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KDE Telepathy

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Claimed by KDE Analyzed 12 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]

157K lines of code

13 current contributors

24 days since last commit

18 users on Open Hub

Moderate Activity
5.0
 
I Use This
Licenses: GPL-2.0+, LGPL-2.1

OpenSIPS

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  Analyzed 12 days 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]

546K lines of code

24 current contributors

22 days since last commit

15 users on Open Hub

High Activity
4.0
   
I Use This

Kazoo Platform

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  Analyzed 16 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]

274K lines of code

39 current contributors

about 1 month since last commit

14 users on Open Hub

High Activity
5.0
 
I Use This

blue.box project

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  Analyzed 10 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

over 4 years since last commit

14 users on Open Hub

Inactive
5.0
 
I Use This

Asterisk-Java

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  Analyzed 16 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]

55.6K lines of code

15 current contributors

about 1 month since last commit

14 users on Open Hub

Moderate Activity
4.875
   
I Use This

sipXecs IP PBX

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  Analyzed almost 2 years ago

sipXecs IP PBX (Enterprise Communications Server) is an open source software implementation of a Session Initiation Protocol (SIP) based unified communications system (IP PBX). Similar to a traditional private branch exchange (PBX), it allows a number of attached telephones to make calls to one ... [More] another, and to connect to other telephone services including the PSTN and SIP Trunking services. sipXecs IP PBX is an open source alternative to commercial PBX offerings. It is easy to use and scales to several thousand users. [Less]

945K lines of code

7 current contributors

almost 3 years since last commit

13 users on Open Hub

Activity Not Available
5.0
 
I Use This

Kamailio (SIP Express Router)

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  Analyzed 11 days 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]

1.81M lines of code

75 current contributors

12 days since last commit

12 users on Open Hub

Very High Activity
5.0
 
I Use This

Linphone

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  Analyzed 18 days ago

Linphone is an open source Voice Over IP phone (or SIP phone) that makes possible to communicate freely with people over the internet, with voice, video, and text instant messaging.

526K lines of code

18 current contributors

23 days since last commit

11 users on Open Hub

Very High Activity
3.5
   
I Use This

Homer SIP Capture

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  Analyzed 14 days ago

HOMER is a robust, carrier-grade, scalable SIP Capture system and Monitoring Application with HEP/HEP2, IP Proto4 (IPIP) encapsulation & port mirroring/monitoring support right out of the box, ready to process & store insane amounts of signaling with instant search, end-to-end analysis and ... [More] drill-down capabilities for ITSPs, VoIP Providers and Trunk Suppliers using SIP signaling [Less]

44.1K lines of code

16 current contributors

17 days since last commit

10 users on Open Hub

Moderate Activity
5.0
 
I Use This

Yate

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  Analyzed 15 days ago

Yet Another Telephony Engine is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine ... [More] , maximizing communications efficiency and minimizing infrastructure costs for businesses. [Less]

376K lines of code

4 current contributors

23 days since last commit

10 users on Open Hub

Moderate Activity
4.25
   
I Use This