Requires permissions to use this class:
* {@link android.Manifest.permission#INTERNET} and
* {@link android.Manifest.permission#USE_SIP}.
*
Requires permissions to {@link #startAudio}:
* {@link android.Manifest.permission#RECORD_AUDIO},
* {@link android.Manifest.permission#ACCESS_WIFI_STATE} and
* {@link android.Manifest.permission#WAKE_LOCK}.
*
Requires permissions to {@link #setSpeakerMode}:
* {@link android.Manifest.permission#MODIFY_AUDIO_SETTINGS}.
*/
public class SipAudioCall {
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 static class Listener {
/**
* Called when the call object is ready to make another call.
* The default implementation calls {@link #onChanged}.
*
* @param call the call object that is ready to make another 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 #onChanged}.
*
* @param call the call object that carries out the audio call
*/
public void onCalling(SipAudioCall call) {
onChanged(call);
}
/**
* Called when a new call comes in.
* 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
*/
public void onRinging(SipAudioCall call, SipProfile caller) {
onChanged(call);
}
/**
* Called when a RINGING response is received for the INVITE request
* sent. The default implementation calls {@link #onChanged}.
*
* @param call the call object that carries out the audio call
*/
public void onRingingBack(SipAudioCall call) {
onChanged(call);
}
/**
* Called when the session is established.
* The default implementation calls {@link #onChanged}.
*
* @param call the call object that carries out the audio call
*/
public void onCallEstablished(SipAudioCall call) {
onChanged(call);
}
/**
* Called when the session is terminated.
* The default implementation calls {@link #onChanged}.
*
* @param call the call object that carries out the audio call
*/
public void onCallEnded(SipAudioCall call) {
onChanged(call);
}
/**
* Called when the peer is busy during session initialization.
* The default implementation calls {@link #onChanged}.
*
* @param call the call object that carries out the audio call
*/
public void onCallBusy(SipAudioCall call) {
onChanged(call);
}
/**
* Called when the call is on hold.
* The default implementation calls {@link #onChanged}.
*
* @param call the call object that carries out the audio call
*/
public void onCallHeld(SipAudioCall call) {
onChanged(call);
}
/**
* 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
*/
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 SipProfile mPendingCallRequest;
private WifiManager mWm;
private WifiManager.WifiLock mWifiHighPerfLock;
private int mErrorCode = SipErrorCode.NO_ERROR;
private String mErrorMessage;
/**
* Creates a call object with the local SIP profile.
* @param context the context for accessing system services such as
* ringtone, audio, WIFI etc
*/
public SipAudioCall(Context context, SipProfile localProfile) {
mContext = context;
mLocalProfile = localProfile;
mWm = (WifiManager) context.getSystemService(Context.WIFI_SERVICE);
}
/**
* Sets the listener to listen to the audio call events. The method calls
* {@code setListener(listener, false)}.
*
* @param listener to listen to the audio call events of this object
* @see #setListener(Listener, boolean)
*/
public void setListener(SipAudioCall.Listener listener) {
setListener(listener, false);
}
/**
* Sets the listener to listen to the audio call events. A
* {@link SipAudioCall} can only hold one listener at a time. Subsequent
* calls to this method override the previous listener.
*
* @param listener to listen to the audio call events of this object
* @param callbackImmediately set to true if the caller wants to be called
* back immediately on the current state
*/
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 boolean isInCall() {
synchronized (this) {
return mInCall;
}
}
/**
* Checks if the call is on hold.
*
* @return true if the call is on hold
*/
public boolean isOnHold() {
synchronized (this) {
return mHold;
}
}
/**
* Closes this object. This object is not usable after being closed.
*/
public void close() {
close(true);
}
private synchronized void close(boolean closeRtp) {
if (closeRtp) stopCall(RELEASE_SOCKET);
mInCall = false;
mHold = false;
mSessionId = System.currentTimeMillis();
mErrorCode = SipErrorCode.NO_ERROR;
mErrorMessage = null;
if (mSipSession != null) {
mSipSession.setListener(null);
mSipSession = null;
}
}
/**
* Gets the local SIP profile.
*
* @return the local SIP profile
*/
public SipProfile getLocalProfile() {
synchronized (this) {
return mLocalProfile;
}
}
/**
* Gets the peer's SIP profile.
*
* @return the peer's SIP profile
*/
public SipProfile getPeerProfile() {
synchronized (this) {
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 int getState() {
synchronized (this) {
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 SipSession getSipSession() {
synchronized (this) {
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);
Listener listener = mListener;
if (listener != null) {
try {
listener.onRingingBack(SipAudioCall.this);
} catch (Throwable t) {
Log.i(TAG, "onRingingBack(): " + t);
}
}
}
@Override
public void onRinging(SipSession session,
SipProfile peerProfile, String sessionDescription) {
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();
}
}
}
@Override
public void onCallEstablished(SipSession session,
String sessionDescription) {
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 void attachCall(SipSession session, String sessionDescription)
throws SipException {
synchronized (this) {
mSipSession = session;
mPeerSd = sessionDescription;
Log.v(TAG, "attachCall()" + mPeerSd);
try {
session.setListener(createListener());
} 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 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.
* @see Listener.onError
* @throws SipException if the SIP service fails to create a session for the
* call
*/
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);
}
}
}
/**
* Ends a call.
* @throws SipException if the SIP service fails to end the call
*/
public void endCall() throws SipException {
synchronized (this) {
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
* called. 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 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
*/
public void holdCall(int timeout) throws SipException {
synchronized (this) {
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
* established within {@code timeout} seconds and
* {@code Listener.onError(SipAudioCall, SipErrorCode.TIME_OUT, String)}
* will be called.
*
* @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
*/
public void answerCall(int timeout) throws SipException {
synchronized (this) {
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,
* {@link Listener#onCallEstablished} is called. 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 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
*/
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() {
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;
}
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);
// 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() {
/* not available in master yet
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 void toggleMute() {
synchronized (this) {
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
*/
public boolean isMuted() {
synchronized (this) {
return mMuted;
}
}
/**
* Puts the device to speaker mode.
*
Requires permission: * {@link android.Manifest.permission#MODIFY_AUDIO_SETTINGS}. */ public void setSpeakerMode(boolean speakerMode) { synchronized (this) { ((AudioManager) mContext.getSystemService(Context.AUDIO_SERVICE)) .setSpeakerphoneOn(speakerMode); } } /** * 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. Value 0 to 15 (inclusive) are valid * inputs. * @see http://tools.ietf.org/html/rfc2833 */ public void sendDtmf(int code) { sendDtmf(code, null); } /** * 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. Value 0 to 15 (inclusive) are valid * inputs. * @param result the result message to send when done */ 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(); } } /** * 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 * @hide */ public AudioStream getAudioStream() { synchronized (this) { return mAudioStream; } } /** * 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. If a group object is set by * {@link #setAudioGroup(AudioGroup)}, then this method returns that object. * * @return the {@link AudioGroup} object or null if the RTP stream has not * yet been set up * @see #getAudioStream * @hide */ public AudioGroup getAudioGroup() { synchronized (this) { if (mAudioGroup != null) return mAudioGroup; return ((mAudioStream == null) ? null : mAudioStream.getGroup()); } } /** * Sets the {@link AudioGroup} object which the {@link AudioStream} object * joins. If {@code audioGroup} is null, then the {@code AudioGroup} object * will be dynamically created when needed. * * @see #getAudioStream * @hide */ public void setAudioGroup(AudioGroup group) { synchronized (this) { if ((mAudioStream != null) && (mAudioStream.getGroup() != null)) { mAudioStream.join(group); } mAudioGroup = group; } } /** * Starts the audio for the established call. This method should be called * after {@link Listener#onCallEstablished} is called. *
Requires permission: * {@link android.Manifest.permission#RECORD_AUDIO}, * {@link android.Manifest.permission#ACCESS_WIFI_STATE} and * {@link android.Manifest.permission#WAKE_LOCK}. */ public void startAudio() { try { startAudioInternal(); } catch (UnknownHostException e) { onError(SipErrorCode.PEER_NOT_REACHABLE, e.getMessage()); } catch (Throwable e) { onError(SipErrorCode.CLIENT_ERROR, e.getMessage()); } } private synchronized void startAudioInternal() throws UnknownHostException { if (mPeerSd == null) { Log.v(TAG, "startAudioInternal() mPeerSd = null"); throw new IllegalStateException("mPeerSd = null"); } stopCall(DONT_RELEASE_SOCKET); mInCall = true; // 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(); } 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(); } }