Open Source 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

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

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

Very fast and very flexible SIP server

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.

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

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. .

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.

A SIP Server for Sanity Testing

Simple test set for SIP server.

ICTCore is an open-source Communications Platform as a Service (CPaaS) designed to empower developers and system integrators to build, deploy, and manage communication-enabled applications with ease. With support for voice, SMS, email, and fax channels, ICTCore provides a programmable communication layer that enables rapid development of ICT-based solutions using standard development skills.Following are few projects developed over ICTCore communications framework ICTFax open source fax server software. ICTDialer open source auto dialer software. Truly unified communications framework ICTCore allow developers to use multiple communication methods / services like Call, FAX, Messaging or Email, plus it also allow them to integrate these services with each other, like SMS notification after each call or email to fax. ICTCore cover all major communication methods Allow integration between different kind of communication methods Uses unified terminology and share

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