Search Results for "voip"

Conceived as an internet radio Shoutcast/Icecast client and DJ console IDJC has two main media players, a background track player, effects buttons, crossfader, webm, aac, ogg, and mp3 streaming, stream automation timers, aux input, voice and VoIP integration. Media file formats include: mp3, ogg, flac, wma, wav, m4a, m3u, xspf, pls, and cue sheet support, IRC track and station announcements, uses jack audio connection kit to provide a flexible audio chain. This list of features is by no means complete. More info at the project homepage.

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. .

Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib

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

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.

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.

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.

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

IHU is a VoIP application for Linux (using Qt and Speex), with low latency, crypted stream, minimal use of bandwith, and without intermediary servers. It is the easiest way to talk real-time with your friends (like phone) on the internet or LAN.

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,

Global mesh development software and hardware platform from developing global wireless mesh network using multiple frequencies.