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Gemini 3 and 200+ AI Models on One Platform
Access Google's best plus Claude, Llama, and Gemma. Fine-tune and deploy from one console.
Build generative AI apps with Vertex AI. Switch between models without switching platforms.
Kablink open team collaboration software uses social networking to unify team workspaces w/ real-time web conferencing. Collaboration for knowledge networking, program management, communities-of-practice, telework, other business process/functional areas
A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.
An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.org.
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.
This project aim to provide a web based application to plan Conference Call using Asterisk application MeetMe(). The engine is based on Vaadin language to improve the portability of the application, Hibernate and some AGI script based on PHP.
XSP is an eXtensible socket-based protocol. XSP 1.0 is an XSP client. The main idea of this project is to connect two devices directly without any servers between them. Chat, file sharing and voice chat are supported now. Authors: m1kc and Solkin.
WebPhone Mural Chat é um projeto que realiza comunicação (chat) entre uma página Web e um Celular. Ambos conectam à um banco de dados MySQL, que provê e armazena os dados do chat, utilizando-se apenas de uma Rede Wireless. O Workflow é Twitter
A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
Sipana is a distributed SIP monitoring tool to monitor the SIP signaling behavior using sequence diagrams and providing SIP end-to-end performance metrics through a centralized Web interface. For more information please visit http://sipana.org/
Chabber is an extensible XMPP-based Instant Messaging application. It is a generic, portable application that follows the standard protocols for Instant Messaging and gives users a lot of advanced functionalities. It provides a platform for plugins.
Tairona is a VOIP phone based on Java with a unique feauture to be launched directly from the website. It has chat and whiteboard among other things and is totally free.
Among the functionality: Include a large variety of codecs (G711, GSM, and SPEEX) - Protocol SIP - Other technical functionalities the support of DTMF (tonalities) although support ENUM (to employ numbers of SIP instead of the addresses of SIP).
Tata messenger is a Java-based instant message sending and receiving tool,most like the microsoft messenger and the Yahoo messenger ,and it will also provides the file transfer ,net meeting(video confrence,video transfer),and file sharing mechanisms
SameDesk is a common workspace for web applications. Supporting applications written in AJAX, Java, Flash, & Server-generated HTML, it lets applications across domains work together for the users via a familiar window GUI delivered via web browsers.
vox-attendant is a telephony voice enabled virtual receptionist designed to work with VoiceXML and includes a distribution for the Voxeo Prophecy platform. It provides a web based interface for managing the directory.
Softphone SIP com voz, vídeo e Mensagem Instantânea.
DTMF, Nat-Traversal, Suporte a Skin(Macromedia Flash), SMS, Historico de Chamadas, Presença, Hold, Auto-Answer e etc.
Sip Softphone with Voice, Video and IM.
The RTP Monitor discovers RTP flows, used by Internet phone applications, and monitors their QoS metrics, observing the interference caused by traffic congestion. It uses distributed collecting agents and provides feedback about the quality in the calls.