Training Session Initiation Protocol (SIP) in Real World Networks


Session Initiation Protocol (SIP) is the fundamental protocol for setting up and managing voice calls, and many text messages. It's used on mobile phones, desk phones, and on links between carriers in the US, Canada, and around the world. Get a deep-dive introduction and have your questions answered with real-world examples demonstrating the important features and troubleshooting methods for SIP in Voice and meeting networks. This course covers SHAKEN/STIR Caller Authentication. 1.5 week / 3 sessions / 6 PDH.

In-Person Training

On-site training at your location


You'll learn the basic architecture of SIP, and how it compares to other protocols like MGCP, MEGACO, ISUP/SS7/C7, and CAS. You'll learn about requests and responses, understanding the essentials of transactions and dialogs. You'll see how SIP registration works with full security as used in real-world networks today. Real-world telephone calls will show you how audio formats (codecs) are selected using Session Description Protocol (SDP), and how calls are routed through networks like those in Cisco BroadWorks, Metaswitch, Asterisk, Sonus, Ribbon and others. You'll also see advanced call flows like call hold and resume, and important PBX / UCaaS services like Busy Lamp Field - Line State Monitoring. Finally, you'll learn how IPv6 is used in SIP, and distinctives of SIP as used in Mobile/IMS networks.

Examples from real-world vendor software platforms, including Polycom, Cisco, Metaswitch, Oracle


Course Content

Voice & Video Network Components

  • Why awareness matters
  • Overview of Key component Types
  • Feature Servers / Application Servers
  • User Agents: SIP Phones, Analog Terminal Adapters (ATAs), PBXs
  • Enterprise Session Border Controllers (e.g., Ribbon EdgeMarc / Audiocodes E-SBC)
  • Redirect / Network Server
  • SIP Registrar Function
  • Session Border Controller (SBC)
  • DNS Servers used in SIP
    • SIP Server Location for SIP Registration
    • Call Routing (e.g., ENUM or URI-Based Routing)
  • Media Servers
  • Multipoint Control Unit (MCU)
  • PSTN Gateways (SIP to SS7/C7)
    • Premise-Based Gateways (Ribbon/Genband and Metaswitch/Microsoft examples)
  • Cloud Services
    • Conferencing
    • Interactive Voice Response
    • Call Recording

Session Initiation Protocol (SIP)

  • SIP is Evolving: Modularity and Extensions
  • SIP Requests, Responses, Transactions and Dialogs

SIP Registration

  • Avoiding Common Troubleshooting Mistakes
  • REGISTER Basics
  • SIP Authentication to fight fraud and Cybersecurity risks
  • How SIP Authentication with CSeq and MD5 Checksum Works

SIP Calling

  • INVITE with SDP
  • Session Description Protocol (SDP) and understanding media flow setup
  • Reading SDP Offer (c=, m=, attributes, and telephone-event)
  • DTMF transfer negotiation with RFC2833 / RFC4733 in SDP
  • SIP response codes like 100, 180, 183, 302, 401, and 503
  • Feature / Application Server / B2BUA Example with SIP Call-ID
  • Redirect / Network Server Operation
  • Call re-routing with SIP redirect
  • PRACK Operation and 100rel 
  • Ending the Call
  • Ladder Diagrams

SIP: Other Dialogs beyond Registration and Calling

  • In-Dialog Invites / Re-INVITE
  • Reading the To-tag in the INVITE
  • Multiple Methods of call hold (a=inactive vs c=
  • Cisco BroadWorks example of Busy Lamp Field with Polycom
  • Microsoft Metaswitch Line State Monitoring  with Cisco
  • Feature Key Synchronization (Do Not Disturb)
  • Call Transfer with REFER
  • Cold (Unattended) and Warm (Attended / Consultative) Transfers
  • Refer-To Header
  • Important Headers for Interop
    • Caller Privacy
    • Call Redirection
    • Identity header for SHAKEN/STIR
  • SIP in IMS (IP Multimedia Subsystem) as used in Mobile / Cellular networks
  • Compressed SIP Headers 
  • Specialised IMS Headers
  • Understand how IPv6 is encoded in SIP


  • How SHAKEN/STIR Works
  • New Identity header and PASSporT

Wireshark Troubleshooting Quick Start

  • Basic Introduction to Wireshark and Packet Capture
  • Sample Files 
  • Setting up Wireshark for VoIP Troubleshooting
  • Filtering and Searching Call Captures

Key Skills

  • Key Telecom network components: Feature / Application Servers, Redirect Servers, Network Servers, SBCs, Media Servers, Enterprise SBCs (e-SBC)
  • SIP Headers, Dialogs, Transactions, Methods, and Responses
  • SIP Calling and Registration used for NNI/Peering/Trunking and Registration
  • Advanced SIP Features for PBX Applications - Line State Monitoring, Line State Synchronization
  • Knowledge of SIP used in IMS and IPv6 networks
  • Wireshark familiarity for Call Troubleshooting