diff --git a/sflphone-client-gnome/src/contacts/calltree.c b/sflphone-client-gnome/src/contacts/calltree.c
index f87206be01e35176d50da316777963fe9ff60d52..d1bf035968d4411f05e90c0fdea0ec5a0e461871 100644
--- a/sflphone-client-gnome/src/contacts/calltree.c
+++ b/sflphone-client-gnome/src/contacts/calltree.c
@@ -412,8 +412,6 @@ calltree_update_call (calltab_t* tab, callable_obj_t * c)
 void calltree_add_history_entry (callable_obj_t * c)
 {
 
-    printf("calltree_add_history_entry\n");
-
     if (dbus_get_history_enabled () == 0)
         return;
 
@@ -473,8 +471,6 @@ void calltree_add_history_entry (callable_obj_t * c)
 void calltree_add_call (calltab_t* tab, callable_obj_t * c)
 {
 
-    printf("calltree_add_call\n");
-
     if (tab == history)
     {
         calltree_add_history_entry (c);
diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index cd9906b0ac370f6f068a001a5211330f43c2ea0b..80731ac081f03ee950a4aca5e6f32151a977c4ed 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -98,7 +98,7 @@ AudioRtp::closeRtpSession () {
         throw;
     }
     AudioLayer* audiolayer = Manager::instance().getAudioDriver();
-    audiolayer->stopStream();
+    // audiolayer->stopStream();
 
     return true;
 }
@@ -398,52 +398,6 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
 
     int compSize = processDataEncode(audiolayer);
 
-    /*
-    // compute codec framesize in ms
-    float fixed_codec_framesize = computeCodecFrameSize(_audiocodec->getFrameSize(), _audiocodec->getClockRate());
-
-    // compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
-    int maxBytesToGet = computeNbByteAudioLayer(fixed_codec_framesize);
-    
-    // available bytes inside ringbuffer
-    int availBytesFromMic = audiolayer->canGetMic();
-
-    // set available byte to maxByteToGet
-    int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
-
-    if (bytesAvail == 0)
-      return;
-
-    // Get bytes from micRingBuffer to data_from_mic
-    int nbSample = audiolayer->getMic( micData , bytesAvail ) / sizeof(SFLDataFormat);
-
-    // nb bytes to be sent over RTP
-    int compSize = 0;
-
-    // test if resampling is required
-    if (_audiocodec->getClockRate() != _layerSampleRate) {
-
-        int nb_sample_up = nbSample;
-        // _debug("_nbSample audiolayer->getMic(): %i \n", nbSample);
-    
-        // Store the length of the mic buffer in samples for recording
-        _nSamplesMic = nbSample;
-
-
-        // int nbSamplesMax = _layerFrameSize * _audiocodec->getClockRate() / 1000; 
-	 nbSample = reSampleData(micData , micDataConverted, _audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
-
-        compSize = _audiocodec->codecEncode( micDataEncoded, micDataConverted, nbSample*sizeof(int16));
-
-    } else {
-        // no resampling required
-
-        // int nbSamplesMax = _codecFrameSize;
-        compSize = _audiocodec->codecEncode( micDataEncoded, micData, nbSample*sizeof(int16));
-
-    }
-    */
-
     // putData put the data on RTP queue, sendImmediate bypass this queue
     if (!_sym) {
         // _sessionSend->putData(timestamp, micDataEncoded, compSize);
@@ -483,54 +437,10 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
     unsigned int size = adu->getSize(); // size in char
 
     processDataDecode(audiolayer, spkrData, size, countTime);
-    /*
-    if (_audiocodec != NULL) {
-
-        // Return the size of data in bytes 
-        int expandedSize = _audiocodec->codecDecode( spkrDataDecoded , spkrData , size );
-
-        // buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
-        int nbSample = expandedSize / sizeof(SFLDataFormat);
-
-        // test if resampling is required 
-        if (_audiocodec->getClockRate() != _layerSampleRate) {
-
-            // Do sample rate conversion
-            int nb_sample_down = nbSample;
-            nbSample = reSampleData(spkrDataDecoded , spkrDataConverted, _codecSampleRate , nb_sample_down, UP_SAMPLING);
-
-            // Store the number of samples for recording
-            _nSamplesSpkr = nbSample;
-
-	    // put data in audio layer, size in byte
-            audiolayer->putMain (spkrDataConverted, nbSample * sizeof(SFLDataFormat));
-
-        } else {
-
-            // Stor the number of samples for recording
-            _nSamplesSpkr = nbSample;
-
-	    // put data in audio layer, size in byte
-            audiolayer->putMain (spkrDataDecoded, nbSample * sizeof(SFLDataFormat));
-        }
-
-        // Notify (with a beep) an incoming call when there is already a call 
-        countTime += time->getSecond();
-        if (Manager::instance().incomingCallWaiting() > 0) {
-            countTime = countTime % 500; // more often...
-            if (countTime == 0) {
-                Manager::instance().notificationIncomingCall();
-            }
-        }
-
-    } else {
-        countTime += time->getSecond();
-    }
-    delete adu; adu = NULL;
-    */
-
+    
 }
 
+
     int 
     AudioRtpRTX::reSampleData(SFLDataFormat *input, SFLDataFormat *output, int sampleRate_codec, int nbSamples, int status)
 {
@@ -634,7 +544,7 @@ AudioRtpRTX::run () {
       
     }
     
-    audiolayer->stopStream();
+    // audiolayer->stopStream();
     _debug("- ARTP Action: Stop call %s\n",_ca->getCallId().c_str());
   //} catch(std::exception &e) {
     //_start.post();
diff --git a/sflphone-common/src/audio/audiostream.cpp b/sflphone-common/src/audio/audiostream.cpp
index 08587fdb310803fe561ae208c025869f6b7ab86b..abfb5ab34f92f598d327899751a6f7ed609d88e7 100644
--- a/sflphone-common/src/audio/audiostream.cpp
+++ b/sflphone-common/src/audio/audiostream.cpp
@@ -49,10 +49,6 @@ AudioStream::connectStream()
 
   if(!_audiostream)
     _audiostream = createStream( _context );
-  else {
-    disconnectStream();
-    _audiostream = createStream( _context );
-  }
 
   return true;
 }
diff --git a/sflphone-common/src/audio/pulselayer.cpp b/sflphone-common/src/audio/pulselayer.cpp
index a5b07f277b2280da0b5fea1ad6131ae5849b73e8..c8410651b536ddddecb21dc5ce513700295430fd 100644
--- a/sflphone-common/src/audio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulselayer.cpp
@@ -166,12 +166,15 @@ bool PulseLayer::createStreams( pa_context* c )
 
     pa_threaded_mainloop_signal(m , 0);
 
+    isCorked = true;
+
     return true;
 }
 
 
 bool PulseLayer::openDevice(int indexIn UNUSED, int indexOut UNUSED, int sampleRate, int frameSize , int stream UNUSED, std::string plugin UNUSED) 
 {
+
     _debug("PulseLayer::openDevice \n");
     _sampleRate = sampleRate;
     _frameSize = frameSize;	
@@ -225,35 +228,46 @@ int PulseLayer::getMic(void *buffer, int toCopy)
 void PulseLayer::startStream (void) 
 {
     // flushMic();
+    if (isCorked) {
 
-    _urgentRingBuffer.flush();
-    _micRingBuffer.flush();
-    _voiceRingBuffer.flush();
+        _urgentRingBuffer.flush();
+        _micRingBuffer.flush();
+        _voiceRingBuffer.flush();
 
-    _debug("PulseLayer::Start stream\n");
-    pa_threaded_mainloop_lock(m);
+        _debug("PulseLayer::Start stream\n");
+        pa_threaded_mainloop_lock(m);
    
-    pa_stream_cork( playback->pulseStream(), 0, NULL, NULL);
-    pa_stream_cork( record->pulseStream(), 0, NULL, NULL);
+        pa_stream_cork( playback->pulseStream(), 0, NULL, NULL);
+        pa_stream_cork( record->pulseStream(), 0, NULL, NULL);
   
-    pa_threaded_mainloop_unlock(m);
+        pa_threaded_mainloop_unlock(m);
+
+	isCorked = false;
+    }
 
 }
 
     void 
 PulseLayer::stopStream (void) 
 {
-     _debug("PulseLayer::Stop stream\n");
-     pa_stream_flush( playback->pulseStream(), NULL, NULL );
-     pa_stream_flush( record->pulseStream(), NULL, NULL );
+
+    if (!isCorked) {
+
+        _debug("PulseLayer::Stop stream\n");
+        pa_stream_flush( playback->pulseStream(), NULL, NULL );
+	pa_stream_flush( record->pulseStream(), NULL, NULL );
     
-    flushMic();
-    flushMain();
-    flushUrgent();
+	flushMic();
+	flushMain();
+	flushUrgent();
 
-    pa_stream_cork( playback->pulseStream(), 1, NULL, NULL);
-    pa_stream_cork( record->pulseStream(), 1, NULL, NULL);
+	pa_stream_cork( playback->pulseStream(), 1, NULL, NULL);
+	pa_stream_cork( record->pulseStream(), 1, NULL, NULL);
 
+	isCorked = true;
+
+    }
+	
 }
 
 
diff --git a/sflphone-common/src/audio/pulselayer.h b/sflphone-common/src/audio/pulselayer.h
index beae9eacc2b5b7400aca9ccd6d25a79b98dcd1c8..698cc9bf6b68b6ff7009b35efcccba3c664d2729 100644
--- a/sflphone-common/src/audio/pulselayer.h
+++ b/sflphone-common/src/audio/pulselayer.h
@@ -189,6 +189,8 @@ class PulseLayer : public AudioLayer {
      */
     AudioStream* record;
 
+    bool isCorked;
+
     int spkrVolume;
     int micVolume;
 
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index e6551144cdc6fd4b65b85582e5d3a5cca64af9d7..28239c5303e4db4c8fa7a5d9c7a5f7e9ec1af6d9 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -143,7 +143,7 @@ ManagerImpl::init()
     }
 
     if (audiolayer == 0)
-      audiolayer->stopStream();
+        audiolayer->stopStream();
 
 
     // Load the history
@@ -326,7 +326,14 @@ ManagerImpl::hangupCall(const CallID& id)
 
     _debug("Stop audio stream\n");
     audiolayer = getAudioDriver();
-    audiolayer->stopStream();
+
+    int nbCalls = getCallList().size();
+
+    _debug("nbCalls %i \n", nbCalls);
+
+    // stop stream 
+    if (!(nbCalls > 1))
+        audiolayer->stopStream();
 
     /* Direct IP to IP call */
     if (getConfigFromCall (id) == Call::IPtoIP) {
@@ -952,11 +959,12 @@ void ManagerImpl::stopTone (bool stopAudio=true)
     hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
     if (!hasToPlayTone)
         return;
-
+    
     if (stopAudio) {
         audiolayer = getAudioDriver();
         if (audiolayer) audiolayer->stopStream();
     }
+    
 
     _toneMutex.enterMutex();
     if (_telephoneTone != 0) {