diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index 10f481487dab6c6f2f837b26e9162a51ee9680c2..2f0005241bb689bd35cbf1f9ef0a7fcc065ed91c 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -110,7 +110,6 @@ AudioRtp::closeRtpSession ()
     // This will make RTP threads finish.
     _debug ("AudioRtp::Stopping rtp session\n");
 
-
     try {
         delete _RTXThread;
         _RTXThread = 0;
@@ -119,9 +118,6 @@ AudioRtp::closeRtpSession ()
         throw;
     }
 
-    // AudioLayer* audiolayer = Manager::instance().getAudioDriver();
-    // audiolayer->stopStream();
-
     _debug ("AudioRtp::Audio rtp stopped\n");
 
     return true;
diff --git a/sflphone-common/src/audio/pulselayer.cpp b/sflphone-common/src/audio/pulselayer.cpp
index ed8b4317774b5b4c1c512ac2ca66434b2278c6ac..39d3ba887d32ab28d818043428f879ea55078013 100644
--- a/sflphone-common/src/audio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulselayer.cpp
@@ -21,8 +21,6 @@
 
 int framesPerBuffer = 2048;
 
-int PulseLayer::streamState;
-
 static  void audioCallback (pa_stream* s, size_t bytes, void* userdata)
 {
     assert (s && bytes);
@@ -39,7 +37,6 @@ PulseLayer::PulseLayer (ManagerImpl* manager)
         , playback()
         , record()
 {
-    PulseLayer::streamState = 0;
     _debug ("PulseLayer::Pulse audio constructor: Create context\n");
 
 }
diff --git a/sflphone-common/src/audio/pulselayer.h b/sflphone-common/src/audio/pulselayer.h
index 698cc9bf6b68b6ff7009b35efcccba3c664d2729..8c1fa083842f9990b73f141730cf849146655280 100644
--- a/sflphone-common/src/audio/pulselayer.h
+++ b/sflphone-common/src/audio/pulselayer.h
@@ -197,7 +197,6 @@ class PulseLayer : public AudioLayer {
     // private:
 
 public: 
-    static int streamState;
 
     friend class AudioLayerTest;
 };
diff --git a/sflphone-common/src/sipvoiplink.cpp b/sflphone-common/src/sipvoiplink.cpp
index aa1e2f9a410b4a267212a636d4249d40bb7d0a79..af41b6d268c760870fcca500204e488df9331432 100644
--- a/sflphone-common/src/sipvoiplink.cpp
+++ b/sflphone-common/src/sipvoiplink.cpp
@@ -601,8 +601,6 @@ SIPVoIPLink::answer (const CallID& id)
         call->setConnectionState (Call::Connected);
         call->setState (Call::Active);
 
-        ;
-
         return true;
     } else {
         // Create and send a 488/Not acceptable here
@@ -617,6 +615,7 @@ SIPVoIPLink::answer (const CallID& id)
         _debug ("SIPVoIPLink::answer: fail terminate call %s \n",call->getCallId().c_str());
         terminateOneCall (call->getCallId());
         removeCall (call->getCallId());
+        _audiortp->closeRtpSession ();
         return false;
     }
 }
@@ -1195,39 +1194,25 @@ SIPVoIPLink::SIPCheckUrl (const std::string& url UNUSED)
 void
 SIPVoIPLink::SIPCallServerFailure (SIPCall *call)
 {
-    //if (!event->response) { return; }
-    //switch(event->response->status_code) {
-    //case SIP_SERVICE_UNAVAILABLE: // 500
-    //case SIP_BUSY_EVRYWHERE:     // 600
-    //case SIP_DECLINE:             // 603
-    //SIPCall* call = findSIPCallWithCid(event->cid);
     if (call != 0) {
         _debug ("Server error!\n");
         CallID id = call->getCallId();
         Manager::instance().callFailure (id);
         terminateOneCall (id);
         removeCall (id);
+        _audiortp->closeRtpSession();
     }
-
-    //break;
-    //}
 }
 
 void
 SIPVoIPLink::SIPCallClosed (SIPCall *call)
 {
-
-
-    // it was without did before
-    //SIPCall* call = findSIPCallWithCid(event->cid);
     if (!call) {
         return;
     }
 
     CallID id = call->getCallId();
 
-    //call->setDid(event->did);
-
     if (Manager::instance().isCurrentCall (id)) {
         call->setAudioStart (false);
         _debug ("* SIP Info: Stopping AudioRTP when closing\n");
@@ -1245,9 +1230,6 @@ SIPVoIPLink::SIPCallClosed (SIPCall *call)
 void
 SIPVoIPLink::SIPCallReleased (SIPCall *call)
 {
-    // do cleanup if exists
-    // only cid because did is always 0 in these case..
-    //SIPCall* call = findSIPCallWithCid(event->cid);
     if (!call) {
         return;
     }
@@ -2050,7 +2032,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
                     break;
 
                 default:
-                    _debug ("sipvoiplink.cpp - line 1635 : Unhandled call state. This is probably a bug.\n");
+                    _debug ("sipvoiplink.cpp - line %d : Unhandled call state. This is probably a bug.\n", __LINE__);
                     break;
             }
         }