protected void setupSIP(Context context, SIP5060ProvisioningRequest req) throws IOException {
AppProperties props = new AppProperties(context);
// Setup the SIP preferences
SharedPreferences settings =
context.getSharedPreferences(RegisterAccount.PREFS_FILE, Context.MODE_PRIVATE);
SharedPreferences sipSettings =
context.getSharedPreferences(Settings.sharedPrefsFile, Context.MODE_PRIVATE);
Editor edSIP = sipSettings.edit();
String num = req.getPhoneNumber();
LumicallDataSource ds = new LumicallDataSource(context);
ds.open();
SIPIdentity sipIdentity = createSIPIdentity(props, settings, req);
for (SIPIdentity s : ds.getSIPIdentities()) {
if (s.getUri().equals(sipIdentity.getUri())) sipIdentity.setId(s.getId());
}
ds.persistSIPIdentity(sipIdentity);
ds.deleteSIP5060ProvisioningRequest(req);
ds.close();
edSIP.putString(Settings.PREF_SIP, Long.toString(sipIdentity.getId()));
if (!sipSettings.contains(Settings.PREF_TEL)) edSIP.putString(Settings.PREF_TEL, "-1");
/* edSIP.putString(Settings.PREF_USERNAME, settings.getString(RegisterAccount.PREF_PHONE_NUMBER, null));
edSIP.putString(Settings.PREF_PASSWORD, settings.getString(RegisterAccount.PREF_SECRET, null));
edSIP.putString(Settings.PREF_SERVER, DEFAULT_SIP_SERVER);
edSIP.putString(Settings.PREF_DOMAIN, DEFAULT_SIP_DOMAIN);
edSIP.putString(Settings.PREF_PROTOCOL, "tcp"); // FIXME - change to TLS
edSIP.putBoolean(Settings.PREF_STUN, true);
edSIP.putString(Settings.PREF_STUN_SERVER, DEFAULT_STUN_SERVER);
edSIP.putString(Settings.PREF_STUN_SERVER_PORT, "" + DEFAULT_STUN_SERVER_PORT); */
edSIP.putBoolean(Settings.PREF_WLAN, true);
edSIP.putBoolean(Settings.PREF_EDGE, true);
edSIP.putBoolean(Settings.PREF_3G, true);
edSIP.putBoolean(Settings.PREF_ON, true);
if (edSIP.commit()) Log.v(TAG, "Configured prefs for number " + num);
else {
Log.e(TAG, "error while committing preferences");
}
// Receiver.engine(context).updateDNS();
Receiver.engine(context).halt();
Receiver.engine(context).StartEngine();
}
public int speaker(int mode) {
int old = speakermode;
if ((Receiver.headset > 0 || Receiver.docked > 0 || Receiver.bluetooth > 0)
&& mode != Receiver.speakermode()) return old;
if (mode == old) return old;
enableBluetooth(false);
saveVolume();
setMode(speakermode = mode);
setCodec();
restoreVolume();
if (mode == AudioManager.MODE_NORMAL && Thread.currentThread().getName().equals("main"))
Toast.makeText(Receiver.mContext, R.string.help_speakerphone, Toast.LENGTH_LONG).show();
return old;
}
/** Runs it in a new Thread. */
public void run() {
boolean nodata =
PreferenceManager.getDefaultSharedPreferences(Receiver.mContext)
.getBoolean(
org.sipdroid.sipua.ui.Settings.PREF_NODATA,
org.sipdroid.sipua.ui.Settings.DEFAULT_NODATA);
keepon =
PreferenceManager.getDefaultSharedPreferences(Receiver.mContext)
.getBoolean(
org.sipdroid.sipua.ui.Settings.PREF_KEEPON,
org.sipdroid.sipua.ui.Settings.DEFAULT_KEEPON);
if (rtp_socket == null) {
if (DEBUG) println("ERROR: RTP socket is null");
return;
}
byte[] buffer = new byte[BUFFER_SIZE + 12];
rtp_packet = new RtpPacket(buffer, 0);
if (DEBUG) println("Reading blocks of max " + buffer.length + " bytes");
running = true;
enableBluetooth(
PreferenceManager.getDefaultSharedPreferences(Receiver.mContext)
.getBoolean(
org.sipdroid.sipua.ui.Settings.PREF_BLUETOOTH,
org.sipdroid.sipua.ui.Settings.DEFAULT_BLUETOOTH));
restored = false;
android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_AUDIO);
am = (AudioManager) Receiver.mContext.getSystemService(Context.AUDIO_SERVICE);
cr = Receiver.mContext.getContentResolver();
saveSettings();
Settings.System.putInt(
cr, Settings.System.WIFI_SLEEP_POLICY, Settings.System.WIFI_SLEEP_POLICY_NEVER);
am.setVibrateSetting(AudioManager.VIBRATE_TYPE_RINGER, AudioManager.VIBRATE_SETTING_OFF);
am.setVibrateSetting(AudioManager.VIBRATE_TYPE_NOTIFICATION, AudioManager.VIBRATE_SETTING_OFF);
if (oldvol == -1) oldvol = am.getStreamVolume(AudioManager.STREAM_MUSIC);
initMode();
setCodec();
short lin[] = new short[BUFFER_SIZE];
short lin2[] = new short[BUFFER_SIZE];
int server, headroom, todo, len = 0, m = 1, expseq, getseq, vm = 1, gap, gseq;
ToneGenerator tg =
new ToneGenerator(
AudioManager.STREAM_VOICE_CALL,
(int) (ToneGenerator.MAX_VOLUME * 2 * org.sipdroid.sipua.ui.Settings.getEarGain()));
track.play();
System.gc();
empty();
lockFirst = true;
while (running) {
lock(true);
if (Receiver.call_state == UserAgent.UA_STATE_HOLD) {
lock(false);
tg.stopTone();
track.pause();
while (running && Receiver.call_state == UserAgent.UA_STATE_HOLD) {
try {
sleep(1000);
} catch (InterruptedException e1) {
}
}
track.play();
System.gc();
timeout = 1;
luser = luser2 = -8000 * mu;
}
try {
rtp_socket.receive(rtp_packet);
if (timeout != 0) {
tg.stopTone();
track.pause();
for (int i = maxjitter * 2; i > 0; i -= BUFFER_SIZE)
write(lin2, 0, i > BUFFER_SIZE ? BUFFER_SIZE : i);
cnt += maxjitter * 2;
track.play();
empty();
}
timeout = 0;
} catch (IOException e) {
if (timeout == 0 && nodata) {
tg.startTone(ToneGenerator.TONE_SUP_RINGTONE);
}
rtp_socket.getDatagramSocket().disconnect();
if (++timeout > 60) {
Receiver.engine(Receiver.mContext).rejectcall();
break;
}
}
if (running && timeout == 0) {
gseq = rtp_packet.getSequenceNumber();
if (seq == gseq) {
m++;
continue;
}
gap = (gseq - seq) & 0xff;
if (gap > 240) continue;
server = track.getPlaybackHeadPosition();
headroom = user - server;
if (headroom > 2 * jitter) cnt += len;
else cnt = 0;
if (lserver == server) cnt2++;
else cnt2 = 0;
if (cnt <= 500 * mu
|| cnt2 >= 2
|| headroom - jitter < len
|| p_type.codec.number() != 8
|| p_type.codec.number() != 0) {
if (rtp_packet.getPayloadType() != p_type.number
&& p_type.change(rtp_packet.getPayloadType())) {
saveVolume();
setCodec();
restoreVolume();
codec = p_type.codec.getTitle();
}
len = p_type.codec.decode(buffer, lin, rtp_packet.getPayloadLength());
// Call recording: Save incoming.
// Data is in buffer lin, from 0 to len.
if (call_recorder != null) call_recorder.writeIncoming(lin, 0, len);
if (speakermode == AudioManager.MODE_NORMAL) calc(lin, 0, len);
else if (gain > 1) calc2(lin, 0, len);
}
avgheadroom = avgheadroom * 0.99 + (double) headroom * 0.01;
if (avgcnt++ > 300)
devheadroom = devheadroom * 0.999 + Math.pow(Math.abs(headroom - avgheadroom), 2) * 0.001;
if (headroom < 250 * mu) {
late++;
newjitter(true);
System.out.println("RTP:underflow " + (int) Math.sqrt(devheadroom));
todo = jitter - headroom;
write(lin2, 0, todo > BUFFER_SIZE ? BUFFER_SIZE : todo);
}
if (cnt > 500 * mu && cnt2 < 2) {
todo = headroom - jitter;
if (todo < len) write(lin, todo, len - todo);
} else write(lin, 0, len);
if (seq != 0) {
getseq = gseq & 0xff;
expseq = ++seq & 0xff;
if (m == RtpStreamSender.m) vm = m;
gap = (getseq - expseq) & 0xff;
if (gap > 0) {
System.out.println("RTP:lost");
if (gap > 100) gap = 1;
loss += gap;
lost += gap;
good += gap - 1;
loss2++;
} else {
if (m < vm) {
loss++;
loss2++;
}
}
good++;
if (good > 110) {
good *= 0.99;
lost *= 0.99;
loss *= 0.99;
loss2 *= 0.99;
late *= 0.99;
}
}
m = 1;
seq = gseq;
if (user >= luser + 8000 * mu
&& (Receiver.call_state == UserAgent.UA_STATE_INCALL
|| Receiver.call_state == UserAgent.UA_STATE_OUTGOING_CALL)) {
if (luser == -8000 * mu || getMode() != speakermode) {
saveVolume();
setMode(speakermode);
restoreVolume();
}
luser = user;
if (user >= luser2 + 160000 * mu) newjitter(false);
}
lserver = server;
}
}
lock(false);
track.stop();
track.release();
tg.stopTone();
tg.release();
saveVolume();
am.setStreamVolume(AudioManager.STREAM_MUSIC, oldvol, 0);
restoreSettings();
enableBluetooth(false);
am.setStreamVolume(AudioManager.STREAM_MUSIC, oldvol, 0);
oldvol = -1;
p_type.codec.close();
rtp_socket.close();
rtp_socket = null;
codec = "";
// Call recording: stop incoming receive.
if (call_recorder != null) {
call_recorder.stopIncoming();
call_recorder = null;
}
if (DEBUG) println("rtp receiver terminated");
cleanupBluetooth();
}