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Original author(s) Savoir-faire Linux Inc.
Stable release 1.3.0 / 14 January 2014; 5 months ago (2014-01-14)
Development status Stable + Development
Written in C / C++
Operating system Linux
Platform i386, amd64, powerpc, sparc
Available in English, French, German, Spanish, Russian, Chinese, Italian, Vietnamese
Type VoIP, telephony, softphone, sip
License GNU General Public License 3
Website sflphone.org

SFLphone is an open-source SIP/IAX2 compatible softphone for Linux. SFLphone is free software released under the GNU General Public License. Packages are available for all major distributions including Debian, openSUSE, Fedora, Mandriva and the latest Ubuntu releases.[1] It is also available in Maemo Community.[2]

SFLphone is one of the few softphones under Linux to support PulseAudio out of the box. The Ubuntu documentation recommends it for enterprise use because of features like conferencing and attended call transfer.[3] It has been named by CIO magazine among the 5 open source VoIP softphones to watch.[4]

It is maintained by Savoir-faire Linux.[5][6]

French documentation is available on Ubuntu-fr website.[7]

SFLphone design[edit]

SFLphone is based on a MVC model: Daemon and client are two separate processes that communicate through D-Bus. The Model is the daemon. Daemon handles all the processing including communication layer (SIP/IAX), audio capture and playback, etc. ... View is the GTK+ graphical user interface. Controller is D-Bus that enables communication between client and server.

Features list[edit]

  • SIP and IAX compatible
  • Unlimited number of calls
  • Call recording
  • Attended call transfer
  • Call hold
  • Multiple audio conferencing (from 0.9.7 version)
  • TLS and ZRTP support (from 0.9.7 version)
  • Audio codecs supported: G711u, G711a, GSM, Speex (8, 16, 32 kHz), CELT, G.722
  • Multiple SIP accounts support
  • STUN support per account (0.9.7)
  • DTMF support (SIP INFO)
  • Instant messaging
  • Call history + search feature
  • Silence detection with Speex audio codec
  • Account assistant wizard
  • Central server providing free SIP/IAX account
  • SIP presence subscription
  • Video multiparty conferencing (EXPERIMENTAL)
  • Multichannel audio support [EXPERIMENTAL]
  • Flac and OGG/Vorbis ringtone support
  • Desktop notification: voicemail number, incoming call, information messages
  • Minimize on start-up
  • Minimize to tray
  • not Direct IP-to-IP SIP call - P2P is not supported by IAx2 according to the documentation
  • SIP Re-invite
  • Address book support: Evolution Data Server integration (for the GNOME client), KABC integration for the KDE client
  • PulseAudio support
  • Native ALSA interface, DMix support
  • Locale settings: French, English, Russian, German, Chinese, Spanish, Italian, Vietnamese
  • Automatic opening of incoming URL

See also[edit]

External links[edit]