Search Results for "c source code" - Page 4
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Showing 156 open source projects for "c source code"
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Tecnotel is a project that seeks security and comodity to the end-user, controlling your house through the telephone line, internet or remote-controller. The software available controls the electronic-equipment and others.
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H323 voice & video proxy appropriated for border and zone controlling (without RAS). Designed for heavy load.
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A 64-bit port of OpenSS7 to SuSE Linux 9.3 2.6.5-7.244-smp stock kernel, and the SuSE Linux Enterprise Server (SLES) 9 2.6.5-7.244 stock kernel.
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KObelisk is a tool that connects to the Asterisk PBX and gives information to the user about incoming calls. The program also provides some basic functions like calllists, ringtones or a callback function.
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Priax is a small graphical multiplatform (Linux and win32) IAX2 client written in C, using libiaxclient and SDL.
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nSIP is an RFC 3261 compliant implementation of SIP (Session Initiation Protocol) for .NET, written in C#. nSDP is an RFC 4566 compliant SDP parser also in C#, and can be used to make payloads to nSIP messages or to parse any SDP message.
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Skype-based wardialer written in C#, can record call results, play a wav file upon the phone being answered. Does full seqential dialing. Saves results to log file.
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GibPhone is a highly extensible VoIP/IM client for the .NET framework that uses a powerful plugin engine to allow for UI extensions and any call control stack / media payload / transport protocol combination.
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Send and receive faxes using spandsp library, record, playback and connect calls (conference) with this commandline linux application for capi the CAPI 2.0 interface. Also runs on Embedded Systems (e.g. AR7 or German FritzBox).
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H.323 Beacon is a tool that can be used to measure/monitor/qualify/troubleshoot performance of H.323 Videoconference sessions both at the network and at the host by using a end-node(client) and core-node(server) architecture.
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Audio/Video Conference software for Instant Messengers. It aims to provide Audio/Video conferencing for as many Instant Messengers as possible through a modular design.
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ISDN CAPI Channel driver (chan_capi) for the Asterisk Open Source VoIP Platform. THIS PROJECT HAS BEEN MOVED TO chan-capi.org
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Asterisk Manage API and FAGI C++ Framework. Implemented with high efficient OS independent Sockets wrapper (ACE). Provide C++ API wrappers for Asterisk Manager API and FAGI. A Manage API relay(proxy) server and a FAGI socket server will be implemented.
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This is a Java wrapper for Ffmpeg. The JMF (Java Media Framework) plugins system lets one use JMStudio or other application to play H.263/rfc2190 streams. Support for H.263+ and MPEG-4 is currently under test, please contact us for further information.
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Tipic Inc. is proud to announce the born of the first Open Source, P2P, VoIP and Video conferencing solution for XMPP/Jabber. The target of this project is to develop a new portable set of codes using high quality libraries to add VoIP and Video to XMPP
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MODU is a linux oriented textual dialup toolkit with support for MS callback and iptables. It is intended to be flexible enough to dialup to many ISP without much configuration.
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Cameloid is a new TCP/UDP based P2P voice and video encryption software, that will provide high level of privacy for all levels of communication using only quality secret-key ciphers to ensure long-term security.
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Phone is a simple IP telephony program allowing voice communication over IP network. Phone is written in C and can be built on UNIX/Linux and MS Windows platforms.
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kallcenter is a Voice over IP Callcenter software for outbound Callcenters. It is based on kphone and uses a mySQL database backend. It was developed and tested on SuSE Linux 9.2. Documentation is currently available in german only.
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Universal Transport For Sip, Including UDP,TCP,HTTP Proxy Tunnelling,Sock4/5,STUN,UPNP,TURN, also a extension in sip for net tunnelling establish.
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SIPatHome was based on the SIP Express Router (SER) which is a well-tested, configurable, free and RFC3261-compliant SIP proxy. The project was providing VoIP server software for the embedded Linux distribution OpenWRT on affordable Linksys WRT hardware.
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Speech profile of Person is created that contains elementary sounds uttered. Profile is 1 time download for listeners. The actual audio sample is encoded based on profile. Decode using the profile stored earlier by User, and the audio can be regenerated.
