We will analyze the steps to make audio & video communications (as SIP Phone WebRTC) into your WebApp, exploiting Asterisk WebRTC techology. The talk shows pros e cons of two different implementations: one using sipML5 library and one with Janus Gateway. Asterisk WebRTC technology open huge scenarios of applications for unified communications. In this session we will look at that technology to make a SIP Phone WebRTC directly integrated into your web browser to provide a real-time audio & video communication WebApp that serves hundreds of contemporary calls. We will consider two different solutions, sipML5 and Janus Gateway, showing pros and cons of both solutions.