Open Source Linux SIP Servers

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

Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.

REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .

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

This is a simple Sip server with UI. GPL Licensed. Available for Windows OS for the time being. Tested with Windows XP,2003,2008,Windows 7,Vista. Support presence,registar,proxy (stateful/stateless),border controller,lawful interception,early media

The FCP deamon controls the netfilter (linux 2.4) firewall of the localhost by parsing its own protocol. The main purpose of this deamon is to support application level gateways (especially SIP server) to bypass firewall and NAT.

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

Web interface for administrating OpenSER SIP Server.

The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for example, HTTP servers). Our aim is to promote analysis of real SIP-server behavior. Therefore, we present SiAnTo, an extended anonymization technique that substitutes session-participant information with matching, but nondescript, labels. This allows for SIP traces to be publicly shared, while keeping interesting traffic-session properties intact.

Very fast and very flexible SIP server

sipsak is a command line tool which can send simple requests to a SIP server. It can run additional tests on a SIP server which are usefull for admins and developers of SIP enviroments. Note: the project is maintained on GitHub https://github.com/nils-ohlmeier/sipsak

A SIP Server for Sanity Testing

Command-line perl script to view PortaSIP SIP server log files in convenient form. It also may be used for other SIP log formats after small changes.

Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server