Linphone 3.5.2

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.

Core features

• SIP user agent compliant with RFC3261


• Supports multiple calls simultaneously with call management features: hold on with music, resume, transfer...


• Audio with the following codecs: speex (narrow band and wideband), G711 (ulaw,alaw), GSM. Through additionals plugins, it also supports AMR and iLBC.


• Video with codecs: H263, H263-1998, MPEG4, theora and H264 (thanks to a plugin based on x264), with resolutions from QCIF(176x144) to SVGA(800x600) provided that network bandwidth and cpu power are sufficient.


• Supports IPv6


• Supports any webcam with a V4L or V4L2 driver under linux and Directshow driver on windows


• Text instant messaging and presence (using the SIMPLE standard)


• Addressbook


• DTMF (telephone tones) support using SIP INFO or RFC2833


• Understands SIP ENUMS (sip phone numbers using the naptr DNS service, without proxy)


• Acoustic echo cancelation using the great echo canceller available in libspeexdsp (works not only with speex codec of course)


• Multiple SIP proxy support: registrar, proxies, outbound proxies


• Digest authentication


• Nat friendly: guesses NAT address for SIP messages, uses STUN for RTP streams


• Sound backends: 
  
  
  
  • Linux: ALSA, OSS, PulseAudio
  
  
  • Windows: waveapi
  
  
  • MacOSX: HAL Audio Unit
  
  
  • iPhone: VoiceProcessing AudioUnit with built-in echo cancellation
  
  
  • Android sound system
  
  
  
  


• 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.


• Can use plugins: to add new codecs, or new core functionalities, such as remote directory search of sip addresses for example.