Tags : Browse Projects

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Signal

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  Analyzed 1 day ago

Free, world-wide, private phone calls for iPhone, Android and Desktop. Private messaging coming soon! Like privacy? Secure your calls for free with Signal. It's that simple. Signal provides end-to-end encryption for your calls, securing your conversations so that nobody can listen in. - ... [More] Signal uses your normal phone number to make and receive calls, so you don't need yet another identifier. - Signal calls are encrypted end-to-end, but function just like you're used to. - Free and Open Source, enabling anyone to verify its security by auditing the code. - Uses wifi or data, not your plan's voice minutes. [Less]

3.06M lines of code

42 current contributors

3 days since last commit

4 users on Open Hub

Very High Activity
0.0
 
I Use This

FreedomBox

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  No analysis available

FreedomBox, a Debian pure blend, is a 100% free software self-hosting web server to deploy social applications on small machines. It provides online communication tools respecting your privacy and data ownership.

0 lines of code

41 current contributors

0 since last commit

3 users on Open Hub

Activity Not Available
5.0
 
I Use This
Mostly written in language not available
Licenses: AGPL3_or_..., gpl3

Sippy SIP B2BUA

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  Analyzed about 18 hours ago

Sippy B2BUA is a RFC3261-compliant SIP Back-to-back user agent (B2BUA). Unlike a SIP proxy server, which only maintains transaction state, the B2BUA maintains complete call state and participates in all call requests. For this reason it can perform number of functions that are not possible to ... [More] implement using SIP proxy, such as for example accurate call accounting, pre-paid rating and billing, fail over call routing etc. Unlike PBX-type solutions such as Asterisk for example, the B2BUA doesn't perform any media relaying or processing, therefore it doesn't introduce any additional packet loss, delay or jitter into the media path. Sippy B2BUA supports RADIUS AAA protocol, which is compatible with Cisco VoIP gateways allowing it to be seamlessly integrated into existing networks. [Less]

10.3K lines of code

1 current contributors

19 days since last commit

3 users on Open Hub

Very Low Activity
0.0
 
I Use This

cgrates

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  Analyzed about 23 hours ago

Rating system for telecom providers

426K lines of code

11 current contributors

4 days since last commit

3 users on Open Hub

High Activity
0.0
 
I Use This

luasofia

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  Analyzed about 17 hours ago

LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.

4.39K lines of code

0 current contributors

about 9 years since last commit

3 users on Open Hub

Inactive
5.0
 
I Use This

ice4j

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  Analyzed about 15 hours ago

A Java implementation of the STUN and TURN and ICE protocols. This project is a continuation of the Stun4J initiative. The goal of the project is to provide VoIP Java developers with a means of handling NATs and firewalls. The project is a joint effort of the SIP Communicator and TurnServer open ... [More] source projects but is entirely based on standards and can be used with any other VoIP client or STUN/TURN server. [Less]

34.1K lines of code

9 current contributors

4 months since last commit

3 users on Open Hub

Very Low Activity
5.0
 
I Use This

Newfies-Dialer

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  No analysis available

Intelligent dialer for Freeswitch This project aims to create a dialer application to deliver message to a high amount of users. This is not a telemarketing application, we aim to provide an easy simple dialer system to reach community of users. You can easily manage your campaign, phonebooks ... [More] , subscriber, gateway and start delivering your messages [Less]

0 lines of code

0 current contributors

0 since last commit

3 users on Open Hub

Activity Not Available
0.0
 
I Use This
Mostly written in language not available
Licenses: mozilla_p...

WebRTC

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  Analyzed about 5 hours ago

WebRTC is an open source project that enables web browsers with Real-Time Communications (RTC) capabilities via simple Javascript APIs. The WebRTC components have been optimized to best serve this purpose. WebRTC implements WHATWG's proposal for video conferencing on the web.

1.02M lines of code

150 current contributors

about 12 hours since last commit

3 users on Open Hub

Very High Activity
0.0
 
I Use This
Licenses: No declared licenses

GNU Gatekeeper (GnuGk)

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  Analyzed 1 day ago

The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions.

95.7K lines of code

2 current contributors

20 days since last commit

3 users on Open Hub

Low Activity
4.5
   
I Use This

erlrtpproxy

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  Analyzed 1 day ago

RTP/RTCP proxy for VoIP and IM applications.

8.26K lines of code

0 current contributors

almost 10 years since last commit

2 users on Open Hub

Inactive
4.0
   
I Use This