Search Results for "two"

Legacy SVN repositories for projects from David Rowe http://rowetel.com Mailing lists for Codec 2 and SM1000.

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...

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.

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.

...Última versión disponible 2.11.19 en la web de VozBox: http://www.vozbox.es/descarga/ Internet connection required during installation. Update version: 2.11.19 1 - Customizing bash_login and bashrc 2 - Upgrade to version libpri-1.4.15 3 - Included new module-manager FOP2-1.0.3 4 - Inclusion of new packages and dependencies. 5 - Correction of errors in the installation. 6 - Upgrade to version webmin-1.710 7 - Correction of errors in the pbx-vpn script. 8- Inclusion of new script pbx-status.

This is a web based instant 1 on 1 private online video conferencing solution. It's a solution for conducting easy to setup face to face meetings without leaving your office or home. It's the easiest and most cost-effective way to meet somebody and discuss one on one, to make a video call just by providing a private room access link.

O OpenVOIP é uma solução Open Source para que empresas possam ter um padrão de comunicação VOIP.

A simple interface to work with SIP numbers of Asterisk. Add, delete, and edit SIP numbers in the MySQL database As well as viewing CDR records Простой интерфейс для работы с SIP номерами сервера Asterisk. Добавление, удаление и редактирование SIP номеров в базе MySQL А так же просмотр CDR записей

The Agispeedy is an AGI Application Server implemention in asterisk and written by Pure PHP. it inconceivable faster than normal agi about 1000%, and improves performance.

SIP over IPv4 and SIP over IPv6 are different worlds, making it impossible to register IPv4-only phones on IPv6-only services, and so on. SIPproxy64 is a proxy that translates between the two worlds. It is small enough to run inside a router.

This is a open source media gateway controller. The protocol used is MEGACO/H.248. In a later version also SIP will be supported.

A $10 Analog Telephone Adapter (ATA) that allows low cost routers such as the WRT54G to connect to an analog telephone. The application is low cost VOIP, in particular for the developing world.

The goal of the project is to create a high-performance, open-source and standards-compliant implementation of a Home-Subscriber-Server (HSS) for use in a IMS context.

Castadiva is a test bed based on low-cost off-the-shelf devices, which is used to test protocols developed for MANETs. Castadiva is completely compatible with the file format used by the ns-2 simulator.

The project "Sippie" is an additional tool for automatically converting Wireshark-Traces to XML - scenarios, which can be easily used as infile for the OpenSource SIPp test tool and traffic generator.

...After three years of development, the project no longer makes sense. The main reason to create the serMyAdmin was the lack of a GUI for OpenSER/OpenSIPS. The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I would like to say thanks to all the people that in some way contributed with the project. The svn code is available to anyone who wants to maintain the project. Most of our development efforts are now focused in a new commercial product based on OpenSIPS called SIP Pulse, completely rewritten in Java and Glassfish www.sippulse.com supporting pre and post paid users with billing and reseller portal.

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.

The system is used to control elements of a smart home remotely, with a web or a telephony interface. Automation can also be created to automatically execute actions based on events.

LibDreamSpeak is an example implementation of the TeamSpeak2 protocol (client-side) in Java. Some day it may be used as base for alternative TeamSpeak2 (mobile?) clients.

PHP / ZeroC ICE based web administration system for Murmur Servers.

OSSEG is an IMS compatible security gateway that adds firewall, VPN, IDS, IPS, and monitoring functionalities to VoIP networks.

An tray icon tool to allow automatic Dial using an XMLRPC framework. The application is based on two separate tools: - a light desktop application - a XMLRPC server witch make an PBX abstraction. (first driver is for Asterisk)

A C++ lib for the Asterisk PBX AGI interface.

Cross-platform Java Teamspeak client