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Linphone is an easy to use, handy internet phone or Voice Over IP phone (VoIP). Mobile phone networks have Internet telephony as a direct competitor, since there are applications that offer a cost-effective alternative to keep in touch with others. Linphone is one of the software solutions in this category, relying on the Voice over IP (VoIP) protocol to allow free communication between users. Linphone allows you to customize the call notification sound file and adjust the resolution during video calls (it supports SVGA, CIF, QCIF, QVGA and more). It bundles a collection of predefined multimedia codecs that allow flawless video and audio transmissions.

This tool will enable you to communicate freely with people over the internet, with voice, video, and text instant messaging.
Linphone also complies to the SIP protocol, an open standart for internet telephony. The application should be able to interoperate with most SIP phones and proxies.

If you subscribe a VoIP to PSTN (=classic telephony) account to a telecom provider, you can reach everyone that has a "classic" phone line. However those calls are not free since PSTN networks are costly.

FEATURES:

Core features

SIP user agent compliant with RFC3261
Decoupling between liblinphone engine and graphical UI: allows to integrate linphone functionnalities in any graphical application.
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 standart)
Addressbook
DTMF (telephone tones) support using SIP INFO or RFC2833
Understands SIP ENUMS (sip phone numbers using the naptr DNS service, without proxy)
Echo cancelation using the great speex echo canceler
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: audio queues
iPhone: AudioUnit
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.
Compliant with open standarts: see the full list there.