Voice over Internet Protocol (VoIP) SIP Phone
  10 reviews

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
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
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)
Latest reviews
apazhe 3 years ago

Could not work on Mint Cinnamon 17.2 64

frank_orellana 4 years ago

Could not open it in 17.2 cinnamon, uninstalled it

lassve 4 years ago

Did not load on mint 17-64,cinnamon

Oznewbie 4 years ago

Could not get it to work in Mint 17-32, Cinnamon, so removed it.

vella 7 years ago

It wasn't easy to set up for sip, not instinctive, but works perfectly, good line sound

phllbit 7 years ago

A very nice piece of software, fully featured, settings can be tricky but works perfectly on many machines

armunro 7 years ago

Used to work well on my old laptop. On my new laptop doesn't pickup the alsa sound card. Tries playing around with the settings. Think it needs some work for alsa2.

LeanderPL 7 years ago

The best SIP software! Works fine, install make profile and... it works! Działa z SIP'em z, dzwoni, odbiera, ma bardzo dobrą jakość rozmów, polecam!

Liszt 9 years ago

Good. Get some echo even with echo cancel on but not sure this can be overcome by the software

funker2010 9 years ago

is good