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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]

2.47M lines of code

62 current contributors

about 12 hours since last commit

169 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 3 months 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]

90.7K lines of code

18 current contributors

5 months since last commit

102 users on Open Hub

Activity Not Available
3.75758
   
0 Reviews
I Use This
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 9 months 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

7 current contributors

10 months since last commit

62 users on Open Hub

Activity Not Available
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 29 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]

2.17M lines of code

41 current contributors

2 months since last commit

54 users on Open Hub

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 Desktop

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Claimed by Atlassian Analyzed about 3 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]

363K lines of code

11 current contributors

about 2 months since last commit

29 users on Open Hub

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

Kamailio (SIP Express Router)

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  Analyzed 2 months ago

Kamailio is an open source SIP server application formerly named OpenSER. Starting with v3.0.0 it has merged its ancestor project, respectively SIP Express Router (SER). The project is managed by its community, released under GPLv2, and focusing to build a flexible and rock solid SIP server. ... [More] Among features: full RFC3261 implementation, support for IPv4, IPv6, UDP, TCP, TLS, SCTP, DNS SRV and NAPTR, Websockets for WebRTC, least cost routing, load balancing, AAA and prepaid billing, presence server, RCS/RCSe, embedded XACP server and MSRP relay, RADIUS, XMLRPC control interface, IMS extensions, MySQL, Postgres and many other DB backends, embedded interpreters for Lua, Perl, Python, C# and other languages, IMS extensions. [Less]

1.88M lines of code

79 current contributors

2 months since last commit

25 users on Open Hub

Activity Not Available
5.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: GPL-2.0+
Tags c carrier chat cluster communication ims kamailio linux messaging openser presence rcs 12 more...

KDE Telepathy

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Claimed by KDE Analyzed 16 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]

158K lines of code

14 current contributors

26 days 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 1 day 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]

573K lines of code

22 current contributors

2 months 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...

blue.box project

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  Analyzed about 1 hour 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

over 5 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 about 7 hours 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]

59.3K lines of code

10 current contributors

22 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