diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index 65516192f168f229d0a1eb64d64d4a6369d28480..70f21a0626f1480e768fd4ea491271dc6978636e 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -33,7 +33,7 @@
 #include "AudioRtpFactory.h"
 #include "AudioZrtpSession.h"
 #include "AudioSrtpSession.h"
-#include "AudioRtpSession.h"
+#include "AudioSymmetricRtpSession.h"
 #include "manager.h"
 #include "sip/sdp.h"
 #include "sip/sipcall.h"
@@ -102,7 +102,7 @@ void AudioRtpFactory::registerAccount(SIPAccount *sipaccount, const std::string&
     _helloHashEnabled = sipaccount->getZrtpHelloHash();
 }
 
-void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
+void AudioRtpFactory::initAudioSymmetricRtpSession (SIPCall * ca)
 {
     ost::MutexLock m (_audioRtpThreadMutex);
 
@@ -138,7 +138,7 @@ void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
         }
     } else {
         _rtpSessionType = Symmetric;
-        _rtpSession = new AudioRtpSession (ca);
+        _rtpSession = new AudioSymmetricRtpSession (ca);
         _debug ("AudioRtpFactory: Starting a symmetric unencrypted rtp session");
     }
 }
@@ -166,8 +166,8 @@ void AudioRtpFactory::start (AudioCodec* audiocodec)
         case Symmetric:
             _debug ("Starting symmetric rtp thread");
 
-            if (static_cast<AudioRtpSession *> (_rtpSession)->startRtpThread (audiocodec) != 0) {
-                throw AudioRtpFactoryException ("AudioRtpFactory: Error: Failed to start AudioRtpSession thread");
+            if (static_cast<AudioSymmetricRtpSession *> (_rtpSession)->startRtpThread (audiocodec) != 0) {
+                throw AudioRtpFactoryException ("AudioRtpFactory: Error: Failed to start AudioSymmetricRtpSession thread");
             }
 
             break;
@@ -202,7 +202,7 @@ void AudioRtpFactory::stop (void)
                 break;
 
             case Symmetric:
-                static_cast<AudioRtpSession *> (_rtpSession)->stopRtpThread();
+                static_cast<AudioSymmetricRtpSession *> (_rtpSession)->stopRtpThread();
                 break;
 
             case Zrtp:
@@ -232,7 +232,7 @@ int AudioRtpFactory::getSessionMedia()
             payloadType = static_cast<AudioSrtpSession *> (_rtpSession)->getCodecPayloadType();
             break;
         case Symmetric:
-            payloadType = static_cast<AudioRtpSession *> (_rtpSession)->getCodecPayloadType();
+            payloadType = static_cast<AudioSymmetricRtpSession *> (_rtpSession)->getCodecPayloadType();
             break;
         case Zrtp:
             payloadType = static_cast<AudioZrtpSession *> (_rtpSession)->getCodecPayloadType();
@@ -255,7 +255,7 @@ void AudioRtpFactory::updateSessionMedia (AudioCodec *audiocodec)
             static_cast<AudioSrtpSession *> (_rtpSession)->updateSessionMedia (audiocodec);
             break;
         case Symmetric:
-            static_cast<AudioRtpSession *> (_rtpSession)->updateSessionMedia (audiocodec);
+            static_cast<AudioSymmetricRtpSession *> (_rtpSession)->updateSessionMedia (audiocodec);
             break;
         case Zrtp:
             static_cast<AudioZrtpSession *> (_rtpSession)->updateSessionMedia (audiocodec);
@@ -281,7 +281,7 @@ void AudioRtpFactory::updateDestinationIpAddress (void)
             break;
 
         case Symmetric:
-            static_cast<AudioRtpSession *> (_rtpSession)->updateDestinationIpAddress();
+            static_cast<AudioSymmetricRtpSession *> (_rtpSession)->updateDestinationIpAddress();
             break;
 
         case Zrtp:
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
index 939eabccbb5f9ba79c0020c2e0ab318c9efd240d..c4840f0ac8dc752b49d6e87da6fe30eae379a6ca 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
@@ -82,20 +82,20 @@ class AudioRtpFactory
          * 	Lazy instantiation method. Create a new RTP session of a given
          * type according to the content of the configuration file.
          * @param ca A pointer on a SIP call
-         * @return A new AudioRtpSession object
+         * @return A new AudioSymmetricRtpSession object
          */
-        void initAudioRtpSession (SIPCall *ca);
+        void initAudioSymmetricRtpSession (SIPCall *ca);
 
         /**
          * Start the audio rtp thread of the type specified in the configuration
-         * file. initAudioRtpSession must have been called prior to that.
+         * file. initAudioSymmetricRtpSession must have been called prior to that.
          * @param None
          */
         void start (AudioCodec*);
 
         /**
          * Stop the audio rtp thread of the type specified in the configuration
-         * file. initAudioRtpSession must have been called prior to that.
+         * file. initAudioSymmetricRtpSession must have been called prior to that.
          * @param None
          */
         void stop();
@@ -119,9 +119,9 @@ class AudioRtpFactory
         /**
          * @param None
          * @return The internal audio rtp thread of the type specified in the configuration
-         * file. initAudioRtpSession must have been called prior to that.
+         * file. initAudioSymmetricRtpSession must have been called prior to that.
          */
-        void * getAudioRtpSession (void) const {
+        void * getAudioSymmetricRtpSession (void) const {
             return _rtpSession;
         }
 
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index 1427eb68d0585ad3669f067773a7f0ee544d8bf9..b44ab585d0f559ccabdd0df72769c370a364e197 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -227,7 +227,7 @@ void AudioRtpRecordHandler::updateNoiseSuppress()
 
     _audioRtpRecord._noiseSuppress = NULL;
 
-    _debug ("AudioRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize());
+    _debug ("AudioSymmetricRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize());
 
     NoiseSuppress *noiseSuppress = new NoiseSuppress (getCodecFrameSize(), getCodecSampleRate());
     AudioProcessing *processing = new AudioProcessing (noiseSuppress);
@@ -249,7 +249,7 @@ void AudioRtpRecordHandler::putDtmfEvent (int digit)
     dtmf->newevent = true;
     dtmf->length = 1000;
     getEventQueue()->push_back (dtmf);
-    _debug ("AudioRtpSession: Put Dtmf Event %d", digit);
+    _debug ("AudioSymmetricRtpSession: Put Dtmf Event %d", digit);
 }
 
 #ifdef DUMP_PROCESS_DATA_ENCODE
diff --git a/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp
index c7f3269bf696cf4386b43c5c2004156611318410..6eee958ebfd1fd2183d4b855361b34fb50fe9d6d 100644
--- a/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioSrtpSession.cpp
@@ -46,7 +46,7 @@ namespace sfl
 {
 
 AudioSrtpSession::AudioSrtpSession (SIPCall * sipcall) :
-    AudioRtpSession (sipcall),
+    AudioSymmetricRtpSession (sipcall),
     _remoteCryptoCtx (NULL),
     _localCryptoCtx (NULL),
     _localCryptoSuite (0),
diff --git a/sflphone-common/src/audio/audiortp/AudioSrtpSession.h b/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
index 86c2718ca82328f1c402448711d5ac50cb7357d4..73e8cd78ef6f89af50d172350ed61e4948239ba0 100644
--- a/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioSrtpSession.h
@@ -30,7 +30,7 @@
 #ifndef __SFL_AUDIO_SRTP_SESSION_H__
 #define __SFL_AUDIO_SRTP_SESSION_H__
 
-#include "AudioRtpSession.h"
+#include "AudioSymmetricRtpSession.h"
 #include "sip/SdesNegotiator.h"
 
 #include <ccrtp/CryptoContext.h>
@@ -66,7 +66,7 @@ class SIPCall;
 namespace sfl
 {
 
-class AudioSrtpSession : public AudioRtpSession
+class AudioSrtpSession : public AudioSymmetricRtpSession
 {
     public:
 
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.cpp
similarity index 71%
rename from sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
rename to sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.cpp
index 59d96332e58f06f6aa670900449ee66654bd7709..cd98dedf5c19d2e3e1abb9e77a85fb52adae6177 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.cpp
@@ -32,14 +32,14 @@
  *  as that of the covered work.
  */
 
-#include "AudioRtpSession.h"
+#include "AudioSymmetricRtpSession.h"
 
 #include "sip/sdp.h"
 #include "audio/audiolayer.h"
 namespace sfl
 {
 
-AudioRtpSession::AudioRtpSession (SIPCall * sipcall) :
+AudioSymmetricRtpSession::AudioSymmetricRtpSession (SIPCall * sipcall) :
     AudioRtpRecordHandler (sipcall)
     , ost::SymmetricRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort())
     , _mainloopSemaphore (0)
@@ -53,37 +53,37 @@ AudioRtpSession::AudioRtpSession (SIPCall * sipcall) :
 {
     assert (_ca);
 
-    _info ("AudioRtpSession: Setting new RTP session with destination %s:%d", _ca->getLocalIp().c_str(), _ca->getLocalAudioPort());
+    _info ("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", _ca->getLocalIp().c_str(), _ca->getLocalAudioPort());
 
     _audioRtpRecord._callId = _ca->getCallId();
 
     setTypeOfService (tosEnhanced);
 }
 
-AudioRtpSession::~AudioRtpSession()
+AudioSymmetricRtpSession::~AudioSymmetricRtpSession()
 {
-    _info ("AudioRtpSession: Delete AudioRtpSession instance");
+    _info ("AudioSymmetricRtpSession: Delete AudioSymmetricRtpSession instance");
 }
 
-void AudioRtpSession::final()
+void AudioSymmetricRtpSession::final()
 {
 
     delete _rtpThread;
 
-    delete static_cast<AudioRtpSession *> (this);
+    delete static_cast<AudioSymmetricRtpSession *> (this);
 }
 
-void AudioRtpSession::setSessionTimeouts (void)
+void AudioSymmetricRtpSession::setSessionTimeouts (void)
 {
-    _debug ("AudioRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout);
+    _debug ("AudioSymmetricRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout);
 
     setSchedulingTimeout (sfl::schedulingTimeout);
     setExpireTimeout (sfl::expireTimeout);
 }
 
-void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec)
+void AudioSymmetricRtpSession::setSessionMedia (AudioCodec *audioCodec)
 {
-    _debug ("AudioRtpSession: Set session media");
+    _debug ("AudioSymmetricRtpSession: Set session media");
 
     // set internal codec info for this session
     setRtpMedia (audioCodec);
@@ -101,19 +101,19 @@ void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec)
         _timestampIncrement = frameSize;
 
     _debug ("AudioRptSession: Codec payload: %d", payloadType);
-    _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate);
-    _debug ("AudioRtpSession: Codec frame size: %d", frameSize);
-    _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement);
+    _debug ("AudioSymmetricRtpSession: Codec sampling rate: %d", smplRate);
+    _debug ("AudioSymmetricRtpSession: Codec frame size: %d", frameSize);
+    _debug ("AudioSymmetricRtpSession: RTP timestamp increment: %d", _timestampIncrement);
 
     if (payloadType == g722PayloadType) {
-        _debug ("AudioRtpSession: Setting G722 payload format");
+        _debug ("AudioSymmetricRtpSession: Setting G722 payload format");
         setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
     } else {
         if (dynamic) {
-            _debug ("AudioRtpSession: Setting dynamic payload format");
+            _debug ("AudioSymmetricRtpSession: Setting dynamic payload format");
             setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
         } else {
-            _debug ("AudioRtpSession: Setting static payload format");
+            _debug ("AudioSymmetricRtpSession: Setting static payload format");
             setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
         }
     }
@@ -121,9 +121,9 @@ void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec)
     _ca->setRecordingSmplRate (getCodecSampleRate());
 }
 
-void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec)
+void AudioSymmetricRtpSession::updateSessionMedia (AudioCodec *audioCodec)
 {
-    _debug ("AudioRtpSession: Update session media");
+    _debug ("AudioSymmetricRtpSession: Update session media");
 
     // Update internal codec for this session
     updateRtpMedia (audioCodec);
@@ -140,19 +140,19 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec)
         _timestampIncrement = frameSize;
 
     _debug ("AudioRptSession: Codec payload: %d", payloadType);
-    _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate);
-    _debug ("AudioRtpSession: Codec frame size: %d", frameSize);
-    _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement);
+    _debug ("AudioSymmetricRtpSession: Codec sampling rate: %d", smplRate);
+    _debug ("AudioSymmetricRtpSession: Codec frame size: %d", frameSize);
+    _debug ("AudioSymmetricRtpSession: RTP timestamp increment: %d", _timestampIncrement);
 
     if (payloadType == g722PayloadType) {
-        _debug ("AudioRtpSession: Setting G722 payload format");
+        _debug ("AudioSymmetricRtpSession: Setting G722 payload format");
         setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
     } else {
         if (dynamic) {
-            _debug ("AudioRtpSession: Setting dynamic payload format");
+            _debug ("AudioSymmetricRtpSession: Setting dynamic payload format");
             setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
         } else {
-            _debug ("AudioRtpSession: Setting static payload format");
+            _debug ("AudioSymmetricRtpSession: Setting static payload format");
             setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
         }
     }
@@ -164,14 +164,14 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec)
 }
 
 
-void AudioRtpSession::setDestinationIpAddress (void)
+void AudioSymmetricRtpSession::setDestinationIpAddress (void)
 {
-    _info ("AudioRtpSession: Setting IP address for the RTP session");
+    _info ("AudioSymmetricRtpSession: Setting IP address for the RTP session");
 
     // Store remote ip in case we would need to forget current destination
     _remote_ip = ost::InetHostAddress (_ca->getLocalSDP()->getRemoteIP().c_str());
     if (!_remote_ip) {
-        _warn ("AudioRtpSession: Target IP address (%s) is not correct!",
+        _warn ("AudioSymmetricRtpSession: Target IP address (%s) is not correct!",
                _ca->getLocalSDP()->getRemoteIP().data());
         return;
     }
@@ -179,24 +179,24 @@ void AudioRtpSession::setDestinationIpAddress (void)
     // Store remote port in case we would need to forget current destination
     _remote_port = (unsigned short) _ca->getLocalSDP()->getRemoteAudioPort();
 
-    _info ("AudioRtpSession: New remote address for session: %s:%d",
+    _info ("AudioSymmetricRtpSession: New remote address for session: %s:%d",
            _ca->getLocalSDP()->getRemoteIP().data(), _remote_port);
 
     if (!addDestination (_remote_ip, _remote_port)) {
-        _warn ("AudioRtpSession: Can't add new destination to session!");
+        _warn ("AudioSymmetricRtpSession: Can't add new destination to session!");
         return;
     }
 }
 
-void AudioRtpSession::updateDestinationIpAddress (void)
+void AudioSymmetricRtpSession::updateDestinationIpAddress (void)
 {
-    _debug ("AudioRtpSession: Update destination ip address");
+    _debug ("AudioSymmetricRtpSession: Update destination ip address");
 
     // Destination address are stored in a list in ccrtp
     // This method remove the current destination entry
 
     if (!forgetDestination (_remote_ip, _remote_port, _remote_port+1)) {
-        _warn ("AudioRtpSession: Could not remove previous destination: %s:%d",
+        _warn ("AudioSymmetricRtpSession: Could not remove previous destination: %s:%d",
         						inet_ntoa(_remote_ip.getAddress()), _remote_port);
     }
 
@@ -205,9 +205,9 @@ void AudioRtpSession::updateDestinationIpAddress (void)
     setDestinationIpAddress();
 }
 
-void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
+void AudioSymmetricRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
 {
-    _debug ("AudioRtpSession: Send Dtmf");
+    _debug ("AudioSymmetricRtpSession: Send Dtmf");
 
     _timestamp += _timestampIncrement;
     dtmf->factor++;
@@ -250,7 +250,7 @@ void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
     }
 }
 
-bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
+bool AudioSymmetricRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
 {
     receiveSpeakerData ();
 
@@ -259,7 +259,7 @@ bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
 
 
 
-void AudioRtpSession::sendMicData()
+void AudioSymmetricRtpSession::sendMicData()
 {
     int compSize = processDataEncode();
 
@@ -277,7 +277,7 @@ void AudioRtpSession::sendMicData()
 }
 
 
-void AudioRtpSession::receiveSpeakerData ()
+void AudioSymmetricRtpSession::receiveSpeakerData ()
 {
 
     const ost::AppDataUnit* adu = NULL;
@@ -304,12 +304,12 @@ void AudioRtpSession::receiveSpeakerData ()
     delete adu;
 }
 
-int AudioRtpSession::startRtpThread (AudioCodec* audiocodec)
+int AudioSymmetricRtpSession::startRtpThread (AudioCodec* audiocodec)
 {
     if (_isStarted)
         return 0;
 
-    _debug ("AudioRtpSession: Starting main thread");
+    _debug ("AudioSymmetricRtpSession: Starting main thread");
 
     _isStarted = true;
     setSessionTimeouts();
@@ -323,26 +323,26 @@ int AudioRtpSession::startRtpThread (AudioCodec* audiocodec)
     return 0;
 }
 
-void AudioRtpSession::stopRtpThread ()
+void AudioSymmetricRtpSession::stopRtpThread ()
 {
-    _debug ("AudioRtpSession: Stoping main thread");
+    _debug ("AudioSymmetricRtpSession: Stoping main thread");
 
     _rtpThread->stopRtpThread();
 
     disableStack();
 }
 
-AudioRtpSession::AudioRtpThread::AudioRtpThread (AudioRtpSession *session) : rtpSession (session), running (true)
+AudioSymmetricRtpSession::AudioRtpThread::AudioRtpThread (AudioSymmetricRtpSession *session) : rtpSession (session), running (true)
 {
-    _debug ("AudioRtpSession: Create new rtp thread");
+    _debug ("AudioSymmetricRtpSession: Create new rtp thread");
 }
 
-AudioRtpSession::AudioRtpThread::~AudioRtpThread()
+AudioSymmetricRtpSession::AudioRtpThread::~AudioRtpThread()
 {
-    _debug ("AudioRtpSession: Delete rtp thread");
+    _debug ("AudioSymmetricRtpSession: Delete rtp thread");
 }
 
-void AudioRtpSession::AudioRtpThread::run()
+void AudioSymmetricRtpSession::AudioRtpThread::run()
 {
     int threadSleep = 20;
 
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
similarity index 91%
rename from sflphone-common/src/audio/audiortp/AudioRtpSession.h
rename to sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
index 4acd2c77a6bed5089b352ecb8995edd20b2a188c..8ac08c60dd14d881c1720c22a21687054105c827 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
@@ -54,17 +54,17 @@ using std::ptrdiff_t;
 namespace sfl
 {
 
-// class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue>
-class AudioRtpSession : public ost::TimerPort, public AudioRtpRecordHandler, public ost::SymmetricRTPSession
+// class AudioSymmetricRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue>
+class AudioSymmetricRtpSession : public ost::TimerPort, public AudioRtpRecordHandler, public ost::SymmetricRTPSession
 {
     public:
         /**
         * Constructor
         * @param sipcall The pointer on the SIP call
         */
-        AudioRtpSession (SIPCall* sipcall);
+        AudioSymmetricRtpSession (SIPCall* sipcall);
 
-        ~AudioRtpSession();
+        ~AudioSymmetricRtpSession();
 
         virtual void final ();
 
@@ -111,7 +111,7 @@ class AudioRtpSession : public ost::TimerPort, public AudioRtpRecordHandler, pub
         class AudioRtpThread : public ost::Thread, public ost::TimerPort
         {
             public:
-                AudioRtpThread (AudioRtpSession *session);
+                AudioRtpThread (AudioSymmetricRtpSession *session);
                 ~AudioRtpThread();
 
                 void stopRtpThread (void) {
@@ -121,7 +121,7 @@ class AudioRtpSession : public ost::TimerPort, public AudioRtpRecordHandler, pub
                 virtual void run();
 
             private:
-                AudioRtpSession *rtpSession;
+                AudioSymmetricRtpSession *rtpSession;
 
                 bool running;
         };
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
index 351e5303e6c3a44c7a630cd9a5cb00f62a258167..c673323cae578b53b63e9f2bd2c8219191ed3e8d 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
@@ -77,7 +77,7 @@ AudioZrtpSession::AudioZrtpSession (SIPCall * sipcall, const std::string& zidFil
 
 AudioZrtpSession::~AudioZrtpSession()
 {
-    _debug ("AudioZrtpSession: Delete AudioRtpSession instance");
+    _debug ("AudioZrtpSession: Delete AudioSymmetricRtpSession instance");
 
     try {
         terminate();
@@ -152,7 +152,7 @@ void AudioZrtpSession::setSessionTimeouts (void)
 
 void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec)
 {
-    _debug ("AudioRtpSession: Set session media");
+    _debug ("AudioSymmetricRtpSession: Set session media");
 
     // set internal codec info for this session
     setRtpMedia (audioCodec);
@@ -176,10 +176,10 @@ void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec)
 
     // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
     if (dynamic) {
-        _debug ("AudioRtpSession: Setting dynamic payload format");
+        _debug ("AudioSymmetricRtpSession: Setting dynamic payload format");
         setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
     } else {
-        _debug ("AudioRtpSession: Setting static payload format");
+        _debug ("AudioSymmetricRtpSession: Setting static payload format");
         setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
     }
 
@@ -187,7 +187,7 @@ void AudioZrtpSession::setSessionMedia (AudioCodec* audioCodec)
 
 void AudioZrtpSession::updateSessionMedia (AudioCodec *audioCodec)
 {
-    _debug ("AudioRtpSession: Update session media");
+    _debug ("AudioSymmetricRtpSession: Update session media");
 
     //
     updateRtpMedia (audioCodec);
@@ -204,16 +204,16 @@ void AudioZrtpSession::updateSessionMedia (AudioCodec *audioCodec)
         _timestampIncrement = frameSize;
 
     _debug ("AudioRptSession: Codec payload: %d", payloadType);
-    _debug ("AudioRtpSession: Codec sampling rate: %d", smplRate);
-    _debug ("AudioRtpSession: Codec frame size: %d", frameSize);
-    _debug ("AudioRtpSession: RTP timestamp increment: %d", _timestampIncrement);
+    _debug ("AudioSymmetricRtpSession: Codec sampling rate: %d", smplRate);
+    _debug ("AudioSymmetricRtpSession: Codec frame size: %d", frameSize);
+    _debug ("AudioSymmetricRtpSession: RTP timestamp increment: %d", _timestampIncrement);
 
     // Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
     if (dynamic) {
-        _debug ("AudioRtpSession: Setting dynamic payload format");
+        _debug ("AudioSymmetricRtpSession: Setting dynamic payload format");
         setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
     } else {
-        _debug ("AudioRtpSession: Setting static payload format");
+        _debug ("AudioSymmetricRtpSession: Setting static payload format");
         setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
     }
 }
diff --git a/sflphone-common/src/audio/audiortp/Makefile.am b/sflphone-common/src/audio/audiortp/Makefile.am
index fd485065caa2cd157d1fce197e83faa612827ae9..d31503edb899bcfe9487af72f5927ee216f2c063 100644
--- a/sflphone-common/src/audio/audiortp/Makefile.am
+++ b/sflphone-common/src/audio/audiortp/Makefile.am
@@ -3,7 +3,7 @@ include $(top_srcdir)/globals.mak
 noinst_LTLIBRARIES = libaudiortp.la
 
 libaudiortp_la_SOURCES = \
-		AudioRtpSession.cpp \
+		AudioSymmetricRtpSession.cpp \
 		AudioRtpRecordHandler.cpp \
 		AudioRtpFactory.cpp \
 		AudioZrtpSession.cpp \
@@ -13,7 +13,7 @@ libaudiortp_la_SOURCES = \
 noinst_HEADERS = \
 		AudioRtpRecordHandler.h \
 		AudioRtpFactory.h \
-		AudioRtpSession.h \
+		AudioSymmetricRtpSession.h \
 		AudioZrtpSession.h \
 		ZrtpSessionCallback.h \
 		AudioSrtpSession.h 
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 30709631100b7d71ecf682898ce1a14876c106ba..76ab7bfe02adaa82d6c73c7a4891a1f39cece9b6 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -716,7 +716,7 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
 	try {
 		_info ("UserAgent: Creating new rtp session");
 		call->getAudioRtp()->initAudioRtpConfig (call);
-		call->getAudioRtp()->initAudioRtpSession (call);
+		call->getAudioRtp()->initAudioSymmetricRtpSession (call);
 		call->getAudioRtp()->initLocalCryptoInfo (call);
 		_info ("UserAgent: Start audio rtp session");
 		call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
@@ -1020,7 +1020,7 @@ SIPVoIPLink::offhold (const CallID& id) throw (VoipLinkException)
         }
 
         call->getAudioRtp()->initAudioRtpConfig (call);
-        call->getAudioRtp()->initAudioRtpSession (call);
+        call->getAudioRtp()->initAudioSymmetricRtpSession (call);
         call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
 
     }
@@ -1790,7 +1790,7 @@ bool SIPVoIPLink::SIPNewIpToIpCall (const CallID& id, const std::string& to)
         // since SDES require crypto attribute.
         try {
             call->getAudioRtp()->initAudioRtpConfig (call);
-            call->getAudioRtp()->initAudioRtpSession (call);
+            call->getAudioRtp()->initAudioSymmetricRtpSession (call);
             call->getAudioRtp()->initLocalCryptoInfo (call);
             call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
         } catch (...) {
@@ -3495,7 +3495,7 @@ void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status)
         SIPAccount *account = (SIPAccount *) Manager::instance().getAccount (accountID);
 
         if (account->getSrtpFallback())
-            call->getAudioRtp()->initAudioRtpSession (call);
+            call->getAudioRtp()->initAudioSymmetricRtpSession (call);
     }
 
     if (!sdpSession)
@@ -3972,7 +3972,7 @@ transaction_request_cb (pjsip_rx_data *rdata)
     try {
         _debug ("UserAgent: Create RTP session for this call");
         call->getAudioRtp()->initAudioRtpConfig (call);
-        call->getAudioRtp()->initAudioRtpSession (call);
+        call->getAudioRtp()->initAudioSymmetricRtpSession (call);
     } catch (...) {
         _warn ("UserAgent: Error: Failed to create rtp thread from answer");
     }
diff --git a/sflphone-common/test/rtptest.cpp b/sflphone-common/test/rtptest.cpp
index 56afefc3b8a899907b21dc64978312d00cb7596a..446aaef714b49d481577f971f2dfe97d0607e4d6 100644
--- a/sflphone-common/test/rtptest.cpp
+++ b/sflphone-common/test/rtptest.cpp
@@ -41,7 +41,7 @@
 #include <time.h>
 
 #include "rtptest.h"
-#include "audio/audiortp/AudioRtpSession.h"
+#include "audio/audiortp/AudioSymmetricRtpSession.h"
 
 #include <unistd.h>
 
diff --git a/sflphone-common/test/rtptest.h b/sflphone-common/test/rtptest.h
index 146d79e4c0995eb5a1b19c6f222c84797cfb9aba..7a5faa3b419a9852f1b171a496d4d020f5e8af52 100644
--- a/sflphone-common/test/rtptest.h
+++ b/sflphone-common/test/rtptest.h
@@ -61,7 +61,7 @@
 
 using namespace sfl;
 
-class AudioRtpSession;
+class AudioSymmetricRtpSession;
 //class AudioRtpFactory;
 class SIPVoIPLink;