Transmitting voice over Internet has been around for decades,
so the principles behind it are not new. In the last 15 years or so
there has also been a huge standardization effort to ensure VoIP
providers could integrate to each other and interoperate with other
telephony systems like PSTN (and GSM).
In the meantime, the Open Source approach has spread all over the
technology spectrum and now almost all the components involved in a VoIP
system have their robust, scalable, Open Source implementation
(noticeable exceptions being the very successful Skype, which led the
market with proprietary solutions).
Whilst Internet and PSTN/GSM have naturally converged (see for example
the IMS approach), until a few years ago VoIP remained basically
isolated from Web browsers, apart from some plug-in solutions.
WebRTC is an Open Source project, mainly led by Google, which aims to
bring VoIP inside the Web browsers, with high quality, secure, audio and
video. This presentation focuses on the main components and principles
behind WebRTC, and how easy the WebRTC APIs are to use
After the presentation, there will be a hands on session where attendees
can experiment with the WebRTC APIs and even build a rudimentary
videochat in less than an hour.
Speaker:
"Giacomo is a developer of Real Time Communication solutions, with a
strong focus on server-side, Open Source solutions in the area of VoIP,
IM and more recently WebRTC. After several years developing the Truphone
Apps (
https://www.truphone.com/us/consumer/app/) platform, in 2015
Giacomo has founded RTCSoft (
http://www.rtcsoft.net/), providing
consulting and development service around Europe."
Twitter:
https://twitter.com/giavac
Slideshare:
http://www.slideshare.net/GiacomoVaccaLunedi link eventrbite per iscrizione