diff --git a/test/mock_voice_engine.h b/test/mock_voice_engine.h
index e388993..45a9f9f 100644
--- a/test/mock_voice_engine.h
+++ b/test/mock_voice_engine.h
@@ -127,44 +127,6 @@
   MOCK_METHOD3(ReceivedRTCPPacket,
                int(int channel, const void* data, size_t length));
 
-  // VoERTP_RTCP
-  MOCK_METHOD2(SetLocalSSRC, int(int channel, unsigned int ssrc));
-  MOCK_METHOD2(GetLocalSSRC, int(int channel, unsigned int& ssrc));
-  MOCK_METHOD2(GetRemoteSSRC, int(int channel, unsigned int& ssrc));
-  MOCK_METHOD3(SetSendAudioLevelIndicationStatus,
-               int(int channel, bool enable, unsigned char id));
-  MOCK_METHOD3(SetReceiveAudioLevelIndicationStatus,
-               int(int channel, bool enable, unsigned char id));
-  MOCK_METHOD3(SetSendAbsoluteSenderTimeStatus,
-               int(int channel, bool enable, unsigned char id));
-  MOCK_METHOD3(SetReceiveAbsoluteSenderTimeStatus,
-               int(int channel, bool enable, unsigned char id));
-  MOCK_METHOD2(SetRTCPStatus, int(int channel, bool enable));
-  MOCK_METHOD2(GetRTCPStatus, int(int channel, bool& enabled));
-  MOCK_METHOD2(SetRTCP_CNAME, int(int channel, const char cName[256]));
-  MOCK_METHOD2(GetRTCP_CNAME, int(int channel, char cName[256]));
-  MOCK_METHOD2(GetRemoteRTCP_CNAME, int(int channel, char cName[256]));
-  MOCK_METHOD7(GetRemoteRTCPData,
-               int(int channel,
-                   unsigned int& NTPHigh,
-                   unsigned int& NTPLow,
-                   unsigned int& timestamp,
-                   unsigned int& playoutTimestamp,
-                   unsigned int* jitter,
-                   unsigned short* fraction_lost));
-  MOCK_METHOD4(GetRTPStatistics,
-               int(int channel,
-                   unsigned int& averageJitterMs,
-                   unsigned int& maxJitterMs,
-                   unsigned int& discardedPackets));
-  MOCK_METHOD2(GetRTCPStatistics, int(int channel, CallStatistics& stats));
-  MOCK_METHOD2(GetRemoteRTCPReportBlocks,
-               int(int channel, std::vector<ReportBlock>* receive_blocks));
-  MOCK_METHOD3(SetREDStatus, int(int channel, bool enable, int redPayloadtype));
-  MOCK_METHOD3(GetREDStatus,
-               int(int channel, bool& enable, int& redPayloadtype));
-  MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets));
-
  private:
   // TODO(ossu): I'm not particularly happy about keeping the decoder factory
   // here, but due to how gmock is implemented, I cannot just keep it in the
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 5f1ae03..600645e 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -19,7 +19,6 @@
     "include/voe_base.h",
     "include/voe_errors.h",
     "include/voe_network.h",
-    "include/voe_rtp_rtcp.h",
     "monitor_module.h",
     "output_mixer.cc",
     "output_mixer.h",
@@ -37,8 +36,6 @@
     "voe_base_impl.h",
     "voe_network_impl.cc",
     "voe_network_impl.h",
-    "voe_rtp_rtcp_impl.cc",
-    "voe_rtp_rtcp_impl.h",
     "voice_engine_defines.h",
     "voice_engine_impl.cc",
     "voice_engine_impl.h",
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 49bc381..a44ccb8 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -39,7 +39,6 @@
 #include "rtc_base/timeutils.h"
 #include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/trace.h"
-#include "voice_engine/include/voe_rtp_rtcp.h"
 #include "voice_engine/output_mixer.h"
 #include "voice_engine/statistics.h"
 #include "voice_engine/utility.h"
@@ -1576,11 +1575,6 @@
   return 0;
 }
 
-int Channel::GetLocalSSRC(unsigned int& ssrc) {
-  ssrc = _rtpRtcpModule->SSRC();
-  return 0;
-}
-
 int Channel::GetRemoteSSRC(unsigned int& ssrc) {
   ssrc = rtp_receiver_->SSRC();
   return 0;
@@ -1671,12 +1665,6 @@
   _rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
 }
 
-int Channel::GetRTCPStatus(bool& enabled) {
-  RtcpMode method = _rtpRtcpModule->RTCP();
-  enabled = (method != RtcpMode::kOff);
-  return 0;
-}
-
 int Channel::SetRTCP_CNAME(const char cName[256]) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetRTCP_CNAME()");
@@ -1689,69 +1677,6 @@
   return 0;
 }
 
-int Channel::GetRemoteRTCP_CNAME(char cName[256]) {
-  if (cName == NULL) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "GetRemoteRTCP_CNAME() invalid CNAME input buffer");
-    return -1;
-  }
-  char cname[RTCP_CNAME_SIZE];
-  const uint32_t remoteSSRC = rtp_receiver_->SSRC();
-  if (_rtpRtcpModule->RemoteCNAME(remoteSSRC, cname) != 0) {
-    _engineStatisticsPtr->SetLastError(
-        VE_CANNOT_RETRIEVE_CNAME, kTraceError,
-        "GetRemoteRTCP_CNAME() failed to retrieve remote RTCP CNAME");
-    return -1;
-  }
-  strcpy(cName, cname);
-  return 0;
-}
-
-int Channel::SendApplicationDefinedRTCPPacket(
-    unsigned char subType,
-    unsigned int name,
-    const char* data,
-    unsigned short dataLengthInBytes) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::SendApplicationDefinedRTCPPacket()");
-  if (!channel_state_.Get().sending) {
-    _engineStatisticsPtr->SetLastError(
-        VE_NOT_SENDING, kTraceError,
-        "SendApplicationDefinedRTCPPacket() not sending");
-    return -1;
-  }
-  if (NULL == data) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SendApplicationDefinedRTCPPacket() invalid data value");
-    return -1;
-  }
-  if (dataLengthInBytes % 4 != 0) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SendApplicationDefinedRTCPPacket() invalid length value");
-    return -1;
-  }
-  RtcpMode status = _rtpRtcpModule->RTCP();
-  if (status == RtcpMode::kOff) {
-    _engineStatisticsPtr->SetLastError(
-        VE_RTCP_ERROR, kTraceError,
-        "SendApplicationDefinedRTCPPacket() RTCP is disabled");
-    return -1;
-  }
-
-  // Create and schedule the RTCP APP packet for transmission
-  if (_rtpRtcpModule->SetRTCPApplicationSpecificData(
-          subType, name, (const unsigned char*)data, dataLengthInBytes) != 0) {
-    _engineStatisticsPtr->SetLastError(
-        VE_SEND_ERROR, kTraceError,
-        "SendApplicationDefinedRTCPPacket() failed to send RTCP packet");
-    return -1;
-  }
-  return 0;
-}
-
 int Channel::GetRemoteRTCPReportBlocks(
     std::vector<ReportBlock>* report_blocks) {
   if (report_blocks == NULL) {
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 5c95a64..48aeafe 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -56,13 +56,37 @@
 class RtpRtcp;
 class RtpTransportControllerSendInterface;
 class TelephoneEventHandler;
-class VoERTPObserver;
 class VoiceEngineObserver;
 
-struct CallStatistics;
-struct ReportBlock;
 struct SenderInfo;
 
+struct CallStatistics {
+  unsigned short fractionLost;
+  unsigned int cumulativeLost;
+  unsigned int extendedMax;
+  unsigned int jitterSamples;
+  int64_t rttMs;
+  size_t bytesSent;
+  int packetsSent;
+  size_t bytesReceived;
+  int packetsReceived;
+  // The capture ntp time (in local timebase) of the first played out audio
+  // frame.
+  int64_t capture_start_ntp_time_ms_;
+};
+
+// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
+struct ReportBlock {
+  uint32_t sender_SSRC;  // SSRC of sender
+  uint32_t source_SSRC;
+  uint8_t fraction_lost;
+  uint32_t cumulative_num_packets_lost;
+  uint32_t extended_highest_sequence_number;
+  uint32_t interarrival_jitter;
+  uint32_t last_SR_timestamp;
+  uint32_t delay_since_last_SR;
+};
+
 namespace voe {
 
 class OutputMixer;
@@ -215,10 +239,8 @@
   int SendTelephoneEventOutband(int event, int duration_ms);
   int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
 
-  // VoERTP_RTCP
+  // RTP+RTCP
   int SetLocalSSRC(unsigned int ssrc);
-  int GetLocalSSRC(unsigned int& ssrc);
-  int GetRemoteSSRC(unsigned int& ssrc);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
@@ -231,13 +253,7 @@
   void ResetSenderCongestionControlObjects();
   void ResetReceiverCongestionControlObjects();
   void SetRTCPStatus(bool enable);
-  int GetRTCPStatus(bool& enabled);
   int SetRTCP_CNAME(const char cName[256]);
-  int GetRemoteRTCP_CNAME(char cName[256]);
-  int SendApplicationDefinedRTCPPacket(unsigned char subType,
-                                       unsigned int name,
-                                       const char* data,
-                                       unsigned short dataLengthInBytes);
   int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
   int GetRTPStatistics(CallStatistics& stats);
   void SetNACKStatus(bool enable, int maxNumberOfPackets);
@@ -343,6 +359,7 @@
  private:
   class ProcessAndEncodeAudioTask;
 
+  int GetRemoteSSRC(unsigned int& ssrc);
   void OnUplinkPacketLossRate(float packet_loss_rate);
   bool InputMute() const;
   bool OnRtpPacketWithHeader(const uint8_t* received_packet,
diff --git a/voice_engine/channel_proxy.h b/voice_engine/channel_proxy.h
index ca11b58..40246aaa 100644
--- a/voice_engine/channel_proxy.h
+++ b/voice_engine/channel_proxy.h
@@ -18,8 +18,8 @@
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/thread_checker.h"
+#include "voice_engine/channel.h"
 #include "voice_engine/channel_manager.h"
-#include "voice_engine/include/voe_rtp_rtcp.h"
 
 #include <memory>
 #include <string>
@@ -42,8 +42,6 @@
 
 namespace voe {
 
-class Channel;
-
 // This class provides the "view" of a voe::Channel that we need to implement
 // webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
 // purposes:
diff --git a/voice_engine/include/voe_rtp_rtcp.h b/voice_engine/include/voe_rtp_rtcp.h
deleted file mode 100644
index e863a28..0000000
--- a/voice_engine/include/voe_rtp_rtcp.h
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
- *  Copyright (c) 2012 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.
- */
-
-// This sub-API supports the following functionalities:
-//
-//  - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
-//  - SSRC handling.
-//  - Transmission of RTCP sender reports.
-//  - Obtaining RTCP data from incoming RTCP sender reports.
-//  - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
-//  - Redundant Coding (RED)
-//  - Writing RTP and RTCP packets to binary files for off-line analysis of
-//    the call quality.
-//
-// Usage example, omitting error checking:
-//
-//  using namespace webrtc;
-//  VoiceEngine* voe = VoiceEngine::Create();
-//  VoEBase* base = VoEBase::GetInterface(voe);
-//  VoERTP_RTCP* rtp_rtcp  = VoERTP_RTCP::GetInterface(voe);
-//  base->Init();
-//  int ch = base->CreateChannel();
-//  ...
-//  rtp_rtcp->SetLocalSSRC(ch, 12345);
-//  ...
-//  base->DeleteChannel(ch);
-//  base->Terminate();
-//  base->Release();
-//  rtp_rtcp->Release();
-//  VoiceEngine::Delete(voe);
-//
-#ifndef VOICE_ENGINE_VOE_RTP_RTCP_H_
-#define VOICE_ENGINE_VOE_RTP_RTCP_H_
-
-#include <vector>
-#include "common_types.h"  // NOLINT(build/include)
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// VoERTPObserver
-class WEBRTC_DLLEXPORT VoERTPObserver {
- public:
-  virtual void OnIncomingCSRCChanged(int channel,
-                                     unsigned int CSRC,
-                                     bool added) = 0;
-
-  virtual void OnIncomingSSRCChanged(int channel, unsigned int SSRC) = 0;
-
- protected:
-  virtual ~VoERTPObserver() {}
-};
-
-// CallStatistics
-struct CallStatistics {
-  unsigned short fractionLost;
-  unsigned int cumulativeLost;
-  unsigned int extendedMax;
-  unsigned int jitterSamples;
-  int64_t rttMs;
-  size_t bytesSent;
-  int packetsSent;
-  size_t bytesReceived;
-  int packetsReceived;
-  // The capture ntp time (in local timebase) of the first played out audio
-  // frame.
-  int64_t capture_start_ntp_time_ms_;
-};
-
-// See section 6.4.1 in http://www.ietf.org/rfc/rfc3550.txt for details.
-struct SenderInfo {
-  uint32_t NTP_timestamp_high;
-  uint32_t NTP_timestamp_low;
-  uint32_t RTP_timestamp;
-  uint32_t sender_packet_count;
-  uint32_t sender_octet_count;
-};
-
-// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
-struct ReportBlock {
-  uint32_t sender_SSRC;  // SSRC of sender
-  uint32_t source_SSRC;
-  uint8_t fraction_lost;
-  uint32_t cumulative_num_packets_lost;
-  uint32_t extended_highest_sequence_number;
-  uint32_t interarrival_jitter;
-  uint32_t last_SR_timestamp;
-  uint32_t delay_since_last_SR;
-};
-
-// VoERTP_RTCP
-class WEBRTC_DLLEXPORT VoERTP_RTCP {
- public:
-  // Factory for the VoERTP_RTCP sub-API. Increases an internal
-  // reference counter if successful. Returns NULL if the API is not
-  // supported or if construction fails.
-  static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine);
-
-  // Releases the VoERTP_RTCP sub-API and decreases an internal
-  // reference counter. Returns the new reference count. This value should
-  // be zero for all sub-API:s before the VoiceEngine object can be safely
-  // deleted.
-  virtual int Release() = 0;
-
-  // Sets the local RTP synchronization source identifier (SSRC) explicitly.
-  virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
-
-  // Gets the local RTP SSRC of a specified |channel|.
-  virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
-
-  // Gets the SSRC of the incoming RTP packets.
-  virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0;
-
-  // Sets the status of rtp-audio-level-indication on a specific |channel|.
-  virtual int SetSendAudioLevelIndicationStatus(int channel,
-                                                bool enable,
-                                                unsigned char id = 1) = 0;
-
-  // Sets the RTCP status on a specific |channel|.
-  virtual int SetRTCPStatus(int channel, bool enable) = 0;
-
-  // Gets the RTCP status on a specific |channel|.
-  virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
-
-  // Sets the canonical name (CNAME) parameter for RTCP reports on a
-  // specific |channel|.
-  virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
-
-  // Gets the canonical name (CNAME) parameter for incoming RTCP reports
-  // on a specific channel.
-  virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0;
-
-  // Gets RTCP statistics for a specific |channel|.
-  virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
-
- protected:
-  VoERTP_RTCP() {}
-  virtual ~VoERTP_RTCP() {}
-};
-
-}  // namespace webrtc
-
-#endif  // #ifndef VOICE_ENGINE_VOE_RTP_RTCP_H_
diff --git a/voice_engine/voe_rtp_rtcp_impl.cc b/voice_engine/voe_rtp_rtcp_impl.cc
deleted file mode 100644
index 15d5022..0000000
--- a/voice_engine/voe_rtp_rtcp_impl.cc
+++ /dev/null
@@ -1,201 +0,0 @@
-/*
- *  Copyright (c) 2012 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 "system_wrappers/include/file_wrapper.h"
-#include "system_wrappers/include/trace.h"
-#include "voice_engine/include/voe_errors.h"
-#include "voice_engine/voe_rtp_rtcp_impl.h"
-#include "voice_engine/voice_engine_impl.h"
-
-#include "voice_engine/channel.h"
-#include "voice_engine/transmit_mixer.h"
-
-namespace webrtc {
-
-VoERTP_RTCP* VoERTP_RTCP::GetInterface(VoiceEngine* voiceEngine) {
-  if (NULL == voiceEngine) {
-    return NULL;
-  }
-  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-  s->AddRef();
-  return s;
-}
-
-VoERTP_RTCPImpl::VoERTP_RTCPImpl(voe::SharedData* shared) : _shared(shared) {
-  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "VoERTP_RTCPImpl::VoERTP_RTCPImpl() - ctor");
-}
-
-VoERTP_RTCPImpl::~VoERTP_RTCPImpl() {
-  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "VoERTP_RTCPImpl::~VoERTP_RTCPImpl() - dtor");
-}
-
-int VoERTP_RTCPImpl::SetLocalSSRC(int channel, unsigned int ssrc) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "SetLocalSSRC(channel=%d, %lu)", channel, ssrc);
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "SetLocalSSRC() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->SetLocalSSRC(ssrc);
-}
-
-int VoERTP_RTCPImpl::GetLocalSSRC(int channel, unsigned int& ssrc) {
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "GetLocalSSRC() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->GetLocalSSRC(ssrc);
-}
-
-int VoERTP_RTCPImpl::GetRemoteSSRC(int channel, unsigned int& ssrc) {
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "GetRemoteSSRC() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->GetRemoteSSRC(ssrc);
-}
-
-int VoERTP_RTCPImpl::SetSendAudioLevelIndicationStatus(int channel,
-                                                       bool enable,
-                                                       unsigned char id) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "SetSendAudioLevelIndicationStatus(channel=%d, enable=%d,"
-               " ID=%u)",
-               channel, enable, id);
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  if (enable && (id < kVoiceEngineMinRtpExtensionId ||
-                 id > kVoiceEngineMaxRtpExtensionId)) {
-    // [RFC5285] The 4-bit id is the local identifier of this element in
-    // the range 1-14 inclusive.
-    _shared->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SetSendAudioLevelIndicationStatus() invalid ID parameter");
-    return -1;
-  }
-
-  // Set state and id for the specified channel.
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(
-        VE_CHANNEL_NOT_VALID, kTraceError,
-        "SetSendAudioLevelIndicationStatus() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->SetSendAudioLevelIndicationStatus(enable, id);
-}
-
-int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "SetRTCPStatus(channel=%d, enable=%d)", channel, enable);
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "SetRTCPStatus() failed to locate channel");
-    return -1;
-  }
-  channelPtr->SetRTCPStatus(enable);
-  return 0;
-}
-
-int VoERTP_RTCPImpl::GetRTCPStatus(int channel, bool& enabled) {
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "GetRTCPStatus() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->GetRTCPStatus(enabled);
-}
-
-int VoERTP_RTCPImpl::SetRTCP_CNAME(int channel, const char cName[256]) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "SetRTCP_CNAME(channel=%d, cName=%s)", channel, cName);
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "SetRTCP_CNAME() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->SetRTCP_CNAME(cName);
-}
-
-int VoERTP_RTCPImpl::GetRemoteRTCP_CNAME(int channel, char cName[256]) {
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "GetRemoteRTCP_CNAME() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->GetRemoteRTCP_CNAME(cName);
-}
-
-int VoERTP_RTCPImpl::GetRTCPStatistics(int channel, CallStatistics& stats) {
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "GetRTPStatistics() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->GetRTPStatistics(stats);
-}
-
-}  // namespace webrtc
diff --git a/voice_engine/voe_rtp_rtcp_impl.h b/voice_engine/voe_rtp_rtcp_impl.h
deleted file mode 100644
index b640bdf..0000000
--- a/voice_engine/voe_rtp_rtcp_impl.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  Copyright (c) 2012 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 VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H_
-#define VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H_
-
-#include "voice_engine/include/voe_rtp_rtcp.h"
-
-#include "voice_engine/shared_data.h"
-
-namespace webrtc {
-
-class VoERTP_RTCPImpl : public VoERTP_RTCP {
- public:
-  // RTCP
-  int SetRTCPStatus(int channel, bool enable) override;
-
-  int GetRTCPStatus(int channel, bool& enabled) override;
-
-  int SetRTCP_CNAME(int channel, const char cName[256]) override;
-
-  int GetRemoteRTCP_CNAME(int channel, char cName[256]) override;
-
-  // SSRC
-  int SetLocalSSRC(int channel, unsigned int ssrc) override;
-
-  int GetLocalSSRC(int channel, unsigned int& ssrc) override;
-
-  int GetRemoteSSRC(int channel, unsigned int& ssrc) override;
-
-  // RTP Header Extension for Client-to-Mixer Audio Level Indication
-  int SetSendAudioLevelIndicationStatus(int channel,
-                                        bool enable,
-                                        unsigned char id) override;
-
-  int GetRTCPStatistics(int channel, CallStatistics& stats) override;
-
- protected:
-  VoERTP_RTCPImpl(voe::SharedData* shared);
-  ~VoERTP_RTCPImpl() override;
-
- private:
-  voe::SharedData* _shared;
-};
-
-}  // namespace webrtc
-
-#endif  // VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H_
diff --git a/voice_engine/voice_engine_defines.h b/voice_engine/voice_engine_defines.h
index c16dd56..bc5eb1b 100644
--- a/voice_engine/voice_engine_defines.h
+++ b/voice_engine/voice_engine_defines.h
@@ -56,12 +56,6 @@
 // Highest minimum playout delay
 enum { kVoiceEngineMaxMinPlayoutDelayMs = 10000 };
 
-// RTP/RTCP
-// Min 4-bit ID for RTP extension (see section 4.2 in RFC 5285)
-enum { kVoiceEngineMinRtpExtensionId = 1 };
-// Max 4-bit ID for RTP extension
-enum { kVoiceEngineMaxRtpExtensionId = 14 };
-
 }  // namespace webrtc
 
 namespace webrtc {
diff --git a/voice_engine/voice_engine_impl.h b/voice_engine/voice_engine_impl.h
index ef03e70..2ede6c3 100644
--- a/voice_engine/voice_engine_impl.h
+++ b/voice_engine/voice_engine_impl.h
@@ -17,7 +17,6 @@
 #include "typedefs.h"  // NOLINT(build/include)
 #include "voice_engine/voe_base_impl.h"
 #include "voice_engine/voe_network_impl.h"
-#include "voice_engine/voe_rtp_rtcp_impl.h"
 
 namespace webrtc {
 namespace voe {
@@ -27,13 +26,11 @@
 class VoiceEngineImpl : public voe::SharedData,  // Must be the first base class
                         public VoiceEngine,
                         public VoENetworkImpl,
-                        public VoERTP_RTCPImpl,
                         public VoEBaseImpl {
  public:
   VoiceEngineImpl()
       : SharedData(),
         VoENetworkImpl(this),
-        VoERTP_RTCPImpl(this),
         VoEBaseImpl(this),
         _ref_count(0) {}
   ~VoiceEngineImpl() override { assert(_ref_count.Value() == 0); }