Hands-On Voice Over IP - VoIP Data and Voice

Course Description

This Hands-On course provides in-depth coverage on Voice over IP (VoIP), which integrates voice and data transmission, is quickly becoming an important factor in network communications.

It promises lower operational costs, greater flexibility, and a variety of enhanced applications. This course provides a thorough foundation to this new technology to help experts in both the data and telephone industries plan and support their VoIP networks.

This Hands-On course will allow participants to work with an in-class intranet infrastructure constructed with Ethernet switches and routers. Upon successfully constructing and testing the in-class Intranet, participants will observe the installation of voice and video hardware and software on PCs. Participants will test for voicevideo quality over the LAN and WAN. These labs are very useful methods to understand the A-Z of VoIP.

Students Will Learn

Explain the basics of telephony and TCPIP
Understand the engineering tools and procedures required for a voice network and the current technologies leading to the integration of voice and data networks
Explain the basics of Voice over IP (VoIP)
Understand existing and emerging standards for VoIP and network architectures to support VoIP
Understanding Carrier Grade VoIP Technologies
Describe the protocols that support VoIP calls and explain how IP works with the PSTN
Identify some of the challenges VoIP faces in today's networks to demonstrate a good understanding of its capabilities
Explore the latest enabling technologies
Explain SoftswicthMGC, Media Gateways, SIP, Megaco, and MGCP
Voice characteristics, compression standa Mean Opinion Scores (MOS)
Learn about the functional components involved in using gateways to deploy VoIP networks
Explain the concepts of quality of service enforcement techniques
Explain performance and voice quality considerations
Discuss VoIP OSSBSS
Review transitioning to the All-VoIP PSTN and VoIP Taxation
Explore project planning process of VoIP
Review successful VoIP deployments for wireline, wireless, and cable operators
Discuss successful and unsuccessful VoIP deployments
Step through a practical process for managing a VoIP deployment project
Explore the current and future market trends
Discuss Video Services Over IP
And More...

Target Audience

This course is designed for anyone who plans on using, evaluating or working with VoIP networks, applications and services.

Prerequisites

Basic telephony and understanding of Basic TCPIP is recommended.

Course Outline

Module I: Introduction

Telephony Terminology
Public Switched Telephone Network (PSTN)
Circuit Switched and Packet Switched
Voice and Video Communications
VoIP and Next- Generation Services
Voice Compression
Performance and voice quality considerations
Voice & Data Convergence
The Numbering Plan for Telephony - E.164
Signaling
Q.931 Messages
Signaling System Number 7 (SS7 )
Examples of the SS7 Functions
SS7 Signaling Architecture
Challenges In Transitioning to an all VoIP
Carrier Grade Services
Regulations and Taxation

Module II: Basics of TCP/IP

Transmission Control Protocol (TCP)
User Datagram Protocol (UDP)
Connection-Oriented Protocol vs. Connectionless Protocols
Basic of IP Addresing
DHCP (Dynamic Host Configuration Protocol)
DNS
Unicast, Broadcast, and Multicast Addresses
Internet Control Message Protocol Technique

Module II: Basic of VoIP

Network Architectures to Support VoIP
Voice over IP Requirements
Support of Real-Time Services
Voice Data Transport
Encapsulation of Voice Data
Voice Encoding and Packetization
Issues in Packet Voice Communication
Real Time Transport Protocol (RTP)
Real-Time Transport Protocol Architecture
Real-time Control Protocol (RTCP)
RTCP QoS
Real-time Streaming Protocol (RTSP)
VoIP Signaling Approaches
H.323
MGCP
Megaco
SIP
Sigtran
SCTP
Evolution of VoIP Signaling Protocols

Module III: Voice Compression

Voice Quality
Voice Sampling
Quantization
Speech-Coding Techniques
Components of a Conversation
Voice Activity Detection
Fixed Sampling
Silence Suppression
Mean Opinion Score (MOS)
Code Excited Linear Prediction
Local Delay Compression
Analysis-by-Synthesis (AbS) Codecs
G.71: Pulse Code Modulation (PCM)
Linear Predictive Coders (LPCs)
G.723.1 and G.729
G.721
G.723.1 ACELP
G.726
G.727
G.728 LD-CELP
G.729A
Selecting Codecs
Cascaded Codecs
ADPCM
LDCELP
CS-ACELP
MPMLQ
FR
EFR
AMR

Module IV: Network Delay Considerations

The IP Datagram
TCP Congestion-Control
Hypothetical Signaling Reference Connection (HSRC)
End-to-end Voice Transmission
Transmission time allocation
Delay Problems
Delay and Jitter
Induced Delay
Processing delay
Network Congestion
Call Setup Delays
Post Dial Delay
Propagation Delay
Handling Delay
Queuing Delay
Lost Packet Compensation
Jitter
IP Precedence
Priority Queuing
Random Early Detection (RED)
Weighted Precedence
Weighted RED
Weighted Fair Queuing

Module V: VoIP Signaling and Protocols

VoIP Signaling Transport
H.323 Architecture
H.245 Control Signaling
H.248/Megaco
Session Description Protocol (SDP)
Session Initiation Protocol (SIP)
SIP Call Process
Call Setup: SIP to ISDN
Call Setup: SIP to SS7
Gateway Protocols
Media Gateway Control Protocol (MGCP)
MGCP and Megaco Protocol Architecture
MGCP and Megaco Protocol Evolution
MGCP Architecture
MGCP Stack
Megaco Features
Megaco and Megaco Enhancements
Session Initiation Protocol vs. Softswitch (MGCP and MEGACO)
MEGACO and H.248
Protocol Comparison
Simple Control Transmission Protocol (SCTP)
Sigtran Architecture
M2UA and M3UA
Simple Conference Control Protocol (SCCP)
Real Time Streaming Protocol (RSTP)
Comparison of VoIP Call Control Protocols

Module VI: QoS Considerations in VoIP Network Design

Voice over Packet (VoIP, VoATM)
Quality of Service (QoS)
QoS Building BlocksVoIP Quality of Service Provisioning
VoIP QOS Requirements Delay, Loss, Blocking
Issues Affecting Voice in an IP Network
Delay in General
Fixed and Variable Delay
Variable Delay Factor: Jitter
Variable Delay Factor: Packet Loss
Variable Delay Factor: Sequence Errors
Variable Delay and QoS Mechanisms
Quality Measurements and Mean Opinion Scores (MOS)
Echo and the Use of Echo Cancelers
Delay Variation and Jitter Buffers
Managing Delay Addressing Quality of Service
Bandwidth Limitations
Traffic Policing
Traffic Shaping
IP QOS Mechanisms
Integrated Services
Differentiated Services
Resource Reservation Protocol (RSVP)
Integrated Services (Int Serv) and QoS
Int Serv Service ClassesIntegrated Services Architecture
Differentiated Services (DiffServ) and QoS
Common Open Policy Server (COPS)
Multi-Protocol Label Switching (MPLS) and QoS
MPLS Operation
MPLS-TE

Module VII: VoIP OAM&P and deployment

Overview of Management Layers
VoIP SLA Management Architecture
VoIP Management Data Sources
Performance and Fualt Management Functions
VoIP Provisioning
Service Control and QoS
Maintaining an SLA
Billing and CRM
Security and Reliability

Module VIII: VoIP Implementations

VoIP Services
VoIP over DSL/Cable Modems
Large-Scale VoIP Service Deployment
Business Operations Solutions
VoIP BSS/OSS and NMS
VoIP E-911 and CALEA
High-Availability VoIP Network
Standards and QoS
Interworking with the Public Switched Telephone Network (PSTN)
Implementing the PSTN Switch/VoIP Gateway Trunk
IP Centrex
Video Bundling With VoIP
Softswitch
Services and Implementation Issues
Security Considerations for VoIP Networks
Network Access Security
Device Security
Using IPSec
VoIP Traffic Cases
VoIP Implementations Lessons Learned
Revenue Assurance for VoIP
Potential hidden pitfalls
Deployment issues and challenges
A project framework for managing a VoIP deployment
Pulling it All Together
Case Studies

Notes

LAB 1 Using Lines and IP Bandwidth Calculation

It can be used to estimate the bandwidth required through an IP based network for a fixed number of voice paths. voip bandwidth calculator calculates the bandwidth requirements for voice over IP. This depends on the voice compression scheme used (e.g. G.711, G.723, G.728, G.729), the packet interval and the transport protocol (RTP, cRTP).

LAB 2 Erlangs to VoIP Bandwidth Calculation

The Erlangs and Bandwidth Calculator can be used to estimate the bandwidth which must be provided through an IP based to satisfactorily transport a given busy hour traffic level.

LAB 3 Build a shared Ethernet and share multimedia files.

LAB 4 Build a switched Ethernet and share multimedia files.

LAB 5 Engage in H.323 audio and video session using your workstation.

LAB 6 View a quality of service (QOS) policy built on a workstation.
Voice Over IP

LAB 7 Construct a router network and view its performance using various delay
measurements.

LAB 8 Configure your gateway for successful voice and fax transmission.

LAB 9 Using router utilities to view a Q.931H.323 protocol connection establishment.

LAB 10 Using your protocol analyzer view multimedia packets in detail.

LAB 11 Configure various VOIP hardware and software products.

And More...

Delivery Method

Instructor-Led with numerous Hands-on Labs and exercises.

Equipment Requirements
(This apply's to our hands-on courses only)

BTS always provides equipment to have a very successful Hands-On course. BTS also encourages all attendees to bring their own equipment to the course. This will provide attendees the opportunity to incorporate their own gear into the labs and gain valuable training using their specific equipment.

Course Length

3 Days