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ApiRTC

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What is Kamailio?

Kamailio® is a powerful open-source SIP server that emerged from the earlier versions of OpenSER and SER, and is released under GPLv2+, designed to efficiently handle thousands of call setups per second. This versatile platform is ideal for building large-scale VoIP systems, facilitating real-time communication, managing presence, supporting WebRTC, enabling instant messaging, and accommodating numerous other applications. Moreover, Kamailio excels in scaling SIP-to-PSTN gateways, PBX systems, and media servers, such as Asterisk™, FreeSWITCH™, or SEMS, with remarkable ease. Its rich feature set encompasses support for asynchronous protocols like TCP, UDP, and SCTP, ensuring secure communication through TLS for voice, video, and text, in addition to compatibility with WebSockets for WebRTC. The server also fully supports both IPv4 and IPv6, offers straightforward instant messaging and presence functionalities via an integrated XCAP server and MSRP relay, and facilitates asynchronous operations effortlessly. Additionally, it includes IMS extensions for VoLTE, ENUM capabilities, along with features for DID and least cost routing, load balancing, and routing failover to enhance reliability. Kamailio's comprehensive suite of accounting, authentication, and authorization features ensures robust security, while it seamlessly integrates with a variety of backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, and Memcached. With all these advanced capabilities, Kamailio stands out as a highly adaptable solution for contemporary communication infrastructures, making it a preferred choice among developers and organizations alike.

What is ApiRTC?

ApiRTC is a user-friendly library that enables developers to effortlessly integrate real-time communication features, including chat, audio, and video, into web and mobile applications. By leveraging WebRTC technology, it guarantees that media streams are securely encrypted and can ideally be transmitted through direct peer-to-peer connections; in cases where this is not feasible, a TURN server can be utilized for media relay. Serving as the core of our platform, ApiRTC is responsible for managing user connections, tracking presence, and setting up media interactions. The library is available in multiple formats to suit a variety of integration requirements, whether for web platforms, hybrid applications utilizing JavaScript, or native apps designed for iOS and Android systems. Additionally, Apizee's offerings for audio and video communication are based on the widely adopted WebRTC standard, which is supported by an expanding array of internet browsers. This increasing compatibility not only improves the performance of our solutions but also guarantees that users can enjoy audio and video streaming on most PCs, tablets, and smartphones, significantly enhancing the accessibility of our communication tools. Moreover, developers can tap into the versatile capabilities of ApiRTC for diverse use cases, making it a crucial component in the toolkit for contemporary application development, thus ensuring that their projects meet modern user demands effectively.