am 08faac3c: Unhide SIP API. (ea445758) · Commits · e / os / android_frameworks_base

api/current.xml

+1950 −0

File changed.

Preview size limit exceeded, changes collapsed.

core/res/AndroidManifest.xml

+7 −0

Original line number	Diff line number	Diff line
		@@ -364,6 +364,13 @@
		android:description="@string/permdesc_nfcLlcp"
		android:label="@string/permlab_nfcLlcp" />

		<!-- Allows an application to use SIP service -->
		<permission android:name="android.permission.USE_SIP"
		android:permissionGroup="android.permission-group.NETWORK"
		android:protectionLevel="dangerous"
		android:description="@string/permdesc_use_sip"
		android:label="@string/permlab_use_sip" />

		<!-- Allows applications to call into AccountAuthenticators. Only
		the system can get this permission. -->
		<permission android:name="android.permission.ACCOUNT_MANAGER"

core/res/res/values/strings.xml

+5 −0

Original line number	Diff line number	Diff line
		@@ -1240,6 +1240,11 @@
		<!-- Description of an application permission, listed so the user can choose whether they want to allow the application to do this. -->
		<string name="permdesc_cache_filesystem">Allows an application to read and write the cache filesystem.</string>

		<!-- Title of an application permission, listed so the user can choose whether they want to allow the application to do this. -->
		<string name="permlab_use_sip">make/receive Internet calls</string>
		<!-- Description of an application permission, listed so the user can choose whether they want to allow the application to do this. -->
		<string name="permdesc_use_sip">Allows an application to use the SIP service to make/receive Internet calls.</string>

		<!-- Policy administration -->

		<!-- Title of policy access to limiting the user's password choices -->

voip/java/android/net/sip/SipAudioCall.java

+166 −113

Original line number	Diff line number	Diff line
		@@ -43,9 +43,11 @@ import java.util.List;
		import java.util.Map;

		/**
		* Class that handles an audio call over SIP.
		* Class that handles an Internet audio call over SIP. {@link SipManager}
		* facilitates instantiating a {@code SipAudioCall} object for making/receiving
		* calls. See {@link SipManager#makeAudioCall} and
		* {@link SipManager#takeAudioCall}.
		*/
		/** @hide */
		public class SipAudioCall {
		private static final String TAG = SipAudioCall.class.getSimpleName();
		private static final boolean RELEASE_SOCKET = true;
		@@ -56,7 +58,7 @@ public class SipAudioCall {
		public static class Listener {
		/**
		* Called when the call object is ready to make another call.
		* The default implementation calls {@link #onChange}.
		* The default implementation calls {@link #onChanged}.
		*
		* @param call the call object that is ready to make another call
		*/
		@@ -66,7 +68,7 @@ public class SipAudioCall {

		/**
		* Called when a request is sent out to initiate a new call.
		* The default implementation calls {@link #onChange}.
		* The default implementation calls {@link #onChanged}.
		*
		* @param call the call object that carries out the audio call
		*/
		@@ -76,7 +78,7 @@ public class SipAudioCall {

		/**
		* Called when a new call comes in.
		* The default implementation calls {@link #onChange}.
		* The default implementation calls {@link #onChanged}.
		*
		* @param call the call object that carries out the audio call
		* @param caller the SIP profile of the caller
		@@ -87,7 +89,7 @@ public class SipAudioCall {

		/**
		* Called when a RINGING response is received for the INVITE request
		* sent. The default implementation calls {@link #onChange}.
		* sent. The default implementation calls {@link #onChanged}.
		*
		* @param call the call object that carries out the audio call
		*/
		@@ -97,7 +99,7 @@ public class SipAudioCall {

		/**
		* Called when the session is established.
		* The default implementation calls {@link #onChange}.
		* The default implementation calls {@link #onChanged}.
		*
		* @param call the call object that carries out the audio call
		*/
		@@ -107,7 +109,7 @@ public class SipAudioCall {

		/**
		* Called when the session is terminated.
		* The default implementation calls {@link #onChange}.
		* The default implementation calls {@link #onChanged}.
		*
		* @param call the call object that carries out the audio call
		*/
		@@ -117,7 +119,7 @@ public class SipAudioCall {

		/**
		* Called when the peer is busy during session initialization.
		* The default implementation calls {@link #onChange}.
		* The default implementation calls {@link #onChanged}.
		*
		* @param call the call object that carries out the audio call
		*/
		@@ -127,7 +129,7 @@ public class SipAudioCall {

		/**
		* Called when the call is on hold.
		* The default implementation calls {@link #onChange}.
		* The default implementation calls {@link #onChanged}.
		*
		* @param call the call object that carries out the audio call
		*/
		@@ -257,18 +259,22 @@ public class SipAudioCall {
		*
		* @return true if the call is established
		*/
		public synchronized boolean isInCall() {
		public boolean isInCall() {
		synchronized (this) {
		return mInCall;
		}
		}

		/**
		* Checks if the call is on hold.
		*
		* @return true if the call is on hold
		*/
		public synchronized boolean isOnHold() {
		public boolean isOnHold() {
		synchronized (this) {
		return mHold;
		}
		}

		/**
		* Closes this object. This object is not usable after being closed.
		@@ -299,18 +305,22 @@ public class SipAudioCall {
		*
		* @return the local SIP profile
		*/
		public synchronized SipProfile getLocalProfile() {
		public SipProfile getLocalProfile() {
		synchronized (this) {
		return mLocalProfile;
		}
		}

		/**
		* Gets the peer's SIP profile.
		*
		* @return the peer's SIP profile
		*/
		public synchronized SipProfile getPeerProfile() {
		public SipProfile getPeerProfile() {
		synchronized (this) {
		return (mSipSession == null) ? null : mSipSession.getPeerProfile();
		}
		}

		/**
		* Gets the state of the {@link SipSession} that carries this call.
		@@ -318,10 +328,12 @@ public class SipAudioCall {
		*
		* @return the session state
		*/
		public synchronized int getState() {
		public int getState() {
		synchronized (this) {
		if (mSipSession == null) return SipSession.State.READY_TO_CALL;
		return mSipSession.getState();
		}
		}


		/**
		@@ -330,9 +342,11 @@ public class SipAudioCall {
		* @return the session object that carries this call
		* @hide
		*/
		public synchronized SipSession getSipSession() {
		public SipSession getSipSession() {
		synchronized (this) {
		return mSipSession;
		}
		}

		private SipSession.Listener createListener() {
		return new SipSession.Listener() {
		@@ -364,10 +378,12 @@ public class SipAudioCall {
		}

		@Override
		public synchronized void onRinging(SipSession session,
		public void onRinging(SipSession session,
		SipProfile peerProfile, String sessionDescription) {
		synchronized (SipAudioCall.this) {
		if ((mSipSession == null) \|\| !mInCall
		\|\| !session.getCallId().equals(mSipSession.getCallId())) {
		\|\| !session.getCallId().equals(
		mSipSession.getCallId())) {
		// should not happen
		session.endCall();
		return;
		@@ -382,6 +398,7 @@ public class SipAudioCall {
		session.endCall();
		}
		}
		}

		@Override
		public void onCallEstablished(SipSession session,
		@@ -508,20 +525,24 @@ public class SipAudioCall {
		* @throws SipException if the SIP service fails to attach this object to
		* the session
		*/
		public synchronized void attachCall(SipSession session,
		String sessionDescription) throws SipException {
		public void attachCall(SipSession session, String sessionDescription)
		throws SipException {
		synchronized (this) {
		mSipSession = session;
		mPeerSd = sessionDescription;
		Log.v(TAG, "attachCall()" + mPeerSd);
		try {
		session.setListener(createListener());

		if (getState() == SipSession.State.INCOMING_CALL) startRinging();
		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
		@@ -529,7 +550,7 @@ public class SipAudioCall {
		* and {@code Listener.onError(SipAudioCall, SipErrorCode.TIME_OUT, String)}
		* will be called.
		*
		* @param callee the SIP profile to make the call to
		* @param peerProfile the SIP profile to make the call to
		* @param sipSession the {@link SipSession} for carrying out the call
		* @param timeout the timeout value in seconds. Default value (defined by
		* SIP protocol) is used if {@code timeout} is zero or negative.
		@@ -537,23 +558,28 @@ public class SipAudioCall {
		* @throws SipException if the SIP service fails to create a session for the
		* call
		*/
		public synchronized void makeCall(SipProfile peerProfile,
		SipSession sipSession, int timeout) throws SipException {
		public void makeCall(SipProfile peerProfile, SipSession sipSession,
		int timeout) throws SipException {
		synchronized (this) {
		mSipSession = sipSession;
		try {
		mAudioStream = new AudioStream(InetAddress.getByName(getLocalIp()));
		mAudioStream = new AudioStream(InetAddress.getByName(
		getLocalIp()));
		sipSession.setListener(createListener());
		sipSession.makeCall(peerProfile, createOffer().encode(), timeout);
		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 synchronized void endCall() throws SipException {
		public void endCall() throws SipException {
		synchronized (this) {
		stopRinging();
		stopCall(RELEASE_SOCKET);
		mInCall = false;
		@@ -561,6 +587,7 @@ public class SipAudioCall {
		// 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
		@@ -574,7 +601,8 @@ public class SipAudioCall {
		* @see Listener.onError
		* @throws SipException if the SIP service fails to hold the call
		*/
		public synchronized void holdCall(int timeout) throws SipException {
		public void holdCall(int timeout) throws SipException {
		synchronized (this) {
		if (mHold) return;
		mSipSession.changeCall(createHoldOffer().encode(), timeout);
		mHold = true;
		@@ -582,6 +610,7 @@ public class SipAudioCall {
		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
		@@ -594,15 +623,18 @@ public class SipAudioCall {
		* @see Listener.onError
		* @throws SipException if the SIP service fails to answer the call
		*/
		public synchronized void answerCall(int timeout) throws SipException {
		public void answerCall(int timeout) throws SipException {
		synchronized (this) {
		stopRinging();
		try {
		mAudioStream = new AudioStream(InetAddress.getByName(getLocalIp()));
		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,
		@@ -616,13 +648,15 @@ public class SipAudioCall {
		* @see Listener.onError
		* @throws SipException if the SIP service fails to unhold the call
		*/
		public synchronized void continueCall(int timeout) throws SipException {
		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 =
		@@ -739,29 +773,36 @@ public class SipAudioCall {
		}

		/** Toggles mute. */
		public synchronized void toggleMute() {
		public void toggleMute() {
		synchronized (this) {
		AudioGroup audioGroup = getAudioGroup();
		if (audioGroup != null) {
		audioGroup.setMode(
		mMuted ? AudioGroup.MODE_NORMAL : AudioGroup.MODE_MUTED);
		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 synchronized boolean isMuted() {
		public boolean isMuted() {
		synchronized (this) {
		return mMuted;
		}
		}

		/** Puts the device to speaker mode. */
		public synchronized void setSpeakerMode(boolean speakerMode) {
		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
		@@ -785,7 +826,8 @@ public class SipAudioCall {
		* inputs.
		* @param result the result message to send when done
		*/
		public synchronized void sendDtmf(int code, Message result) {
		public void sendDtmf(int code, Message result) {
		synchronized (this) {
		AudioGroup audioGroup = getAudioGroup();
		if ((audioGroup != null) && (mSipSession != null)
		&& (SipSession.State.IN_CALL == getState())) {
		@@ -794,6 +836,7 @@ public class SipAudioCall {
		}
		if (result != null) result.sendToTarget();
		}
		}

		/**
		* Gets the {@link AudioStream} object used in this call. The object
		@@ -806,9 +849,11 @@ public class SipAudioCall {
		* yet been set up
		* @hide
		*/
		public synchronized AudioStream getAudioStream() {
		public AudioStream getAudioStream() {
		synchronized (this) {
		return mAudioStream;
		}
		}

		/**
		* Gets the {@link AudioGroup} object which the {@link AudioStream} object
		@@ -824,10 +869,12 @@ public class SipAudioCall {
		* @see #getAudioStream
		* @hide
		*/
		public synchronized AudioGroup getAudioGroup() {
		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
		@@ -837,12 +884,14 @@ public class SipAudioCall {
		* @see #getAudioStream
		* @hide
		*/
		public synchronized void setAudioGroup(AudioGroup group) {
		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
		@@ -981,18 +1030,22 @@ public class SipAudioCall {
		*
		* @param enabled true to enable; false to disable
		*/
		public synchronized void setRingbackToneEnabled(boolean enabled) {
		public void setRingbackToneEnabled(boolean enabled) {
		synchronized (this) {
		mRingbackToneEnabled = enabled;
		}
		}

		/**
		* Enables/disables the ring tone.
		*
		* @param enabled true to enable; false to disable
		*/
		public synchronized void setRingtoneEnabled(boolean enabled) {
		public void setRingtoneEnabled(boolean enabled) {
		synchronized (this) {
		mRingtoneEnabled = enabled;
		}
		}

		private void startRingbackTone() {
		if (!mRingbackToneEnabled) return;

voip/java/android/net/sip/SipErrorCode.java

+3 −4

Original line number	Diff line number	Diff line
		@@ -19,10 +19,9 @@ package android.net.sip;
		/**
		* Defines error code returned in
		* {@link SipRegistrationListener#onRegistrationFailed},
		* {@link ISipSessionListener#onError},
		* {@link ISipSessionListener#onCallChangeFailed} and
		* {@link ISipSessionListener#onRegistrationFailed}.
		* @hide
		* {@link SipSession.Listener#onError},
		* {@link SipSession.Listener#onCallChangeFailed} and
		* {@link SipSession.Listener#onRegistrationFailed}.
		*/
		public class SipErrorCode {
		/** Not an error. */