[kde-announce] Twinkle 0.5

Name: Twinkle
Version: 0.5
Type: Other Software
Depend: Qt 3.x
License: GPL
Homepage: http://www.twinklephone.com
More Info:
http://www.kde-apps.org/content/show.php?content=26926

Description:
 Twinkle is a soft phone for your voice over IP
communcations using the SIP protocol. You can use
it for direct IP phone to IP phone communication
or in a network using a SIP proxy to route your
calls.
Some of the features offered are call waiting,
call hold, 3-way conference call, call transfer,
and call reject. It supports STUN or a statically
configured public IP address for NAT traversal.
When using STUN, it will send keep-alive packets
to keep NAT bindings alive.

Changelog:
 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


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