Clouds

WebRTC Deployment

Essential Skills Gained

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Understand the fundamentals of SIP, including its role in VoIP systems.

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Configure and troubleshoot SIP registrars, proxies, and endpoints.

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Analyze SIP call flows and packets using tools like Wireshark and Termshark.

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Integrate WebRTC with SIP for real-time communications.

Format

5 day course with lecture and hands-on labs.

Audience

Network Engineers

VoIP Administrators

Telecommunications Professionals

DevOps and Site Reliability Engineers

Developers working with SIP or WebRTC

Description

SIP and WebRTC: Comprehensive VoIP Training is a 5-day hands-on course designed to teach participants how to deploy, secure, and troubleshoot Voice over IP (VoIP) systems using SIP (Session Initiation Protocol) and WebRTC (Web Real-Time Communications). Combining focused lectures with practical labs, this course provides a deep dive into SIP messages, headers, call flows, and the integration of WebRTC for real-time communications. Students will work with tools like Wireshark and Termshark to analyze SIP packets, set up SIP registrars and proxies, and configure VoIP systems using platforms like Kamailio, RTPEngine, and Asterisk. Additional topics include NAT traversal, SIP security best practices, and media transport protocols like RTP and SDP. By the end of the course, participants will be equipped with the skills to implement and troubleshoot enterprise-grade VoIP systems.

Summary

  • 💻 Register for Poll

Introduction

  • 💻 Welcome to Alta3 Labs

  • 💻 Navigation

Getting Started

  • 💻 Vim: A Modal Text Editor

  • 💻 Efficient CLI Usage with Tmux

AI LLM Toolkit

  • 💻 Large Language Model toolkit for AI Solution Assistance

Software Control Management

  • 💻 SCM Option #1 - GitHub

Installing WebRTC Playground

  • 💻 Introducing the WebRTC Playground

  • 💬 RTPEngine

  • 💻 Install RTPEngine

  • 💬 Kamailio

  • 💻 Install Kamailio

  • 💬 SIP-JS

  • 💻 Install SIP-JS

  • 💻 Install nginx

  • 💬 STUN/TURN

  • 💻 DEMO-Install coturn

  • 💻 SIP REGISTER

  • 💻 SIP Domains

  • 💻 Dial Plan-PDT

  • 💻 DialPlan module

  • 💻 IP Tables

  • 💻 IP Table testing

  • 💻 Analyzing websockets

Installing a SIP B2B-UA

  • 💻 Install Asterisk

SIP Fundamentals

  • 💻 Introduction to VoIP

  • 💻 Termshark

SIP Registrars

  • 💻 SIP Architecture

  • 💻 Successful REGISTER by a User Agent

  • 💻 REGISTER Fails Auth

  • 💻 deREGISTER Log Out

SIP INVITE

  • 💻 Regular Expression

  • 💻 Routing the INVITE

  • 💻 The SIP INVITE

  • 💻 SIP INVITE Packet Analysis with Wireshark

Establishing Calls

  • 💻 SIP Dialog

  • 💬 SIP Entities

Call Flows

  • 💻 Basic SIP Call Flows

  • 💻 SIP 3xx Redirection

SIP Proxies

  • 💻 Call Routing

  • 💻 INVITE Relay by SIP Proxies

  • 💻 No Record Routes

  • 💻 SIP URIs

  • 💻 CANCELed SIP call

  • 💻 Global Failures or 6xx responses

Supporting Systems

  • 💻 SIP and the DNS

SIP Headers

  • 💻 Common SIP Headers

Session Description Protocol

  • 💻 Session Description Protocol

  • 💻 Session Description Protocol

  • 💻 SDP Video Call Setup

  • 💻 SDP Video Call Setup Fails

Real-Time Transport Protocol

  • 💻 Real-time Transport Protocol

  • 💻 One-Way Media

Dual Tone Multi Frequency

  • 💻 Transmitting DTMF

  • 💻 Methods for Transport of DTMF

SIP Timers

  • 💻 SIP Timers

SIP Security

  • 💻 SIP Security

NAT Issues

Your Team has Unique Training Needs.

Your team deserves training as unique as they are.

Let us tailor the course to your needs at no extra cost.