Search Results for "linux-debian"
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Showing 9 open source projects for "linux-debian"
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jphonelite
Java VoIP Softphone (SIP) (replaced by jfPhone)
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 Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone. -
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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).
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IMS-Communicator is an IMS softphone based on the old version of the sip-communicator java project, implemented on top of the JAIN-SIP stack and the Java Media Framework API. It can be used with Open IMS Core (http://www.openimscore.org).
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Asterisk based callcenter system. Administration interface to manage queues, agents, accounts, skills - track calls, create statistics. Realtime queue monitor. Agent web interface with build in IAX softphone.
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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).
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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).
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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
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The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit
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