diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index 720e5a9e9a0a573b17b479b3d10486a46bb33b50..bc5d4b0728dcaa1729db3fc0ab4bff92bd75c2d8 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -31,7 +31,8 @@
 #include <samplerate.h>
 
 #include "../global.h"
-/** maximum of byte inside an incoming packet 
+
+/** maximum bytes inside an incoming packet 
  *  8000 sampling/s * 20s/1000 = 160
  */
 #define RTP_20S_8KHZ_MAX 160
diff --git a/src/call.h b/src/call.h
index c12e1bac85a5e417178d835aa07bef9e16e34f4a..7f64b56c173c893051f8419da70f5842768c62ce 100644
--- a/src/call.h
+++ b/src/call.h
@@ -33,9 +33,26 @@ typedef std::string CallID;
  */
 class Call{
 public:
-    enum CallType {Incoming, Outgoing};
-    enum ConnectionState {Disconnected, Trying, Progressing, Ringing, Connected };
-    enum CallState {Inactive, Active, Hold, Busy, Refused, Error};
+  /**
+   * This determines if the call originated from the local user (Outgoing)
+   * or from some remote peer (Incoming).
+   */
+  enum CallType {Incoming, Outgoing};
+  
+  /**
+   * Tell where we're at with the call. The call gets Connected when we know
+   * from the other end what happened with out call. A call can be 'Connected'
+   * even if the call state is Busy, Refused, or Error.
+   *
+   * Audio should be transmitted when ConnectionState = Connected AND
+   * CallState = Active.
+   */
+  enum ConnectionState {Disconnected, Trying, Progressing, Ringing, Connected};
+
+  /**
+   * The Call State.
+   */
+  enum CallState {Inactive, Active, Hold, Busy, Refused, Error};
 
     /**
      * Constructor of a call
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index d6d9106935edc39842af6e3aa031974168889e60..2d604cf8cf1352c8979c44bb6748c69ea10d74d4 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -1436,7 +1436,7 @@ SIPVoIPLink::findSIPCallWithCid(int cid)
 {
   if (cid < 1) {
     _debug("! SIP Error: Not enough information for this event\n");
-    return 0;
+    return NULL;
   }
   ost::MutexLock m(_callMapMutex);
   SIPCall* call = 0;
@@ -1448,7 +1448,7 @@ SIPVoIPLink::findSIPCallWithCid(int cid)
     }
     iter++;
   }
-  return 0;
+  return NULL;
 }
 
 SIPCall* 
@@ -1456,7 +1456,7 @@ SIPVoIPLink::findSIPCallWithCidDid(int cid, int did)
 {
   if (cid < 1 && did < -1) {
     _debug("! SIP Error: Not enough information for this event\n");
-    return 0;
+    return NULL;
   }
   ost::MutexLock m(_callMapMutex);
   SIPCall* call = 0;
@@ -1468,7 +1468,7 @@ SIPVoIPLink::findSIPCallWithCidDid(int cid, int did)
     }
     iter++;
   }
-  return 0;
+  return NULL;
 }
 
 SIPCall*
@@ -1478,7 +1478,7 @@ SIPVoIPLink::getSIPCall(const CallID& id)
   if (call) {
     return dynamic_cast<SIPCall*>(call);
   }
-  return 0;
+  return NULL;
 }
 
 /**