SIP Training Course


In this hands‐on workshop you will develop an expert knowledge of Session Initiation Protocol. Learn the latest, up to the minute developments with this suite of critically important Internet Engineering Task Force standards. Join your colleagues for an intense, hands‐on workshop covering the latest SIP developments.

TrainingCity was the original global pioneer in SIP & VoIP training over a decade ago.  We've been at this a long time, and we know this stuff!  Join our team of experienced VoIP/SIP experts and learn SIP the right way, real world hands on labs, an instructor who's seen it all and can answer your toughest questions.  Call today to speak to one of our SIP experts and discuss your unique VoIP/SIP learning objectives.

Who Needs to Attend

Anyone working with VoIP, Video, or emerging VoLTE, IMS services needs to understand SIP.  SIP is the core signaling protocol that drives all modern IP based communications systems.  The good news is that you can become a SIP Expert.  We can help you!


TCP/IP Training Fundamentals is essential for all attendees. In general, it is highly recommended that all attendees complete the VoIP hands On Workshop.  Attendees who work at major carrier clients such as VerizonAT&T, Rogers, Bell Canada, Hawaiian Telcom, etc and software designers with a specific interest in SIP, can usually bypass the VoIP class if they are specifically focused on SIP Trunking related issues.

Detailed Course Outline

Module One: Introduction to SIP
  1. The role of Signalling in Multimedia Communications Networks
  2. Introduction to Session Initiation Protocol
  3. Basic SIP Call Flow
  4. Role of Session Description Protocol: SDP
  5. IETF Standards Development
  6. Current Trends in SIP Standards development
  7. Introduction to RFC 3261
  8. Core relevant SIP RFCs Reviewed
  9. Signalling vs Transport
  10. Multimedia Transport Considerations
  11. Layer 2/IP/UDP/RTP/Codec
  12. Nature of RTP Stream
  13. RTP socket selection
  14. Role of RTCP
  15. NAT Traversal Considerations
  16. Other Signalling Protocols
  17. H.323, MGCP, IAX2, Vendor Specific
  18. Signalling in the PSTN
  19. Overview of SS7 Operation
  20. SS7 Call Flow examples
  • Lab 1.0: Install & configure Wireshark™ Protocol Analyzer
  • Lab 1.1: Review core Telephony Capture Functions of Wireshark
  • Lab 1.2: Capture & analyze various TCP/IP Traffic
  • Lab 1.3: Capture Multimedia Codec/Real Time Protocol
Module Two: SIP Operational Overview
  1. SIP + SDP
  2. Application Layer Protocol
  3. Comparison with HTTP
  4. SIP User Agent
  5. UAC/UAS model
  6. SIP Servers
  7. The need for a Proxy Server
  8. Registration Server Function
  9. Redirect & Location Server
  10. Examples of SIP devices
    1. SIP Terminals & Phones
    2. SIP Servers & IP PBX devices
    3. SIP PSTN Gateways
    4. Session Border Controllers
  11. B2BUA Considerations
  12. NAT Traversal Challenge & Options
  13. ICE/STUN/TURN Explained
  14. DNS & SIP
    1. DNS Operations with SIP
    2. SRV RR
    3. NAPTR RR
    4. A/AAAA RR
  • Lab 2.0: Case Study: Enterprise SIP Deployment Architecture
  • Lab 2.1: Setup SIP UA to SIP Proxy Server NAT Traversal
Model Three SIP Architecture
  1. The Request / Response Model
  2. Information Contained in the SDP
  3. The nature of SIP Methods and numbered response messages
  4. SIP Header Overview
    1. SIP Request / Status Line
    2. Required Header Fields
      1. Via, To/From/Contact, Call‐ID, etc.
  5. SIP Message
    1. Call Flow Diagram of REGISTER Method with MD5 Authentication Process
    1. Map Call Flows for various INVITE Methods, including basic call setup, transfer, third party
    2. call control, etc. Examples of SDP carried by INVITE & 200OK
  9. Operational use of other SIP Methods
  10. SIP Response Messages
    1. 100, 180 Ringing, 182, 183
    2. 200OK
    3. 300, 302 Redirect operational call flow examples, others
    4. 400, 500, 600 various responses
  • Lab 3.0: Setup & configure an Open Source SIP Server in Linux Ubuntu Distro
  • Lab 3.1: Configure SIP User Agents (SIP Phones) to Register with SIP Server (IP PBX) over IPv4
  • Lab 3.2: Capture REGISTER METHOD in Wireshark. Examine Authentication process using MD5
  • Lab 3.3: Capture INVITE METHOD SIP Call Flow. Analyze Call Flow Diagrams
  • Lab 3.4: Capture Transfer Call Flow to include exchange of SDP with ACK Method
Module Four: SIP Call Flows
  1. SIP Call Flow Diagrams
  2. PBX Feature Replication
  3. SIP Call Flows for Core PBX features
  4. Transfer, Hold, VM, MoH, Etc.
  5. Third Party Call Control
  • Lab 4.0: Capture & Analyze various SIP Call Flow Diagrams using Wireshark
  • Lab 4.1: Troubleshooting Multi‐vendor SIP Call Flows
Module Five SIP & SS7 Integration
  1. SS7 Overview
  2. Q.Sig, Q.931, CAS considerations
  3. Issues with T‐1 vs PRI connectivity at PSTN Gateway
  4. E911 Implementation Considerations & Troubleshooting
  5. SIP / PSTN Call Flows
  6. SIP to SS7, SS7 to SIP
  • Lab 5.0: Capture & troubleshoot SIP to SS7 Call Flows
Module Six: SIP Trunking
  1. Practical Implementation of SIP Trunking
  2. Regulatory & E911
  3. IP PBX Deployment Example
  4. SIP Addressing for PSTN Integration
  5. E.164 Number integration
  6. SIP Trunking Considerations
  7. Privacy & SIP Proxy Servers
  8. Trusted Domain
  9. RFC 3325 Overview
  10. Privacy Header
  11. P‐Asserted‐Identity Header
  12. P‐Preferred‐Identity Header
  13. Remote‐Party‐ID
  14. NAT Traversal
  15. The role of the ITSP
  • Lab 6.0: Establish SIP Trunk to ITSP
  • Lab 6.1: Capture & analyse SIP Trunking traffic with ISTP
Module Seven: SIP Security
  1. Default use of MD5 Authentication
  2. Role in preventing Replay Attacks
  3. How to Determine Level of Exposure
  4. Use of ACL with SIP
  5. Authenticated Identity Body Applications for Security in RFC 3893
  6. Signalling & Media Security Considerations
  7. SIPS, TLS, IPsec
  • Lab 7.1: Establish SIPS Sessions
  • Lab 7.2: Implement IPsec ESP SIP Signalling
  • Lab 7.3: Troubleshoot Hacking Scenarios

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