diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 2aaacc75a0..be534a0c37 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -99,6 +99,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":audio",
+      "../api:simulated_network_api",
       "../system_wrappers:system_wrappers",
       "../test:test_common",
       "../test:test_support",
@@ -175,6 +176,7 @@ if (rtc_include_tests) {
 
       deps = [
         ":audio_end_to_end_test",
+        "../api:simulated_network_api",
         "../common_audio",
         "../rtc_base:rtc_base_approved",
         "../system_wrappers",
@@ -246,6 +248,7 @@ if (rtc_include_tests) {
       "test/audio_bwe_integration_test.h",
     ]
     deps = [
+      "../api:simulated_network_api",
       "../common_audio",
       "../rtc_base:rtc_base_approved",
       "../system_wrappers",
diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc
index 4d8ac804c8..6640f6d2ac 100644
--- a/audio/test/audio_bwe_integration_test.cc
+++ b/audio/test/audio_bwe_integration_test.cc
@@ -115,8 +115,8 @@ class NoBandwidthDropAfterDtx : public AudioBweTest {
     return test::ResourcePath("voice_engine/audio_dtx16", "wav");
   }
 
-  FakeNetworkPipe::Config GetNetworkPipeConfig() override {
-    FakeNetworkPipe::Config pipe_config;
+  DefaultNetworkSimulationConfig GetNetworkPipeConfig() override {
+    DefaultNetworkSimulationConfig pipe_config;
     pipe_config.link_capacity_kbps = 50;
     pipe_config.queue_length_packets = 1500;
     pipe_config.queue_delay_ms = 300;
diff --git a/audio/test/audio_bwe_integration_test.h b/audio/test/audio_bwe_integration_test.h
index a88b047993..c70ad1c91b 100644
--- a/audio/test/audio_bwe_integration_test.h
+++ b/audio/test/audio_bwe_integration_test.h
@@ -13,6 +13,7 @@
 #include <memory>
 #include <string>
 
+#include "api/test/simulated_network.h"
 #include "test/call_test.h"
 #include "test/single_threaded_task_queue.h"
 
@@ -26,7 +27,7 @@ class AudioBweTest : public test::EndToEndTest {
  protected:
   virtual std::string AudioInputFile() = 0;
 
-  virtual FakeNetworkPipe::Config GetNetworkPipeConfig() = 0;
+  virtual DefaultNetworkSimulationConfig GetNetworkPipeConfig() = 0;
 
   size_t GetNumVideoStreams() const override;
   size_t GetNumAudioStreams() const override;
diff --git a/audio/test/audio_end_to_end_test.cc b/audio/test/audio_end_to_end_test.cc
index 820b46444e..8ba9a047f3 100644
--- a/audio/test/audio_end_to_end_test.cc
+++ b/audio/test/audio_end_to_end_test.cc
@@ -26,8 +26,8 @@ constexpr int kSampleRate = 48000;
 AudioEndToEndTest::AudioEndToEndTest()
     : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
 
-FakeNetworkPipe::Config AudioEndToEndTest::GetNetworkPipeConfig() const {
-  return FakeNetworkPipe::Config();
+DefaultNetworkSimulationConfig AudioEndToEndTest::GetNetworkPipeConfig() const {
+  return DefaultNetworkSimulationConfig();
 }
 
 size_t AudioEndToEndTest::GetNumVideoStreams() const {
diff --git a/audio/test/audio_end_to_end_test.h b/audio/test/audio_end_to_end_test.h
index cbe25f5b3a..9dda834d22 100644
--- a/audio/test/audio_end_to_end_test.h
+++ b/audio/test/audio_end_to_end_test.h
@@ -14,6 +14,7 @@
 #include <string>
 #include <vector>
 
+#include "api/test/simulated_network.h"
 #include "test/call_test.h"
 
 namespace webrtc {
@@ -28,7 +29,7 @@ class AudioEndToEndTest : public test::EndToEndTest {
   const AudioSendStream* send_stream() const { return send_stream_; }
   const AudioReceiveStream* receive_stream() const { return receive_stream_; }
 
-  virtual FakeNetworkPipe::Config GetNetworkPipeConfig() const;
+  virtual DefaultNetworkSimulationConfig GetNetworkPipeConfig() const;
 
   size_t GetNumVideoStreams() const override;
   size_t GetNumAudioStreams() const override;
diff --git a/audio/test/audio_stats_test.cc b/audio/test/audio_stats_test.cc
index ee225c0b50..019679e078 100644
--- a/audio/test/audio_stats_test.cc
+++ b/audio/test/audio_stats_test.cc
@@ -32,8 +32,8 @@ class NoLossTest : public AudioEndToEndTest {
 
   NoLossTest() = default;
 
-  FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
-    FakeNetworkPipe::Config pipe_config;
+  DefaultNetworkSimulationConfig GetNetworkPipeConfig() const override {
+    DefaultNetworkSimulationConfig pipe_config;
     pipe_config.queue_delay_ms = kRttMs / 2;
     return pipe_config;
   }
diff --git a/audio/test/low_bandwidth_audio_test.cc b/audio/test/low_bandwidth_audio_test.cc
index 169f4bf592..16a3b91e87 100644
--- a/audio/test/low_bandwidth_audio_test.cc
+++ b/audio/test/low_bandwidth_audio_test.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "audio/test/audio_end_to_end_test.h"
 #include "rtc_base/flags.h"
 #include "system_wrappers/include/sleep.h"
@@ -88,8 +89,8 @@ class Mobile2GNetworkTest : public AudioQualityTest {
          {{"maxaveragebitrate", "6000"}, {"ptime", "60"}, {"stereo", "1"}}});
   }
 
-  FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
-    FakeNetworkPipe::Config pipe_config;
+  DefaultNetworkSimulationConfig GetNetworkPipeConfig() const override {
+    DefaultNetworkSimulationConfig pipe_config;
     pipe_config.link_capacity_kbps = 12;
     pipe_config.queue_length_packets = 1500;
     pipe_config.queue_delay_ms = 400;
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 796509746b..3ec2a2c1ae 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -261,6 +261,17 @@ rtc_source_set("simulated_network") {
   ]
 }
 
+rtc_source_set("simulated_packet_receiver") {
+  sources = [
+    "simulated_packet_receiver.h",
+  ]
+  deps = [
+    ":call_interfaces",
+    "../api:simulated_network_api",
+    "../modules:module_api",
+  ]
+}
+
 rtc_source_set("fake_network") {
   sources = [
     "fake_network_pipe.cc",
@@ -269,6 +280,7 @@ rtc_source_set("fake_network") {
   deps = [
     ":call_interfaces",
     ":simulated_network",
+    ":simulated_packet_receiver",
     "..:webrtc_common",
     "../api:simulated_network_api",
     "../api:transport_api",
@@ -360,6 +372,7 @@ if (rtc_include_tests) {
       ":call_interfaces",
       ":video_stream_api",
       "..:webrtc_common",
+      "../api:simulated_network_api",
       "../api/audio_codecs:builtin_audio_encoder_factory",
       "../api/video:video_bitrate_allocation",
       "../api/video_codecs:video_codecs_api",
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 54e01b1232..00cfada83a 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -15,6 +15,7 @@
 
 #include "absl/memory/memory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/test/simulated_network.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video_codecs/video_encoder_config.h"
 #include "call/call.h"
@@ -60,7 +61,7 @@ class CallPerfTest : public test::CallTest {
 
   void TestMinTransmitBitrate(bool pad_to_min_bitrate);
 
-  void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
+  void TestCaptureNtpTime(const DefaultNetworkSimulationConfig& net_config,
                           int threshold_ms,
                           int start_time_ms,
                           int run_time_ms);
@@ -148,7 +149,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
 
-  FakeNetworkPipe::Config audio_net_config;
+  DefaultNetworkSimulationConfig audio_net_config;
   audio_net_config.queue_delay_ms = 500;
   audio_net_config.loss_percent = 5;
 
@@ -207,13 +208,13 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
     video_send_transport = absl::make_unique<test::PacketTransport>(
         &task_queue_, sender_call_.get(), &observer,
         test::PacketTransport::kSender, video_pt_map,
-        FakeNetworkPipe::Config());
+        DefaultNetworkSimulationConfig());
     video_send_transport->SetReceiver(receiver_call_->Receiver());
 
     receive_transport = absl::make_unique<test::PacketTransport>(
         &task_queue_, receiver_call_.get(), &observer,
         test::PacketTransport::kReceiver, payload_type_map_,
-        FakeNetworkPipe::Config());
+        DefaultNetworkSimulationConfig());
     receive_transport->SetReceiver(sender_call_->Receiver());
 
     CreateSendConfig(1, 0, 0, video_send_transport.get());
@@ -323,14 +324,15 @@ TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) {
                      DriftingClock::PercentsSlower(30.0f), "_video_faster");
 }
 
-void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
-                                      int threshold_ms,
-                                      int start_time_ms,
-                                      int run_time_ms) {
+void CallPerfTest::TestCaptureNtpTime(
+    const DefaultNetworkSimulationConfig& net_config,
+    int threshold_ms,
+    int start_time_ms,
+    int run_time_ms) {
   class CaptureNtpTimeObserver : public test::EndToEndTest,
                                  public rtc::VideoSinkInterface<VideoFrame> {
    public:
-    CaptureNtpTimeObserver(const FakeNetworkPipe::Config& net_config,
+    CaptureNtpTimeObserver(const DefaultNetworkSimulationConfig& net_config,
                            int threshold_ms,
                            int start_time_ms,
                            int run_time_ms)
@@ -441,7 +443,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
     }
 
     rtc::CriticalSection crit_;
-    const FakeNetworkPipe::Config net_config_;
+    const DefaultNetworkSimulationConfig net_config_;
     Clock* const clock_;
     int threshold_ms_;
     int start_time_ms_;
@@ -461,7 +463,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
 // Flaky tests, disabled on Mac due to webrtc:8291.
 #if !(defined(WEBRTC_MAC))
 TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
-  FakeNetworkPipe::Config net_config;
+  DefaultNetworkSimulationConfig net_config;
   net_config.queue_delay_ms = 100;
   // TODO(wu): lower the threshold as the calculation/estimatation becomes more
   // accurate.
@@ -472,7 +474,7 @@ TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
 }
 
 TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
-  FakeNetworkPipe::Config net_config;
+  DefaultNetworkSimulationConfig net_config;
   net_config.queue_delay_ms = 100;
   net_config.delay_standard_deviation_ms = 10;
   // TODO(wu): lower the threshold as the calculation/estimatation becomes more
@@ -828,8 +830,8 @@ void CallPerfTest::TestMinAudioVideoBitrate(
           max_bwe_(max_bwe) {}
 
    protected:
-    FakeNetworkPipe::Config GetFakeNetworkPipeConfig() {
-      FakeNetworkPipe::Config pipe_config;
+    DefaultNetworkSimulationConfig GetFakeNetworkPipeConfig() {
+      DefaultNetworkSimulationConfig pipe_config;
       pipe_config.link_capacity_kbps = test_bitrate_from_;
       return pipe_config;
     }
@@ -856,7 +858,7 @@ void CallPerfTest::TestMinAudioVideoBitrate(
                ? test_bitrate <= test_bitrate_to_
                : test_bitrate >= test_bitrate_to_;
            test_bitrate += test_bitrate_step_) {
-        FakeNetworkPipe::Config pipe_config;
+        DefaultNetworkSimulationConfig pipe_config;
         pipe_config.link_capacity_kbps = test_bitrate;
         send_transport_->SetConfig(pipe_config);
         receive_transport_->SetConfig(pipe_config);
diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index a46cd75f18..546014d39d 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h
@@ -22,8 +22,8 @@
 #include "api/call/transport.h"
 #include "api/test/simulated_network.h"
 #include "call/call.h"
+#include "call/simulated_packet_receiver.h"
 #include "common_types.h"  // NOLINT(build/include)
-#include "modules/include/module.h"
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/thread_annotations.h"
@@ -87,7 +87,8 @@ class NetworkPacket {
 
 // Class faking a network link, internally is uses an implementation of a
 // SimulatedNetworkInterface to simulate network behavior.
-class FakeNetworkPipe : public Transport, public PacketReceiver, public Module {
+class FakeNetworkPipe : public webrtc::SimulatedPacketReceiverInterface,
+                        public Transport {
  public:
   using Config = NetworkSimulationInterface::SimulatedNetworkConfig;
 
@@ -135,15 +136,16 @@ class FakeNetworkPipe : public Transport, public PacketReceiver, public Module {
 
   ~FakeNetworkPipe() override;
 
-  void SetClockOffset(int64_t offset_ms);
+  // Deprecated. DO NOT USE. Will be removed soon.
+  void SetClockOffset(int64_t offset_ms) override;
 
   // Deprecated. DO NOT USE. Hold direct reference on NetworkSimulationInterface
   // instead and call SetConfig on that object directly. Will be removed soon.
   // Sets a new configuration. This won't affect packets already in the pipe.
-  void SetConfig(const FakeNetworkPipe::Config& config);
+  void SetConfig(const DefaultNetworkSimulationConfig& config) override;
 
   // Must not be called in parallel with DeliverPacket or Process.
-  void SetReceiver(PacketReceiver* receiver);
+  void SetReceiver(PacketReceiver* receiver) override;
 
   // Implements Transport interface. When/if packets are delivered, they will
   // be passed to the transport instance given in SetReceiverTransport(). These
@@ -174,7 +176,7 @@ class FakeNetworkPipe : public Transport, public PacketReceiver, public Module {
 
   // Get statistics.
   float PercentageLoss();
-  int AverageDelay();
+  int AverageDelay() override;
   size_t DroppedPackets();
   size_t SentPackets();
   void ResetStats();
diff --git a/call/rampup_tests.h b/call/rampup_tests.h
index a22dfc9934..5fd089fb27 100644
--- a/call/rampup_tests.h
+++ b/call/rampup_tests.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "api/test/simulated_network.h"
 #include "call/call.h"
 #include "logging/rtc_event_log/rtc_event_log.h"
 #include "rtc_base/event.h"
@@ -64,7 +65,7 @@ class RampUpTester : public test::EndToEndTest {
 
   rtc::Event stop_event_;
   Clock* const clock_;
-  FakeNetworkPipe::Config forward_transport_config_;
+  DefaultNetworkSimulationConfig forward_transport_config_;
   const size_t num_video_streams_;
   const size_t num_audio_streams_;
   const size_t num_flexfec_streams_;
diff --git a/call/simulated_packet_receiver.h b/call/simulated_packet_receiver.h
new file mode 100644
index 0000000000..6630427b1a
--- /dev/null
+++ b/call/simulated_packet_receiver.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2018 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.
+ */
+
+#ifndef CALL_SIMULATED_PACKET_RECEIVER_H_
+#define CALL_SIMULATED_PACKET_RECEIVER_H_
+
+#include "api/test/simulated_network.h"
+#include "call/packet_receiver.h"
+#include "modules/include/module.h"
+
+namespace webrtc {
+
+// Private API that is fixing surface between DirectTransport and underlying
+// network conditions simulation implementation.
+class SimulatedPacketReceiverInterface : public PacketReceiver, public Module {
+ public:
+  // Must not be called in parallel with DeliverPacket or Process.
+  // Destination receiver will be injected with this method
+  virtual void SetReceiver(PacketReceiver* receiver) = 0;
+
+  // Reports average packet delay.
+  virtual int AverageDelay() = 0;
+
+  // Deprecated. DO NOT USE. Temporary added to be able to introduce
+  // SimulatedPacketReceiverInterface into DirectTransport instead of
+  // FakeNetworkPipe, will be removed soon.
+  virtual void SetClockOffset(int64_t offset_ms) = 0;
+
+  // Deprecated. DO NOT USE. Temporary added to be able to introduce
+  // SimulatedPacketReceiverInterface into DirectTransport instead of
+  // FakeNetworkPipe, will be removed soon.
+  virtual void SetConfig(const DefaultNetworkSimulationConfig& config) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_SIMULATED_PACKET_RECEIVER_H_
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 9b4ba418f7..8431cf27c1 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -459,8 +459,10 @@ rtc_source_set("direct_transport") {
   }
   deps = [
     "..:webrtc_common",
+    "../api:simulated_network_api",
     "../api:transport_api",
     "../call:call_interfaces",
+    "../call:simulated_packet_receiver",
     "../modules/rtp_rtcp",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:sequenced_task_checker",
@@ -538,6 +540,7 @@ rtc_source_set("test_common") {
     ":video_test_common",
     "..:webrtc_common",
     "../api:libjingle_peerconnection_api",
+    "../api:simulated_network_api",
     "../api:transport_api",
     "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/audio_codecs:builtin_audio_encoder_factory",
@@ -548,6 +551,7 @@ rtc_source_set("test_common") {
     "../call",
     "../call:call_interfaces",
     "../call:rtp_sender",
+    "../call:simulated_packet_receiver",
     "../call:video_stream_api",
     "../common_video",
     "../logging:rtc_event_log_api",
diff --git a/test/call_test.cc b/test/call_test.cc
index 4df054b5cc..18a5b9a5fa 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -730,14 +730,14 @@ test::PacketTransport* BaseTest::CreateSendTransport(
     Call* sender_call) {
   return new PacketTransport(
       task_queue, sender_call, this, test::PacketTransport::kSender,
-      CallTest::payload_type_map_, FakeNetworkPipe::Config());
+      CallTest::payload_type_map_, DefaultNetworkSimulationConfig());
 }
 
 test::PacketTransport* BaseTest::CreateReceiveTransport(
     SingleThreadedTaskQueueForTesting* task_queue) {
   return new PacketTransport(
       task_queue, nullptr, this, test::PacketTransport::kReceiver,
-      CallTest::payload_type_map_, FakeNetworkPipe::Config());
+      CallTest::payload_type_map_, DefaultNetworkSimulationConfig());
 }
 
 size_t BaseTest::GetNumVideoStreams() const {
diff --git a/test/direct_transport.cc b/test/direct_transport.cc
index 93221a6055..351210ee5c 100644
--- a/test/direct_transport.cc
+++ b/test/direct_transport.cc
@@ -11,6 +11,7 @@
 
 #include "absl/memory/memory.h"
 #include "call/call.h"
+#include "call/fake_network_pipe.h"
 #include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "system_wrappers/include/clock.h"
 #include "test/single_threaded_task_queue.h"
@@ -40,13 +41,13 @@ DirectTransport::DirectTransport(
     Call* send_call,
     const std::map<uint8_t, MediaType>& payload_type_map)
     : DirectTransport(task_queue,
-                      FakeNetworkPipe::Config(),
+                      DefaultNetworkSimulationConfig(),
                       send_call,
                       payload_type_map) {}
 
 DirectTransport::DirectTransport(
     SingleThreadedTaskQueueForTesting* task_queue,
-    const FakeNetworkPipe::Config& config,
+    const DefaultNetworkSimulationConfig& config,
     Call* send_call,
     const std::map<uint8_t, MediaType>& payload_type_map)
     : send_call_(send_call),
@@ -59,7 +60,7 @@ DirectTransport::DirectTransport(
 
 DirectTransport::DirectTransport(
     SingleThreadedTaskQueueForTesting* task_queue,
-    std::unique_ptr<FakeNetworkPipe> pipe,
+    std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
     Call* send_call,
     const std::map<uint8_t, MediaType>& payload_type_map)
     : send_call_(send_call),
@@ -81,7 +82,7 @@ void DirectTransport::SetClockOffset(int64_t offset_ms) {
   fake_network_->SetClockOffset(offset_ms);
 }
 
-void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
+void DirectTransport::SetConfig(const DefaultNetworkSimulationConfig& config) {
   fake_network_->SetConfig(config);
 }
 
diff --git a/test/direct_transport.h b/test/direct_transport.h
index 2d473073e9..c5c2d94322 100644
--- a/test/direct_transport.h
+++ b/test/direct_transport.h
@@ -15,8 +15,9 @@
 #include <memory>
 
 #include "api/call/transport.h"
+#include "api/test/simulated_network.h"
 #include "call/call.h"
-#include "call/fake_network_pipe.h"
+#include "call/simulated_packet_receiver.h"
 #include "rtc_base/sequenced_task_checker.h"
 #include "rtc_base/thread_annotations.h"
 #include "test/single_threaded_task_queue.h"
@@ -46,12 +47,12 @@ class DirectTransport : public Transport {
                   const std::map<uint8_t, MediaType>& payload_type_map);
 
   DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
-                  const FakeNetworkPipe::Config& config,
+                  const DefaultNetworkSimulationConfig& config,
                   Call* send_call,
                   const std::map<uint8_t, MediaType>& payload_type_map);
 
   DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
-                  std::unique_ptr<FakeNetworkPipe> pipe,
+                  std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
                   Call* send_call,
                   const std::map<uint8_t, MediaType>& payload_type_map);
 
@@ -59,7 +60,7 @@ class DirectTransport : public Transport {
 
   void SetClockOffset(int64_t offset_ms);
 
-  void SetConfig(const FakeNetworkPipe::Config& config);
+  void SetConfig(const DefaultNetworkSimulationConfig& config);
 
   RTC_DEPRECATED void StopSending();
 
@@ -86,7 +87,7 @@ class DirectTransport : public Transport {
       RTC_GUARDED_BY(&sequence_checker_);
 
   const Demuxer demuxer_;
-  const std::unique_ptr<FakeNetworkPipe> fake_network_;
+  const std::unique_ptr<SimulatedPacketReceiverInterface> fake_network_;
 
   rtc::SequencedTaskChecker sequence_checker_;
 };
diff --git a/test/layer_filtering_transport.cc b/test/layer_filtering_transport.cc
index 1f36619bdd..d2861db591 100644
--- a/test/layer_filtering_transport.cc
+++ b/test/layer_filtering_transport.cc
@@ -22,7 +22,7 @@ namespace test {
 
 LayerFilteringTransport::LayerFilteringTransport(
     SingleThreadedTaskQueueForTesting* task_queue,
-    const FakeNetworkPipe::Config& config,
+    const DefaultNetworkSimulationConfig& config,
     Call* send_call,
     uint8_t vp8_video_payload_type,
     uint8_t vp9_video_payload_type,
@@ -42,7 +42,7 @@ LayerFilteringTransport::LayerFilteringTransport(
 
 LayerFilteringTransport::LayerFilteringTransport(
     SingleThreadedTaskQueueForTesting* task_queue,
-    std::unique_ptr<FakeNetworkPipe> pipe,
+    std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
     Call* send_call,
     uint8_t vp8_video_payload_type,
     uint8_t vp9_video_payload_type,
@@ -62,7 +62,7 @@ LayerFilteringTransport::LayerFilteringTransport(
 
 LayerFilteringTransport::LayerFilteringTransport(
     SingleThreadedTaskQueueForTesting* task_queue,
-    const FakeNetworkPipe::Config& config,
+    const DefaultNetworkSimulationConfig& config,
     Call* send_call,
     uint8_t vp8_video_payload_type,
     uint8_t vp9_video_payload_type,
@@ -80,7 +80,7 @@ LayerFilteringTransport::LayerFilteringTransport(
 
 LayerFilteringTransport::LayerFilteringTransport(
     SingleThreadedTaskQueueForTesting* task_queue,
-    std::unique_ptr<FakeNetworkPipe> pipe,
+    std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
     Call* send_call,
     uint8_t vp8_video_payload_type,
     uint8_t vp9_video_payload_type,
diff --git a/test/layer_filtering_transport.h b/test/layer_filtering_transport.h
index bd5f2e8ab9..7d2fe4f2c7 100644
--- a/test/layer_filtering_transport.h
+++ b/test/layer_filtering_transport.h
@@ -12,8 +12,9 @@
 
 #include <map>
 
+#include "api/test/simulated_network.h"
 #include "call/call.h"
-#include "call/fake_network_pipe.h"
+#include "call/simulated_packet_receiver.h"
 #include "test/direct_transport.h"
 #include "test/single_threaded_task_queue.h"
 
@@ -24,7 +25,7 @@ namespace test {
 class LayerFilteringTransport : public test::DirectTransport {
  public:
   LayerFilteringTransport(SingleThreadedTaskQueueForTesting* task_queue,
-                          const FakeNetworkPipe::Config& config,
+                          const DefaultNetworkSimulationConfig& config,
                           Call* send_call,
                           uint8_t vp8_video_payload_type,
                           uint8_t vp9_video_payload_type,
@@ -34,31 +35,33 @@ class LayerFilteringTransport : public test::DirectTransport {
                           uint32_t ssrc_to_filter_min,
                           uint32_t ssrc_to_filter_max);
   LayerFilteringTransport(SingleThreadedTaskQueueForTesting* task_queue,
-                          const FakeNetworkPipe::Config& config,
-                          Call* send_call,
-                          uint8_t vp8_video_payload_type,
-                          uint8_t vp9_video_payload_type,
-                          int selected_tl,
-                          int selected_sl,
-                          const std::map<uint8_t, MediaType>& payload_type_map);
-  LayerFilteringTransport(SingleThreadedTaskQueueForTesting* task_queue,
-                          std::unique_ptr<FakeNetworkPipe> pipe,
-                          Call* send_call,
-                          uint8_t vp8_video_payload_type,
-                          uint8_t vp9_video_payload_type,
-                          int selected_tl,
-                          int selected_sl,
-                          const std::map<uint8_t, MediaType>& payload_type_map,
-                          uint32_t ssrc_to_filter_min,
-                          uint32_t ssrc_to_filter_max);
-  LayerFilteringTransport(SingleThreadedTaskQueueForTesting* task_queue,
-                          std::unique_ptr<FakeNetworkPipe> pipe,
+                          const DefaultNetworkSimulationConfig& config,
                           Call* send_call,
                           uint8_t vp8_video_payload_type,
                           uint8_t vp9_video_payload_type,
                           int selected_tl,
                           int selected_sl,
                           const std::map<uint8_t, MediaType>& payload_type_map);
+  LayerFilteringTransport(
+      SingleThreadedTaskQueueForTesting* task_queue,
+      std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+      Call* send_call,
+      uint8_t vp8_video_payload_type,
+      uint8_t vp9_video_payload_type,
+      int selected_tl,
+      int selected_sl,
+      const std::map<uint8_t, MediaType>& payload_type_map,
+      uint32_t ssrc_to_filter_min,
+      uint32_t ssrc_to_filter_max);
+  LayerFilteringTransport(
+      SingleThreadedTaskQueueForTesting* task_queue,
+      std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+      Call* send_call,
+      uint8_t vp8_video_payload_type,
+      uint8_t vp9_video_payload_type,
+      int selected_tl,
+      int selected_sl,
+      const std::map<uint8_t, MediaType>& payload_type_map);
   bool DiscardedLastPacket() const;
   bool SendRtp(const uint8_t* data,
                size_t length,
diff --git a/test/rtp_rtcp_observer.h b/test/rtp_rtcp_observer.h
index d946b5f675..aa6fc490f0 100644
--- a/test/rtp_rtcp_observer.h
+++ b/test/rtp_rtcp_observer.h
@@ -14,6 +14,8 @@
 #include <memory>
 #include <vector>
 
+#include "api/test/simulated_network.h"
+#include "call/simulated_packet_receiver.h"
 #include "call/video_send_stream.h"
 #include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/criticalsection.h"
@@ -96,7 +98,7 @@ class PacketTransport : public test::DirectTransport {
                   RtpRtcpObserver* observer,
                   TransportType transport_type,
                   const std::map<uint8_t, MediaType>& payload_type_map,
-                  const FakeNetworkPipe::Config& configuration)
+                  const DefaultNetworkSimulationConfig& configuration)
       : test::DirectTransport(task_queue,
                               configuration,
                               send_call,
@@ -109,7 +111,7 @@ class PacketTransport : public test::DirectTransport {
                   RtpRtcpObserver* observer,
                   TransportType transport_type,
                   const std::map<uint8_t, MediaType>& payload_type_map,
-                  std::unique_ptr<FakeNetworkPipe> nw_pipe)
+                  std::unique_ptr<SimulatedPacketReceiverInterface> nw_pipe)
       : test::DirectTransport(task_queue,
                               std::move(nw_pipe),
                               send_call,
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 5f9455c500..5901fb98b0 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -424,6 +424,7 @@ if (rtc_include_tests) {
       ":video",
       ":video_mocks",
       ":video_stream_encoder_impl",
+      "../api:simulated_network_api",
       "../api/video:video_frame",
       "../api/video:video_frame_i420",
       "../api/video_codecs:video_codecs_api",
diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index 3b1df3c590..e33feca478 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "rtc_base/rate_limiter.h"
 #include "system_wrappers/include/sleep.h"
@@ -164,7 +165,7 @@ TEST_P(BandwidthEndToEndTest, RembWithSendSideBwe) {
         test::SingleThreadedTaskQueueForTesting* task_queue) override {
       receive_transport_ = new test::PacketTransport(
           task_queue, nullptr, this, test::PacketTransport::kReceiver,
-          payload_type_map_, FakeNetworkPipe::Config());
+          payload_type_map_, DefaultNetworkSimulationConfig());
       return receive_transport_;
     }
 
diff --git a/video/end_to_end_tests/fec_tests.cc b/video/end_to_end_tests/fec_tests.cc
index 4493bde019..043f0cbec2 100644
--- a/video/end_to_end_tests/fec_tests.cc
+++ b/video/end_to_end_tests/fec_tests.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "test/call_test.h"
@@ -239,7 +240,7 @@ class FlexfecRenderObserver : public test::EndToEndTest,
       Call* sender_call) override {
     // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
     const int kNetworkDelayMs = 100;
-    FakeNetworkPipe::Config config;
+    DefaultNetworkSimulationConfig config;
     config.queue_delay_ms = kNetworkDelayMs;
     return new test::PacketTransport(task_queue, sender_call, this,
                                      test::PacketTransport::kSender,
@@ -424,7 +425,7 @@ TEST_P(FecEndToEndTest, ReceivedUlpfecPacketsNotNacked) {
       // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
       // Configure some network delay.
       const int kNetworkDelayMs = 50;
-      FakeNetworkPipe::Config config;
+      DefaultNetworkSimulationConfig config;
       config.queue_delay_ms = kNetworkDelayMs;
       return new test::PacketTransport(task_queue, sender_call, this,
                                        test::PacketTransport::kSender,
diff --git a/video/end_to_end_tests/multi_codec_receive_tests.cc b/video/end_to_end_tests/multi_codec_receive_tests.cc
index d4ab3ee194..3999c83eb0 100644
--- a/video/end_to_end_tests/multi_codec_receive_tests.cc
+++ b/video/end_to_end_tests/multi_codec_receive_tests.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "modules/video_coding/codecs/h264/include/h264.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "modules/video_coding/codecs/vp9/include/vp9.h"
@@ -149,13 +150,13 @@ class MultiCodecReceiveTest : public test::CallTest {
       send_transport_.reset(new test::PacketTransport(
           &task_queue_, sender_call_.get(), &observer_,
           test::PacketTransport::kSender, kPayloadTypeMap,
-          FakeNetworkPipe::Config()));
+          DefaultNetworkSimulationConfig()));
       send_transport_->SetReceiver(receiver_call_->Receiver());
 
       receive_transport_.reset(new test::PacketTransport(
           &task_queue_, receiver_call_.get(), &observer_,
           test::PacketTransport::kReceiver, kPayloadTypeMap,
-          FakeNetworkPipe::Config()));
+          DefaultNetworkSimulationConfig()));
       receive_transport_->SetReceiver(sender_call_->Receiver());
     });
   }
diff --git a/video/end_to_end_tests/probing_tests.cc b/video/end_to_end_tests/probing_tests.cc
index 4d0009f41c..0520c6f53a 100644
--- a/video/end_to_end_tests/probing_tests.cc
+++ b/video/end_to_end_tests/probing_tests.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "test/call_test.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
@@ -220,7 +221,7 @@ TEST_P(ProbingEndToEndTest, ProbeOnVideoEncoderReconfiguration) {
         Call* sender_call) override {
       send_transport_ = new test::PacketTransport(
           task_queue, sender_call, this, test::PacketTransport::kSender,
-          CallTest::payload_type_map_, FakeNetworkPipe::Config());
+          CallTest::payload_type_map_, DefaultNetworkSimulationConfig());
       return send_transport_;
     }
 
@@ -239,7 +240,7 @@ TEST_P(ProbingEndToEndTest, ProbeOnVideoEncoderReconfiguration) {
             // bitrate).
             if (stats.send_bandwidth_bps >= 250000 &&
                 stats.send_bandwidth_bps <= 350000) {
-              FakeNetworkPipe::Config config;
+              DefaultNetworkSimulationConfig config;
               config.link_capacity_kbps = 200;
               send_transport_->SetConfig(config);
 
@@ -254,7 +255,7 @@ TEST_P(ProbingEndToEndTest, ProbeOnVideoEncoderReconfiguration) {
             break;
           case 1:
             if (stats.send_bandwidth_bps <= 210000) {
-              FakeNetworkPipe::Config config;
+              DefaultNetworkSimulationConfig config;
               config.link_capacity_kbps = 5000;
               send_transport_->SetConfig(config);
 
diff --git a/video/end_to_end_tests/receive_time_tests.cc b/video/end_to_end_tests/receive_time_tests.cc
index 9e025cb553..c676d666b6 100644
--- a/video/end_to_end_tests/receive_time_tests.cc
+++ b/video/end_to_end_tests/receive_time_tests.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/timeutils.h"
 #include "test/call_test.h"
@@ -93,7 +94,8 @@ class ReportedReceiveTimeTester : public test::EndToEndTest {
       Call* sender_call) override {
     return send_transport_ = new test::PacketTransport(
                task_queue, sender_call, this, test::PacketTransport::kSender,
-               test::CallTest::payload_type_map_, FakeNetworkPipe::Config());
+               test::CallTest::payload_type_map_,
+               DefaultNetworkSimulationConfig());
   }
   void PerformTest() override {
     observation_complete_.Wait(test::CallTest::kDefaultTimeoutMs);
diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc
index 7145b02211..8f30f14b00 100644
--- a/video/end_to_end_tests/retransmission_tests.cc
+++ b/video/end_to_end_tests/retransmission_tests.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/call_test.h"
@@ -137,7 +138,7 @@ TEST_P(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
         test::SingleThreadedTaskQueueForTesting* task_queue) override {
       test::PacketTransport* receive_transport = new test::PacketTransport(
           task_queue, nullptr, this, test::PacketTransport::kReceiver,
-          payload_type_map_, FakeNetworkPipe::Config());
+          payload_type_map_, DefaultNetworkSimulationConfig());
       receive_transport_ = receive_transport;
       return receive_transport;
     }
diff --git a/video/end_to_end_tests/rtp_rtcp_tests.cc b/video/end_to_end_tests/rtp_rtcp_tests.cc
index c8eb802ce9..c9740650f6 100644
--- a/video/end_to_end_tests/rtp_rtcp_tests.cc
+++ b/video/end_to_end_tests/rtp_rtcp_tests.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "test/call_test.h"
 #include "test/gtest.h"
@@ -277,10 +278,10 @@ void RtpRtcpEndToEndTest::TestRtpStatePreservation(
     send_transport = absl::make_unique<test::PacketTransport>(
         &task_queue_, sender_call_.get(), &observer,
         test::PacketTransport::kSender, payload_type_map_,
-        FakeNetworkPipe::Config());
+        DefaultNetworkSimulationConfig());
     receive_transport = absl::make_unique<test::PacketTransport>(
         &task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
-        payload_type_map_, FakeNetworkPipe::Config());
+        payload_type_map_, DefaultNetworkSimulationConfig());
     send_transport->SetReceiver(receiver_call_->Receiver());
     receive_transport->SetReceiver(sender_call_->Receiver());
 
@@ -465,7 +466,7 @@ TEST_F(RtpRtcpEndToEndTest, TestFlexfecRtpStatePreservation) {
   task_queue_.SendTask([&]() {
     CreateCalls();
 
-    FakeNetworkPipe::Config lossy_delayed_link;
+    DefaultNetworkSimulationConfig lossy_delayed_link;
     lossy_delayed_link.loss_percent = 2;
     lossy_delayed_link.queue_delay_ms = 50;
 
@@ -474,7 +475,7 @@ TEST_F(RtpRtcpEndToEndTest, TestFlexfecRtpStatePreservation) {
         test::PacketTransport::kSender, payload_type_map_, lossy_delayed_link);
     send_transport->SetReceiver(receiver_call_->Receiver());
 
-    FakeNetworkPipe::Config flawless_link;
+    DefaultNetworkSimulationConfig flawless_link;
     receive_transport = absl::make_unique<test::PacketTransport>(
         &task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
         payload_type_map_, flawless_link);
diff --git a/video/end_to_end_tests/stats_tests.cc b/video/end_to_end_tests/stats_tests.cc
index ae259111d8..c1b45b0bc7 100644
--- a/video/end_to_end_tests/stats_tests.cc
+++ b/video/end_to_end_tests/stats_tests.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/simulated_network.h"
 #include "modules/rtp_rtcp/source/rtp_utility.h"
 #include "modules/video_coding/include/video_coding_defines.h"
 #include "system_wrappers/include/metrics.h"
@@ -228,7 +229,7 @@ TEST_F(StatsEndToEndTest, GetStats) {
     test::PacketTransport* CreateSendTransport(
         test::SingleThreadedTaskQueueForTesting* task_queue,
         Call* sender_call) override {
-      FakeNetworkPipe::Config network_config;
+      DefaultNetworkSimulationConfig network_config;
       network_config.loss_percent = 5;
       return new test::PacketTransport(task_queue, sender_call, this,
                                        test::PacketTransport::kSender,
@@ -707,7 +708,7 @@ TEST_F(StatsEndToEndTest, CallReportsRttForSender) {
   std::unique_ptr<test::DirectTransport> receiver_transport;
 
   task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
-    FakeNetworkPipe::Config config;
+    DefaultNetworkSimulationConfig config;
     config.queue_delay_ms = kSendDelayMs;
     CreateCalls();
     sender_transport = absl::make_unique<test::DirectTransport>(
diff --git a/video/picture_id_tests.cc b/video/picture_id_tests.cc
index 3c43d7d112..60f11ca4e6 100644
--- a/video/picture_id_tests.cc
+++ b/video/picture_id_tests.cc
@@ -7,6 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+
+#include "api/test/simulated_network.h"
 #include "media/engine/internalencoderfactory.h"
 #include "media/engine/simulcast_encoder_adapter.h"
 #include "modules/rtp_rtcp/source/rtp_format.h"
@@ -290,7 +292,7 @@ void PictureIdTest::SetupEncoder(VideoEncoderFactory* encoder_factory,
     send_transport_.reset(new test::PacketTransport(
         &task_queue_, sender_call_.get(), observer_.get(),
         test::PacketTransport::kSender, payload_type_map_,
-        FakeNetworkPipe::Config()));
+        DefaultNetworkSimulationConfig()));
 
     CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport_.get());
     GetVideoSendConfig()->encoder_settings.encoder_factory = encoder_factory;
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 4bf5b99035..7a2ded0e15 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -11,6 +11,7 @@
 #include <memory>
 #include <vector>
 
+#include "api/test/simulated_network.h"
 #include "call/call.h"
 #include "call/rtp_transport_controller_send.h"
 #include "common_video/include/frame_callback.h"
@@ -524,7 +525,7 @@ class UlpfecObserver : public test::EndToEndTest {
     // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
     // Configure some network delay.
     const int kNetworkDelayMs = 100;
-    FakeNetworkPipe::Config config;
+    DefaultNetworkSimulationConfig config;
     config.loss_percent = 5;
     config.queue_delay_ms = kNetworkDelayMs;
     return new test::PacketTransport(
@@ -711,7 +712,7 @@ class FlexfecObserver : public test::EndToEndTest {
     // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
     // Therefore we need some network delay.
     const int kNetworkDelayMs = 100;
-    FakeNetworkPipe::Config config;
+    DefaultNetworkSimulationConfig config;
     config.loss_percent = 5;
     config.queue_delay_ms = kNetworkDelayMs;
     return new test::PacketTransport(
@@ -1455,7 +1456,7 @@ TEST_F(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) {
         test::SingleThreadedTaskQueueForTesting* task_queue,
         Call* sender_call) override {
       const int kNetworkDelayMs = 50;
-      FakeNetworkPipe::Config config;
+      DefaultNetworkSimulationConfig config;
       config.loss_percent = 10;
       config.link_capacity_kbps = kCapacityKbps;
       config.queue_delay_ms = kNetworkDelayMs;