diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index e532515d604c30dac6e10e9e3a0fe8c3ce2fe2f7..646aa89b95d65652bb446f860364e011cc0ec92f 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -125,14 +125,14 @@ void AudioRtpFactory::initAudioRtpSession (SIPCall * ca)
 void AudioRtpFactory::start (void)
 {
     if (_rtpSession == NULL) {
-        throw AudioRtpFactoryException ("_rtpSession was null when trying to start audio thread");
+        throw AudioRtpFactoryException ("RTP: Error: _rtpSession was null when trying to start audio thread");
     }
 
     switch (_rtpSessionType) {
 
         case Sdes:
 	    if (static_cast<AudioSrtpSession *> (_rtpSession)->startRtpThread() != 0) {
-                throw AudioRtpFactoryException ("Failed to start AudioSRtpSession thread");
+                throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioSRtpSession thread");
             }
 	    break;
 
@@ -140,7 +140,7 @@ void AudioRtpFactory::start (void)
             _debug ("Starting symmetric rtp thread");
 
             if (static_cast<AudioSymmetricRtpSession *> (_rtpSession)->startRtpThread() != 0) {
-                throw AudioRtpFactoryException ("Failed to start AudioSymmetricRtpSession thread");
+                throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioSymmetricRtpSession thread");
             }
 
             break;
@@ -148,9 +148,8 @@ void AudioRtpFactory::start (void)
         case Zrtp:
 
             if (static_cast<AudioZrtpSession *> (_rtpSession)->startRtpThread() != 0) {
-                throw AudioRtpFactoryException ("Failed to start AudioZrtpSession thread");
+                throw AudioRtpFactoryException ("RTP: Error: Failed to start AudioZrtpSession thread");
             }
-
             break;
     }
 }
@@ -158,10 +157,10 @@ void AudioRtpFactory::start (void)
 void AudioRtpFactory::stop (void)
 {
     ost::MutexLock mutex (_audioRtpThreadMutex);
-    _debug ("Stopping audio rtp session");
+    _info("RTP: Stopping audio rtp session");
 
     if (_rtpSession == NULL) {
-        _debugException ("_rtpSession is null when trying to stop. Returning.");
+        _debugException ("RTP: Error: _rtpSession is null when trying to stop. Returning.");
         return;
     }
 
@@ -183,16 +182,16 @@ void AudioRtpFactory::stop (void)
 
         _rtpSession = NULL;
     } catch (...) {
-        _debugException ("Exception caught when stopping the audio rtp session");
-        throw AudioRtpFactoryException("caught exception in AudioRtpFactory::stop");
+        _debugException ("RTP: Error: Exception caught when stopping the audio rtp session");
+        throw AudioRtpFactoryException("RTP: Error: caught exception in AudioRtpFactory::stop");
     }
 }
 
 void AudioRtpFactory::updateDestinationIpAddress (void)
 {
-    _debug ("Updating IP address");
+    _info ("RTP: Updating IP address");
     if (_rtpSession == NULL) {
-        throw AudioRtpFactoryException ("_rtpSession was null when trying to update IP address");
+        throw AudioRtpFactoryException ("RTP: Error: _rtpSession was null when trying to update IP address");
     }
 
     switch (_rtpSessionType) {
@@ -216,19 +215,20 @@ sfl::AudioZrtpSession * AudioRtpFactory::getAudioZrtpSession()
     if ( (_rtpSessionType == Zrtp) && (_rtpSessionType != NULL)) {
         return static_cast<AudioZrtpSession *> (_rtpSession);
     } else {
-        throw AudioRtpFactoryException("_rtpSession is NULL in getAudioZrtpSession");
+        throw AudioRtpFactoryException("RTP: Error: _rtpSession is NULL in getAudioZrtpSession");
     }
 }
 
-  void AudioRtpFactory::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
+void AudioRtpFactory::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
 {
     if ( _rtpSession && _rtpSessionType && (_rtpSessionType == Sdes)) {
         static_cast<AudioSrtpSession *> (_rtpSession)->setRemoteCryptoInfo(nego);
     }
     else {
-        throw AudioRtpFactoryException("_rtpSession is NULL in setRemoteCryptoInfo");
+        throw AudioRtpFactoryException("RTP: Error: _rtpSession is NULL in setRemoteCryptoInfo");
     }
 }
+
 }
 
 
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
index cfa14165f1cf0f85960eb775b4b09fd476df7062..0076c968c8262ccd743a56b53d421439b1878dd4 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.h
@@ -63,27 +63,27 @@ namespace sfl {
 
 	void initAudioRtpConfig(SIPCall *ca);
 
-        /**
-         * 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
-         */
-        void initAudioRtpSession(SIPCall *ca);
-        
-        /**
-         * Start the audio rtp thread of the type specified in the configuration
-         * file. initAudioRtpSession must have been called prior to that.
-         * @param None
-         */
-        void start();
-     
-         /**
-         * Stop the audio rtp thread of the type specified in the configuration
-         * file. initAudioRtpSession must have been called prior to that.
-         * @param None
-         */
-        void stop();
+	/**
+	 * 	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
+	 */
+	void initAudioRtpSession(SIPCall *ca);
+
+	/**
+	 * Start the audio rtp thread of the type specified in the configuration
+	 * file. initAudioRtpSession must have been called prior to that.
+	 * @param None
+	 */
+	void start();
+
+	/**
+	 * Stop the audio rtp thread of the type specified in the configuration
+	 * file. initAudioRtpSession must have been called prior to that.
+	 * @param None
+	 */
+	void stop();
 
 	/**
          * Update current RTP destination address with one stored in call 
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
index 5b6b67101f740635b5eb7f9ebcc77343689eba93..cb82fe29e227fe0071d48c6aeda2cab3be5e9008 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.cpp
@@ -38,7 +38,6 @@ AudioZrtpSession::AudioZrtpSession (ManagerImpl * manager, SIPCall * sipcall, co
 {
     _debug ("AudioZrtpSession initialized");
     initializeZid();
-    // startZrtp();
 }
 
 void AudioZrtpSession::initializeZid (void)
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
index 2e41e04b45b42086f903e551207563cdd9fdfba7..138bd2684733881494e2e26f8e1af01c85fe931c 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
@@ -38,8 +38,8 @@ namespace sfl {
     class AudioZrtpSession : public ost::SymmetricZRTPSession, public AudioRtpSession<AudioZrtpSession> 
     {
         public:
-        AudioZrtpSession(ManagerImpl * manager, SIPCall * sipcall, const std::string& zidFilename);          
-            
+			AudioZrtpSession(ManagerImpl * manager, SIPCall * sipcall, const std::string& zidFilename);
+
         private:
             void initializeZid(void);
             std::string _zidFilename;
diff --git a/sflphone-common/src/audio/audiortp/ZrtpSessionCallback.cpp b/sflphone-common/src/audio/audiortp/ZrtpSessionCallback.cpp
index f866b39e007a7b9ffd11fb4c5307a05c6b9f0547..145792e0bce09f97977772b4602a12559e2d4566 100644
--- a/sflphone-common/src/audio/audiortp/ZrtpSessionCallback.cpp
+++ b/sflphone-common/src/audio/audiortp/ZrtpSessionCallback.cpp
@@ -150,7 +150,6 @@ ZrtpSessionCallback::showMessage (GnuZrtpCodes::MessageSeverity sev, int32_t sub
         msg = _infoMap[subCode];
 
         if (msg != NULL) {
-            _debug ("ZRTP Debug:");
         }
     }
 
@@ -158,7 +157,6 @@ ZrtpSessionCallback::showMessage (GnuZrtpCodes::MessageSeverity sev, int32_t sub
         msg = _warningMap[subCode];
 
         if (msg != NULL) {
-            _debug ("ZRTP Debug:");
         }
     }
 
@@ -166,10 +164,11 @@ ZrtpSessionCallback::showMessage (GnuZrtpCodes::MessageSeverity sev, int32_t sub
         msg = _severeMap[subCode];
 
         if (msg != NULL) {
-            _debug ("ZRTP Debug:");
         }
     }
 
+
+
     if (sev == ZrtpError) {
         if (subCode < 0) {  // received an error packet from peer
             subCode *= -1;
@@ -181,7 +180,7 @@ ZrtpSessionCallback::showMessage (GnuZrtpCodes::MessageSeverity sev, int32_t sub
         msg = _zrtpMap[subCode];
 
         if (msg != NULL) {
-            _debug ("ZRTP Debug: %s", msg->c_str());
+
         }
     }
 }
diff --git a/sflphone-common/src/sip/sdp.cpp b/sflphone-common/src/sip/sdp.cpp
index ba4e856b301a3009fd1a222f60b157bcbd67fa27..886dc1430f588511d7b3794f3cf77b96a6a31c83 100644
--- a/sflphone-common/src/sip/sdp.cpp
+++ b/sflphone-common/src/sip/sdp.cpp
@@ -489,9 +489,6 @@ void Sdp::set_negotiated_sdp (const pjmedia_sdp_session *sdp)
 
             pjmedia_sdp_attr_to_rtpmap (_pool, attribute, &rtpmap);
 
-            // _debug("================== set_negociated_offer ===================== %i", pj_strtoul(&rtpmap->pt));
-            // _debug("================== set_negociated_offer ===================== %s", current->desc.fmt[j].ptr);
-            // _debug("================== set_negociated_offer ===================== %i", atoi(current->desc.fmt[j].ptr));
             iter = codecs_list.find ( (AudioCodecType) pj_strtoul (&rtpmap->pt));
 
             if (iter==codecs_list.end())
@@ -627,7 +624,7 @@ void Sdp::set_remote_ip_from_sdp (const pjmedia_sdp_session *r_sdp)
 {
 
     std::string remote_ip (r_sdp->conn->addr.ptr, r_sdp->conn->addr.slen);
-    _debug ("            Remote IP from fetching SDP: %s", remote_ip.c_str());
+    _info ("SDP: Remote IP from fetching SDP: %s",  remote_ip.c_str());
     this->set_remote_ip (remote_ip);
 }
 
@@ -637,21 +634,21 @@ void Sdp::set_remote_audio_port_from_sdp (pjmedia_sdp_media *r_media)
     int remote_port;
 
     remote_port = r_media->desc.port;
-    _debug ("            Remote Audio Port from fetching SDP: %d", remote_port);
+    _info ("SDP: Remote Audio Port from fetching SDP: %d", remote_port);
     this->set_remote_audio_port (remote_port);
 }
 
 void Sdp::set_media_transport_info_from_remote_sdp (const pjmedia_sdp_session *remote_sdp)
 {
 
-    _debug ("Fetching media from sdp");
+    _info ("SDP: Fetching media from sdp");
 
     pjmedia_sdp_media *r_media;
 
     this->get_remote_sdp_media_from_offer (remote_sdp, &r_media);
 
     if (r_media==NULL) {
-        _debug ("SDP Failure: no remote sdp media found in the remote offer");
+        _warn ("SDP: Error: no remote sdp media found in the remote offer");
         return;
     }