Voice over IP using SIP

3 Day Course
Hands On
Code PDN023

Book Now - 3 Delivery Methods Available:

Scheduled Online Onsite

Overview

SIP is fast becoming the Voice over IP protocol of choice. During this 3-day course delegates will examine SIP technology and architecture and learn how a functioning VoIP service can be established.

Delegates will be able to take part in demonstrations of working SIP calls and see captured SIP traffic over a classroom network. They will also be able to connect their own Laptop or classroom PCs to the network and learn to capture traffic and examine sizing networks using spreadsheets.

This course is aimed at technicians, engineers and managers who need an understanding of SIP but do not need to go down to the development programming level. Should you require a more detailed course, these are available.

Objectives

When you have completed this course you will be able to:

  • Enter into discussion of products, services and technology that use VoIP
  • Plan and size a SIP based VoIP solution
  • Describe in overview the use of layered protocols for packet services
  • Distinguish the important Internet protocols
  • Compare and contrast IETF SIP with other approaches to VoIP
  • Compare the business case for using VoIP and for Circuit Switching carriage of voice

Modules

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Introduction and Background (17 topics)

  • Evolution of Telecommunications
  • Circuit Switched voice
  • Packet Switching Data
  • Motivation: Why use VoIP
  • Comparison between current voice and data networks
  • One Integrated Network
  • Sharing resources
  • Migration
  • Where VoIP can be deployed
  • Integration at the PBX
  • Integration at the PC
  • Integration at the desk with IP phones
  • Which IP Network
  • Internet Telephony
  • VoIP over an Intranet
  • Internet Telephony Service Providers
  • Demonstration 1: A working VoIP SIP call

VoIP Architectures (25 topics)

  • Source of VoIP standards
  • ITU H323 and IETF SIP evolution compared
  • Multimedia conference over packet network
  • What counts as Multimedia
  • Voice
  • Video
  • Conference
  • Sources and mixes
  • How does a normal phone call get connected
  • Call Map
  • Conversion to digital
  • Dialing and Signaling
  • Alerting and Call Progress Tones
  • Carrying voice over IP
  • Encoding voice using codecs
  • Preserving timing
  • Impact of Jitter
  • Removing Jitter
  • RTP
  • RTCP
  • How does A VoIP call get connected
  • Separation of signalling and media streams
  • TCP/UDP Port numbers
  • Impact of packet loss on signalling
  • Demonstration 2: Capturing signalling on working VoIP SIP call

VoIP using IETF Architecture SIP (27 topics)

  • Why has SIP become important?
  • SIP Components
  • SIP Addressing - the URI
  • Connection signaling
  • Signaling when the phone rings
  • Capabilities exchange
  • Closing calls
  • SIP Message Format
  • SIP Registration Services
  • Identification of users and phones
  • SIP Registrar
  • Mapping Identities
  • SIP Proxy Services
  • Redirection
  • Impact of NAT
  • SIP Location Services
  • Forwarding registrations
  • Canceling Forwards
  • Security in SIP exchanges
  • SIPS
  • Authentication
  • Encryption
  • Demonstration Session 3: SIP Proxy Controlled Services
  • 1. Setup a SIP Proxy controlled VoIP service
  • 2. Configure a SIP application
  • 3. Observe and capture SIP Registrar interactions and call connections
  • 4. Observe Network Performance Using Netmeter

Quality of the Voice (21 topics)

  • What Constitutes Quality
  • Delay and Availability
  • Understanding the speech and recognizing the person speaking
  • Quality Measures
  • Mean end to end delay
  • Mean Opinion Scores
  • Codecs
  • Companded PCM
  • ADPCM
  • CELP
  • G.711,G.726, G.728, G.729, G.723.1
  • Delivering QOS with VoIP
  • Mixing Voice and Data
  • RSVP
  • Diffserv
  • Weighted Fair Queues
  • Sizing VoIP Services
  • Demonstration 4: Quality of Service Planning
  • 1. Size a VoIP service using spreadsheets
  • 2. Predict Delay and QOS performance
  • 3. Mix voice and Data over a low speed Router-Router Link to deliver QOS in practice

Review and evaluation

Prerequisites

The course assumes a basic understanding of PCs, Windows, simple TCP/IP networking using PCs and elementary telecommunications. It does not assume any prior knowledge of VoIP or packet voice technology.

Additional Learning

The courses below may help you meet the knowledge level required to take this course. If you are unsure please ask a training advisor .

Scheduled Dates

Please select from the dates below to make an enquiry or booking.

Pricing

Different pricing structures are available including special offers. These include early bird, late availability, multi-place, corporate volume and self-funding rates. Please arrange a discussion with a training advisor to discover your most cost effective option.

Code Location Duration Price Nov Dec Jan Feb Mar Apr
PDN023

(Classroom or Online)
3 Days $2,085

What Our Customers Say

The training was delivered with a high level of expertise and excellence. Instructor was highly knowledgeable.”

Technical Trainer, Aviat Networks

Overall the course was really good, the trainer really understood the material and was very approachable.”

Customer Training Manager, Aviat Networks

Excellent course, informative and well-paced.”

CSE, Cisco

Course was very well outlined. Topics were great and bridged many gaps.”

System Engineer, Cable & Wireless

An excellent intro to video encoding & MPEG transport streams - I would definitely recommend it.”

Broadcast Engineer, Cisco

Definitely an excellent intro. Left me interested in learning more.”

Broadcast Engineer, Eircom

Excellent training course with real examples and practical classroom demonstrations.”

Transport Designer, Orange

Instructor knowledge and experience was excellent.”

Solutions Engineer, Akamai

Excellent course, very clear and well organised. Course content delivery was very good.”

Assistant Engineer, Dhiraagu

Very informative and appropriate.”

Network Support Technician, BT

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