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Voice over Internet Protocol (VoIP) SIP Phone

Soft-phone for making telephone calls using SIP over an IP network.

Twinkle supports direct IP phone to IP phone communication or a network using a SIP proxy to route your calls.

In addition to making basic voice calls Twinkle provides you the following features regardless of the services that your VoIP service provider might offer.

2 call appearances (lines) Multiple active call identities Custom ring tones 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 Blind call transfer Call transfer with consultation (attended call transfer) (new) Reject call transfer request Call reject Repeat last call Do not disturb Auto answer Message Waiting Indication Voice mail speed dial User definable scripts triggered on call events E.g. to implement selective call reject or distinctive ringing RFC 2833 DTMF events In-band DTMF Out-of-band DTMF (SIP INFO) STUN support for NAT traversal Send NAT keep alive packets when using STUN NAT traversal through static provisioning Missed call indication History of call detail records for incoming, outgoing, successful and missed DNS SRV support Automatic fail-over to an alternate server if a server is unavailable Other programs can originate a call via Twinkle, e.g. call from address book System tray icon System tray menu to originate and answer calls while Twinkle stays hidden User definable number conversion rules Simple address book Support for UDP and TCP (new) as transport for SIP Presence Instant messaging Simple file transfer with instant message Instant message composition indication Command line interface (CLI)

VoIP security Secure voice communication by ZRTP/SRTP MD5 digest authentication support for all SIP requests AKAv1-MD5 digest authentication support for all SIP requests (new) Identity hiding

Audio codecs G.711 A-law (64 kbps payload, 8 kHz sampling rate) G.711 u-law (64 kbps payload, 8 kHz sampling rate) GSM (13 kbps payload, 8 kHz sampling rate) Speex narrow band (15.2 kbps payload, 8 kHz sampling rate) Speex wide band (28 kbps payload, 16 kHz sampling rate) Speex ultra wide band (36 kbps payload, 32 kHz sampling rate) G.726 (16, 24, 32 or 40 kbps payload, 8 kHz sampling rate)

For all codecs the following preprocessing options are available to improve quality at the far end of a call. Automatic gain control (AGC) (new) Noise reduction (new) Voice activity detection (VAD) (new) Acoustic echo control (AEC) [experimental] (new)

Package version:1:1.4.2-2+b2


twinkle for Debian

This package has been configured to use the Debian system shared library 
libgsm, rather than libgsm included with twinkle.

 -- Mark Purcell <>, Sun, 29 May 2005 09:40:22 +0100


Twinkle is a SIP based VoIP client.

Release 0.5 notes
In this release the SIP UDP port and RTP port settings have been
moved from the user profile to the system settings. If you made
any changes to the default port values in your user profiles, then
these changes will be lost.

Library requirements
To compile Twinkle you need the following libraries:



TWINKLE(1)                  User Commands                 TWINKLE(1)

       Twinkle - Voice over Internet Protocol (VoIP) SIP Phone

       twinkle [options]

       -c     Run in command line interface (CLI) mode

       --share <dir>
              Set the share directory.

       -f <profile>
              Startup  with  a  specific  profile.  You  will not be


twinkle (1:1.4.2-2+b2) unstable; urgency=low

  * Binary-only non-maintainer upload for i386; no sou


25 february 2009 - 1.4.2
- Integration with Diamondcard Worldwide Communica


Author of Twinkle:

Michel de Boer <> designed and implemented Twinkle.



Thanks to the following people for testing and finding all
those lovely bugs:

Richard Bos
Schelte B


This package was debianized by Mark Purcell <> on
Sun, 29 May 2005 09:40:22 +0100.


Browse inside twinkle_1.4.2-2+b2_i386.deb

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