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diff --git a/media/webrtc/signaling/gtest/MockCall.h b/media/webrtc/signaling/gtest/MockCall.h
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+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MOCK_CALL_H_
+#define MOCK_CALL_H_
+
+#include "gmock/gmock.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/Maybe.h"
+#include "mozilla/media/MediaUtils.h"
+#include "WebrtcCallWrapper.h"
+#include "PeerConnectionCtx.h"
+
+// libwebrtc
+#include "api/call/audio_sink.h"
+#include "call/call.h"
+
+namespace test {
+class MockCallWrapper;
+
+class MockAudioSendStream : public webrtc::AudioSendStream {
+ public:
+  explicit MockAudioSendStream(RefPtr<MockCallWrapper> aCallWrapper)
+      : mCallWrapper(std::move(aCallWrapper)) {}
+
+  const webrtc::AudioSendStream::Config& GetConfig() const override;
+
+  void Reconfigure(const Config& config) override;
+
+  void Start() override {}
+
+  void Stop() override {}
+
+  void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override {
+  }
+
+  bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
+                          int duration_ms) override {
+    return true;
+  }
+
+  void SetMuted(bool muted) override {}
+
+  Stats GetStats() const override { return mStats; }
+
+  Stats GetStats(bool has_remote_tracks) const override { return mStats; }
+
+  virtual ~MockAudioSendStream() {}
+
+  const RefPtr<MockCallWrapper> mCallWrapper;
+  webrtc::AudioSendStream::Stats mStats;
+};
+
+class MockAudioReceiveStream : public webrtc::AudioReceiveStreamInterface {
+ public:
+  explicit MockAudioReceiveStream(RefPtr<MockCallWrapper> aCallWrapper)
+      : mCallWrapper(std::move(aCallWrapper)) {}
+
+  void Start() override {}
+
+  void Stop() override {}
+
+  bool IsRunning() const override { return true; }
+
+  bool transport_cc() const override { return false; }
+
+  Stats GetStats(bool get_and_clear_legacy_stats) const override {
+    return mStats;
+  }
+
+  void SetSink(webrtc::AudioSinkInterface* sink) override {}
+
+  void SetGain(float gain) override {}
+
+  std::vector<webrtc::RtpSource> GetSources() const override {
+    return mRtpSources;
+  }
+
+  virtual void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override {
+    // Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure
+    // method.
+    MOZ_ASSERT(false);
+  }
+  virtual void SetDecoderMap(
+      std::map<int, webrtc::SdpAudioFormat> decoder_map) override;
+  virtual void SetTransportCc(bool use_transport_cc) override {
+    // Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure
+    // method.
+    MOZ_ASSERT(false);
+  }
+  virtual void SetNackHistory(int history_ms) override {
+    // Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure
+    // method.
+    MOZ_ASSERT(false);
+  }
+  virtual void SetNonSenderRttMeasurement(bool enabled) override {}
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override {}
+  void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
+  const std::vector<webrtc::RtpExtension>& GetRtpExtensions() const override;
+  webrtc::RtpHeaderExtensionMap GetRtpExtensionMap() const override;
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override { return false; }
+  int GetBaseMinimumPlayoutDelayMs() const override { return 0; }
+  uint32_t remote_ssrc() const override { return 0; }
+
+  virtual ~MockAudioReceiveStream() {}
+
+  const RefPtr<MockCallWrapper> mCallWrapper;
+  webrtc::AudioReceiveStreamInterface::Stats mStats;
+  std::vector<webrtc::RtpSource> mRtpSources;
+};
+
+class MockVideoSendStream : public webrtc::VideoSendStream {
+ public:
+  explicit MockVideoSendStream(RefPtr<MockCallWrapper> aCallWrapper)
+      : mCallWrapper(std::move(aCallWrapper)) {}
+
+  void Start() override {}
+
+  void Stop() override {}
+
+  bool started() override { return false; }
+
+  void SetSource(
+      rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
+      const webrtc::DegradationPreference& degradation_preference) override {}
+
+  void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) override;
+
+  Stats GetStats() override { return mStats; }
+
+  void UpdateActiveSimulcastLayers(
+      const std::vector<bool> active_layers) override {}
+
+  void AddAdaptationResource(
+      rtc::scoped_refptr<webrtc::Resource> resource) override {}
+
+  std::vector<rtc::scoped_refptr<webrtc::Resource>> GetAdaptationResources()
+      override {
+    return std::vector<rtc::scoped_refptr<webrtc::Resource>>();
+  }
+
+  virtual ~MockVideoSendStream() {}
+
+  const RefPtr<MockCallWrapper> mCallWrapper;
+  webrtc::VideoSendStream::Stats mStats;
+};
+
+class MockVideoReceiveStream : public webrtc::VideoReceiveStreamInterface {
+ public:
+  explicit MockVideoReceiveStream(RefPtr<MockCallWrapper> aCallWrapper)
+      : mCallWrapper(std::move(aCallWrapper)) {}
+
+  void Start() override {}
+
+  void Stop() override {}
+
+  bool transport_cc() const override { return false; }
+  void SetTransportCc(bool use_transport_cc) override {}
+
+  Stats GetStats() const override { return mStats; }
+
+  std::vector<webrtc::RtpSource> GetSources() const override {
+    return std::vector<webrtc::RtpSource>();
+  }
+
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override { return false; }
+
+  int GetBaseMinimumPlayoutDelayMs() const override { return 0; }
+
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override {}
+
+  void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override {}
+
+  RecordingState SetAndGetRecordingState(RecordingState state,
+                                         bool generate_key_frame) override {
+    return {};
+  }
+
+  void GenerateKeyFrame() override {}
+
+  void SetRtcpMode(webrtc::RtcpMode mode) override {}
+
+  void SetFlexFecProtection(webrtc::RtpPacketSinkInterface* flexfec_sink) override {}
+
+  void SetLossNotificationEnabled(bool enabled) override {}
+
+  void SetNackHistory(webrtc::TimeDelta history) override {}
+
+  void SetProtectionPayloadTypes(int red_payload_type,
+                                 int ulpfec_payload_type) override {}
+
+  void SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) override {}
+
+  virtual void SetAssociatedPayloadTypes(
+      std::map<int, int> associated_payload_types) override {}
+
+  void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override {
+  }
+  webrtc::RtpHeaderExtensionMap GetRtpExtensionMap() const override;
+
+  virtual ~MockVideoReceiveStream() {}
+
+  const RefPtr<MockCallWrapper> mCallWrapper;
+  webrtc::VideoReceiveStreamInterface::Stats mStats;
+};
+
+class MockCall : public webrtc::Call {
+ public:
+  explicit MockCall(RefPtr<MockCallWrapper> aCallWrapper)
+      : mCallWrapper(std::move(aCallWrapper)) {}
+
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override {
+    MOZ_RELEASE_ASSERT(!mAudioSendConfig);
+    mAudioSendConfig = mozilla::Some(config);
+    return new MockAudioSendStream(mCallWrapper);
+  }
+
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override {
+    mAudioSendConfig = mozilla::Nothing();
+    delete static_cast<MockAudioSendStream*>(send_stream);
+  }
+
+  webrtc::AudioReceiveStreamInterface* CreateAudioReceiveStream(
+      const webrtc::AudioReceiveStreamInterface::Config& config) override {
+    MOZ_RELEASE_ASSERT(!mAudioReceiveConfig);
+    mAudioReceiveConfig = mozilla::Some(config);
+    return new MockAudioReceiveStream(mCallWrapper);
+  }
+  void DestroyAudioReceiveStream(
+      webrtc::AudioReceiveStreamInterface* receive_stream) override {
+    mAudioReceiveConfig = mozilla::Nothing();
+    delete static_cast<MockAudioReceiveStream*>(receive_stream);
+  }
+
+  webrtc::VideoSendStream* CreateVideoSendStream(
+      webrtc::VideoSendStream::Config config,
+      webrtc::VideoEncoderConfig encoder_config) override {
+    MOZ_RELEASE_ASSERT(!mVideoSendConfig);
+    MOZ_RELEASE_ASSERT(!mVideoSendEncoderConfig);
+    mVideoSendConfig = mozilla::Some(std::move(config));
+    mVideoSendEncoderConfig = mozilla::Some(std::move(encoder_config));
+    return new MockVideoSendStream(mCallWrapper);
+  }
+
+  void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override {
+    mVideoSendConfig = mozilla::Nothing();
+    mVideoSendEncoderConfig = mozilla::Nothing();
+    delete static_cast<MockVideoSendStream*>(send_stream);
+  }
+
+  webrtc::VideoReceiveStreamInterface* CreateVideoReceiveStream(
+      webrtc::VideoReceiveStreamInterface::Config configuration) override {
+    MOZ_RELEASE_ASSERT(!mVideoReceiveConfig);
+    mVideoReceiveConfig = mozilla::Some(std::move(configuration));
+    return new MockVideoReceiveStream(mCallWrapper);
+  }
+
+  void DestroyVideoReceiveStream(
+      webrtc::VideoReceiveStreamInterface* receive_stream) override {
+    mVideoReceiveConfig = mozilla::Nothing();
+    delete static_cast<MockVideoReceiveStream*>(receive_stream);
+  }
+
+  webrtc::FlexfecReceiveStream* CreateFlexfecReceiveStream(
+      const webrtc::FlexfecReceiveStream::Config config) override {
+    return nullptr;
+  }
+
+  void DestroyFlexfecReceiveStream(
+      webrtc::FlexfecReceiveStream* receive_stream) override {}
+
+  void AddAdaptationResource(
+      rtc::scoped_refptr<webrtc::Resource> resource) override {}
+
+  webrtc::PacketReceiver* Receiver() override { return nullptr; }
+
+  webrtc::RtpTransportControllerSendInterface* GetTransportControllerSend()
+      override {
+    return nullptr;
+  }
+
+  Stats GetStats() const override { return mStats; }
+
+  void SignalChannelNetworkState(webrtc::MediaType media,
+                                 webrtc::NetworkState state) override {}
+
+  void OnAudioTransportOverheadChanged(
+      int transport_overhead_per_packet) override {}
+
+  void OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
+                          uint32_t local_ssrc) override {}
+  void OnLocalSsrcUpdated(webrtc::VideoReceiveStreamInterface& stream,
+                          uint32_t local_ssrc) override {}
+  void OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream,
+                          uint32_t local_ssrc) override {}
+
+  void OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream,
+                         absl::string_view sync_group) override {}
+
+  void OnSentPacket(const rtc::SentPacket& sent_packet) override {}
+
+  void SetClientBitratePreferences(
+      const webrtc::BitrateSettings& preferences) override {}
+
+  std::vector<webrtc::VideoStream> CreateEncoderStreams(int width, int height) {
+    return mVideoSendEncoderConfig->video_stream_factory->CreateEncoderStreams(
+        width, height, *mVideoSendEncoderConfig);
+  }
+
+  virtual const webrtc::WebRtcKeyValueConfig& trials() const override {
+    return mUnusedConfig;
+  }
+
+  virtual webrtc::TaskQueueBase* network_thread() const override {
+    return nullptr;
+  }
+
+  virtual webrtc::TaskQueueBase* worker_thread() const override {
+    return nullptr;
+  }
+
+  virtual ~MockCall(){};
+
+  const RefPtr<MockCallWrapper> mCallWrapper;
+  mozilla::Maybe<webrtc::AudioReceiveStreamInterface::Config>
+      mAudioReceiveConfig;
+  mozilla::Maybe<webrtc::AudioSendStream::Config> mAudioSendConfig;
+  mozilla::Maybe<webrtc::VideoReceiveStreamInterface::Config>
+      mVideoReceiveConfig;
+  mozilla::Maybe<webrtc::VideoSendStream::Config> mVideoSendConfig;
+  mozilla::Maybe<webrtc::VideoEncoderConfig> mVideoSendEncoderConfig;
+  webrtc::Call::Stats mStats;
+  webrtc::NoTrialsConfig mUnusedConfig;
+};
+
+class MockCallWrapper : public mozilla::WebrtcCallWrapper {
+ public:
+  MockCallWrapper(
+      RefPtr<mozilla::SharedWebrtcState> aSharedState,
+      mozilla::UniquePtr<webrtc::VideoBitrateAllocatorFactory>
+          aVideoBitrateAllocatorFactory,
+      mozilla::UniquePtr<webrtc::RtcEventLog> aEventLog,
+      mozilla::UniquePtr<webrtc::TaskQueueFactory> aTaskQueueFactory,
+      const mozilla::dom::RTCStatsTimestampMaker& aTimestampMaker,
+      mozilla::UniquePtr<mozilla::media::ShutdownBlockingTicket>
+          aShutdownTicket)
+      : mozilla::WebrtcCallWrapper(
+            std::move(aSharedState), std::move(aVideoBitrateAllocatorFactory),
+            std::move(aEventLog), std::move(aTaskQueueFactory), aTimestampMaker,
+            std::move(aShutdownTicket)) {}
+
+  static RefPtr<MockCallWrapper> Create() {
+    auto state = mozilla::MakeRefPtr<mozilla::SharedWebrtcState>(
+        mozilla::AbstractThread::GetCurrent(), webrtc::AudioState::Config(),
+        nullptr, nullptr);
+    auto wrapper = mozilla::MakeRefPtr<MockCallWrapper>(
+        state, nullptr, nullptr, nullptr,
+        mozilla::dom::RTCStatsTimestampMaker(), nullptr);
+    wrapper->SetCall(mozilla::WrapUnique(new MockCall(wrapper)));
+    return wrapper;
+  }
+
+  MockCall* GetMockCall() { return static_cast<MockCall*>(Call()); }
+};
+
+}  // namespace test
+#endif