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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.
No port forwarding, DMZ, and others are required, Trate allows other applications - almost all applications(VNC,WinSCP,BitTorrent,Filezilla,ssh,Web/Ftp etc) - to auto-create dynamic connections between two peers behind two separate NATs and firewalls.
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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)
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.
Utilities enhance AMP, A@H, Asterisk and Cisco XML. Speed Dialing revised using web interface. Paging using custom meetme routines. Intercom by dialing 0 + Ext #. Browsing AMP's extensions on your phones LCD. Backup patch for Asterisk@Home maint screen.
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,
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