Full-stack observability with actually useful AI | Grafana Cloud
Our generous forever free tier includes the full platform, including the AI Assistant, for 3 users with 10k metrics, 50GB logs, and 50GB traces.
Built 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.
Create free account
Add Two Lines of Code. Get Full APM.
AppSignal installs in minutes and auto-configures dashboards, alerts, and error tracking.
Works out of the box for Rails, Django, Express, Phoenix, and more. Monitoring exceptions and performance in no time.
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...
Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.
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/
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.
A JNI wrapper for pjsip. You can use this wrapper to develop Java applications using the pjsip library. At the moment only the pjsua API is implemented. If you would like to obtain a commercial license, or need customisations, please contact us.
Jabbin is an Open Source social application that combines VoIP, Instant Messaging and Social Networking, allowing you to focus on what you really care about: your friends.
Give your IT, operations, and business teams the ability to deliver exceptional services—without the complexity.
Freshservice is an intuitive, AI-powered platform that helps IT, operations, and business teams deliver exceptional service without the usual complexity. Automate repetitive tasks, resolve issues faster, and provide seamless support across the organization. From managing incidents and assets to driving smarter decisions, Freshservice makes it easy to stay efficient and scale with confidence.
Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com
Apreta is about sharing information in real-time meetings. It provides desktop sharing, presentation sharing, a shared whiteboard, and audio conferencing.
A command line SIP/H323 softphone capable of sending and receiving audio files as well as sending out of band DTMF digits. Supports a BNF format configuration language for scripting call scenarios. Useful for example system testing.
Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.
This is an Interactive Voice Response System using TAPI (MS TAPI 2.x).You can use this as an answering machine.But most of all this is meant as an illustration of how you can use TAPI API for developing IVR applications in VC++
A Simple Middlebox Configuration Protocol (RFC 4540) implementation for Linux.
SIMCO is a signaling protocol that can be used by applications (such as SIP B2BUAs) to dynamically control firewalls and Network Address Translators (NATs).
This C++ library has been designed as a Chrome SIP stack.
Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services.
The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to...