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KDE Telepathy

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Claimed by KDE Analyzed about 12 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

7 months since last commit

17 users on Open Hub

Very Low Activity
5.0
 
I Use This

Kazoo Platform

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  Analyzed about 18 hours 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

about 1 month since last commit

15 users on Open Hub

Very Low Activity
5.0
 
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OpenSIPS

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  Analyzed about 1 hour 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]

651K lines of code

25 current contributors

about 15 hours since last commit

15 users on Open Hub

High Activity
4.0
   
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Asterisk-Java

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  Analyzed about 24 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]

51.7K lines of code

11 current contributors

5 months since last commit

14 users on Open Hub

Very Low Activity
4.875
   
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blue.box project

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  Analyzed about 9 hours 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

almost 11 years since last commit

14 users on Open Hub

Inactive
5.0
 
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Linphone

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  Analyzed about 2 hours 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.

511K lines of code

25 current contributors

about 17 hours since last commit

11 users on Open Hub

Very High Activity
3.6
   
I Use This

Yate

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  No analysis available

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]

0 lines of code

5 current contributors

0 since last commit

10 users on Open Hub

Activity Not Available
4.25
   
I Use This
Mostly written in language not available
Licenses: gpl

A2Billing

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  No analysis available

A2Billing complements the Asterisk project by enabling the following features on both TDM and VoIP calls: Traditional calling card services Credit limit on both pre-paid and post-paid customers Callback services Residential VoIP services Wholesale minutes termination Monthly/weekly free ... [More] calling packages Invoicing Paypal, Moneybookers and Authorize.net integration. The project is easy to use and is frequently seen on FreePBX installations to bring accountability to small offices' phone usage. For ITSP and traditional telco wholesale usage it has been seen to easily scale to millions of minutes per month, with 100,000s rates across many trunks. Work is in progress to further enhance A2Billing's scalability and availability. [Less]

0 lines of code

0 current contributors

0 since last commit

10 users on Open Hub

Activity Not Available
4.0
   
I Use This
Mostly written in language not available
Licenses: gpl

Restcomm

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Claimed by TeleStax Analyzed 4 months ago

Restcomm is the only full Stack Communications Platform as a Service (cPaaS). It enables you to create, deploy and manage services and applications integrating voice, video and data across a range of IP and legacy communications networks. It drives convergence with the following key enablers ... [More] : Communications API Layer and Visual Designer in Restcomm Connect, WebRTC SDKs for Web and Mobile, SMSC, USSD Gateway, GMLC for GeoLocation. It also offers middleware telecom infrastructure with JAIN-SLEE, SIP Servlets, SS7 Stack, SIP Stack, Diameter Stack and SMPP Stack Restcomm is supported by TeleStax [Less]

5.54M lines of code

20 current contributors

about 2 years since last commit

10 users on Open Hub

Activity Not Available
4.375
   
I Use This

Homer SIP Capture

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  Analyzed about 16 hours 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]

60.8K lines of code

2 current contributors

10 months since last commit

10 users on Open Hub

Very Low Activity
5.0
 
I Use This