Ekiga Changelog

New in version 4.0.1

February 21st, 2013
  • User-visible fixes:
  • Fix crash when quitting ekiga while receiving presence information
  • Fix crash when quitting ekiga right after starting it (before STUN ending)
  • Fix crash when disabling an account while icons in roster are changing
  • Fix crash when receiving call a second time
  • Fix crash in XML parsing in case of malicious code
  • Fix increasing CPU usage afer hours of usage caused by endless OPTIONS
  • Several fixes for H.323:
  • fix H.323 parsing
  • add the username in authentication
  • fix unregistering the gatekeeper
  • fix registration
  • assign gk_name only if success
  • do not propose adding an H.323 account if the protocol is not built-in
  • Fix registration for registrars accepting the last Contact item offered
  • Allow to change the REGISTER compatibility mode of an existing registration
  • Fix impossibility to hangup active call after a missed call
  • Fix busy or call forwarding on busy occuring when connection is released
  • Fix subscribing/unsubscribing when enabling and disabling SIP accounts
  • Do not show is-typing messages sent by other programs during chatting
  • Stop ongoing registration when remove account
  • Use meaningful names for ALSA sub-devices
  • Allow to enter contact addresses without host part, and choose the host later
  • Increase number of characters shown in device names
  • Use a better icon for call history in addressbook
  • Show the address instead of "telephoneNumber" in addressbook
  • Deactivate NullAudio ptlib's device for audio input too
  • Do not send OPTIONS messages once the account is disabled
  • Hide the main window immediately on exit
  • Handle xa status as away
  • Fix debugging message when registering
  • Fix race condition leading to duplicate entry in call history
  • Fix incoming call if two INVITE's in a fork arrive very close together
  • Use correct username in OPTIONS messages
  • Allow to have message waiting indication even if asterisk's vmexten is off
  • Send OPTION only on the right interface
  • Fix buttons direction in dialpad for RTL languages
  • Fix aborting RTP receiver with Polycom HDX8000
  • Fix possible incorrect jitter calculation for RTCP
  • Only kill REGISTER/SUBSCRIBE forks if a "try again" response is received
  • Various other fixes
  • Build fixes:
  • Fix building opal when java SDK installed and swig is not
  • Some code cleanup
  • Translation updates:
  • Update translations: fr, ml, pt_BR
  • Update help translations: pt_BR

New in version 4.0.0 (November 27th, 2012)

  • A major overhaul of the main window, a re-adding of the H264 and H263+ video plugins.
  • A new pulse audio plugin (in ptlib), new audio codecs: SILK (used by skype), G.722.1 (aka Siren 7), and G.722.2 (aka GSM-AMR Wide band), fixed H.323 gatekeeper support, call auto-answer, support for handling multiple video streams (H.239), unregister of accounts when quitting, faster startup by not retrieving the canonical name of servers, fixes for the Contact field during registration, Jabber/XMPP and GStreamer fixes, and support for mingw-w64 on Windows using its own headers.

New in version 3.3.2 (August 23rd, 2011)

  • User-visible fixes:
  • Re-add back H264 and H263+ video plugins
  • Fix presence in case of several tuples
  • Fix presence when receiving old presentity type without note
  • Re-add contact limited compatibility mode for bogus registrars
  • Fix leaking the opal account store in the assistant
  • Make Enter key in roster (contact list) make a call
  • Fix infinite loops in the loudmouth presentity code
  • Add mnemonics to buttons in Accounts dialog box
  • Fix _dl_close failed assertion at quit
  • Distributor-visible changes:
  • Re-add back H264 and H263+ video plugins
  • Build fixes:
  • Fix generation of po translation files
  • Fix compilation error with older GTK on GDK_KEY_KP_Enter
  • Replace some deprecated symbols in gtk 2.18
  • Fix build with -Wl,-z,defs of xcap plugin
  • Check for ptlib expat support during configure
  • Translation updates:
  • Updated translations: cs, de, es, fr, ru, sl

New in version 3.3.0 (December 22nd, 2010)

  • This version brings several changes (bugs and protocol fixes, but not so many features and no security fixes).
  • It is not a replacement for the stable release 3.2.7, but should be used by people having problems with the stable release (e.g. a pulse plugin is included in this release) and by people who want to help ekiga.
  • Major changes include a new pulse audio plugin (in ptlib), various SIP/H.323 fixes, many code cleanups, and the replacement of various deprecated code sections with current equivalents.

New in version 3.2.7 (May 31st, 2010)

  • User-visible fixes:
  • Use NAT ports instead of local ports for RTP, fixing many connection issues
  • Add workaround for "In some cases ekiga answers always Busy Here"
  • Fix crash in the avahi code
  • Set the default audio and video devices
  • Fix Call buttons do not work
  • Add bpp16 (RGB16) support
  • Add partial support for IPv6
  • Fix number of users found on ekiga.net LDAP directory
  • Fix connection type setting in assistant
  • Use a decent connection type by default (DSL 128kb/s uplink)
  • Check off iLBC, CELT32, CELT48 and G722 audio codecs by default
  • Fix possible crash when PTLib code accessed by thread not created by PTlib
  • Fix issue when deleting a safe object
  • Fix possible NULL pointer access if get multiple NOTIFY messages on a REFER
  • Protocol fixes:
  • Fix strange issue with SIP call diversion (302 response) to the same endpoint
  • Change to only unregister those contact addresses we successfully registered
  • Fix incorrect change to AlertingPhase in H323
  • Change authentication failure to be fatal and SIP handler removed
  • Fix double authorisation failure leaving SIP handler in the wrong state
  • Fix SIP REFER completion on receiving NOTIFY with id parameter
  • Fix authentication issue when can get to SIP server via two interfaces
  • Fix address translation of REGISTER contact fields when STUN is not used
  • Fix locating correct authorisation credentials for INVITE
  • Fix correct test handling RFC3261/8.2.2.2 merged requests and other
  • "multi-path" received INVITE requests
  • other minor fixes
  • Build fixes:
  • Fix gdu (gnome-doc-utils) configure option
  • Replace deprecated functions in gtk >= 2.18
  • Fix build with binutils-gold
  • other minor build fixes
  • Windows port fixes:
  • Fix major bug where the uninstaller could remove non-Ekiga files
  • Fix starting Ekiga from last page of installer
  • Fix language selection during installation
  • Update version of dependent libraries
  • Allow to create executable from release or from git/svn
  • Add Dutch and Romanian translations of Windows installer
  • Add all help localisation files to the installer
  • Fix compilation error on std::freopen on newer gcc
  • other build fixes
  • Translation fixes:
  • Updated translations: bn_IN, eu, gu, kn
  • New translations: ast
  • Updated help translations.

New in version 3.2.6 (September 23rd, 2009)

  • This release fixes various crashes and freezes.
  • It adds a workaround for registrars that refuse registration when several contacts are given.
  • It fixes "Contacts never go offline".
  • It makes sure the output device is set before a call.
  • It fixes port handling when registering to a proxy with a non-standard port.
  • It is possible to register e.g. user@a.b as a username as required by some providers.
  • There are many improvements to the Windows port, which is now in beta status.
  • There are many other fixes.

New in version 3.2.5 (July 7th, 2009)

  • Fixed very low rate of sent images during video conversations
  • Fixed crash in LDAP
  • Fixed crash in presence
  • Fixed simultaneous reads from different threads in jitter code
  • Fixed crash during idle times
  • Fixed crash when accepting a call
  • Fixed deadlock in SIP handler
  • Fixed freeze upon calling and showing of pc:udp$... in the URI bar
  • Fixed crash in G726-16 audio codec
  • Allows multiple registrations with the same registrar using different user names
  • Allows user to choose system iLBC
  • Fixed window resizing in chat when entering very long words
  • Fixed linking of sbc plugin with libsamplerate
  • Fixed compilation with gtk 2.12
  • Fixed compilation with gcc 4.4
  • Windows and Solaris specific fixes
  • Added a small script, ekiga-debug-analyser, not installed, which retrieves only the packets exchanged from a Ekiga debug output
  • Other minor fixes
  • Updated translations: bn_IN, el, or
  • Updated help translations: el
  • Changes in ekiga 3.2.4 (2009-05-20)
  • Fixed OPAL and PTLIB recommended versions
  • Changes in ekiga 3.2.3 (2009-05-20)
  • Fixed remote uri not being shown in the uri bar when dialing out
  • Changes in ekiga 3.2.2 (2009-05-20)
  • Fixed a crash on some calls

New in version 3.2.1 (May 20th, 2009)

  • It contains a fix for invalid OPAL and PTLIB recommended versions.

New in version 3.2.1 (May 20th, 2009)

  • 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 unusefulRequestTerminated code, but the actual error
  • In assistant, fill user name field, if empty, with user name
  • In preferences, audio/video devices, remove unused FFMPEG andWAVFile 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 SIPregistrar
  • Fixed problem with not waiting till ACK arrives, someimplementations get offended if the ACK gets a transaction does notexist error. Thanks hongsion for the report
  • Fixed bug where if a non-INVITE transaction gets a 1xx response, butthen the 2xx (or above) response is lost, the command is notretransmitted
  • Added fix for video plug in shared library loading, current codewould 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 differentlocal user name to the original call
  • Fixed bug where opal tries to install plugins even if they have beendisabled
  • 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 inputframes
  • 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 neveroffered then abort the call as this is SO not to specification!
  • Fixed possible assertion if the soundcard blocks and prevents thedevice to be closed
  • Fixed possible path through unsubscribe/unregister code that couldlead to a NULL pointer being used
  • Fixed issue in SIP registering, if both a full AOR and a registrarhost name is provided then we would normally disable all registrarsearches (e.g. SRV record lookup) and just use the host namespecified
  • 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 usingINVITE with a replace header
  • Fixed being able to switch off jitter buffer while still a threadreading from it
  • Fixed bug with "hairpin" SIP calls, subsequent commands to INVITEare 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 asession 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 401/407authentication required error, the re-transmitted INVITE was not are-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 IPaddress/port for RTP, then we did not change our RTP sendaddresss/port
  • Add numerous boundary checks to H.263 codec
  • Discard out of order packets, mode A frames that don't begin with astart code, and frames that don't begin with a start code in H.263codec
  • Fixed initial H.323 call set up honouring the auto-startconfiguration 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 backdevices, Fixed issue with empty strings for Windows sound devicesbeing 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