diff --git a/CMakeLists.txt b/CMakeLists.txt
index bfedefa..69b0338 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -59,7 +59,7 @@ set(LIBDATACHANNEL_SOURCES
 	${CMAKE_CURRENT_SOURCE_DIR}/src/message.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/peerconnection.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/logcounter.cpp
-	${CMAKE_CURRENT_SOURCE_DIR}/src/rtcp.cpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpreceivingsession.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/sctptransport.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/threadpool.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/tls.cpp
@@ -67,7 +67,7 @@ set(LIBDATACHANNEL_SOURCES
 	${CMAKE_CURRENT_SOURCE_DIR}/src/processor.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/capi.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/rtppacketizationconfig.cpp
-	${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpsenderreportable.cpp
+	${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpsenderreporter.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/rtppacketizer.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/opusrtppacketizer.cpp
 	${CMAKE_CURRENT_SOURCE_DIR}/src/opuspacketizationhandler.cpp
@@ -92,7 +92,8 @@ set(LIBDATACHANNEL_HEADERS
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/configuration.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/datachannel.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/description.hpp
-	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcp.hpp
+	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcphandler.hpp
+	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcpreceivingsession.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/include.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/init.hpp
 	${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/log.hpp
@@ -311,9 +312,9 @@ if(NOT NO_EXAMPLES AND NOT CMAKE_SYSTEM_NAME STREQUAL "WindowsStore")
 	set(JSON_BuildTests OFF CACHE INTERNAL "")
 	add_subdirectory(deps/json)
 	add_subdirectory(examples/client)
+if(NOT NO_MEDIA)
 	add_subdirectory(examples/media)
 	add_subdirectory(examples/sfu-media)
-if(NOT NO_MEDIA)
     add_subdirectory(examples/streamer)
 endif()
 	add_subdirectory(examples/copy-paste)
diff --git a/examples/streamer/helpers.cpp b/examples/streamer/helpers.cpp
index a6b063e..be51e45 100644
--- a/examples/streamer/helpers.cpp
+++ b/examples/streamer/helpers.cpp
@@ -60,7 +60,7 @@ int gettimeofday(struct timeval *tv, struct timezone *tz)
 using namespace std;
 using namespace rtc;
 
-ClientTrackData::ClientTrackData(shared_ptr<Track> track, shared_ptr<RTCPSenderReportable> sender) {
+ClientTrackData::ClientTrackData(shared_ptr<Track> track, shared_ptr<RtcpSenderReporter> sender) {
     this->track = track;
     this->sender = sender;
 }
diff --git a/examples/streamer/helpers.hpp b/examples/streamer/helpers.hpp
index 557a897..4d09cef 100644
--- a/examples/streamer/helpers.hpp
+++ b/examples/streamer/helpers.hpp
@@ -23,9 +23,9 @@
 
 struct ClientTrackData {
     std::shared_ptr<rtc::Track> track;
-    std::shared_ptr<rtc::RTCPSenderReportable> sender;
+    std::shared_ptr<rtc::RtcpSenderReporter> sender;
 
-    ClientTrackData(std::shared_ptr<rtc::Track> track, std::shared_ptr<rtc::RTCPSenderReportable> sender);
+    ClientTrackData(std::shared_ptr<rtc::Track> track, std::shared_ptr<rtc::RtcpSenderReporter> sender);
 };
 
 struct Client {
diff --git a/examples/streamer/main.cpp b/examples/streamer/main.cpp
index e0a3986..b9c2954 100644
--- a/examples/streamer/main.cpp
+++ b/examples/streamer/main.cpp
@@ -217,9 +217,9 @@ shared_ptr<ClientTrackData> addVideo(const shared_ptr<PeerConnection> pc, const
     video.addSSRC(ssrc, cname, msid);
     auto track = pc->addTrack(video);
     // create RTP configuration
-    auto rtpConfig = shared_ptr<RTPPacketizationConfig>(new RTPPacketizationConfig(ssrc, cname, payloadType, H264RTPPacketizer::defaultClockRate));
+    auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate));
     // create packetizer
-    auto packetizer = shared_ptr<H264RTPPacketizer>(new H264RTPPacketizer(rtpConfig));
+    auto packetizer = shared_ptr<H264RtpPacketizer>(new H264RtpPacketizer(rtpConfig));
     // create H264 and RTCP SP handler
     shared_ptr<H264PacketizationHandler> h264Handler(new H264PacketizationHandler(H264PacketizationHandler::Separator::Length, packetizer));
     // set handler
@@ -235,9 +235,9 @@ shared_ptr<ClientTrackData> addAudio(const shared_ptr<PeerConnection> pc, const
     audio.addSSRC(ssrc, cname, msid);
     auto track = pc->addTrack(audio);
     // create RTP configuration
-    auto rtpConfig = shared_ptr<RTPPacketizationConfig>(new RTPPacketizationConfig(ssrc, cname, payloadType, OpusRTPPacketizer::defaultClockRate));
+    auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate));
     // create packetizer
-    auto packetizer = make_shared<OpusRTPPacketizer>(rtpConfig);
+    auto packetizer = make_shared<OpusRtpPacketizer>(rtpConfig);
     // create opus and RTCP SP handler
     auto opusHandler = make_shared<OpusPacketizationHandler>(packetizer);
     // set handler
@@ -449,8 +449,8 @@ void addToStream(shared_ptr<Client> client, bool isAddingVideo) {
         auto currentTime_s = currentTime_us / (1000 * 1000);
 
         // set start time of stream
-        video->sender->rtpConfig->setStartTime(currentTime_s, RTPPacketizationConfig::EpochStart::T1970);
-        audio->sender->rtpConfig->setStartTime(currentTime_s, RTPPacketizationConfig::EpochStart::T1970);
+        video->sender->rtpConfig->setStartTime(currentTime_s, RtpPacketizationConfig::EpochStart::T1970);
+        audio->sender->rtpConfig->setStartTime(currentTime_s, RtpPacketizationConfig::EpochStart::T1970);
 
         // start stat recording of RTCP SR
         video->sender->startRecording();
diff --git a/include/rtc/h264packetizationhandler.hpp b/include/rtc/h264packetizationhandler.hpp
index c0bcf97..06b50bb 100644
--- a/include/rtc/h264packetizationhandler.hpp
+++ b/include/rtc/h264packetizationhandler.hpp
@@ -16,22 +16,22 @@
  * along with this program; If not, see <http://www.gnu.org/licenses/>.
  */
 
-#ifndef H264PacketizationHandler_hpp
-#define H264PacketizationHandler_hpp
+#ifndef H264_PACKETIZATION_HANDLER_H
+#define H264_PACKETIZATION_HANDLER_H
 
 #if RTC_ENABLE_MEDIA
 
 #include "h264rtppacketizer.hpp"
 #include "nalunit.hpp"
-#include "rtcp.hpp"
-#include "rtcpsenderreportable.hpp"
+#include "rtcphandler.hpp"
+#include "rtcpsenderreporter.hpp"
 
 namespace rtc {
 
 /// Handler for H264 packetization
-class RTC_CPP_EXPORT H264PacketizationHandler : public RtcpHandler, public RTCPSenderReportable {
+class RTC_CPP_EXPORT H264PacketizationHandler : public RtcpHandler, public RtcpSenderReporter {
 	/// RTP packetizer for H264
-	const std::shared_ptr<H264RTPPacketizer> packetizer;
+	const std::shared_ptr<H264RtpPacketizer> packetizer;
 
 	const uint16_t maximumFragmentSize;
 
@@ -49,7 +49,7 @@ public:
 	/// Construct handler for H264 packetization.
 	/// @param separator Nal units separator
 	/// @param packetizer RTP packetizer for h264
-	H264PacketizationHandler(Separator separator, std::shared_ptr<H264RTPPacketizer> packetizer,
+	H264PacketizationHandler(Separator separator, std::shared_ptr<H264RtpPacketizer> packetizer,
 	                         uint16_t maximumFragmentSize = NalUnits::defaultMaximumFragmentSize);
 
 	/// Returns message unchanged
@@ -73,4 +73,4 @@ private:
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* H264PacketizationHandler_hpp */
+#endif /* H264_PACKETIZATION_HANDLER_H */
diff --git a/include/rtc/h264rtppacketizer.hpp b/include/rtc/h264rtppacketizer.hpp
index 3e2e804..0741208 100644
--- a/include/rtc/h264rtppacketizer.hpp
+++ b/include/rtc/h264rtppacketizer.hpp
@@ -16,8 +16,8 @@
  * along with this program; If not, see <http://www.gnu.org/licenses/>.
  */
 
-#ifndef H264RTPPacketizer_hpp
-#define H264RTPPacketizer_hpp
+#ifndef H264RtpPacketizer_hpp
+#define H264RtpPacketizer_hpp
 
 #if RTC_ENABLE_MEDIA
 
@@ -26,7 +26,7 @@
 namespace rtc {
 
 /// RTP packetization of h264 payload
-class RTC_CPP_EXPORT H264RTPPacketizer : public rtc::RTPPacketizer {
+class RTC_CPP_EXPORT H264RtpPacketizer : public RtpPacketizer {
 
 public:
 	/// Default clock rate for H264 in RTP
@@ -36,11 +36,11 @@ public:
 	/// @note RTP configuration is used in packetization process which may change some configuration
 	/// properties such as sequence number.
 	/// @param rtpConfig  RTP configuration
-	H264RTPPacketizer(std::shared_ptr<rtc::RTPPacketizationConfig> rtpConfig);
+	H264RtpPacketizer(std::shared_ptr<RtpPacketizationConfig> rtpConfig);
 };
 
 } // namespace rtc
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* H264RTPPacketizer_hpp */
+#endif /* H264RtpPacketizer_hpp */
diff --git a/include/rtc/nalunit.hpp b/include/rtc/nalunit.hpp
index 1ea40e3..e309e49 100644
--- a/include/rtc/nalunit.hpp
+++ b/include/rtc/nalunit.hpp
@@ -60,22 +60,22 @@ private:
 #pragma pack(pop)
 
 /// Nal unit
-struct RTC_CPP_EXPORT NalUnit : rtc::binary {
+struct RTC_CPP_EXPORT NalUnit : binary {
 	NalUnit(const NalUnit &unit) = default;
 	NalUnit(size_t size, bool includingHeader = true)
-	    : rtc::binary(size + (includingHeader ? 0 : 1)) {}
+	    : binary(size + (includingHeader ? 0 : 1)) {}
 
 	template <typename Iterator>
-	NalUnit(Iterator begin_, Iterator end_) : rtc::binary(begin_, end_) {}
+	NalUnit(Iterator begin_, Iterator end_) : binary(begin_, end_) {}
 
-	NalUnit(rtc::binary &&data) : rtc::binary(std::move(data)) {}
+	NalUnit(binary &&data) : binary(std::move(data)) {}
 
-	NalUnit() : rtc::binary(1) {}
+	NalUnit() : binary(1) {}
 
 	bool forbiddenBit() { return header()->forbiddenBit(); }
 	uint8_t nri() { return header()->nri(); }
 	uint8_t unitType() { return header()->unitType(); }
-	rtc::binary payload() {
+	binary payload() {
 		assert(size() >= 1);
 		return {begin() + 1, end()};
 	}
@@ -83,7 +83,7 @@ struct RTC_CPP_EXPORT NalUnit : rtc::binary {
 	void setForbiddenBit(bool isSet) { header()->setForbiddenBit(isSet); }
 	void setNRI(uint8_t nri) { header()->setNRI(nri); }
 	void setUnitType(uint8_t type) { header()->setUnitType(type); }
-	void setPayload(rtc::binary payload) {
+	void setPayload(binary payload) {
 		assert(size() >= 1);
 		erase(begin() + 1, end());
 		insert(end(), payload.begin(), payload.end());
@@ -101,13 +101,13 @@ struct RTC_CPP_EXPORT NalUnitFragmentA : NalUnit {
 	enum class FragmentType { Start, Middle, End };
 
 	NalUnitFragmentA(FragmentType type, bool forbiddenBit, uint8_t nri, uint8_t unitType,
-	                 rtc::binary data);
+	                 binary data);
 
 	static std::vector<NalUnitFragmentA> fragmentsFrom(NalUnit nalu, uint16_t maximumFragmentSize);
 
 	uint8_t unitType() { return fragmentHeader()->unitType(); }
 
-	rtc::binary payload() {
+	binary payload() {
 		assert(size() >= 2);
 		return {begin() + 2, end()};
 	}
@@ -124,7 +124,7 @@ struct RTC_CPP_EXPORT NalUnitFragmentA : NalUnit {
 
 	void setUnitType(uint8_t type) { fragmentHeader()->setUnitType(type); }
 
-	void setPayload(rtc::binary payload) {
+	void setPayload(binary payload) {
 		assert(size() >= 2);
 		erase(begin() + 2, end());
 		insert(end(), payload.begin(), payload.end());
@@ -145,7 +145,7 @@ protected:
 class RTC_CPP_EXPORT NalUnits : public std::vector<NalUnit> {
 public:
 	static const uint16_t defaultMaximumFragmentSize = 1400;
-	std::vector<rtc::binary>
+	std::vector<binary>
 	generateFragments(uint16_t maximumFragmentSize = NalUnits::defaultMaximumFragmentSize);
 };
 
diff --git a/include/rtc/opuspacketizationhandler.hpp b/include/rtc/opuspacketizationhandler.hpp
index 7191a76..ff9c89a 100644
--- a/include/rtc/opuspacketizationhandler.hpp
+++ b/include/rtc/opuspacketizationhandler.hpp
@@ -16,26 +16,26 @@
  * along with this program; If not, see <http://www.gnu.org/licenses/>.
  */
 
-#ifndef OpusPacketizationHandler_hpp
-#define OpusPacketizationHandler_hpp
+#ifndef RTC_OPUS_PACKETIZATION_HANDLER_H
+#define RTC_OPUS_PACKETIZATION_HANDLER_H
 
 #if RTC_ENABLE_MEDIA
 
 #include "opusrtppacketizer.hpp"
-#include "rtcp.hpp"
-#include "rtcpsenderreportable.hpp"
+#include "rtcphandler.hpp"
+#include "rtcpsenderreporter.hpp"
 
 namespace rtc {
 
 /// Handler for opus packetization
-class RTC_CPP_EXPORT OpusPacketizationHandler : public RtcpHandler, public RTCPSenderReportable {
+class RTC_CPP_EXPORT OpusPacketizationHandler : public RtcpHandler, public RtcpSenderReporter {
 	/// RTP packetizer for opus
-	const std::shared_ptr<OpusRTPPacketizer> packetizer;
+	const std::shared_ptr<OpusRtpPacketizer> packetizer;
 
 public:
 	/// Construct handler for opus packetization.
 	/// @param packetizer RTP packetizer for opus
-	OpusPacketizationHandler(std::shared_ptr<OpusRTPPacketizer> packetizer);
+	OpusPacketizationHandler(std::shared_ptr<OpusRtpPacketizer> packetizer);
 
 	/// Returns message unchanged
 	/// @param ptr message
@@ -49,4 +49,4 @@ public:
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* OpusPacketizationHandler_hpp */
+#endif /* RTC_OPUS_PACKETIZATION_HANDLER_H */
diff --git a/include/rtc/opusrtppacketizer.hpp b/include/rtc/opusrtppacketizer.hpp
index 510b020..b47b90d 100644
--- a/include/rtc/opusrtppacketizer.hpp
+++ b/include/rtc/opusrtppacketizer.hpp
@@ -16,8 +16,8 @@
  * along with this program; If not, see <http://www.gnu.org/licenses/>.
  */
 
-#ifndef OpusRTPPacketizer_hpp
-#define OpusRTPPacketizer_hpp
+#ifndef RTC_OPUS_RTP_PACKETIZER_H
+#define RTC_OPUS_RTP_PACKETIZER_H
 
 #if RTC_ENABLE_MEDIA
 
@@ -26,7 +26,7 @@
 namespace rtc {
 
 /// RTP packetizer for opus
-class RTC_CPP_EXPORT OpusRTPPacketizer : public rtc::RTPPacketizer {
+class RTC_CPP_EXPORT OpusRtpPacketizer : public RtpPacketizer {
 
 public:
 	/// default clock rate used in opus RTP communication
@@ -36,17 +36,17 @@ public:
 	/// @note RTP configuration is used in packetization process which may change some configuration
 	/// properties such as sequence number.
 	/// @param rtpConfig  RTP configuration
-	OpusRTPPacketizer(std::shared_ptr<rtc::RTPPacketizationConfig> rtpConfig);
+	OpusRtpPacketizer(std::shared_ptr<RtpPacketizationConfig> rtpConfig);
 
 	/// Creates RTP packet for given payload based on `rtpConfig`.
 	/// @note This function increase sequence number after packetization.
 	/// @param payload RTP payload
 	/// @param setMark This needs to be `false` for all RTP packets with opus payload
-	rtc::message_ptr packetize(rtc::binary payload, bool setMark) override;
+	message_ptr packetize(binary payload, bool setMark) override;
 };
 
 } // namespace rtc
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* OpusRTPPacketizer_hpp */
+#endif /* RTC_OPUS_RTP_PACKETIZER_H */
diff --git a/include/rtc/peerconnection.hpp b/include/rtc/peerconnection.hpp
index 4f4b5bc..e80528f 100644
--- a/include/rtc/peerconnection.hpp
+++ b/include/rtc/peerconnection.hpp
@@ -31,6 +31,7 @@
 #include "track.hpp"
 
 #include <atomic>
+#include <chrono>
 #include <functional>
 #include <future>
 #include <list>
diff --git a/include/rtc/rtc.h b/include/rtc/rtc.h
index 27650d2..52ec440 100644
--- a/include/rtc/rtc.h
+++ b/include/rtc/rtc.h
@@ -282,7 +282,7 @@ int rtcSetTrackRTPTimestamp(int id, uint32_t timestamp);
 /// @param timestamp Pointer for result
 int rtcGetPreviousTrackSenderReportTimestamp(int id, uint32_t * timestamp);
 
-/// Set `NeedsToReport` flag in RTCPSenderReportable handler identified by given track id
+/// Set `NeedsToReport` flag in RtcpSenderReporter handler identified by given track id
 /// @param id Track id
 int rtcSetNeedsToSendRTCPSR(int id);
 
diff --git a/include/rtc/rtc.hpp b/include/rtc/rtc.hpp
index f945817..1019c78 100644
--- a/include/rtc/rtc.hpp
+++ b/include/rtc/rtc.hpp
@@ -27,11 +27,14 @@
 
 #if RTC_ENABLE_MEDIA
 
-// opus/h264 streaming
+// RTCP handling
+#include "rtcpreceivingsession.hpp"
+
+// Opus/h264 streaming
 #include "h264packetizationhandler.hpp"
 #include "opuspacketizationhandler.hpp"
 
-#endif /* RTC_ENABLE_MEDIA */
+#endif // RTC_ENABLE_MEDIA
 
 // C API
 #include "rtc.h"
diff --git a/include/rtc/rtcphandler.hpp b/include/rtc/rtcphandler.hpp
new file mode 100644
index 0000000..2c7ee3e
--- /dev/null
+++ b/include/rtc/rtcphandler.hpp
@@ -0,0 +1,50 @@
+/**
+ * Copyright (c) 2020 Staz Modrzynski
+ * Copyright (c) 2020 Paul-Louis Ageneau
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef RTC_RTCP_HANDLER_H
+#define RTC_RTCP_HANDLER_H
+
+#include "include.hpp"
+#include "message.hpp"
+
+namespace rtc {
+
+class RTC_CPP_EXPORT RtcpHandler {
+protected:
+	// Use this callback when trying to send custom data (such as RTCP) to the client.
+	synchronized_callback<message_ptr> outgoingCallback;
+
+public:
+	// Called when there is traffic coming from the peer
+	virtual message_ptr incoming(message_ptr ptr) = 0;
+
+	// Called when there is traffic that needs to be sent to the peer
+	virtual message_ptr outgoing(message_ptr ptr) = 0;
+
+	// This callback is used to send traffic back to the peer.
+	void onOutgoing(const std::function<void(message_ptr)> &cb) {
+		this->outgoingCallback = synchronized_callback<message_ptr>(cb);
+	}
+
+	virtual bool requestKeyframe() { return false; }
+};
+
+} // namespace rtc
+
+#endif // RTC_RTCP_HANDLER_H
diff --git a/include/rtc/rtcp.hpp b/include/rtc/rtcpreceivingsession.hpp
similarity index 57%
rename from include/rtc/rtcp.hpp
rename to include/rtc/rtcpreceivingsession.hpp
index 04aed8f..2e2595a 100644
--- a/include/rtc/rtcp.hpp
+++ b/include/rtc/rtcpreceivingsession.hpp
@@ -17,58 +17,24 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef RTC_RTCP_H
-#define RTC_RTCP_H
+#ifndef RTC_RTCP_RECEIVING_SESSION_H
+#define RTC_RTCP_RECEIVING_SESSION_H
 
-#include <utility>
+#if RTC_ENABLE_MEDIA
 
 #include "include.hpp"
-#include "log.hpp"
+#include "rtcphandler.hpp"
 #include "message.hpp"
 #include "rtp.hpp"
 
 namespace rtc {
 
-class RTC_CPP_EXPORT RtcpHandler {
-protected:
-	/**
-	 * Use this callback when trying to send custom data (such as RTCP) to the client.
-	 */
-	synchronized_callback<rtc::message_ptr> outgoingCallback;
-
-public:
-	/**
-	 * Called when there is traffic coming from the peer
-	 * @param ptr
-	 * @return
-	 */
-	virtual rtc::message_ptr incoming(rtc::message_ptr ptr) = 0;
-
-	/**
-	 * Called when there is traffic that needs to be sent to the peer
-	 * @param ptr
-	 * @return
-	 */
-	virtual rtc::message_ptr outgoing(rtc::message_ptr ptr) = 0;
-
-	/**
-	 * This callback is used to send traffic back to the peer.
-	 * This callback skips calling the track's methods.
-	 * @param cb
-	 */
-	void onOutgoing(const std::function<void(rtc::message_ptr)> &cb);
-
-	virtual bool requestKeyframe() { return false; }
-};
-
-class Track;
-
 // An RtcpSession can be plugged into a Track to handle the whole RTCP session
 class RTC_CPP_EXPORT RtcpReceivingSession : public RtcpHandler {
 public:
-	rtc::message_ptr incoming(rtc::message_ptr ptr) override;
-	rtc::message_ptr outgoing(rtc::message_ptr ptr) override;
-	bool send(rtc::message_ptr ptr);
+	message_ptr incoming(message_ptr ptr) override;
+	message_ptr outgoing(message_ptr ptr) override;
+	bool send(message_ptr ptr);
 
 	void requestBitrate(unsigned int newBitrate);
 
@@ -88,4 +54,6 @@ protected:
 
 } // namespace rtc
 
-#endif // RTC_RTCP_H
+#endif // RTC_ENABLE_MEDIA
+
+#endif // RTC_RTCP_RECEIVING_SESSION_H
diff --git a/include/rtc/rtcpsenderreportable.hpp b/include/rtc/rtcpsenderreporter.hpp
similarity index 90%
rename from include/rtc/rtcpsenderreportable.hpp
rename to include/rtc/rtcpsenderreporter.hpp
index 9303ac3..2aad1c2 100644
--- a/include/rtc/rtcpsenderreportable.hpp
+++ b/include/rtc/rtcpsenderreporter.hpp
@@ -16,8 +16,8 @@
  * along with this program; If not, see <http://www.gnu.org/licenses/>.
  */
 
-#ifndef RTCPSenderReporter_hpp
-#define RTCPSenderReporter_hpp
+#ifndef RTC_RTCP_SENDER_REPORTABLE_H
+#define RTC_RTCP_SENDER_REPORTABLE_H
 
 #if RTC_ENABLE_MEDIA
 
@@ -27,7 +27,7 @@
 namespace rtc {
 
 /// Class for sending RTCP SR
-class RTC_CPP_EXPORT RTCPSenderReportable {
+class RTC_CPP_EXPORT RtcpSenderReporter {
 	bool needsToReport = false;
 
 	uint32_t packetCount = 0;
@@ -50,9 +50,9 @@ public:
 	const uint32_t &previousReportedTimestamp = _previousReportedTimestamp;
 
 	/// RTP configuration
-	const std::shared_ptr<RTPPacketizationConfig> rtpConfig;
+	const std::shared_ptr<RtpPacketizationConfig> rtpConfig;
 
-	RTCPSenderReportable(std::shared_ptr<RTPPacketizationConfig> rtpConfig);
+	RtcpSenderReporter(std::shared_ptr<RtpPacketizationConfig> rtpConfig);
 
 	/// Set `needsToReport` flag. Sender report will be sent before next RTP packet with same
 	/// timestamp.
@@ -90,4 +90,4 @@ protected:
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* RTCPSenderReporter_hpp */
+#endif /* RTC_RTCP_SENDER_REPORTABLE_H */
diff --git a/include/rtc/rtppacketizationconfig.hpp b/include/rtc/rtppacketizationconfig.hpp
index 5a871b6..0f5ed0e 100644
--- a/include/rtc/rtppacketizationconfig.hpp
+++ b/include/rtc/rtppacketizationconfig.hpp
@@ -16,8 +16,8 @@
  * along with this program; If not, see <http://www.gnu.org/licenses/>.
  */
 
-#ifndef RTPPacketizationConfig_hpp
-#define RTPPacketizationConfig_hpp
+#ifndef RTC_RTP_PACKETIZATION_CONFIG_H
+#define RTC_RTP_PACKETIZATION_CONFIG_H
 
 #if RTC_ENABLE_MEDIA
 
@@ -26,10 +26,10 @@
 namespace rtc {
 
 /// RTP configuration used in packetization process
-class RTC_CPP_EXPORT RTPPacketizationConfig {
+class RTC_CPP_EXPORT RtpPacketizationConfig {
 	uint32_t _startTimestamp = 0;
 	double _startTime_s = 0;
-	RTPPacketizationConfig(const RTPPacketizationConfig &) = delete;
+	RtpPacketizationConfig(const RtpPacketizationConfig &) = delete;
 
 public:
 	const SSRC ssrc;
@@ -65,7 +65,7 @@ public:
 	/// @param sequenceNumber Initial sequence number of RTP packets (random number is choosed if
 	/// nullopt)
 	/// @param timestamp Initial timastamp of RTP packets (random number is choosed if nullopt)
-	RTPPacketizationConfig(SSRC ssrc, std::string cname, uint8_t payloadType, uint32_t clockRate,
+	RtpPacketizationConfig(SSRC ssrc, std::string cname, uint8_t payloadType, uint32_t clockRate,
 	                       std::optional<uint16_t> sequenceNumber = std::nullopt,
 	                       std::optional<uint32_t> timestamp = std::nullopt);
 
@@ -92,4 +92,4 @@ public:
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* RTPPacketizationConfig_hpp */
+#endif /* RTC_RTP_PACKETIZATION_CONFIG_H */
diff --git a/include/rtc/rtppacketizer.hpp b/include/rtc/rtppacketizer.hpp
index 4ebdd82..85fabda 100644
--- a/include/rtc/rtppacketizer.hpp
+++ b/include/rtc/rtppacketizer.hpp
@@ -16,8 +16,8 @@
  * along with this program; If not, see <http://www.gnu.org/licenses/>.
  */
 
-#ifndef RTPPacketizer_hpp
-#define RTPPacketizer_hpp
+#ifndef RTC_RTP_PACKETIZER_H
+#define RTC_RTP_PACKETIZER_H
 
 #if RTC_ENABLE_MEDIA
 
@@ -26,19 +26,19 @@
 
 namespace rtc {
 
-/// Class responsizble for rtp packetization
-class RTC_CPP_EXPORT RTPPacketizer {
+/// Class responsible for RTP packetization
+class RTC_CPP_EXPORT RtpPacketizer {
 	static const auto rtpHeaderSize = 12;
 
 public:
-	// rtp configuration
-	const std::shared_ptr<RTPPacketizationConfig> rtpConfig;
+	// RTP configuration
+	const std::shared_ptr<RtpPacketizationConfig> rtpConfig;
 
 	/// Constructs packetizer with given RTP configuration.
 	/// @note RTP configuration is used in packetization process which may change some configuration
 	/// properties such as sequence number.
 	/// @param rtpConfig  RTP configuration
-	RTPPacketizer(std::shared_ptr<RTPPacketizationConfig> rtpConfig);
+	RtpPacketizer(std::shared_ptr<RtpPacketizationConfig> rtpConfig);
 
 	/// Creates RTP packet for given payload based on `rtpConfig`.
 	/// @note This function increase sequence number after packetization.
@@ -51,4 +51,4 @@ public:
 
 #endif /* RTC_ENABLE_MEDIA */
 
-#endif /* RTPPacketizer_hpp */
+#endif /* RTC_RTP_PACKETIZER_H */
diff --git a/include/rtc/track.hpp b/include/rtc/track.hpp
index 21d9a07..a6c0636 100644
--- a/include/rtc/track.hpp
+++ b/include/rtc/track.hpp
@@ -24,7 +24,7 @@
 #include "include.hpp"
 #include "message.hpp"
 #include "queue.hpp"
-#include "rtcp.hpp"
+#include "rtcphandler.hpp"
 
 #include <atomic>
 #include <variant>
diff --git a/src/capi.cpp b/src/capi.cpp
index f409be8..2247ee0 100644
--- a/src/capi.cpp
+++ b/src/capi.cpp
@@ -53,8 +53,8 @@ std::unordered_map<int, shared_ptr<PeerConnection>> peerConnectionMap;
 std::unordered_map<int, shared_ptr<DataChannel>> dataChannelMap;
 std::unordered_map<int, shared_ptr<Track>> trackMap;
 #if RTC_ENABLE_MEDIA
-std::unordered_map<int, shared_ptr<RTCPSenderReportable>> rtcpSenderMap;
-std::unordered_map<int, shared_ptr<RTPPacketizationConfig>> rtpConfigMap;
+std::unordered_map<int, shared_ptr<RtcpSenderReporter>> rtcpSenderMap;
+std::unordered_map<int, shared_ptr<RtpPacketizationConfig>> rtpConfigMap;
 #endif
 #if RTC_ENABLE_WEBSOCKET
 std::unordered_map<int, shared_ptr<WebSocket>> webSocketMap;
@@ -149,20 +149,20 @@ void eraseTrack(int tr) {
 
 #if RTC_ENABLE_MEDIA
 
-shared_ptr<RTCPSenderReportable> getRTCPSender(int id) {
+shared_ptr<RtcpSenderReporter> getRTCPSender(int id) {
 	std::lock_guard lock(mutex);
 	if (auto it = rtcpSenderMap.find(id); it != rtcpSenderMap.end())
 		return it->second;
 	else
-		throw std::invalid_argument("RTCPSenderReportable ID does not exist");
+		throw std::invalid_argument("RtcpSenderReporter ID does not exist");
 }
 
-void emplaceRTCPSender(shared_ptr<RTCPSenderReportable> ptr, int tr) {
+void emplaceRTCPSender(shared_ptr<RtcpSenderReporter> ptr, int tr) {
 	std::lock_guard lock(mutex);
 	rtcpSenderMap.emplace(std::make_pair(tr, ptr));
 }
 
-shared_ptr<RTPPacketizationConfig> getRTPConfig(int id) {
+shared_ptr<RtpPacketizationConfig> getRTPConfig(int id) {
 	std::lock_guard lock(mutex);
 	if (auto it = rtpConfigMap.find(id); it != rtpConfigMap.end())
 		return it->second;
@@ -170,7 +170,7 @@ shared_ptr<RTPPacketizationConfig> getRTPConfig(int id) {
 		throw std::invalid_argument("RTPConfiguration ID does not exist");
 }
 
-void emplaceRTPConfig(shared_ptr<RTPPacketizationConfig> ptr, int tr) {
+void emplaceRTPConfig(shared_ptr<RtpPacketizationConfig> ptr, int tr) {
 	std::lock_guard lock(mutex);
 	rtpConfigMap.emplace(std::make_pair(tr, ptr));
 }
@@ -190,14 +190,14 @@ Description::Direction rtcDirectionToDirection(rtcDirection direction) {
 	}
 }
 
-shared_ptr<RTPPacketizationConfig>
-getNewRTPPacketizationConfig(uint32_t ssrc, const char *cname, uint8_t payloadType,
+shared_ptr<RtpPacketizationConfig>
+getNewRtpPacketizationConfig(uint32_t ssrc, const char *cname, uint8_t payloadType,
                              uint32_t clockRate, uint16_t sequenceNumber, uint32_t timestamp) {
 	if (!cname) {
 		throw std::invalid_argument("Unexpected null pointer for cname");
 	}
 
-	return std::make_shared<RTPPacketizationConfig>(ssrc, cname, payloadType, clockRate,
+	return std::make_shared<RtpPacketizationConfig>(ssrc, cname, payloadType, clockRate,
 	                                                sequenceNumber, timestamp);
 }
 
@@ -533,10 +533,10 @@ int rtcSetH264PacketizationHandler(int tr, uint32_t ssrc, const char *cname, uin
 	return WRAP({
 		auto track = getTrack(tr);
 		// create RTP configuration
-		auto rtpConfig = getNewRTPPacketizationConfig(ssrc, cname, payloadType, clockRate,
+		auto rtpConfig = getNewRtpPacketizationConfig(ssrc, cname, payloadType, clockRate,
 		                                              sequenceNumber, timestamp);
 		// create packetizer
-		auto packetizer = shared_ptr<H264RTPPacketizer>(new H264RTPPacketizer(rtpConfig));
+		auto packetizer = shared_ptr<H264RtpPacketizer>(new H264RtpPacketizer(rtpConfig));
 		// create H264 and RTCP SP handler
 		shared_ptr<H264PacketizationHandler> h264Handler(
 		    new H264PacketizationHandler(H264PacketizationHandler::Separator::Length, packetizer, maxFragmentSize));
@@ -553,10 +553,10 @@ int rtcSetOpusPacketizationHandler(int tr, uint32_t ssrc, const char *cname, uin
 	return WRAP({
 		auto track = getTrack(tr);
 		// create RTP configuration
-		auto rtpConfig = getNewRTPPacketizationConfig(ssrc, cname, payloadType, clockRate,
+		auto rtpConfig = getNewRtpPacketizationConfig(ssrc, cname, payloadType, clockRate,
 		                                              sequenceNumber, timestamp);
 		// create packetizer
-		auto packetizer = shared_ptr<OpusRTPPacketizer>(new OpusRTPPacketizer(rtpConfig));
+		auto packetizer = shared_ptr<OpusRtpPacketizer>(new OpusRtpPacketizer(rtpConfig));
 		// create Opus and RTCP SP handler
 		shared_ptr<OpusPacketizationHandler> opusHandler(new OpusPacketizationHandler(packetizer));
 		emplaceRTCPSender(opusHandler, tr);
@@ -570,9 +570,9 @@ int rtcSetRtpConfigurationStartTime(int id, double startTime_s, bool timeInterva
                                     uint32_t timestamp) {
 	return WRAP({
 		auto config = getRTPConfig(id);
-		auto epoch = RTPPacketizationConfig::EpochStart::T1900;
+		auto epoch = RtpPacketizationConfig::EpochStart::T1900;
 		if (timeIntervalSince1970) {
-			epoch = RTPPacketizationConfig::EpochStart::T1970;
+			epoch = RtpPacketizationConfig::EpochStart::T1970;
 		}
 		config->setStartTime(startTime_s, epoch, timestamp);
 	});
diff --git a/src/datachannel.cpp b/src/datachannel.cpp
index 93c1e95..98b0d45 100644
--- a/src/datachannel.cpp
+++ b/src/datachannel.cpp
@@ -28,10 +28,11 @@
 #include <arpa/inet.h>
 #endif
 
-rtc::LogCounter COUNTER_USERNEG_OPEN_MESSAGE(
-    plog::warning, "Number of open messages for a user-negotiated DataChannel received");
 namespace rtc {
 
+LogCounter COUNTER_USERNEG_OPEN_MESSAGE(
+    plog::warning, "Number of open messages for a user-negotiated DataChannel received");
+
 using std::shared_ptr;
 using std::weak_ptr;
 using std::chrono::milliseconds;
diff --git a/src/dtlssrtptransport.cpp b/src/dtlssrtptransport.cpp
index 3232c36..aa02a75 100644
--- a/src/dtlssrtptransport.cpp
+++ b/src/dtlssrtptransport.cpp
@@ -29,29 +29,29 @@ using std::shared_ptr;
 using std::to_integer;
 using std::to_string;
 
-static rtc::LogCounter COUNTER_MEDIA_TRUNCATED(plog::warning,
+namespace rtc {
+
+static LogCounter COUNTER_MEDIA_TRUNCATED(plog::warning,
                                                "Number of truncated SRT(C)P packets received");
-static rtc::LogCounter
+static LogCounter
     COUNTER_UNKNOWN_PACKET_TYPE(plog::warning,
                                 "Number of RTP packets received with an unknown packet type");
-static rtc::LogCounter COUNTER_SRTCP_REPLAY(plog::warning,
+static LogCounter COUNTER_SRTCP_REPLAY(plog::warning,
                                             "Number of SRTCP replay packets received");
-static rtc::LogCounter
+static LogCounter
     COUNTER_SRTCP_AUTH_FAIL(plog::warning,
                             "Number of SRTCP packets received that failed authentication checks");
-static rtc::LogCounter
+static LogCounter
     COUNTER_SRTCP_FAIL(plog::warning,
                        "Number of SRTCP packets received that had an unknown libSRTP failure");
-static rtc::LogCounter COUNTER_SRTP_REPLAY(plog::warning, "Number of SRTP replay packets received");
-static rtc::LogCounter
+static LogCounter COUNTER_SRTP_REPLAY(plog::warning, "Number of SRTP replay packets received");
+static LogCounter
     COUNTER_SRTP_AUTH_FAIL(plog::warning,
                            "Number of SRTP packets received that failed authentication checks");
 static rtc::LogCounter
     COUNTER_SRTP_FAIL(plog::warning,
                       "Number of SRTP packets received that had an unknown libSRTP failure");
 
-namespace rtc {
-
 void DtlsSrtpTransport::Init() { srtp_init(); }
 
 void DtlsSrtpTransport::Cleanup() { srtp_shutdown(); }
diff --git a/src/h264packetizationhandler.cpp b/src/h264packetizationhandler.cpp
index a0208d2..a391de9 100644
--- a/src/h264packetizationhandler.cpp
+++ b/src/h264packetizationhandler.cpp
@@ -76,7 +76,7 @@ NalUnitStartSequenceMatch StartSequenceMatchSucc(NalUnitStartSequenceMatch match
 
 message_ptr H264PacketizationHandler::incoming(message_ptr ptr) { return ptr; }
 
-shared_ptr<NalUnits> H264PacketizationHandler::splitMessage(rtc::message_ptr message) {
+shared_ptr<NalUnits> H264PacketizationHandler::splitMessage(message_ptr message) {
 	auto nalus = make_shared<NalUnits>();
 	if (separator == Separator::Length) {
 		unsigned long long index = 0;
@@ -155,9 +155,9 @@ message_ptr H264PacketizationHandler::outgoing(message_ptr ptr) {
 }
 
 H264PacketizationHandler::H264PacketizationHandler(Separator separator,
-                                                   std::shared_ptr<H264RTPPacketizer> packetizer,
+                                                   std::shared_ptr<H264RtpPacketizer> packetizer,
                                                    uint16_t maximumFragmentSize)
-    : RtcpHandler(), rtc::RTCPSenderReportable(packetizer->rtpConfig), packetizer(packetizer),
+    : RtcpHandler(), RtcpSenderReporter(packetizer->rtpConfig), packetizer(packetizer),
       maximumFragmentSize(maximumFragmentSize), separator(separator) {
 
 	senderReportOutgoingCallback = [this](message_ptr msg) { outgoingCallback(msg); };
diff --git a/src/h264rtppacketizer.cpp b/src/h264rtppacketizer.cpp
index 3eabef8..0c2e681 100644
--- a/src/h264rtppacketizer.cpp
+++ b/src/h264rtppacketizer.cpp
@@ -22,8 +22,8 @@
 
 namespace rtc {
 
-H264RTPPacketizer::H264RTPPacketizer(std::shared_ptr<RTPPacketizationConfig> rtpConfig)
-    : RTPPacketizer(rtpConfig) {}
+H264RtpPacketizer::H264RtpPacketizer(std::shared_ptr<RtpPacketizationConfig> rtpConfig)
+    : RtpPacketizer(rtpConfig) {}
 
 } // namespace rtc
 
diff --git a/src/logcounter.cpp b/src/logcounter.cpp
index ae3e562..dd226f5 100644
--- a/src/logcounter.cpp
+++ b/src/logcounter.cpp
@@ -18,15 +18,17 @@
 
 #include "logcounter.hpp"
 
-rtc::LogCounter::LogCounter(plog::Severity severity, const std::string &text,
-                            std::chrono::seconds duration) {
+namespace rtc {
+
+LogCounter::LogCounter(plog::Severity severity, const std::string &text,
+                       std::chrono::seconds duration) {
 	mData = std::make_shared<LogData>();
 	mData->mDuration = duration;
 	mData->mSeverity = severity;
 	mData->mText = text;
 }
 
-rtc::LogCounter &rtc::LogCounter::operator++(int) {
+LogCounter &LogCounter::operator++(int) {
 	if (mData->mCount++ == 0) {
 		ThreadPool::Instance().schedule(
 		    mData->mDuration,
@@ -43,4 +45,6 @@ rtc::LogCounter &rtc::LogCounter::operator++(int) {
 		    mData);
 	}
 	return *this;
-}
\ No newline at end of file
+}
+
+} // namespace rtc
diff --git a/src/logcounter.hpp b/src/logcounter.hpp
index d34407a..cd0e0bd 100644
--- a/src/logcounter.hpp
+++ b/src/logcounter.hpp
@@ -16,13 +16,17 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef WEBRTC_SERVER_LOGCOUNTER_HPP
-#define WEBRTC_SERVER_LOGCOUNTER_HPP
+#ifndef RTC_SERVER_LOGCOUNTER_HPP
+#define RTC_SERVER_LOGCOUNTER_HPP
 
 #include "include.hpp"
 #include "threadpool.hpp"
 
+#include <atomic>
+#include <chrono>
+
 namespace rtc {
+
 class LogCounter {
 private:
 	struct LogData {
@@ -41,6 +45,7 @@ public:
 
 	LogCounter &operator++(int);
 };
+
 } // namespace rtc
 
-#endif // WEBRTC_SERVER_LOGCOUNTER_HPP
+#endif // RTC_SERVER_LOGCOUNTER_HPP
diff --git a/src/opuspacketizationhandler.cpp b/src/opuspacketizationhandler.cpp
index b14987f..e058fdd 100644
--- a/src/opuspacketizationhandler.cpp
+++ b/src/opuspacketizationhandler.cpp
@@ -22,8 +22,8 @@
 
 namespace rtc {
 
-OpusPacketizationHandler::OpusPacketizationHandler(std::shared_ptr<OpusRTPPacketizer> packetizer)
-    : RtcpHandler(), RTCPSenderReportable(packetizer->rtpConfig), packetizer(packetizer) {
+OpusPacketizationHandler::OpusPacketizationHandler(std::shared_ptr<OpusRtpPacketizer> packetizer)
+    : RtcpHandler(), RtcpSenderReporter(packetizer->rtpConfig), packetizer(packetizer) {
 	senderReportOutgoingCallback = [this](message_ptr msg) { outgoingCallback(msg); };
 }
 
diff --git a/src/opusrtppacketizer.cpp b/src/opusrtppacketizer.cpp
index 0c0fcec..aa9fce0 100644
--- a/src/opusrtppacketizer.cpp
+++ b/src/opusrtppacketizer.cpp
@@ -22,12 +22,12 @@
 
 namespace rtc {
 
-OpusRTPPacketizer::OpusRTPPacketizer(std::shared_ptr<RTPPacketizationConfig> rtpConfig)
-    : RTPPacketizer(rtpConfig) {}
+OpusRtpPacketizer::OpusRtpPacketizer(std::shared_ptr<RtpPacketizationConfig> rtpConfig)
+    : RtpPacketizer(rtpConfig) {}
 
-message_ptr OpusRTPPacketizer::packetize(binary payload, bool setMark) {
+message_ptr OpusRtpPacketizer::packetize(binary payload, bool setMark) {
 	assert(!setMark);
-	return RTPPacketizer::packetize(payload, false);
+	return RtpPacketizer::packetize(payload, false);
 }
 
 } // namespace rtc
diff --git a/src/peerconnection.cpp b/src/peerconnection.cpp
index 626205e..3dc9f1a 100644
--- a/src/peerconnection.cpp
+++ b/src/peerconnection.cpp
@@ -23,6 +23,7 @@
 #include "logcounter.hpp"
 #include "processor.hpp"
 #include "threadpool.hpp"
+#include "rtp.hpp"
 
 #include "dtlstransport.hpp"
 #include "icetransport.hpp"
diff --git a/src/rtcp.cpp b/src/rtcpreceivingsession.cpp
similarity index 72%
rename from src/rtcp.cpp
rename to src/rtcpreceivingsession.cpp
index 1f92658..b4ae27f 100644
--- a/src/rtcp.cpp
+++ b/src/rtcpreceivingsession.cpp
@@ -17,10 +17,12 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "rtcp.hpp"
+#if RTC_ENABLE_MEDIA
 
+#include "rtcpreceivingsession.hpp"
 #include "logcounter.hpp"
 #include "track.hpp"
+
 #include <cmath>
 #include <utility>
 
@@ -30,18 +32,17 @@
 #include <arpa/inet.h>
 #endif
 
-static rtc::LogCounter COUNTER_BAD_RTP_HEADER(plog::warning, "Number of malformed RTP headers");
-static rtc::LogCounter COUNTER_UNKNOWN_PPID(plog::warning, "Number of Unknown PPID messages");
-static rtc::LogCounter COUNTER_BAD_NOTIF_LEN(plog::warning, "Number of Bad-Lengthed notifications");
-static rtc::LogCounter COUNTER_BAD_SCTP_STATUS(plog::warning,
-                                               "Number of unknown SCTP_STATUS errors");
-
 namespace rtc {
 
-rtc::message_ptr RtcpReceivingSession::outgoing(rtc::message_ptr ptr) { return ptr; }
+static LogCounter COUNTER_BAD_RTP_HEADER(plog::warning, "Number of malformed RTP headers");
+static LogCounter COUNTER_UNKNOWN_PPID(plog::warning, "Number of Unknown PPID messages");
+static LogCounter COUNTER_BAD_NOTIF_LEN(plog::warning, "Number of Bad-Lengthed notifications");
+static LogCounter COUNTER_BAD_SCTP_STATUS(plog::warning, "Number of unknown SCTP_STATUS errors");
 
-rtc::message_ptr RtcpReceivingSession::incoming(rtc::message_ptr ptr) {
-	if (ptr->type == rtc::Message::Type::Binary) {
+message_ptr RtcpReceivingSession::outgoing(message_ptr ptr) { return ptr; }
+
+message_ptr RtcpReceivingSession::incoming(message_ptr ptr) {
+	if (ptr->type == Message::Type::Binary) {
 		auto rtp = reinterpret_cast<const RTP *>(ptr->data());
 
 		// https://tools.ietf.org/html/rfc3550#appendix-A.1
@@ -65,7 +66,7 @@ rtc::message_ptr RtcpReceivingSession::incoming(rtc::message_ptr ptr) {
 		return ptr;
 	}
 
-	assert(ptr->type == rtc::Message::Type::Control);
+	assert(ptr->type == Message::Type::Control);
 	auto rr = reinterpret_cast<const RTCP_RR *>(ptr->data());
 	if (rr->header.payloadType() == 201) {
 		// RR
@@ -95,8 +96,7 @@ void RtcpReceivingSession::requestBitrate(unsigned int newBitrate) {
 }
 
 void RtcpReceivingSession::pushREMB(unsigned int bitrate) {
-	rtc::message_ptr msg =
-	    rtc::make_message(RTCP_REMB::sizeWithSSRCs(1), rtc::Message::Type::Control);
+	message_ptr msg = make_message(RTCP_REMB::sizeWithSSRCs(1), Message::Type::Control);
 	auto remb = reinterpret_cast<RTCP_REMB *>(msg->data());
 	remb->preparePacket(mSsrc, 1, bitrate);
 	remb->setSsrc(0, mSsrc);
@@ -105,7 +105,7 @@ void RtcpReceivingSession::pushREMB(unsigned int bitrate) {
 }
 
 void RtcpReceivingSession::pushRR(unsigned int lastSR_delay) {
-	auto msg = rtc::make_message(RTCP_RR::sizeWithReportBlocks(1), rtc::Message::Type::Control);
+	auto msg = make_message(RTCP_RR::sizeWithReportBlocks(1), Message::Type::Control);
 	auto rr = reinterpret_cast<RTCP_RR *>(msg->data());
 	rr->preparePacket(mSsrc, 1);
 	rr->getReportBlock(0)->preparePacket(mSsrc, 0, 0, uint16_t(mGreatestSeqNo), 0, 0, mSyncNTPTS,
@@ -131,13 +131,12 @@ bool RtcpReceivingSession::requestKeyframe() {
 }
 
 void RtcpReceivingSession::pushPLI() {
-	auto msg = rtc::make_message(rtc::RTCP_PLI::size(), rtc::Message::Type::Control);
-	auto *pli = reinterpret_cast<rtc::RTCP_PLI *>(msg->data());
+	auto msg = make_message(RTCP_PLI::size(), Message::Type::Control);
+	auto *pli = reinterpret_cast<RTCP_PLI *>(msg->data());
 	pli->preparePacket(mSsrc);
 	send(msg);
 }
 
-void RtcpHandler::onOutgoing(const std::function<void(rtc::message_ptr)> &cb) {
-	this->outgoingCallback = synchronized_callback<rtc::message_ptr>(cb);
-}
 } // namespace rtc
+
+#endif // RTC_ENABLE_MEDIA
diff --git a/src/rtcpsenderreportable.cpp b/src/rtcpsenderreporter.cpp
similarity index 80%
rename from src/rtcpsenderreportable.cpp
rename to src/rtcpsenderreporter.cpp
index f48e751..2fcc175 100644
--- a/src/rtcpsenderreportable.cpp
+++ b/src/rtcpsenderreporter.cpp
@@ -18,37 +18,37 @@
 
 #if RTC_ENABLE_MEDIA
 
-#include "rtcpsenderreportable.hpp"
+#include "rtcpsenderreporter.hpp"
 
 namespace rtc {
 
-void RTCPSenderReportable::startRecording() {
+void RtcpSenderReporter::startRecording() {
 	_previousReportedTimestamp = rtpConfig->timestamp;
 	timeOffset = rtpConfig->startTime_s - rtpConfig->timestampToSeconds(rtpConfig->timestamp);
 }
 
-void RTCPSenderReportable::sendReport(uint32_t timestamp) {
+void RtcpSenderReporter::sendReport(uint32_t timestamp) {
 	auto sr = getSenderReport(timestamp);
 	_previousReportedTimestamp = timestamp;
 	senderReportOutgoingCallback(move(sr));
 }
 
-void RTCPSenderReportable::addToReport(RTP *rtp, uint32_t rtpSize) {
+void RtcpSenderReporter::addToReport(RTP *rtp, uint32_t rtpSize) {
 	packetCount += 1;
 	assert(!rtp->padding());
 	payloadOctets += rtpSize - rtp->getSize();
 }
 
-RTCPSenderReportable::RTCPSenderReportable(std::shared_ptr<RTPPacketizationConfig> rtpConfig)
+RtcpSenderReporter::RtcpSenderReporter(std::shared_ptr<RtpPacketizationConfig> rtpConfig)
     : rtpConfig(rtpConfig) {}
 
-uint64_t RTCPSenderReportable::secondsToNTP(double seconds) {
+uint64_t RtcpSenderReporter::secondsToNTP(double seconds) {
 	return std::round(seconds * double(uint64_t(1) << 32));
 }
 
-void RTCPSenderReportable::setNeedsToReport() { needsToReport = true; }
+void RtcpSenderReporter::setNeedsToReport() { needsToReport = true; }
 
-message_ptr RTCPSenderReportable::getSenderReport(uint32_t timestamp) {
+message_ptr RtcpSenderReporter::getSenderReport(uint32_t timestamp) {
 	auto srSize = RTCP_SR::size(0);
 	auto msg = make_message(srSize + RTCP_SDES::size({{uint8_t(rtpConfig->cname.size())}}),
 	                        Message::Type::Control);
diff --git a/src/rtppacketizationconfig.cpp b/src/rtppacketizationconfig.cpp
index 6f1dcd4..09c1fd9 100644
--- a/src/rtppacketizationconfig.cpp
+++ b/src/rtppacketizationconfig.cpp
@@ -22,7 +22,7 @@
 
 namespace rtc {
 
-RTPPacketizationConfig::RTPPacketizationConfig(SSRC ssrc, string cname, uint8_t payloadType,
+RtpPacketizationConfig::RtpPacketizationConfig(SSRC ssrc, string cname, uint8_t payloadType,
                                                uint32_t clockRate,
                                                std::optional<uint16_t> sequenceNumber,
                                                std::optional<uint32_t> timestamp)
@@ -42,7 +42,7 @@ RTPPacketizationConfig::RTPPacketizationConfig(SSRC ssrc, string cname, uint8_t
 	this->_startTimestamp = this->timestamp;
 }
 
-void RTPPacketizationConfig::setStartTime(double startTime_s, EpochStart epochStart,
+void RtpPacketizationConfig::setStartTime(double startTime_s, EpochStart epochStart,
                                           std::optional<uint32_t> startTimestamp) {
 	this->_startTime_s = startTime_s + static_cast<unsigned long long>(epochStart);
 	if (startTimestamp.has_value()) {
@@ -53,20 +53,20 @@ void RTPPacketizationConfig::setStartTime(double startTime_s, EpochStart epochSt
 	}
 }
 
-double RTPPacketizationConfig::getSecondsFromTimestamp(uint32_t timestamp, uint32_t clockRate) {
+double RtpPacketizationConfig::getSecondsFromTimestamp(uint32_t timestamp, uint32_t clockRate) {
 	return double(timestamp) / double(clockRate);
 }
 
-double RTPPacketizationConfig::timestampToSeconds(uint32_t timestamp) {
-	return RTPPacketizationConfig::getSecondsFromTimestamp(timestamp, clockRate);
+double RtpPacketizationConfig::timestampToSeconds(uint32_t timestamp) {
+	return RtpPacketizationConfig::getSecondsFromTimestamp(timestamp, clockRate);
 }
 
-uint32_t RTPPacketizationConfig::getTimestampFromSeconds(double seconds, uint32_t clockRate) {
+uint32_t RtpPacketizationConfig::getTimestampFromSeconds(double seconds, uint32_t clockRate) {
 	return uint32_t(seconds * clockRate);
 }
 
-uint32_t RTPPacketizationConfig::secondsToTimestamp(double seconds) {
-	return RTPPacketizationConfig::getTimestampFromSeconds(seconds, clockRate);
+uint32_t RtpPacketizationConfig::secondsToTimestamp(double seconds) {
+	return RtpPacketizationConfig::getTimestampFromSeconds(seconds, clockRate);
 }
 
 } // namespace rtc
diff --git a/src/rtppacketizer.cpp b/src/rtppacketizer.cpp
index d8163f2..1e92c10 100644
--- a/src/rtppacketizer.cpp
+++ b/src/rtppacketizer.cpp
@@ -22,10 +22,10 @@
 
 namespace rtc {
 
-RTPPacketizer::RTPPacketizer(std::shared_ptr<RTPPacketizationConfig> rtpConfig)
+RtpPacketizer::RtpPacketizer(std::shared_ptr<RtpPacketizationConfig> rtpConfig)
     : rtpConfig(rtpConfig) {}
 
-message_ptr RTPPacketizer::packetize(binary payload, bool setMark) {
+message_ptr RtpPacketizer::packetize(binary payload, bool setMark) {
 	auto msg = make_message(rtpHeaderSize + payload.size());
 	auto *rtp = (RTP *)msg->data();
 	rtp->setPayloadType(rtpConfig->payloadType);
diff --git a/src/sctptransport.cpp b/src/sctptransport.cpp
index 8e01563..994eb20 100644
--- a/src/sctptransport.cpp
+++ b/src/sctptransport.cpp
@@ -56,15 +56,15 @@ using namespace std::chrono;
 
 using std::shared_ptr;
 
-static rtc::LogCounter COUNTER_UNKNOWN_PPID(plog::warning,
-                                            "Number of SCTP packets received with an unknown PPID");
-static rtc::LogCounter
+namespace rtc {
+
+static LogCounter COUNTER_UNKNOWN_PPID(plog::warning,
+                                       "Number of SCTP packets received with an unknown PPID");
+static LogCounter
     COUNTER_BAD_NOTIF_LEN(plog::warning,
                           "Number of SCTP packets received with an bad notification length");
-static rtc::LogCounter COUNTER_BAD_SCTP_STATUS(plog::warning,
-                                               "Number of SCTP packets received with a bad status");
-
-namespace rtc {
+static LogCounter COUNTER_BAD_SCTP_STATUS(plog::warning,
+                                          "Number of SCTP packets received with a bad status");
 
 std::unordered_set<SctpTransport *> SctpTransport::Instances;
 std::shared_mutex SctpTransport::InstancesMutex;