Showing 11 open source projects for "live streaming php"

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  • 1
    MCU Media Server

    MCU Media Server

    SIP Video Multiconference Media Server with WebRTC support.

    REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .
    Downloads: 0 This Week
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  • 2

    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
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  • 3
    Libre Software to manage audio/video connection from a PC to a libre hardware real time stethoscope hardware, developed to allow doctors make a remote diagnose in real time in isolated rural areas of developing countries or for medicine teaching
    Downloads: 0 This Week
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  • 4

    red

    Red simulates a lossy and limited network over two ethernet interfaces

    Downloads: 0 This Week
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  • 5
    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.
    Downloads: 0 This Week
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  • 6
    Video Applications for Asterisk (libh43m, app_h324m, app_mp4, app_rtsp, app_transcode, app_swf) H324m stack, mp4 record and playblack, rtsp support, video transcoding and swf file playback.
    Downloads: 0 This Week
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  • 7
    Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.
    Downloads: 0 This Week
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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
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  • 9
    JSR180 is the standard API spec by jcp.org which is been implemented over Resiprocate Stack. Our purpose is to enable "Open Source Code with Open APIs". By implementing JSR180 we could easily access the SIP Stack by the API documented in the JSR180 Spe
    Downloads: 0 This Week
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  • 10
    RUMPI: Real Time Multi Peer Interaction. RUMPI want to be as ubiquitous as Flash-browser-plugin, but enabling browsers for real time communication between peers: VoIP (IAX2) and Real-time video Streaming. RUMPI is not a plattform, is a Browser plugin,
    Downloads: 0 This Week
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  • 11
    Global mesh development software and hardware platform from developing global wireless mesh network using multiple frequencies.
    Downloads: 0 This Week
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