
|
(rand() % kMaxMessageSize) + 1; // NOLINT (rand_r instead of rand)
|
|
std::string message;
|
|
ASSERT_TRUE(rtc::CreateRandomString(length, &message));
|
|
caller()->data_channel()->Send(DataBuffer(message));
|
|
callee()->data_channel()->Send(DataBuffer(message));
|
|
sent_messages.push_back(message);
|
|
}
|
|
|
|
// Wait for all messages to be received.
|
|
EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
|
|
caller()->data_observer()->received_message_count(),
|
|
kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
|
|
callee()->data_observer()->received_message_count(),
|
|
kDefaultTimeout);
|
|
|
|
// Sort and compare to make sure none of the messages were corrupted.
|
|
std::vector<std::string> caller_received_messages =
|
|
caller()->data_observer()->messages();
|
|
std::vector<std::string> callee_received_messages =
|
|
callee()->data_observer()->messages();
|
|
absl::c_sort(sent_messages);
|
|
absl::c_sort(caller_received_messages);
|
|
absl::c_sort(callee_received_messages);
|
|
EXPECT_EQ(sent_messages, caller_received_messages);
|
|
EXPECT_EQ(sent_messages, callee_received_messages);
|
|
}
|
|
|
|
// This test sets up a call between two parties with audio, and video. When
|
|
// audio and video are setup and flowing, an SCTP data channel is negotiated.
|
|
TEST_P(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Do initial offer/answer with audio/video.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Create data channel and do new offer and answer.
|
|
caller()->CreateDataChannel();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Caller data channel should already exist (it created one). Callee data
|
|
// channel may not exist yet, since negotiation happens in-band, not in SDP.
|
|
ASSERT_NE(nullptr, caller()->data_channel());
|
|
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
// Ensure data can be sent in both directions.
|
|
std::string data = "hello world";
|
|
caller()->data_channel()->Send(DataBuffer(data));
|
|
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
callee()->data_channel()->Send(DataBuffer(data));
|
|
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
}
|
|
|
|
// Set up a connection initially just using SCTP data channels, later upgrading
|
|
// to audio/video, ensuring frames are received end-to-end. Effectively the
|
|
// inverse of the test above.
|
|
// This was broken in M57; see https://crbug.com/711243
|
|
TEST_P(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Do initial offer/answer with just data channel.
|
|
caller()->CreateDataChannel();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Wait until data can be sent over the data channel.
|
|
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
|
|
// Do subsequent offer/answer with two-way audio and video. Audio and video
|
|
// should end up bundled on the DTLS/ICE transport already used for data.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) {
|
|
cricket::SctpDataContentDescription* dcd_offer =
|
|
GetFirstSctpDataContentDescription(desc);
|
|
// See https://crbug.com/webrtc/11211 - this function is a no-op
|
|
ASSERT_TRUE(dcd_offer);
|
|
dcd_offer->set_use_sctpmap(false);
|
|
dcd_offer->set_protocol("UDP/DTLS/SCTP");
|
|
}
|
|
|
|
// Test that the data channel works when a spec-compliant SCTP m= section is
|
|
// offered (using "a=sctp-port" instead of "a=sctpmap", and using
|
|
// "UDP/DTLS/SCTP" as the protocol).
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
DataChannelWorksWhenSpecCompliantSctpOfferReceived) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->SetGeneratedSdpMunger(MakeSpecCompliantSctpOffer);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
|
|
// Ensure data can be sent in both directions.
|
|
std::string data = "hello world";
|
|
caller()->data_channel()->Send(DataBuffer(data));
|
|
EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
callee()->data_channel()->Send(DataBuffer(data));
|
|
EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
|
|
kDefaultTimeout);
|
|
}
|
|
|
|
#endif // HAVE_SCTP
|
|
|
|
// Test that the ICE connection and gathering states eventually reach
|
|
// "complete".
|
|
TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Do normal offer/answer.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
|
|
caller()->ice_gathering_state(), kMaxWaitForFramesMs);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
|
|
callee()->ice_gathering_state(), kMaxWaitForFramesMs);
|
|
// After the best candidate pair is selected and all candidates are signaled,
|
|
// the ICE connection state should reach "complete".
|
|
// TODO(deadbeef): Currently, the ICE "controlled" agent (the
|
|
// answerer/"callee" by default) only reaches "connected". When this is
|
|
// fixed, this test should be updated.
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kDefaultTimeout);
|
|
}
|
|
|
|
constexpr int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
|
|
cricket::PORTALLOCATOR_DISABLE_RELAY |
|
|
cricket::PORTALLOCATOR_DISABLE_TCP;
|
|
|
|
// Use a mock resolver to resolve the hostname back to the original IP on both
|
|
// sides and check that the ICE connection connects.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
IceStatesReachCompletionWithRemoteHostname) {
|
|
auto caller_resolver_factory =
|
|
std::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
|
|
auto callee_resolver_factory =
|
|
std::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
|
|
NiceMock<rtc::MockAsyncResolver> callee_async_resolver;
|
|
NiceMock<rtc::MockAsyncResolver> caller_async_resolver;
|
|
|
|
// This also verifies that the injected AsyncResolverFactory is used by
|
|
// P2PTransportChannel.
|
|
EXPECT_CALL(*caller_resolver_factory, Create())
|
|
.WillOnce(Return(&caller_async_resolver));
|
|
webrtc::PeerConnectionDependencies caller_deps(nullptr);
|
|
caller_deps.async_resolver_factory = std::move(caller_resolver_factory);
|
|
|
|
EXPECT_CALL(*callee_resolver_factory, Create())
|
|
.WillOnce(Return(&callee_async_resolver));
|
|
webrtc::PeerConnectionDependencies callee_deps(nullptr);
|
|
callee_deps.async_resolver_factory = std::move(callee_resolver_factory);
|
|
|
|
PeerConnectionInterface::RTCConfiguration config;
|
|
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
|
|
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
|
|
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
|
|
config, std::move(caller_deps), config, std::move(callee_deps)));
|
|
|
|
caller()->SetRemoteAsyncResolver(&callee_async_resolver);
|
|
callee()->SetRemoteAsyncResolver(&caller_async_resolver);
|
|
|
|
// Enable hostname candidates with mDNS names.
|
|
caller()->SetMdnsResponder(
|
|
std::make_unique<webrtc::FakeMdnsResponder>(network_thread()));
|
|
callee()->SetMdnsResponder(
|
|
std::make_unique<webrtc::FakeMdnsResponder>(network_thread()));
|
|
|
|
SetPortAllocatorFlags(kOnlyLocalPorts, kOnlyLocalPorts);
|
|
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kDefaultTimeout);
|
|
|
|
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
|
|
"WebRTC.PeerConnection.CandidatePairType_UDP",
|
|
webrtc::kIceCandidatePairHostNameHostName));
|
|
}
|
|
|
|
// Test that firewalling the ICE connection causes the clients to identify the
|
|
// disconnected state and then removing the firewall causes them to reconnect.
|
|
class PeerConnectionIntegrationIceStatesTest
|
|
: public PeerConnectionIntegrationBaseTest,
|
|
public ::testing::WithParamInterface<
|
|
std::tuple<SdpSemantics, std::tuple<std::string, uint32_t>>> {
|
|
protected:
|
|
PeerConnectionIntegrationIceStatesTest()
|
|
: PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())) {
|
|
port_allocator_flags_ = std::get<1>(std::get<1>(GetParam()));
|
|
}
|
|
|
|
void StartStunServer(const SocketAddress& server_address) {
|
|
stun_server_.reset(
|
|
cricket::TestStunServer::Create(network_thread(), server_address));
|
|
}
|
|
|
|
bool TestIPv6() {
|
|
return (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
}
|
|
|
|
void SetPortAllocatorFlags() {
|
|
PeerConnectionIntegrationBaseTest::SetPortAllocatorFlags(
|
|
port_allocator_flags_, port_allocator_flags_);
|
|
}
|
|
|
|
std::vector<SocketAddress> CallerAddresses() {
|
|
std::vector<SocketAddress> addresses;
|
|
addresses.push_back(SocketAddress("1.1.1.1", 0));
|
|
if (TestIPv6()) {
|
|
addresses.push_back(SocketAddress("1111:0:a:b:c:d:e:f", 0));
|
|
}
|
|
return addresses;
|
|
}
|
|
|
|
std::vector<SocketAddress> CalleeAddresses() {
|
|
std::vector<SocketAddress> addresses;
|
|
addresses.push_back(SocketAddress("2.2.2.2", 0));
|
|
if (TestIPv6()) {
|
|
addresses.push_back(SocketAddress("2222:0:a:b:c:d:e:f", 0));
|
|
}
|
|
return addresses;
|
|
}
|
|
|
|
void SetUpNetworkInterfaces() {
|
|
// Remove the default interfaces added by the test infrastructure.
|
|
caller()->network_manager()->RemoveInterface(kDefaultLocalAddress);
|
|
callee()->network_manager()->RemoveInterface(kDefaultLocalAddress);
|
|
|
|
// Add network addresses for test.
|
|
for (const auto& caller_address : CallerAddresses()) {
|
|
caller()->network_manager()->AddInterface(caller_address);
|
|
}
|
|
for (const auto& callee_address : CalleeAddresses()) {
|
|
callee()->network_manager()->AddInterface(callee_address);
|
|
}
|
|
}
|
|
|
|
private:
|
|
uint32_t port_allocator_flags_;
|
|
std::unique_ptr<cricket::TestStunServer> stun_server_;
|
|
};
|
|
|
|
// Ensure FakeClockForTest is constructed first (see class for rationale).
|
|
class PeerConnectionIntegrationIceStatesTestWithFakeClock
|
|
: public FakeClockForTest,
|
|
public PeerConnectionIntegrationIceStatesTest {};
|
|
|
|
// Tests that the PeerConnection goes through all the ICE gathering/connection
|
|
// states over the duration of the call. This includes Disconnected and Failed
|
|
// states, induced by putting a firewall between the peers and waiting for them
|
|
// to time out.
|
|
TEST_P(PeerConnectionIntegrationIceStatesTestWithFakeClock, VerifyIceStates) {
|
|
const SocketAddress kStunServerAddress =
|
|
SocketAddress("99.99.99.1", cricket::STUN_SERVER_PORT);
|
|
StartStunServer(kStunServerAddress);
|
|
|
|
PeerConnectionInterface::RTCConfiguration config;
|
|
PeerConnectionInterface::IceServer ice_stun_server;
|
|
ice_stun_server.urls.push_back(
|
|
"stun:" + kStunServerAddress.HostAsURIString() + ":" +
|
|
kStunServerAddress.PortAsString());
|
|
config.servers.push_back(ice_stun_server);
|
|
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
|
|
ConnectFakeSignaling();
|
|
SetPortAllocatorFlags();
|
|
SetUpNetworkInterfaces();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
|
|
// Initial state before anything happens.
|
|
ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew,
|
|
caller()->ice_gathering_state());
|
|
ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
|
|
caller()->ice_connection_state());
|
|
ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
|
|
caller()->standardized_ice_connection_state());
|
|
|
|
// Start the call by creating the offer, setting it as the local description,
|
|
// then sending it to the peer who will respond with an answer. This happens
|
|
// asynchronously so that we can watch the states as it runs in the
|
|
// background.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kDefaultTimeout,
|
|
FakeClock());
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->standardized_ice_connection_state(),
|
|
kDefaultTimeout, FakeClock());
|
|
|
|
// Verify that the observer was notified of the intermediate transitions.
|
|
EXPECT_THAT(caller()->ice_connection_state_history(),
|
|
ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
|
|
PeerConnectionInterface::kIceConnectionConnected,
|
|
PeerConnectionInterface::kIceConnectionCompleted));
|
|
EXPECT_THAT(caller()->standardized_ice_connection_state_history(),
|
|
ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
|
|
PeerConnectionInterface::kIceConnectionConnected,
|
|
PeerConnectionInterface::kIceConnectionCompleted));
|
|
EXPECT_THAT(
|
|
caller()->peer_connection_state_history(),
|
|
ElementsAre(PeerConnectionInterface::PeerConnectionState::kConnecting,
|
|
PeerConnectionInterface::PeerConnectionState::kConnected));
|
|
EXPECT_THAT(caller()->ice_gathering_state_history(),
|
|
ElementsAre(PeerConnectionInterface::kIceGatheringGathering,
|
|
PeerConnectionInterface::kIceGatheringComplete));
|
|
|
|
// Block connections to/from the caller and wait for ICE to become
|
|
// disconnected.
|
|
for (const auto& caller_address : CallerAddresses()) {
|
|
firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
|
|
}
|
|
RTC_LOG(LS_INFO) << "Firewall rules applied";
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
|
|
caller()->ice_connection_state(), kDefaultTimeout,
|
|
FakeClock());
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
|
|
caller()->standardized_ice_connection_state(),
|
|
kDefaultTimeout, FakeClock());
|
|
|
|
// Let ICE re-establish by removing the firewall rules.
|
|
firewall()->ClearRules();
|
|
RTC_LOG(LS_INFO) << "Firewall rules cleared";
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kDefaultTimeout,
|
|
FakeClock());
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->standardized_ice_connection_state(),
|
|
kDefaultTimeout, FakeClock());
|
|
|
|
// According to RFC7675, if there is no response within 30 seconds then the
|
|
// peer should consider the other side to have rejected the connection. This
|
|
// is signaled by the state transitioning to "failed".
|
|
constexpr int kConsentTimeout = 30000;
|
|
for (const auto& caller_address : CallerAddresses()) {
|
|
firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
|
|
}
|
|
RTC_LOG(LS_INFO) << "Firewall rules applied again";
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
|
|
caller()->ice_connection_state(), kConsentTimeout,
|
|
FakeClock());
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
|
|
caller()->standardized_ice_connection_state(),
|
|
kConsentTimeout, FakeClock());
|
|
}
|
|
|
|
// Tests that if the connection doesn't get set up properly we eventually reach
|
|
// the "failed" iceConnectionState.
|
|
TEST_P(PeerConnectionIntegrationIceStatesTestWithFakeClock,
|
|
IceStateSetupFailure) {
|
|
// Block connections to/from the caller and wait for ICE to become
|
|
// disconnected.
|
|
for (const auto& caller_address : CallerAddresses()) {
|
|
firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
|
|
}
|
|
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
SetPortAllocatorFlags();
|
|
SetUpNetworkInterfaces();
|
|
caller()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
|
|
// According to RFC7675, if there is no response within 30 seconds then the
|
|
// peer should consider the other side to have rejected the connection. This
|
|
// is signaled by the state transitioning to "failed".
|
|
constexpr int kConsentTimeout = 30000;
|
|
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
|
|
caller()->standardized_ice_connection_state(),
|
|
kConsentTimeout, FakeClock());
|
|
}
|
|
|
|
// Tests that the best connection is set to the appropriate IPv4/IPv6 connection
|
|
// and that the statistics in the metric observers are updated correctly.
|
|
TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
SetPortAllocatorFlags();
|
|
SetUpNetworkInterfaces();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kDefaultTimeout);
|
|
|
|
// TODO(bugs.webrtc.org/9456): Fix it.
|
|
const int num_best_ipv4 = webrtc::metrics::NumEvents(
|
|
"WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
|
|
const int num_best_ipv6 = webrtc::metrics::NumEvents(
|
|
"WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
|
|
if (TestIPv6()) {
|
|
// When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
|
|
// connection.
|
|
EXPECT_METRIC_EQ(0, num_best_ipv4);
|
|
EXPECT_METRIC_EQ(1, num_best_ipv6);
|
|
} else {
|
|
EXPECT_METRIC_EQ(1, num_best_ipv4);
|
|
EXPECT_METRIC_EQ(0, num_best_ipv6);
|
|
}
|
|
|
|
EXPECT_METRIC_EQ(0, webrtc::metrics::NumEvents(
|
|
"WebRTC.PeerConnection.CandidatePairType_UDP",
|
|
webrtc::kIceCandidatePairHostHost));
|
|
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
|
|
"WebRTC.PeerConnection.CandidatePairType_UDP",
|
|
webrtc::kIceCandidatePairHostPublicHostPublic));
|
|
}
|
|
|
|
constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
|
|
cricket::PORTALLOCATOR_DISABLE_STUN |
|
|
cricket::PORTALLOCATOR_DISABLE_RELAY;
|
|
constexpr uint32_t kFlagsIPv6NoStun =
|
|
cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_STUN |
|
|
cricket::PORTALLOCATOR_ENABLE_IPV6 | cricket::PORTALLOCATOR_DISABLE_RELAY;
|
|
constexpr uint32_t kFlagsIPv4Stun =
|
|
cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_RELAY;
|
|
|
|
INSTANTIATE_TEST_SUITE_P(
|
|
PeerConnectionIntegrationTest,
|
|
PeerConnectionIntegrationIceStatesTest,
|
|
Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
|
|
Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
|
|
std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
|
|
std::make_pair("IPv4 with STUN", kFlagsIPv4Stun))));
|
|
|
|
INSTANTIATE_TEST_SUITE_P(
|
|
PeerConnectionIntegrationTest,
|
|
PeerConnectionIntegrationIceStatesTestWithFakeClock,
|
|
Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
|
|
Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
|
|
std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
|
|
std::make_pair("IPv4 with STUN", kFlagsIPv4Stun))));
|
|
|
|
// This test sets up a call between two parties with audio and video.
|
|
// During the call, the caller restarts ICE and the test verifies that
|
|
// new ICE candidates are generated and audio and video still can flow, and the
|
|
// ICE state reaches completed again.
|
|
TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Do normal offer/answer and wait for ICE to complete.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
|
|
// To verify that the ICE restart actually occurs, get
|
|
// ufrag/password/candidates before and after restart.
|
|
// Create an SDP string of the first audio candidate for both clients.
|
|
const webrtc::IceCandidateCollection* audio_candidates_caller =
|
|
caller()->pc()->local_description()->candidates(0);
|
|
const webrtc::IceCandidateCollection* audio_candidates_callee =
|
|
callee()->pc()->local_description()->candidates(0);
|
|
ASSERT_GT(audio_candidates_caller->count(), 0u);
|
|
ASSERT_GT(audio_candidates_callee->count(), 0u);
|
|
std::string caller_candidate_pre_restart;
|
|
ASSERT_TRUE(
|
|
audio_candidates_caller->at(0)->ToString(&caller_candidate_pre_restart));
|
|
std::string callee_candidate_pre_restart;
|
|
ASSERT_TRUE(
|
|
audio_candidates_callee->at(0)->ToString(&callee_candidate_pre_restart));
|
|
const cricket::SessionDescription* desc =
|
|
caller()->pc()->local_description()->description();
|
|
std::string caller_ufrag_pre_restart =
|
|
desc->transport_infos()[0].description.ice_ufrag;
|
|
desc = callee()->pc()->local_description()->description();
|
|
std::string callee_ufrag_pre_restart =
|
|
desc->transport_infos()[0].description.ice_ufrag;
|
|
|
|
EXPECT_EQ(caller()->ice_candidate_pair_change_history().size(), 1u);
|
|
// Have the caller initiate an ICE restart.
|
|
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
|
|
// Grab the ufrags/candidates again.
|
|
audio_candidates_caller = caller()->pc()->local_description()->candidates(0);
|
|
audio_candidates_callee = callee()->pc()->local_description()->candidates(0);
|
|
ASSERT_GT(audio_candidates_caller->count(), 0u);
|
|
ASSERT_GT(audio_candidates_callee->count(), 0u);
|
|
std::string caller_candidate_post_restart;
|
|
ASSERT_TRUE(
|
|
audio_candidates_caller->at(0)->ToString(&caller_candidate_post_restart));
|
|
std::string callee_candidate_post_restart;
|
|
ASSERT_TRUE(
|
|
audio_candidates_callee->at(0)->ToString(&callee_candidate_post_restart));
|
|
desc = caller()->pc()->local_description()->description();
|
|
std::string caller_ufrag_post_restart =
|
|
desc->transport_infos()[0].description.ice_ufrag;
|
|
desc = callee()->pc()->local_description()->description();
|
|
std::string callee_ufrag_post_restart =
|
|
desc->transport_infos()[0].description.ice_ufrag;
|
|
// Sanity check that an ICE restart was actually negotiated in SDP.
|
|
ASSERT_NE(caller_candidate_pre_restart, caller_candidate_post_restart);
|
|
ASSERT_NE(callee_candidate_pre_restart, callee_candidate_post_restart);
|
|
ASSERT_NE(caller_ufrag_pre_restart, caller_ufrag_post_restart);
|
|
ASSERT_NE(callee_ufrag_pre_restart, callee_ufrag_post_restart);
|
|
EXPECT_GT(caller()->ice_candidate_pair_change_history().size(), 1u);
|
|
|
|
// Ensure that additional frames are received after the ICE restart.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Verify that audio/video can be received end-to-end when ICE renomination is
|
|
// enabled.
|
|
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
|
|
PeerConnectionInterface::RTCConfiguration config;
|
|
config.enable_ice_renomination = true;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
|
|
ConnectFakeSignaling();
|
|
// Do normal offer/answer and wait for some frames to be received in each
|
|
// direction.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Sanity check that ICE renomination was actually negotiated.
|
|
const cricket::SessionDescription* desc =
|
|
caller()->pc()->local_description()->description();
|
|
for (const cricket::TransportInfo& info : desc->transport_infos()) {
|
|
ASSERT_THAT(info.description.transport_options, Contains("renomination"));
|
|
}
|
|
desc = callee()->pc()->local_description()->description();
|
|
for (const cricket::TransportInfo& info : desc->transport_infos()) {
|
|
ASSERT_THAT(info.description.transport_options, Contains("renomination"));
|
|
}
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// With a max bundle policy and RTCP muxing, adding a new media description to
|
|
// the connection should not affect ICE at all because the new media will use
|
|
// the existing connection.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
AddMediaToConnectedBundleDoesNotRestartIce) {
|
|
PeerConnectionInterface::RTCConfiguration config;
|
|
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
|
|
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(
|
|
config, PeerConnectionInterface::RTCConfiguration()));
|
|
ConnectFakeSignaling();
|
|
|
|
caller()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kDefaultTimeout);
|
|
|
|
caller()->clear_ice_connection_state_history();
|
|
|
|
caller()->AddVideoTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
EXPECT_EQ(0u, caller()->ice_connection_state_history().size());
|
|
}
|
|
|
|
// This test sets up a call between two parties with audio and video. It then
|
|
// renegotiates setting the video m-line to "port 0", then later renegotiates
|
|
// again, enabling video.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
VideoFlowsAfterMediaSectionIsRejectedAndRecycled) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
// Do initial negotiation, only sending media from the caller. Will result in
|
|
// video and audio recvonly "m=" sections.
|
|
caller()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Negotiate again, disabling the video "m=" section (the callee will set the
|
|
// port to 0 due to offer_to_receive_video = 0).
|
|
if (sdp_semantics_ == SdpSemantics::kPlanB) {
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_video = 0;
|
|
callee()->SetOfferAnswerOptions(options);
|
|
} else {
|
|
callee()->SetRemoteOfferHandler([this] {
|
|
callee()
|
|
->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
|
|
->StopInternal();
|
|
});
|
|
}
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Sanity check that video "m=" section was actually rejected.
|
|
const ContentInfo* answer_video_content = cricket::GetFirstVideoContent(
|
|
callee()->pc()->local_description()->description());
|
|
ASSERT_NE(nullptr, answer_video_content);
|
|
ASSERT_TRUE(answer_video_content->rejected);
|
|
|
|
// Enable video and do negotiation again, making sure video is received
|
|
// end-to-end, also adding media stream to callee.
|
|
if (sdp_semantics_ == SdpSemantics::kPlanB) {
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_video = 1;
|
|
callee()->SetOfferAnswerOptions(options);
|
|
} else {
|
|
// The caller's transceiver is stopped, so we need to add another track.
|
|
auto caller_transceiver =
|
|
caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
|
|
EXPECT_EQ(nullptr, caller_transceiver.get());
|
|
caller()->AddVideoTrack();
|
|
}
|
|
callee()->AddVideoTrack();
|
|
callee()->SetRemoteOfferHandler(nullptr);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Verify the caller receives frames from the newly added stream, and the
|
|
// callee receives additional frames from the re-enabled video m= section.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio();
|
|
media_expectations.ExpectBidirectionalVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// This tests that if we negotiate after calling CreateSender but before we
|
|
// have a track, then set a track later, frames from the newly-set track are
|
|
// received end-to-end.
|
|
TEST_F(PeerConnectionIntegrationTestPlanB,
|
|
MediaFlowsAfterEarlyWarmupWithCreateSender) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
auto caller_audio_sender =
|
|
caller()->pc()->CreateSender("audio", "caller_stream");
|
|
auto caller_video_sender =
|
|
caller()->pc()->CreateSender("video", "caller_stream");
|
|
auto callee_audio_sender =
|
|
callee()->pc()->CreateSender("audio", "callee_stream");
|
|
auto callee_video_sender =
|
|
callee()->pc()->CreateSender("video", "callee_stream");
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
|
|
// Wait for ICE to complete, without any tracks being set.
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
// Now set the tracks, and expect frames to immediately start flowing.
|
|
EXPECT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
|
|
EXPECT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
|
|
EXPECT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
|
|
EXPECT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// This tests that if we negotiate after calling AddTransceiver but before we
|
|
// have a track, then set a track later, frames from the newly-set tracks are
|
|
// received end-to-end.
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
MediaFlowsAfterEarlyWarmupWithAddTransceiver) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
auto audio_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
|
|
ASSERT_EQ(RTCErrorType::NONE, audio_result.error().type());
|
|
auto caller_audio_sender = audio_result.MoveValue()->sender();
|
|
auto video_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
|
|
ASSERT_EQ(RTCErrorType::NONE, video_result.error().type());
|
|
auto caller_video_sender = video_result.MoveValue()->sender();
|
|
callee()->SetRemoteOfferHandler([this] {
|
|
ASSERT_EQ(2u, callee()->pc()->GetTransceivers().size());
|
|
callee()->pc()->GetTransceivers()[0]->SetDirectionWithError(
|
|
RtpTransceiverDirection::kSendRecv);
|
|
callee()->pc()->GetTransceivers()[1]->SetDirectionWithError(
|
|
RtpTransceiverDirection::kSendRecv);
|
|
});
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
|
|
// Wait for ICE to complete, without any tracks being set.
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
|
caller()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
// Now set the tracks, and expect frames to immediately start flowing.
|
|
auto callee_audio_sender = callee()->pc()->GetSenders()[0];
|
|
auto callee_video_sender = callee()->pc()->GetSenders()[1];
|
|
ASSERT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
|
|
ASSERT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
|
|
ASSERT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
|
|
ASSERT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// This test verifies that a remote video track can be added via AddStream,
|
|
// and sent end-to-end. For this particular test, it's simply echoed back
|
|
// from the caller to the callee, rather than being forwarded to a third
|
|
// PeerConnection.
|
|
TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
// Just send a video track from the caller.
|
|
caller()->AddVideoTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
|
|
ASSERT_EQ(1U, callee()->remote_streams()->count());
|
|
|
|
// Echo the stream back, and do a new offer/anwer (initiated by callee this
|
|
// time).
|
|
callee()->pc()->AddStream(callee()->remote_streams()->at(0));
|
|
callee()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Test that we achieve the expected end-to-end connection time, using a
|
|
// fake clock and simulated latency on the media and signaling paths.
|
|
// We use a TURN<->TURN connection because this is usually the quickest to
|
|
// set up initially, especially when we're confident the connection will work
|
|
// and can start sending media before we get a STUN response.
|
|
//
|
|
// With various optimizations enabled, here are the network delays we expect to
|
|
// be on the critical path:
|
|
// 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then
|
|
// signaling answer (with DTLS fingerprint).
|
|
// 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when
|
|
// using TURN<->TURN pair, and DTLS exchange is 4 packets,
|
|
// the first of which should have arrived before the answer.
|
|
TEST_P(PeerConnectionIntegrationTestWithFakeClock,
|
|
EndToEndConnectionTimeWithTurnTurnPair) {
|
|
static constexpr int media_hop_delay_ms = 50;
|
|
static constexpr int signaling_trip_delay_ms = 500;
|
|
// For explanation of these values, see comment above.
|
|
static constexpr int required_media_hops = 9;
|
|
static constexpr int required_signaling_trips = 2;
|
|
// For internal delays (such as posting an event asychronously).
|
|
static constexpr int allowed_internal_delay_ms = 20;
|
|
static constexpr int total_connection_time_ms =
|
|
media_hop_delay_ms * required_media_hops +
|
|
signaling_trip_delay_ms * required_signaling_trips +
|
|
allowed_internal_delay_ms;
|
|
|
|
static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
|
|
0};
|
|
static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
|
|
0};
|
|
cricket::TestTurnServer* turn_server_1 = CreateTurnServer(
|
|
turn_server_1_internal_address, turn_server_1_external_address);
|
|
|
|
cricket::TestTurnServer* turn_server_2 = CreateTurnServer(
|
|
turn_server_2_internal_address, turn_server_2_external_address);
|
|
// Bypass permission check on received packets so media can be sent before
|
|
// the candidate is signaled.
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_1] {
|
|
turn_server_1->set_enable_permission_checks(false);
|
|
});
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_2] {
|
|
turn_server_2->set_enable_permission_checks(false);
|
|
});
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_1_config;
|
|
webrtc::PeerConnectionInterface::IceServer ice_server_1;
|
|
ice_server_1.urls.push_back("turn:88.88.88.0:3478");
|
|
ice_server_1.username = "test";
|
|
ice_server_1.password = "test";
|
|
client_1_config.servers.push_back(ice_server_1);
|
|
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
client_1_config.presume_writable_when_fully_relayed = true;
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_2_config;
|
|
webrtc::PeerConnectionInterface::IceServer ice_server_2;
|
|
ice_server_2.urls.push_back("turn:99.99.99.0:3478");
|
|
ice_server_2.username = "test";
|
|
ice_server_2.password = "test";
|
|
client_2_config.servers.push_back(ice_server_2);
|
|
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
client_2_config.presume_writable_when_fully_relayed = true;
|
|
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
|
|
// Set up the simulated delays.
|
|
SetSignalingDelayMs(signaling_trip_delay_ms);
|
|
ConnectFakeSignaling();
|
|
virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
|
|
virtual_socket_server()->UpdateDelayDistribution();
|
|
|
|
// Set "offer to receive audio/video" without adding any tracks, so we just
|
|
// set up ICE/DTLS with no media.
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_audio = 1;
|
|
options.offer_to_receive_video = 1;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
EXPECT_TRUE_SIMULATED_WAIT(DtlsConnected(), total_connection_time_ms,
|
|
FakeClock());
|
|
// Closing the PeerConnections destroys the ports before the ScopedFakeClock.
|
|
// If this is not done a DCHECK can be hit in ports.cc, because a large
|
|
// negative number is calculated for the rtt due to the global clock changing.
|
|
ClosePeerConnections();
|
|
}
|
|
|
|
// Verify that a TurnCustomizer passed in through RTCConfiguration
|
|
// is actually used by the underlying TURN candidate pair.
|
|
// Note that turnport_unittest.cc contains more detailed, lower-level tests.
|
|
TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
|
|
static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
|
|
0};
|
|
static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
|
|
0};
|
|
CreateTurnServer(turn_server_1_internal_address,
|
|
turn_server_1_external_address);
|
|
CreateTurnServer(turn_server_2_internal_address,
|
|
turn_server_2_external_address);
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_1_config;
|
|
webrtc::PeerConnectionInterface::IceServer ice_server_1;
|
|
ice_server_1.urls.push_back("turn:88.88.88.0:3478");
|
|
ice_server_1.username = "test";
|
|
ice_server_1.password = "test";
|
|
client_1_config.servers.push_back(ice_server_1);
|
|
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
auto* customizer1 = CreateTurnCustomizer();
|
|
client_1_config.turn_customizer = customizer1;
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_2_config;
|
|
webrtc::PeerConnectionInterface::IceServer ice_server_2;
|
|
ice_server_2.urls.push_back("turn:99.99.99.0:3478");
|
|
ice_server_2.username = "test";
|
|
ice_server_2.password = "test";
|
|
client_2_config.servers.push_back(ice_server_2);
|
|
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
auto* customizer2 = CreateTurnCustomizer();
|
|
client_2_config.turn_customizer = customizer2;
|
|
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
|
|
ConnectFakeSignaling();
|
|
|
|
// Set "offer to receive audio/video" without adding any tracks, so we just
|
|
// set up ICE/DTLS with no media.
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_audio = 1;
|
|
options.offer_to_receive_video = 1;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
|
|
|
|
ExpectTurnCustomizerCountersIncremented(customizer1);
|
|
ExpectTurnCustomizerCountersIncremented(customizer2);
|
|
}
|
|
|
|
// Verifies that you can use TCP instead of UDP to connect to a TURN server and
|
|
// send media between the caller and the callee.
|
|
TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
|
|
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
|
|
|
|
// Enable TCP for the fake turn server.
|
|
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
|
|
cricket::PROTO_TCP);
|
|
|
|
webrtc::PeerConnectionInterface::IceServer ice_server;
|
|
ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
|
|
ice_server.username = "test";
|
|
ice_server.password = "test";
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_1_config;
|
|
client_1_config.servers.push_back(ice_server);
|
|
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_2_config;
|
|
client_2_config.servers.push_back(ice_server);
|
|
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
|
|
|
|
// Do normal offer/answer and wait for ICE to complete.
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kMaxWaitForFramesMs);
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
EXPECT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Verify that a SSLCertificateVerifier passed in through
|
|
// PeerConnectionDependencies is actually used by the underlying SSL
|
|
// implementation to determine whether a certificate presented by the TURN
|
|
// server is accepted by the client. Note that openssladapter_unittest.cc
|
|
// contains more detailed, lower-level tests.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
SSLCertificateVerifierUsedForTurnConnections) {
|
|
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
|
|
|
|
// Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
|
|
// that host name verification passes on the fake certificate.
|
|
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
|
|
cricket::PROTO_TLS, "88.88.88.0");
|
|
|
|
webrtc::PeerConnectionInterface::IceServer ice_server;
|
|
ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
|
|
ice_server.username = "test";
|
|
ice_server.password = "test";
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_1_config;
|
|
client_1_config.servers.push_back(ice_server);
|
|
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_2_config;
|
|
client_2_config.servers.push_back(ice_server);
|
|
// Setting the type to kRelay forces the connection to go through a TURN
|
|
// server.
|
|
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
|
|
// Get a copy to the pointer so we can verify calls later.
|
|
rtc::TestCertificateVerifier* client_1_cert_verifier =
|
|
new rtc::TestCertificateVerifier();
|
|
client_1_cert_verifier->verify_certificate_ = true;
|
|
rtc::TestCertificateVerifier* client_2_cert_verifier =
|
|
new rtc::TestCertificateVerifier();
|
|
client_2_cert_verifier->verify_certificate_ = true;
|
|
|
|
// Create the dependencies with the test certificate verifier.
|
|
webrtc::PeerConnectionDependencies client_1_deps(nullptr);
|
|
client_1_deps.tls_cert_verifier =
|
|
std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
|
|
webrtc::PeerConnectionDependencies client_2_deps(nullptr);
|
|
client_2_deps.tls_cert_verifier =
|
|
std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
|
|
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
|
|
client_1_config, std::move(client_1_deps), client_2_config,
|
|
std::move(client_2_deps)));
|
|
ConnectFakeSignaling();
|
|
|
|
// Set "offer to receive audio/video" without adding any tracks, so we just
|
|
// set up ICE/DTLS with no media.
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_audio = 1;
|
|
options.offer_to_receive_video = 1;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
|
|
|
|
EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
|
|
EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection) {
|
|
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
|
|
|
|
// Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
|
|
// that host name verification passes on the fake certificate.
|
|
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
|
|
cricket::PROTO_TLS, "88.88.88.0");
|
|
|
|
webrtc::PeerConnectionInterface::IceServer ice_server;
|
|
ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
|
|
ice_server.username = "test";
|
|
ice_server.password = "test";
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_1_config;
|
|
client_1_config.servers.push_back(ice_server);
|
|
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
|
|
PeerConnectionInterface::RTCConfiguration client_2_config;
|
|
client_2_config.servers.push_back(ice_server);
|
|
// Setting the type to kRelay forces the connection to go through a TURN
|
|
// server.
|
|
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
|
|
// Get a copy to the pointer so we can verify calls later.
|
|
rtc::TestCertificateVerifier* client_1_cert_verifier =
|
|
new rtc::TestCertificateVerifier();
|
|
client_1_cert_verifier->verify_certificate_ = false;
|
|
rtc::TestCertificateVerifier* client_2_cert_verifier =
|
|
new rtc::TestCertificateVerifier();
|
|
client_2_cert_verifier->verify_certificate_ = false;
|
|
|
|
// Create the dependencies with the test certificate verifier.
|
|
webrtc::PeerConnectionDependencies client_1_deps(nullptr);
|
|
client_1_deps.tls_cert_verifier =
|
|
std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
|
|
webrtc::PeerConnectionDependencies client_2_deps(nullptr);
|
|
client_2_deps.tls_cert_verifier =
|
|
std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
|
|
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
|
|
client_1_config, std::move(client_1_deps), client_2_config,
|
|
std::move(client_2_deps)));
|
|
ConnectFakeSignaling();
|
|
|
|
// Set "offer to receive audio/video" without adding any tracks, so we just
|
|
// set up ICE/DTLS with no media.
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_audio = 1;
|
|
options.offer_to_receive_video = 1;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
bool wait_res = true;
|
|
// TODO(bugs.webrtc.org/9219): When IceConnectionState is implemented
|
|
// properly, should be able to just wait for a state of "failed" instead of
|
|
// waiting a fixed 10 seconds.
|
|
WAIT_(DtlsConnected(), kDefaultTimeout, wait_res);
|
|
ASSERT_FALSE(wait_res);
|
|
|
|
EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
|
|
EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
|
|
}
|
|
|
|
// Test that the injected ICE transport factory is used to create ICE transports
|
|
// for WebRTC connections.
|
|
TEST_P(PeerConnectionIntegrationTest, IceTransportFactoryUsedForConnections) {
|
|
PeerConnectionInterface::RTCConfiguration default_config;
|
|
PeerConnectionDependencies dependencies(nullptr);
|
|
auto ice_transport_factory = std::make_unique<MockIceTransportFactory>();
|
|
EXPECT_CALL(*ice_transport_factory, RecordIceTransportCreated()).Times(1);
|
|
dependencies.ice_transport_factory = std::move(ice_transport_factory);
|
|
auto wrapper = CreatePeerConnectionWrapper("Caller", nullptr, &default_config,
|
|
std::move(dependencies), nullptr,
|
|
/*reset_encoder_factory=*/false,
|
|
/*reset_decoder_factory=*/false);
|
|
ASSERT_TRUE(wrapper);
|
|
wrapper->CreateDataChannel();
|
|
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
|
|
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
|
|
wrapper->pc()->SetLocalDescription(observer,
|
|
wrapper->CreateOfferAndWait().release());
|
|
}
|
|
|
|
// Test that audio and video flow end-to-end when codec names don't use the
|
|
// expected casing, given that they're supposed to be case insensitive. To test
|
|
// this, all but one codec is removed from each media description, and its
|
|
// casing is changed.
|
|
//
|
|
// In the past, this has regressed and caused crashes/black video, due to the
|
|
// fact that code at some layers was doing case-insensitive comparisons and
|
|
// code at other layers was not.
|
|
TEST_P(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
|
|
// Remove all but one audio/video codec (opus and VP8), and change the
|
|
// casing of the caller's generated offer.
|
|
caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) {
|
|
cricket::AudioContentDescription* audio =
|
|
GetFirstAudioContentDescription(description);
|
|
ASSERT_NE(nullptr, audio);
|
|
auto audio_codecs = audio->codecs();
|
|
audio_codecs.erase(std::remove_if(audio_codecs.begin(), audio_codecs.end(),
|
|
[](const cricket::AudioCodec& codec) {
|
|
return codec.name != "opus";
|
|
}),
|
|
audio_codecs.end());
|
|
ASSERT_EQ(1u, audio_codecs.size());
|
|
audio_codecs[0].name = "OpUs";
|
|
audio->set_codecs(audio_codecs);
|
|
|
|
cricket::VideoContentDescription* video =
|
|
GetFirstVideoContentDescription(description);
|
|
ASSERT_NE(nullptr, video);
|
|
auto video_codecs = video->codecs();
|
|
video_codecs.erase(std::remove_if(video_codecs.begin(), video_codecs.end(),
|
|
[](const cricket::VideoCodec& codec) {
|
|
return codec.name != "VP8";
|
|
}),
|
|
video_codecs.end());
|
|
ASSERT_EQ(1u, video_codecs.size());
|
|
video_codecs[0].name = "vP8";
|
|
video->set_codecs(video_codecs);
|
|
});
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Verify frames are still received end-to-end.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Wait for one audio frame to be received by the callee.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio(1);
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
|
|
auto receiver = callee()->pc()->GetReceivers()[0];
|
|
ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
|
|
auto sources = receiver->GetSources();
|
|
ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
|
|
EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
|
|
sources[0].source_id());
|
|
EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddVideoTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Wait for one video frame to be received by the callee.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeVideo(1);
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
|
|
auto receiver = callee()->pc()->GetReceivers()[0];
|
|
ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_VIDEO);
|
|
auto sources = receiver->GetSources();
|
|
ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
|
|
ASSERT_GT(sources.size(), 0u);
|
|
EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
|
|
sources[0].source_id());
|
|
EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
|
|
}
|
|
|
|
// Test that if a track is removed and added again with a different stream ID,
|
|
// the new stream ID is successfully communicated in SDP and media continues to
|
|
// flow end-to-end.
|
|
// TODO(webrtc.bugs.org/8734): This test does not work for Unified Plan because
|
|
// it will not reuse a transceiver that has already been sending. After creating
|
|
// a new transceiver it tries to create an offer with two senders of the same
|
|
// track ids and it fails.
|
|
TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
// Add track using stream 1, do offer/answer.
|
|
rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
|
|
caller()->CreateLocalAudioTrack();
|
|
rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
|
|
caller()->AddTrack(track, {"stream_1"});
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio(1);
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
// Remove the sender, and create a new one with the new stream.
|
|
caller()->pc()->RemoveTrack(sender);
|
|
sender = caller()->AddTrack(track, {"stream_2"});
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Wait for additional audio frames to be received by the callee.
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
auto output = std::make_unique<testing::NiceMock<MockRtcEventLogOutput>>();
|
|
ON_CALL(*output, IsActive()).WillByDefault(::testing::Return(true));
|
|
ON_CALL(*output, Write(::testing::_)).WillByDefault(::testing::Return(true));
|
|
EXPECT_CALL(*output, Write(::testing::_)).Times(::testing::AtLeast(1));
|
|
EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
|
|
std::move(output), webrtc::RtcEventLog::kImmediateOutput));
|
|
|
|
caller()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
}
|
|
|
|
// Test that if candidates are only signaled by applying full session
|
|
// descriptions (instead of using AddIceCandidate), the peers can connect to
|
|
// each other and exchange media.
|
|
TEST_P(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
// Each side will signal the session descriptions but not candidates.
|
|
ConnectFakeSignalingForSdpOnly();
|
|
|
|
// Add audio video track and exchange the initial offer/answer with media
|
|
// information only. This will start ICE gathering on each side.
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
|
|
// Wait for all candidates to be gathered on both the caller and callee.
|
|
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
|
|
caller()->ice_gathering_state(), kDefaultTimeout);
|
|
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
|
|
callee()->ice_gathering_state(), kDefaultTimeout);
|
|
|
|
// The candidates will now be included in the session description, so
|
|
// signaling them will start the ICE connection.
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Ensure that media flows in both directions.
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Test that SetAudioPlayout can be used to disable audio playout from the
|
|
// start, then later enable it. This may be useful, for example, if the caller
|
|
// needs to play a local ringtone until some event occurs, after which it
|
|
// switches to playing the received audio.
|
|
TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
// Set up audio-only call where audio playout is disabled on caller's side.
|
|
caller()->pc()->SetAudioPlayout(false);
|
|
caller()->AddAudioTrack();
|
|
callee()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Pump messages for a second.
|
|
WAIT(false, 1000);
|
|
// Since audio playout is disabled, the caller shouldn't have received
|
|
// anything (at the playout level, at least).
|
|
EXPECT_EQ(0, caller()->audio_frames_received());
|
|
// As a sanity check, make sure the callee (for which playout isn't disabled)
|
|
// did still see frames on its audio level.
|
|
ASSERT_GT(callee()->audio_frames_received(), 0);
|
|
|
|
// Enable playout again, and ensure audio starts flowing.
|
|
caller()->pc()->SetAudioPlayout(true);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
double GetAudioEnergyStat(PeerConnectionWrapper* pc) {
|
|
auto report = pc->NewGetStats();
|
|
auto track_stats_list =
|
|
report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
|
|
const webrtc::RTCMediaStreamTrackStats* remote_track_stats = nullptr;
|
|
for (const auto* track_stats : track_stats_list) {
|
|
if (track_stats->remote_source.is_defined() &&
|
|
*track_stats->remote_source) {
|
|
remote_track_stats = track_stats;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!remote_track_stats->total_audio_energy.is_defined()) {
|
|
return 0.0;
|
|
}
|
|
return *remote_track_stats->total_audio_energy;
|
|
}
|
|
|
|
// Test that if audio playout is disabled via the SetAudioPlayout() method, then
|
|
// incoming audio is still processed and statistics are generated.
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
DisableAudioPlayoutStillGeneratesAudioStats) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
// Set up audio-only call where playout is disabled but audio-processing is
|
|
// still active.
|
|
caller()->AddAudioTrack();
|
|
callee()->AddAudioTrack();
|
|
caller()->pc()->SetAudioPlayout(false);
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Wait for the callee to receive audio stats.
|
|
EXPECT_TRUE_WAIT(GetAudioEnergyStat(caller()) > 0, kMaxWaitForFramesMs);
|
|
}
|
|
|
|
// Test that SetAudioRecording can be used to disable audio recording from the
|
|
// start, then later enable it. This may be useful, for example, if the caller
|
|
// wants to ensure that no audio resources are active before a certain state
|
|
// is reached.
|
|
TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
|
|
// Set up audio-only call where audio recording is disabled on caller's side.
|
|
caller()->pc()->SetAudioRecording(false);
|
|
caller()->AddAudioTrack();
|
|
callee()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Pump messages for a second.
|
|
WAIT(false, 1000);
|
|
// Since caller has disabled audio recording, the callee shouldn't have
|
|
// received anything.
|
|
EXPECT_EQ(0, callee()->audio_frames_received());
|
|
// As a sanity check, make sure the caller did still see frames on its
|
|
// audio level since audio recording is enabled on the calle side.
|
|
ASSERT_GT(caller()->audio_frames_received(), 0);
|
|
|
|
// Enable audio recording again, and ensure audio starts flowing.
|
|
caller()->pc()->SetAudioRecording(true);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Test that after closing PeerConnections, they stop sending any packets (ICE,
|
|
// DTLS, RTP...).
|
|
TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
|
|
// Set up audio/video/data, wait for some frames to be received.
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
#ifdef HAVE_SCTP
|
|
caller()->CreateDataChannel();
|
|
#endif
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
// Close PeerConnections.
|
|
ClosePeerConnections();
|
|
// Pump messages for a second, and ensure no new packets end up sent.
|
|
uint32_t sent_packets_a = virtual_socket_server()->sent_packets();
|
|
WAIT(false, 1000);
|
|
uint32_t sent_packets_b = virtual_socket_server()->sent_packets();
|
|
EXPECT_EQ(sent_packets_a, sent_packets_b);
|
|
}
|
|
|
|
// Test that transport stats are generated by the RTCStatsCollector for a
|
|
// connection that only involves data channels. This is a regression test for
|
|
// crbug.com/826972.
|
|
#ifdef HAVE_SCTP
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
TransportStatsReportedForDataChannelOnlyConnection) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
|
|
|
|
auto caller_report = caller()->NewGetStats();
|
|
EXPECT_EQ(1u, caller_report->GetStatsOfType<RTCTransportStats>().size());
|
|
auto callee_report = callee()->NewGetStats();
|
|
EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
|
|
}
|
|
#endif // HAVE_SCTP
|
|
|
|
TEST_P(PeerConnectionIntegrationTest,
|
|
IceEventsGeneratedAndLoggedInRtcEventLog) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithFakeRtcEventLog());
|
|
ConnectFakeSignaling();
|
|
PeerConnectionInterface::RTCOfferAnswerOptions options;
|
|
options.offer_to_receive_audio = 1;
|
|
caller()->SetOfferAnswerOptions(options);
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
|
|
ASSERT_NE(nullptr, caller()->event_log_factory());
|
|
ASSERT_NE(nullptr, callee()->event_log_factory());
|
|
webrtc::FakeRtcEventLog* caller_event_log =
|
|
static_cast<webrtc::FakeRtcEventLog*>(
|
|
caller()->event_log_factory()->last_log_created());
|
|
webrtc::FakeRtcEventLog* callee_event_log =
|
|
static_cast<webrtc::FakeRtcEventLog*>(
|
|
callee()->event_log_factory()->last_log_created());
|
|
ASSERT_NE(nullptr, caller_event_log);
|
|
ASSERT_NE(nullptr, callee_event_log);
|
|
int caller_ice_config_count = caller_event_log->GetEventCount(
|
|
webrtc::RtcEvent::Type::IceCandidatePairConfig);
|
|
int caller_ice_event_count = caller_event_log->GetEventCount(
|
|
webrtc::RtcEvent::Type::IceCandidatePairEvent);
|
|
int callee_ice_config_count = callee_event_log->GetEventCount(
|
|
webrtc::RtcEvent::Type::IceCandidatePairConfig);
|
|
int callee_ice_event_count = callee_event_log->GetEventCount(
|
|
webrtc::RtcEvent::Type::IceCandidatePairEvent);
|
|
EXPECT_LT(0, caller_ice_config_count);
|
|
EXPECT_LT(0, caller_ice_event_count);
|
|
EXPECT_LT(0, callee_ice_config_count);
|
|
EXPECT_LT(0, callee_ice_event_count);
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) {
|
|
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
|
|
|
|
CreateTurnServer(turn_server_internal_address, turn_server_external_address);
|
|
|
|
webrtc::PeerConnectionInterface::IceServer ice_server;
|
|
ice_server.urls.push_back("turn:88.88.88.0:3478");
|
|
ice_server.username = "test";
|
|
ice_server.password = "test";
|
|
|
|
PeerConnectionInterface::RTCConfiguration caller_config;
|
|
caller_config.servers.push_back(ice_server);
|
|
caller_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
|
|
caller_config.surface_ice_candidates_on_ice_transport_type_changed = true;
|
|
|
|
PeerConnectionInterface::RTCConfiguration callee_config;
|
|
callee_config.servers.push_back(ice_server);
|
|
callee_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
|
|
callee_config.surface_ice_candidates_on_ice_transport_type_changed = true;
|
|
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));
|
|
|
|
// Do normal offer/answer and wait for ICE to complete.
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
// Since we are doing continual gathering, the ICE transport does not reach
|
|
// kIceGatheringComplete (see
|
|
// P2PTransportChannel::OnCandidatesAllocationDone), and consequently not
|
|
// kIceConnectionComplete.
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
caller()->ice_connection_state(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
|
callee()->ice_connection_state(), kDefaultTimeout);
|
|
// Note that we cannot use the metric
|
|
// |WebRTC.PeerConnection.CandidatePairType_UDP| in this test since this
|
|
// metric is only populated when we reach kIceConnectionComplete in the
|
|
// current implementation.
|
|
EXPECT_EQ(cricket::RELAY_PORT_TYPE,
|
|
caller()->last_candidate_gathered().type());
|
|
EXPECT_EQ(cricket::RELAY_PORT_TYPE,
|
|
callee()->last_candidate_gathered().type());
|
|
|
|
// Loosen the caller's candidate filter.
|
|
caller_config = caller()->pc()->GetConfiguration();
|
|
caller_config.type = webrtc::PeerConnectionInterface::kAll;
|
|
caller()->pc()->SetConfiguration(caller_config);
|
|
// We should have gathered a new host candidate.
|
|
EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
|
|
caller()->last_candidate_gathered().type(), kDefaultTimeout);
|
|
|
|
// Loosen the callee's candidate filter.
|
|
callee_config = callee()->pc()->GetConfiguration();
|
|
callee_config.type = webrtc::PeerConnectionInterface::kAll;
|
|
callee()->pc()->SetConfiguration(callee_config);
|
|
EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
|
|
callee()->last_candidate_gathered().type(), kDefaultTimeout);
|
|
|
|
// Create an offer and verify that it does not contain an ICE restart (i.e new
|
|
// ice credentials).
|
|
std::string caller_ufrag_pre_offer = caller()
|
|
->pc()
|
|
->local_description()
|
|
->description()
|
|
->transport_infos()[0]
|
|
.description.ice_ufrag;
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
std::string caller_ufrag_post_offer = caller()
|
|
->pc()
|
|
->local_description()
|
|
->description()
|
|
->transport_infos()[0]
|
|
.description.ice_ufrag;
|
|
EXPECT_EQ(caller_ufrag_pre_offer, caller_ufrag_post_offer);
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, OnIceCandidateError) {
|
|
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
|
|
3478};
|
|
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
|
|
|
|
CreateTurnServer(turn_server_internal_address, turn_server_external_address);
|
|
|
|
webrtc::PeerConnectionInterface::IceServer ice_server;
|
|
ice_server.urls.push_back("turn:88.88.88.0:3478");
|
|
ice_server.username = "test";
|
|
ice_server.password = "123";
|
|
|
|
PeerConnectionInterface::RTCConfiguration caller_config;
|
|
caller_config.servers.push_back(ice_server);
|
|
caller_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
|
|
|
|
PeerConnectionInterface::RTCConfiguration callee_config;
|
|
callee_config.servers.push_back(ice_server);
|
|
callee_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
|
|
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));
|
|
|
|
// Do normal offer/answer and wait for ICE to complete.
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(401, caller()->error_event().error_code, kDefaultTimeout);
|
|
EXPECT_EQ("Unauthorized", caller()->error_event().error_text);
|
|
EXPECT_EQ("turn:88.88.88.0:3478?transport=udp", caller()->error_event().url);
|
|
EXPECT_NE(caller()->error_event().address, "");
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationTest, OnIceCandidateErrorWithEmptyAddress) {
|
|
webrtc::PeerConnectionInterface::IceServer ice_server;
|
|
ice_server.urls.push_back("turn:127.0.0.1:3478?transport=tcp");
|
|
ice_server.username = "test";
|
|
ice_server.password = "test";
|
|
|
|
PeerConnectionInterface::RTCConfiguration caller_config;
|
|
caller_config.servers.push_back(ice_server);
|
|
caller_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
|
|
|
|
PeerConnectionInterface::RTCConfiguration callee_config;
|
|
callee_config.servers.push_back(ice_server);
|
|
callee_config.type = webrtc::PeerConnectionInterface::kRelay;
|
|
callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
|
|
|
|
ASSERT_TRUE(
|
|
CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));
|
|
|
|
// Do normal offer/answer and wait for ICE to complete.
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
EXPECT_EQ_WAIT(701, caller()->error_event().error_code, kDefaultTimeout);
|
|
EXPECT_EQ(caller()->error_event().address, "");
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
AudioKeepsFlowingAfterImplicitRollback) {
|
|
PeerConnectionInterface::RTCConfiguration config;
|
|
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
|
|
config.enable_implicit_rollback = true;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
callee()->AddAudioTrack();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
SetSignalIceCandidates(false); // Workaround candidate outrace sdp.
|
|
caller()->AddVideoTrack();
|
|
callee()->AddVideoTrack();
|
|
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
|
|
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
|
|
callee()->pc()->SetLocalDescription(observer,
|
|
callee()->CreateOfferAndWait().release());
|
|
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
|
|
caller()->CreateAndSetAndSignalOffer(); // Implicit rollback.
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
ImplicitRollbackVisitsStableState) {
|
|
RTCConfiguration config;
|
|
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
|
|
config.enable_implicit_rollback = true;
|
|
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
|
|
|
|
rtc::scoped_refptr<MockSetSessionDescriptionObserver> sld_observer(
|
|
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
|
|
callee()->pc()->SetLocalDescription(sld_observer,
|
|
callee()->CreateOfferAndWait().release());
|
|
EXPECT_TRUE_WAIT(sld_observer->called(), kDefaultTimeout);
|
|
EXPECT_EQ(sld_observer->error(), "");
|
|
|
|
rtc::scoped_refptr<MockSetSessionDescriptionObserver> srd_observer(
|
|
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
|
|
callee()->pc()->SetRemoteDescription(
|
|
srd_observer, caller()->CreateOfferAndWait().release());
|
|
EXPECT_TRUE_WAIT(srd_observer->called(), kDefaultTimeout);
|
|
EXPECT_EQ(srd_observer->error(), "");
|
|
|
|
EXPECT_THAT(callee()->peer_connection_signaling_state_history(),
|
|
ElementsAre(PeerConnectionInterface::kHaveLocalOffer,
|
|
PeerConnectionInterface::kStable,
|
|
PeerConnectionInterface::kHaveRemoteOffer));
|
|
}
|
|
|
|
INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest,
|
|
PeerConnectionIntegrationTest,
|
|
Values(SdpSemantics::kPlanB,
|
|
SdpSemantics::kUnifiedPlan));
|
|
|
|
INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest,
|
|
PeerConnectionIntegrationTestWithFakeClock,
|
|
Values(SdpSemantics::kPlanB,
|
|
SdpSemantics::kUnifiedPlan));
|
|
|
|
// Tests that verify interoperability between Plan B and Unified Plan
|
|
// PeerConnections.
|
|
class PeerConnectionIntegrationInteropTest
|
|
: public PeerConnectionIntegrationBaseTest,
|
|
public ::testing::WithParamInterface<
|
|
std::tuple<SdpSemantics, SdpSemantics>> {
|
|
protected:
|
|
// Setting the SdpSemantics for the base test to kDefault does not matter
|
|
// because we specify not to use the test semantics when creating
|
|
// PeerConnectionWrappers.
|
|
PeerConnectionIntegrationInteropTest()
|
|
: PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB),
|
|
caller_semantics_(std::get<0>(GetParam())),
|
|
callee_semantics_(std::get<1>(GetParam())) {}
|
|
|
|
bool CreatePeerConnectionWrappersWithSemantics() {
|
|
return CreatePeerConnectionWrappersWithSdpSemantics(caller_semantics_,
|
|
callee_semantics_);
|
|
}
|
|
|
|
const SdpSemantics caller_semantics_;
|
|
const SdpSemantics callee_semantics_;
|
|
};
|
|
|
|
TEST_P(PeerConnectionIntegrationInteropTest, NoMediaLocalToNoMediaRemote) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
|
|
ConnectFakeSignaling();
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationInteropTest, OneAudioLocalToNoMediaRemote) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
|
|
ConnectFakeSignaling();
|
|
auto audio_sender = caller()->AddAudioTrack();
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Verify that one audio receiver has been created on the remote and that it
|
|
// has the same track ID as the sending track.
|
|
auto receivers = callee()->pc()->GetReceivers();
|
|
ASSERT_EQ(1u, receivers.size());
|
|
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, receivers[0]->media_type());
|
|
EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id());
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationInteropTest, OneAudioOneVideoToNoMediaRemote) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
|
|
ConnectFakeSignaling();
|
|
auto video_sender = caller()->AddVideoTrack();
|
|
auto audio_sender = caller()->AddAudioTrack();
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Verify that one audio and one video receiver have been created on the
|
|
// remote and that they have the same track IDs as the sending tracks.
|
|
auto audio_receivers =
|
|
callee()->GetReceiversOfType(cricket::MEDIA_TYPE_AUDIO);
|
|
ASSERT_EQ(1u, audio_receivers.size());
|
|
EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id());
|
|
auto video_receivers =
|
|
callee()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO);
|
|
ASSERT_EQ(1u, video_receivers.size());
|
|
EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id());
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationInteropTest,
|
|
OneAudioOneVideoLocalToOneAudioOneVideoRemote) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.ExpectBidirectionalAudioAndVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
TEST_P(PeerConnectionIntegrationInteropTest,
|
|
ReverseRolesOneAudioLocalToOneVideoRemote) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
|
|
ConnectFakeSignaling();
|
|
caller()->AddAudioTrack();
|
|
callee()->AddVideoTrack();
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Verify that only the audio track has been negotiated.
|
|
EXPECT_EQ(0u, caller()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO).size());
|
|
// Might also check that the callee's NegotiationNeeded flag is set.
|
|
|
|
// Reverse roles.
|
|
callee()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CallerExpectsSomeVideo();
|
|
media_expectations.CalleeExpectsSomeAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
INSTANTIATE_TEST_SUITE_P(
|
|
PeerConnectionIntegrationTest,
|
|
PeerConnectionIntegrationInteropTest,
|
|
Values(std::make_tuple(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
|
|
std::make_tuple(SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB)));
|
|
|
|
// Test that if the Unified Plan side offers two video tracks then the Plan B
|
|
// side will only see the first one and ignore the second.
|
|
TEST_F(PeerConnectionIntegrationTestPlanB, TwoVideoUnifiedPlanToNoMediaPlanB) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithSdpSemantics(
|
|
SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB));
|
|
ConnectFakeSignaling();
|
|
auto first_sender = caller()->AddVideoTrack();
|
|
caller()->AddVideoTrack();
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
|
|
// Verify that there is only one receiver and it corresponds to the first
|
|
// added track.
|
|
auto receivers = callee()->pc()->GetReceivers();
|
|
ASSERT_EQ(1u, receivers.size());
|
|
EXPECT_TRUE(receivers[0]->track()->enabled());
|
|
EXPECT_EQ(first_sender->track()->id(), receivers[0]->track()->id());
|
|
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
// Test that if the initial offer tagged BUNDLE section is rejected due to its
|
|
// associated RtpTransceiver being stopped and another transceiver is added,
|
|
// then renegotiation causes the callee to receive the new video track without
|
|
// error.
|
|
// This is a regression test for bugs.webrtc.org/9954
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
ReOfferWithStoppedBundleTaggedTransceiver) {
|
|
RTCConfiguration config;
|
|
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
|
|
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
|
|
ConnectFakeSignaling();
|
|
auto audio_transceiver_or_error =
|
|
caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
|
|
ASSERT_TRUE(audio_transceiver_or_error.ok());
|
|
auto audio_transceiver = audio_transceiver_or_error.MoveValue();
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeAudio();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
|
|
audio_transceiver->StopInternal();
|
|
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack());
|
|
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
{
|
|
MediaExpectations media_expectations;
|
|
media_expectations.CalleeExpectsSomeVideo();
|
|
ASSERT_TRUE(ExpectNewFrames(media_expectations));
|
|
}
|
|
}
|
|
|
|
#ifdef HAVE_SCTP
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
EndToEndCallWithBundledSctpDataChannel) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->AddAudioVideoTracks();
|
|
callee()->AddAudioVideoTracks();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_EQ_WAIT(SctpTransportState::kConnected,
|
|
caller()->pc()->GetSctpTransport()->Information().state(),
|
|
kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
EndToEndCallWithDataChannelOnlyConnects) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
ASSERT_TRUE(caller()->data_observer()->IsOpen());
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan, DataChannelClosesWhenClosed) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
caller()->data_channel()->Close();
|
|
ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
DataChannelClosesWhenClosedReverse) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
callee()->data_channel()->Close();
|
|
ASSERT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
}
|
|
|
|
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
|
|
DataChannelClosesWhenPeerConnectionClosed) {
|
|
ASSERT_TRUE(CreatePeerConnectionWrappers());
|
|
ConnectFakeSignaling();
|
|
caller()->CreateDataChannel();
|
|
caller()->CreateAndSetAndSignalOffer();
|
|
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
|
|
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
caller()->pc()->Close();
|
|
ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
|
|
}
|
|
|
|
#endif // HAVE_SCTP
|
|
|
|
} // namespace
|
|
} // namespace webrtc
|
|
|
|
#endif // if !defined(THREAD_SANITIZER)
|