Search Results for "audio video streaming server source code"

JavaComm is a client–server communication platform offering chat, telephony, screen sharing, conferencing, a media player, and configurable media streams. Users can define and manage their own streams for flexible playback. All server-based communication runs over HTTPS (SSL/TLS), while file transfers are handled directly between users without passing through the server. The JavaComm Server runs on any Jakarta EE Web Profile 10 platform with the required extensions, using Tomcat 10.1.52 as...

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. .

JAVIER is a JAvascript Voicexml InterpretER, designed (but not restricted) to run inside a web browser, its main engine has less than 1000 lines of code. It's maybe, the tiniest but (almost) FULL VoiceXML implementation.

Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip

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

client-server full-java chat, provided with audio&video (JMF), custom rooms, custom forums, instant- messaging and file sharing

OfficeSIP Softphone and Messenger are two enterprise VoIP SIP clients written in C# in .NET Framework. The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server. See also open source, cross-platform: 1) simple messenger Brief Msg at http://briefmsg.org 2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list

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.

Cairo sets out to provide an enterprise grade, MRCPv2 compliant speech solution utilizing existing open source speech resources such as FreeTTS and Sphinx-4.

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.

This project aims to build a software reference implementation of the EBU standard for the transmission of high quality, low latency, audio streams over IP networks. (EBU-tech 3326)

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

The MRCPv2 protocol is designed to allow client devices to control media processing resources, such as speech recognition engines. MRCP4J provides a Java API that encapsulates the MRCPv2 protocol and can be used to implement MRCP clients and/or servers.

IM,WebIM and Sip voip tool. Client is Win32, Server is CentOS 5. support video,audio, file trans, sip call.