TransNexus supports the following open source projects:
Asterisk is software that turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and more. It is used by small businesses, large businesses, call centers, carriers and governments worldwide. Asterisk is free and open source. Asterisk is sponsored by Digium. TransNexus has been an active contributor to Asterisk since 2004. Asterisk V1.2.0 and later versions supports OSP peering for SIP communications. Asterisk V1.6 supports OSP peering for SIP, H323 and IAX communications.
Debian began in 1993 by Ian Murdock as a new distribution which would be made openly, in the spirit of Linux and GNU. Debian is meant to be carefully and conscientiously put together, and to be maintained and supported with similar care. It started as a small, tightly-knit group of free software hackers, and gradually grew to become a large, well-organized community of developers and users.
FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.
Kamailio, formerly OpenSER, is an open source SIP server released under GPL, able to handle thousands of call setups per second. Among features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video), SIMPLE instant messaging and presence, ENUM, least cost routing, load balancing, routing fail-over, accounting, authentication and authorization against MySQL, Postgres, Oracle, Radius, LDAP, XMLRPC control interface, SNMP monitoring. It can be used to build large VoIP servicing platforms or to scale up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.
OpenSIPS is a mature open source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server, is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable design. What OpenSIPS has to offer comes in a reliable and high-performance flavor. OpenSIPS is one of the fastest SIP servers, with a throughput that confirms it as a solution up to enterprise or carrier-grade class.