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Showing 8 open source projects for "java video codecs"

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  • 1
    jphonelite

    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.
    4 Reviews
    Downloads: 1 This Week
    Last Update:
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  • 2
    GNU Gatekeeper (GnuGk)

    GNU Gatekeeper (GnuGk)

    H.323 Gatekeeper for VoIP and videconferencing

    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/
    2 Reviews
    Downloads: 2 This Week
    Last Update:
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  • 3
    jpbxlite

    jpbxlite

    Java VoIP/SIP PBX system (replaced by jfPBX)

    jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net
    Downloads: 0 This Week
    Last Update:
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  • 4
    Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
    10 Reviews
    Downloads: 0 This Week
    Last Update:
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  • 5

    SMPTE 2022-1 library for VLC & co

    Optimized and cross platform SMPTE 2022 FEC library in C, Python, Java

    Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib
    Downloads: 0 This Week
    Last Update:
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  • 6
    Zanzibar is a complete, standards based IVR. It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.
    1 Review
    Downloads: 0 This Week
    Last Update:
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  • 7
    Google Contacts to Grandstream xml Phonebook Format Converter
    2 Reviews
    Downloads: 0 This Week
    Last Update:
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  • 8
    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.
    Downloads: 0 This Week
    Last Update:
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