Search Results for "java telephony"

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 Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone.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/

The Bria Desktop API has moved to github at https://github.com/CounterPathAPI. Additional samples for Java and Javascript can be found there. The Bria Desktop allows third-party applications to control Bria for Windows softphone clients. By leveraging the Application API, third-party applications can perform commands such as starting an audio or video call, answering a call or placing a call on hold. CounterPath is now encouraging third-party-application developers to integrate...

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 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.

TIPcon1 is a standards-compliant soft phone using text, voice and video with accessibility features.

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

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.

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).

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

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).

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).

With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.

HATEFNA هاتفنا .....http://www.hatefna.org ..... (PC&Mobile)IP SoftPhone over SIP and ASTERISK using JAVA for PC and Symbian C++ for Symbian Mobiles Phones ..... برمجيات للمهاتفة ببروتوكول الإنترنت

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.

VoiceXML Voice Mail Appliation written in Java. Inlcudes a distribution for the Voxeo Prophecy platform.

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.

Jabast connects XMPP Clients to a asterisk PBX to allow extension listening, calls and presence controle for Agents.

Wavelet EMP is a web-based J2EE ERP system running on PostgreSQL DB. It is in production status and used by more than 100 shops in South Eash Asia. The software team is in Malaysia. Powered by WAVELET.BIZ. For demo/help, email vincent@wavelet.biz

TiVonage allows Vonage customers to manage their voice mail, including: * Check for new messages * Listen to messages * View caller ID and address info * Return calls via Click2Call ...all from the comfort of their TiVo remotes.