diff --git a/sflphone-common/src/call.cpp b/sflphone-common/src/call.cpp
index 2705bc9501c9d605ba83eae7323c906def1212d2..6d6523856133f50f063c866bb6e2cd5c86ec1494 100644
--- a/sflphone-common/src/call.cpp
+++ b/sflphone-common/src/call.cpp
@@ -34,7 +34,6 @@
 
 Call::Call (const CallID& id, Call::CallType type)
     : _callMutex()
-    , _audioStarted (false)
     , _localIPAddress ("")
     , _localAudioPort (0)
     , _localExternalAudioPort (0)
@@ -149,21 +148,6 @@ Call::getLocalAudioPort()
     return _localAudioPort;
 }
 
-void
-Call::setAudioStart (bool start)
-{
-    ost::MutexLock m (_callMutex);
-    _audioStarted = start;
-}
-
-bool
-Call::isAudioStarted()
-{
-    ost::MutexLock m (_callMutex);
-    return _audioStarted;
-}
-
-
 bool
 Call::setRecording()
 {
diff --git a/sflphone-common/src/call.h b/sflphone-common/src/call.h
index 0e5749a07ef55f485b8443369293f451ddf68760..846b6b51c3d1cc2eabbaa782fc55649de89d68af 100644
--- a/sflphone-common/src/call.h
+++ b/sflphone-common/src/call.h
@@ -208,20 +208,6 @@ class Call: public Recordable
             return _callConfig;
         }
 
-        /**
-         * Set the audio start boolean (protected by mutex)
-         * @param start true if we start the audio
-         *		    false otherwise
-         */
-        void setAudioStart (bool start);
-
-        /**
-         * Tell if the audio is started (protected by mutex)
-         * @return true if it's already started
-         *	      false otherwise
-         */
-        bool isAudioStarted();
-
         /**
          * Set my IP [not protected]
          * @param ip  The local IP address
@@ -280,8 +266,6 @@ class Call: public Recordable
         /** Protect every attribute that can be changed by two threads */
         ost::Mutex _callMutex;
 
-        bool _audioStarted;
-
         // Informations about call socket / audio
 
         /** My IP address */
@@ -325,8 +309,6 @@ class Call: public Recordable
 
         /** File name for his call, should be peer number */
         std::string _filename;
-
-
 };
 
 #endif
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index f67313a4d453141107b16db3470db032d76b1399..fd734a73561c9366582dc4e8d26fc55ebd276a0c 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -942,7 +942,6 @@ SIPVoIPLink::onhold (const CallID& id) throw (VoipLinkException)
     }
 
     // Stop sound
-    call->setAudioStart (false);
     call->setState (Call::Hold);
 
     try {
@@ -1668,7 +1667,6 @@ SIPVoIPLink::SIPCallClosed (SIPCall *call)
     CallID id = call->getCallId();
 
     if (Manager::instance().isCurrentCall (id)) {
-        call->setAudioStart (false);
         _debug ("UserAgent: Stopping AudioRTP when closing");
         call->getAudioRtp()->stop();
     }
@@ -3518,8 +3516,6 @@ void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status)
     AudioCodecType pl = (AudioCodecType) sessionMedia->getPayloadType();
 
     try {
-        call->setAudioStart (true);
-
         Manager::instance().audioLayerMutexLock();
         Manager::instance().getAudioDriver()->startStream();
         Manager::instance().audioLayerMutexUnlock();