Search Results for "c# source code"
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20 projects for "c# source code" 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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Gammu
Cellular manager for mobile phones/modems
Gammu is a cellular manager for mobile phones/modems. It contains libraries and functions for ringtones,logos,phonebook,SMS,etc. (used by external software), a command line version (with backup/restore) and SMS gateway (with MySQL and PostgreSQL supp -
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. -
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pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/
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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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The OSP Toolkit is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321). The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.
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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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Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.
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A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.
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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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H323 voice & video proxy appropriated for border and zone controlling (without RAS). Designed for heavy load.
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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.
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SCMxx
Management tool for Siemens Mobile phones
SCMxx allows you to exchange certain types of data with mobile phones made by Siemens Mobile. It works with the S25, S35i, M35i, C35i, SL45, S45 and ME45 and probably others. -
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Bayonne is supported from Savanna.gnu.org. Please visit there. Sourceforge is generally depreciated except for mailing lists, but updated releases will be posted here from time to time.
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meicd - UNIX command line run daemon that connects to Nortel Meridian MAX MEI TCP/IP ports. Records PBX event information for use in generating reports and and tracking call flow. Simply decodes the protocol and dumps the data into flat files.
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geskoctrl is a application for download/upload firmware for PBXes and to control these PBXes from the german company GESKO.
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US mirror of answerd, a software answering machine. (Records a voice telephone message and emails it to you as a sound file.) The author of answerd is Vyacheslav Zamyatin (slawa@medinf.chuvashia.su)
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Open Source SIP Back to Back User Agent) and Billing system. Has an Embeded TCL interpreter making session control and session mediation limitless in flexibility. Useful for: Routing Engine, Calling Cards, SBC, Peering, Wholesale and more!
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The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit
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