Ring (software)

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Ring VOIP logo.svg
Original author(s) Savoir-faire Linux Inc.
Stable release 1.4.1 / 18 September 2014; 12 months ago (2014-09-18)
Development status Stable + Development
Written in C / C++
Operating system Linux Windows OS X
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 ring.cx

Ring (formerly SFLphone) is an open-source SIP/IAX2 compatible softphone for Linux, Windows and OS X. Ring 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]

Ring 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.[2] It has been named by CIO magazine among the 5 open source VoIP softphones to watch.[3]

It is maintained by Savoir-faire Linux.[4][5]

French documentation is available on Ubuntu-fr website.[6]

Ring design[edit]

Ring 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 compatible
  • Unlimited number of calls
  • Call recording
  • Attended call transfer
  • Automatic call answering
  • Call hold
  • Multiple audio conferencing (from 0.9.7 version)
  • TLS and ZRTP support
  • Audio codecs supported: G711u, G711a, GSM, Speex (8, 16, 32 kHz), Opus, G.722
  • Multiple SIP accounts support
  • STUN support per account
  • DTMF 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 support
  • Jack Audio Connection Kit 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]