SIP & Wireshark - Public - Guaranteed to Run

Learn more about Session Initiation Protocol (SIP) and Wireshark in this 5 day public online event, always guaranteed to run! Priced to ensure you end up in the class, join YouTube’s CodeWithFeeser (Russell Zachary Feeser) as we explore the protocol that started the VoIP revolution, SIP!

Today, SIP serves as the backbone of nearly every modern communication management system—powering platforms like Cisco Call Manager, Avaya Session and Communication Manager, Avaya IP Office, Oracle Session Border Controllers, Ericsson IMS cores, Asterisk, ShoreTel, and Mitel solutions.

In this course, you'll dive into live call analysis using tools such as Wireshark and TCPDump. Through hands-on work with both your own packet captures (PCAPs) and a comprehensive library of pre-captured traffic, you’ll gain a deep understanding of how SIP initiates calls, how RTP transmits real-time voice and video, and how to identify and troubleshoot communication errors effectively.

The course content is highly technical and structured to provide a clear learning path. Students will receive an official SIP certification upon completion.

Next Class

May 19-23, 2025

10:00 AM - 6:00 PM

Virtual: Online - US/Eastern

Course Outline

VOIP Fundamentals

• 💬 Introduction to VoIP

Packet Captures

• 💻 Upload PCAPs for analysis (OPTIONAL)

• 💻 Introduction to Wireshark

• 💻 Termshark

SIP Registrars

• 💬 SIP Architecture

• 💻 Successful REGISTER by a User Agent

• 💻 REGISTER Fails Auth

• 💻 deREGISTER Log Out

SIP INVITE

• 💬 Regular Expression

• 💻 Building a Dial Plan with RegEx

• 💬 Routing the INVITE

• 💻 The SIP INVITE

• 💻 SIP INVITE Packet Analysis with Wireshark

Establishing Calls

• 💬 SIP Dialog

• 💻 Troubleshooting Common SIP Failures with Wireshark

• 💬 SIP Entities

Call Flows

• 💬 Basic SIP Call Flows

• 💬 SIP 3xx Redirection

• 💻 Call Forwarding or 3xx responses

• 💬 SIP REFER

• 💻 SIP REFER for Call Transfer

• 💬 SIP PRACK 100rel

• 💻 SIP PRACK 100rel

• 💬 Call Forking

• 💻 Call Forking

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

• 💬 ENUM

• 💬 Interop with the PSTN

SIP Tools

• 💻 Install Asterisk

• 💻 Linux Fundamentals

• 💻 Using vim

• 💻 Making pcaps with tcpdump

• 💻 Making pcaps with tshark

• 💬 SIPp

• 💻 SIPp SIP Tester

• 💻 SIP Swiss Army Knife

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

Fax

• 💬 Fax Handling

Presence

• 💬 Presence

• 💻 SIP PUBLISH

• 💻 Presence and IM Exchange

SIP Timers

• 💬 SIP Timers

SIP Security

• 💬 SIP Security

NAT Issues

• 💬 NAT