SFLphone
Original author(s) Savoir-faire Linux Inc.
Stable release 1.4.1 / 18 September 2014; 5 months ago (2014-09-18 )
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]
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] [ 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+ or KDE graphical user interface. Controller is D-Bus that enables communication between client and server.
§ Features list [ edit ]
SIP and IAX compatibleUnlimited number of calls
Call recording
Attended call transfer
Automatic call answering
Call hold
Multiple audio conferencing (from 0.9.7 version)
TLS and ZRTP supportAudio codecs supported: G711u, G711a, GSM, Speex (8, 16, 32 kHz), Opus , G.722
Multiple SIP accounts support
STUN support per accountDTMF support (SIP INFO)Instant messaging
Call history + search feature
Silence detection with Speex audio codec
Automatic Gain Control
Account assistant wizard
Global keyboard shortcuts
SIP presence subscription
Video calls
Streaming of video and audio files during a call
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 supportJack Audio Connection Kit supportNative ALSA interface, DMix support
Locale settings: French, English, Russian, German, Chinese, Spanish, Italian, Vietnamese
Automatic opening of incoming URL
§ External links [ edit ] § References [ edit ] .svg){kind=logo}
