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Asterisk

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  Analyzed about 18 hours 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]

2.53M lines of code

59 current contributors

about 2 months since last commit

171 users on Open Hub

High Activity
4.38775
   
0 Reviews
I Use This
Mostly written in C
Licenses: gpl
Tags asterisk c conferencing googletalk h323 iax iax2 isdn ivr jabber jingle mgcp 12 more...

Empathy

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  Analyzed over 1 year 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]

181K lines of code

12 current contributors

almost 2 years since last commit

100 users on Open Hub

Activity Not Available
3.75758
   
0 Reviews
I Use This
Mostly written in C
Licenses: gpl
Tags aim chat client gnome google_talk icq im instant_messaging linux messaging messenger msn 6 more...

Ekiga

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  Analyzed 1 day 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

1 current contributors

about 1 year since last commit

61 users on Open Hub

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

FreeSWITCH

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  Analyzed about 7 hours 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]

1.75M lines of code

44 current contributors

3 days since last commit

58 users on Open Hub

Moderate Activity
4.875
   
0 Reviews
I Use This
Mostly written in C
Licenses: AFL1-1, apache_ish, Apache_Li...
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 1 day 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]

343K lines of code

9 current contributors

2 months since last commit

28 users on Open Hub

Very Low Activity
4.46154
   
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 1 day 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]

1.03M lines of code

73 current contributors

2 days since last commit

26 users on Open Hub

Very High Activity
5.0
 
0 Reviews
I Use This
Mostly written in C
Licenses: gpl
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 about 17 hours 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]

95.9K lines of code

6 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, lgpl21
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]

610K lines of code

25 current contributors

3 days since last commit

15 users on Open Hub

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

Kazoo Platform

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  Analyzed 1 day 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]

327K lines of code

43 current contributors

2 months since last commit

15 users on Open Hub

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

Asterisk-Java

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  Analyzed 1 day 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]

63K lines of code

11 current contributors

2 months since last commit

14 users on Open Hub

Low Activity
4.875
   
0 Reviews
I Use This
Mostly written in Java
Licenses: apache_2
Tags asterisk iax java library sip voip