Search Results for "multi-system"
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Showing 10 open source projects for "multi-system"
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SipLine
Free, native Windows SIP softphone with HD audio, SRTP encryption
...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 -
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ictcore
ICTCore: Unified Communications Framework for web developers
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. ... -
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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
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
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. . -
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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
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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
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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.
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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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