Linphone 3.5.2000

Linphone is an easy to use, handy internet phone or Voice Over IP phone (VoIP). This tool will enable you to communicate freely with people over the internet, with voice, video, and text instant messaging.

Linphone 3.5.2000

Linphone is an easy to use, handy internet phone or Voice Over IP phone (VoIP). 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.

   



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