Search Results for "interactive c"
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57 projects for "interactive c" with 2 filters applied:
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/ -
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GetSmart is a download manager that supports Multi-Connection downloads. With Multi-Mirror search it can download different parts from these mirror servers simultaneously. It can 'act as proxy' to serve multiple clients. It can also works as a Daemon.
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TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.
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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt
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GNU Telephony
GNU Telephony Services and Applications
GNU Telephony intends anyone to use free as in freedom software for telephony, and to do real-time collaboration freely and privately over the Internet, with the freedom to do so on any platform they choose to use. -
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KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
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Stop vibe-debugging.AppSignal's MCP server hands Claude, Cursor, or Zed your real errors, traces, and the deploy that shipped them. AI writes the fix; you review the diff.
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dtmfbox is a small softswitch application (SIP/CAPI), that can be used to control different tasks over telephone keyboard via DTMF. Mostly, it was made to run on the AVM FRITZ!Box 7170 (mipsel) but works under Unix/Linux and Windows, too.
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Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home/corporate/operator level markets.
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GSM Multiplexer Daemon is intended to be used for GSM Modens. It implements (a subset of) the GSM 07.10 standard, it can be used to multiplex one or more logical channels over one physical serial channel to the modem.
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This project is aim to develop module named app_avi.so for asterisk to add the ability of play avi file.
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The goal of the project is to create a high-performance, open-source and standards-compliant implementation of a Home-Subscriber-Server (HSS) for use in a IMS context.
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Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.
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Quintum Tenor VoIP Gateway Call Detail Record (CDR) Advanced Server. This CDR Server can collect Call Detail Records (CDRs) from multiple Tenors. The data is collected simultaneously and continuously. Supported MySQL and file storages.
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AMS is a suite of software intended to make day to day administration and monitoring of an Asterisk PBX server easier. It contains a daemon that acts as a proxy to Asterisk's Manger Interface and a GTK GUI application for monitoring and administration.
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BladeWareVXML is a portable VoiceXML 2.1 interpreter that is an enhanced version (performance, usability and integration) of OpenVXI. A commercial version, with documentation, sample code, and support options, is available from the Commetrex Website.
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Emergency Context Resolution with Internet Technologies The project is based on a draft from files draft-ietf-ecrit-* Developed by IETF ECRIT Working Group
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This is a least cost routing programme (LCR) for the Fritz!Box fon. It uses billiger-telefonieren.de to determine the current cheapest rates (i.e. prefixes like 01013) and the box's web interface to update the rates. Currently useful for Germans, only.
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This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/) The OpenH323 project provides full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
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rtnppd is able to route TNPP 3.8 (Telocator Network Paging Protocol) packets between serial links and other rtnppd programs (over an IP-network). Also includes rtapd daemon for routing packets to rtnppd.
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The next generation of the Openh323 Gatekeeper project that provides high availability and scalibility for carrier grade, available freely under GNU GPL.
