We're sorry but this page doesn't work properly without JavaScript enabled. Please enable it to continue.
Feedback

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

685
 Cite
 Share
 Download
 Purchase
Logo of FOSDEM VZWFOSDEM VZW
 Polidori, Alessandro

Formal Metadata

Title
Asterisk WebRTC frontier: make client SIP Phone with sipML5 - Janus Gateway
Subtitle
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
Title of Series
Number of Parts
561
Author
License
CC Attribution 2.0 Belgium:
You are free to use, adapt and copy, distribute and transmit the work or content in adapted or unchanged form for any legal purpose as long as the work is attributed to the author in the manner specified by the author or licensor.
Identifiers
Publisher
Release Date
Language

Content Metadata

Subject Area
Genre
Abstract
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.