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Asterisk

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  Analyzed about 1 hour ago

Asterisk is a complete PBX and telephony toolkit in software. It runs on Linux, *BSD, MacOSX, and Solaris. It provides all of the features you would expect from a PBX and more as it enables developers to build customized voice applications of many types. Asterisk does voice over IP in many ... [More] protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. [Less]

835K lines of code

73 current contributors

about 16 hours since last commit

170 users on Open Hub

Very High Activity
4.375
   
0 Reviews
I Use This
Mostly written in C
Licenses: GPL-2.0+
Tags asterisk c conferencing googletalk h323 iax iax2 isdn ivr jabber jingle mgcp 12 more...

Empathy

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  Analyzed 20 days ago

Empathy consists of a rich set of reusable instant messaging widgets, and a GNOME client using those widgets. It uses Telepathy and Nokia's Mission Control, and reuses Gossip's UI. The main goal is to permit desktop integration by providing libempathy and libempathy-gtk libraries. libempathy-gtk is ... [More] a set of powerful widgets that can be embeded into any GNOME application. [Less]

90.9K lines of code

42 current contributors

3 months since last commit

105 users on Open Hub

Low Activity
3.75758
   
0 Reviews
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Mostly written in C
Licenses: GPL-2.0+
Tags aim chat client gnome google_talk icq im instant_messaging linux messaging messenger msn 6 more...

Ekiga

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  Analyzed 3 days ago

Ekiga (formely known as GnomeMeeting) is an open source VoIP and video conferencing application for GNOME. Ekiga uses both the H.323 and SIP protocols. It supports many audio and video codecs, and is interoperable with other SIP compliant software and also with Microsoft NetMeeting.

74.2K lines of code

12 current contributors

5 days since last commit

63 users on Open Hub

Very Low Activity
3.7619
   
0 Reviews
I Use This
Mostly written in C++
Licenses: GPL-2.0+
Tags client conferencing instant_messenger sip softphone video voip

FreeSWITCH

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  Analyzed 22 days ago

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a ... [More] stable telephony platform on which many applications can be developed using a wide range of free tools. [Less]

2.38M lines of code

74 current contributors

22 days since last commit

52 users on Open Hub

Very High Activity
4.86667
   
0 Reviews
I Use This
Mostly written in C
Licenses: No declared licenses
Tags c codecs communication conferencing freeswitch googletalk isdn ivr javascript jingle lua media 20 more...

Jitsi

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Claimed by Atlassian Analyzed about 22 hours ago

Jitsi (previously SIP Communicator) is a cross-platform VOIP, videoconference, desktop sharing and chat client. Protocols supported include SIP, XMPP/Google Talk/Jabber/Facebook chat, AIM/ICQ, Windows Live, Yahoo! Messenger and Bonjour. Other features include: Call recording, Call & chat ... [More] encryption, Noise suppression, Echo cancellation, File transfer, multi-user chat, Desktop streaming, Presence, Conference calls, Integration with Microsoft Outlook and Apple Address Book, Support for LDAP directories, Support for Google Contacts, On-line provisioning, Systray notifications, IPv6 support, Spell checker and many more. [Less]

362K lines of code

14 current contributors

1 day since last commit

28 users on Open Hub

Moderate Activity
4.41667
   
0 Reviews
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Mostly written in Java
Licenses: LGPL
Tags aim aol_instant_messenger bonjour callrecording chat chatting client communication conference desktop-sharing encryption facebookchat 27 more...

KDE Telepathy

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Claimed by KDE Analyzed 28 days ago

Real time Communication has traditionally been a detached feature of Desktop Computing, provided via stand-alone Instant Messaging clients with poor integration into the desktop experience. One of the primary goals of the KDE 4 series is to tighten integration between different components of the ... [More] environment. The KDE Telepathy (KTp) project aims to tackle just this. Our aims are: * To integrate Real Time Communication deeply into the KDE Workspaces and Applications * To provide a infrastructure to aid development of Collaborative features for KDE applications. [Less]

161K lines of code

11 current contributors

3 months since last commit

18 users on Open Hub

Moderate Activity
5.0
 
0 Reviews
I Use This
Mostly written in C++
Licenses: GPL-2.0+, LGPL-2.1
Tags aim bonjour desktop-sharing facebookchat filetransfer googletalk google_talk groupchat gtalk icq im instantmessaging 16 more...

OpenSIPS

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  Analyzed about 10 hours ago

OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server, is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing ... [More] engine, OpenSIPS 'unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable (modular) design. What OpenSIPS has to offer, comes in a reliable and high-performance flavour - OpenSIPS is one of the fastest SIP servers, with a throughput that confirms it as a solution up to enterprise or carrier-grade class. [Less]

543K lines of code

26 current contributors

about 13 hours since last commit

15 users on Open Hub

High Activity
4.0
   
1 Review
I Use This
Mostly written in C
Licenses: GPL-2.0+
Tags chat communication linux nat openser presence router server sip telephony video voice 1 more...

Kazoo Platform

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  Analyzed 3 days ago

Kazoo is a scalable, distributed, cloud-based telephony platform that allows you to build powerful telephony applications with a rich set of APIs. Designed to handle anything from large carrier to small countries, the Whistle infrastructure can do it all. There are no lock-ins and the software is ... [More] open-source to give you complete freedom. Services include: - Complete redundancy and failover between data centers - Complete replication of all data - Use of Map/Reduce algorithms inside NoSQL databases - Multi-master replication and caching of registrations, active channels and call lookups - Load balancing built-in - Event driven messaging for managing and using calls - A complete REST interface for implementing call flow features [Less]

263K lines of code

33 current contributors

4 days since last commit

14 users on Open Hub

High Activity
5.0
 
1 Review
I Use This
Mostly written in Erlang
Licenses: MPL-1.1
Tags 2600 2600hertz 2600hz bigcouch cloud cloudplatform cluster clustering comet communication conferencing couchdb 40 more...

blue.box project

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  Analyzed 6 days ago

blue.box is an open-source configuration and management software enabling the use of the FreeSWITCH and Asterisk switching libraries. It supports multi-tenancy, skinning and is completely open-source. It can be used with database and file replication to scale up to thousands of registered devices ... [More] and simultaneous phone calls. It can operate in the cloud or on the premise. blue.box development is hosted and funded by the 2600hz project. http://www.2600hz.org [Less]

460K lines of code

0 current contributors

about 4 years since last commit

14 users on Open Hub

Inactive
5.0
 
2 Reviews
I Use This
Mostly written in PHP
Licenses: MPL-1.1
Tags 2600 2600hertz 2600hz asterisk communication conferencing distro framework freeswitch googletalk h323 iax 21 more...

Asterisk-Java

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  Analyzed 4 days ago

Asterisk-Java, a free Java library for Asterisk PBX integration, consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and ... [More] the Manager API. The FastAGI implementation supports all commands currently available from Asterisk. The Manager API implementation supports receiving events from the Asterisk server (e.g. call progess, registered peers, channel state) and sending actions to Asterisk (e.g. originate call, agent login/logoff, start/stop voice recording). [Less]

55.1K lines of code

11 current contributors

4 days since last commit

14 users on Open Hub

Moderate Activity
4.875
   
0 Reviews
I Use This
Mostly written in Java
Licenses: Apache-2.0
Tags asterisk iax java library sip voip