FOSDEM 2019
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Schedule
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Events
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Developer rooms
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Real Time Communications (RTC)
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Asterisk WebRTC frontier: make client SIP Phone with sipML5 and Janus Gateway

Asterisk WebRTC frontier: make client SIP Phone with sipML5 and Janus Gateway

Analyzing a real project on production (www.nethvoice.it) we will look at two different implementations of a SIP Phone WebRTC of NethCTI Web App. We will see great code examples, WebRTC technologies and a real demo of an audio/video call

Track: Real Time Communications (RTC) devroom
Room: H.1309 (Van Rijn)
Day: Sunday
Start: 09:00
End: 09:20

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.