Open Source Internet Phone
By admin on Mar 17, 2013 in Android, C/C++, open source
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.
Signaling
- SIP user agent compliant with RFC3261
- SIP/UDP, SIP/TCP, SIP/TLS
- 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
Media
- 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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