resolved conflicts for merge of ee813bae to master

Change-Id: I84ca961fb18b29313b5ec6041a79ca87a1a1cd68

1 files changed, 830 insertions, 125 deletions

diff --git a/voip/java/android/net/sip/SipAudioCall.java b/voip/java/android/net/sip/SipAudioCall.java
index 0069fe065284..2f4fd90e7019 100644
--- a/voip/java/android/net/sip/SipAudioCall.java
+++ b/voip/java/android/net/sip/SipAudioCall.java
@@ -16,120 +16,184 @@
 
 package android.net.sip;
 
+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.SimpleSessionDescription.Media;
+import android.net.wifi.WifiManager;
 import android.os.Message;
+import android.os.RemoteException;
+import android.os.Vibrator;
+import android.provider.Settings;
+import android.util.Log;
+
+import java.io.IOException;
+import java.net.InetAddress;
+import java.net.UnknownHostException;
+import java.util.ArrayList;
+import java.util.HashMap;
+import java.util.List;
+import java.util.Map;
 
 /**
- * Interface for making audio calls over SIP.
- * @hide
+ * Class that handles an audio call over SIP.
  */
-public interface SipAudioCall {
+/** @hide */
+public class SipAudioCall extends SipSessionAdapter {
+    private static final String TAG = SipAudioCall.class.getSimpleName();
+    private static final boolean RELEASE_SOCKET = true;
+    private static final boolean DONT_RELEASE_SOCKET = false;
+    private static final int SESSION_TIMEOUT = 5; // in seconds
+
     /** Listener class for all event callbacks. */
-    public interface Listener {
+    public static class Listener {
         /**
          * Called when the call object is ready to make another call.
+         * The default implementation calls {@link #onChange}.
          *
          * @param call the call object that is ready to make another call
          */
-        void onReadyToCall(SipAudioCall call);
+        public void onReadyToCall(SipAudioCall call) {
+            onChanged(call);
+        }
 
         /**
          * Called when a request is sent out to initiate a new call.
+         * The default implementation calls {@link #onChange}.
          *
          * @param call the call object that carries out the audio call
          */
-        void onCalling(SipAudioCall call);
+        public void onCalling(SipAudioCall call) {
+            onChanged(call);
+        }
 
         /**
          * Called when a new call comes in.
+         * The default implementation calls {@link #onChange}.
          *
          * @param call the call object that carries out the audio call
          * @param caller the SIP profile of the caller
          */
-        void onRinging(SipAudioCall call, SipProfile caller);
+        public void onRinging(SipAudioCall call, SipProfile caller) {
+            onChanged(call);
+        }
 
         /**
-         * Called when a RINGING response is received for the INVITE request sent
+         * Called when a RINGING response is received for the INVITE request
+         * sent. The default implementation calls {@link #onChange}.
          *
          * @param call the call object that carries out the audio call
          */
-        void onRingingBack(SipAudioCall call);
+        public void onRingingBack(SipAudioCall call) {
+            onChanged(call);
+        }
 
         /**
          * Called when the session is established.
+         * The default implementation calls {@link #onChange}.
          *
          * @param call the call object that carries out the audio call
          */
-        void onCallEstablished(SipAudioCall call);
+        public void onCallEstablished(SipAudioCall call) {
+            onChanged(call);
+        }
 
         /**
          * Called when the session is terminated.
+         * The default implementation calls {@link #onChange}.
          *
          * @param call the call object that carries out the audio call
          */
-        void onCallEnded(SipAudioCall call);
+        public void onCallEnded(SipAudioCall call) {
+            onChanged(call);
+        }
 
         /**
          * Called when the peer is busy during session initialization.
+         * The default implementation calls {@link #onChange}.
          *
          * @param call the call object that carries out the audio call
          */
-        void onCallBusy(SipAudioCall call);
+        public void onCallBusy(SipAudioCall call) {
+            onChanged(call);
+        }
 
         /**
          * Called when the call is on hold.
+         * The default implementation calls {@link #onChange}.
          *
          * @param call the call object that carries out the audio call
          */
-        void onCallHeld(SipAudioCall call);
+        public void onCallHeld(SipAudioCall call) {
+            onChanged(call);
+        }
 
         /**
-         * Called when an error occurs.
+         * Called when an error occurs. The default implementation is no op.
          *
          * @param call the call object that carries out the audio call
          * @param errorCode error code of this error
          * @param errorMessage error message
          * @see SipErrorCode
          */
-        void onError(SipAudioCall call, int errorCode, String errorMessage);
+        public void onError(SipAudioCall call, int errorCode,
+                String errorMessage) {
+            // no-op
+        }
+
+        /**
+         * Called when an event occurs and the corresponding callback is not
+         * overridden. The default implementation is no op. Error events are
+         * not re-directed to this callback and are handled in {@link #onError}.
+         */
+        public void onChanged(SipAudioCall call) {
+            // no-op
+        }
     }
 
+    private Context mContext;
+    private SipProfile mLocalProfile;
+    private SipAudioCall.Listener mListener;
+    private SipSession mSipSession;
+
+    private long mSessionId = System.currentTimeMillis();
+    private String mPeerSd;
+
+    private AudioStream mAudioStream;
+    private AudioGroup mAudioGroup;
+
+    private boolean mInCall = false;
+    private boolean mMuted = false;
+    private boolean mHold = false;
+
+    private boolean mRingbackToneEnabled = true;
+    private boolean mRingtoneEnabled = true;
+    private Ringtone mRingtone;
+    private ToneGenerator mRingbackTone;
+
+    private SipProfile mPendingCallRequest;
+    private WifiManager mWm;
+    private WifiManager.WifiLock mWifiHighPerfLock;
+
+    private int mErrorCode = SipErrorCode.NO_ERROR;
+    private String mErrorMessage;
+
     /**
-     * The adapter class for {@link Listener}. The default implementation of
-     * all callback methods is no-op.
+     * Creates a call object with the local SIP profile.
+     * @param context the context for accessing system services such as
+     *        ringtone, audio, WIFI etc
      */
-    public class Adapter implements Listener {
-        protected void onChanged(SipAudioCall call) {
-        }
-        public void onReadyToCall(SipAudioCall call) {
-            onChanged(call);
-        }
-        public void onCalling(SipAudioCall call) {
-            onChanged(call);
-        }
-        public void onRinging(SipAudioCall call, SipProfile caller) {
-            onChanged(call);
-        }
-        public void onRingingBack(SipAudioCall call) {
-            onChanged(call);
-        }
-        public void onCallEstablished(SipAudioCall call) {
-            onChanged(call);
-        }
-        public void onCallEnded(SipAudioCall call) {
-            onChanged(call);
-        }
-        public void onCallBusy(SipAudioCall call) {
-            onChanged(call);
-        }
-        public void onCallHeld(SipAudioCall call) {
-            onChanged(call);
-        }
-        public void onError(SipAudioCall call, int errorCode,
-                String errorMessage) {
-            onChanged(call);
-        }
+    public SipAudioCall(Context context, SipProfile localProfile) {
+        mContext = context;
+        mLocalProfile = localProfile;
+        mWm = (WifiManager) context.getSystemService(Context.WIFI_SERVICE);
     }
 
     /**
@@ -139,7 +203,9 @@ public interface SipAudioCall {
      * @param listener to listen to the audio call events of this object
      * @see #setListener(Listener, boolean)
      */
-    void setListener(Listener listener);
+    public void setListener(SipAudioCall.Listener listener) {
+        setListener(listener, false);
+    }
 
     /**
      * Sets the listener to listen to the audio call events. A
@@ -150,44 +216,355 @@ public interface SipAudioCall {
      * @param callbackImmediately set to true if the caller wants to be called
      *      back immediately on the current state
      */
-    void setListener(Listener listener, boolean callbackImmediately);
+    public void setListener(SipAudioCall.Listener listener,
+            boolean callbackImmediately) {
+        mListener = listener;
+        try {
+            if ((listener == null) || !callbackImmediately) {
+                // do nothing
+            } else if (mErrorCode != SipErrorCode.NO_ERROR) {
+                listener.onError(this, mErrorCode, mErrorMessage);
+            } else if (mInCall) {
+                if (mHold) {
+                    listener.onCallHeld(this);
+                } else {
+                    listener.onCallEstablished(this);
+                }
+            } else {
+                int state = getState();
+                switch (state) {
+                    case SipSession.State.READY_TO_CALL:
+                        listener.onReadyToCall(this);
+                        break;
+                    case SipSession.State.INCOMING_CALL:
+                        listener.onRinging(this, getPeerProfile());
+                        break;
+                    case SipSession.State.OUTGOING_CALL:
+                        listener.onCalling(this);
+                        break;
+                    case SipSession.State.OUTGOING_CALL_RING_BACK:
+                        listener.onRingingBack(this);
+                        break;
+                }
+            }
+        } catch (Throwable t) {
+            Log.e(TAG, "setListener()", t);
+        }
+    }
+
+    /**
+     * Checks if the call is established.
+     *
+     * @return true if the call is established
+     */
+    public synchronized boolean isInCall() {
+        return mInCall;
+    }
+
+    /**
+     * Checks if the call is on hold.
+     *
+     * @return true if the call is on hold
+     */
+    public synchronized boolean isOnHold() {
+        return mHold;
+    }
 
     /**
      * Closes this object. This object is not usable after being closed.
      */
-    void close();
+    public void close() {
+        close(true);
+    }
+
+    private synchronized void close(boolean closeRtp) {
+        if (closeRtp) stopCall(RELEASE_SOCKET);
+        stopRingbackTone();
+        stopRinging();
+
+        mInCall = false;
+        mHold = false;
+        mSessionId = System.currentTimeMillis();
+        mErrorCode = SipErrorCode.NO_ERROR;
+        mErrorMessage = null;
+
+        if (mSipSession != null) {
+            mSipSession.setListener(null);
+            mSipSession = null;
+        }
+    }
 
     /**
-     * Initiates an audio call to the specified profile. The attempt will be
-     * timed out if the call is not established within {@code timeout} seconds
-     * and {@code Listener.onError(SipAudioCall, SipErrorCode.TIME_OUT, String)}
-     * will be called.
+     * Gets the local SIP profile.
      *
-     * @param callee the SIP profile to make the call to
-     * @param sipManager the {@link SipManager} object to help make call with
-     * @param timeout the timeout value in seconds
-     * @see Listener.onError
+     * @return the local SIP profile
      */
-    void makeCall(SipProfile callee, SipManager sipManager, int timeout)
-            throws SipException;
+    public synchronized SipProfile getLocalProfile() {
+        return mLocalProfile;
+    }
 
     /**
-     * Starts the audio for the established call. This method should be called
-     * after {@link Listener#onCallEstablished} is called.
+     * Gets the peer's SIP profile.
+     *
+     * @return the peer's SIP profile
      */
-    void startAudio();
+    public synchronized SipProfile getPeerProfile() {
+        return (mSipSession == null) ? null : mSipSession.getPeerProfile();
+    }
+
+    /**
+     * Gets the state of the {@link SipSession} that carries this call.
+     * The value returned must be one of the states in {@link SipSession.State}.
+     *
+     * @return the session state
+     */
+    public synchronized int getState() {
+        if (mSipSession == null) return SipSession.State.READY_TO_CALL;
+        return mSipSession.getState();
+    }
+
+
+    /**
+     * Gets the {@link SipSession} that carries this call.
+     *
+     * @return the session object that carries this call
+     * @hide
+     */
+    public synchronized SipSession getSipSession() {
+        return mSipSession;
+    }
+
+    private SipSession.Listener createListener() {
+        return new SipSession.Listener() {
+            @Override
+            public void onCalling(SipSession session) {
+                Log.d(TAG, "calling... " + session);
+                Listener listener = mListener;
+                if (listener != null) {
+                    try {
+                        listener.onCalling(SipAudioCall.this);
+                    } catch (Throwable t) {
+                        Log.i(TAG, "onCalling(): " + t);
+                    }
+                }
+            }
+
+            @Override
+            public void onRingingBack(SipSession session) {
+                Log.d(TAG, "sip call ringing back: " + session);
+                if (!mInCall) startRingbackTone();
+                Listener listener = mListener;
+                if (listener != null) {
+                    try {
+                        listener.onRingingBack(SipAudioCall.this);
+                    } catch (Throwable t) {
+                        Log.i(TAG, "onRingingBack(): " + t);
+                    }
+                }
+            }
+
+            @Override
+            public synchronized void onRinging(SipSession session,
+                    SipProfile peerProfile, String sessionDescription) {
+                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();
+                }
+            }
+
+            @Override
+            public void onCallEstablished(SipSession session,
+                    String sessionDescription) {
+                stopRingbackTone();
+                stopRinging();
+                mPeerSd = sessionDescription;
+                Log.v(TAG, "onCallEstablished()" + mPeerSd);
+
+                Listener listener = mListener;
+                if (listener != null) {
+                    try {
+                        if (mHold) {
+                            listener.onCallHeld(SipAudioCall.this);
+                        } else {
+                            listener.onCallEstablished(SipAudioCall.this);
+                        }
+                    } catch (Throwable t) {
+                        Log.i(TAG, "onCallEstablished(): " + t);
+                    }
+                }
+            }
+
+            @Override
+            public void onCallEnded(SipSession session) {
+                Log.d(TAG, "sip call ended: " + session);
+                Listener listener = mListener;
+                if (listener != null) {
+                    try {
+                        listener.onCallEnded(SipAudioCall.this);
+                    } catch (Throwable t) {
+                        Log.i(TAG, "onCallEnded(): " + t);
+                    }
+                }
+                close();
+            }
+
+            @Override
+            public void onCallBusy(SipSession session) {
+                Log.d(TAG, "sip call busy: " + session);
+                Listener listener = mListener;
+                if (listener != null) {
+                    try {
+                        listener.onCallBusy(SipAudioCall.this);
+                    } catch (Throwable t) {
+                        Log.i(TAG, "onCallBusy(): " + t);
+                    }
+                }
+                close(false);
+            }
+
+            @Override
+            public void onCallChangeFailed(SipSession session, int errorCode,
+                    String message) {
+                Log.d(TAG, "sip call change failed: " + message);
+                mErrorCode = errorCode;
+                mErrorMessage = message;
+                Listener listener = mListener;
+                if (listener != null) {
+                    try {
+                        listener.onError(SipAudioCall.this, mErrorCode,
+                                message);
+                    } catch (Throwable t) {
+                        Log.i(TAG, "onCallBusy(): " + t);
+                    }
+                }
+            }
+
+            @Override
+            public void onError(SipSession session, int errorCode,
+                    String message) {
+                SipAudioCall.this.onError(errorCode, message);
+            }
+
+            @Override
+            public void onRegistering(SipSession session) {
+                // irrelevant
+            }
+
+            @Override
+            public void onRegistrationTimeout(SipSession session) {
+                // irrelevant
+            }
+
+            @Override
+            public void onRegistrationFailed(SipSession session, int errorCode,
+                    String message) {
+                // irrelevant
+            }
+
+            @Override
+            public void onRegistrationDone(SipSession session, int duration) {
+                // irrelevant
+            }
+        };
+    }
+
+    private void onError(int errorCode, String message) {
+        Log.d(TAG, "sip session error: "
+                + SipErrorCode.toString(errorCode) + ": " + message);
+        mErrorCode = errorCode;
+        mErrorMessage = message;
+        Listener listener = mListener;
+        if (listener != null) {
+            try {
+                listener.onError(this, errorCode, message);
+            } catch (Throwable t) {
+                Log.i(TAG, "onError(): " + t);
+            }
+        }
+        synchronized (this) {
+            if ((errorCode == SipErrorCode.DATA_CONNECTION_LOST)
+                    || !isInCall()) {
+                close(true);
+            }
+        }
+    }
 
     /**
      * Attaches an incoming call to this call object.
      *
      * @param session the session that receives the incoming call
      * @param sessionDescription the session description of the incoming call
+     * @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());
+
+            if (getState() == SipSession.State.INCOMING_CALL) startRinging();
+        } catch (Throwable e) {
+            Log.e(TAG, "attachCall()", e);
+            throwSipException(e);
+        }
+    }
+
+    /**
+     * Initiates an audio call to the specified profile. The attempt will be
+     * timed out if the call is not established within {@code timeout} seconds
+     * and {@code Listener.onError(SipAudioCall, SipErrorCode.TIME_OUT, String)}
+     * will be called.
+     *
+     * @param callee the SIP profile to make the call to
+     * @param sipManager the {@link SipManager} object to help make call with
+     * @param timeout the timeout value in seconds. Default value (defined by
+     *        SIP protocol) is used if {@code timeout} is zero or negative.
+     * @see Listener.onError
+     * @throws SipException if the SIP service fails to create a session for the
+     *        call
      */
-    void attachCall(ISipSession session, String sessionDescription)
-            throws SipException;
+    public synchronized void makeCall(SipProfile peerProfile,
+        SipManager sipManager, int timeout) throws SipException {
+        SipSession s = mSipSession = sipManager.createSipSession(
+                mLocalProfile, createListener());
+        if (s == null) {
+            throw new SipException(
+                    "Failed to create SipSession; network available?");
+        }
+        try {
+            mAudioStream = new AudioStream(InetAddress.getByName(getLocalIp()));
+            s.makeCall(peerProfile, createOffer().encode(), timeout);
+        } catch (IOException e) {
+            throw new SipException("makeCall()", e);
+        }
+    }
 
-    /** Ends a call. */
-    void endCall() throws SipException;
+    /**
+     * 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;
+
+        // perform the above local ops first and then network op
+        if (mSipSession != null) mSipSession.endCall();
+    }
 
     /**
      * Puts a call on hold.  When succeeds, {@link Listener#onCallHeld} is
@@ -196,10 +573,19 @@ public interface SipAudioCall {
      * {@code Listener.onError(SipAudioCall, SipErrorCode.TIME_OUT, String)}
      * will be called.
      *
-     * @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.
      * @see Listener.onError
+     * @throws SipException if the SIP service fails to hold the call
      */
-    void holdCall(int timeout) throws SipException;
+    public synchronized void holdCall(int timeout) throws SipException {
+        if (mHold) return;
+        mSipSession.changeCall(createHoldOffer().encode(), timeout);
+        mHold = true;
+
+        AudioGroup audioGroup = getAudioGroup();
+        if (audioGroup != null) audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
+    }
 
     /**
      * Answers a call. The attempt will be timed out if the call is not
@@ -207,10 +593,20 @@ public interface SipAudioCall {
      * {@code Listener.onError(SipAudioCall, SipErrorCode.TIME_OUT, String)}
      * will be called.
      *
-     * @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.
      * @see Listener.onError
+     * @throws SipException if the SIP service fails to answer the call
      */
-    void answerCall(int timeout) throws SipException;
+    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);
+        }
+    }
 
     /**
      * Continues a call that's on hold. When succeeds,
@@ -219,45 +615,189 @@ public interface SipAudioCall {
      * {@code Listener.onError(SipAudioCall, SipErrorCode.TIME_OUT, String)}
      * will be called.
      *
-     * @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.
      * @see Listener.onError
+     * @throws SipException if the SIP service fails to unhold the call
      */
-    void continueCall(int timeout) throws SipException;
+    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);
+    }
 
-    /** Puts the device to speaker mode. */
-    void setSpeakerMode(boolean speakerMode);
+    private SimpleSessionDescription createOffer() {
+        SimpleSessionDescription offer =
+                new SimpleSessionDescription(mSessionId, getLocalIp());
+        AudioCodec[] codecs = AudioCodec.getCodecs();
+        Media media = offer.newMedia(
+                "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
+        for (AudioCodec codec : AudioCodec.getCodecs()) {
+            media.setRtpPayload(codec.type, codec.rtpmap, codec.fmtp);
+        }
+        media.setRtpPayload(127, "telephone-event/8000", "0-15");
+        return offer;
+    }
 
-    /** Toggles mute. */
-    void toggleMute();
+    private SimpleSessionDescription createAnswer(String offerSd) {
+        SimpleSessionDescription offer =
+                new SimpleSessionDescription(offerSd);
+        SimpleSessionDescription answer =
+                new SimpleSessionDescription(mSessionId, getLocalIp());
+        AudioCodec codec = null;
+        for (Media media : offer.getMedia()) {
+            if ((codec == null) && (media.getPort() > 0)
+                    && "audio".equals(media.getType())
+                    && "RTP/AVP".equals(media.getProtocol())) {
+                // Find the first audio codec we supported.
+                for (int type : media.getRtpPayloadTypes()) {
+                    codec = AudioCodec.getCodec(type, media.getRtpmap(type),
+                            media.getFmtp(type));
+                    if (codec != null) {
+                        break;
+                    }
+                }
+                if (codec != null) {
+                    Media reply = answer.newMedia(
+                            "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
+                    reply.setRtpPayload(codec.type, codec.rtpmap, codec.fmtp);
 
-    /**
-     * Checks if the call is on hold.
-     *
-     * @return true if the call is on hold
-     */
-    boolean isOnHold();
+                    // Check if DTMF is supported in the same media.
+                    for (int type : media.getRtpPayloadTypes()) {
+                        String rtpmap = media.getRtpmap(type);
+                        if ((type != codec.type) && (rtpmap != null)
+                                && rtpmap.startsWith("telephone-event")) {
+                            reply.setRtpPayload(
+                                    type, rtpmap, media.getFmtp(type));
+                        }
+                    }
+
+                    // Handle recvonly and sendonly.
+                    if (media.getAttribute("recvonly") != null) {
+                        answer.setAttribute("sendonly", "");
+                    } else if(media.getAttribute("sendonly") != null) {
+                        answer.setAttribute("recvonly", "");
+                    } else if(offer.getAttribute("recvonly") != null) {
+                        answer.setAttribute("sendonly", "");
+                    } else if(offer.getAttribute("sendonly") != null) {
+                        answer.setAttribute("recvonly", "");
+                    }
+                    continue;
+                }
+            }
+            // Reject the media.
+            Media reply = answer.newMedia(
+                    media.getType(), 0, 1, media.getProtocol());
+            for (String format : media.getFormats()) {
+                reply.setFormat(format, null);
+            }
+        }
+        if (codec == null) {
+            throw new IllegalStateException("Reject SDP: no suitable codecs");
+        }
+        return answer;
+    }
+
+    private SimpleSessionDescription createHoldOffer() {
+        SimpleSessionDescription offer = createContinueOffer();
+        offer.setAttribute("sendonly", "");
+        return offer;
+    }
+
+    private SimpleSessionDescription createContinueOffer() {
+        SimpleSessionDescription offer =
+                new SimpleSessionDescription(mSessionId, getLocalIp());
+        Media media = offer.newMedia(
+                "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
+        AudioCodec codec = mAudioStream.getCodec();
+        media.setRtpPayload(codec.type, codec.rtpmap, codec.fmtp);
+        int dtmfType = mAudioStream.getDtmfType();
+        if (dtmfType != -1) {
+            media.setRtpPayload(dtmfType, "telephone-event/8000", "0-15");
+        }
+        return offer;
+    }
+
+    private void grabWifiHighPerfLock() {
+        if (mWifiHighPerfLock == null) {
+            Log.v(TAG, "acquire wifi high perf lock");
+            mWifiHighPerfLock = ((WifiManager)
+                    mContext.getSystemService(Context.WIFI_SERVICE))
+                    .createWifiLock(WifiManager.WIFI_MODE_FULL_HIGH_PERF, TAG);
+            mWifiHighPerfLock.acquire();
+        }
+    }
+
+    private void releaseWifiHighPerfLock() {
+        if (mWifiHighPerfLock != null) {
+            Log.v(TAG, "release wifi high perf lock");
+            mWifiHighPerfLock.release();
+            mWifiHighPerfLock = null;
+        }
+    }
+
+    private boolean isWifiOn() {
+        return (mWm.getConnectionInfo().getBSSID() == null) ? false : true;
+    }
+
+    /** Toggles mute. */
+    public synchronized void toggleMute() {
+        AudioGroup audioGroup = getAudioGroup();
+        if (audioGroup != null) {
+            audioGroup.setMode(
+                    mMuted ? AudioGroup.MODE_NORMAL : AudioGroup.MODE_MUTED);
+            mMuted = !mMuted;
+        }
+    }
 
     /**
      * Checks if the call is muted.
      *
      * @return true if the call is muted
      */
-    boolean isMuted();
+    public synchronized boolean isMuted() {
+        return mMuted;
+    }
+
+    /** Puts the device to speaker mode. */
+    public synchronized void setSpeakerMode(boolean speakerMode) {
+        ((AudioManager) mContext.getSystemService(Context.AUDIO_SERVICE))
+                .setSpeakerphoneOn(speakerMode);
+    }
 
     /**
-     * Sends a DTMF code.
+     * Sends a DTMF code. According to RFC2833, event 0--9 maps to decimal
+     * value 0--9, '*' to 10, '#' to 11, event 'A'--'D' to 12--15, and event
+     * flash to 16. Currently, event flash is not supported.
      *
-     * @param code the DTMF code to send
+     * @param code the DTMF code to send. Value 0 to 15 (inclusive) are valid
+     *        inputs.
+     * @see http://tools.ietf.org/html/rfc2833
      */
-    void sendDtmf(int code);
+    public void sendDtmf(int code) {
+        sendDtmf(code, null);
+    }
 
     /**
-     * Sends a DTMF code.
+     * Sends a DTMF code. According to RFC2833, event 0--9 maps to decimal
+     * value 0--9, '*' to 10, '#' to 11, event 'A'--'D' to 12--15, and event
+     * flash to 16. Currently, event flash is not supported.
      *
-     * @param code the DTMF code to send
+     * @param code the DTMF code to send. Value 0 to 15 (inclusive) are valid
+     *        inputs.
      * @param result the result message to send when done
      */
-    void sendDtmf(int code, Message result);
+    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);
+        }
+        if (result != null) result.sendToTarget();
+    }
 
     /**
      * Gets the {@link AudioStream} object used in this call. The object
@@ -268,8 +808,11 @@ public interface SipAudioCall {
      *
      * @return the {@link AudioStream} object or null if the RTP stream has not
      *      yet been set up
+     * @hide
      */
-    AudioStream getAudioStream();
+    public synchronized AudioStream getAudioStream() {
+        return mAudioStream;
+    }
 
     /**
      * Gets the {@link AudioGroup} object which the {@link AudioStream} object
@@ -283,8 +826,12 @@ public interface SipAudioCall {
      * @return the {@link AudioGroup} object or null if the RTP stream has not
      *      yet been set up
      * @see #getAudioStream
+     * @hide
      */
-    AudioGroup getAudioGroup();
+    public synchronized AudioGroup getAudioGroup() {
+        if (mAudioGroup != null) return mAudioGroup;
+        return ((mAudioStream == null) ? null : mAudioStream.getGroup());
+    }
 
     /**
      * Sets the {@link AudioGroup} object which the {@link AudioStream} object
@@ -292,56 +839,214 @@ public interface SipAudioCall {
      * will be dynamically created when needed.
      *
      * @see #getAudioStream
+     * @hide
      */
-    void setAudioGroup(AudioGroup audioGroup);
+    public synchronized void setAudioGroup(AudioGroup group) {
+        if ((mAudioStream != null) && (mAudioStream.getGroup() != null)) {
+            mAudioStream.join(group);
+        }
+        mAudioGroup = group;
+    }
 
     /**
-     * Checks if the call is established.
-     *
-     * @return true if the call is established
+     * Starts the audio for the established call. This method should be called
+     * after {@link Listener#onCallEstablished} is called.
      */
-    boolean isInCall();
+    public void startAudio() {
+        try {
+            startAudioInternal();
+        } catch (UnknownHostException e) {
+            onError(SipErrorCode.PEER_NOT_REACHABLE, e.getMessage());
+        } catch (Throwable e) {
+            onError(SipErrorCode.CLIENT_ERROR, e.getMessage());
+        }
+    }
 
-    /**
-     * Gets the local SIP profile.
-     *
-     * @return the local SIP profile
-     */
-    SipProfile getLocalProfile();
+    private synchronized void startAudioInternal() throws UnknownHostException {
+        if (mPeerSd == null) {
+            Log.v(TAG, "startAudioInternal() mPeerSd = null");
+            throw new IllegalStateException("mPeerSd = null");
+        }
 
-    /**
-     * Gets the peer's SIP profile.
-     *
-     * @return the peer's SIP profile
-     */
-    SipProfile getPeerProfile();
+        stopCall(DONT_RELEASE_SOCKET);
+        mInCall = true;
 
-    /**
-     * Gets the state of the {@link ISipSession} that carries this call.
-     * The value returned must be one of the states in {@link SipSessionState}.
-     *
-     * @return the session state
-     */
-    int getState();
+        // Run exact the same logic in createAnswer() to setup mAudioStream.
+        SimpleSessionDescription offer =
+                new SimpleSessionDescription(mPeerSd);
+        AudioStream stream = mAudioStream;
+        AudioCodec codec = null;
+        for (Media media : offer.getMedia()) {
+            if ((codec == null) && (media.getPort() > 0)
+                    && "audio".equals(media.getType())
+                    && "RTP/AVP".equals(media.getProtocol())) {
+                // Find the first audio codec we supported.
+                for (int type : media.getRtpPayloadTypes()) {
+                    codec = AudioCodec.getCodec(
+                            type, media.getRtpmap(type), media.getFmtp(type));
+                    if (codec != null) {
+                        break;
+                    }
+                }
+
+                if (codec != null) {
+                    // Associate with the remote host.
+                    String address = media.getAddress();
+                    if (address == null) {
+                        address = offer.getAddress();
+                    }
+                    stream.associate(InetAddress.getByName(address),
+                            media.getPort());
+
+                    stream.setDtmfType(-1);
+                    stream.setCodec(codec);
+                    // Check if DTMF is supported in the same media.
+                    for (int type : media.getRtpPayloadTypes()) {
+                        String rtpmap = media.getRtpmap(type);
+                        if ((type != codec.type) && (rtpmap != null)
+                                && rtpmap.startsWith("telephone-event")) {
+                            stream.setDtmfType(type);
+                        }
+                    }
+
+                    // Handle recvonly and sendonly.
+                    if (mHold) {
+                        stream.setMode(RtpStream.MODE_NORMAL);
+                    } else if (media.getAttribute("recvonly") != null) {
+                        stream.setMode(RtpStream.MODE_SEND_ONLY);
+                    } else if(media.getAttribute("sendonly") != null) {
+                        stream.setMode(RtpStream.MODE_RECEIVE_ONLY);
+                    } else if(offer.getAttribute("recvonly") != null) {
+                        stream.setMode(RtpStream.MODE_SEND_ONLY);
+                    } else if(offer.getAttribute("sendonly") != null) {
+                        stream.setMode(RtpStream.MODE_RECEIVE_ONLY);
+                    } else {
+                        stream.setMode(RtpStream.MODE_NORMAL);
+                    }
+                    break;
+                }
+            }
+        }
+        if (codec == null) {
+            throw new IllegalStateException("Reject SDP: no suitable codecs");
+        }
+
+        if (isWifiOn()) grabWifiHighPerfLock();
+
+        if (!mHold) {
+            /* The recorder volume will be very low if the device is in
+             * IN_CALL mode. Therefore, we have to set the mode to NORMAL
+             * in order to have the normal microphone level.
+             */
+            ((AudioManager) mContext.getSystemService
+                    (Context.AUDIO_SERVICE))
+                    .setMode(AudioManager.MODE_NORMAL);
+        }
+
+        // AudioGroup logic:
+        AudioGroup audioGroup = getAudioGroup();
+        if (mHold) {
+            if (audioGroup != null) {
+                audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
+            }
+            // don't create an AudioGroup here; doing so will fail if
+            // there's another AudioGroup out there that's active
+        } else {
+            if (audioGroup == null) audioGroup = new AudioGroup();
+            stream.join(audioGroup);
+            if (mMuted) {
+                audioGroup.setMode(AudioGroup.MODE_MUTED);
+            } else {
+                audioGroup.setMode(AudioGroup.MODE_NORMAL);
+            }
+        }
+    }
+
+    private void stopCall(boolean releaseSocket) {
+        Log.d(TAG, "stop audiocall");
+        releaseWifiHighPerfLock();
+        if (mAudioStream != null) {
+            mAudioStream.join(null);
+
+            if (releaseSocket) {
+                mAudioStream.release();
+                mAudioStream = null;
+            }
+        }
+    }
+
+    private String getLocalIp() {
+        return mSipSession.getLocalIp();
+    }
 
-    /**
-     * Gets the {@link ISipSession} that carries this call.
-     *
-     * @return the session object that carries this call
-     */
-    ISipSession getSipSession();
 
     /**
      * Enables/disables the ring-back tone.
      *
      * @param enabled true to enable; false to disable
      */
-    void setRingbackToneEnabled(boolean enabled);
+    public synchronized void setRingbackToneEnabled(boolean enabled) {
+        mRingbackToneEnabled = enabled;
+    }
 
     /**
      * Enables/disables the ring tone.
      *
      * @param enabled true to enable; false to disable
      */
-    void setRingtoneEnabled(boolean enabled);
+    public synchronized void setRingtoneEnabled(boolean enabled) {
+        mRingtoneEnabled = enabled;
+    }
+
+    private void startRingbackTone() {
+        if (!mRingbackToneEnabled) return;
+        if (mRingbackTone == null) {
+            // The volume relative to other sounds in the stream
+            int toneVolume = 80;
+            mRingbackTone = new ToneGenerator(
+                    AudioManager.STREAM_VOICE_CALL, toneVolume);
+        }
+        mRingbackTone.startTone(ToneGenerator.TONE_CDMA_LOW_PBX_L);
+    }
+
+    private void stopRingbackTone() {
+        if (mRingbackTone != null) {
+            mRingbackTone.stopTone();
+            mRingbackTone.release();
+            mRingbackTone = null;
+        }
+    }
+
+    private void startRinging() {
+        if (!mRingtoneEnabled) return;
+        ((Vibrator) mContext.getSystemService(Context.VIBRATOR_SERVICE))
+                .vibrate(new long[] {0, 1000, 1000}, 1);
+        AudioManager am = (AudioManager)
+                mContext.getSystemService(Context.AUDIO_SERVICE);
+        if (am.getStreamVolume(AudioManager.STREAM_RING) > 0) {
+            String ringtoneUri =
+                    Settings.System.DEFAULT_RINGTONE_URI.toString();
+            mRingtone = RingtoneManager.getRingtone(mContext,
+                    Uri.parse(ringtoneUri));
+            mRingtone.play();
+        }
+    }
+
+    private void stopRinging() {
+        ((Vibrator) mContext.getSystemService(Context.VIBRATOR_SERVICE))
+                .cancel();
+        if (mRingtone != null) mRingtone.stop();
+    }
+
+    private void throwSipException(Throwable throwable) throws SipException {
+        if (throwable instanceof SipException) {
+            throw (SipException) throwable;
+        } else {
+            throw new SipException("", throwable);
+        }
+    }
+
+    private SipProfile getPeerProfile(SipSession session) {
+        return session.getPeerProfile();
+    }
 }