Search Results for "video audio frontend"

VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP...

jfPhone is a VoIP/SIP SoftPhone for Desktops.

Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.

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

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.

A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.

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

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.

With MultiMediaBox (MMBox) you can play video and sound files, send SMS, enter in P2P networks like Gnutella, Kazaa and OpenFT and talk with friends in MSN. All this features in a single application. You can get more functions downloading new MMBox Panel

This is an extension to the project OpenGatekeeper. It enable the routing through the gatekeeper of the RTP (audio and video) and T.120 (data) traffic so no traffic is directly exchanged between communication endpoints.