/* * Copyright (C) 2010 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ package android.net.sip; import android.content.Context; import android.media.AudioManager; 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.telephony.Rlog; import android.text.TextUtils; import java.io.IOException; import java.net.InetAddress; import java.net.UnknownHostException; /** * Handles an Internet audio call over SIP. You can instantiate this class with {@link SipManager}, * using {@link SipManager#makeAudioCall makeAudioCall()} and {@link SipManager#takeAudioCall * takeAudioCall()}. * *

Note: Using this class require the * {@link android.Manifest.permission#INTERNET} and * {@link android.Manifest.permission#USE_SIP} permissions. In addition, {@link * #startAudio} requires the * {@link android.Manifest.permission#RECORD_AUDIO}, * {@link android.Manifest.permission#ACCESS_WIFI_STATE}, and * {@link android.Manifest.permission#WAKE_LOCK} permissions; and {@link #setSpeakerMode * setSpeakerMode()} requires the * {@link android.Manifest.permission#MODIFY_AUDIO_SETTINGS} permission.

* *
*

Developer Guides

*

For more information about using SIP, read the * Session Initiation Protocol * developer guide.

*
*/ public class SipAudioCall { private static final String LOG_TAG = SipAudioCall.class.getSimpleName(); private static final boolean DBG = true; private static final boolean RELEASE_SOCKET = true; private static final boolean DONT_RELEASE_SOCKET = false; private static final int SESSION_TIMEOUT = 5; // in seconds private static final int TRANSFER_TIMEOUT = 15; // in seconds /** Listener for events relating to a SIP call, such as when a call is being * recieved ("on ringing") or a call is outgoing ("on calling"). *

Many of these events are also received by {@link SipSession.Listener}.

*/ 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 SipSession mTransferringSession; 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 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 * {@link #setListener 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) { loge("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 synchronized void transferToNewSession() { if (mTransferringSession == null) return; SipSession origin = mSipSession; mSipSession = mTransferringSession; mTransferringSession = null; // stop the replaced call. if (mAudioStream != null) { mAudioStream.join(null); } else { try { mAudioStream = new AudioStream(InetAddress.getByName( getLocalIp())); } catch (Throwable t) { loge("transferToNewSession():", t); } } if (origin != null) origin.endCall(); startAudio(); } private SipSession.Listener createListener() { return new SipSession.Listener() { @Override public void onCalling(SipSession session) { if (DBG) log("onCalling: session=" + session); Listener listener = mListener; if (listener != null) { try { listener.onCalling(SipAudioCall.this); } catch (Throwable t) { loge("onCalling():", t); } } } @Override public void onRingingBack(SipSession session) { if (DBG) log("onRingingBackk: " + session); Listener listener = mListener; if (listener != null) { try { listener.onRingingBack(SipAudioCall.this); } catch (Throwable t) { loge("onRingingBack():", t); } } } @Override public void onRinging(SipSession session, SipProfile peerProfile, String sessionDescription) { // this callback is triggered only for reinvite. 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) { loge("onRinging():", e); session.endCall(); } } } @Override public void onCallEstablished(SipSession session, String sessionDescription) { mPeerSd = sessionDescription; if (DBG) log("onCallEstablished(): " + mPeerSd); // TODO: how to notify the UI that the remote party is changed if ((mTransferringSession != null) && (session == mTransferringSession)) { transferToNewSession(); return; } Listener listener = mListener; if (listener != null) { try { if (mHold) { listener.onCallHeld(SipAudioCall.this); } else { listener.onCallEstablished(SipAudioCall.this); } } catch (Throwable t) { loge("onCallEstablished(): ", t); } } } @Override public void onCallEnded(SipSession session) { if (DBG) log("onCallEnded: " + session + " mSipSession:" + mSipSession); // reset the trasnferring session if it is the one. if (session == mTransferringSession) { mTransferringSession = null; return; } // or ignore the event if the original session is being // transferred to the new one. if ((mTransferringSession != null) || (session != mSipSession)) return; Listener listener = mListener; if (listener != null) { try { listener.onCallEnded(SipAudioCall.this); } catch (Throwable t) { loge("onCallEnded(): ", t); } } close(); } @Override public void onCallBusy(SipSession session) { if (DBG) log("onCallBusy: " + session); Listener listener = mListener; if (listener != null) { try { listener.onCallBusy(SipAudioCall.this); } catch (Throwable t) { loge("onCallBusy(): ", t); } } close(false); } @Override public void onCallChangeFailed(SipSession session, int errorCode, String message) { if (DBG) log("onCallChangedFailed: " + message); mErrorCode = errorCode; mErrorMessage = message; Listener listener = mListener; if (listener != null) { try { listener.onError(SipAudioCall.this, mErrorCode, message); } catch (Throwable t) { loge("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 } @Override public void onCallTransferring(SipSession newSession, String sessionDescription) { if (DBG) log("onCallTransferring: mSipSession=" + mSipSession + " newSession=" + newSession); mTransferringSession = newSession; try { if (sessionDescription == null) { newSession.makeCall(newSession.getPeerProfile(), createOffer().encode(), TRANSFER_TIMEOUT); } else { String answer = createAnswer(sessionDescription).encode(); newSession.answerCall(answer, SESSION_TIMEOUT); } } catch (Throwable e) { loge("onCallTransferring()", e); newSession.endCall(); } } }; } private void onError(int errorCode, String message) { if (DBG) log("onError: " + SipErrorCode.toString(errorCode) + ": " + message); mErrorCode = errorCode; mErrorMessage = message; Listener listener = mListener; if (listener != null) { try { listener.onError(this, errorCode, message); } catch (Throwable t) { loge("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 or VOIP API is not supported by the device * @see SipManager#isVoipSupported */ public void attachCall(SipSession session, String sessionDescription) throws SipException { if (!SipManager.isVoipSupported(mContext)) { throw new SipException("VOIP API is not supported"); } synchronized (this) { mSipSession = session; mPeerSd = sessionDescription; if (DBG) log("attachCall(): " + mPeerSd); try { session.setListener(createListener()); } catch (Throwable e) { loge("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 {@link Listener#onError 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 or VOIP API is not supported by the device * @see SipManager#isVoipSupported */ public void makeCall(SipProfile peerProfile, SipSession sipSession, int timeout) throws SipException { if (DBG) log("makeCall: " + peerProfile + " session=" + sipSession + " timeout=" + timeout); if (!SipManager.isVoipSupported(mContext)) { throw new SipException("VOIP API is not supported"); } synchronized (this) { mSipSession = sipSession; try { mAudioStream = new AudioStream(InetAddress.getByName( getLocalIp())); sipSession.setListener(createListener()); sipSession.makeCall(peerProfile, createOffer().encode(), timeout); } catch (IOException e) { loge("makeCall:", 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 { if (DBG) log("endCall: mSipSession" + mSipSession); 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 * {@link Listener#onError 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 { if (DBG) log("holdCall: mSipSession" + mSipSession + " timeout=" + timeout); synchronized (this) { if (mHold) return; if (mSipSession == null) { loge("holdCall:"); throw new SipException("Not in a call to hold call"); } mSipSession.changeCall(createHoldOffer().encode(), timeout); mHold = true; setAudioGroupMode(); } } /** * Answers a call. The attempt will be timed out if the call is not * established within {@code timeout} seconds and * {@link Listener#onError 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 { if (DBG) log("answerCall: mSipSession" + mSipSession + " timeout=" + timeout); synchronized (this) { if (mSipSession == null) { throw new SipException("No call to answer"); } try { mAudioStream = new AudioStream(InetAddress.getByName( getLocalIp())); mSipSession.answerCall(createAnswer(mPeerSd).encode(), timeout); } catch (IOException e) { loge("answerCall:", 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 * {@link Listener#onError 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 { if (DBG) log("continueCall: mSipSession" + mSipSession + " timeout=" + timeout); synchronized (this) { if (!mHold) return; mSipSession.changeCall(createContinueOffer().encode(), timeout); mHold = false; setAudioGroupMode(); } } 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"); if (DBG) log("createOffer: offer=" + offer); return offer; } private SimpleSessionDescription createAnswer(String offerSd) { if (TextUtils.isEmpty(offerSd)) return createOffer(); 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) { loge("createAnswer: no suitable codes"); throw new IllegalStateException("Reject SDP: no suitable codecs"); } if (DBG) log("createAnswer: answer=" + answer); return answer; } private SimpleSessionDescription createHoldOffer() { SimpleSessionDescription offer = createContinueOffer(); offer.setAttribute("sendonly", ""); if (DBG) log("createHoldOffer: offer=" + offer); return offer; } private SimpleSessionDescription createContinueOffer() { if (DBG) log("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) { if (DBG) log("grabWifiHighPerfLock:"); mWifiHighPerfLock = ((WifiManager) mContext.getSystemService(Context.WIFI_SERVICE)) .createWifiLock(WifiManager.WIFI_MODE_FULL_HIGH_PERF, LOG_TAG); mWifiHighPerfLock.acquire(); } } private void releaseWifiHighPerfLock() { if (mWifiHighPerfLock != null) { if (DBG) log("releaseWifiHighPerfLock:"); mWifiHighPerfLock.release(); mWifiHighPerfLock = null; } } private boolean isWifiOn() { return (mWm.getConnectionInfo().getBSSID() == null) ? false : true; } /** Toggles mute. */ public void toggleMute() { synchronized (this) { mMuted = !mMuted; setAudioGroupMode(); } } /** * 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. *

Note: Requires the * {@link android.Manifest.permission#MODIFY_AUDIO_SETTINGS} permission.

* * @param speakerMode set true to enable speaker mode; false to disable */ public void setSpeakerMode(boolean speakerMode) { synchronized (this) { ((AudioManager) mContext.getSystemService(Context.AUDIO_SERVICE)) .setSpeakerphoneOn(speakerMode); setAudioGroupMode(); } } private boolean isSpeakerOn() { return ((AudioManager) mContext.getSystemService(Context.AUDIO_SERVICE)) .isSpeakerphoneOn(); } /** * Sends a DTMF code. According to RFC 2883, * 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. */ public void sendDtmf(int code) { sendDtmf(code, null); } /** * Sends a DTMF code. According to RFC 2883, * 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())) { if (DBG) log("sendDtmf: code=" + code + " result=" + result); 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. Note that the mode of the * {@code AudioGroup} is not changed according to the audio settings (i.e., * hold, mute, speaker phone) of this object. This is mainly used to merge * multiple {@code SipAudioCall} objects to form a conference call. The * settings of the first object (that merges others) override others'. * * @see #getAudioStream * @hide */ public void setAudioGroup(AudioGroup group) { synchronized (this) { if (DBG) log("setAudioGroup: group=" + group); 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. *

Note: Requires the * {@link android.Manifest.permission#RECORD_AUDIO}, * {@link android.Manifest.permission#ACCESS_WIFI_STATE} and * {@link android.Manifest.permission#WAKE_LOCK} permissions.

*/ 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 (DBG) loge("startAudioInternal: mPeerSd=" + mPeerSd); if (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(); // AudioGroup logic: AudioGroup audioGroup = getAudioGroup(); if (mHold) { // 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); } setAudioGroupMode(); } // set audio group mode based on current audio configuration private void setAudioGroupMode() { AudioGroup audioGroup = getAudioGroup(); if (DBG) log("setAudioGroupMode: audioGroup=" + audioGroup); if (audioGroup != null) { if (mHold) { audioGroup.setMode(AudioGroup.MODE_ON_HOLD); } else if (mMuted) { audioGroup.setMode(AudioGroup.MODE_MUTED); } else if (isSpeakerOn()) { audioGroup.setMode(AudioGroup.MODE_ECHO_SUPPRESSION); } else { audioGroup.setMode(AudioGroup.MODE_NORMAL); } } } private void stopCall(boolean releaseSocket) { if (DBG) log("stopCall: releaseSocket=" + releaseSocket); 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 void log(String s) { Rlog.d(LOG_TAG, s); } private void loge(String s) { Rlog.e(LOG_TAG, s); } private void loge(String s, Throwable t) { Rlog.e(LOG_TAG, s, t); } }