Twinkle 1.4.2

A soft phone for your voice over IP communcations using the SIP protocol.
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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.

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

Main features:

  • 2 call appearances (lines)
  • Multiple active call identities
  • Custom ring tones (new)
  • 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
  • Reject call transfer request
  • Call reject
  • Repeat last call
  • Do not disturb
  • Auto answer
  • User defineable scripts to handle incoming calls (new) E.g. to implement selective call reject or distinctive ringing
  • Send DTMF digits (RFC 2833) to navigate IVR systems
  • STUN support for NAT traversal
  • Send NAT keep alive packets when using STUN
  • NAT traversal through static provisioning
  • Missed call indication (new)
  • History of call detail records for incoming, outgoing, successful and missed calls
  • DNS SRV support
  • Automatic failover to an alternate server if a server is unavailable
  • Other programs can originate a SIP call via Twinkle, e.g. call from address book (new)
  • System tray icon (now also on non-KDE builts)
  • System tray menu to quickly originate and answer calls while Twinkle stays hidden
  • Audio codecs:
  • Twinkle supports the following audio codecs.
  • G.711 A-law (64 kbps payload)
  • G.711 μ-law (64 kbps payload)
  • GSM (13 kbps payload)
  • For audio playing Open Sound System (OSS) is used.
  • Standards support:
  • Twinkle implements the following standards.
  • 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 3265 - Session Initiation Protocol (SIP)-Specific Event Notification (new)
  • RFC 3420 - Internet Media Type message/sipfrag (new)
  • RFC 3489 - Simple Traversal of UDP Through Network Address Translators (NATs) (new)
  • RFC 3515 - The Session Initiation Protocol (SIP) Refer Method (new)
  • RFC 3581 - An extension to SIP for Symmetric Response Routing
  • RFC 3550 - RTP: A Transport Protocol for Real-Time Applications
  • RFC 3892 - The Session Initiation Protocol (SIP) Referred-By Mechanism (new)
  • 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

last updated on:
February 26th, 2009, 7:42 GMT
license type:
GPL (GNU General Public License) 
developed by:
Michel de Boer
ROOT \ Communications \ Internet Phone
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What's New in This Release:
  • This version provides integration with Diamondcard Worldwide Communication services for making calls to regular and cell phones and sending SMS messages.
  • Furthermore, the call history now gives details on the total number of calls and call duration.
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