diff --git a/media/base/media_engine.h b/media/base/media_engine.h
index dc8579a517..0d10248ed0 100644
--- a/media/base/media_engine.h
+++ b/media/base/media_engine.h
@@ -98,6 +98,28 @@ class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
   // TODO(solenberg): Remove once VoE API refactoring is done.
   virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
 
+  virtual std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) {
+    // TODO(hta): Make pure virtual when all downstream has updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual std::unique_ptr<VoiceMediaReceiveChannelInterface>
+  CreateReceiveChannel(webrtc::Call* call,
+                       const MediaConfig& config,
+                       const AudioOptions& options,
+                       const webrtc::CryptoOptions& crypto_options,
+                       webrtc::AudioCodecPairId codec_pair_id) {
+    // TODO(hta): Make pure virtual when all downstream has updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
   // MediaChannel creation
   // Creates a voice media channel. Returns NULL on failure.
   virtual VoiceMediaChannel* CreateMediaChannel(
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index c09e1374ea..0dc4b7a8a6 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -433,6 +433,28 @@ rtc::scoped_refptr<webrtc::AudioState> WebRtcVoiceEngine::GetAudioState()
   return audio_state_;
 }
 
+std::unique_ptr<VoiceMediaSendChannelInterface>
+WebRtcVoiceEngine::CreateSendChannel(
+    webrtc::Call* call,
+    const MediaConfig& config,
+    const AudioOptions& options,
+    const webrtc::CryptoOptions& crypto_options,
+    webrtc::AudioCodecPairId codec_pair_id) {
+  return std::make_unique<WebRtcVoiceSendChannel>(
+      this, config, options, crypto_options, call, codec_pair_id);
+}
+
+std::unique_ptr<VoiceMediaReceiveChannelInterface>
+WebRtcVoiceEngine::CreateReceiveChannel(
+    webrtc::Call* call,
+    const MediaConfig& config,
+    const AudioOptions& options,
+    const webrtc::CryptoOptions& crypto_options,
+    webrtc::AudioCodecPairId codec_pair_id) {
+  return std::make_unique<WebRtcVoiceReceiveChannel>(
+      this, config, options, crypto_options, call, codec_pair_id);
+}
+
 VoiceMediaChannel* WebRtcVoiceEngine::CreateMediaChannel(
     MediaChannel::Role role,
     webrtc::Call* call,
@@ -444,13 +466,13 @@ VoiceMediaChannel* WebRtcVoiceEngine::CreateMediaChannel(
   std::unique_ptr<VoiceMediaSendChannelInterface> send_channel;
   std::unique_ptr<VoiceMediaReceiveChannelInterface> receive_channel;
   if (role == MediaChannel::Role::kSend || role == MediaChannel::Role::kBoth) {
-    send_channel = std::make_unique<WebRtcVoiceSendChannel>(
-        this, config, options, crypto_options, call, codec_pair_id);
+    send_channel =
+        CreateSendChannel(call, config, options, crypto_options, codec_pair_id);
   }
   if (role == MediaChannel::Role::kReceive ||
       role == MediaChannel::Role::kBoth) {
-    receive_channel = std::make_unique<WebRtcVoiceReceiveChannel>(
-        this, config, options, crypto_options, call, codec_pair_id);
+    receive_channel = CreateReceiveChannel(call, config, options,
+                                           crypto_options, codec_pair_id);
   }
   return new VoiceMediaShimChannel(std::move(send_channel),
                                    std::move(receive_channel));
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index d7661a61d2..327beed3d0 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -105,8 +105,22 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
 
   // Does initialization that needs to occur on the worker thread.
   void Init() override;
-
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
+
+  std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) override;
+
+  std::unique_ptr<VoiceMediaReceiveChannelInterface> CreateReceiveChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) override;
+
   VoiceMediaChannel* CreateMediaChannel(
       MediaChannel::Role role,
       webrtc::Call* call,