| /* |
| * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <stdint.h> |
| |
| #include <algorithm> |
| #include <memory> |
| #include <string> |
| #include <tuple> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/types/optional.h" |
| #include "api/async_resolver_factory.h" |
| #include "api/candidate.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/dtmf_sender_interface.h" |
| #include "api/ice_transport_interface.h" |
| #include "api/jsep.h" |
| #include "api/media_stream_interface.h" |
| #include "api/media_types.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/rtc_error.h" |
| #include "api/rtc_event_log/rtc_event.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/rtc_event_log_output.h" |
| #include "api/rtp_parameters.h" |
| #include "api/rtp_receiver_interface.h" |
| #include "api/rtp_sender_interface.h" |
| #include "api/rtp_transceiver_direction.h" |
| #include "api/rtp_transceiver_interface.h" |
| #include "api/scoped_refptr.h" |
| #include "api/stats/rtc_stats.h" |
| #include "api/stats/rtc_stats_report.h" |
| #include "api/stats/rtcstats_objects.h" |
| #include "api/transport/rtp/rtp_source.h" |
| #include "api/uma_metrics.h" |
| #include "api/units/time_delta.h" |
| #include "api/video/video_rotation.h" |
| #include "logging/rtc_event_log/fake_rtc_event_log.h" |
| #include "logging/rtc_event_log/fake_rtc_event_log_factory.h" |
| #include "media/base/codec.h" |
| #include "media/base/media_constants.h" |
| #include "media/base/stream_params.h" |
| #include "p2p/base/mock_async_resolver.h" |
| #include "p2p/base/port.h" |
| #include "p2p/base/port_allocator.h" |
| #include "p2p/base/port_interface.h" |
| #include "p2p/base/stun_server.h" |
| #include "p2p/base/test_stun_server.h" |
| #include "p2p/base/test_turn_customizer.h" |
| #include "p2p/base/test_turn_server.h" |
| #include "p2p/base/transport_description.h" |
| #include "p2p/base/transport_info.h" |
| #include "pc/media_session.h" |
| #include "pc/peer_connection.h" |
| #include "pc/peer_connection_factory.h" |
| #include "pc/session_description.h" |
| #include "pc/test/fake_periodic_video_source.h" |
| #include "pc/test/integration_test_helpers.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "rtc_base/fake_clock.h" |
| #include "rtc_base/fake_mdns_responder.h" |
| #include "rtc_base/fake_network.h" |
| #include "rtc_base/firewall_socket_server.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/helpers.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/socket_address.h" |
| #include "rtc_base/ssl_certificate.h" |
| #include "rtc_base/ssl_fingerprint.h" |
| #include "rtc_base/ssl_identity.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/test_certificate_verifier.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/time_utils.h" |
| #include "rtc_base/virtual_socket_server.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| 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(PeerConnectionIntegrationWrapper* sender, |
| PeerConnectionIntegrationWrapper* 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)); |
| } |
| |
| // 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<PeerConnectionIntegrationWrapper> 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<PeerConnectionIntegrationWrapper> 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)); |
| } |
| |
| // Used for the test below. |
| void RemoveBundleGroupSsrcsAndMidExtension(cricket::SessionDescription* desc) { |
| RemoveSsrcsAndKeepMsids(desc); |
| desc->RemoveGroupByName("BUNDLE"); |
| for (ContentInfo& content : desc->contents()) { |
| cricket::MediaContentDescription* media = content.media_description(); |
| cricket::RtpHeaderExtensions extensions = media->rtp_header_extensions(); |
| extensions.erase(std::remove_if(extensions.begin(), extensions.end(), |
| [](const RtpExtension& extension) { |
| return extension.uri == |
| RtpExtension::kMidUri; |
| }), |
| extensions.end()); |
| media->set_rtp_header_extensions(extensions); |
| } |
| } |
| |
| // Tests that video flows between multiple video tracks when BUNDLE is not used, |
| // SSRCs are not signaled and the MID RTP header extension is not used. This |
| // relies on demuxing by payload type, which normally doesn't work if you have |
| // multiple media sections using the same payload type, but which should work as |
| // long as the media sections aren't bundled. |
| // Regression test for: http://crbug.com/webrtc/12023 |
| TEST_F(PeerConnectionIntegrationTestUnifiedPlan, |
| EndToEndCallWithTwoVideoTracksNoBundleNoSignaledSsrcAndNoMid) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddVideoTrack(); |
| caller()->AddVideoTrack(); |
| callee()->AddVideoTrack(); |
| callee()->AddVideoTrack(); |
| caller()->SetReceivedSdpMunger(&RemoveBundleGroupSsrcsAndMidExtension); |
| callee()->SetReceivedSdpMunger(&RemoveBundleGroupSsrcsAndMidExtension); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| ASSERT_EQ(2u, caller()->pc()->GetReceivers().size()); |
| ASSERT_EQ(2u, callee()->pc()->GetReceivers().size()); |
| // Make sure we are not bundled. |
| ASSERT_NE(caller()->pc()->GetSenders()[0]->dtls_transport(), |
| caller()->pc()->GetSenders()[1]->dtls_transport()); |
| |
| // Expect video to be received in both directions on both tracks. |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalVideo(); |
| EXPECT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // Used for the test below. |
| void ModifyPayloadTypesAndRemoveMidExtension( |
| cricket::SessionDescription* desc) { |
| int pt = 96; |
| for (ContentInfo& content : desc->contents()) { |
| cricket::MediaContentDescription* media = content.media_description(); |
| cricket::RtpHeaderExtensions extensions = media->rtp_header_extensions(); |
| extensions.erase(std::remove_if(extensions.begin(), extensions.end(), |
| [](const RtpExtension& extension) { |
| return extension.uri == |
| RtpExtension::kMidUri; |
| }), |
| extensions.end()); |
| media->set_rtp_header_extensions(extensions); |
| cricket::VideoContentDescription* video = media->as_video(); |
| ASSERT_TRUE(video != nullptr); |
| std::vector<cricket::VideoCodec> codecs = {{pt++, "VP8"}}; |
| video->set_codecs(codecs); |
| } |
| } |
| |
| // Tests that two video tracks can be demultiplexed by payload type alone, by |
| // using different payload types for the same codec in different m= sections. |
| // This practice is discouraged but historically has been supported. |
| // Regression test for: http://crbug.com/webrtc/12029 |
| TEST_F(PeerConnectionIntegrationTestUnifiedPlan, |
| EndToEndCallWithTwoVideoTracksDemultiplexedByPayloadType) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddVideoTrack(); |
| caller()->AddVideoTrack(); |
| callee()->AddVideoTrack(); |
| callee()->AddVideoTrack(); |
| caller()->SetGeneratedSdpMunger(&ModifyPayloadTypesAndRemoveMidExtension); |
| callee()->SetGeneratedSdpMunger(&ModifyPayloadTypesAndRemoveMidExtension); |
| // We can't remove SSRCs from the generated SDP because then no send streams |
| // would be created. |
| 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()); |
| // Make sure we are bundled. |
| ASSERT_EQ(caller()->pc()->GetSenders()[0]->dtls_transport(), |
| caller()->pc()->GetSenders()[1]->dtls_transport()); |
| |
| // 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)); |
| } |
| |
| // 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); |
| } |
| |
| #if !defined(THREAD_SANITIZER) |
| // This test provokes TSAN errors. See bugs.webrtc.org/3608 |
| |
| 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. |
| // TODO(bugs.webrtc.org/12590): Flaky on Windows and on Linux MSAN. |
| #if defined(WEBRTC_WIN) || defined(WEBRTC_LINUX) |
| #define MAYBE_IceStatesReachCompletionWithRemoteHostname \ |
| DISABLED_IceStatesReachCompletionWithRemoteHostname |
| #else |
| #define MAYBE_IceStatesReachCompletionWithRemoteHostname \ |
| IceStatesReachCompletionWithRemoteHostname |
| #endif |
| TEST_P(PeerConnectionIntegrationTest, |
| MAYBE_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)); |
| } |
| |
| #endif // !defined(THREAD_SANITIZER) |
| |
| // 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(firewall(), 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 {}; |
| |
| #if !defined(THREAD_SANITIZER) |
| // This test provokes TSAN errors. bugs.webrtc.org/11282 |
| |
| // 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()); |
| } |
| |
| #endif // !defined(THREAD_SANITIZER) |
| |
| // Tests that the best connection is set to the appropriate IPv4/IPv6 connection |
| // and that the statistics in the metric observers are updated correctly. |
| // TODO(bugs.webrtc.org/12591): Flaky on Windows. |
| #if defined(WEBRTC_WIN) |
| #define MAYBE_VerifyBestConnection DISABLED_VerifyBestConnection |
| #else |
| #define MAYBE_VerifyBestConnection VerifyBestConnection |
| #endif |
| TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_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. |
| // TODO(bugs.webrtc.org/12538): Fails on tsan. |
| #if defined(THREAD_SANITIZER) |
| #define MAYBE_AddMediaToConnectedBundleDoesNotRestartIce \ |
| DISABLED_AddMediaToConnectedBundleDoesNotRestartIce |
| #else |
| #define MAYBE_AddMediaToConnectedBundleDoesNotRestartIce \ |
| AddMediaToConnectedBundleDoesNotRestartIce |
| #endif |
| TEST_P(PeerConnectionIntegrationTest, |
| MAYBE_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)); |
| } |
| |
| #if !defined(THREAD_SANITIZER) |
| // This test provokes TSAN errors. bugs.webrtc.org/11282 |
| |
| // 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(); |
| } |
| |
| #endif // !defined(THREAD_SANITIZER) |
| |
| // 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(); |
| auto observer = rtc::make_ref_counted<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(); |
| |
|