Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

webrtc/ortc/ortcfactory_integrationtest.cc

+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <utility>  // For std::pair, std::move.
+
+#include "webrtc/api/ortc/ortcfactoryinterface.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/fakenetwork.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/physicalsocketserver.h"
+#include "webrtc/base/virtualsocketserver.h"
+#include "webrtc/ortc/testrtpparameters.h"
+#include "webrtc/p2p/base/udptransport.h"
+#include "webrtc/pc/test/fakeaudiocapturemodule.h"
+#include "webrtc/pc/test/fakeperiodicvideocapturer.h"
+#include "webrtc/pc/test/fakevideotrackrenderer.h"
+
+namespace {
+
+const int kDefaultTimeout = 10000;  // 10 seconds.
+// Default number of audio/video frames to wait for before considering a test a
+// success.
+const int kDefaultNumFrames = 3;
+const rtc::IPAddress kIPv4LocalHostAddress =
+    rtc::IPAddress(0x7F000001);  // 127.0.0.1
+
+}  // namespace
+
+namespace webrtc {
+
+// Used to test that things work end-to-end when using the default
+// implementations of threads/etc. provided by OrtcFactory, with the exception
+// of using a virtual network.
+//
+// By default, the virtual network manager doesn't enumerate any networks, but
+// sockets can still be created in this state.
+class OrtcFactoryIntegrationTest : public testing::Test {
+ public:
+  OrtcFactoryIntegrationTest()
+      : virtual_socket_server_(&physical_socket_server_),
+        network_thread_(&virtual_socket_server_),
+        fake_audio_capture_module1_(FakeAudioCaptureModule::Create()),
+        fake_audio_capture_module2_(FakeAudioCaptureModule::Create()) {
+    // Sockets are bound to the ANY address, so this is needed to tell the
+    // virtual network which address to use in this case.
+    virtual_socket_server_.SetDefaultRoute(kIPv4LocalHostAddress);
+    network_thread_.Start();
+    // Need to create after network thread is started.
+    ortc_factory1_ = OrtcFactoryInterface::Create(
+                         &network_thread_, nullptr, &fake_network_manager_,
+                         nullptr, fake_audio_capture_module1_)
+                         .MoveValue();
+    ortc_factory2_ = OrtcFactoryInterface::Create(
+                         &network_thread_, nullptr, &fake_network_manager_,
+                         nullptr, fake_audio_capture_module2_)
+                         .MoveValue();
+  }
+
+ protected:
+  typedef std::pair<std::unique_ptr<UdpTransportInterface>,
+                    std::unique_ptr<UdpTransportInterface>>
+      UdpTransportPair;
+  typedef std::pair<std::unique_ptr<RtpTransportInterface>,
+                    std::unique_ptr<RtpTransportInterface>>
+      RtpTransportPair;
+  typedef std::pair<std::unique_ptr<RtpTransportControllerInterface>,
+                    std::unique_ptr<RtpTransportControllerInterface>>
+      RtpTransportControllerPair;
+
+  // Helper function that creates one UDP transport each for |ortc_factory1_|
+  // and |ortc_factory2_|, and connects them.
+  UdpTransportPair CreateAndConnectUdpTransportPair() {
+    auto transport1 = ortc_factory1_->CreateUdpTransport(AF_INET).MoveValue();
+    auto transport2 = ortc_factory2_->CreateUdpTransport(AF_INET).MoveValue();
+    transport1->SetRemoteAddress(
+        rtc::SocketAddress(virtual_socket_server_.GetDefaultRoute(AF_INET),
+                           transport2->GetLocalAddress().port()));
+    transport2->SetRemoteAddress(
+        rtc::SocketAddress(virtual_socket_server_.GetDefaultRoute(AF_INET),
+                           transport1->GetLocalAddress().port()));
+    return {std::move(transport1), std::move(transport2)};
+  }
+
+  // Creates one transport controller each for |ortc_factory1_| and
+  // |ortc_factory2_|.
+  RtpTransportControllerPair CreateRtpTransportControllerPair() {
+    return {ortc_factory1_->CreateRtpTransportController().MoveValue(),
+            ortc_factory2_->CreateRtpTransportController().MoveValue()};
+  }
+
+  // Helper function that creates a pair of RtpTransports between
+  // |ortc_factory1_| and |ortc_factory2_|. Expected to be called with the
+  // result of CreateAndConnectUdpTransportPair. |rtcp_udp_transports| can be
+  // empty if RTCP muxing is used. |transport_controllers| can be empty if
+  // these transports are being created using a default transport controller.
+  RtpTransportPair CreateRtpTransportPair(
+      const RtcpParameters& rtcp_parameters,
+      const UdpTransportPair& rtp_udp_transports,
+      const UdpTransportPair& rtcp_udp_transports,
+      const RtpTransportControllerPair& transport_controllers) {
+    auto transport_result1 = ortc_factory1_->CreateRtpTransport(
+        rtcp_parameters, rtp_udp_transports.first.get(),
+        rtcp_udp_transports.first.get(), transport_controllers.first.get());
+    auto transport_result2 = ortc_factory2_->CreateRtpTransport(
+        rtcp_parameters, rtp_udp_transports.second.get(),
+        rtcp_udp_transports.second.get(), transport_controllers.second.get());
+    return {transport_result1.MoveValue(), transport_result2.MoveValue()};
+  }
+
+  // For convenience when |rtcp_udp_transports| and |transport_controllers|
+  // aren't needed.
+  RtpTransportPair CreateRtpTransportPair(
+      const RtcpParameters& rtcp_parameters,
+      const UdpTransportPair& rtp_udp_transports) {
+    return CreateRtpTransportPair(rtcp_parameters, rtp_udp_transports,
+                                  UdpTransportPair(),
+                                  RtpTransportControllerPair());
+  }
+
+  // Ends up using fake audio capture module, which was passed into OrtcFactory
+  // on creation.
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
+      const std::string& id,
+      OrtcFactoryInterface* ortc_factory) {
+    // Disable echo cancellation to make test more efficient.
+    cricket::AudioOptions options;
+    options.echo_cancellation.emplace(true);
+    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+        ortc_factory->CreateAudioSource(options);
+    return ortc_factory->CreateAudioTrack(id, source);
+  }
+
+  // Stores created capturer in |fake_video_capturers_|.
+  rtc::scoped_refptr<webrtc::VideoTrackInterface>
+  CreateLocalVideoTrackAndFakeCapturer(const std::string& id,
+                                       OrtcFactoryInterface* ortc_factory) {
+    cricket::FakeVideoCapturer* fake_capturer =
+        new webrtc::FakePeriodicVideoCapturer();
+    fake_video_capturers_.push_back(fake_capturer);
+    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
+        ortc_factory->CreateVideoSource(
+            std::unique_ptr<cricket::VideoCapturer>(fake_capturer));
+    return rtc::scoped_refptr<webrtc::VideoTrackInterface>(
+        ortc_factory->CreateVideoTrack(id, source));
+  }
+
+  rtc::PhysicalSocketServer physical_socket_server_;
+  rtc::VirtualSocketServer virtual_socket_server_;
+  rtc::Thread network_thread_;
+  rtc::FakeNetworkManager fake_network_manager_;
+  rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module1_;
+  rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module2_;
+  std::unique_ptr<OrtcFactoryInterface> ortc_factory1_;
+  std::unique_ptr<OrtcFactoryInterface> ortc_factory2_;
+  // Actually owned by video tracks.
+  std::vector<cricket::FakeVideoCapturer*> fake_video_capturers_;
+};
+
+// Very basic end-to-end test with a single pair of audio RTP sender and
+// receiver.
+//
+// Uses muxed RTCP, and minimal parameters with a hard-coded config that's
+// known to work.
+TEST_F(OrtcFactoryIntegrationTest, BasicOneWayAudioRtpSenderAndReceiver) {
+  auto udp_transports = CreateAndConnectUdpTransportPair();
+  auto rtp_transports =
+      CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
+
+  auto sender_result = ortc_factory1_->CreateRtpSender(
+      cricket::MEDIA_TYPE_AUDIO, rtp_transports.first.get());
+  auto receiver_result = ortc_factory2_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_AUDIO, rtp_transports.second.get());
+  ASSERT_TRUE(sender_result.ok());
+  ASSERT_TRUE(receiver_result.ok());
+  auto sender = sender_result.MoveValue();
+  auto receiver = receiver_result.MoveValue();
+
+  RTCError error =
+      sender->SetTrack(CreateLocalAudioTrack("audio", ortc_factory1_.get()));
+  EXPECT_TRUE(error.ok());
+
+  RtpParameters opus_parameters = MakeMinimalOpusParameters();
+  EXPECT_TRUE(receiver->Receive(opus_parameters).ok());
+  EXPECT_TRUE(sender->Send(opus_parameters).ok());
+  // Sender and receiver are connected and configured; audio frames should be
+  // able to flow at this point.
+  EXPECT_TRUE_WAIT(
+      fake_audio_capture_module2_->frames_received() > kDefaultNumFrames,
+      kDefaultTimeout);
+}
+
+// Very basic end-to-end test with a single pair of video RTP sender and
+// receiver.
+//
+// Uses muxed RTCP, and minimal parameters with a hard-coded config that's
+// known to work.
+TEST_F(OrtcFactoryIntegrationTest, BasicOneWayVideoRtpSenderAndReceiver) {
+  auto udp_transports = CreateAndConnectUdpTransportPair();
+  auto rtp_transports =
+      CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
+
+  auto sender_result = ortc_factory1_->CreateRtpSender(
+      cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
+  auto receiver_result = ortc_factory2_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
+  ASSERT_TRUE(sender_result.ok());
+  ASSERT_TRUE(receiver_result.ok());
+  auto sender = sender_result.MoveValue();
+  auto receiver = receiver_result.MoveValue();
+
+  RTCError error = sender->SetTrack(
+      CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
+  EXPECT_TRUE(error.ok());
+
+  RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
+  EXPECT_TRUE(receiver->Receive(vp8_parameters).ok());
+  EXPECT_TRUE(sender->Send(vp8_parameters).ok());
+  FakeVideoTrackRenderer fake_renderer(
+      static_cast<VideoTrackInterface*>(receiver->GetTrack().get()));
+  // Sender and receiver are connected and configured; video frames should be
+  // able to flow at this point.
+  EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames,
+                   kDefaultTimeout);
+}
+
+// Test that if the track is changed while sending, the sender seamlessly
+// transitions to sending it and frames are received end-to-end.
+//
+// Only doing this for video, since given that audio is sourced from a single
+// fake audio capture module, the audio track is just a dummy object.
+// TODO(deadbeef): Change this when possible.
+TEST_F(OrtcFactoryIntegrationTest, SetTrackWhileSending) {
+  auto udp_transports = CreateAndConnectUdpTransportPair();
+  auto rtp_transports =
+      CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
+
+  auto sender_result = ortc_factory1_->CreateRtpSender(
+      cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
+  auto receiver_result = ortc_factory2_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
+  ASSERT_TRUE(sender_result.ok());
+  ASSERT_TRUE(receiver_result.ok());
+  auto sender = sender_result.MoveValue();
+  auto receiver = receiver_result.MoveValue();
+
+  RTCError error = sender->SetTrack(
+      CreateLocalVideoTrackAndFakeCapturer("video_1", ortc_factory1_.get()));
+  EXPECT_TRUE(error.ok());
+  RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
+  EXPECT_TRUE(receiver->Receive(vp8_parameters).ok());
+  EXPECT_TRUE(sender->Send(vp8_parameters).ok());
+  FakeVideoTrackRenderer fake_renderer(
+      static_cast<VideoTrackInterface*>(receiver->GetTrack().get()));
+  // Expect for some initial number of frames to be received.
+  EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames,
+                   kDefaultTimeout);
+  // Stop the old capturer, set a new track, and verify new frames are received
+  // from the new track. Stopping the old capturer ensures that we aren't
+  // actually still getting frames from it.
+  fake_video_capturers_[0]->Stop();
+  int prev_num_frames = fake_renderer.num_rendered_frames();
+  error = sender->SetTrack(
+      CreateLocalVideoTrackAndFakeCapturer("video_2", ortc_factory1_.get()));
+  EXPECT_TRUE(error.ok());
+  EXPECT_TRUE_WAIT(
+      fake_renderer.num_rendered_frames() > kDefaultNumFrames + prev_num_frames,
+      kDefaultTimeout);
+}
+
+// End-to-end test with two pairs of RTP senders and receivers, for audio and
+// video.
+//
+// Uses muxed RTCP, and minimal parameters with hard-coded configs that are
+// known to work.
+TEST_F(OrtcFactoryIntegrationTest,
+       BasicTwoWayAudioVideoRtpSendersAndReceivers) {
+  auto udp_transports = CreateAndConnectUdpTransportPair();
+  auto rtp_transports =
+      CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
+
+  // Create all the senders and receivers (four per endpoint).
+  auto audio_sender_result1 = ortc_factory1_->CreateRtpSender(
+      cricket::MEDIA_TYPE_AUDIO, rtp_transports.first.get());
+  auto video_sender_result1 = ortc_factory1_->CreateRtpSender(
+      cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
+  auto audio_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_AUDIO, rtp_transports.first.get());
+  auto video_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
+  ASSERT_TRUE(audio_sender_result1.ok());
+  ASSERT_TRUE(video_sender_result1.ok());
+  ASSERT_TRUE(audio_receiver_result1.ok());
+  ASSERT_TRUE(video_receiver_result1.ok());
+  auto audio_sender1 = audio_sender_result1.MoveValue();
+  auto video_sender1 = video_sender_result1.MoveValue();
+  auto audio_receiver1 = audio_receiver_result1.MoveValue();
+  auto video_receiver1 = video_receiver_result1.MoveValue();
+
+  auto audio_sender_result2 = ortc_factory2_->CreateRtpSender(
+      cricket::MEDIA_TYPE_AUDIO, rtp_transports.second.get());
+  auto video_sender_result2 = ortc_factory2_->CreateRtpSender(
+      cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
+  auto audio_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_AUDIO, rtp_transports.second.get());
+  auto video_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
+  ASSERT_TRUE(audio_sender_result2.ok());
+  ASSERT_TRUE(video_sender_result2.ok());
+  ASSERT_TRUE(audio_receiver_result2.ok());
+  ASSERT_TRUE(video_receiver_result2.ok());
+  auto audio_sender2 = audio_sender_result2.MoveValue();
+  auto video_sender2 = video_sender_result2.MoveValue();
+  auto audio_receiver2 = audio_receiver_result2.MoveValue();
+  auto video_receiver2 = video_receiver_result2.MoveValue();
+
+  // Add fake tracks.
+  RTCError error = audio_sender1->SetTrack(
+      CreateLocalAudioTrack("audio", ortc_factory1_.get()));
+  EXPECT_TRUE(error.ok());
+  error = video_sender1->SetTrack(
+      CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
+  EXPECT_TRUE(error.ok());
+  error = audio_sender2->SetTrack(
+      CreateLocalAudioTrack("audio", ortc_factory2_.get()));
+  EXPECT_TRUE(error.ok());
+  error = video_sender2->SetTrack(
+      CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory2_.get()));
+  EXPECT_TRUE(error.ok());
+
+  // "sent_X_parameters1" are the parameters that endpoint 1 sends with and
+  // endpoint 2 receives with.
+  RtpParameters sent_opus_parameters1 =
+      MakeMinimalOpusParametersWithSsrc(0xdeadbeef);
+  RtpParameters sent_vp8_parameters1 =
+      MakeMinimalVp8ParametersWithSsrc(0xbaadfeed);
+  RtpParameters sent_opus_parameters2 =
+      MakeMinimalOpusParametersWithSsrc(0x13333337);
+  RtpParameters sent_vp8_parameters2 =
+      MakeMinimalVp8ParametersWithSsrc(0x12345678);
+
+  // Configure the senders' and receivers' parameters.
+  EXPECT_TRUE(audio_receiver1->Receive(sent_opus_parameters2).ok());
+  EXPECT_TRUE(video_receiver1->Receive(sent_vp8_parameters2).ok());
+  EXPECT_TRUE(audio_receiver2->Receive(sent_opus_parameters1).ok());
+  EXPECT_TRUE(video_receiver2->Receive(sent_vp8_parameters1).ok());
+  EXPECT_TRUE(audio_sender1->Send(sent_opus_parameters1).ok());
+  EXPECT_TRUE(video_sender1->Send(sent_vp8_parameters1).ok());
+  EXPECT_TRUE(audio_sender2->Send(sent_opus_parameters2).ok());
+  EXPECT_TRUE(video_sender2->Send(sent_vp8_parameters2).ok());
+
+  FakeVideoTrackRenderer fake_video_renderer1(
+      static_cast<VideoTrackInterface*>(video_receiver1->GetTrack().get()));
+  FakeVideoTrackRenderer fake_video_renderer2(
+      static_cast<VideoTrackInterface*>(video_receiver2->GetTrack().get()));
+
+  // Senders and receivers are connected and configured; audio and video frames
+  // should be able to flow at this point.
+  EXPECT_TRUE_WAIT(
+      fake_audio_capture_module1_->frames_received() > kDefaultNumFrames &&
+          fake_video_renderer1.num_rendered_frames() > kDefaultNumFrames &&
+          fake_audio_capture_module2_->frames_received() > kDefaultNumFrames &&
+          fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames,
+      kDefaultTimeout);
+}
+
+// End-to-end test with two pairs of RTP senders and receivers, for audio and
+// video. Unlike the test above, this attempts to make the parameters as
+// complex as possible.
+//
+// Uses non-muxed RTCP, with separate audio/video transports, and a full set of
+// parameters, as would normally be used in a PeerConnection.
+//
+// TODO(deadbeef): Update this test as more audio/video features become
+// supported.
+TEST_F(OrtcFactoryIntegrationTest, FullTwoWayAudioVideoRtpSendersAndReceivers) {
+  // We want four pairs of UDP transports for this test, for audio/video and
+  // RTP/RTCP.
+  auto audio_rtp_udp_transports = CreateAndConnectUdpTransportPair();
+  auto audio_rtcp_udp_transports = CreateAndConnectUdpTransportPair();
+  auto video_rtp_udp_transports = CreateAndConnectUdpTransportPair();
+  auto video_rtcp_udp_transports = CreateAndConnectUdpTransportPair();
+
+  // Since we have multiple RTP transports on each side, we need an RTP
+  // transport controller.
+  auto transport_controllers = CreateRtpTransportControllerPair();
+
+  RtcpParameters audio_rtcp_parameters;
+  audio_rtcp_parameters.mux = false;
+  auto audio_rtp_transports =
+      CreateRtpTransportPair(audio_rtcp_parameters, audio_rtp_udp_transports,
+                             audio_rtcp_udp_transports, transport_controllers);
+
+  RtcpParameters video_rtcp_parameters;
+  video_rtcp_parameters.mux = false;
+  video_rtcp_parameters.reduced_size = true;
+  auto video_rtp_transports =
+      CreateRtpTransportPair(video_rtcp_parameters, video_rtp_udp_transports,
+                             video_rtcp_udp_transports, transport_controllers);
+
+  // Create all the senders and receivers (four per endpoint).
+  auto audio_sender_result1 = ortc_factory1_->CreateRtpSender(
+      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transports.first.get());
+  auto video_sender_result1 = ortc_factory1_->CreateRtpSender(
+      cricket::MEDIA_TYPE_VIDEO, video_rtp_transports.first.get());
+  auto audio_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transports.first.get());
+  auto video_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_VIDEO, video_rtp_transports.first.get());
+  ASSERT_TRUE(audio_sender_result1.ok());
+  ASSERT_TRUE(video_sender_result1.ok());
+  ASSERT_TRUE(audio_receiver_result1.ok());
+  ASSERT_TRUE(video_receiver_result1.ok());
+  auto audio_sender1 = audio_sender_result1.MoveValue();
+  auto video_sender1 = video_sender_result1.MoveValue();
+  auto audio_receiver1 = audio_receiver_result1.MoveValue();
+  auto video_receiver1 = video_receiver_result1.MoveValue();
+
+  auto audio_sender_result2 = ortc_factory2_->CreateRtpSender(
+      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transports.second.get());
+  auto video_sender_result2 = ortc_factory2_->CreateRtpSender(
+      cricket::MEDIA_TYPE_VIDEO, video_rtp_transports.second.get());
+  auto audio_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transports.second.get());
+  auto video_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
+      cricket::MEDIA_TYPE_VIDEO, video_rtp_transports.second.get());
+  ASSERT_TRUE(audio_sender_result2.ok());
+  ASSERT_TRUE(video_sender_result2.ok());
+  ASSERT_TRUE(audio_receiver_result2.ok());
+  ASSERT_TRUE(video_receiver_result2.ok());
+  auto audio_sender2 = audio_sender_result2.MoveValue();
+  auto video_sender2 = video_sender_result2.MoveValue();
+  auto audio_receiver2 = audio_receiver_result2.MoveValue();
+  auto video_receiver2 = video_receiver_result2.MoveValue();
+
+  RTCError error = audio_sender1->SetTrack(
+      CreateLocalAudioTrack("audio", ortc_factory1_.get()));
+  EXPECT_TRUE(error.ok());
+  error = video_sender1->SetTrack(
+      CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
+  EXPECT_TRUE(error.ok());
+  error = audio_sender2->SetTrack(
+      CreateLocalAudioTrack("audio", ortc_factory2_.get()));
+  EXPECT_TRUE(error.ok());
+  error = video_sender2->SetTrack(
+      CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory2_.get()));
+  EXPECT_TRUE(error.ok());
+
+  // Use different codecs in different directions for extra challenge.
+  RtpParameters opus_send_parameters = MakeFullOpusParameters();
+  RtpParameters isac_send_parameters = MakeFullIsacParameters();
+  RtpParameters vp8_send_parameters = MakeFullVp8Parameters();
+  RtpParameters vp9_send_parameters = MakeFullVp9Parameters();
+
+  // Remove "payload_type" from receive parameters. Receiver will need to
+  // discern the payload type from packets received.
+  RtpParameters opus_receive_parameters = opus_send_parameters;
+  RtpParameters isac_receive_parameters = isac_send_parameters;
+  RtpParameters vp8_receive_parameters = vp8_send_parameters;
+  RtpParameters vp9_receive_parameters = vp9_send_parameters;
+  opus_receive_parameters.encodings[0].codec_payload_type.reset();
+  isac_receive_parameters.encodings[0].codec_payload_type.reset();
+  vp8_receive_parameters.encodings[0].codec_payload_type.reset();
+  vp9_receive_parameters.encodings[0].codec_payload_type.reset();
+
+  // Configure the senders' and receivers' parameters.
+  //
+  // Note: Intentionally, the top codec in the receive parameters does not
+  // match the codec sent by the other side. If "Receive" is called with a list
+  // of codecs, the receiver should be prepared to receive any of them, not
+  // just the one on top.
+  EXPECT_TRUE(audio_receiver1->Receive(opus_receive_parameters).ok());
+  EXPECT_TRUE(video_receiver1->Receive(vp8_receive_parameters).ok());
+  EXPECT_TRUE(audio_receiver2->Receive(isac_receive_parameters).ok());
+  EXPECT_TRUE(video_receiver2->Receive(vp9_receive_parameters).ok());
+  EXPECT_TRUE(audio_sender1->Send(opus_send_parameters).ok());
+  EXPECT_TRUE(video_sender1->Send(vp8_send_parameters).ok());
+  EXPECT_TRUE(audio_sender2->Send(isac_send_parameters).ok());
+  EXPECT_TRUE(video_sender2->Send(vp9_send_parameters).ok());
+
+  FakeVideoTrackRenderer fake_video_renderer1(
+      static_cast<VideoTrackInterface*>(video_receiver1->GetTrack().get()));
+  FakeVideoTrackRenderer fake_video_renderer2(
+      static_cast<VideoTrackInterface*>(video_receiver2->GetTrack().get()));
+
+  // Senders and receivers are connected and configured; audio and video frames
+  // should be able to flow at this point.
+  EXPECT_TRUE_WAIT(
+      fake_audio_capture_module1_->frames_received() > kDefaultNumFrames &&
+          fake_video_renderer1.num_rendered_frames() > kDefaultNumFrames &&
+          fake_audio_capture_module2_->frames_received() > kDefaultNumFrames &&
+          fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames,
+      kDefaultTimeout);
+}
+
+// TODO(deadbeef): End-to-end test for multiple senders/receivers of the same
+// media type, once that's supported. Currently, it is not because the
+// BaseChannel model relies on there being a single VoiceChannel and
+// VideoChannel, and these only support a single set of codecs/etc. per
+// send/receive direction.
+
+// TODO(deadbeef): End-to-end test for simulcast, once that's supported by this
+// API.
+
+}  // namespace webrtc