Fonimo Integrations

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#1 Open Source VoIP Billing Solution ASTPP has been empowering various VoIP businesses and service providers with its amazing Open Source VoIP solutions. It is regarded as one of the top VoIP billing solutions and the #1 open source FreeSWITCH-driven Class 4 and Class 5 Softswitch Solutions, which automates invoicing and billing processes completely. ASTPP is one of few solutions that support the N-level reseller structure. It uses cutting edge technologies like FreeSWITCH and OpenSIPs as core technologies to provide the most robust, reliable, and scalable solutions. ASTPP is a smart telephony platform, which has a suite of smart VoIP telephony solutions to offer to service providers and businesses, including, but not limited to: Class 4 Softswitch Class 5 Softswitch VoIP Billing Calling Cards Multi Tenant IP PBX DID Management FreeSWITCH Billing FoIP (Fax Over IP) STIR/SHAKEN Call Forwarding SIP Trunking MVNO Billing Mobile Dialer

OpenSIPS is an open-source SIP proxy/server for voice, video, IM, presence, and any other SIP extensions. OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 residential platforms, trunking/wholesale, enterprise/virtual PBX solutions, session border controllers, application servers, front-end load balancers, IMS platforms, call centers, and many others. OpenSIPS is recommended for any kind of SIP scenario/service by the high throughput, tens of thousands of CPS, millions of ‏simultaneous calls (see official tests), the flexibility of routing and integration, routing scripts for implementing custom routing logics, several interfacing APIs. The effective application building, more than 120 modules to provide features, for SIP handling, backend operations, integration, the routing logic. OpenSIPS 3.2 release was the subject of a security audit, making everything safer and more secure for everybody.

Kamailio® (successor of former OpenSER and SER) is an open-source SIP server released under GPLv2+, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and real-time communications, presence, WebRTC, Instant messaging, and other applications. Moreover, it can be easily used for scaling up SIP-to-PSTN gateways, PBX systems, or media servers like Asterisk™, FreeSWITCH™, or SEMS. Among the powerful features, are asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video, text); WebSocket support for WebRTC; IPv4 and IPv6; simple instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached.