Merge changes I9adc67d2,I32dd22af

* changes:
  SIP: fix a recursion bug when local IP becomes invalid (network disconnected).
  SIP telephony: integrate with new RTP stack and other fixes.

7 files changed, 229 insertions, 125 deletions

diff --git a/phone/src2/com/android/internal/telephony/sip/SipCallBase.java b/phone/src2/com/android/internal/telephony/sip/SipCallBase.java
index 67f72d3..e7eda4f 100644
--- a/phone/src2/com/android/internal/telephony/sip/SipCallBase.java
+++ b/phone/src2/com/android/internal/telephony/sip/SipCallBase.java
@@ -92,7 +92,7 @@ abstract class SipCallBase extends Call {
         return connections.size() == MAX_CONNECTIONS_PER_CALL;
     }
 
-    protected void clearDisconnected() {
+    void clearDisconnected() {
         for (Iterator<Connection> it = connections.iterator(); it.hasNext(); ) {
             Connection c = it.next();
             if (c.getState() == State.DISCONNECTED) it.remove();
diff --git a/phone/src2/com/android/internal/telephony/sip/SipPhone.java b/phone/src2/com/android/internal/telephony/sip/SipPhone.java
index 49fe7db..1af5200 100755
--- a/phone/src2/com/android/internal/telephony/sip/SipPhone.java
+++ b/phone/src2/com/android/internal/telephony/sip/SipPhone.java
@@ -20,6 +20,7 @@ import android.content.ContentValues;
 import android.content.Context;
 import android.content.SharedPreferences;
 import android.net.Uri;
+import android.net.rtp.AudioGroup;
 import android.net.sip.SipAudioCall;
 import android.net.sip.SipManager;
 import android.net.sip.SipProfile;
@@ -211,10 +212,6 @@ public class SipPhone extends SipPhoneBase {
         }
     }
 
-    private void switchWaitingOrHoldingAndActive() {
-        // TODO
-    }
-
     public boolean canConference() {
         //TODO
         //return mCT.canConference();
@@ -321,8 +318,6 @@ public class SipPhone extends SipPhoneBase {
     }
 
     private class SipCall extends SipCallBase {
-        SipAudioCall mSipAudioCall;
-
         void switchWith(SipCall that) {
             synchronized (SipPhone.class) {
                 SipCall tmp = new SipCall();
@@ -335,16 +330,17 @@ public class SipPhone extends SipPhoneBase {
         private void takeOver(SipCall that) {
             connections = that.connections;
             state = that.state;
-            mSipAudioCall = that.mSipAudioCall;
             for (Connection c : connections) {
                 ((SipConnection) c).changeOwner(this);
             }
         }
 
+        @Override
         public Phone getPhone() {
             return SipPhone.this;
         }
 
+        @Override
         public List<Connection> getConnections() {
             synchronized (SipPhone.class) {
                 // FIXME should return Collections.unmodifiableList();
@@ -352,14 +348,13 @@ public class SipPhone extends SipPhoneBase {
             }
         }
 
-        public Connection dial(String calleeSipUri) throws SipException {
+        Connection dial(String calleeSipUri) throws SipException {
             try {
                 SipProfile callee =
                         new SipProfile.Builder(calleeSipUri).build();
                 SipConnection c = new SipConnection(this, callee);
                 connections.add(c);
                 c.dial();
-                mSipAudioCall = c.mSipAudioCall;
                 setState(Call.State.DIALING);
                 return c;
             } catch (ParseException e) {
@@ -370,6 +365,7 @@ public class SipPhone extends SipPhoneBase {
 
         // TODO: if this is the foreground call and a background call exists,
         // resume the background call
+        @Override
         public void hangup() throws CallStateException {
             Log.v(LOG_TAG, "hang up call: " + getState() + ": " + this
                     + " on phone " + getPhone());
@@ -385,8 +381,7 @@ public class SipPhone extends SipPhoneBase {
             setState(State.DISCONNECTING);
         }
 
-        public void take(SipAudioCall sipAudioCall, boolean makeCallWait) {
-            mSipAudioCall = sipAudioCall;
+        void take(SipAudioCall sipAudioCall, boolean makeCallWait) {
             SipProfile callee = sipAudioCall.getPeerProfile();
             SipConnection c = new SipConnection(this, callee);
             connections.add(c);
@@ -398,61 +393,74 @@ public class SipPhone extends SipPhoneBase {
             notifyNewRingingConnectionP(c);
         }
 
-        public void rejectCall() throws CallStateException {
+        void rejectCall() throws CallStateException {
             hangup();
         }
 
-        public void acceptCall() throws CallStateException {
-            try {
-                mSipAudioCall.answerCall();
-            } catch (SipException e) {
-                throw new CallStateException("acceptCall(): " + e);
+        void acceptCall() throws CallStateException {
+            if (this != foregroundCall) {
+                throw new CallStateException("acceptCall() in a non-fg call");
+            }
+            if (connections.size() != 1) {
+                throw new CallStateException("acceptCall() in a conf call");
             }
+            ((SipConnection) connections.get(0)).acceptCall();
         }
 
-        public void hold() {
-            if (mSipAudioCall == null) return;
-            try {
-                mSipAudioCall.holdCall();
-                setState(State.HOLDING);
-            } catch (SipException e) {
-                // FIXME: ignore?
-            }
+        void hold() throws CallStateException {
+            AudioGroup audioGroup = getAudioGroup();
+            if (audioGroup == null) return;
+            audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
+            setState(State.HOLDING);
+            for (Connection c : connections) ((SipConnection) c).hold();
         }
 
-        public void unhold() {
-            if (mSipAudioCall == null) return;
-            try {
-                mSipAudioCall.continueCall();
-                setState(State.ACTIVE);
-            } catch (SipException e) {
-                // FIXME: ignore?
-            }
+        void unhold() throws CallStateException {
+            AudioGroup audioGroup = getAudioGroup();
+            if (audioGroup == null) return;
+            audioGroup.setMode(AudioGroup.MODE_NORMAL);
+            setState(State.ACTIVE);
+            for (Connection c : connections) ((SipConnection) c).unhold();
         }
 
-        public void setMute(boolean muted) {
-            if (mSipAudioCall == null) return;
-            if (mSipAudioCall.isMuted() != muted) {
-                mSipAudioCall.toggleMute();
-            }
+        void setMute(boolean muted) {
+            AudioGroup audioGroup = getAudioGroup();
+            if (audioGroup == null) return;
+            audioGroup.setMode(
+                    muted ? AudioGroup.MODE_MUTED : AudioGroup.MODE_NORMAL);
         }
 
-        public boolean getMute() {
-            if (mSipAudioCall == null) return false;
-            return mSipAudioCall.isMuted();
+        boolean getMute() {
+            AudioGroup audioGroup = getAudioGroup();
+            if (audioGroup == null) return false;
+            return (audioGroup.getMode() == AudioGroup.MODE_MUTED);
         }
 
-        public void sendDtmf(char c) {
-            if (mSipAudioCall != null) {
-                mSipAudioCall.sendDtmf((int) (c - '0'));
-            }
+        void sendDtmf(char c) {
+            AudioGroup audioGroup = getAudioGroup();
+            if (audioGroup == null) return;
+            audioGroup.sendDtmf(convertDtmf(c));
         }
 
-        protected void clearDisconnected() {
-            super.clearDisconnected();
-            if (state == State.IDLE) mSipAudioCall = null;
+        private int convertDtmf(char c) {
+            int code = c - '0';
+            if ((code < 0) || (code > 9)) {
+                switch (c) {
+                    case '*': return 10;
+                    case '#': return 11;
+                    case 'A': return 12;
+                    case 'B': return 13;
+                    case 'C': return 14;
+                    case 'D': return 15;
+                    default:
+                        throw new IllegalArgumentException(
+                                "invalid DTMF char: " + (int) c);
+                }
+            }
+            return code;
         }
 
+        @Override
         protected void setState(State newState) {
             if (state != newState) {
                 Log.v(LOG_TAG, "++******++ call state changed: " + state
@@ -471,14 +479,14 @@ public class SipPhone extends SipPhoneBase {
             }
         }
 
-        private void onConnectionStateChanged(SipConnection conn) {
+        void onConnectionStateChanged(SipConnection conn) {
             // this can be called back when a conf call is formed
             if (state != State.ACTIVE) {
                 setState(conn.getState());
             }
         }
 
-        private void onConnectionEnded(SipConnection conn) {
+        void onConnectionEnded(SipConnection conn) {
             // set state to DISCONNECTED only when all conns are disconnected
             if (state != State.DISCONNECTED) {
                 boolean allConnectionsDisconnected = true;
@@ -492,6 +500,11 @@ public class SipPhone extends SipPhoneBase {
             }
             notifyDisconnectP(conn);
         }
+
+        private AudioGroup getAudioGroup() {
+            if (connections.isEmpty()) return null;
+            return ((SipConnection) connections.get(0)).getAudioGroup();
+        }
     }
 
     private class SipConnection extends SipConnectionBase {
@@ -502,6 +515,7 @@ public class SipPhone extends SipPhoneBase {
         private boolean mIncoming = false;
 
         private SipAudioCallAdapter mAdapter = new SipAudioCallAdapter() {
+            @Override
             protected void onCallEnded(DisconnectCause cause) {
                 if (getDisconnectCause() != DisconnectCause.LOCAL) {
                     setDisconnectCause(cause);
@@ -517,6 +531,7 @@ public class SipPhone extends SipPhoneBase {
                 }
             }
 
+            @Override
             public void onChanged(SipAudioCall call) {
                 synchronized (SipPhone.class) {
                     Call.State newState = getCallStateFrom(call);
@@ -540,17 +555,54 @@ public class SipPhone extends SipPhoneBase {
             mPeer = callee;
         }
 
-        public void take(SipAudioCall sipAudioCall, Call.State newState) {
+        void take(SipAudioCall sipAudioCall, Call.State newState) {
             mState = newState;
             mSipAudioCall = sipAudioCall;
             sipAudioCall.setListener(mAdapter); // call back to set state
             mIncoming = true;
         }
 
-        public void changeOwner(SipCall owner) {
+        void acceptCall() throws CallStateException {
+            try {
+                mSipAudioCall.answerCall();
+            } catch (SipException e) {
+                throw new CallStateException("acceptCall(): " + e);
+            }
+        }
+
+        void changeOwner(SipCall owner) {
             mOwner = owner;
         }
 
+        AudioGroup getAudioGroup() {
+            if (mSipAudioCall == null) return null;
+            return mSipAudioCall.getAudioGroup();
+        }
+
+        void dial() throws SipException {
+            mState = Call.State.DIALING;
+            mSipAudioCall = mSipManager.makeAudioCall(mContext, mProfile,
+                    mPeer, null);
+            mSipAudioCall.setRingbackToneEnabled(false);
+            mSipAudioCall.setListener(mAdapter);
+        }
+
+        void hold() throws CallStateException {
+            try {
+                mSipAudioCall.holdCall();
+            } catch (SipException e) {
+                throw new CallStateException("hold(): " + e);
+            }
+        }
+
+        void unhold() throws CallStateException {
+            try {
+                mSipAudioCall.continueCall();
+            } catch (SipException e) {
+                throw new CallStateException("unhold(): " + e);
+            }
+        }
+
         @Override
         public Call.State getState() {
             return mState;
@@ -566,14 +618,6 @@ public class SipPhone extends SipPhoneBase {
             return mPeer.getUriString();
         }
 
-        public void dial() throws SipException {
-            mState = Call.State.DIALING;
-            mSipAudioCall = mSipManager.makeAudioCall(mContext, mProfile,
-                    mPeer, null);
-            mSipAudioCall.setRingbackToneEnabled(false);
-            mSipAudioCall.setListener(mAdapter);
-        }
-
         @Override
         public SipCall getCall() {
             return mOwner;
@@ -632,6 +676,7 @@ public class SipPhone extends SipPhoneBase {
     }
 
     private static Call.State getCallStateFrom(SipAudioCall sipAudioCall) {
+        if (sipAudioCall.isOnHold()) return Call.State.HOLDING;
         SipSessionState sessionState = sipAudioCall.getState();
         switch (sessionState) {
             case READY_TO_CALL:            return Call.State.IDLE;
@@ -652,14 +697,17 @@ public class SipPhone extends SipPhoneBase {
 
         protected abstract void onCallEnded(Connection.DisconnectCause cause);
 
+        @Override
         public void onCallEnded(SipAudioCall call) {
             onCallEnded(Connection.DisconnectCause.NORMAL);
         }
 
+        @Override
         public void onCallBusy(SipAudioCall call) {
             onCallEnded(Connection.DisconnectCause.BUSY);
         }
 
+        @Override
         public void onError(SipAudioCall call, String errorMessage) {
             mError = new SipException(errorMessage);
             // FIXME: specify error
diff --git a/src/android/net/sip/SipAudioCall.java b/src/android/net/sip/SipAudioCall.java
index 79f4481..abdc9d7 100644
--- a/src/android/net/sip/SipAudioCall.java
+++ b/src/android/net/sip/SipAudioCall.java
@@ -16,6 +16,8 @@
 
 package android.net.sip;
 
+import android.net.rtp.AudioGroup;
+import android.net.rtp.AudioStream;
 import android.os.Message;
 
 import javax.sip.SipException;
@@ -174,13 +176,19 @@ public interface SipAudioCall {
     /** Ends a call. */
     void endCall() throws SipException;
 
-    /** Puts a call on hold. */
+    /**
+     * Puts a call on hold.  When succeeds,
+     * {@link #Listener#onCallHeld(SipAudioCall)} is called.
+     */
     void holdCall() throws SipException;
 
     /** Answers a call. */
     void answerCall() throws SipException;
 
-    /** Continues a call that's on hold. */
+    /**
+     * Continues a call that's on hold. When succeeds,
+     * {@link #Listener#onCallEstablished(SipAudioCall)} is called.
+     */
     void continueCall() throws SipException;
 
     /** Puts the device to in-call mode. */
@@ -222,6 +230,32 @@ public interface SipAudioCall {
     void sendDtmf(int code, Message result);
 
     /**
+     * Gets the {@link AudioStream} object used in this call. The object
+     * represents the RTP stream that carries the audio data to and from the
+     * peer. The object may not be created before the call is established. And
+     * it is undefined after the call ends or the {@link #close} method is
+     * called.
+     *
+     * @return the {@link AudioStream} object or null if the RTP stream has not
+     *      yet been set up
+     */
+    AudioStream getAudioStream();
+
+    /**
+     * Gets the {@link AudioGroup} object which the {@link AudioStream} object
+     * joins. The group object may not exist before the call is established.
+     * Also, the {@code AudioStream} may change its group during a call (e.g.,
+     * after the call is held/un-held). Finally, the {@code AudioGroup} object
+     * returned by this method is undefined after the call ends or the
+     * {@link #close} method is called.
+     *
+     * @return the {@link AudioGroup} object or null if the RTP stream has not
+     *      yet been set up
+     * @see #getAudioStream
+     */
+    AudioGroup getAudioGroup();
+
+    /**
      * Checks if the call is established.
      *
      * @return true if the call is established
diff --git a/src/com/android/sip/FLog.java b/src/com/android/sip/FLog.java
index 2cc1827..460d759 100644
--- a/src/com/android/sip/FLog.java
+++ b/src/com/android/sip/FLog.java
@@ -58,6 +58,7 @@ class FLog {
     }
 
     static synchronized void write(String type, String tag, String msg) {
+        if (writer == null) return;
         try {
             writer.println(String.format("%s %s %s| %s| %s", type,
                     getDate(), getTime(), tag, msg));
@@ -69,6 +70,7 @@ class FLog {
 
     static synchronized void write(String type, String tag, String msg,
             Throwable t) {
+        if (writer == null) return;
         try {
             writer.println(String.format("%s %s| %s| %s", type,
                     getDate(), getTime(), tag, msg));
diff --git a/src/com/android/sip/SipAudioCallImpl.java b/src/com/android/sip/SipAudioCallImpl.java
index 15fc726..ffecb16 100644
--- a/src/com/android/sip/SipAudioCallImpl.java
+++ b/src/com/android/sip/SipAudioCallImpl.java
@@ -18,16 +18,16 @@ package com.android.sip;
 
 import gov.nist.javax.sdp.fields.SDPKeywords;
 
-import com.android.sip.rtp.AudioCodec;
-import com.android.sip.rtp.AudioStream;
-import com.android.sip.rtp.RtpSocket;
-
 import android.content.Context;
 import android.media.AudioManager;
 import android.media.Ringtone;
 import android.media.RingtoneManager;
 import android.media.ToneGenerator;
 import android.net.Uri;
+import android.net.rtp.AudioCodec;
+import android.net.rtp.AudioGroup;
+import android.net.rtp.AudioStream;
+import android.net.rtp.RtpStream;
 import android.net.sip.ISipSession;
 import android.net.sip.SdpSessionDescription;
 import android.net.sip.SessionDescription;
@@ -71,9 +71,7 @@ public class SipAudioCallImpl extends SipSessionAdapter
 
     private AudioStream mRtpSession;
     private SdpSessionDescription.AudioCodec mCodec;
-    private RtpSocket mMediaSocket;
     private long mSessionId = -1L; // SDP session ID
-    private boolean mChangingSession = false;
     private boolean mInCall = false;
     private boolean mMuted = false;
     private boolean mHold = false;
@@ -138,7 +136,6 @@ public class SipAudioCallImpl extends SipSessionAdapter
         mSipSession = null;
         mInCall = false;
         mHold = false;
-        mChangingSession = false;
         mSessionId = -1L;
     }
 
@@ -222,7 +219,6 @@ public class SipAudioCallImpl extends SipSessionAdapter
     private synchronized void establishCall(byte[] sessionDescription) {
         stopRingbackTone();
         stopRinging();
-        mChangingSession = false;
         try {
             SdpSessionDescription sd =
                     new SdpSessionDescription(sessionDescription);
@@ -241,7 +237,11 @@ public class SipAudioCallImpl extends SipSessionAdapter
         Listener listener = mListener;
         if (listener != null) {
             try {
-                listener.onCallEstablished(SipAudioCallImpl.this);
+                if (mHold) {
+                    listener.onCallHeld(SipAudioCallImpl.this);
+                } else {
+                    listener.onCallEstablished(SipAudioCallImpl.this);
+                }
             } catch (Throwable t) {
                 Log.e(TAG, "onCallEstablished()", t);
             }
@@ -280,16 +280,13 @@ public class SipAudioCallImpl extends SipSessionAdapter
     public void onCallChangeFailed(ISipSession session,
             String className, String message) {
         // TODO:
-        mChangingSession = false;
     }
 
     @Override
     public void onError(ISipSession session, String className,
             String message) {
         Log.d(TAG, "sip session error: " + className + ": " + message);
-        // don't stop RTP session on SIP error
-        // TODO: what to do if call is on hold
-        close(false);
+        close(true);
         Listener listener = mListener;
         if (listener != null) {
             try {
@@ -335,20 +332,23 @@ public class SipAudioCallImpl extends SipSessionAdapter
         try {
             stopRinging();
             if (mSipSession != null) mSipSession.endCall();
+            stopCall(true);
         } catch (Throwable e) {
             throwSipException(e);
         }
     }
 
     public synchronized void holdCall() throws SipException {
+        if (mHold) return;
         try {
-            if (mChangingSession) return;
-            mChangingSession = true;
-            mHold = true;
             mSipSession.changeCall(createHoldSessionDescription());
+            mHold = true;
         } catch (Throwable e) {
             throwSipException(e);
         }
+
+        AudioGroup audioGroup = getAudioGroup();
+        if (audioGroup != null) audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
     }
 
     public synchronized void answerCall() throws SipException {
@@ -362,14 +362,16 @@ public class SipAudioCallImpl extends SipSessionAdapter
     }
 
     public synchronized void continueCall() throws SipException {
+        if (!mHold) return;
         try {
-            if (mChangingSession) return;
-            mChangingSession = true;
             mHold = false;
             mSipSession.changeCall(createContinueSessionDescription());
         } catch (Throwable e) {
             throwSipException(e);
         }
+
+        AudioGroup audioGroup = getAudioGroup();
+        if (audioGroup != null) audioGroup.setMode(AudioGroup.MODE_NORMAL);
     }
 
     private SessionDescription createOfferSessionDescription() {
@@ -456,12 +458,10 @@ public class SipAudioCallImpl extends SipSessionAdapter
     }
 
     public synchronized void toggleMute() {
-        if (mRtpSession != null) {
-            if (mMuted) {
-                mRtpSession.startSending();
-            } else {
-                mRtpSession.stopSending();
-            }
+        AudioGroup audioGroup = getAudioGroup();
+        if (audioGroup != null) {
+            audioGroup.setMode(
+                    mMuted ? AudioGroup.MODE_NORMAL : AudioGroup.MODE_MUTED);
             mMuted = !mMuted;
         }
     }
@@ -485,13 +485,23 @@ public class SipAudioCallImpl extends SipSessionAdapter
     }
 
     public synchronized void sendDtmf(int code, Message result) {
-        if ((mSipSession != null) && (SipSessionState.IN_CALL == getState())) {
+        AudioGroup audioGroup = getAudioGroup();
+        if ((audioGroup != null) && (mSipSession != null)
+                && (SipSessionState.IN_CALL == getState())) {
             Log.v(TAG, "send DTMF: " + code);
-            mRtpSession.sendDtmf(code);
+            audioGroup.sendDtmf(code);
         }
         if (result != null) result.sendToTarget();
     }
 
+    public synchronized AudioStream getAudioStream() {
+        return mRtpSession;
+    }
+
+    public synchronized AudioGroup getAudioGroup() {
+        return ((mRtpSession == null) ? null : mRtpSession.getAudioGroup());
+    }
+
     private SdpSessionDescription.AudioCodec getCodec(SdpSessionDescription sd) {
         HashMap<String, AudioCodec> acceptableCodecs =
                 new HashMap<String, AudioCodec>();
@@ -539,30 +549,39 @@ public class SipAudioCallImpl extends SipSessionAdapter
         int frameSize = sampleRate / 50; // 160
         try {
             // TODO: get sample rate from sdp
-            mMediaSocket.associate(InetAddress.getByName(peerMediaAddress),
-                    peerMediaPort);
             mCodec = getCodec(peerSd);
-            mRtpSession = new AudioStream(mMediaSocket);
-            mRtpSession.setCodec(convert(mCodec), mCodec.payloadType);
-            mRtpSession.setDtmf(DTMF);
-            mRtpSession.prepare();
+
+            AudioStream audioStream = mRtpSession;
+            audioStream.associate(InetAddress.getByName(peerMediaAddress),
+                    peerMediaPort);
+            audioStream.setCodec(convert(mCodec), mCodec.payloadType);
+            audioStream.setDtmfType(DTMF);
             Log.d(TAG, "start media: localPort=" + localPort + ", peer="
                     + peerMediaAddress + ":" + peerMediaPort);
-            if (mHold) {
-                Log.d(TAG, "   on hold");
-                mRtpSession.stopSending();
-                mRtpSession.stopReceiving();
-            } else {
-                if (peerSd.isSending(AUDIO)) {
-                    Log.d(TAG, "   start receiving");
-                    mRtpSession.startReceiving();
+
+            audioStream.setMode(RtpStream.MODE_NORMAL);
+            if (!mHold) {
+                // FIXME: won't work if peer is not sending nor receiving
+                if (!peerSd.isSending(AUDIO)) {
+                    Log.d(TAG, "   not receiving");
+                    audioStream.setMode(RtpStream.MODE_SEND_ONLY);
                 }
-                if (peerSd.isReceiving(AUDIO)) {
-                    Log.d(TAG, "   start sending");
-                    mRtpSession.startSending();
+                if (!peerSd.isReceiving(AUDIO)) {
+                    Log.d(TAG, "   not sending");
+                    audioStream.setMode(RtpStream.MODE_RECEIVE_ONLY);
                 }
             }
             setInCallMode();
+
+            AudioGroup audioGroup = new AudioGroup();
+            audioStream.join(audioGroup);
+            if (mHold) {
+                audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
+            } else if (mMuted) {
+                audioGroup.setMode(AudioGroup.MODE_MUTED);
+            } else {
+                audioGroup.setMode(AudioGroup.MODE_NORMAL);
+            }
         } catch (Exception e) {
             Log.e(TAG, "call()", e);
         }
@@ -571,21 +590,22 @@ public class SipAudioCallImpl extends SipSessionAdapter
     private void stopCall(boolean releaseSocket) {
         Log.d(TAG, "stop audiocall");
         if (mRtpSession != null) {
-            mRtpSession.close();
+            mRtpSession.join(null);
+
             if (releaseSocket) {
-                mMediaSocket = null;
+                mRtpSession.release();
+                mRtpSession = null;
             }
         }
         setInCallMode();
     }
 
     private int getLocalMediaPort() {
-        if (mMediaSocket != null) return mMediaSocket.getLocalPort();
+        if (mRtpSession != null) return mRtpSession.getLocalPort();
         try {
-            RtpSocket s = mMediaSocket =
-                    new RtpSocket(InetAddress.getByName(getLocalIp()));
-            int localPort = s.getLocalPort();
-            return localPort;
+            AudioStream s = mRtpSession =
+                    new AudioStream(InetAddress.getByName(getLocalIp()));
+            return s.getLocalPort();
         } catch (IOException e) {
             Log.w(TAG, "getLocalMediaPort(): " + e);
             throw new RuntimeException(e);
diff --git a/src/com/android/sip/SipServiceImpl.java b/src/com/android/sip/SipServiceImpl.java
index baf7e4c..f926905 100644
--- a/src/com/android/sip/SipServiceImpl.java
+++ b/src/com/android/sip/SipServiceImpl.java
@@ -411,7 +411,7 @@ class SipServiceImpl extends ISipService.Stub {
                 // network disconnected
                 FLog.w(TAG, "resetGroup(): network disconnected?");
                 if (localIp != null) {
-                    resetGroup(localIp);
+                    resetGroup(null); // reset w/o local IP
                 } else {
                     // recursive
                     Log.wtf(TAG, "impossible!");
diff --git a/src/com/android/sip/SipSessionListenerProxy.java b/src/com/android/sip/SipSessionListenerProxy.java
index c640df4..531e394 100644
--- a/src/com/android/sip/SipSessionListenerProxy.java
+++ b/src/com/android/sip/SipSessionListenerProxy.java
@@ -49,7 +49,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onCalling(session);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onCalling(): " + t);
+                    Log.w(TAG, "onCalling()", t);
                 }
             }
         });
@@ -63,7 +63,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onRinging(session, caller, sessionDescription);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onRinging(): " + t);
+                    Log.w(TAG, "onRinging()", t);
                 }
             }
         });
@@ -76,7 +76,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onRingingBack(session);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onRingingBack(): " + t);
+                    Log.w(TAG, "onRingingBack()", t);
                 }
             }
         });
@@ -90,7 +90,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onCallEstablished(session, sessionDescription);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onCallEstablished(): " + t);
+                    Log.w(TAG, "onCallEstablished()", t);
                 }
             }
         });
@@ -103,7 +103,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onCallEnded(session);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onCallEnded(): " + t);
+                    Log.w(TAG, "onCallEnded()", t);
                 }
             }
         });
@@ -116,7 +116,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onCallBusy(session);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onCallBusy(): " + t);
+                    Log.w(TAG, "onCallBusy()", t);
                 }
             }
         });
@@ -130,7 +130,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onCallChangeFailed(session, className, message);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onCallChangeFailed(): " + t);
+                    Log.w(TAG, "onCallChangeFailed()", t);
                 }
             }
         });
@@ -144,7 +144,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onError(session, className, message);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onError(): " + t);
+                    Log.w(TAG, "onError()", t);
                 }
             }
         });
@@ -157,7 +157,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onRegistering(session);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onRegistering(): " + t);
+                    Log.w(TAG, "onRegistering()", t);
                 }
             }
         });
@@ -185,7 +185,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onRegistrationFailed(session, className, message);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onRegistrationFailed(): " + t);
+                    Log.w(TAG, "onRegistrationFailed()", t);
                 }
             }
         });
@@ -198,7 +198,7 @@ class SipSessionListenerProxy extends ISipSessionListener.Stub {
                 try {
                     mListener.onRegistrationTimeout(session);
                 } catch (Throwable t) {
-                    Log.w(TAG, "onRegistrationTimeout(): " + t);
+                    Log.w(TAG, "onRegistrationTimeout()", t);
                 }
             }
         });