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Chan-SCCP channel driver for Asterisk
Analyzed 20 minutes ago
Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like.
Licenses:
GNU_Gener...
FreedomBox
Analyzed about 24 hours ago
FreedomBox, a Debian pure blend, is a 100% free software self-hosting web server to deploy social applications on small machines. It provides online communication tools respecting your privacy and data ownership.
Licenses:
AGPL3_or_..., gpl3
Sippy SIP B2BUA
S
Analyzed 1 day ago
Sippy B2BUA is a RFC3261-compliant SIP Back-to-back user agent (B2BUA).
Unlike a SIP proxy server, which only maintains transaction state, the B2BUA maintains complete call state and participates in all call requests. For this reason it can perform number of functions that are not possible to
luasofia
Analyzed about 7 hours ago
LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.
Licenses:
lgpl21_or...
ice4j
Analyzed 1 day ago
A Java implementation of the STUN and TURN and ICE protocols. This project is a continuation of the Stun4J initiative. The goal of the project is to provide VoIP Java developers with a means of handling NATs and firewalls.
The project is a joint effort of the SIP Communicator and TurnServer open
Newfies-Dialer
No analysis available
Intelligent dialer for Freeswitch
This project aims to create a dialer application to deliver message to a high amount of users. This is not a telemarketing application, we aim to provide an easy simple dialer system to reach community of users.
You can easily manage your campaign, phonebooks
Mostly written in language not available
Licenses:
mozilla_p...
WebRTC
W
Analyzed 1 day ago
WebRTC is an open source project that enables web browsers with Real-Time Communications (RTC) capabilities via simple Javascript APIs. The WebRTC components have been optimized to best serve this purpose.
WebRTC implements WHATWG's proposal for video conferencing on the web.
Licenses:
No declared licenses
baresip
Analyzed about 21 hours ago
Baresip is a portable and modular SIP User-Agent with audio and video support.
Features:
Audio codecs: AMR, BV32, G.711, G.722, G.722.1, G.726, GSM, iLBC, iSAC, L16, OPUS, Silk, Speex
Video codecs: H.263, H.264, H.265, MPEG4, VP8
Audio drivers: Alsa, Coreaudio, Gstreamer, OpenSLES, OSS
Licenses:
BSD-3-Clause
GNU Gatekeeper (GnuGk)
Analyzed about 21 hours ago
The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions.