SIP Phone WebRTC into your browser: a real story from sipML5 to Janus
Submitted by Alessandro Polidori (@alepolidori) on Monday, 16 July 2018
Nowadays, WebRTC is a new technology that is opening up very huge scenario in distributed architectures to realize real-time audio/video communications, but it is still underused. This talk could be useful to all developers that are interested in realizing audio and video communication in their own web application.
You can not fall down not considering audio & video in your business. After this talk, you will be able to easily integrate an audio / video platform into your web application !
Do you want to realize a telephone into your Web App ? We will look at the technological choices behind the construction of a SIP Phone WebRTC integrated into your browser to provide a Unified Communication solution by WebApp. We will consider the pros and cons of two different solutions, from sipML5 to Janus Gateway in a real scenario in production on 3000 customers.
Software Engineer, with more than 7 years of experience in web technologies, distributed architectures and agile methodologies.
Expert in Node.js I am involved in design, implementation of code that powers core services, REST APIs, WebSocket communications and client side Web Applications with real-time WebRTC audio and video, applied to Asterisk PBX system.
Open source enthusiast (personally and for work) I spoke at Universal JS Day 2018 conference in Italy.