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SIP Essentials

Course Code: 6002
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Overview

This course helps participants gain a thorough understanding of what SIP is, how it works and how to practically use it. During the course, participants get to explore how SIP interoperates in the current telecommunications network, while also being able to understand the protocol beyond the basic fundamentals. The course covers not only the Session Initiation Protocol but also other protocols that are related to the SIP implementation.

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Trivera

Course Delivery

This course is available in the following formats:

Live Classroom
Duration: 5 days

Live Virtual Classroom
Duration: 5 days

What You'll learn

  • Introduction to SIP
  • Understanding the SIP architecture
  • Exploring the regular expressions
  • Learning how to route the SIP INVITE
  • Exploring the SIP dialog and the SIP entities
  • Understanding the SIP URIs, message headers and SDPs
  • Exploring SIP and the DNS
  • Learning about ENUM, legacy, RTP and RTCP
  • Handling DTMF and Fax

Outline

Module 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

Module 2: SIP Architecture

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

Module 3: Regular Expression

  • Metacharacters
  • Substitution
  • REGEX Modifications

Module 4: Routing the SIP INVITE

  • Proxy Routing
  • Via and Record-Route

Module 5: The SIP Dialog

  • SIP Dialog
  • The reINVITE

Module 6: SIP Entities

  • SIP Topology
  • SIP Proxy
  • B2BUA
  • Outbound Proxy

Module 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

Module 8: SIP Call Routing

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

Module 9: SIP Uniform Resource Indicators (URIs)

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

Module 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

Module 11: Session Description Protocol (SDP)

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

Module 12: SIP and the DNS

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

Module 13: ENUM

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

Module 14: Legacy

  • Early Media
  • SIP-T and SIP-I

Module 15: RTP and Real-Time Control Protocol (RTCP)

  • RTP Headers
  • RTP Dejitter
  • Conferencing
  • RTCP

Module 16: DTMF Handling

  • DTMF
  • SIP INFO
  • RFC 2833

Module 17: Fax Handling

  • 30
  • 38
  • SDP RFC 3407

Module 18: Presence

  • Presence overview
  • PIDF XML example
  • Rich Presence
  • Presence Message Flow
  • Instant Messaging

Module 19: SIP Timers

  • Standard Timer Values
  • Session-Expires

Module 20: SIP Security

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

Module 21: SIP NAT Traversal

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

Participants need to have an understanding of TCP/IP networking and VoIP.

Who Should Attend

The course is highly recommended for –

  • VoIP engineers
  • Asterisk developers
  • VoIP software developers
  • VoIP network engineers

Interested in this course? Let’s connect!

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