Search Results for "server security system"
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Showing 37 open source projects for "server security system"
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AI-generated apps that pass security reviewRetool lets you generate dashboards, admin panels, and workflows directly on your data. Type something like “Build me a revenue dashboard on my Stripe data” and get a working app with security, permissions, and compliance built in from day one. Whether on our cloud or self-hosted, create the internal software your team needs without compromising enterprise standards or control.
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Synapse
Matrix reference homeserver
Matrix is an ambitious new ecosystem for open federated Instant Messaging and VoIP. Everything in Matrix happens in a room. Rooms are distributed and do not exist on any single server. Rooms can be located using convenience aliases like #matrix:matrix.org or #test:localhost:8448. Synapse is currently in rapid development, but as of version 0.5 we believe it is sufficiently stable to be run as an internet-facing service for real usage! Create and manage fully distributed chat rooms with no... -
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or... -
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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... -
Application Monitoring That Won't Slow Your App DownFull APM with errors, performance, logs, and uptime monitoring. 99.999% uptime SLA on the platform itself.
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ViciStack - VICIdial Call Center Tools
VICIdial call center optimization tools, monitoring utilities, and per
VICIdial call center optimization tools, scripts, and monitoring utilities for system administrators and call center managers. Includes performance tuning guides, AMD (Answering Machine Detection) analysis tools, compliance automation scripts, and real-time monitoring dashboards for VICIdial-based contact centers running on Asterisk PBX. Key features: - Server performance monitoring and alerting - AMD accuracy tuning and analysis - Campaign optimization scripts - Compliance and DNC automation - Asterisk dialplan utilities - Real-time agent and queue dashboards Visit https://vicistack.com for documentation, guides, and community resources. -
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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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ictcore
ICTCore: Unified Communications Framework for web developers
ICTCore is an open-source Communications Platform as a Service (CPaaS) designed to empower developers and system integrators to build, deploy, and manage communication-enabled applications with ease. With support for voice, SMS, email, and fax channels, ICTCore provides a programmable communication layer that enables rapid development of ICT-based solutions using standard development skills.Following are few projects developed over ICTCore communications framework ICTFax open source fax server software. ... -
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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... -
Forever Free Full-Stack Observability | Grafana CloudBuilt on open standards like Prometheus and OpenTelemetry, Grafana Cloud includes Kubernetes Monitoring, Application Observability, Incident Response, plus the AI-powered Grafana Assistant. Get started with our generous free tier today.
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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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Kazoo Live Medium
Kazoo Live Medium. From Null to a Full Featured PBX on minutes
From Null to a Full Featured PBX in matter of minutes!! If you want: - To boot or install quickly Kazoo on your server. - To test about how it works, without installing on your hard drive. - To install on cluster environment - To install if your previous platform go downtime. This is a suitable option. What is included? Monster-UI Open Source Apps, like SmartPBX, CallFlows, PBX Connector, Voicemails, Faxes, Accounts and Number Manager Kazoo v4.3.x Kamailio ... -
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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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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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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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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. . -
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VoiceOne gives you the ability to install and compleatly configure a pbx platform based on Asterisk 1.8 with an easy to use web GUI, which would be a framework to build a communication server adding various plugins.
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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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Sipura2NCID
SIPURA brand ATA to ncidd gateway
This is a SIPURA/LinkSys/Cisco brand ATA (analog telephone adapter) to NCIDD (network caller-ID daemon) gateway. sipura2ncid uses the ATA's debug message output to a syslog server. No need for packet snooping or network rearrangement. -
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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
