Ring (software)

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Ring
Ring VOIP logo.svg
Original author(s) Savoir-faire Linux Inc.
Stable release none
Preview release 1.0.0 - dab80ab8a0 / 14 March 2016; 4 months ago (2016-03-14)
Development status Beta + Development
Written in C / C++
Operating system Linux, FreeBSD, Microsoft Windows, OS X, Android
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/en

Ring (formerly SFLphone) is an open-source SIP-compatible softphone and instant messenger for Linux, Microsoft Windows, OS X and Android that can work without a central server. Ring is free software released under the GNU General Public License. Packages are available for all major Linux distributions including Debian, Fedora, and Ubuntu.[1] Separate GNOME and KDE versions are available.[2]

By adopting distributed hash table technology (as used, for instance, within the BitTorrent network), Ring creates its own network over which it can distribute directory functions, authentication and encryption across all systems connected to it.[3]

Ring is developed and maintained by Savoir-faire Linux,[4][5] a Canadian company, with the help of a global community of users and contributors; it positions itself as a potential free Skype replacement.[6] Documentation is available on Ring's Tuleap wiki.[7]

History[edit]

SFLphone was one of the few softphones under Linux to support PulseAudio out of the box. The Ubuntu documentation recommended it for enterprise use because of features like conferencing and attended call transfer.[8] In 2009, CIO magazine listed SFLphone among the top five open-source VoIP softphones to watch.[9]

One step beyond SFLphone, Ring retained SIP compatibility and support, while adding a new communication platform that does not require a centralized server to establish communication.

Design[edit]

Ring is based on a MVC model, with a daemon (the model) a client (the view) communicating. The daemon handles all the processing including communication layer (SIP/IAX), audio capture and playback, and so on. The client is a graphical user interface. D-Bus can act as the controller enabling communication between the client and the daemon.

Features[edit]

  • SIP-compatible with OpenDHT support[2][10]
  • Unlimited number of calls
  • Instant messaging
  • Searchable call history
  • Call recording[2]
  • Attended call transfer
  • Automatic call answering
  • Call holding
  • Audio and video calls with multi-party audio[2] and experimentally video conferencing[11]
  • Multi-channel audio support (experimental)
  • Streaming of video and audio files during a call
  • TLS and ZRTP support
  • Multiple[2] audio codecs supported: G711u, G711a, GSM, Speex (8, 16, 32 kHz), Opus, G.722 (silence detection supported with Speex)
  • Multiple SIP accounts support, with per-account STUN support and SIP presence subscription
  • DTMF support
  • Automatic Gain Control
  • Account assistant wizard
  • Global keyboard shortcuts
  • Flac and Vorbis ringtone support[11]
  • Desktop notification: voicemail number, incoming call, information messages
  • SIP Re-invite
  • Address book integration in GNOME and KDE
  • PulseAudio support
  • Jack Audio Connection Kit support
  • Locale settings: French, English, Russian, German, Chinese, Spanish, Italian, Vietnamese
  • Automatic opening of incoming URL

See also[edit]

References[edit]

External links[edit]