Twinkle SIP Softphone for Linux


Twinkle is a softphone for your voice over IP and instant messaging 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 and messages.

Simply install an asterisk server and create an extensions. Install twinkle in your linux desktop and configure for your sip accounts, then make a calls.

Twinkle Features: 
  • 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)
  • Reject call transfer request
  • Call reject
  • Repeat last call
  • Do not disturb
  • Auto answer
  • Message Waiting Inidication
  • Voice mail speed dial
  • User defineable scripts triggered on call events
    E.g. to implement selective call reject or distinctive ringing
  • RFC 2833 DTMF events
  • Inband 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
  • Persistent TCP connections for NAT traversal
  • Missed call indication
  • 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
  • System tray icon
  • System tray menu to quickly originate and answer calls while Twinkle stays hidden
  • User defineable number conversion rules
  • Simple address book
  • Support for UDP and TCP 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
  • Identity hiding
 Audio Codecs Supported:
  • G.711 A-law (64 kbps payload, 8 kHz sampling rate)
  • G.711 μ-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)
  • iLBC (13.3 or 15.2 kbps payload, 8 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)
  • Noise reduction
  • Voice activity detection (VAD)
  • Acoustic echo control (AEC) [experimental]
Presence:

Twinkle supports basic presence functionality using a presence agent that must be offered by your SIP provider.
Only the basic online and offline presence states (PIDF) are supported. Your buddy lists are stored locally on your computer.

 Instant Messaging: 

Twinkle has basic instant messaging capabilties. You can send plain text messages. You can receive plain text or rich text (html) messages.

 Twinkle is available for Linux only (GPL license).

Article from http://www.twinklephone.com/

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