SIP Essentials

In this course, students learn Session Initiation Protocol and important protocols related to SIP implementations. This course thoroughly explains what SIP is, how it works, and also provides a practical guide on how to use it. The lessons in this course are clear and very technical. In this course, students will examine how SIP interoperates in the current telecommunications network, going beyond the basics of the protocol and getting a big picture understanding of how it all fits together.

Days : 5
Price :

CAD$2,855.00

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Description

At the end of the course, students will receive access to the Alta3 Research SIP certification exam. Upon successful completion of the exam, students will be awarded a SIP certificate.

A vendor-agnostic study of Session Initiation Protocol designed for the person that must make SIP work.

Who should attend

Any company or individual who wants to advance their comprehension of VoIP and SIP.

Prerequisites

None Required

Follow-On Courses:

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Outline: SIP Essentials (SIP)

1. SIP Introduction

  • SIP Message Format
  • Legacy Call Control
  • Compare SIP
  • Packetizing Voice
  • SIP Call Flow
  • How SIP Routes Media
  • SIP Call Control
  • SIP in 4G

2. SIP Architecture

  • SIP UA
  • SIP Requests
  • SIP Response
  • SIP URI
  • SIP Architecture
  • SIP Domain
  • SIP Registration
  • SIP Call Routing
  • Loose Routing

3. Regular Expression

  • Metacharacters
  • Substitution
  • REGEX Modifications

4. Routing the SIP INVITE

  • Proxy Routing
  • Via and Record-Route

5. The SIP Dialog

  • SIP Dialog
  • The reINVITE

6. SIP Entities

  • SIP Topology
  • SIP Proxy
  • B2BUA
  • Outbound Proxy

7. SIP Call Flow Examples

  • Wireshark Colors
  • Wireshark Preferences
  • SIP Stack
  • REGISTER with Authentication
  • Wireshark Analysis of SIP Dialog
  • SIP Redirect
  • CFNA
  • REFER and Call Transfer

8. SIP Call Routing

  • PRACK 100-rel
  • Call Forking
  • Loop and Spiral
  • Third Party Call Control
  • Path Minimalization
  • SIP in the PLMN
  • OPTIONS Method

9. SIP Uniform Resource Indicators (URIs)

  • URI vs. URL vs. URN
  • SIP URI Examples
  • URI Delimeters
  • SIP and SIPs
  • tel URI
  • URI Escape Codes

10. SIP Message Headers

  • SIP Header Overview
  • Dialog ID Headers
  • User-Agent
  • SIPp Header Modification
  • Proxy-Authenticate
  • Allow and Supported
  • History Info
  • Join
  • Session Expires
  • PPI and PIA
  • Establish Service Path
  • IMS Hosted
  • Content-Type

11. Session Description Protocol (SDP)

  • SDP Background
  • SDP Format
  • SIP = one way?
  • SDP Lines
  • SDP Offer/Answer
  • Call Hold

12. SIP and the DNS

  • Zone File
  • SOA and NS Records
  • A-Record
  • SRV Record
  • NAPTR Record
  • Locating SIP Servers

13. ENUM

  • ENUM Database Example
  • ENUM Query and Response
  • ENUM REGEX
  • Post ENUM Routing

14. Legacy

  • Early Media
  • SIP-T and SIP-I

15. RTP and Real-Time Control Protocol (RTCP)

  • RTP Headers
  • RTP Dejitter
  • Conferencing
  • RTCP

16. DTMF Handling

  • DTMF
  • SIP INFO
  • RFC 2833

17. Fax Handling

  • T.30
  • T.38
  • SDP RFC 3407

18. Presence

  • Presence Overview
  • PIDF XML Example
  • Rich Presence
  • Presence Message Flow
  • Instant Messaging

19. SIP Timers

  • Standard Timer Values
  • Session-Expires

20. SIP Security

  • Security for Call Setup
  • Authentication
  • S/MIME
  • TLS

21. SIP NAT Traversal

  • NAT
  • NAT Types
  • STUN & TURN

SIP Essentials Labs

  • Understanding the Network
  • Lab 1: Construct & Enable a VoIP Network
  • Lab 2: SIP User Agent Configuration
  • Lab 3: Direct UA to UA Routing with No Proxy
  • Lab 4: Proxy Based SIP Routing
  • Lab 5: Adding Authorized UAs to a Domain
  • Lab 6: Intra Domain Routing (SIP in the same domain)
  • Lab 7: SIP REGISTER – Registering a SIP UA
  • Lab 8: Registering a SIP UA Softclient
  • Lab 9: Registering a SIP UA Client to a Mobile Device
  • Lab 10: Inter Domain Routing (SIP in different domains)
  • Lab 11: Strip off the Leading ‘9’
  • Lab 12: PDT Management
  • Lab 13: Using Wireshark
  • Lab 14: Capture a SIP Registration via Wireshark
  • Lab 15: Capture a ‘Normal’ SIP Call via Wireshark
  • Lab 16: Capture a Call to a Vacant Number via Wireshark
  • Lab 17: Capture a SIP Call to Busy Number via Wireshark
  • Lab 18: Capture a Call Forward via Wireshark
  • Lab 19: Via, Record Route, and Route Headers
  • Lab 20: Examining ‘Max Forwards’
  • Lab 21: INVITE with SDP – sendonly vs. sendrecv
  • Lab 22: Silence Suppression
  • Lab 23: DTMF RFC 2833 and SIP INFO
  • Lab 24: SIP B2BUA Configuration Example
  • Lab 25: Register Linksys SIP Phone with Asterisk PBX
  • Lab 26: SIP Presence (NOTIFY)
  • Lab 27: RTP Relay
  • Lab 28: Direct RTP Flow Between Two UAs – 3PCC