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authorMichal Kubecek <mkubecek@suse.cz>2015-04-13 09:21:39 +0200
committerMichal Kubecek <mkubecek@suse.cz>2015-04-13 09:21:39 +0200
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treea40b171be1d859c2232ccc94f758010f9ae54d3c /NEWS
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+25 february 2009 - 1.4.2
+========================
+- Integration with Diamondcard Worldwide Communication Service
+ (worldwide calls to regular and cell phones and SMS).
+- Show number of calls and total call duration in call history.
+- Show message size while typing an instant message.
+- Show "referred by" party for an incoming transferred call in systray popup.
+- Option to allow call transfer while consultation call is still in progress.
+- Improved lock file checking. No more stale lock files.
+
+Bug fixes:
+----------
+- Opening an IM attachment did not work anymore.
+
+Build fixes:
+------------
+- Link with ncurses library
+
+
+31 january 2009 - 1.4.1
+=======================
+Bug fixes:
+----------
+- No sound when Twinkle is compiled without speex support.
+
+Build fixes:
+------------
+- Compiling without KDE sometimes failed (cannot find -lqt-mt).
+- Configure script did not correctly check for the readline-devel package.
+
+
+25 january 2009 - 1.4
+=====================
+- Service route discovery during registration.
+- Codec preprocessing: automatic gain control, voice activation detection,
+ noise reduction, acoustic echo cancellation (experimental).
+- Support tel-URI as destination address for a call or instant message.
+- User profile option to expand a telephone number to a tel-URI instead
+ of a sip-URI.
+- Add descending q-value to contacts in 3XX responses for the redirection
+ services.
+- AKAv1-MD5 authentication.
+- Command line editing, history, auto-completion.
+- Ignore wrong formatted domain-parameter in digest challenge.
+- Match tel-URI in incoming call to address book.
+- Determine RTP IP address for SDP answer from RTP IP address in SDP offer.
+- Show context menu's when pressing the right mouse button instead of
+ after clicking.
+- Swedish translation
+- Resampled ringback tone from 8287 Hz to 8000 Hz
+
+Bug fixes
+---------
+- Text line edit in the message form looses focus after sending an IM.
+- Twinkle does not escape reserved symbols when dialing.
+- Deregister all function causes a crash.
+- Twinkle crashes at startup in CLI mode.
+- Twinkle may freeze when an ALSA error is detected when starting
+ the ringback tone and the outgoing call gets answered very fast.
+- User profile editor did not allow spaces in a user name.
+
+New RFC's
+---------
+RFC 3608 - Session Initiation Protocol (SIP) Extension Header Field
+ for Service Route Discovery During Registration
+
+
+24 august 2008 - 1.3.2
+======================
+- Fix in non-KDE version for gcc 4.3
+
+23 august 2008 - 1.3.1
+======================
+- Disable file attachment button in message window when destination
+ address is not filled in
+- Updated russian translation
+
+Build fixes
+-----------
+- Fixes for gcc 4.3 (missing includes)
+- non-KDE version failed to build
+
+
+18 august 2008 - 1.3
+====================
+- Send file attachment with instant message.
+- Show timestamp with instant messages.
+- Instant message composition indication (RFC 3994).
+- Persistent TCP connections with keep alive.
+- Do not try to send SIP messages larger than 64K via UDP.
+- Integration with libzrtcpp-1.3.0
+- Xsession support to restore Twinkle after system shutdown/startup.
+- Call snd_pcm_state to determine jitter buffer exhaustion (some ALSA
+ implementations gave problems with the old method).
+- SDP parser allows SDP body without terminating CRLF.
+- Russian translation.
+
+Bug fixes
+---------
+- SIP parser did not allow white space between header name and colon.
+- With "send in-dialog requests to proxy" enabled and transport
+ mode set to "auto", in-dialog requests are wrongly sent via TCP.
+- Crash when a too large message is received.
+- Comparison of authentication parameters (e.g. algorithm) were case-sensitive.
+ These comparisons must be case-insensitive.
+- SDP parser could not parse other media transports than RTP/AVP.
+- Twinkle sent 415 response instead of 200 OK on in-dialog INFO without body.
+- Twinkle responds with 513 Message too large on an incoming call.
+- ICMP error on STUN request causes Twinkle to crash.
+- Add received-parameter to Via header of an incoming request if it contains
+ an empty rport parameter (RFC 3581)
+- Twinkle did not add Contact header and copy Record-Route header
+ to 180 response.
+
+New RFC's
+---------
+RFC 3994 - Indication of Message Composition for Instant Messaging
+
+
+8 march 2008 - 1.2
+==================
+- SIP over TCP
+- Automatic selection of IP address.
+ * On a multi-homed machine you do not have to select an IP address/NIC
+ anymore.
+- Support for sending a q-value in a registration contact.
+- Send DTMF on an early media stream.
+- Choose auth over auth-int qop when server supports both for authentication.
+ This avoids problems with SIP ALGs.
+- Support tel-URI in From and To headers in incoming SIP messages.
+- Print a log rotation message at end of log when a log file is full.
+- Remove 20 character limit on profile names.
+- Reject an incoming MESSAGE with 603 if max. sessions == 0
+- Delivery notification when a 202 response is received on a MESSAGE.
+
+Bug fixes
+---------
+- When you deactivate a profile that has MWI active, but MWI subscription failed,
+ and subsequently activate this profile again, then Twinkle does not subscribe to
+ MWI.
+- The max redirection value was always set to 1.
+- Leading space in the body of a SIP message causes a parse failure
+- Twinkle crashes with SIGABRT when it receives an INVITE with
+ a CSeq header that contains an invalid method.
+- Latest release of lrelease corrupted translation files.
+- Twinkle crashes on 'twinkle --cmd line'
+- If an MWI NOTIFY does not contain a voice msg summary, twinkle
+ shows a random number for the amount of messages waiting.
+- Depending on the locale Twinkle encoded a q-value with a comma
+ instead of a dot as decimal point.
+
+Build changes
+-------------
+- Modifications for gcc 4.3.
+- Remove fast sequence of open/close calls for ALSA to avoid
+ problems with bluez.
+
+
+21 july 2007 - 1.1
+==================
+- French translation
+- Presence
+- Instant messaging
+- New CLI commands: presence, message
+
+Bug fixes
+---------
+- If a session was on-hold and Twinkle received a re-INVITE without
+ SDP, it would offer SDP on-hold in the 200 OK, instead of a brand
+ new SDP offer.
+- Twinkle refused to change to another profile with the same user name
+ as the current active profile.
+- ICMP processing did not work most times (uninitialized data).
+- Replace strerror by strerror_r (caused rare SIGSEGV crashes)
+- Fix deadlock in timekeeper (caused rare freezes)
+
+New RFC's
+---------
+RFC 3428 - Session Initiation Protocol (SIP) Extension for Instant Messaging
+RFC 3856 - A Presence Event Package for the Session Initiation Protocol (SIP)
+RFC 3863 - Presence Information Data Format (PIDF)
+RFC 3903 - Session Initiation Protocol (SIP) Extension
+ for Event State Publication
+
+
+19 may 2007 - 1.0.1
+===================
+- Czech translation
+- Check on user profiles having the same contact name at startup.
+- When comparing an incoming INVITE request-URI with the contact-name,
+ ignore the host part to avoid NAT problems.
+- A call to voice mail will not be attached to the "redial" button.
+- Added voice mail entry to services and systray menu.
+- New command line options: --show, --hide
+- TWINKLE_LINE environment variable in scripts. This variable contains
+ the line number (starting at 1) associated with a trigger.
+- Preload KAddressbook at startup.
+- Allow multiple occurrences of the display_msg parameter in the incoming call
+ script to create multi-line messages.
+- Handle SIP forking and early media interaction
+
+Bug fixes
+---------
+- Fix conference call
+- If lock file still exists when you start Twinkle, Twinkle asks
+ if it should start anyway. When you click 'yes', Twinkle does not start.
+- Audio validation opened soundcard in stereo instead of mono
+- When quitting Twinkle while the call history window is open, a segfault occurs
+- When an incoming call is rejected when only unsupported codecs are offered,
+ it does not show as a missed call in the call history.
+- Segfault when the remote party establishes an early media session without
+ sending a to-tag in the 1XX response (some Cisco devices).
+- in_call_failed trigger was not called when the call failed before ringing.
+- Escape double quote with backslash in display name.
+- On some system Twinkle occasionally crashed at startup with the following
+ error: Xlib: unexpected async reply
+
+Build Changes
+-------------
+- Remove AC_CHECK_HEADERS([]) from configure.in
+- Configure checks for lrelease.
+
+Other
+-----
+- Very small part of the comments has been formatted now for automatic
+ documentation generation with doxygen.
+
+
+22 jan 2007 - 1.0
+=================
+- Local address book
+- Message waiting indication (MWI)
+ * Sollicted MWI as specified by RFC 3842
+ * Unsollicited MWI as implemented by Asterisk
+- Voice mail speed dial
+- Call transfer with consultation
+ * This is a combination of a consultation call on the other line
+ followed by a blind transfer.
+- Attended call transfer
+ * This is a combination of a consultation call on the other line
+ followed by a replacement from B to C of the call on the first line.
+ This is only possible if the C-party supports "replaces".
+ If "replaces" is not supported, then twinkle automatically falls
+ back to "transfer with consultation".
+- User identity hiding
+- Multi language support
+ This version contains Dutch and German translations
+- Send BYE when a CANCEL/2XX INVITE glare occurs.
+- When call release was not immediate due to network problems or protocol errors,
+ the line would be locked for some time. Now Twinkle releases a call in the
+ background immediately freeing the line for new calls.
+- Escape reserved symbols in a URI by their hex-notation (%hex).
+- Changed key binding for Bye from F7 to ESC
+- When a lock file exists at startup, Twinkle asks if you want to override it
+- New command line options: --force, --sip-port, --rtp-port
+- Ring tone and speaker device list now also shows playback only devices
+- Microphone device list now also shows capture only devices
+- Validate audio device settings on startup, before making a call, before
+ answering a call.
+- SIP_FROM_USER, SIP_FROM_HOST, SIP_TO_USER, SIP_TO_HOST variables for call scripts.
+- display_msg parameter added to incoming call script
+- User profile options to indicate which codec preference to follow
+- Twinkle now asks permission for an incoming REFER asynchronously. This
+ prevents blocking of the transaction layer.
+- Highlight missed calls in call history
+- Support for G.726 ATM AAL2 codeword packing
+- replaces SIP extension (RFC 3891)
+- norefesub SIP extension (RFC 4488)
+- SIP parser supports IPv6 addresses in SIP URI's and Via headers
+ (Note: Twinkle does not support transport over IPv6)
+- Support mid-call change of SSRC
+- Handling of SIGCHLD, SIGTERM and SIGINT on platforms implementing
+ LinuxThreads instead of NPTL threading (e.g. sparc)
+
+Bug fixes
+---------
+- Invalid speex payload when setting ptime=30 for G.711
+- When editing the active user profile via File -> Change User -> Edit
+ QObject::connect: No such slot MphoneForm::displayUser(t_user*)
+- 32 s after call setup the DTMF button gets disabled.
+- 4XX response on INVITE does not get properly handled.
+ From/To/Subject labels are not cleared. No call history record is made.
+- The dial combobox accepted a newline through copy/past. This corrupted
+ the system settings.
+- When a far-end responds with no supported codecs, Twinkle automatically
+ releases the call. If the far-end sends an invalid response on this
+ release and the user pressed the BYE button, Twinkle crashed.
+- When using STUN the private port was put in the Via header instead of
+ the public port.
+- Twinkle crashes once in a while, while it is just sitting idle.
+
+Build changes
+-------------
+- If libbind exists then link with libbind, otherwise link with libresolv
+ This solves GLIBC_PRIVATE errors on Fedora
+- Link with libboost_regex or libboost_regex-gcc
+
+New RFC's
+---------
+RFC 3323 - A Privacy Mechanism for the Session Initiation Protocol (SIP)
+RFC 3325 - Private Extensions to the Session Initiation Protocol (SIP) for
+ Asserted Identity within Trusted Networks
+RFC 3842 - A Message Summary and Message Waiting Indication Event Package
+ for the Session Initiation Protocol (SIP)
+RFC 3891 - The Session Initiation Protocol (SIP) "Replaces" Header
+RFC 4488 - Suppression of Session Initiation Protocol (SIP)
+ REFER Method Implicit Subscription
+
+
+01 oct 2006 - Release 0.9
+=========================
+- Supports Phil Zimmermann's ZRTP and SRTP for
+ secure voice communication.
+ ZRTP/SRTP is provided by the latest version (1.5.0) of the
+ GNU ccRTP library.
+ The implementation is interoperable with Zfone beta2
+- SIP INFO method (RFC 2976)
+- DTMF via SIP INFO
+- G.726 codec (16, 24, 32 and 48 kbps modes)
+- Option to hide display
+- CLI command "answerbye" to answer an incoming or hangup an established call
+- Switch lines from system tray menu
+- Answer or reject a call from the KDE systray popup on incoming call
+- Icons to indicate line status
+- Default NIC option in system settings
+- Accept SDP offer without m= lines (RFC 3264 section 5, RFC 3725 flow IV)
+
+Bug fixes
+---------
+- t_audio::open did not return a value
+- segmentation fault when quitting Twinkle in transient call state
+- Twinkle did not accept message/sipfrag body with a single CRLF at the end
+- user profile could not be changed on service redirect dialog
+- Twinkle did not react to 401/407 authentication challenges for
+ PRACK, REFER, SUBSCRIBE and NOTIFY
+
+Build changes
+-------------
+- For ZRTP support you need to install libzrtpcpp first. This library
+ comes as an extension library with ccRTP.
+
+
+09 jul 2006 - Release 0.8.1
+===========================
+- Removed iLBC source code from Twinkle. To use iLBC you can
+ link Twinkle with the ilbc library (ilbc package). When you
+ have the ilbc library installed on your system, then Twinkle's
+ configure script will automatically setup the Makefiles to
+ link with the library.
+
+Bug fixes
+---------
+- Name and photo lookups in KAddressbook on incoming calls may
+ freeze Twinkle.
+
+Build improvements
+------------------
+- Added missing includes to userprofile.ui and addressfinder.h
+- Configure has new --without-speex option
+
+
+01 jul 2006 - Release 0.8
+=========================
+- iLBC
+- Make supplementary service settings persistent
+- Lookup name in address book for incoming call
+- Display photo from address book of caller on incoming call
+- Number conversion rules
+- Always popup systray notification (KDE only) on incoming call
+- Add organization and subject to incoming call popup
+- New call script trigger points: incoming call answered, incoming call failed,
+ outgoing call, outgoing call answered, outgoing call failed, local release,
+ remote release.
+- Added 'end' parameter for the incoming call script
+- Option to provision ALSA and OSS devices that are not in the standard list
+ of devices.
+- Option to auto show main window on incoming call
+- Resized the user profile window such that it fits on an 800x600 display
+- Popup the user profile selection window, when the SIP UDP port is occupied
+ during startup of Twinkle, so the user can change to another port.
+- Skip unsupported codecs in user profile during startup
+
+Bug fixes
+---------
+- Sometimes the NAT discovery window never closed
+- When RTP timestamps wrap around some RTP packets may be discarded
+- When the dial history contains an entry of insane length, the
+ main window becomes insanely large on next startup
+- On rare occasions, Twinkle could respond to an incoming call for
+ a deactivated user profile.
+- Credentials cache did not get erased when a failure response other
+ than 401/407 was received on a REGISTER with credentials.
+- G.711 enocders amplified soft noise from the microphone.
+
+Newly supported RFC's
+---------------------
+RFC 3951 - Internet Low Bit Rate Codec (iLBC)
+RFC 3952 - Real-time Transport Protocol (RTP) Payload Format
+ for internet Low Bit Rate Codec (iLBC) Speech
+
+Build notes
+-----------
+- New dependency on libboost-regex (boost package)
+
+
+07 may 2006 - Release 0.7.1
+===========================
+- Check that --call and --cmd arguments are not empty
+- When DTMF transport is "inband", then do not signal RFC2833 support in SDP
+
+Bug fixes
+---------
+- CLI and non-KDE version hang when stopping ring tone
+- The GUI allowed payload type 96-255 for DTMF and Speex, while
+ maximum value is only 127
+- When a dynamic codec change takes place at the same time as a re-INVITE
+ Twinkle sometimes freezes.
+- Sending RFC 2833 DTMF events fails when codec is speex-wb or speex-uwb
+
+
+29 apr 2006 - Release 0.7
+=========================
+- Speex support (narrow, wide and ultra wide band)
+- Support for dynamic payload numbers for audio codecs in SDP
+- Inband DTMF (option for DTMF transport in user profile)
+- UTF-8 support to properly display non-ASCII characters
+- --cmd command line option to remotely execute CLI commands
+- --immediate command line option to perform --call and --cmd without user
+ confirmation.
+- --set-profile command line option to set the active profile.
+- Support "?subject=<subject>" as part of address for --call
+- The status icon are always displayed: gray -> inactive, full color -> active
+- Clicking the registration status icon fetches current registration status
+- Clicking the service icons enables/disables the service
+- Fancier popup from KDE system tray on incoming call.
+- Popup from system tray shows as long as the phone is ringing.
+- Reload button on address form
+- Remove special phone number symbols from dialed strings.
+ This option can be enabled/disabled via the user profile.
+- Remove duplicate entries from the dial history drop down box
+- Specify in the user profile what symbols are special symbols to remove.
+- Changed default for "use domain to create unique contact header value" to
+ "no"
+- New SIP protocol option: allow SDP change in INVITE responses
+- Do not ask username and password when authentication for an
+ automatic re-regsitration fails. The user may not be at his desk, and
+ the authentication dialog stalls Twinkle.
+- Ask authentication password when user profile contains authentication
+ name, but no password.
+- Improved handling of socket errors when interface goes down temporarily.
+
+Bug fixes
+---------
+- If the far end holds a call and then resumes a call while Twinkle has
+ been put locally on-hold, then Twinkle will start recording sound from
+ the mic and send it to the far-end while indicating that the call is
+ still on-hold.
+- Crash on no-op SDP in re-INVITE
+- Twinkle exits when it receives SIGSTOP followed by SIGCONT
+- call release cause in history is incorrect for incoming calls.
+
+Build improvements
+------------------
+- Break dependency on X11/xpm.h
+
+
+26 feb 2006 - Release 0.6.2
+===========================
+- Graceful termination on reception of SIGINT and SIGTERM
+
+Bug fixes
+---------
+- If the URI in a received To-header is not enclosed by '<' and '>', then
+ the tag parameter is erronesouly parsed as a URI parameter instead of a
+ header parameter. This causes failing call setup, tear down, when
+ communicating with a far-end that does not enclose the URI in angle
+ brackets in the To-header.
+- Function to flush OSS buffers flushed a random amount of samples that
+ could cause sound clipping (at start of call and after call hold) when
+ using OSS.
+- In some cases Twinkle added "user=phone" to a URI when the URI already
+ had a user parameter.
+
+
+11 feb 2006 - Release 0.6.1
+===========================
+- action=autoanswer added to call script actions
+- Performance improvement of --call parameter
+- Synchronized dial history drop downs on main window and call dialog
+- Dial history drop down lists are stored persistently
+- Redial information is stored persistently
+
+Bug fixes
+---------
+- When using STUN Twinkle freezes when making a call and the STUN
+ server does not respond within 200 ms (since version 0.2)
+- Some malformed SIP messages triggered a memory leak in the
+ parser code generated by bison (since version 0.1)
+- The lexical scanner jammed on rubbish input (since version 0.1)
+
+
+05 feb 2006 - Release 0.6
+=========================
+- Custom ring tones (package libsndfile is needed)
+- Twinkle can call a user defineable script for each incoming call.
+ With this script the user can:
+ * reject, redirect or accept a call
+ * define a specific ring tone (distinctive ringing)
+- Missed call indication
+- Call directly from the main window
+- DTMF keys can by typed directly from the keyboard at the main window.
+ Letters are converted to the corresponding digits.
+- Letters can be typed in the DTMF window. They are converted to digits.
+- Call duration in call history
+- Call duration timer while call is established
+- Added --call parameter to command line to instruct Twinkle to make
+ a call
+- Increased expiry timer for outgoing RTP packets to 160 ms
+ With this setting slow sound cards should give better sound quality
+ for the mic.
+- System setting to disable call waiting.
+- System setting to modify hangup behaviour of 3-way call. Hang up both
+ lines or only the active line.
+- Replace dots with underscores in contact value
+- Silently discard packets on the SIP port smaller than 10 bytes
+- User profile option to disable the usage of the domain name in the
+ contact header.
+- Graceful release of calls when quitting Twinkle
+- Changed call hold default from RFC2543 to RFC3264
+
+Bug fixes
+---------
+- An '=' in a value of a user profile or system settings parameter
+ caused a syntax error
+- If a default startup profile was renamed, the default startup list
+ was not updated
+- When call was put on-hold using RFC2543 method, the host in the
+ SDP o= line was erroneously set to 0.0.0.0
+- When a response with wrong tags but correct branch was received, a
+ line would hang forever (RFC 3261 did not specify this scenario).
+- If far end responds with 200 OK to CANCEL, but never sends 487 on
+ INVITE as mandated by RFC 3261, then a line would hang forever
+- CPU load was sometimes excessive when using ALSA
+
+
+01 jan 2006 - Release 0.5
+=========================
+- Run multiple user profiles in parallel
+- Add/remove users while Twinkle is running
+- The SIP UDP port and RTP port settings have been moved from the user
+ profile to system settings. Changes of the default values in the user
+ profile will be lost.
+- DNS SRV support for SIP and STUN
+- ICMP processing
+- SIP failover on 503 response
+- SIP and STUN failover on ICMP error
+- When a call is originated from the call history, copy the subject to the
+ call window (prefixed with "Re:" when replying to a call).
+- Remove '/' from a phone number taken from KAddressbook. / is used in
+ Germany to separate the area code from the local number.
+- Queue incoming in-dialog request if ACK has not been received yet.
+- Clear credentials cache when user changes realm, username or password
+- Added micro seconds to timestamps in log
+- Detecting a soundcard playing out at slightly less than 8000 samples per
+ second is now done on the RTP queue status. This seems to be more reliable
+ than checking the ALSA sound buffer filling.
+- OSS fragment size and ALSA period size are now changeable via the system
+ settings. Some soundcard problems may be solved by changing these values.
+- Default ALSA period size for capturing lowered from 128 to 32. This seems
+ to give better performance on some sound cards.
+
+Bug fixes
+---------
+- With certain ALSA settings (eg. mic=default, speaker=plughw), the ALSA
+ device got locked up after 1 call.
+- The ports used for NAT discovery via STUN stayed open.
+- When a STUN transaction for a media port failed, the GUI did not clear
+ the line information fields.
+- Sending DTMF events took many unnecessary CPU cycles
+- Parse failure when Server or User-Agent header contained comment only
+
+Newly supported RFC's
+---------------------
+RFC 2782 - A DNS RR for specifying the location of services (DNS SRV)
+RFC 3263 - Session Initiation Protocol (SIP): Locating SIP Servers
+
+
+28 nov 2005 - Release 0.4.2
+===========================
+- Microphone noise reduction (can be disabled in system settings)
+- System tray icon shows status of active line and enabled services
+- Call history option added to system tray menu
+
+Bug fixes
+---------
+- Twinkle crashes at startup when the systray icon is disabled in the system settings.
+- Line stays forever in dialing state when pressing ESC in the call window
+
+
+19 nov 2005 - Release 0.4.1
+===========================
+- Fixed build problems with gcc-4.0.2 and qt3-3.4.4
+
+18 nov 2005 - Release 0.4
+=========================
+- Interface to KAddressbook
+- History of incoming and outgoing calls (successful and missed calls)
+- History of 10 last calls on call dialog window for redialling
+- Call and service menu options added to KDE sys tray icon
+- Allow a missing mandatory Expires header in a 2XX response on SUBSCRIBE
+- Big Endian support for sound playing (eg. PPC platforms)
+- System setting to start Twinkle hidden in system tray
+- System setting to start with a default profile
+- System setting to start on a default IP address
+- Command line option (-i) for IP address
+
+Bug fixes
+---------
+- send a 500 failure response on a request that is received out of order
+ instead of discarding the request.
+- 64bit fix in events.cpp
+- race condition on starting/stopping audio threads could cause a crash
+- segmentation fault when RTP port could not be opened.
+- CLI looped forever on reaching EOF
+- 64bit fix in events.cpp
+- ALSA lib pcm_hw.c:590:(snd_pcm_hw_pause) SNDRV_PCM_IOCTL_PAUSE failed
+- sometimes when quitting Twinkle a segmentation fault occurred
+
+Build improvements
+------------------
+- Removed platform dependent code from stunRand() in stun.cxx
+- It should be possible to build Twinkle without the KDE addons on a
+ non-KDE system
+- new option --without-kde added to configure to build a non-KDE version
+ of Twinkle
+
+
+22 oct 2005 - Release 0.3.2
+===========================
+- Fixed several build problams with KDE include files and
+ libraries.
+
+If you already succesfully installed release 0.3.1 then there is
+no need to upgrade to 0.3.2 as there is no new functionality.
+
+16 oct 2005 - Release 0.3.1
+===========================
+This is a minor bug fix release.
+
+Bug fixes:
+----------
+- Command line options -f and -share were broken in release 0.3
+ This release fixes the command line options.
+
+
+09 oct 2005 - Release 0.3
+=========================
+
+New functionality:
+------------------
+- ALSA support
+- System tray icon
+- Send NAT keep alive packets when Twinkle sits behind a symmetric firewall
+ (discovered via STUN)
+- Allow missing or wrong Contact header in a 200 OK response on a REGISTER
+
+Bug fixes:
+----------
+- Hostnames in Via and Warning headers were erroneously converted to lower case.
+- t_ts_non_invite::timeout assert( t==TIMER_J ) when ACK is received
+ for a non-INVITE request that had INVITE as method in the CSeq header.
+- The SIP/SDP parser accepted a port number > 65535. This caused an assert
+- Segmentation fault on some syntax errors in SIP headers
+- Line got stuck when CSeq sequence nr 0 was received. RFC 3261 allows 0.
+- With 100rel required, every 1XX after the first 1XX response were discarded.
+- Fixed build problems on 64-bit architectures.
+- Dead lock due to logging in UDP sender.
+- Segmentation fault when packet loss occurred while the sequence
+ number in the RTP packets wrapped around.
+- Route set was not recomputed on reception of a 2XX response, when a 1XX
+ repsonse before already contained a Record-Route header.
+
+
+30 jul 2005 - Release 0.2.1
+===========================
+
+New functionality:
+------------------
+- Clear button on log view window.
+
+Bug fixes:
+----------
+- The system settings window confused the speaker and mic settings.
+- Log view window sometimes opened behind other windows.
+- Segmentation fault when SUBSCRIBE with expires=0 was received to end
+ a refer subscription.
+- When a call transfer fails, the original call is received. If the line
+ for this call is not the active call however, the call should stay
+ on-hold.
+- On rare occasions a segmentation fault occurred when the ring tone
+ was stopped.
+- Log view window sometimes caused deadlock.
+
+
+24 jul 2005 - Release 0.2
+=========================
+
+New functionality:
+------------------
+- STUN support for NAT traversal
+- Blind call transfer service
+- Reject call transfer request
+- Auto answer service
+- REFER, NOTIFY and SUBSCRIBE support for call transfer scenario's
+ * REFER is sent for blind call transfer. Twinkle accpets incoming
+ NOTIFY messages about the transfer progress.
+ Twinkle can send SUBSCRIBE to extend refer event subscription
+ * Incoming REFER within dialog is handled by Twinkle
+ Twinkle sends NOTIFY messages during transfer.
+ Incoming SUBSCRIBE to extend refer event subscription is granted.
+- Retry re-INVITE after a glare (491 response, RFC 3261 14.1)
+- Respond with 416 if a request with a non-sip URI is received
+- Multiple sound card support for playing ring tone to a different
+ device than speech
+- The To-tag in a 200 OK on a CANCEL was different than the To-tag in a provisional
+ response on the INVITE. RFC 3261 recommends that these To-tags are the same.
+ Twinkle now uses the same To-tag.
+- Show error messages to user when trying to submit invalid values on the
+ user profile
+- DTMF volume configurable via user profile
+- Log viewer
+- User profile wizard
+- Help texts for many input fields (e.g. in user profile). Help can be accessed
+ by pressing Ctrl+F1 or using the question mark from the title bar.
+
+Bug fixes:
+----------
+- A retransmission of an incoming INVITE after a 2XX has been sent
+ was seen as a new INVITE.
+- If an OPTIONS request timed out then the GUI did not release its
+ lock causing a deadlock.
+- If the URI in a To, From, Contact or Reply-To header is not
+ enclosed by < and >, then the parameters (separated by a semi-colon)
+ belong to the header, NOT to the URI.
+ They were parsed as parameters of the URI. This could cause the
+ loss of a tag-parameter causing call setup failures.
+- Do not resize window when setting a long string in to, from or subject
+
+Newly supported RFC's
+---------------------
+RFC 3265 - Session Initiation Protocol (SIP)-Specific Event Notification
+RFC 3420 - Internet Media Type message/sipfrag
+RFC 3489 - Simple Traversal of User Datagram Protocol (UDP)
+ Through Network Address Translators (NATs)
+RFC 3515 - The Session Initiation Protocol (SIP) Refer Method
+RFC 3892 - The Session Initiation Protocol (SIP) Referred-By Mechanism
+
+
+27 apr 2005 - Release 0.1
+=========================
+
+First release of Twinkle, a SIP VoIP client.
+
+- Basic calls
+- 2 call appearances (lines)
+- Call Waiting
+- Call Hold
+- 3-way conference calling
+- Mute
+- Call redirection on demand
+- Call redirection unconditional
+- Call redirection when busy
+- Call redirection no answer
+- Reject call redirection request
+- Call reject
+- Do not disturb
+- Send DTMF digits to navigate IVR systems
+- NAT traversal through static provisioning
+- Audio codecs: G.711 A-law, G.711 u-law, GSM
+
+Supported RFC's
+---------------
+- RFC 2327 - SDP: Session Description Protocol
+- RFC 2833 - RTP Payload for DTMF Digits
+- RFC 3261 - SIP: Session Initiation Protocol
+- RFC 3262 - Reliability of Provisional Responses in SIP
+- RFC 3264 - An Offer/Answer Model with the Session Description Protocol (SDP)
+- RFC 3581 - An extension to SIP for Symmetric Response Routing
+- RFC 3550 - RTP: A Transport Protocol for Real-Time Applications
+
+RFC 3261 is not fully implemented yet.
+
+- No TCP transport support, only UDP
+- No DNS SRV support, only DNS A-record lookup
+- Only plain SDP bodies are supported, no multi-part MIME or S/MIME
+- Only sip: URI support, no sips: URI support