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diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 0000000..c8916bb --- /dev/null +++ b/ChangeLog @@ -0,0 +1,796 @@ +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 |