SIP Essentials

Learn all about the Session Initiation Protocol and other protocols related to SIP implementations

Course Code : 6002

$1795

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.

Schedule Classes

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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

  • SIP Message Format
  • Legacy Call Control
  • Compare SIP
  • Packetizing Voice
  • SIP Call Flow
  • How SIP Routes Media
  • SIP Call Control
  • SIP in 4G
  • SIP UA
  • SIP Requests
  • SIP Response
  • SIP URI
  • SIP Architecture
  • SIP Domain
  • SIP Registration
  • SIP Call Routing
  • Loose Routing
  • Metacharacters
  • Substitution
  • REGEX Modifications
  • Proxy Routing
  • Via and Record-Route
  • SIP Dialog
  • The reINVITE
  • SIP Topology
  • SIP Proxy
  • B2BUA
  • Outbound Proxy
  • Wireshark Colors
  • Wireshark Preferences
  • SIP Stack
  • REGISTER with Authentication
  • Wireshark Analysis of SIP Dialog
  • SIP Redirect
  • CFNA
  • REFER and Call Transfer
  • PRACK 100-rel
  • Call Forking
  • Loop and Spiral
  • Third Party Call Control
  • Path Minimalization
  • SIP in the PLMN
  • OPTIONS Method
  • URI vs. URL vs. URN
  • SIP URI Examples
  • URI Delimeters
  • SIP and SIPs
  • tel URI
  • URI Escape Codes
  • 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
  • SDP Background
  • SDP Format
  • SIP = one way?
  • SDP Lines
  • SDP Offer/Answer
  • Call Hold
  • Zone File
  • SOA and NS Records
  • A-Record
  • SRV Record
  • NAPTR Record
  • Locating SIP Servers
  • ENUM Database Example
  • ENUM Query and Response
  • ENUM REGEX
  • Post ENUM Routing
  • Early Media
  • SIP-T and SIP-I
  • RTP Headers
  • RTP Dejitter
  • Conferencing
  • RTCP
  • DTMF
  • SIP INFO
  • RFC 2833
  • 30
  • 38
  • SDP RFC 3407
  • Presence overview
  • PIDF XML example
  • Rich Presence
  • Presence Message Flow
  • Instant Messaging
  • Standard Timer Values
  • Session-Expires
  • Security for Call Setup
  • Authentication
  • S/MIME
  • TLS
  • 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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