[Ekiga-devel-list] ANNOUNCE - Ekiga 3.2.1 [STABLE] available

[Eugen Dedu <Eugen Dedu pu-pm univ-fcomte fr>]

This is the first stable release of the 3.2 release series of Ekiga.


* What is it ?
==============

Ekiga is a free Voice over IP softphone allowing you to do free audio 
and video calls over the Internet.

Ekiga is the first Open Source application to support both H.323 and
SIP, as well as audio and video.  Ekiga was formerly known as
GnomeMeeting.

More information can be found at http://www.ekiga.org


* Where to get it ?
===================

Ekiga is available at:
ftp://ftp.gnome.org/pub/gnome/sources/ekiga/3.2 (take 3.2.1)

Required librairies can be found at:
ftp://ftp.gnome.org/pub/gnome/sources/opal/3.6/ (take 3.6.2)
ftp://ftp.gnome.org/pub/gnome/sources/ptlib/2.6/ (take 2.6.2)


* What's changed ?
==================

- Fixed various crashes on shutdown
- Fixed crash when opening preferences or assistant
- Fixed crash when no account
- Fixed SIP registration
- Fixed DTMF mode for SIP endpoint
- Migrate ekiga.net configuration from 3.0 to 3.2
- Maintain window position on hiding/showing the main window
- On some failed registration, do not show the unuseful
  RequestTerminated code, but the actual error
- In assistant, fill user name field, if empty, with user name
- In preferences, audio/video devices, remove unused FFMPEG and
  WAVFile modules
- Fixed recognition of cameras with non-ascii characters
- Fixed compilation with --disable-tracing
- Various fixes during configuration
- Fixed issue with having multiple registrations with the same SIP registrar
- Fixed problem with not waiting till ACK arrives, some
  implementations get offended if the ACK gets a transaction does not
  exist error. Thanks hongsion for the report
- Fixed bug where if a non-INVITE transaction gets a 1xx response, but
  then the 2xx (or above) response is lost, the command is not
  retransmitted
- Added fix for video plug in shared library loading, current code
  would not look anywhere but default path
- Fixed compiling G722 plug in on SUN
- Fixed correct value for remote party address
- Fixed compilation on NetBSD
- Fixed INVITE sent in response to a REFER command using a different
  local user name to the original call
- Fixed bug where opal tries to install plugins even if they have been 
disabled
- Fixed crash in PStandardColourConverter::YUY2toYUV420PWithResize
- Fixed include path overrides package include path
- Fixed search for connection matching replaces header dialog info,
  broken during changes to make calls back into the same stack
- Fixed from/to fields reversed in call dialog identifier information,
  needed for a INVITE with replaces header
- Fixed thread leaks
- Fixed video I-frame detection
- Fixed media format matching option additions
- Fixed advanced rate controller support
- Fixed popping frames problem when rate controller skips input frames
- Fixed missing re-lock of mutex on jitter buffer shut down
- Fixed gatekeeper discovery
- Added YUV2 support to DirectX code
- Fixed crash in PInterfaceMonitor::Stop
- If SIP answer to our offer contains only media formats we never
  offered then abort the call as this is SO not to specification!
- Fixed possible assertion if the soundcard blocks and prevents the
  device to be closed
- Fixed possible path through unsubscribe/unregister code that could
  lead to a NULL pointer being used
- Fixed issue in SIP registering, if both a full AOR and a registrar
  host name is provided then we would normally disable all registrar
  searches (e.g. SRV record lookup) and just use the host name
  specified
- Change default TSTO in H.263 to give better quality
- Fixed issue with SIP call hairpinning back into the same stack
- Fixed possibility of closing a channel twice
- Fixed intermittent problem with losing an audio channel when using
  INVITE with a replace header
- Fixed being able to switch off jitter buffer while still a thread
  reading from it
- Fixed bug with "hairpin" SIP calls, subsequent commands to INVITE
  are not routed to the correct connection instance
- H.224 should not be enabled when H.323 is disabled
- Various Solaris build fixes
- Fixed RFC3890 support
- Don't stop a call from clearing due to lack of media just because a
  session has not received any packets
- Fixed memory leaks in the plugins code
- Improved the RTP stack performances
- Fixed various video payload problems
- Fixed issue with outgoing re-INVITE that gets a 401/407
  authentication required error, the re-transmitted INVITE was not a
  re-INVITE but another normal INVITE, so "hold" doesn't work
- Fixed issue with incoming re-INVITE that has no SDp in the INVITE,
  if the eventual ACK has the same streams but only changed the IP
  address/port for RTP, then we did not change our RTP send
  addresss/port
- Add numerous boundary checks to H.263 codec
- Discard out of order packets, mode A frames that don't begin with a
  start code, and frames that don't begin with a start code in H.263
  codec
- Fixed initial H.323 call set up honouring the auto-start
  configuration for "don't offer"
- Fixes for gcc 4.4.0
- Fixed compilation with video, h.323 or sip disabled
- Windows port:
  - DirectX fixes
  - Better LDAP support
  - Add back devices
  - Fixed issue with empty strings for Windows sound devices being
    returned when being used over a Remote Desktop connection
  - Fixed G.722 compilation
  - Fixed linker problems
- Other minor fixes
- Updates translations: ar, as, crh, es, kn, nb, or, zh_CN
- Updated help translation: el, es

Special thanks to Julien Puydt, Michael Rickmann, Mounir Lamouri,
Eugen Dedu, Jan Schampera and Yannick Defais for their continuous work
on Ekiga.

On behalf of Ekiga team,
Eugen Dedu