Ekiga 3.2.1 available

This is the first stable update of Ekiga.



Here is the list of changes:

  • 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, Added 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.

Very special thanks to Robert J