blob: b2401022c46a557376041044c33fcdf4c8200d3c [file] [log] [blame]
/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <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;