WebRTC Deployment

Specs

  • 5 Day Course

  • Lecture and Labs (with Instructor led Training)

Learn how to build, deploy and troubleshoot web-rtc services using all open source components. Finish the course with a working system available by cloning the Alta3 "sipgate" github repository which provides a working framework, ready to provide the foundation for production web-rtc applications.

  1. NAT Essentials - Understand the technical details of traversing Full cone, ip restricted, port restricted, and symetitrical NATs

  2. STUN - A detailed analsysis of the STUN protocol

  3. TURN - A detailed analysis of the TURN protocol

  4. ICE - Learn how ICE manages NAT traversal.

  5. STUN/TURN/ICE labs - configure and deply coturn.

  6. NGINX - Learn how to configure NGINX as a reverse proxy, SSL edge, and preread forwarder.

  7. websocket essentials - Deploy, manage and troubleshoot websocket based services, especially as deployed in web-rtc.

  8. Kamailio ws (web socket) - Deploy a secure proxy to act as carrier grade websocket to SIP gateway.

  9. RTP Engine - Learn how to manage, troubleshoot and deploy the portion of NAT traversal logic that performs the work of RTP relay.

  10. js.sip - Learn the essentials of js.sip, arguably the most popular open javascript client in common use today.

  • A valid domain: sip.alta3.com

  • NTP config update

  • A SSL certificate: lets-encrypt

  • SIP client: js.sip

  • The web server: NGINX

  • Kamailio Secure Web socket to SIP gateway: WSS to SIP

  • siremix: kamailio DB manager

  • NAT Traversal RTP proxy: rtpengine

  • A turn server for NAT traversal: coturn

  • A SIP target to call: asterisk server

labs diagram
  • None required

  • Any company or individual who wants to advance their comprehension of web based real time communication

  • Session Initiation Protocol

  • VoLTE and the IMS

  • 5G Essentials

Overview

Objectives

Contact us to schedule!