Search Results for "server security system"
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Showing 21 open source projects for "server security system"
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MongoDB Atlas runs apps anywhereMongoDB Atlas gives you the freedom to build and run modern applications anywhere—across AWS, Azure, and Google Cloud. With global availability in over 115 regions, Atlas lets you deploy close to your users, meet compliance needs, and scale with confidence across any geography.
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BlackBelt Privacy Tor/i2p+WASTE+VidVoIP
Browse on Tor/i2p, Anon p2p Chat / FileTx, Conf / Video VoIP
Open Source - GPLv3 inc images. *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer... -
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SipLine
Free, native Windows SIP softphone with HD audio, SRTP encryption
SipLine is a modern, lightweight SIP softphone for Windows 10/11 built with .NET 9 WPF. It provides enterprise-grade VoIP communication with a focus on performance, security, and extensibility. Features Ultra-Fast: Startup in <0.5s, minimal CPU usage HD Audio: Opus and G.711 codecs with adaptive jitter buffer Security: TLS transport, SDES-SRTP encryption, Windows DPAPI credential protection Multi-Account: Up to 5 simultaneous SIP accounts Plugin SDK: C# SDK for custom integrations... -
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ictagent
A WebRTC-Enabled Chrome Extension for Seamless Call Center
...An Embedded Web Phone. It is Operating System Independent, we just need a browser. Click 2 Call. Agent can access contents, contacts and communications from a single application(browser). It will make agent performance much better. It can harvest contacts from random pages / websites/ web applications. CRM Integration. Inbound and Outbound Calls. -
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Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
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VitalPBX
Unified Communications System - PBX System
VitalPBX is a free telephone and communications PBX system for companies. It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. VitalPBX will help you implement a secure... -
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on... -
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CISCO IP Phone Scanner
Scan all CISCO phones in your LAN to get their information
*** Project moved to https://majed.xyz/coding/desktop-application/cisco-ip-phone-scanner-2/ ****** Scan all CISCO phones in your LAN to get their information(Extension,SN,Model,IP,MAC,etc..) The tool is multi-threaded with the capability of Exporting the results to Excel, the down hosts will also be retrieved under the Error List. Search for an entire Class A network The ability to include Switch info (IP/Port-Name) Include name of Phone user (works with CUCM) Export results to... -
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Accent Tool Suite
Accent interfaces and controls systems through goals and policies
...As most of Accent in in Java, it needs at least a Java Runtime Environment but more likely a Java Software Development Kit. To interact with the real world, Accent needs relevant hardware such as a communications server (softswitch) or home automation equipment. -
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
The project has moved! Please find current versions at https://www.gnugk.org/ 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. The project has moved! Please find current versions at https://www.gnugk.org/ -
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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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AVAYA CMS API Server
CMS API Server with web site
.... - Basic security. -
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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mumb1e
A Murmur-Server Webinterface written in PHP for Mumble
Administration Tool for Mumble -/- Murmur Servers. Mumb1e Admin Plugin, also named "MAP" is a powerful Murmur web interface application written with PHP that can create useres, edit themself and a lot more for Mumble server. MAP is developed over m -
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Eolos VoipAudit
Universal Framework to audit Voip protocols's security
Visual Framework Tool to scan/sniff address space, enumerate users, crack credentials, pattern based dial spoofing and security reporting for Voip protocols. This software is intended to give a general framework to build and plug Voip protocol analizers in order to fix security issues and enhance VoiP platforms confidence. It IS NOT intended to be a cracking tool for malicious system breakers, but a really software MUST for security people to assure Voip platform security. -
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Turn.Server is .NET TURN (Traversal Using Relay NAT) server written in C#. Compatible with [MS-TURN] extension and clients based on Microsoft UCCAPI SDK.
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OfficeSIP Softphone and Messenger are two enterprise VoIP SIP clients written in C# in .NET Framework. The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server. See also open source, cross-platform: 1) simple messenger Brief Msg at http://briefmsg.org 2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list
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Plug and play provisioning server for snom VoIP phones written in Python. The server supports the plug and play provisioning feature embedded in snom phones of the snom 3xx, snom8xx and snom MeetingPoint series.
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The project is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.
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The Java Teamspeak 3 Server Query Software Development Kit is a Java Library that provides Classes to access the Teamspeak Server Query for easy integreating TSSQ into Java Desktop- or Web-Applications.
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MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.
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Acv is a PHP program which allows you to monitor Asterisk servers by viewing active calls, channels, and overall server performance numbers.
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