This layer provides support for building liblinphone and associated packages necessary for running linphonec or linphone-daemon and making successful audio and video SIP calls. This is the official layer written by Belledonne Communications, responsible of Linphone.

Setup information Mailing list

Git repository

git://git.linphone.org/meta-bc.git web repo

Last commit: 6 months, 1 week ago (master branch)

Maintainers

Dependencies

The meta-bc layer depends upon:

Recipe name Version Description
antlr3c-bc git_ac1069cf214b15b86272cbc6ded5916d7d2f85ec Linphone version of antlr3
bcg729 git_9ada79d7ff53815e85432e7442810a2fd49dbd0e Belledonne Communications G729 codec library and mediastreamer2 pluginc
bctoolbox git_01285f4b49e4c06534058aa80dc6e9567cca9b16 SIP stack from Belledonne Communications
bcunit git_d8d2f4b40209e06b400f893cce58e4c6ba73341d BCunit library
belcard git_b9e1951be4575c62e326d761a7f7c79c5cce9cb9 Belcard is a C++ library to manipulate VCard standard format
belle-sip git_d8c5e9e08b3bd6640898e46850333f1ad900c8d2 SIP stack from Belledonne Communications
belr git_fdce52526089e88c98f19b0d36483cc3d31ef9bd Belr is Belledonne Communications' language recognition library.
bzrtp git_1ab274fa74f889769a656a7650f7aca5ea48679c Opensource implementation of ZRTP keys exchange protocol
flexisip git_640cea0528fecef1c649c8b82575ed41b8976067 General purpose SIP proxy with media capabilities from Belledonne Communications
libgsm git_0f8822b5326c76bb9dc4c6b552631f51792c3982 GSM Audio Library
linphone git_AUTOINC Audio/video SIP-based IP phone (console edition)
linphone-plugins 1.0 Plugins for linphone to have additional codecs.
mediastreamer2 git_eb2af04948a1a0e0f3384f0e46c3513b0aa51e95 Powerful and lightweight streaming engine specialized for voice and video telephony applications
msamr git+wb Mediastreamer2 plugin adding support for AMR codec
msamr git Mediastreamer2 plugin adding support for AMR codec
msopenh264 git Mediastreamer2 plugin adding support for H264 codec
mswebrtc git Mediastreamer2 plugin adding support for WebRTC features (iSAC codec, AEC...)
openh264 1.5.1 Cisco Open Source H.264 Codec library which supports H.264 encoding and decoding.
opus-bc git_35b371a85bf2cf21ab4b12b5475c76a2775b25d1 Opus is a totally open, royalty-free, highly versatile audio codec
ortp git_4cea5e6e7b6da329030b4f34b66a6168b864bb1d Real-time transport protocol (RFC 3550) library
polarssl-bc git_9864c92b71b81dd1dda885eae108cc3fc9a0cf4b SSL/TLS library
sofia-sip-ua-bc git_97c1824a4769c34add85e6ff306bf3aa34116c75 Linphone version of sofia-sip
spandsp 0.0.6-pre18 A library of many DSP functions for telephony.
speex-bc 1.2rc1 Audio codec speex
srtp-bc git_d79ae95126baa3cec83097469e97525a1d9e2d50 SRTP is a security profile for RTP that adds confidentiality, message authentication, and replay protection to that protocol.
tunnel git_fbf1f7bc516cfbfa9b2eacb950b9d6c245c893fe Linphone tunnel library