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Training Course Outline

Data Networking Courses   Voice over IP VOIP Training Courses

Voice over IP using SIP

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Course Duration: 3 days

Course Code: PDN023

Course Description:

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 other PCs to the classroom network and take part in capturing traffic and sizing networks using spreadsheets.

The course will be 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.

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

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.

Prerequisite courses:

Hands On TCP/IP and Internet Protocols

ICND2: Interconnecting Cisco Network Devices Part 2

Follow on courses:

IPTV: Internet Protocol Services for TV Distribution and Switching

Understanding MultiService Access and SIP

IP Multicasting for Multimedia Distribution Networks

This course includes the following modules:

Introduction and Background

• 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

• 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

• 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

• 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

Revue and evaluation

 

Location	Duration	RRP	May	Jun	Jul	Aug	Sep	Oct
Click on course date to make a booking or check availability.
London	3 days	£1495			28 - 30			27 - 29

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