Search Results for "open java projects"

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.

jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange. Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone. Please go to...

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/

jlibrtp aims to create a library that makes it easy to support RTP (RFC 3550,3551) in Java applications. SRTP (RFC 3771) has been delayed in favor of RFC 4585.

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

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 projects intend to port to Windows / Interix the famous and performant SIP Express Router from IPTEL (SER)

The goal of SkyypReader is to create printable files from Skype logs. Currently SkyypReader can read callmember256.dbb files.

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.

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

T.140 Presentation Library is an implementation of the presentation routines described in ITU-T T.140.

this is to create an chatting module which is easy to implement

VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z

Java SIP stack, originally based on mjsip.

This application permit user reading RSS (web feed) on his mobile device.

pvoim = Parilo Voim : voice over internet messaging XMPP Client that supports text and sound messages

The Cisco IP Phone Idle Display Logo Engine project or CIP IDLE A framework for displaying custom logos on Cisco IP Phones 79XX series. Also provides a Java alternative to cip2gif.exe and gif2cip.exe

talkLock is voice encryption software for your cell phone. talkLock includes a server component so that you own all points of encryption and decryption of your communications.

Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).

An implementation of the iax protocol using java. Only alaw, ulaw codec Supported.

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.

RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has source code for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive

trakkcor tracks your position via gps and submits the position data to your server. All you need is a j2me-compliant mobilephone which supports JSR-82 and an external gps receiver.

VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).