Unhide SIP API.

Change-Id: I09468e3149a242a3b1e085ad220eb74f84ac6c68

7 files changed, 208 insertions, 144 deletions

diff --git a/voip/java/android/net/sip/SipAudioCall.java b/voip/java/android/net/sip/SipAudioCall.java
index c23da20..f55bade 100644
--- a/voip/java/android/net/sip/SipAudioCall.java
+++ b/voip/java/android/net/sip/SipAudioCall.java
@@ -43,9 +43,11 @@ import java.util.List;
 import java.util.Map;
 
 /**
- * Class that handles an audio call over SIP.
+ * Class that handles an Internet audio call over SIP. {@link SipManager}
+ * facilitates instantiating a {@code SipAudioCall} object for making/receiving
+ * calls. See {@link SipManager#makeAudioCall} and
+ * {@link SipManager#takeAudioCall}.
  */
-/** @hide */
 public class SipAudioCall {
     private static final String TAG = SipAudioCall.class.getSimpleName();
     private static final boolean RELEASE_SOCKET = true;
@@ -56,7 +58,7 @@ public class SipAudioCall {
     public static class Listener {
         /**
          * Called when the call object is ready to make another call.
-         * The default implementation calls {@link #onChange}.
+         * The default implementation calls {@link #onChanged}.
          *
          * @param call the call object that is ready to make another call
          */
@@ -66,7 +68,7 @@ public class SipAudioCall {
 
         /**
          * Called when a request is sent out to initiate a new call.
-         * The default implementation calls {@link #onChange}.
+         * The default implementation calls {@link #onChanged}.
          *
          * @param call the call object that carries out the audio call
          */
@@ -76,7 +78,7 @@ public class SipAudioCall {
 
         /**
          * Called when a new call comes in.
-         * The default implementation calls {@link #onChange}.
+         * The default implementation calls {@link #onChanged}.
          *
          * @param call the call object that carries out the audio call
          * @param caller the SIP profile of the caller
@@ -87,7 +89,7 @@ public class SipAudioCall {
 
         /**
          * Called when a RINGING response is received for the INVITE request
-         * sent. The default implementation calls {@link #onChange}.
+         * sent. The default implementation calls {@link #onChanged}.
          *
          * @param call the call object that carries out the audio call
          */
@@ -97,7 +99,7 @@ public class SipAudioCall {
 
         /**
          * Called when the session is established.
-         * The default implementation calls {@link #onChange}.
+         * The default implementation calls {@link #onChanged}.
          *
          * @param call the call object that carries out the audio call
          */
@@ -107,7 +109,7 @@ public class SipAudioCall {
 
         /**
          * Called when the session is terminated.
-         * The default implementation calls {@link #onChange}.
+         * The default implementation calls {@link #onChanged}.
          *
          * @param call the call object that carries out the audio call
          */
@@ -117,7 +119,7 @@ public class SipAudioCall {
 
         /**
          * Called when the peer is busy during session initialization.
-         * The default implementation calls {@link #onChange}.
+         * The default implementation calls {@link #onChanged}.
          *
          * @param call the call object that carries out the audio call
          */
@@ -127,7 +129,7 @@ public class SipAudioCall {
 
         /**
          * Called when the call is on hold.
-         * The default implementation calls {@link #onChange}.
+         * The default implementation calls {@link #onChanged}.
          *
          * @param call the call object that carries out the audio call
          */
@@ -257,8 +259,10 @@ public class SipAudioCall {
      *
      * @return true if the call is established
      */
-    public synchronized boolean isInCall() {
-        return mInCall;
+    public boolean isInCall() {
+        synchronized (this) {
+            return mInCall;
+        }
     }
 
     /**
@@ -266,8 +270,10 @@ public class SipAudioCall {
      *
      * @return true if the call is on hold
      */
-    public synchronized boolean isOnHold() {
-        return mHold;
+    public boolean isOnHold() {
+        synchronized (this) {
+            return mHold;
+        }
     }
 
     /**
@@ -299,8 +305,10 @@ public class SipAudioCall {
      *
      * @return the local SIP profile
      */
-    public synchronized SipProfile getLocalProfile() {
-        return mLocalProfile;
+    public SipProfile getLocalProfile() {
+        synchronized (this) {
+            return mLocalProfile;
+        }
     }
 
     /**
@@ -308,8 +316,10 @@ public class SipAudioCall {
      *
      * @return the peer's SIP profile
      */
-    public synchronized SipProfile getPeerProfile() {
-        return (mSipSession == null) ? null : mSipSession.getPeerProfile();
+    public SipProfile getPeerProfile() {
+        synchronized (this) {
+            return (mSipSession == null) ? null : mSipSession.getPeerProfile();
+        }
     }
 
     /**
@@ -318,9 +328,11 @@ public class SipAudioCall {
      *
      * @return the session state
      */
-    public synchronized int getState() {
-        if (mSipSession == null) return SipSession.State.READY_TO_CALL;
-        return mSipSession.getState();
+    public int getState() {
+        synchronized (this) {
+            if (mSipSession == null) return SipSession.State.READY_TO_CALL;
+            return mSipSession.getState();
+        }
     }
 
 
@@ -330,8 +342,10 @@ public class SipAudioCall {
      * @return the session object that carries this call
      * @hide
      */
-    public synchronized SipSession getSipSession() {
-        return mSipSession;
+    public SipSession getSipSession() {
+        synchronized (this) {
+            return mSipSession;
+        }
     }
 
     private SipSession.Listener createListener() {
@@ -364,22 +378,25 @@ public class SipAudioCall {
             }
 
             @Override
-            public synchronized void onRinging(SipSession session,
+            public void onRinging(SipSession session,
                     SipProfile peerProfile, String sessionDescription) {
-                if ((mSipSession == null) || !mInCall
-                        || !session.getCallId().equals(mSipSession.getCallId())) {
-                    // should not happen
-                    session.endCall();
-                    return;
-                }
+                synchronized (SipAudioCall.this) {
+                    if ((mSipSession == null) || !mInCall
+                            || !session.getCallId().equals(
+                                    mSipSession.getCallId())) {
+                        // should not happen
+                        session.endCall();
+                        return;
+                    }
 
-                // session changing request
-                try {
-                    String answer = createAnswer(sessionDescription).encode();
-                    mSipSession.answerCall(answer, SESSION_TIMEOUT);
-                } catch (Throwable e) {
-                    Log.e(TAG, "onRinging()", e);
-                    session.endCall();
+                    // session changing request
+                    try {
+                        String answer = createAnswer(sessionDescription).encode();
+                        mSipSession.answerCall(answer, SESSION_TIMEOUT);
+                    } catch (Throwable e) {
+                        Log.e(TAG, "onRinging()", e);
+                        session.endCall();
+                    }
                 }
             }
 
@@ -508,18 +525,22 @@ public class SipAudioCall {
      * @throws SipException if the SIP service fails to attach this object to
      *        the session
      */
-    public synchronized void attachCall(SipSession session,
-            String sessionDescription) throws SipException {
-        mSipSession = session;
-        mPeerSd = sessionDescription;
-        Log.v(TAG, "attachCall()" + mPeerSd);
-        try {
-            session.setListener(createListener());
+    public void attachCall(SipSession session, String sessionDescription)
+            throws SipException {
+        synchronized (this) {
+            mSipSession = session;
+            mPeerSd = sessionDescription;
+            Log.v(TAG, "attachCall()" + mPeerSd);
+            try {
+                session.setListener(createListener());
 
-            if (getState() == SipSession.State.INCOMING_CALL) startRinging();
-        } catch (Throwable e) {
-            Log.e(TAG, "attachCall()", e);
-            throwSipException(e);
+                if (getState() == SipSession.State.INCOMING_CALL) {
+                    startRinging();
+                }
+            } catch (Throwable e) {
+                Log.e(TAG, "attachCall()", e);
+                throwSipException(e);
+            }
         }
     }
 
@@ -529,7 +550,7 @@ public class SipAudioCall {
      * and {@code Listener.onError(SipAudioCall, SipErrorCode.TIME_OUT, String)}
      * will be called.
      *
-     * @param callee the SIP profile to make the call to
+     * @param peerProfile the SIP profile to make the call to
      * @param sipSession the {@link SipSession} for carrying out the call
      * @param timeout the timeout value in seconds. Default value (defined by
      *        SIP protocol) is used if {@code timeout} is zero or negative.
@@ -537,15 +558,19 @@ public class SipAudioCall {
      * @throws SipException if the SIP service fails to create a session for the
      *        call
      */
-    public synchronized void makeCall(SipProfile peerProfile,
-            SipSession sipSession, int timeout) throws SipException {
-        mSipSession = sipSession;
-        try {
-            mAudioStream = new AudioStream(InetAddress.getByName(getLocalIp()));
-            sipSession.setListener(createListener());
-            sipSession.makeCall(peerProfile, createOffer().encode(), timeout);
-        } catch (IOException e) {
-            throw new SipException("makeCall()", e);
+    public void makeCall(SipProfile peerProfile, SipSession sipSession,
+            int timeout) throws SipException {
+        synchronized (this) {
+            mSipSession = sipSession;
+            try {
+                mAudioStream = new AudioStream(InetAddress.getByName(
+                        getLocalIp()));
+                sipSession.setListener(createListener());
+                sipSession.makeCall(peerProfile, createOffer().encode(),
+                        timeout);
+            } catch (IOException e) {
+                throw new SipException("makeCall()", e);
+            }
         }
     }
 
@@ -553,13 +578,15 @@ public class SipAudioCall {
      * Ends a call.
      * @throws SipException if the SIP service fails to end the call
      */
-    public synchronized void endCall() throws SipException {
-        stopRinging();
-        stopCall(RELEASE_SOCKET);
-        mInCall = false;
+    public void endCall() throws SipException {
+        synchronized (this) {
+            stopRinging();
+            stopCall(RELEASE_SOCKET);
+            mInCall = false;
 
-        // perform the above local ops first and then network op
-        if (mSipSession != null) mSipSession.endCall();
+            // perform the above local ops first and then network op
+            if (mSipSession != null) mSipSession.endCall();
+        }
     }
 
     /**
@@ -574,13 +601,15 @@ public class SipAudioCall {
      * @see Listener.onError
      * @throws SipException if the SIP service fails to hold the call
      */
-    public synchronized void holdCall(int timeout) throws SipException {
+    public void holdCall(int timeout) throws SipException {
+        synchronized (this) {
         if (mHold) return;
-        mSipSession.changeCall(createHoldOffer().encode(), timeout);
-        mHold = true;
+            mSipSession.changeCall(createHoldOffer().encode(), timeout);
+            mHold = true;
 
-        AudioGroup audioGroup = getAudioGroup();
-        if (audioGroup != null) audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
+            AudioGroup audioGroup = getAudioGroup();
+            if (audioGroup != null) audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
+        }
     }
 
     /**
@@ -594,13 +623,16 @@ public class SipAudioCall {
      * @see Listener.onError
      * @throws SipException if the SIP service fails to answer the call
      */
-    public synchronized void answerCall(int timeout) throws SipException {
-        stopRinging();
-        try {
-            mAudioStream = new AudioStream(InetAddress.getByName(getLocalIp()));
-            mSipSession.answerCall(createAnswer(mPeerSd).encode(), timeout);
-        } catch (IOException e) {
-            throw new SipException("answerCall()", e);
+    public void answerCall(int timeout) throws SipException {
+        synchronized (this) {
+            stopRinging();
+            try {
+                mAudioStream = new AudioStream(InetAddress.getByName(
+                        getLocalIp()));
+                mSipSession.answerCall(createAnswer(mPeerSd).encode(), timeout);
+            } catch (IOException e) {
+                throw new SipException("answerCall()", e);
+            }
         }
     }
 
@@ -616,12 +648,14 @@ public class SipAudioCall {
      * @see Listener.onError
      * @throws SipException if the SIP service fails to unhold the call
      */
-    public synchronized void continueCall(int timeout) throws SipException {
-        if (!mHold) return;
-        mSipSession.changeCall(createContinueOffer().encode(), timeout);
-        mHold = false;
-        AudioGroup audioGroup = getAudioGroup();
-        if (audioGroup != null) audioGroup.setMode(AudioGroup.MODE_NORMAL);
+    public void continueCall(int timeout) throws SipException {
+        synchronized (this) {
+            if (!mHold) return;
+            mSipSession.changeCall(createContinueOffer().encode(), timeout);
+            mHold = false;
+            AudioGroup audioGroup = getAudioGroup();
+            if (audioGroup != null) audioGroup.setMode(AudioGroup.MODE_NORMAL);
+        }
     }
 
     private SimpleSessionDescription createOffer() {
@@ -739,12 +773,15 @@ public class SipAudioCall {
     }
 
     /** Toggles mute. */
-    public synchronized void toggleMute() {
-        AudioGroup audioGroup = getAudioGroup();
-        if (audioGroup != null) {
-            audioGroup.setMode(
-                    mMuted ? AudioGroup.MODE_NORMAL : AudioGroup.MODE_MUTED);
-            mMuted = !mMuted;
+    public void toggleMute() {
+        synchronized (this) {
+            AudioGroup audioGroup = getAudioGroup();
+            if (audioGroup != null) {
+                audioGroup.setMode(mMuted
+                        ? AudioGroup.MODE_NORMAL
+                        : AudioGroup.MODE_MUTED);
+                mMuted = !mMuted;
+            }
         }
     }
 
@@ -753,14 +790,18 @@ public class SipAudioCall {
      *
      * @return true if the call is muted
      */
-    public synchronized boolean isMuted() {
-        return mMuted;
+    public boolean isMuted() {
+        synchronized (this) {
+            return mMuted;
+        }
     }
 
     /** Puts the device to speaker mode. */
-    public synchronized void setSpeakerMode(boolean speakerMode) {
-        ((AudioManager) mContext.getSystemService(Context.AUDIO_SERVICE))
-                .setSpeakerphoneOn(speakerMode);
+    public void setSpeakerMode(boolean speakerMode) {
+        synchronized (this) {
+            ((AudioManager) mContext.getSystemService(Context.AUDIO_SERVICE))
+                    .setSpeakerphoneOn(speakerMode);
+        }
     }
 
     /**
@@ -785,14 +826,16 @@ public class SipAudioCall {
      *        inputs.
      * @param result the result message to send when done
      */
-    public synchronized void sendDtmf(int code, Message result) {
-        AudioGroup audioGroup = getAudioGroup();
-        if ((audioGroup != null) && (mSipSession != null)
-                && (SipSession.State.IN_CALL == getState())) {
-            Log.v(TAG, "send DTMF: " + code);
-            audioGroup.sendDtmf(code);
+    public void sendDtmf(int code, Message result) {
+        synchronized (this) {
+            AudioGroup audioGroup = getAudioGroup();
+            if ((audioGroup != null) && (mSipSession != null)
+                    && (SipSession.State.IN_CALL == getState())) {
+                Log.v(TAG, "send DTMF: " + code);
+                audioGroup.sendDtmf(code);
+            }
+            if (result != null) result.sendToTarget();
         }
-        if (result != null) result.sendToTarget();
     }
 
     /**
@@ -806,8 +849,10 @@ public class SipAudioCall {
      *      yet been set up
      * @hide
      */
-    public synchronized AudioStream getAudioStream() {
-        return mAudioStream;
+    public AudioStream getAudioStream() {
+        synchronized (this) {
+            return mAudioStream;
+        }
     }
 
     /**
@@ -824,9 +869,11 @@ public class SipAudioCall {
      * @see #getAudioStream
      * @hide
      */
-    public synchronized AudioGroup getAudioGroup() {
-        if (mAudioGroup != null) return mAudioGroup;
-        return ((mAudioStream == null) ? null : mAudioStream.getGroup());
+    public AudioGroup getAudioGroup() {
+        synchronized (this) {
+            if (mAudioGroup != null) return mAudioGroup;
+            return ((mAudioStream == null) ? null : mAudioStream.getGroup());
+        }
     }
 
     /**
@@ -837,11 +884,13 @@ public class SipAudioCall {
      * @see #getAudioStream
      * @hide
      */
-    public synchronized void setAudioGroup(AudioGroup group) {
-        if ((mAudioStream != null) && (mAudioStream.getGroup() != null)) {
-            mAudioStream.join(group);
+    public void setAudioGroup(AudioGroup group) {
+        synchronized (this) {
+            if ((mAudioStream != null) && (mAudioStream.getGroup() != null)) {
+                mAudioStream.join(group);
+            }
+            mAudioGroup = group;
         }
-        mAudioGroup = group;
     }
 
     /**
@@ -981,8 +1030,10 @@ public class SipAudioCall {
      *
      * @param enabled true to enable; false to disable
      */
-    public synchronized void setRingbackToneEnabled(boolean enabled) {
-        mRingbackToneEnabled = enabled;
+    public void setRingbackToneEnabled(boolean enabled) {
+        synchronized (this) {
+            mRingbackToneEnabled = enabled;
+        }
     }
 
     /**
@@ -990,8 +1041,10 @@ public class SipAudioCall {
      *
      * @param enabled true to enable; false to disable
      */
-    public synchronized void setRingtoneEnabled(boolean enabled) {
-        mRingtoneEnabled = enabled;
+    public void setRingtoneEnabled(boolean enabled) {
+        synchronized (this) {
+            mRingtoneEnabled = enabled;
+        }
     }
 
     private void startRingbackTone() {
diff --git a/voip/java/android/net/sip/SipErrorCode.java b/voip/java/android/net/sip/SipErrorCode.java
index a55ab25..6aee5f1 100644
--- a/voip/java/android/net/sip/SipErrorCode.java
+++ b/voip/java/android/net/sip/SipErrorCode.java
@@ -19,10 +19,9 @@ package android.net.sip;
 /**
  * Defines error code returned in
  * {@link SipRegistrationListener#onRegistrationFailed},
- * {@link ISipSessionListener#onError},
- * {@link ISipSessionListener#onCallChangeFailed} and
- * {@link ISipSessionListener#onRegistrationFailed}.
- * @hide
+ * {@link SipSession.Listener#onError},
+ * {@link SipSession.Listener#onCallChangeFailed} and
+ * {@link SipSession.Listener#onRegistrationFailed}.
  */
 public class SipErrorCode {
     /** Not an error. */
diff --git a/voip/java/android/net/sip/SipException.java b/voip/java/android/net/sip/SipException.java
index f0d846b..225b94f 100644
--- a/voip/java/android/net/sip/SipException.java
+++ b/voip/java/android/net/sip/SipException.java
@@ -18,7 +18,6 @@ package android.net.sip;
 
 /**
  * General SIP-related exception class.
- * @hide
  */
 public class SipException extends Exception {
     public SipException() {
diff --git a/voip/java/android/net/sip/SipManager.java b/voip/java/android/net/sip/SipManager.java
index 8c32aa0..ee0e3cd 100644
--- a/voip/java/android/net/sip/SipManager.java
+++ b/voip/java/android/net/sip/SipManager.java
@@ -48,7 +48,8 @@ import java.text.ParseException;
  * <li>process SIP events directly with a {@link SipSession} created by
  *      {@link #createSipSession}.</li>
  * </ul>
- * @hide
+ * {@code SipManager} can only be instantiated if SIP API is supported by the
+ * device. (See {@link #isApiSupported}).
  */
 public class SipManager {
     /**
@@ -58,10 +59,17 @@ public class SipManager {
      */
     public static final int INCOMING_CALL_RESULT_CODE = 101;
 
-    /** Part of the incoming call intent. */
+    /**
+     * Key to retrieve the call ID from an incoming call intent.
+     * @see #open(SipProfile, PendingIntent, SipRegistrationListener)
+     */
     public static final String EXTRA_CALL_ID = "android:sipCallID";
 
-    /** Part of the incoming call intent. */
+    /**
+     * Key to retrieve the offered session description from an incoming call
+     * intent.
+     * @see #open(SipProfile, PendingIntent, SipRegistrationListener)
+     */
     public static final String EXTRA_OFFER_SD = "android:sipOfferSD";
 
     /**
@@ -178,7 +186,11 @@ public class SipManager {
      * make subsequent calls through {@link #makeAudioCall}. If the
      * auto-registration option is enabled in the profile, the SIP service
      * will register the profile to the corresponding SIP provider periodically
-     * in order to receive calls from the provider.
+     * in order to receive calls from the provider. When the SIP service
+     * receives a new call, it will send out an intent with the provided action
+     * string. The intent contains a call ID extra and an offer session
+     * description string extra. Use {@link #getCallId} and
+     * {@link #getOfferSessionDescription} to retrieve those extras.
      *
      * @param localProfile the SIP profile to receive incoming calls for
      * @param incomingCallPendingIntent When an incoming call is received, the
@@ -194,6 +206,9 @@ public class SipManager {
      * @throws NullPointerException if {@code incomingCallPendingIntent} is null
      * @throws SipException if the profile contains incorrect settings or
      *      calling the SIP service results in an error
+     * @see #isIncomingCallIntent
+     * @see #getCallId
+     * @see #getOfferSessionDescription
      */
     public void open(SipProfile localProfile,
             PendingIntent incomingCallPendingIntent,
@@ -291,7 +306,8 @@ public class SipManager {
      * @param peerProfile the SIP profile to make the call to
      * @param listener to listen to the call events from {@link SipAudioCall};
      *      can be null
-     * @param timeout the timeout value in seconds
+     * @param timeout the timeout value in seconds. Default value (defined by
+     *        SIP protocol) is used if {@code timeout} is zero or negative.
      * @return a {@link SipAudioCall} object
      * @throws SipException if calling the SIP service results in an error
      * @see SipAudioCall.Listener.onError
@@ -321,7 +337,8 @@ public class SipManager {
      * @param peerProfileUri URI of the SIP profile to make the call to
      * @param listener to listen to the call events from {@link SipAudioCall};
      *      can be null
-     * @param timeout the timeout value in seconds
+     * @param timeout the timeout value in seconds. Default value (defined by
+     *        SIP protocol) is used if {@code timeout} is zero or negative.
      * @return a {@link SipAudioCall} object
      * @throws SipException if calling the SIP service results in an error
      * @see SipAudioCall.Listener.onError
@@ -489,7 +506,7 @@ public class SipManager {
     }
 
     /**
-     * Gets the {@link ISipSession} that handles the incoming call. For audio
+     * Gets the {@link SipSession} that handles the incoming call. For audio
      * calls, consider to use {@link SipAudioCall} to handle the incoming call.
      * See {@link #takeAudioCall}. Note that the method may be called only once
      * for the same intent. For subsequent calls on the same intent, the method
@@ -498,11 +515,12 @@ public class SipManager {
      * @param incomingCallIntent the incoming call broadcast intent
      * @return the session object that handles the incoming call
      */
-    public ISipSession getSessionFor(Intent incomingCallIntent)
+    public SipSession getSessionFor(Intent incomingCallIntent)
             throws SipException {
         try {
             String callId = getCallId(incomingCallIntent);
-            return mSipService.getPendingSession(callId);
+            ISipSession s = mSipService.getPendingSession(callId);
+            return new SipSession(s);
         } catch (RemoteException e) {
             throw new SipException("getSessionFor()", e);
         }
@@ -514,8 +532,8 @@ public class SipManager {
     }
 
     /**
-     * Creates a {@link ISipSession} with the specified profile. Use other
-     * methods, if applicable, instead of interacting with {@link ISipSession}
+     * Creates a {@link SipSession} with the specified profile. Use other
+     * methods, if applicable, instead of interacting with {@link SipSession}
      * directly.
      *
      * @param localProfile the SIP profile the session is associated with
diff --git a/voip/java/android/net/sip/SipProfile.java b/voip/java/android/net/sip/SipProfile.java
index 6d5cb3c..dddb07d 100644
--- a/voip/java/android/net/sip/SipProfile.java
+++ b/voip/java/android/net/sip/SipProfile.java
@@ -33,7 +33,6 @@ import javax.sip.address.URI;
 
 /**
  * Class containing a SIP account, domain and server information.
- * @hide
  */
 public class SipProfile implements Parcelable, Serializable, Cloneable {
     private static final long serialVersionUID = 1L;
diff --git a/voip/java/android/net/sip/SipRegistrationListener.java b/voip/java/android/net/sip/SipRegistrationListener.java
index 37c9ce2..e1f35ad 100644
--- a/voip/java/android/net/sip/SipRegistrationListener.java
+++ b/voip/java/android/net/sip/SipRegistrationListener.java
@@ -18,7 +18,6 @@ package android.net.sip;
 
 /**
  * Listener class to listen to SIP registration events.
- * @hide
  */
 public interface SipRegistrationListener {
     /**
diff --git a/voip/java/android/net/sip/SipSession.java b/voip/java/android/net/sip/SipSession.java
index 0cc7206..9c08e46 100644
--- a/voip/java/android/net/sip/SipSession.java
+++ b/voip/java/android/net/sip/SipSession.java
@@ -22,14 +22,12 @@ import android.util.Log;
 /**
  * A SIP session that is associated with a SIP dialog or a standalone
  * transaction not within a dialog.
- * @hide
  */
 public final class SipSession {
     private static final String TAG = "SipSession";
 
     /**
      * Defines {@link SipSession} states.
-     * @hide
      */
     public static class State {
         /** When session is ready to initiate a call or transaction. */
@@ -101,7 +99,6 @@ public final class SipSession {
 
     /**
      * Listener class that listens to {@link SipSession} events.
-     * @hide
      */
     public static class Listener {
         /**
@@ -281,7 +278,7 @@ public final class SipSession {
 
     /**
      * Gets the session state. The value returned must be one of the states in
-     * {@link SipSessionState}.
+     * {@link State}.
      *
      * @return the session state
      */
@@ -339,7 +336,7 @@ public final class SipSession {
      * Performs registration to the server specified by the associated local
      * profile. The session listener is called back upon success or failure of
      * registration. The method is only valid to call when the session state is
-     * in {@link SipSessionState#READY_TO_CALL}.
+     * in {@link State#READY_TO_CALL}.
      *
      * @param duration duration in second before the registration expires
      * @see Listener
@@ -357,7 +354,7 @@ public final class SipSession {
      * profile. Unregistration is technically the same as registration with zero
      * expiration duration. The session listener is called back upon success or
      * failure of unregistration. The method is only valid to call when the
-     * session state is in {@link SipSessionState#READY_TO_CALL}.
+     * session state is in {@link State#READY_TO_CALL}.
      *
      * @see Listener
      */
@@ -372,7 +369,7 @@ public final class SipSession {
     /**
      * Initiates a call to the specified profile. The session listener is called
      * back upon defined session events. The method is only valid to call when
-     * the session state is in {@link SipSessionState#READY_TO_CALL}.
+     * the session state is in {@link State#READY_TO_CALL}.
      *
      * @param callee the SIP profile to make the call to
      * @param sessionDescription the session description of this call
@@ -393,7 +390,7 @@ public final class SipSession {
     /**
      * Answers an incoming call with the specified session description. The
      * method is only valid to call when the session state is in
-     * {@link SipSessionState#INCOMING_CALL}.
+     * {@link State#INCOMING_CALL}.
      *
      * @param sessionDescription the session description to answer this call
      * @param timeout the session will be timed out if the call is not
@@ -411,10 +408,10 @@ public final class SipSession {
     /**
      * Ends an established call, terminates an outgoing call or rejects an
      * incoming call. The method is only valid to call when the session state is
-     * in {@link SipSessionState#IN_CALL},
-     * {@link SipSessionState#INCOMING_CALL},
-     * {@link SipSessionState#OUTGOING_CALL} or
-     * {@link SipSessionState#OUTGOING_CALL_RING_BACK}.
+     * in {@link State#IN_CALL},
+     * {@link State#INCOMING_CALL},
+     * {@link State#OUTGOING_CALL} or
+     * {@link State#OUTGOING_CALL_RING_BACK}.
      */
     public void endCall() {
         try {
@@ -426,7 +423,7 @@ public final class SipSession {
 
     /**
      * Changes the session description during a call. The method is only valid
-     * to call when the session state is in {@link SipSessionState#IN_CALL}.
+     * to call when the session state is in {@link State#IN_CALL}.
      *
      * @param sessionDescription the new session description
      * @param timeout the session will be timed out if the call is not