Search Results for "sofia-sip"

SipLine is a modern, lightweight SIP softphone for Windows 10/11 built with .NET 9 WPF. It provides enterprise-grade VoIP communication with a focus on performance, security, and extensibility. Features Ultra-Fast: Startup in <0.5s, minimal CPU usage HD Audio: Opus and G.711 codecs with adaptive jitter buffer Security: TLS transport, SDES-SRTP encryption, Windows DPAPI credential protection Multi-Account: Up to 5 simultaneous SIP accounts Plugin SDK: C# SDK for custom integrations (CRM, call recording, automation) Plugin Marketplace: Community and premium extensions Headset Support: Jabra, Poly, Sennheiser, Logitech with HID controls Call Quality: Real-time MOS Score, Jitter, Packet Loss and RTT monitoring Enterprise: Silent MSI installer, GPO support, JSON centralized config Compatibility: 3CX, Asterisk, FreePBX, OVH, Twilio, RingCentral, and more Language packages : https://sipline.feelautom.fr/languages

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...

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

...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. .

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

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). ...

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

bv-tapi is a SIP TAPI bridge. It supports click to dial and screen popping. It is compatible with Freeswitch and babblevoice.com, but potentially should work with any SIP server which supports RFC 4235 (not Asterisk!). Source code now hosted at github. https://github.com/tinpotnick/bv-tapi.git

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.

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

Very fast and very flexible SIP server

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

Simple test set for SIP server.

Web interface for administrating OpenSER SIP Server.

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.

A SIP Server for Sanity Testing

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