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Description

OpenSIPS is an open-source SIP proxy/server designed for handling voice, video, instant messaging, presence, and various other SIP extensions. This versatile signaling SIP server caters to carriers, telecom operators, and ITSPs, providing solutions for diverse applications such as Class 4 and Class 5 residential platforms, trunking and wholesale services, enterprise and virtual PBX systems, session border controllers, application servers, load balancers, IMS platforms, call centers, and much more. Its high throughput capability supports tens of thousands of calls per second and millions of simultaneous connections, making it a preferred choice for various SIP scenarios and services. OpenSIPS excels due to its flexible routing and integration options, which allow for custom routing logic through routing scripts and numerous interfacing APIs. With over 120 modules available, it facilitates effective application development for SIP handling, backend operations, and routing logic. Additionally, the 3.2 release of OpenSIPS underwent a thorough security audit, enhancing its safety and reliability for users across the board. This robust framework not only meets current demands but also continuously evolves to address future challenges in the telecommunications landscape.

Description

Unified Communications, OTT, and telecom providers can configure, launch, and monetize any type communication service in IP networks. This includes Unified Communications, Hosted IP PBX / IP Centrex and SIP Trunking. DID Management, DID Management, OTT Voice and calling cards. VoLTE/VoWiFI are some of the key functional modules. Key functional modules: Class 4 & 5 SIP softswitch, for residential and enterprise telephony service; convergent online billing, service provisioning and charging (optional).