Merge branch '1.0-local' into trunk-local

* 1.0-local:
  Fixes Issue #696 : Wrong VIA in INVITE when hold/unhold calls. Fixes Issue #694 : Disable DirectSound. Contributes to Issue #637 by removing use of raw types. Removes CallParticipantChangeListener from CallSessionImpl since it doesn't seem to serve any purpose.

src/net/java/sip/communicator/impl/media/CallSessionImpl.java

@@ -73,14 +73,12 @@
  */
 public class CallSessionImpl
     extends PropertyChangeNotifier
-    implements CallSession
-               , CallParticipantListener
-               , CallChangeListener
-               , ReceiveStreamListener
-               , SendStreamListener
-               , SessionListener
-               , ControllerListener
-//             , SecureEventListener
+    implements CallSession,
+               CallChangeListener,
+               ReceiveStreamListener,
+               SendStreamListener,
+               SessionListener,
+               ControllerListener
 {
     private static final Logger logger
         = Logger.getLogger(CallSessionImpl.class);
@@ -1207,18 +1205,24 @@ private String processSdpStr(CallParticipant participant,
         }
         else
         {
-            //create an sdp answer.
+            //create the SDP answer.
             CallParticipantState participantState = participant.getState();
 
             if (CallParticipantState.CONNECTED.equals(participantState)
                     || CallParticipantState.isOnHold(participantState))
+            {
+                //if the call is already connected then this is a
+                //reinitialization (e.g. placing the call on/off hold)
                 sdpAnswer =
                     createSessionDescriptionForHold(
                         sdpStr,
                         (getSdpOfferMediaFlags(sdpStr)
                             & CallSession.ON_HOLD_REMOTELY) != 0);
+            }
             else
+            {
                 sdpAnswer = createSessionDescription(sdp, null);
+            }
         }
 
         //extract media descriptions
@@ -1237,7 +1241,6 @@ private String processSdpStr(CallParticipant participant,
         //add the RTP targets
         initStreamTargets(sdp.getConnection(), mediaDescriptions);
 
-
         /*
          * The media descriptions of the offer may list encodings which are not
          * in our answer i.e. we may end up streaming in an encoding disabled by
@@ -1609,7 +1612,7 @@ private SessionDescription createSessionDescription(
 
             /*
              * Only allocate ports if this is a call establishing event. The
-             * opposite could happen for example, when issuing a Request.INVITE
+             * opposite could happen for example, when issuing a reINVITE
              * that would put a CallParticipant on hold.
              */
             boolean allocateMediaPorts = false;
@@ -2596,19 +2599,6 @@ private void disposePlayer(Player player)
                 VideoEvent.REMOTE);
     }
 
-    /**
-     * Indicates that a change has occurred in the status of the source
-     * CallParticipant.
-     *
-     * @param evt The <tt>CallParticipantChangeEvent</tt> instance containing
-     * the source event as well as its previous and its new status.
-     */
-    public void participantStateChanged(CallParticipantChangeEvent evt)
-    {
-        /** @todo implement participantStateChanged() */
-        /** @todo remove target for participant. */
-    }
-
     /**
      * Indicates that a new call participant has joined the source call.
      * @param evt the <tt>CallParticipantEvent</tt> containing the source call
@@ -2616,8 +2606,6 @@ public void participantStateChanged(CallParticipantChangeEvent evt)
      */
     public synchronized void callParticipantAdded(CallParticipantEvent evt)
     {
-        CallParticipant sourceParticipant = evt.getSourceCallParticipant();
-        sourceParticipant.addCallParticipantListener(this);
     }
 
     /**
@@ -2630,45 +2618,6 @@ public void callParticipantRemoved(CallParticipantEvent evt)
 
     }
 
-    //-------- dummy implementations of listener methods that we don't need
-    /**
-     * Ignore - we're not concerned by this event inside a call session.
-     *
-     * @param evt ignore.
-     */
-    public void participantImageChanged(CallParticipantChangeEvent evt)
-    {
-    }
-
-    /**
-     * Ignore - we're not concerned by this event inside a call session.
-     *
-     * @param evt ignore.
-     */
-    public void participantDisplayNameChanged(CallParticipantChangeEvent evt)
-    {
-    }
-
-    /**
-     * Ignore - we're not concerned by this event inside a call session.
-     *
-     * @param evt ignore.
-     */
-    public void participantTransportAddressChanged(
-                                CallParticipantChangeEvent evt)
-    {
-        /** @todo i am not sure we should be ignoring this one ... */
-    }
-
-    /**
-     * Ignore - we're not concerned by this event inside a call session.
-     *
-     * @param evt ignore.
-     */
-    public void participantAddressChanged(CallParticipantChangeEvent evt)
-    {
-    }
-
     //implementation of jmf listener methods
     /**
      * Method called back in the SessionListener to notify

src/net/java/sip/communicator/impl/media/device/JmfDeviceDetector.java

@@ -114,8 +114,11 @@ private void initialize()
             }
         }
 
-        detectDirectAudio();
-        detectS8DirectAudio();
+        // Issues #693 and #524:
+        // Disable DirectSound since it fails on multiple installations
+        //
+        // detectDirectAudio();
+        // detectS8DirectAudio();
         detectCaptureDevices();
     }
 
@@ -127,15 +130,19 @@ private void detectCaptureDevices()
     {
         logger.info("Looking for Audio capturer");
 
-        //First check if DirectSound capture is available
-        try
-        {
-            new DirectSoundAuto();
-        }
-        catch (Throwable exc)
-        {
-            logger.debug("No direct sound detected: " + exc.getMessage());
-        }
+        // Issues #693 and #524:
+        // Disable DirectSound since it fails on multiple installations
+        //
+        // //First check if DirectSound capture is available
+        // try
+        // {
+        //
+        //    new DirectSoundAuto();
+        // }
+        // catch (Throwable exc)
+        // {
+        //     logger.debug("No direct sound detected: " + exc.getMessage());
+        // }
 
         // check if JavaSound capture is available
         try

src/net/java/sip/communicator/impl/protocol/sip/OperationSetBasicTelephonySipImpl.java

@@ -1555,10 +1555,12 @@ private void processInvite(SipProvider sourceProvider,
 
             if (replacesHeader == null)
             {
+                //this is not a transfered call so start ringing
                 statusCode = Response.RINGING;
             }
             else
             {
+                //this is a transfered call
                 List<CallParticipantSipImpl> callParticipantsToReplace =
                     activeCallsRepository.findCallParticipants(
                         replacesHeader.getCallId(), replacesHeader.getToTag(),
@@ -1582,6 +1584,7 @@ private void processInvite(SipProvider sourceProvider,
         }
         else
         {
+            //this is a reINVITE - so we'll OK it without ringing
             statusCode = Response.OK;
         }
 
@@ -1662,9 +1665,8 @@ private void processInvite(SipProvider sourceProvider,
         logger.debug("Invite seems ok, we'll say " + statusCodeString + ".");
         try
         {
-            response =
-                protocolProvider.getMessageFactory().createResponse(statusCode,
-                    invite);
+            response =  protocolProvider.getMessageFactory()
+                .createResponse(statusCode, invite);
             protocolProvider.attachToTag(response, dialog);
             response.setHeader(protocolProvider.getSipCommUserAgentHeader());
 
@@ -2568,6 +2570,18 @@ private Request createRequest(Dialog dialog, String method)
                 + " request.", OperationFailedException.INTERNAL_ERROR, ex);
         }
 
+        //override the via header as jain-sip is generating one from the
+        //listening point which is 0.0.0.0 or ::0
+        ArrayList<ViaHeader> viaHeaders
+            = protocolProvider.getLocalViaHeaders(dialog.getRemoteParty());
+        request.setHeader(viaHeaders.get(0));
+
+        // User Agent
+        UserAgentHeader userAgentHeader =
+            protocolProvider.getSipCommUserAgentHeader();
+        if (userAgentHeader != null)
+            request.addHeader(userAgentHeader);
+
         /*
          * The authorization-related headers are the responsibility of the
          * application (according to the Javadoc of JAIN-SIP).

test/net/java/sip/communicator/slick/protocol/sip/TestAccountInstallation.java

@@ -78,9 +78,9 @@ public void testInstallAccount()
 
         //Prepare the properties of the first sip account.
 
-        Hashtable sipAccount1Properties = getAccountProperties(
+        Hashtable<String,String> sipAccount1Properties = getAccountProperties(
             SipProtocolProviderServiceLick.ACCOUNT_1_PREFIX);
-        Hashtable sipAccount2Properties = getAccountProperties(
+        Hashtable<String, String> sipAccount2Properties = getAccountProperties(
             SipProtocolProviderServiceLick.ACCOUNT_2_PREFIX);
 
         //try to install an account with a null account id
@@ -166,9 +166,9 @@ public void testInstallAccount()
      * @return a Hashtable that can be used when creating the account in a
      * protocol provider factory.
      */
-    private Hashtable getAccountProperties(String accountPrefix)
+    private Hashtable<String, String> getAccountProperties(String accountPrefix)
     {
-        Hashtable table = new Hashtable();
+        Hashtable<String, String> table = new Hashtable<String, String>();
 
         String userID = System.getProperty(
             accountPrefix + ProtocolProviderFactory.USER_ID, null);