CHANGELOG:
7-3-2017 : v1.12.0 released
  - fix : online help URL has moved
3-7-2017 : v1.11.0 released
  - updated ffmpeg support
6-2-2015 : v1.10.0 released
  - added GSM codec (bug #37)
  - fix : if call comes in on line that failed to register, mark the line as active to allow user to answer the call
    - this is a rare case where server may have been unreachable briefly
9-24-2015 : v1.9.9 released
  - added display name to sip accounts
  - added lock file to prevent running multiple copies
9-12-2015 : v1.9.8 released
  - fixes to register()ing
    - this should fix bug #31
7-9-2015 : v1.9.7 released
  - just minor fixes for linux users
    - if native libraries aren't loaded the app will not exit but the video button will be disabled
    - now includes 32bit linux libraries
    - this should fix bug #29
6-30-2015 : v1.9.6 released
  - fixed support for QOP auth and Record-Route fields (RFC 2543 6.29).
    - flowroute.com is now supported
6-15-2015 : v1.9.5 released
  - fixed handling auth string with spaces in quotes (created custom String.split())
4-7-2015 : v1.9.4 released
  - fixed some memory leaks
3-26-2015 : v1.9.3 released
  - added Online Help button to configure window (see ticket #25)
2-9-2015 : v1.9.3 beta 1 (not released)
  - added support for inbound calls where SDP is not in INVITE but in ACK instead (see RFC 3665 sec 3.6)
    see ticket #24
10-25-2014 : v1.9.2 released
  - removed 'Keep Audio Open' option
    audio is now start()ed only when needed (both output and input)
    this should keep audio synced better (see bug #22)
  - the audio recording has been moved to a seperate thread
9-15-2014 : v1.9.1 released
  - new option : exit when closed
  - new tray icon (Windows 7 theme)
7-8-2014 : v1.9.0 released
  - new codec : G.722 wideband : 16Khz sampling rate @ 14bits/sample (64Kbps)
    - with a different sampling rate I had to rewrite a lot of code to support different sampling rates.
    - everything is now interpolated up to 44.1KHz internally (requires a bit more cpu power)
    - JPL is now ready for more HD codecs
      - although I've looked at some like G.711.1 and doing so would be very difficult
        - do you remember your FFTs from university?
7-5-2014 : v1.8.1 released
  - fixed video codec selection (works better with Asterisk now)
    - I was using the offered codecs instead of the accepted ones
      and by default VP8 is used which Asterisk doesn't support yet
4-22-2014 : v1.8.0 released
  - new : a different ringtone can be assigned for inbound and outbound(ringback) tones. (user requested feature)
3-12-2014 : v1.7.1 released
  - fix : Java8 compatible
  - new : added checkAuth() and checkForReplay() for SRTP (added security)
2-1-2014 : v1.7.0 released
  - new : SRTP key exchange thru DTLS (tested with jPBXlite/0.16 only)
  - this should finish everything I have planned lately and jPhoneLite should quiet down for a while unless bugs are found
1-31-2014 : v1.6.0 released
  - new : SIP signaling now supports TCP and TLS transports
    - with TLS you must configure Asterisk to not verify certificate: sip.conf add : tlsdontverifyserver=yes
1-30-2014 : v1.5.0 released
  - new : added SRTP media encryption, tested with Asterisk 1.8 (11.7.0)
  - up next : secure signaling and DTLS
12-31-2013 : v1.4.3 released
  - fix : WinSound would cause NPE in stop() if start() failed
  - fix : if TURN failed the line was unusable
12-17-2013 : v1.4.2 released
  - fix : when a call is placed on hold the RTPChannel thinks the stream is inactive and generates an rtpInactive event which ends the call (this logic was recently added)
    - if the streams mode doesn't include recv then the inactive event is not triggered
  - fix : properly send back the SDP mode (sendrecv, inactive, etc.) for each stream as requested in invite
  - new : when the audio stream is on hold (no recv) an intermittent beep is generated locally
12-12-2013 : v1.4.1 released
  - fix : removed requestPublicIP in javaforce.voip.RTP - a quick add in v1.4.0 which is not needed and I didn't wait for reply which caused harmless stun exceptions
  - new : now support using the LocalCamera on multiple lines at the same time (as long as they all use the same codec)
12-9-2013 : v1.4.0 released
  - added support for multiple video channels.  Video conferencing with jPBXlite/0.12 is now possible.
    - see jpbxlite.sf.net/VideoConference.txt for more details
  - video with XLite 4.x (H.263-1998) now works. (although Xlite's video decoder seems to have problems and even crashed a few times).
  - many bug fixes
11-30-2013 : v1.3.0 released
  - added VP8 video codec support (RFC Draft : http://tools.ietf.org/html/draft-ietf-payload-vp8-10)
11-27-2013 : v1.2.0 released
  - added STUN,TURN,ICE support
    - STUN is used to detect public IP/port
    - TURN is used to relay RTP data when direct connecting is not possible
      - some PBXes do NOT proxy the RTP data so TURN can be useful in these cases
    - tested with resiprocate reTurn server (http://resiprocate.org/ReTurn_Overview)
  - added SIP Expires to EditSettings dialog box (default = 3600 seconds) (was previously undocumented setting)
  - added SIP/RTP Port Ranges
  - added check for "received" & "rport" tag in "Via" field returned from server (RFC 3581)
    - this is just as good as STUN (if the server supports it)
  - removed dyndns support to detect public IP (lagged really bad sometimes with no proper timeout)
11-22-2013 : v1.1.9 released
  - update status when transfer is requested incase PBX doesn't respond
  - perform sip registration in a new thread after config is done
    - makes gui more responsive since registration can sometimes take several seconds
  - other minor fixes
11-21-2013 : v1.1.8 released
  - calls are no longer cut off after a transfer, the PBX will issue a BYE when the transfer is complete (blind or non-blind)
    - this avoids failed transfers leaving a call leg dangeling (which would eventually get dropped)
  - during xfer process you can double-click a contact or recent call log to enter as target
11-14-2013 : v1.1.7 released
  - fixed sound uninit order
  - fixed SIP registration using qop,nc (no quotes)
  - fixed some other minor SIP handling issues
  - added ESC/ENTER hotkeys to dialog boxes
10-14-2013 : v1.1.6 released
  - undo bug #14 but add a break;
10-01-2013 : v1.1.5 released
  - fixed ACK missing Tag sometimes (see bug #13)
  - fixed NOTIFY not decoding messages sometimes (see bug #14)
9-27-2013 : v1.1.4 released
  - fixed msvcr100.dll not found on msi package
9-26-2013 : v1.1.3 released
  - fixed sipexpires option not being used in reregister()
  - fixed possible crash in codecpack
9-25-2013 : v1.1.2 released
  - JPL is now UTF-8 compatible
  - Settings includes a 'sipexpires' entry if you want to change it manually
      Default = 3600 seconds (1 hr)
      min=300 (5 mins) max = 3600 (1hr)
  - added option to download CodecPack from EditSettings
9-19-2013 : v1.1.1 released
  - fixed WinSound issues
9-16-2013 : v1.1.0 released
  - H.263 RFC2190 still incomplete - maybe fix later
  - tested with jPBXlite/0.9 including video
9-13-2013 : v1.1.0 BETA #3 released
  - new - all three audio systems provide selecting the output/input hardware (java/win/lnx)
  - new - the Win32/64 loader is now 32/64bit universal (32bit loader will launch 64bit if needed)
        - makes for just one MSI package (less confusion)
  - if there are no more bug fixes this will become the next stable release
9-12-2013 : v1.1.0 BETA #2 released
  - new - finished Linux native API using JNA for Sound/Camera
        - camera may not support all pixelformats until I get ffmpeg/libavdevice working
9-7-2013 : v1.1.0 BETA #1 released
  - new - all JNI code has been ported to JNA
    - linux api has not been coded yet (so no camera yet in linux)
  - new - H.263/264 is now supported with FFMPEG
    - H.263 requires 1998/2000 versions.
    - The original H.263 using RFC2190 is not supported yet (see src/javaforce/voip/RTPH263.java)
  - new - JPL is now packaged in MSI files for Windows users
7-28-2013 : v1.0.7 released
  - fix - auto hold feature will not place a line on hold that is in a conference
  - fix - an inbound reINVITE while still ringing would send a 200 back in error
7-25-2013 : v1.0.6 released
  - fix - was sending an ACK for a 183 (ringback).  ACK should only be sent for 200 once call is complete.
  - hopefully this fixes all bugs related to new upgrades
7-24-2013 : v1.0.5 released
  - fix - onCancel() now fully ends a call which can happen in a ringback phase of a call
7-23-2013 : v1.0.4 released
  - fix - fix inbound call again - a reINVITE while line is still ringing could cause a second line to start ringing for the same call
  - fix - hold function now works more than once (cseq was not incremented - which I had a comment not to do so - not sure why??? - oh well, works now).
  - new - option:Auto Hold/Unhold - this will automatically hold/unhold lines as you switch between them
  - new - ringback tones are now supported (SIP code 183).  If you call certain phones (mostly mobiles) they may have custom ring tones for you to hear.
  - new - vol settings are now saved in config
7-22-2013 : v1.0.3 released
  - fix : if a line is ringing from inbound call and another inbound call comes in on the same SIP account while the first call is still rining,
            the new call will overtake the same line and the first call was lost
5-8-2013 : v1.0.2 released
  - fix : SUBSCRIBE was not working (extra headers were removed before 401 was processed)
  - new : can now perform non-blind transfers.
    - when you want to transfer someone pick another line first
    - call the transfer target on the other line and let them know you are going to transfer a call to them
    - go back to the first line and press XFR and then the other line
    - if all goes well the original line will be transfered to the other line
3-1-2013 : v1.0.1 released
  - fix : timestamp on g729a should be 160 like all the other codecs
12-29-2012 : v1.0 released !!!
  - no code change (a few renamed classes)
  - Thanks to all over the world for sending in bug reports, network traps, etc. to help make jPhoneLite successful!!!
12-5-2012 : v0.99.1 released
  - fix : RTP buffering issue (now uses cyclical buffer)
12-4-2012 : v0.99 released
  - new : added a recording button (to WAV file)
  - new : optimized codec encoding/decoding by avoiding memory reallocation (now assumes 20ms blocks)
  - fix : g711a codec : encoding algo was wrong (copied incorrectly from asterisk)
  - note : the next version will be 1.00 if I don't fix any bug in the next week or two
12-3-2012 : v0.92 released
  - fix : I think I really fixed the windows audio this time.  Tested on systems that were having problems before.
          Increased number of buffers used and cleaned up the code by avoiding any memory re-allocation in the timers.
11-30-2012 : v0.91 released
  - new : added disable video option per line configuration.
          I've noticed some SIP providers get confused when they see both audio and video codecs.
  - fix : properly end a call when codec negotation fails
  - new : can now config video FPS and resolution (keep it low or audio gets choppy)
  - removed : Windows VFW support - was just crashing - and unable to scale window properly (obsolete api)
  - fix : use proper RGB order when using videoInput library
11-28-2012 : v0.90 released
  - removed : flash support - audio had unacceptable latency, video was non-standard codec
  - new : added Windows Camera support using VFW and DirectShow (using videoInput library)
  - fix : codec negotation when receiving a call
11-16-2012 : v0.85 released
  - new : added g711a codec (European format)
  - fix : improved windows output quality (keep buffers full)
11-7-2012 : v0.84 released
  - new : added windows native API for audio support
  - fix : the timestamp in the SDP packet is incremented by the # of bytes in the packet, not the ms of duration.
          This effected g711u, which should be 160, not 20.
          g729a just happened to be 20ms and 20bytes per packet so not effected.
          This fixes callcentric.com support, when using g711u
          I think Asterisk ignores the timestamp so this was never an issue before.
11-5-2012 : v0.83 released
  - added full javascript support.  see jphonelite-javascript.html and jphonelite-javascript.php for example usage.
  - added disable g729a option
?-?-2012 : v0.82 released
  - ??? fixed something ???
7-10-2012 : v0.81 released
  - video under linux is now possible with GStreamer [requires jfLinux.org]
6-30-2012 : v0.70 released
  - now supports native Linux sound
4-15-2012 : v0.60 released
  - no code change, just restructured for JavaForce/6.0
10-23-2011 : v0.54 released
  - new : mini/micro editions for webpages
  - note : everything seems to work fine with Java 7u1 SDK (although there are 30 warnings)
8-3-2011 : v0.53 released [just a few hours later]
  - fix : fix for Asterisk 1.8 broke 1.6/1.4 - fixed it again.
  - note : JPL doesn't not seem to be compatible with Java 7 SDK (although compiled with JDK 6 seems to run fine in JRE 7) - I'll try and fix that later
8-3-2011 : v0.52 released
  - fix : Asterisk 1.8 compatible (improved 401/407 processing)
  - fix : Flash video works again
    - using javascript from Java applet to start flash was not working anymore, so I forced Flash to keep reconnecting until Java RTMP server starts
5-2-2011 : v0.51 released
  - fix : Wav loader supports Microsoft WAV format (ignore 'fact' header)
  - fix : use different icons for Windows/Linux
4-18-2011 : v0.50 released
  - new : now uses Flash for video (H.264) (Applet only)
    - removed old JMF video (crud)
    - removed old enhanced windows-specific features (crud)
3-17-2011 : v0.23 released
  - fix : timing issue when a line registered before same lines were not ready yet
11-9-2010 : v0.22 released
  - new : keys F1-F6 will select lines 1-6
  - note : dropped "BETA" designation
9-27-2010 : v0.21 BETA released
  - new : support for non-REGISTERed mode (just leave password blank)
  - fix : pressing digits generates DTMF tones locally again (java:KeyMaps were not working - used dispatchKeyEvent() instead - it's a lot less coding)
9-19-2010 : v0.20 BETA released
  - new : video now support H.263 codec and is compatible with Asterisk PBX  (requires JMF)
  - new : enhanced features (Instant messaging, Share Desktop, File transfer) are compatible with Asterisk PBX
    - enhanded features are not compatible with older versions of JPL
  - fix : many bugs
8-9-2010 : v0.19 BETA released
  - new : added speaker phone button (toggles mic on/off when other person is talking to avoid echo)
8-4-2010 : v0.18 BETA released
  - new : added ringtone option to settings (used on inbound calls)
8-2-2010 : v0.17 BETA released
  - fix : MD5 qop=auth is now properly supported (qop=auth-int is not supported)
7-14-2010 : v0.16 BETA released
  - new : contacts can now be monitored (SIP:SUBSCRIBE)
  - new : presence publish available in general options (SIP:PUBLISH) (Note : not supported by Asterisk)
  - fix : process SIP:403 properly for INVITEs (was assuming for REGISTER only before)
  - fix : unREGISTERs properly now
7-7-2010 : v0.15 BETA released
  - new : passwords are now encrypted in XML config and are not visible in settings editor.
        - old unencrypted passwords in XML config are still loaded, and encrypted on next save
  - new : in software volume control mode the volume can be amplified (up to +10dB)
  - fix : sound meters now show highest level instead of an average (which didn't make much sense for an audio signal)
  - fix : SIP:CANCEL handled properly  (487, 200)
  - fix : zero playback meter when audio is stop()ed.
6-22-2010 : v0.14 BETA released
  - new : (user requested) added ability to load XML config in Applet from website before jPhoneLite starts (see php files)
  - fix : VNC was not closing files while loading icons so they could only be loaded once
  - fix : placed sip.register()/unregister() in try {} catch {} block
  - fix : only enable ShareDesktop/Video buttons if DLL was successfully loaded
6-21-2010 : v0.13.1 BETA released
  - fix : opps - make sure Windows API is only called on Windows.  Was unable to open config screen on non-windows systems.
6-21-2010 : v0.13 BETA released
  - new : LET THERE BE VIDEO (windows only)
          I started to try out JMF for video but not liking it too much.  Since JMF just uses VFW I decided to write my own native code.
          Video code is much like VNC code and uses same JPEG/PNG codec.  I'll try to add better compression later.
          Just like other advanced features it uses custom RTP packets which Asterisk blocks.
          I've tested it with callcentric.com and jPBXlite.
          What? Never heard of jPBXlite !?!  Well that's because I haven't released it YET.
          Available soon at http://jpbxlite.sourceforge.net
  - fix : everything is compiled in Java 1.5 so it should work with Java 5 systems.
  - fix : VNC code improved a little, works best with WinXP (still has issues with Vista and multi-monitor systems)
6-18-2010 : v0.12.1 BETA released
  - new : just added NetBeans projects files for those who want to compile from the IDE.
6-16-2010 : v0.12 BETA released
  - new : added Applet as requested
  - new : added key binding ENTER to click Call
6-11-2010 : v0.11 BETA released
  - new : added message waiting (voicemail) indicator (flashing orange light)
  - fix : keepAudioOpen option was preventing ringing sound from being played (introduced in v0.9.1)
  - new : added option 'Use smaller font' for JVMs with different font sizes (in case text in buttons is missing)
6-10-2010 : v0.10 BETA released
  - new : send file feature (just like ShareDesktop and Instant Messaging it requires custom RTP packets)
  - new : the handling of custom RTP packets has changed and is *NOT* compatible with previous versions.
          RTP packets are now properly seperated into channels using the ssrc id.
          So Instant Messaging and ShareDesktop will not work with older versions.
  - fix : properly flush output to close a ShareDesktop session.
  - fix : ShareDesktop client can now close the session.
6-1-2010 : v0.9.1 BETA released
  - new : improved share desktop (VNC) code (viewer also includes a toolbar for some basic functions)
  - new (user requested feature) : new option 'Keep Audio Open' - you can disable it so output audio is only start()ed
    when actually needed (default is to keep it always running to avoid popping sounds when audio is started and stopped).
5-26-2010 : v0.9 BETA released
  - new : share desktop - allow other side to see and control your desktop. (works only on Windows systems)
5-10-2010 : v0.8.3 BETA released
  - new : added a system tray icon that shows incoming calls if jPhoneLite is not active window
  - new : new options : 'hide on minimize' and 'always on top'
  - new : can now ignore an incoming call (just press End)
5-6-2010 : v0.8.2 BETA released
  - fix : check available() before read()ing from mic to avoid blocking in timer - improves audio quality
  - fix : another bug in local ip detection
4-30-2010 : v0.8.1 BETA released
  - fix : log rollover didn't work
  - new : added new option 'Disable Logging'
4-28-2010 : v0.8 BETA released
  - new : added new feature : Instant Messaging (IM)
    if server allows any RTP data thru then the IM button is enabled which popups a simple
    instant messaging window.
    Doesn't seem to work with Asterisk but callcentric.com works.
4-26-2010 : v0.7 BETA released
  - new : finished writing audio input/output selection in the settings
  - note : most source now includes lots of javadoc comments.
    Try running 'ant javadoc' in the root folder and in /projects/jphonelite
    This will generate 'javadoc' folders, in there open index.html for the javadocs.
  - fix : jnlp now installs start menu and desktop shortcuts.
  - fix : other minor fixes (tries to register a line 5 times before giving up)
4-19-2010 : v0.6 BETA released
  - new : now supports callcentric.com (wasn't easy) (hold doesn't work - use mute instead)
    callcentric is VERY strict on their sip syntax
  - new : added authorization username (if different from normal username)
  - fix : improved local IP detection
  - new : checks for update on startup
4-13-2010 : v0.5 BETA released
  - fixed : time critical timers must use scheduleAtFixedRate() or else choppy sound was occuring on some (most?) systems
  - new : retry register if reregister failed during expired reregisters
  - fixed : other minor improvements
4-9-2010 : v0.4 BETA released
  - fixed : codec negotiation
    if both codecs are accepted by remote then reINVITE with just g729a to make sure correct codec is used (outbound)
    only send back one codec for inbound calls
  - new : reregister after connection expires (3600 seconds)
  - new : reregister if not registered in 1 sec (happens a lot actually)
  - new : hosts can include port spec (example.com:5060)
  - new : hungup shows error code (503, etc.)
4-8-2010 : v0.3 BETA released
  - handle reINVITEs now (was just hanging up before)
  - generate DTMF tones locally (those generated locally and remotely) (only generated while in a call)
  - fixed bug : was issuing 180 AFTER 200 for an INVITE if AA was enabled (harmless?)
4-7-2010 : v0.2 BETA released
  - added g729a codec
  - other minor fixes
4-6-2010 : v0.1 BETA released
  - first public release