Integrating telephony directly into web applications has become increasingly accessible with Web Real Time Communications (WebRTC), a technology that allows developers to enable web-based calls by accessing microphones and cameras within a browser. WebRTC, coupled with a robust interconnect to the Public Switched Telephone Network (PSTN), facilitates secure data transmission to a media server, enabling seamless audio and video sessions. The process involves setting up a server for call control and managing participants, who can be web-based or PSTN-based, through sessions that handle media streams. By using straightforward scripting with tools like NodeJS and JavaScript, developers can create sessions, add participants, and connect audio streams with minimal code. The integration of a SIP Interconnect allows PSTN callers to join WebRTC sessions, ensuring all components are managed server-side. This approach is exemplified by Bandwidth’s WebRTC platform, which simplifies call setup and management, making it easier for developers to implement in-browser calling functionalities.