Bring phone calls to your web browser with Bandwidth and WebRTC

The text discusses the integration of telephony into web applications using Web Real-Time Communications (WebRTC). It explains that WebRTC allows developers to access microphones and cameras from within a browser, enabling users to make or receive calls directly from their web apps. The process involves setting up clients and servers, managing streams and subscriptions, and handling media flows. The text also mentions the use of SIP Interconnect for translating between SIP protocol and proprietary signaling used by WebRTC platforms. It provides a sample code in NodeJS and Javascript to demonstrate how to set up a call with both web-based and PSTN-based participants.