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Open Source Internet Phone

Linphone is an internet phone or Voice Over IP phone (VoIP).

  • With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging.
  • Linphone makes use of the SIP protocol , an open standard for internet telephony. You can use Linphone with any SIP VoIP operator, including our free SIP audio/video service.
  • linphone is free-software (or open-source), you can download and redistribute it freely.
  • Linphone is available for desktop computers: Linux, Windows, MacOSX, and for mobile phones: Android, iPhone, Blackberry.
  • SIP user agent compliant with RFC3261
  • Supports IPv6
  • Digest authentication
  • Supports multiple calls simultaneously with call management features: hold on with music, resume, transfer…
  • Multiple SIP proxy support: registrar, proxies, outbound proxies
  • Text instant messaging with delivery notification 
  • Presence using the SIMPLE standard in peer to peer mode
  • DTMF (telephone tones) support using SIP INFO or RFC2833
  • Audio with the following codecs: speex (narrow band and wideband), G711 (ulaw,alaw), GSM, G722. Through additionals plugins, it also supports AMR-NB, SILK, G729 and iLBC.
  • Video with codecs: VP8 (WebM), H263, H263-1998, MPEG4, theora and H264 (thanks to a plugin based on x264), with resolutions from QCIF(176×144) to SVGA(800×600) provided that network bandwidth and cpu power are sufficient.
  • Audio conferencing
  • Supports SRTP and zRTP (encryption of voice and video)
  • ICE support (RFC5246) to allow peer to peer audio & video connections without media relay server, but for other products you will need choosing a dedicated server.
  • Supports any webcam with a V4L or V4L2 driver under linux and Directshow driver on windows
  • Acoustic echo cancelation using the great echo canceller available in libspeexdsp (works not only with speex codec of course)
  • Efficient bandwidth management: the bandwidth limitations are signaled using SDP (b=AS…), resulting in audio and video session established with bitrates that fits the user’s network capabilities.
  • Low bandwidth mode: make audio calls over EDGE
  • Adaptive audio & video bitrate algorithm to adapt to available network bandwidth.
  • Sound backends: 
    • Linux: ALSA, OSS, PulseAudio
    • Windows: waveapi
    • MacOSX: HAL Audio Unit
    • iPhone: VoiceProcessing AudioUnit with built-in echo cancellation
    • Android sound system
    • JSR135 on BlackBerryCan use plugins: to add new codecs, or new core functionalities, such as remote directory search of sip addresses for example.


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