diff --git a/api/BUILD.gn b/api/BUILD.gn
index f8a6532866..b736f93885 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -1223,3 +1223,10 @@ if (rtc_include_tests) {
     ]
   }
 }
+
+rtc_source_set("webrtc_key_value_config") {
+  visibility = [ "*" ]
+  sources = [ "webrtc_key_value_config.h" ]
+  deps = [ "../rtc_base/system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 30955273b0..fb0fcf7a64 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -48,7 +48,10 @@ rtc_library("network_control") {
 rtc_source_set("webrtc_key_value_config") {
   visibility = [ "*" ]
   sources = [ "webrtc_key_value_config.h" ]
-  deps = [ "../../rtc_base/system:rtc_export" ]
+  deps = [
+    "../../api:webrtc_key_value_config",
+    "../../rtc_base/system:rtc_export",
+  ]
   absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
 
diff --git a/api/transport/webrtc_key_value_config.h b/api/transport/webrtc_key_value_config.h
index 5666a82783..f666873cfe 100644
--- a/api/transport/webrtc_key_value_config.h
+++ b/api/transport/webrtc_key_value_config.h
@@ -10,24 +10,8 @@
 #ifndef API_TRANSPORT_WEBRTC_KEY_VALUE_CONFIG_H_
 #define API_TRANSPORT_WEBRTC_KEY_VALUE_CONFIG_H_
 
-#include <string>
-
-#include "absl/strings/string_view.h"
-#include "rtc_base/system/rtc_export.h"
-
-namespace webrtc {
-
-// An interface that provides a key-value mapping for configuring internal
-// details of WebRTC. Note that there's no guarantess that the meaning of a
-// particular key value mapping will be preserved over time and no announcements
-// will be made if they are changed. It's up to the library user to ensure that
-// the behavior does not break.
-class RTC_EXPORT WebRtcKeyValueConfig {
- public:
-  virtual ~WebRtcKeyValueConfig() = default;
-  // The configured value for the given key. Defaults to an empty string.
-  virtual std::string Lookup(absl::string_view key) const = 0;
-};
-}  // namespace webrtc
+// TODO(bugs.webrtc.org/10335): Remove once all migrated to
+// "api/webrtc_key_value_config.h".
+#include "api/webrtc_key_value_config.h"
 
 #endif  // API_TRANSPORT_WEBRTC_KEY_VALUE_CONFIG_H_
diff --git a/api/webrtc_key_value_config.h b/api/webrtc_key_value_config.h
new file mode 100644
index 0000000000..32a2e47eed
--- /dev/null
+++ b/api/webrtc_key_value_config.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2019 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 API_WEBRTC_KEY_VALUE_CONFIG_H_
+#define API_WEBRTC_KEY_VALUE_CONFIG_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// An interface that provides a key-value mapping for configuring internal
+// details of WebRTC. Note that there's no guarantess that the meaning of a
+// particular key value mapping will be preserved over time and no announcements
+// will be made if they are changed. It's up to the library user to ensure that
+// the behavior does not break.
+class RTC_EXPORT WebRtcKeyValueConfig {
+ public:
+  virtual ~WebRtcKeyValueConfig() = default;
+  // The configured value for the given key. Defaults to an empty string.
+  virtual std::string Lookup(absl::string_view key) const = 0;
+
+  bool IsEnabled(absl::string_view key) const {
+    return Lookup(key).find("Enabled") == 0;
+  }
+
+  bool IsDisabled(absl::string_view key) const {
+    return Lookup(key).find("Disabled") == 0;
+  }
+};
+}  // namespace webrtc
+
+#endif  // API_WEBRTC_KEY_VALUE_CONFIG_H_
diff --git a/call/bitrate_allocator.cc b/call/bitrate_allocator.cc
index 1693661ef5..2684a1650e 100644
--- a/call/bitrate_allocator.cc
+++ b/call/bitrate_allocator.cc
@@ -23,7 +23,6 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_minmax.h"
 #include "system_wrappers/include/clock.h"
-#include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 
 namespace webrtc {
diff --git a/call/call.cc b/call/call.cc
index bb62365cd4..2638b22b31 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -61,7 +61,6 @@
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/cpu_info.h"
-#include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 #include "video/call_stats2.h"
 #include "video/send_delay_stats.h"
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/modules/congestion_controller/rtp/transport_feedback_adapter.cc
index 87691bf263..87e917e978 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter.cc
@@ -22,7 +22,6 @@
 #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
diff --git a/modules/pacing/paced_sender_unittest.cc b/modules/pacing/paced_sender_unittest.cc
index 53cc1c42ed..7abd532417 100644
--- a/modules/pacing/paced_sender_unittest.cc
+++ b/modules/pacing/paced_sender_unittest.cc
@@ -19,7 +19,6 @@
 #include "modules/pacing/packet_router.h"
 #include "modules/utility/include/mock/mock_process_thread.h"
 #include "system_wrappers/include/clock.h"
-#include "system_wrappers/include/field_trial.h"
 #include "test/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/modules/remote_bitrate_estimator/bwe_defines.cc b/modules/remote_bitrate_estimator/bwe_defines.cc
index 6afbe133e2..22c605e82d 100644
--- a/modules/remote_bitrate_estimator/bwe_defines.cc
+++ b/modules/remote_bitrate_estimator/bwe_defines.cc
@@ -10,8 +10,6 @@
 
 #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
 
-#include "system_wrappers/include/field_trial.h"
-
 namespace webrtc {
 
 const char kBweTypeHistogram[] = "WebRTC.BWE.Types";
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index a9c059bed3..66da247ade 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -91,6 +91,7 @@ rtc_library("rtc_p2p") {
     "../api:rtc_error",
     "../api:scoped_refptr",
     "../api:sequence_checker",
+    "../api:webrtc_key_value_config",
     "../api:wrapping_async_dns_resolver",
     "../api/crypto:options",
     "../api/rtc_event_log",
@@ -240,6 +241,7 @@ if (rtc_include_tests) {
       "../api:mock_async_dns_resolver",
       "../api:packet_socket_factory",
       "../api:scoped_refptr",
+      "../api:webrtc_key_value_config",
       "../api/transport:stun_types",
       "../api/units:time_delta",
       "../rtc_base",
@@ -259,6 +261,7 @@ if (rtc_include_tests) {
       "../system_wrappers:metrics",
       "../test:field_trial",
       "../test:rtc_expect_death",
+      "../test:scoped_key_value_config",
       "../test:test_support",
       "//testing/gtest",
     ]
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index 5d3552b7f6..f864150969 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -34,7 +34,6 @@
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/third_party/base64/base64.h"
 #include "rtc_base/trace_event.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace {
 
diff --git a/p2p/base/stun_request.cc b/p2p/base/stun_request.cc
index 09a7a8345e..ed94ccb5fc 100644
--- a/p2p/base/stun_request.cc
+++ b/p2p/base/stun_request.cc
@@ -19,7 +19,6 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/string_encode.h"
 #include "rtc_base/time_utils.h"  // For TimeMillis
-#include "system_wrappers/include/field_trial.h"
 
 namespace cricket {
 
diff --git a/p2p/base/transport_description_factory.cc b/p2p/base/transport_description_factory.cc
index e46114ed83..18c4a28c2b 100644
--- a/p2p/base/transport_description_factory.cc
+++ b/p2p/base/transport_description_factory.cc
@@ -21,8 +21,9 @@
 
 namespace cricket {
 
-TransportDescriptionFactory::TransportDescriptionFactory()
-    : secure_(SEC_DISABLED) {}
+TransportDescriptionFactory::TransportDescriptionFactory(
+    const webrtc::WebRtcKeyValueConfig& field_trials)
+    : secure_(SEC_DISABLED), field_trials_(field_trials) {}
 
 TransportDescriptionFactory::~TransportDescriptionFactory() = default;
 
diff --git a/p2p/base/transport_description_factory.h b/p2p/base/transport_description_factory.h
index 0be7f32929..46f1c2f9fa 100644
--- a/p2p/base/transport_description_factory.h
+++ b/p2p/base/transport_description_factory.h
@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "api/webrtc_key_value_config.h"
 #include "p2p/base/ice_credentials_iterator.h"
 #include "p2p/base/transport_description.h"
 #include "rtc_base/rtc_certificate.h"
@@ -37,7 +38,8 @@ struct TransportOptions {
 class TransportDescriptionFactory {
  public:
   // Default ctor; use methods below to set configuration.
-  TransportDescriptionFactory();
+  explicit TransportDescriptionFactory(
+      const webrtc::WebRtcKeyValueConfig& field_trials);
   ~TransportDescriptionFactory();
 
   SecurePolicy secure() const { return secure_; }
@@ -73,12 +75,15 @@ class TransportDescriptionFactory {
       const TransportDescription* current_description,
       IceCredentialsIterator* ice_credentials) const;
 
+  const webrtc::WebRtcKeyValueConfig& trials() const { return field_trials_; }
+
  private:
   bool SetSecurityInfo(TransportDescription* description,
                        ConnectionRole role) const;
 
   SecurePolicy secure_;
   rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
+  const webrtc::WebRtcKeyValueConfig& field_trials_;
 };
 
 }  // namespace cricket
diff --git a/p2p/base/transport_description_factory_unittest.cc b/p2p/base/transport_description_factory_unittest.cc
index 77a56eff26..75e6e8ee09 100644
--- a/p2p/base/transport_description_factory_unittest.cc
+++ b/p2p/base/transport_description_factory_unittest.cc
@@ -25,6 +25,7 @@
 #include "rtc_base/ssl_identity.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 using cricket::TransportDescription;
 using cricket::TransportDescriptionFactory;
@@ -36,6 +37,8 @@ class TransportDescriptionFactoryTest : public ::testing::Test {
  public:
   TransportDescriptionFactoryTest()
       : ice_credentials_({}),
+        f1_(field_trials_),
+        f2_(field_trials_),
         cert1_(rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
             new rtc::FakeSSLIdentity("User1")))),
         cert2_(rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
@@ -156,6 +159,7 @@ class TransportDescriptionFactoryTest : public ::testing::Test {
     }
   }
 
+  webrtc::test::ScopedKeyValueConfig field_trials_;
   cricket::IceCredentialsIterator ice_credentials_;
   TransportDescriptionFactory f1_;
   TransportDescriptionFactory f2_;
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 32d114444f..f88226a51c 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -143,6 +143,7 @@ rtc_library("rtc_pc_base") {
     "../api:scoped_refptr",
     "../api:sequence_checker",
     "../api:video_track_source_constraints",
+    "../api:webrtc_key_value_config",
     "../api/crypto:options",
     "../api/rtc_event_log",
     "../api/task_queue",
@@ -813,6 +814,7 @@ rtc_source_set("stats_collector") {
     "../api:rtp_parameters",
     "../api:scoped_refptr",
     "../api:sequence_checker",
+    "../api:webrtc_key_value_config",
     "../api/audio_codecs:audio_codecs_api",
     "../api/video:video_rtp_headers",
     "../call:call_interfaces",
@@ -829,7 +831,6 @@ rtc_source_set("stats_collector") {
     "../rtc_base:stringutils",
     "../rtc_base:threading",
     "../rtc_base:timeutils",
-    "../system_wrappers:field_trial",
   ]
   absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
@@ -906,6 +907,7 @@ rtc_source_set("webrtc_session_description_factory") {
     "webrtc_session_description_factory.h",
   ]
   deps = [
+    ":connection_context",
     ":rtc_pc_base",
     ":sdp_state_provider",
     ":session_description",
@@ -1581,7 +1583,8 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base/task_utils:to_queued_task",
       "../rtc_base/third_party/sigslot",
       "../system_wrappers:metrics",
-      "../test:field_trial",
+      "../test:explicit_key_value_config",
+      "../test:scoped_key_value_config",
       "../test:test_common",
       "../test:test_main",
       "../test:test_support",
@@ -1767,6 +1770,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:rtc_error",
       "../api:rtp_transceiver_direction",
       "../api:scoped_refptr",
+      "../api:webrtc_key_value_config",
       "../api/adaptation:resource_adaptation_api",
       "../api/audio:audio_mixer_api",
       "../api/crypto:frame_decryptor_interface",
@@ -1817,9 +1821,10 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base/third_party/sigslot",
       "../system_wrappers:field_trial",
       "../system_wrappers:metrics",
-      "../test:field_trial",
+      "../test:explicit_key_value_config",
       "../test:fileutils",
       "../test:rtp_test_utils",
+      "../test:scoped_key_value_config",
       "../test:test_common",
       "../test/pc/sctp:fake_sctp_transport",
       "./scenario_tests:pc_scenario_tests",
@@ -1994,9 +1999,10 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base/third_party/base64",
       "../rtc_base/third_party/sigslot",
       "../system_wrappers:metrics",
-      "../test:field_trial",
+      "../test:explicit_key_value_config",
       "../test:fileutils",
       "../test:rtp_test_utils",
+      "../test:scoped_key_value_config",
       "../test:test_support",
       "../test/pc/sctp:fake_sctp_transport",
     ]
@@ -2054,10 +2060,12 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:rtc_stats_api",
       "../api:scoped_refptr",
       "../api:sequence_checker",
+      "../api:webrtc_key_value_config",
       "../api/audio:audio_mixer_api",
       "../api/audio_codecs:audio_codecs_api",
       "../api/task_queue",
       "../api/task_queue:default_task_queue_factory",
+      "../api/transport:webrtc_key_value_config",
       "../api/video:builtin_video_bitrate_allocator_factory",
       "../api/video:video_frame",
       "../api/video:video_rtp_headers",
@@ -2084,6 +2092,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base/synchronization:mutex",
       "../rtc_base/task_utils:repeating_task",
       "../rtc_base/third_party/sigslot",
+      "../test:scoped_key_value_config",
       "../test:test_support",
       "../test:video_test_common",
     ]
diff --git a/pc/channel_manager_unittest.cc b/pc/channel_manager_unittest.cc
index 9503243a09..7d7d3e475f 100644
--- a/pc/channel_manager_unittest.cc
+++ b/pc/channel_manager_unittest.cc
@@ -24,6 +24,7 @@
 #include "rtc_base/location.h"
 #include "rtc_base/thread.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 namespace cricket {
 namespace {
@@ -88,6 +89,7 @@ class ChannelManagerTest : public ::testing::Test {
       video_bitrate_allocator_factory_;
   std::unique_ptr<cricket::ChannelManager> cm_;
   cricket::FakeCall fake_call_;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
 };
 
 TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
@@ -123,7 +125,8 @@ TEST_F(ChannelManagerTest, CreateDestroyChannels) {
       "fake_dtls_transport", cricket::ICE_CANDIDATE_COMPONENT_RTP,
       network_.get());
   auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
-      /*rtcp_mux_required=*/true);
+      /*rtcp_mux_required=*/true, field_trials_);
+
   network_->Invoke<void>(
       RTC_FROM_HERE, [&rtp_dtls_transport, &dtls_srtp_transport] {
         dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport.get(),
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 3bf9556403..463f60bf35 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -46,6 +46,7 @@
 #include "rtc_base/task_utils/to_queued_task.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 using cricket::DtlsTransportInternal;
 using cricket::FakeVoiceMediaChannel;
@@ -317,7 +318,7 @@ class ChannelTest : public ::testing::Test, public sigslot::has_slots<> {
       cricket::DtlsTransportInternal* rtp_dtls_transport,
       cricket::DtlsTransportInternal* rtcp_dtls_transport) {
     auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
-        rtcp_dtls_transport == nullptr);
+        rtcp_dtls_transport == nullptr, field_trials_);
 
     network_thread_->Invoke<void>(
         RTC_FROM_HERE,
@@ -1441,6 +1442,7 @@ class ChannelTest : public ::testing::Test, public sigslot::has_slots<> {
   rtc::Buffer rtcp_packet_;
   cricket::CandidatePairInterface* last_selected_candidate_pair_;
   rtc::UniqueRandomIdGenerator ssrc_generator_;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
 };
 
 template <>
diff --git a/pc/connection_context.cc b/pc/connection_context.cc
index d093ee3cf5..31810bca62 100644
--- a/pc/connection_context.cc
+++ b/pc/connection_context.cc
@@ -73,7 +73,8 @@ rtc::Thread* MaybeWrapThread(rtc::Thread* signaling_thread,
 
 std::unique_ptr<SctpTransportFactoryInterface> MaybeCreateSctpFactory(
     std::unique_ptr<SctpTransportFactoryInterface> factory,
-    rtc::Thread* network_thread) {
+    rtc::Thread* network_thread,
+    const WebRtcKeyValueConfig& field_trials) {
   if (factory) {
     return factory;
   }
@@ -102,15 +103,15 @@ ConnectionContext::ConnectionContext(
                                             owned_worker_thread_)),
       signaling_thread_(MaybeWrapThread(dependencies->signaling_thread,
                                         wraps_current_thread_)),
+      trials_(dependencies->trials ? std::move(dependencies->trials)
+                                   : std::make_unique<FieldTrialBasedConfig>()),
       network_monitor_factory_(
           std::move(dependencies->network_monitor_factory)),
       call_factory_(std::move(dependencies->call_factory)),
       sctp_factory_(
           MaybeCreateSctpFactory(std::move(dependencies->sctp_factory),
-                                 network_thread())),
-      trials_(dependencies->trials
-                  ? std::move(dependencies->trials)
-                  : std::make_unique<FieldTrialBasedConfig>()) {
+                                 network_thread(),
+                                 *trials_.get())) {
   signaling_thread_->AllowInvokesToThread(worker_thread_);
   signaling_thread_->AllowInvokesToThread(network_thread_);
   worker_thread_->AllowInvokesToThread(network_thread_);
diff --git a/pc/connection_context.h b/pc/connection_context.h
index 2aaa840df1..05f838fac4 100644
--- a/pc/connection_context.h
+++ b/pc/connection_context.h
@@ -112,6 +112,10 @@ class ConnectionContext final
   rtc::Thread* const network_thread_;
   rtc::Thread* const worker_thread_;
   rtc::Thread* const signaling_thread_;
+
+  // Accessed both on signaling thread and worker thread.
+  std::unique_ptr<WebRtcKeyValueConfig> const trials_;
+
   // channel_manager is accessed both on signaling thread and worker thread.
   std::unique_ptr<cricket::ChannelManager> channel_manager_;
   std::unique_ptr<rtc::NetworkMonitorFactory> const network_monitor_factory_
@@ -124,8 +128,6 @@ class ConnectionContext final
   std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_
       RTC_GUARDED_BY(signaling_thread_);
   std::unique_ptr<SctpTransportFactoryInterface> const sctp_factory_;
-  // Accessed both on signaling thread and worker thread.
-  std::unique_ptr<WebRtcKeyValueConfig> const trials_;
 };
 
 }  // namespace webrtc
diff --git a/pc/data_channel_integrationtest.cc b/pc/data_channel_integrationtest.cc
index c76a4d97b0..6d96251ac1 100644
--- a/pc/data_channel_integrationtest.cc
+++ b/pc/data_channel_integrationtest.cc
@@ -39,7 +39,6 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/virtual_socket_server.h"
-#include "system_wrappers/include/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -845,7 +844,7 @@ TEST_P(DataChannelIntegrationTest,
   EXPECT_GT(202u, callee()->data_observer()->received_message_count());
   EXPECT_LE(2u, callee()->data_observer()->received_message_count());
   // Then, check that observed behavior (lose some messages) has not changed
-  if (!webrtc::field_trial::IsDisabled("WebRTC-DataChannel-Dcsctp")) {
+  if (!trials().IsDisabled("WebRTC-DataChannel-Dcsctp")) {
     // DcSctp loses all messages. This is correct.
     EXPECT_EQ(2u, callee()->data_observer()->received_message_count());
   } else {
diff --git a/pc/dtls_srtp_transport.cc b/pc/dtls_srtp_transport.cc
index 9a4135bb2f..9ec14f530b 100644
--- a/pc/dtls_srtp_transport.cc
+++ b/pc/dtls_srtp_transport.cc
@@ -27,8 +27,9 @@ static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
 
 namespace webrtc {
 
-DtlsSrtpTransport::DtlsSrtpTransport(bool rtcp_mux_enabled)
-    : SrtpTransport(rtcp_mux_enabled) {}
+DtlsSrtpTransport::DtlsSrtpTransport(bool rtcp_mux_enabled,
+                                     const WebRtcKeyValueConfig& field_trials)
+    : SrtpTransport(rtcp_mux_enabled, field_trials) {}
 
 void DtlsSrtpTransport::SetDtlsTransports(
     cricket::DtlsTransportInternal* rtp_dtls_transport,
diff --git a/pc/dtls_srtp_transport.h b/pc/dtls_srtp_transport.h
index c2c51c22f2..2ee6e02e30 100644
--- a/pc/dtls_srtp_transport.h
+++ b/pc/dtls_srtp_transport.h
@@ -32,7 +32,8 @@ namespace webrtc {
 // configures the SrtpSessions in the base class.
 class DtlsSrtpTransport : public SrtpTransport {
  public:
-  explicit DtlsSrtpTransport(bool rtcp_mux_enabled);
+  DtlsSrtpTransport(bool rtcp_mux_enabled,
+                    const WebRtcKeyValueConfig& field_trials);
 
   // Set P2P layer RTP/RTCP DtlsTransports. When using RTCP-muxing,
   // `rtcp_dtls_transport` is null.
diff --git a/pc/dtls_srtp_transport_unittest.cc b/pc/dtls_srtp_transport_unittest.cc
index 76d9c30c5e..30f09e740c 100644
--- a/pc/dtls_srtp_transport_unittest.cc
+++ b/pc/dtls_srtp_transport_unittest.cc
@@ -31,6 +31,7 @@
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/third_party/sigslot/sigslot.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 using cricket::FakeDtlsTransport;
 using cricket::FakeIceTransport;
@@ -59,7 +60,7 @@ class DtlsSrtpTransportTest : public ::testing::Test,
       FakeDtlsTransport* rtcp_dtls,
       bool rtcp_mux_enabled) {
     auto dtls_srtp_transport =
-        std::make_unique<DtlsSrtpTransport>(rtcp_mux_enabled);
+        std::make_unique<DtlsSrtpTransport>(rtcp_mux_enabled, field_trials_);
 
     dtls_srtp_transport->SetDtlsTransports(rtp_dtls, rtcp_dtls);
 
@@ -256,6 +257,7 @@ class DtlsSrtpTransportTest : public ::testing::Test,
   webrtc::TransportObserver transport_observer2_;
 
   int sequence_number_ = 0;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
 };
 
 // Tests that if RTCP muxing is enabled and transports are set after RTP
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index e63742aaee..6c69130a38 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -62,6 +62,7 @@ JsepTransportController::JsepTransportController(
   RTC_DCHECK(config_.rtcp_handler);
   RTC_DCHECK(config_.ice_transport_factory);
   RTC_DCHECK(config_.on_dtls_handshake_error_);
+  RTC_DCHECK(config_.field_trials);
 }
 
 JsepTransportController::~JsepTransportController() {
@@ -477,8 +478,8 @@ JsepTransportController::CreateSdesTransport(
     cricket::DtlsTransportInternal* rtp_dtls_transport,
     cricket::DtlsTransportInternal* rtcp_dtls_transport) {
   RTC_DCHECK_RUN_ON(network_thread_);
-  auto srtp_transport =
-      std::make_unique<webrtc::SrtpTransport>(rtcp_dtls_transport == nullptr);
+  auto srtp_transport = std::make_unique<webrtc::SrtpTransport>(
+      rtcp_dtls_transport == nullptr, *config_.field_trials);
   RTC_DCHECK(rtp_dtls_transport);
   srtp_transport->SetRtpPacketTransport(rtp_dtls_transport);
   if (rtcp_dtls_transport) {
@@ -497,7 +498,7 @@ JsepTransportController::CreateDtlsSrtpTransport(
     cricket::DtlsTransportInternal* rtcp_dtls_transport) {
   RTC_DCHECK_RUN_ON(network_thread_);
   auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
-      rtcp_dtls_transport == nullptr);
+      rtcp_dtls_transport == nullptr, *config_.field_trials);
   if (config_.enable_external_auth) {
     dtls_srtp_transport->EnableExternalAuth();
   }
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index ed4d20ba84..44d5981684 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -137,6 +137,9 @@ class JsepTransportController : public sigslot::has_slots<> {
     // Factory for SCTP transports.
     SctpTransportFactoryInterface* sctp_factory = nullptr;
     std::function<void(rtc::SSLHandshakeError)> on_dtls_handshake_error_;
+
+    // Field trials.
+    const webrtc::WebRtcKeyValueConfig* field_trials;
   };
 
   // The ICE related events are fired on the `network_thread`.
diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc
index 622a9b90e3..ee306f093a 100644
--- a/pc/jsep_transport_controller_unittest.cc
+++ b/pc/jsep_transport_controller_unittest.cc
@@ -33,6 +33,7 @@
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/thread.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 using cricket::Candidate;
 using cricket::Candidates;
@@ -100,6 +101,7 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
     config.ice_transport_factory = fake_ice_transport_factory_.get();
     config.dtls_transport_factory = fake_dtls_transport_factory_.get();
     config.on_dtls_handshake_error_ = [](rtc::SSLHandshakeError s) {};
+    config.field_trials = &field_trials_;
     transport_controller_ = std::make_unique<JsepTransportController>(
         network_thread, port_allocator, nullptr /* async_resolver_factory */,
         config);
@@ -378,6 +380,7 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
   // Transport controller needs to be destroyed first, because it may issue
   // callbacks that modify the changed_*_by_mid in the destructor.
   std::unique_ptr<JsepTransportController> transport_controller_;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
 };
 
 TEST_F(JsepTransportControllerTest, GetRtpTransport) {
diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc
index 2ec96d2af7..c63dc496db 100644
--- a/pc/jsep_transport_unittest.cc
+++ b/pc/jsep_transport_unittest.cc
@@ -37,6 +37,7 @@
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/third_party/sigslot/sigslot.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 namespace cricket {
 namespace {
@@ -89,7 +90,7 @@ class JsepTransport2Test : public ::testing::Test, public sigslot::has_slots<> {
       rtc::PacketTransportInternal* rtp_packet_transport,
       rtc::PacketTransportInternal* rtcp_packet_transport) {
     auto srtp_transport = std::make_unique<webrtc::SrtpTransport>(
-        rtcp_packet_transport == nullptr);
+        rtcp_packet_transport == nullptr, field_trials_);
 
     srtp_transport->SetRtpPacketTransport(rtp_packet_transport);
     if (rtcp_packet_transport) {
@@ -102,7 +103,7 @@ class JsepTransport2Test : public ::testing::Test, public sigslot::has_slots<> {
       cricket::DtlsTransportInternal* rtp_dtls_transport,
       cricket::DtlsTransportInternal* rtcp_dtls_transport) {
     auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
-        rtcp_dtls_transport == nullptr);
+        rtcp_dtls_transport == nullptr, field_trials_);
     dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport,
                                            rtcp_dtls_transport);
     return dtls_srtp_transport;
@@ -192,6 +193,8 @@ class JsepTransport2Test : public ::testing::Test, public sigslot::has_slots<> {
   // The SrtpTransport is owned by `jsep_transport_`. Keep a raw pointer here
   // for testing.
   webrtc::SrtpTransport* sdes_transport_ = nullptr;
+
+  webrtc::test::ScopedKeyValueConfig field_trials_;
 };
 
 // The parameterized tests cover both cases when RTCP mux is enable and
diff --git a/pc/media_session.cc b/pc/media_session.cc
index 2a3a698874..9aeae708d7 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -39,7 +39,6 @@
 #include "rtc_base/string_encode.h"
 #include "rtc_base/third_party/base64/base64.h"
 #include "rtc_base/unique_id_generator.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace {
 
@@ -301,7 +300,8 @@ static StreamParams CreateStreamParamsForNewSenderWithSsrcs(
     const std::string& rtcp_cname,
     bool include_rtx_streams,
     bool include_flexfec_stream,
-    UniqueRandomIdGenerator* ssrc_generator) {
+    UniqueRandomIdGenerator* ssrc_generator,
+    const webrtc::WebRtcKeyValueConfig& field_trials) {
   StreamParams result;
   result.id = sender.track_id;
 
@@ -313,8 +313,7 @@ static StreamParams CreateStreamParamsForNewSenderWithSsrcs(
            "a single media streams. This session has multiple "
            "media streams however, so no FlexFEC SSRC will be generated.";
   }
-  if (include_flexfec_stream &&
-      !webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03")) {
+  if (include_flexfec_stream && !field_trials.IsEnabled("WebRTC-FlexFEC-03")) {
     include_flexfec_stream = false;
     RTC_LOG(LS_WARNING)
         << "WebRTC-FlexFEC trial is not enabled, not sending FlexFEC";
@@ -396,12 +395,12 @@ static void AddSimulcastToMediaDescription(
 // content_description.
 // `current_params` - All currently known StreamParams of any media type.
 template <class C>
-static bool AddStreamParams(
-    const std::vector<SenderOptions>& sender_options,
-    const std::string& rtcp_cname,
-    UniqueRandomIdGenerator* ssrc_generator,
-    StreamParamsVec* current_streams,
-    MediaContentDescriptionImpl<C>* content_description) {
+static bool AddStreamParams(const std::vector<SenderOptions>& sender_options,
+                            const std::string& rtcp_cname,
+                            UniqueRandomIdGenerator* ssrc_generator,
+                            StreamParamsVec* current_streams,
+                            MediaContentDescriptionImpl<C>* content_description,
+                            const webrtc::WebRtcKeyValueConfig& field_trials) {
   // SCTP streams are not negotiated using SDP/ContentDescriptions.
   if (IsSctpProtocol(content_description->protocol())) {
     return true;
@@ -423,7 +422,7 @@ static bool AddStreamParams(
               // Signal SSRCs and legacy simulcast (if requested).
               CreateStreamParamsForNewSenderWithSsrcs(
                   sender, rtcp_cname, include_rtx_streams,
-                  include_flexfec_stream, ssrc_generator)
+                  include_flexfec_stream, ssrc_generator, field_trials)
               :
               // Signal RIDs and spec-compliant simulcast (if requested).
               CreateStreamParamsForNewSenderWithRids(sender, rtcp_cname);
@@ -711,11 +710,12 @@ static bool CreateMediaContentOffer(
     const RtpHeaderExtensions& rtp_extensions,
     UniqueRandomIdGenerator* ssrc_generator,
     StreamParamsVec* current_streams,
-    MediaContentDescriptionImpl<C>* offer) {
+    MediaContentDescriptionImpl<C>* offer,
+    const webrtc::WebRtcKeyValueConfig& field_trials) {
   offer->AddCodecs(codecs);
   if (!AddStreamParams(media_description_options.sender_options,
                        session_options.rtcp_cname, ssrc_generator,
-                       current_streams, offer)) {
+                       current_streams, offer, field_trials)) {
     return false;
   }
 
@@ -1351,7 +1351,8 @@ static bool SetCodecsInAnswer(
     const MediaSessionOptions& session_options,
     UniqueRandomIdGenerator* ssrc_generator,
     StreamParamsVec* current_streams,
-    MediaContentDescriptionImpl<C>* answer) {
+    MediaContentDescriptionImpl<C>* answer,
+    const webrtc::WebRtcKeyValueConfig& field_trials) {
   std::vector<C> negotiated_codecs;
   NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs,
                   media_description_options.codec_preferences.empty());
@@ -1359,7 +1360,7 @@ static bool SetCodecsInAnswer(
   answer->set_protocol(offer->protocol());
   if (!AddStreamParams(media_description_options.sender_options,
                        session_options.rtcp_cname, ssrc_generator,
-                       current_streams, answer)) {
+                       current_streams, answer, field_trials)) {
     return false;  // Something went seriously wrong.
   }
   return true;
@@ -2325,11 +2326,11 @@ bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
   std::vector<std::string> crypto_suites;
   GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options,
                                         &crypto_suites);
-  if (!CreateMediaContentOffer(media_description_options, session_options,
-                               filtered_codecs, sdes_policy,
-                               GetCryptos(current_content), crypto_suites,
-                               audio_rtp_extensions, ssrc_generator_,
-                               current_streams, audio.get())) {
+  if (!CreateMediaContentOffer(
+          media_description_options, session_options, filtered_codecs,
+          sdes_policy, GetCryptos(current_content), crypto_suites,
+          audio_rtp_extensions, ssrc_generator_, current_streams, audio.get(),
+          transport_desc_factory_->trials())) {
     return false;
   }
 
@@ -2434,11 +2435,11 @@ bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
   std::vector<std::string> crypto_suites;
   GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options,
                                         &crypto_suites);
-  if (!CreateMediaContentOffer(media_description_options, session_options,
-                               filtered_codecs, sdes_policy,
-                               GetCryptos(current_content), crypto_suites,
-                               video_rtp_extensions, ssrc_generator_,
-                               current_streams, video.get())) {
+  if (!CreateMediaContentOffer(
+          media_description_options, session_options, filtered_codecs,
+          sdes_policy, GetCryptos(current_content), crypto_suites,
+          video_rtp_extensions, ssrc_generator_, current_streams, video.get(),
+          transport_desc_factory_->trials())) {
     return false;
   }
 
@@ -2622,8 +2623,8 @@ bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
       audio_transport->secure() ? cricket::SEC_DISABLED : secure();
   if (!SetCodecsInAnswer(offer_audio_description, filtered_codecs,
                          media_description_options, session_options,
-                         ssrc_generator_, current_streams,
-                         audio_answer.get())) {
+                         ssrc_generator_, current_streams, audio_answer.get(),
+                         transport_desc_factory_->trials())) {
     return false;
   }
   if (!CreateMediaContentAnswer(
@@ -2748,8 +2749,8 @@ bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
       video_transport->secure() ? cricket::SEC_DISABLED : secure();
   if (!SetCodecsInAnswer(offer_video_description, filtered_codecs,
                          media_description_options, session_options,
-                         ssrc_generator_, current_streams,
-                         video_answer.get())) {
+                         ssrc_generator_, current_streams, video_answer.get(),
+                         transport_desc_factory_->trials())) {
     return false;
   }
   if (!CreateMediaContentAnswer(
diff --git a/pc/media_session.h b/pc/media_session.h
index aa24f015de..d6c7474fe6 100644
--- a/pc/media_session.h
+++ b/pc/media_session.h
@@ -22,6 +22,7 @@
 #include "api/media_types.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
+#include "api/webrtc_key_value_config.h"
 #include "media/base/media_constants.h"
 #include "media/base/rid_description.h"
 #include "media/base/stream_params.h"
@@ -35,6 +36,13 @@
 #include "pc/simulcast_description.h"
 #include "rtc_base/unique_id_generator.h"
 
+namespace webrtc {
+
+// Forward declaration due to circular dependecy.
+class ConnectionContext;
+
+}  // namespace webrtc
+
 namespace cricket {
 
 class ChannelManager;
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index 6bc842d222..f06ddc7586 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -43,9 +43,9 @@
 #include "rtc_base/string_encode.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/unique_id_generator.h"
-#include "test/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 #define ASSERT_CRYPTO(cd, s, cs)      \
   ASSERT_EQ(s, cd->cryptos().size()); \
@@ -431,7 +431,10 @@ void PreferGcmCryptoParameters(CryptoParamsVec* cryptos) {
 class MediaSessionDescriptionFactoryTest : public ::testing::Test {
  public:
   MediaSessionDescriptionFactoryTest()
-      : f1_(&tdf1_, &ssrc_generator1), f2_(&tdf2_, &ssrc_generator2) {
+      : tdf1_(field_trials),
+        tdf2_(field_trials),
+        f1_(&tdf1_, &ssrc_generator1),
+        f2_(&tdf2_, &ssrc_generator2) {
     f1_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs1),
                          MAKE_VECTOR(kAudioCodecs1));
     f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1),
@@ -791,12 +794,13 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
   }
 
  protected:
+  webrtc::test::ScopedKeyValueConfig field_trials;
   UniqueRandomIdGenerator ssrc_generator1;
   UniqueRandomIdGenerator ssrc_generator2;
-  MediaSessionDescriptionFactory f1_;
-  MediaSessionDescriptionFactory f2_;
   TransportDescriptionFactory tdf1_;
   TransportDescriptionFactory tdf2_;
+  MediaSessionDescriptionFactory f1_;
+  MediaSessionDescriptionFactory f2_;
 };
 
 // Create a typical audio offer, and ensure it matches what we expect.
@@ -3260,8 +3264,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateMultipleRtxSsrcs) {
 // Test that, when the FlexFEC codec is added, a FlexFEC ssrc is created
 // together with a FEC-FR grouping. Guarded by WebRTC-FlexFEC-03 trial.
 TEST_F(MediaSessionDescriptionFactoryTest, GenerateFlexfecSsrc) {
-  webrtc::test::ScopedFieldTrials override_field_trials(
-      "WebRTC-FlexFEC-03/Enabled/");
+  webrtc::test::ScopedKeyValueConfig override_field_trials(
+      field_trials, "WebRTC-FlexFEC-03/Enabled/");
   MediaSessionOptions opts;
   AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
                              RtpTransceiverDirection::kSendRecv, kActive,
@@ -3303,8 +3307,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, GenerateFlexfecSsrc) {
 // TODO(brandtr): Remove this test when we support simulcast, either through
 // multiple FlexfecSenders, or through multistream protection.
 TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateNoFlexfecSsrcs) {
-  webrtc::test::ScopedFieldTrials override_field_trials(
-      "WebRTC-FlexFEC-03/Enabled/");
+  webrtc::test::ScopedKeyValueConfig override_field_trials(
+      field_trials, "WebRTC-FlexFEC-03/Enabled/");
   MediaSessionOptions opts;
   AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
                              RtpTransceiverDirection::kSendRecv, kActive,
@@ -4330,7 +4334,10 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 class MediaProtocolTest : public ::testing::TestWithParam<const char*> {
  public:
   MediaProtocolTest()
-      : f1_(&tdf1_, &ssrc_generator1), f2_(&tdf2_, &ssrc_generator2) {
+      : tdf1_(field_trials_),
+        tdf2_(field_trials_),
+        f1_(&tdf1_, &ssrc_generator1),
+        f2_(&tdf2_, &ssrc_generator2) {
     f1_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs1),
                          MAKE_VECTOR(kAudioCodecs1));
     f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1),
@@ -4350,10 +4357,11 @@ class MediaProtocolTest : public ::testing::TestWithParam<const char*> {
   }
 
  protected:
-  MediaSessionDescriptionFactory f1_;
-  MediaSessionDescriptionFactory f2_;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
   TransportDescriptionFactory tdf1_;
   TransportDescriptionFactory tdf2_;
+  MediaSessionDescriptionFactory f1_;
+  MediaSessionDescriptionFactory f2_;
   UniqueRandomIdGenerator ssrc_generator1;
   UniqueRandomIdGenerator ssrc_generator2;
 };
@@ -4389,7 +4397,8 @@ INSTANTIATE_TEST_SUITE_P(MediaProtocolDtlsPatternTest,
                          ::testing::ValuesIn(kMediaProtocolsDtls));
 
 TEST_F(MediaSessionDescriptionFactoryTest, TestSetAudioCodecs) {
-  TransportDescriptionFactory tdf;
+  webrtc::test::ScopedKeyValueConfig field_trials;
+  TransportDescriptionFactory tdf(field_trials);
   UniqueRandomIdGenerator ssrc_generator;
   MediaSessionDescriptionFactory sf(&tdf, &ssrc_generator);
   std::vector<AudioCodec> send_codecs = MAKE_VECTOR(kAudioCodecs1);
@@ -4459,7 +4468,8 @@ bool CodecsMatch(const std::vector<Codec>& codecs1,
 }
 
 void TestAudioCodecsOffer(RtpTransceiverDirection direction) {
-  TransportDescriptionFactory tdf;
+  webrtc::test::ScopedKeyValueConfig field_trials;
+  TransportDescriptionFactory tdf(field_trials);
   UniqueRandomIdGenerator ssrc_generator;
   MediaSessionDescriptionFactory sf(&tdf, &ssrc_generator);
   const std::vector<AudioCodec> send_codecs = MAKE_VECTOR(kAudioCodecs1);
@@ -4556,11 +4566,13 @@ std::vector<T> VectorFromIndices(const T* array, const int (&indices)[IDXS]) {
 void TestAudioCodecsAnswer(RtpTransceiverDirection offer_direction,
                            RtpTransceiverDirection answer_direction,
                            bool add_legacy_stream) {
-  TransportDescriptionFactory offer_tdf;
-  TransportDescriptionFactory answer_tdf;
+  webrtc::test::ScopedKeyValueConfig field_trials;
+  TransportDescriptionFactory offer_tdf(field_trials);
+  TransportDescriptionFactory answer_tdf(field_trials);
   UniqueRandomIdGenerator ssrc_generator1, ssrc_generator2;
   MediaSessionDescriptionFactory offer_factory(&offer_tdf, &ssrc_generator1);
   MediaSessionDescriptionFactory answer_factory(&answer_tdf, &ssrc_generator2);
+
   offer_factory.set_audio_codecs(
       VectorFromIndices(kOfferAnswerCodecs, kOfferSendCodecs),
       VectorFromIndices(kOfferAnswerCodecs, kOfferRecvCodecs));
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index a315ac510f..ec6bef32f7 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -711,6 +711,8 @@ JsepTransportController* PeerConnection::InitializeTransportController_n(
         }
       };
 
+  config.field_trials = &context_->trials();
+
   transport_controller_.reset(
       new JsepTransportController(network_thread(), port_allocator_.get(),
                                   async_dns_resolver_factory_.get(), config));
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index cd4af9e420..a6e4510b47 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -451,6 +451,8 @@ class PeerConnection : public PeerConnectionInternal,
   }
   void RequestUsagePatternReportForTesting();
 
+  const WebRtcKeyValueConfig& trials() override { return context_->trials(); }
+
  protected:
   // Available for rtc::scoped_refptr creation
   PeerConnection(rtc::scoped_refptr<ConnectionContext> context,
diff --git a/pc/peer_connection_internal.h b/pc/peer_connection_internal.h
index 16caade6c9..eaff20a0d3 100644
--- a/pc/peer_connection_internal.h
+++ b/pc/peer_connection_internal.h
@@ -178,6 +178,8 @@ class PeerConnectionInternal : public PeerConnectionInterface,
   virtual void NoteDataAddedEvent() {}
   // Handler for the "channel closed" signal
   virtual void OnSctpDataChannelClosed(DataChannelInterface* channel) {}
+
+  virtual const WebRtcKeyValueConfig& trials() = 0;
 };
 
 }  // namespace webrtc
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index c0b09e39c3..a7b9c9ab68 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -70,6 +70,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/run_loop.h"
+#include "test/scoped_key_value_config.h"
 
 using ::testing::_;
 using ::testing::ContainerEq;
@@ -187,7 +188,7 @@ class RtpSenderReceiverTest
 
   std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
     auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
-        /*rtcp_mux_required=*/true);
+        /*rtcp_mux_required=*/true, field_trials_);
     dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
                                            /*rtcp_dtls_transport=*/nullptr);
     return dtls_srtp_transport;
@@ -544,6 +545,7 @@ class RtpSenderReceiverTest
   rtc::scoped_refptr<VideoTrackInterface> video_track_;
   rtc::scoped_refptr<AudioTrackInterface> audio_track_;
   bool audio_sender_destroyed_signal_fired_ = false;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
 };
 
 // Test that `voice_channel_` is updated when an audio track is associated
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 69588350ab..a9501caa5e 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -63,7 +63,6 @@
 #include "rtc_base/string_encode.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/trace_event.h"
-#include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 
 using cricket::ContentInfo;
@@ -1186,13 +1185,14 @@ std::unique_ptr<SdpOfferAnswerHandler> SdpOfferAnswerHandler::Create(
     PeerConnectionDependencies& dependencies,
     ConnectionContext* context) {
   auto handler = absl::WrapUnique(new SdpOfferAnswerHandler(pc, context));
-  handler->Initialize(configuration, dependencies);
+  handler->Initialize(configuration, dependencies, context);
   return handler;
 }
 
 void SdpOfferAnswerHandler::Initialize(
     const PeerConnectionInterface::RTCConfiguration& configuration,
-    PeerConnectionDependencies& dependencies) {
+    PeerConnectionDependencies& dependencies,
+    ConnectionContext* context) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   video_options_.screencast_min_bitrate_kbps =
       configuration.screencast_min_bitrate;
@@ -1222,9 +1222,8 @@ void SdpOfferAnswerHandler::Initialize(
 
   webrtc_session_desc_factory_ =
       std::make_unique<WebRtcSessionDescriptionFactory>(
-          signaling_thread(), channel_manager(), this, pc_->session_id(),
-          pc_->dtls_enabled(), std::move(dependencies.cert_generator),
-          certificate,
+          context, this, pc_->session_id(), pc_->dtls_enabled(),
+          std::move(dependencies.cert_generator), certificate,
           [this](const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
             RTC_DCHECK_RUN_ON(signaling_thread());
             transport_controller_s()->SetLocalCertificate(certificate);
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index 67ead47242..a32ece930b 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -220,7 +220,8 @@ class SdpOfferAnswerHandler : public SdpStateProvider,
   // once. Modifies dependencies.
   void Initialize(
       const PeerConnectionInterface::RTCConfiguration& configuration,
-      PeerConnectionDependencies& dependencies);
+      PeerConnectionDependencies& dependencies,
+      ConnectionContext* context);
 
   rtc::Thread* signaling_thread() const;
   rtc::Thread* network_thread() const;
diff --git a/pc/srtp_session.cc b/pc/srtp_session.cc
index a81f2415a5..b5b244265c 100644
--- a/pc/srtp_session.cc
+++ b/pc/srtp_session.cc
@@ -18,6 +18,7 @@
 #include "absl/base/attributes.h"
 #include "absl/base/const_init.h"
 #include "api/array_view.h"
+#include "api/webrtc_key_value_config.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "pc/external_hmac.h"
 #include "rtc_base/byte_order.h"
@@ -26,7 +27,6 @@
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/string_encode.h"
 #include "rtc_base/time_utils.h"
-#include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 #include "third_party/libsrtp/include/srtp.h"
 #include "third_party/libsrtp/include/srtp_priv.h"
@@ -40,8 +40,10 @@ using ::webrtc::ParseRtpSequenceNumber;
 // in srtp.h.
 constexpr int kSrtpErrorCodeBoundary = 28;
 
-SrtpSession::SrtpSession() {
-  dump_plain_rtp_ = webrtc::field_trial::IsEnabled("WebRTC-Debugging-RtpDump");
+SrtpSession::SrtpSession() {}
+
+SrtpSession::SrtpSession(const webrtc::WebRtcKeyValueConfig& field_trials) {
+  dump_plain_rtp_ = field_trials.IsEnabled("WebRTC-Debugging-RtpDump");
 }
 
 SrtpSession::~SrtpSession() {
diff --git a/pc/srtp_session.h b/pc/srtp_session.h
index d88eaae319..c3979aae7a 100644
--- a/pc/srtp_session.h
+++ b/pc/srtp_session.h
@@ -18,6 +18,7 @@
 
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/synchronization/mutex.h"
 
 // Forward declaration to avoid pulling in libsrtp headers here
@@ -35,6 +36,7 @@ void ProhibitLibsrtpInitialization();
 class SrtpSession {
  public:
   SrtpSession();
+  explicit SrtpSession(const webrtc::WebRtcKeyValueConfig& field_trials);
   ~SrtpSession();
 
   SrtpSession(const SrtpSession&) = delete;
diff --git a/pc/srtp_session_unittest.cc b/pc/srtp_session_unittest.cc
index dc08c2e908..16a840a307 100644
--- a/pc/srtp_session_unittest.cc
+++ b/pc/srtp_session_unittest.cc
@@ -21,6 +21,7 @@
 #include "system_wrappers/include/metrics.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 #include "third_party/libsrtp/include/srtp.h"
 
 using ::testing::ElementsAre;
@@ -32,7 +33,9 @@ std::vector<int> kEncryptedHeaderExtensionIds;
 
 class SrtpSessionTest : public ::testing::Test {
  public:
-  SrtpSessionTest() { webrtc::metrics::Reset(); }
+  SrtpSessionTest() : s1_(field_trials_), s2_(field_trials_) {
+    webrtc::metrics::Reset();
+  }
 
  protected:
   virtual void SetUp() {
@@ -69,6 +72,7 @@ class SrtpSessionTest : public ::testing::Test {
     EXPECT_EQ(expected_len, out_len);
     EXPECT_EQ(0, memcmp(rtcp_packet_, kRtcpReport, out_len));
   }
+  webrtc::test::ScopedKeyValueConfig field_trials_;
   cricket::SrtpSession s1_;
   cricket::SrtpSession s2_;
   char rtp_packet_[sizeof(kPcmuFrame) + 10];
diff --git a/pc/srtp_transport.cc b/pc/srtp_transport.cc
index 230c1a347b..b59fc1d1d2 100644
--- a/pc/srtp_transport.cc
+++ b/pc/srtp_transport.cc
@@ -34,8 +34,9 @@
 
 namespace webrtc {
 
-SrtpTransport::SrtpTransport(bool rtcp_mux_enabled)
-    : RtpTransport(rtcp_mux_enabled) {}
+SrtpTransport::SrtpTransport(bool rtcp_mux_enabled,
+                             const WebRtcKeyValueConfig& field_trials)
+    : RtpTransport(rtcp_mux_enabled), field_trials_(field_trials) {}
 
 RTCError SrtpTransport::SetSrtpSendKey(const cricket::CryptoParams& params) {
   if (send_params_) {
@@ -324,13 +325,13 @@ bool SrtpTransport::SetRtcpParams(int send_cs,
     return false;
   }
 
-  send_rtcp_session_.reset(new cricket::SrtpSession());
+  send_rtcp_session_.reset(new cricket::SrtpSession(field_trials_));
   if (!send_rtcp_session_->SetSend(send_cs, send_key, send_key_len,
                                    send_extension_ids)) {
     return false;
   }
 
-  recv_rtcp_session_.reset(new cricket::SrtpSession());
+  recv_rtcp_session_.reset(new cricket::SrtpSession(field_trials_));
   if (!recv_rtcp_session_->SetRecv(recv_cs, recv_key, recv_key_len,
                                    recv_extension_ids)) {
     return false;
@@ -361,8 +362,8 @@ void SrtpTransport::ResetParams() {
 }
 
 void SrtpTransport::CreateSrtpSessions() {
-  send_session_.reset(new cricket::SrtpSession());
-  recv_session_.reset(new cricket::SrtpSession());
+  send_session_.reset(new cricket::SrtpSession(field_trials_));
+  recv_session_.reset(new cricket::SrtpSession(field_trials_));
   if (external_auth_enabled_) {
     send_session_->EnableExternalAuth();
   }
diff --git a/pc/srtp_transport.h b/pc/srtp_transport.h
index 4bc028d68e..2fec29eb01 100644
--- a/pc/srtp_transport.h
+++ b/pc/srtp_transport.h
@@ -21,6 +21,7 @@
 #include "absl/types/optional.h"
 #include "api/crypto_params.h"
 #include "api/rtc_error.h"
+#include "api/webrtc_key_value_config.h"
 #include "p2p/base/packet_transport_internal.h"
 #include "pc/rtp_transport.h"
 #include "pc/srtp_session.h"
@@ -36,7 +37,8 @@ namespace webrtc {
 // parameters for the SrtpSession underneath.
 class SrtpTransport : public RtpTransport {
  public:
-  explicit SrtpTransport(bool rtcp_mux_enabled);
+  SrtpTransport(bool rtcp_mux_enabled,
+                const WebRtcKeyValueConfig& field_trials);
 
   virtual ~SrtpTransport() = default;
 
@@ -167,6 +169,8 @@ class SrtpTransport : public RtpTransport {
   int rtp_abs_sendtime_extn_id_ = -1;
 
   int decryption_failure_count_ = 0;
+
+  const WebRtcKeyValueConfig& field_trials_;
 };
 
 }  // namespace webrtc
diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc
index 980ebca08a..4e643935a9 100644
--- a/pc/srtp_transport_unittest.cc
+++ b/pc/srtp_transport_unittest.cc
@@ -27,6 +27,7 @@
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/third_party/sigslot/sigslot.h"
 #include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
 
 using rtc::kSrtpAeadAes128Gcm;
 using rtc::kTestKey1;
@@ -57,8 +58,10 @@ class SrtpTransportTest : public ::testing::Test, public sigslot::has_slots<> {
     rtp_packet_transport1_->SetDestination(rtp_packet_transport2_.get(),
                                            asymmetric);
 
-    srtp_transport1_ = std::make_unique<SrtpTransport>(rtcp_mux_enabled);
-    srtp_transport2_ = std::make_unique<SrtpTransport>(rtcp_mux_enabled);
+    srtp_transport1_ =
+        std::make_unique<SrtpTransport>(rtcp_mux_enabled, field_trials_);
+    srtp_transport2_ =
+        std::make_unique<SrtpTransport>(rtcp_mux_enabled, field_trials_);
 
     srtp_transport1_->SetRtpPacketTransport(rtp_packet_transport1_.get());
     srtp_transport2_->SetRtpPacketTransport(rtp_packet_transport2_.get());
@@ -333,6 +336,7 @@ class SrtpTransportTest : public ::testing::Test, public sigslot::has_slots<> {
   TransportObserver rtp_sink2_;
 
   int sequence_number_ = 0;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
 };
 
 class SrtpTransportTestWithExternalAuth
diff --git a/pc/stats_collector.cc b/pc/stats_collector.cc
index 6b1cda3ad5..927e993203 100644
--- a/pc/stats_collector.cc
+++ b/pc/stats_collector.cc
@@ -31,6 +31,7 @@
 #include "api/sequence_checker.h"
 #include "api/video/video_content_type.h"
 #include "api/video/video_timing.h"
+#include "api/webrtc_key_value_config.h"
 #include "call/call.h"
 #include "media/base/media_channel.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
@@ -55,7 +56,6 @@
 #include "rtc_base/thread.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/trace_event.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 namespace {
@@ -543,7 +543,7 @@ StatsCollector::StatsCollector(PeerConnectionInternal* pc)
     : pc_(pc),
       stats_gathering_started_(0),
       use_standard_bytes_stats_(
-          webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
+          pc->trials().IsEnabled(kUseStandardBytesStats)) {
   RTC_DCHECK(pc_);
 }
 
diff --git a/pc/stats_collector.h b/pc/stats_collector.h
index 751a2de09c..794437e9eb 100644
--- a/pc/stats_collector.h
+++ b/pc/stats_collector.h
@@ -30,6 +30,7 @@
 #include "api/peer_connection_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/stats_types.h"
+#include "api/webrtc_key_value_config.h"
 #include "p2p/base/connection_info.h"
 #include "p2p/base/port.h"
 #include "pc/peer_connection_internal.h"
diff --git a/pc/test/fake_peer_connection_base.h b/pc/test/fake_peer_connection_base.h
index 7d64ab8180..2b4dd73913 100644
--- a/pc/test/fake_peer_connection_base.h
+++ b/pc/test/fake_peer_connection_base.h
@@ -18,7 +18,9 @@
 #include <vector>
 
 #include "api/sctp_transport_interface.h"
+#include "api/webrtc_key_value_config.h"
 #include "pc/peer_connection_internal.h"
+#include "test/scoped_key_value_config.h"
 
 namespace webrtc {
 
@@ -358,7 +360,10 @@ class FakePeerConnectionBase : public PeerConnectionInternal {
   void SetSctpDataMid(const std::string& mid) override {}
   void ResetSctpDataMid() override {}
 
+  const WebRtcKeyValueConfig& trials() override { return field_trials_; }
+
  protected:
+  webrtc::test::ScopedKeyValueConfig field_trials_;
   sigslot::signal1<SctpDataChannel*> SignalSctpDataChannelCreated_;
 };
 
diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index 99ad8af4b5..00417d96a5 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -120,8 +120,8 @@
 #include "rtc_base/time_utils.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "system_wrappers/include/metrics.h"
-#include "test/field_trial.h"
 #include "test/gmock.h"
+#include "test/scoped_key_value_config.h"
 
 namespace webrtc {
 
@@ -1366,9 +1366,9 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
         fss_(new rtc::FirewallSocketServer(ss_.get())),
         network_thread_(new rtc::Thread(fss_.get())),
         worker_thread_(rtc::Thread::Create()),
-        field_trials_(field_trials.has_value()
-                          ? new test::ScopedFieldTrials(*field_trials)
-                          : nullptr) {
+        // TODO(bugs.webrtc.org/10335): Pass optional ScopedKeyValueConfig.
+        field_trials_(new test::ScopedKeyValueConfig(
+            field_trials.has_value() ? *field_trials : "")) {
     network_thread_->SetName("PCNetworkThread", this);
     worker_thread_->SetName("PCWorkerThread", this);
     RTC_CHECK(network_thread_->Start());
@@ -1855,6 +1855,8 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
                               expected_cipher_suite);
   }
 
+  const WebRtcKeyValueConfig& trials() const { return *field_trials_.get(); }
+
  protected:
   SdpSemantics sdp_semantics_;
 
@@ -1874,7 +1876,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
   std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_;
   std::unique_ptr<PeerConnectionIntegrationWrapper> caller_;
   std::unique_ptr<PeerConnectionIntegrationWrapper> callee_;
-  std::unique_ptr<test::ScopedFieldTrials> field_trials_;
+  std::unique_ptr<WebRtcKeyValueConfig> field_trials_;
 };
 
 }  // namespace webrtc
diff --git a/pc/webrtc_session_description_factory.cc b/pc/webrtc_session_description_factory.cc
index 82ba849544..09f2ee2f00 100644
--- a/pc/webrtc_session_description_factory.cc
+++ b/pc/webrtc_session_description_factory.cc
@@ -24,6 +24,7 @@
 #include "api/jsep_session_description.h"
 #include "api/rtc_error.h"
 #include "api/sequence_checker.h"
+#include "pc/connection_context.h"
 #include "pc/sdp_state_provider.h"
 #include "pc/session_description.h"
 #include "rtc_base/checks.h"
@@ -127,8 +128,7 @@ void WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
 }
 
 WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
-    rtc::Thread* signaling_thread,
-    cricket::ChannelManager* channel_manager,
+    ConnectionContext* context,
     const SdpStateProvider* sdp_info,
     const std::string& session_id,
     bool dtls_enabled,
@@ -136,8 +136,10 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
     const rtc::scoped_refptr<rtc::RTCCertificate>& certificate,
     std::function<void(const rtc::scoped_refptr<rtc::RTCCertificate>&)>
         on_certificate_ready)
-    : signaling_thread_(signaling_thread),
-      session_desc_factory_(channel_manager, &transport_desc_factory_),
+    : signaling_thread_(context->signaling_thread()),
+      transport_desc_factory_(context->trials()),
+      session_desc_factory_(context->channel_manager(),
+                            &transport_desc_factory_),
       // RFC 4566 suggested a Network Time Protocol (NTP) format timestamp
       // as the session id and session version. To simplify, it should be fine
       // to just use a random number as session id and start version from
diff --git a/pc/webrtc_session_description_factory.h b/pc/webrtc_session_description_factory.h
index 6a6e8efa51..79171f797b 100644
--- a/pc/webrtc_session_description_factory.h
+++ b/pc/webrtc_session_description_factory.h
@@ -79,8 +79,7 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
   // asynchronously. If a certificate is given, will use that for identifying
   // over DTLS. If neither is specified, DTLS is disabled.
   WebRtcSessionDescriptionFactory(
-      rtc::Thread* signaling_thread,
-      cricket::ChannelManager* channel_manager,
+      ConnectionContext* context,
       const SdpStateProvider* sdp_info,
       const std::string& session_id,
       bool dtls_enabled,
diff --git a/test/BUILD.gn b/test/BUILD.gn
index f49dbd485a..82d909d1ae 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -214,7 +214,6 @@ rtc_library("rtp_test_utils") {
 
 rtc_library("field_trial") {
   visibility = [ "*" ]
-  testonly = true
   sources = [
     "field_trial.cc",
     "field_trial.h",
@@ -233,9 +232,23 @@ rtc_library("explicit_key_value_config") {
   ]
 
   deps = [
-    "../api/transport:webrtc_key_value_config",
+    ":field_trial",
+    "../api:webrtc_key_value_config",
+    "../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+}
+
+rtc_library("scoped_key_value_config") {
+  sources = [
+    "scoped_key_value_config.cc",
+    "scoped_key_value_config.h",
+  ]
+
+  deps = [
+    ":field_trial",
+    "../api:webrtc_key_value_config",
     "../rtc_base:checks",
-    "../system_wrappers:field_trial",
   ]
   absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
 }
diff --git a/test/explicit_key_value_config.cc b/test/explicit_key_value_config.cc
index 69f725a9e2..6a561fa209 100644
--- a/test/explicit_key_value_config.cc
+++ b/test/explicit_key_value_config.cc
@@ -10,9 +10,8 @@
 
 #include "test/explicit_key_value_config.h"
 
-#include "api/transport/webrtc_key_value_config.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/checks.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 namespace test {
diff --git a/test/explicit_key_value_config.h b/test/explicit_key_value_config.h
index 9a3bc84f60..0a728ab536 100644
--- a/test/explicit_key_value_config.h
+++ b/test/explicit_key_value_config.h
@@ -15,7 +15,7 @@
 #include <string>
 
 #include "absl/strings/string_view.h"
-#include "api/transport/webrtc_key_value_config.h"
+#include "api/webrtc_key_value_config.h"
 
 namespace webrtc {
 namespace test {
diff --git a/test/peer_scenario/BUILD.gn b/test/peer_scenario/BUILD.gn
index a08b4acbeb..d3ac0da606 100644
--- a/test/peer_scenario/BUILD.gn
+++ b/test/peer_scenario/BUILD.gn
@@ -49,6 +49,8 @@ if (rtc_include_tests) {
       "../../rtc_base",
       "../../rtc_base:null_socket_server",
       "../../rtc_base:stringutils",
+      "../../test:explicit_key_value_config",
+      "../../test:scoped_key_value_config",
       "../logging:log_writer",
       "../network:emulated_network",
       "../scenario",
diff --git a/test/peer_scenario/scenario_connection.cc b/test/peer_scenario/scenario_connection.cc
index dac21d5586..07da453b8a 100644
--- a/test/peer_scenario/scenario_connection.cc
+++ b/test/peer_scenario/scenario_connection.cc
@@ -132,6 +132,7 @@ JsepTransportController::Config ScenarioIceConnectionImpl::CreateJsepConfig() {
     RTC_DCHECK_RUN_ON(network_thread_);
     observer_->OnPacketReceived(packet);
   };
+  config.field_trials = &field_trials;
   return config;
 }
 
diff --git a/test/peer_scenario/scenario_connection.h b/test/peer_scenario/scenario_connection.h
index f43b3d39d4..e8cef527c5 100644
--- a/test/peer_scenario/scenario_connection.h
+++ b/test/peer_scenario/scenario_connection.h
@@ -19,6 +19,7 @@
 #include "api/jsep.h"
 #include "p2p/base/transport_description.h"
 #include "test/network/network_emulation_manager.h"
+#include "test/scoped_key_value_config.h"
 
 namespace webrtc {
 
@@ -56,6 +57,8 @@ class ScenarioIceConnection {
   virtual EmulatedEndpoint* endpoint() = 0;
   virtual const cricket::TransportDescription& transport_description()
       const = 0;
+
+  webrtc::test::ScopedKeyValueConfig field_trials;
 };
 
 }  // namespace webrtc
diff --git a/test/scoped_key_value_config.cc b/test/scoped_key_value_config.cc
new file mode 100644
index 0000000000..fe3f42635d
--- /dev/null
+++ b/test/scoped_key_value_config.cc
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2020 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 "test/scoped_key_value_config.h"
+
+#include "api/webrtc_key_value_config.h"
+#include "rtc_base/checks.h"
+#include "test/field_trial.h"
+
+namespace {
+
+// This part is copied from system_wrappers/field_trial.cc.
+void InsertIntoMap(std::map<std::string, std::string>& key_value_map,
+                   const std::string& s) {
+  std::string::size_type field_start = 0;
+  while (field_start < s.size()) {
+    std::string::size_type separator_pos = s.find('/', field_start);
+    RTC_CHECK_NE(separator_pos, std::string::npos)
+        << "Missing separator '/' after field trial key.";
+    RTC_CHECK_GT(separator_pos, field_start)
+        << "Field trial key cannot be empty.";
+    std::string key = s.substr(field_start, separator_pos - field_start);
+    field_start = separator_pos + 1;
+
+    RTC_CHECK_LT(field_start, s.size())
+        << "Missing value after field trial key. String ended.";
+    separator_pos = s.find('/', field_start);
+    RTC_CHECK_NE(separator_pos, std::string::npos)
+        << "Missing terminating '/' in field trial string.";
+    RTC_CHECK_GT(separator_pos, field_start)
+        << "Field trial value cannot be empty.";
+    std::string value = s.substr(field_start, separator_pos - field_start);
+    field_start = separator_pos + 1;
+
+    key_value_map[key] = value;
+  }
+  // This check is technically redundant due to earlier checks.
+  // We nevertheless keep the check to make it clear that the entire
+  // string has been processed, and without indexing past the end.
+  RTC_CHECK_EQ(field_start, s.size());
+}
+
+}  // namespace
+
+namespace webrtc {
+namespace test {
+
+ScopedKeyValueConfig::ScopedKeyValueConfig()
+    : ScopedKeyValueConfig(nullptr, "") {}
+
+ScopedKeyValueConfig::ScopedKeyValueConfig(const std::string& s)
+    : ScopedKeyValueConfig(nullptr, s) {}
+
+ScopedKeyValueConfig::ScopedKeyValueConfig(ScopedKeyValueConfig& parent,
+                                           const std::string& s)
+    : ScopedKeyValueConfig(&parent, s) {}
+
+ScopedKeyValueConfig::ScopedKeyValueConfig(ScopedKeyValueConfig* parent,
+                                           const std::string& s)
+    : parent_(parent), leaf_(nullptr) {
+  InsertIntoMap(key_value_map_, s);
+  // Also store field trials in global string (until we get rid of it).
+  scoped_field_trials_ = std::make_unique<ScopedFieldTrials>(s);
+
+  if (parent == nullptr) {
+    // We are root, set leaf_.
+    leaf_ = this;
+  } else {
+    // Link root to new leaf.
+    GetRoot(parent)->leaf_ = this;
+    RTC_DCHECK(leaf_ == nullptr);
+  }
+}
+
+ScopedKeyValueConfig::~ScopedKeyValueConfig() {
+  if (parent_) {
+    GetRoot(parent_)->leaf_ = parent_;
+  }
+}
+
+ScopedKeyValueConfig* ScopedKeyValueConfig::GetRoot(ScopedKeyValueConfig* n) {
+  while (n->parent_ != nullptr) {
+    n = n->parent_;
+  }
+  return n;
+}
+
+std::string ScopedKeyValueConfig::Lookup(absl::string_view key) const {
+  if (parent_ == nullptr) {
+    return leaf_->LookupRecurse(key);
+  } else {
+    return LookupRecurse(key);
+  }
+}
+
+std::string ScopedKeyValueConfig::LookupRecurse(absl::string_view key) const {
+  auto it = key_value_map_.find(std::string(key));
+  if (it != key_value_map_.end())
+    return it->second;
+
+  if (parent_) {
+    return parent_->LookupRecurse(key);
+  }
+
+  return "";
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/test/scoped_key_value_config.h b/test/scoped_key_value_config.h
new file mode 100644
index 0000000000..a00f61ac60
--- /dev/null
+++ b/test/scoped_key_value_config.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2020 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 TEST_SCOPED_KEY_VALUE_CONFIG_H_
+#define TEST_SCOPED_KEY_VALUE_CONFIG_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/webrtc_key_value_config.h"
+#include "test/field_trial.h"
+
+namespace webrtc {
+namespace test {
+
+class ScopedKeyValueConfig : public WebRtcKeyValueConfig {
+ public:
+  virtual ~ScopedKeyValueConfig();
+  ScopedKeyValueConfig();
+  explicit ScopedKeyValueConfig(const std::string& s);
+  ScopedKeyValueConfig(ScopedKeyValueConfig& parent, const std::string& s);
+
+  std::string Lookup(absl::string_view key) const override;
+
+ private:
+  ScopedKeyValueConfig(ScopedKeyValueConfig* parent, const std::string& s);
+  ScopedKeyValueConfig* GetRoot(ScopedKeyValueConfig* n);
+  std::string LookupRecurse(absl::string_view key) const;
+
+  ScopedKeyValueConfig* const parent_;
+
+  // The leaf in a list of stacked ScopedKeyValueConfig.
+  // Only set on root (e.g with parent_ == nullptr).
+  const ScopedKeyValueConfig* leaf_;
+
+  std::map<std::string, std::string> key_value_map_;
+  std::unique_ptr<ScopedFieldTrials> scoped_field_trials_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCOPED_KEY_VALUE_CONFIG_H_
diff --git a/video/encoder_bitrate_adjuster.cc b/video/encoder_bitrate_adjuster.cc
index 9f4d8551c4..8ed16a7565 100644
--- a/video/encoder_bitrate_adjuster.cc
+++ b/video/encoder_bitrate_adjuster.cc
@@ -17,7 +17,6 @@
 #include "rtc_base/experiments/rate_control_settings.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/time_utils.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 namespace {