Twinkle 0.9 review

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Twinkle is a soft phone for your voice over IP communcations using the SIP protocol

License: GPL (GNU General Public License)
File size: 988K
Developer: Michel de Boer
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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.

Here are some key features of "Twinkle":
2 call appearances (lines)
Multiple active call identities
Custom ring tones (new)
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
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


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