Overview

This comprehensive hands-on practical VoIP course has been developed and proven over many years to deliver an excellent, up-to-date coverage of present-day VoIP systems. Attendees learn about the three main VoIP architectures: SIP, H.323 and H.248/Megaco, and the course examines in-depth voice coding, media coding and call signalling, with hands-on protocol analysis. Attendees also learn about IP Quality of Service implementation, the markets, key VoIP vendors and their products.

Duration

3 days

Audience

  • People responsible for evaluating, planning, installing, configuring, administrating or supporting VoIP products and networks.
  • Personnel wanting to move into the field of modern IP-based voice communications and media streaming.
  • Level 2 and Level 3 support staff responsible for diagnosing and troubleshooting issues with VoIP systems.

Prerequisites

Delegates should ideally have a good understanding of the TCP/IP protocol suite and IP addressing prior to attending this course.

Objectives

Discuss VoIP products, services and VoIP network deployments with customers or suppliers.
Plan VoIP solutions, including capacity planning and system sizing.
Configure VoIP phones, applications, SIP converters and more.
Understand the different protocol interactions for call/session signalling, media streams and QoE reporting.
Capture and analyse VoIP traffic using a network protocol analyser.
Analyse and troubleshoot Quality of Service (QoS) issues in a VoIP network.
Compare and contrast ITU H.323, IETF SIP and proprietary approaches to VoIP implementation.

Content Headings

Overview of VoIP
Voice and data convergence
Components of a VoIP system
Standards employed in current VoIP solutions
The role of Voice Processing
The speech encoding process
Sampling, Quantisation, Coding, Framing
Silence suppression
Voice coding and compression standards
Adaptive encoding techniques
Coding fax signals
Voice codecs: G.711, G.722, G.721, G.723, G.726, G.723
Assessing voice quality
Mean Opinion Scores (MOS)
Detecting flaws in transmitted voice
Employing MOS ratings for codecs and real networks
Assessing Voice Quality
Measurable components
What to test and measure
P.800 / P.861 recommendations
PESQ

Operating Voice over IP
The issues when operating Voice over IP
Delay, Talker overlap, Echo
Jitter, Packet loss
Out of Order Delivery
The role of Voice Processing and DSP
Real-time Transport Protocol (RTP)
The role of RTP
RTP header in detail
RTP payload types
Real-time Transport Control Protocol (RTCP)
Conclusions

Introduction to Voice over IP signalling
Overview of signalling in PSTN networks
Overview of private network signalling
The major architectures and standards for Voice over IP
ITU H.323
IETF SIP
MGCP and Megaco/H.248
Cisco SCCP (Skinny)
VOIP in the enterprise
VOIP in PSTN Emulation Service (PES)

ITU / IETF Megaco / H.248
MGCP and Megaco
The Media Gateway Reference Architecture
End-to-End call setup
IETF Megaco
Megaco Terminations and Contexts
Megaco Commands
Megaco Packages
Megaco IP phone Media Gateway

Overview of IP QoS
Classifying IP traffic
Review of the IPv4 Datagram format
IPv4 Service (TOS) field
Precedence bits
DTR(C) bits
Characteristics of RTP media flows
Classifying packets in IPv6 networks
The need for QoS
IP Differentiated Services (Diff-Serv)
Queuing and Scheduling mechanisms
First-In First-Out (FIFO), Strict priority scheduling, Fair Queuing, Weighted-Fair Queuing (WFQ), Class-Based Queuing, Hierarchical Class Based Queuing (CBQ)
Coping with packet loss
Controlling admission
Employing Random Early Detection
Employing traffic shaping
IEEE 802.1p/Q
Operating IP over ATM networks
Overview of MPLS

SIP Overview
Introduction
SIP design requirements
The development of SIP
SIP and VOIP
SIP Vs. H.323 and H.248/MEGACO
What SIP does
– SIP and Next Generation Networks, NGN
– SIP and mobility
Why we would deploy SIP
The role of SIP within:
– Mobile
– Provider VOIP
– NGN solutions
– Unified Communications
– Multimedia
– SIP based contact centres and using SIP for contact centre hosting
– SIP Trunking
– Mini case studies

SIP Architecture and Components
SIP User Agents
SIP Registrar
SIP Proxies
SIP Location server
SIP Redirect Server
SIP Back to Back User Agent (B2BUA)
SIP PBX

Overview of SIP Operation
The SIP User Agent client and server
The SIP URI
SIP Methods and Responses
SIP message exchange
– SIP Messages involved setting up a simple SIP call
– INVITE method
– 100 Trying
– 180 Ringing
– 200 OK
– ACK
– BYE method
Overview of other widely used SIP messages
– SUBSCRIBE
– OPTIONS
– NOTIFY
– REINVITE
– PUBLISH
– INFO
– PRACK

SDP Description and Role
Role of SDP
Structure of SDP
SDP operation

SIP Registration and Location Servers
Role of the SIP Registration Server
SIP Registration Method
SIP Registration/location to provide roaming/mobility

SIP Proxy
SIP Stateful Proxy
SIP Call Stateful Proxy
SIP Stateless Proxy
SIP Stateful/stateless proxies

SIP URI and DNS
Mapping E.164 dialled digits to SIP using DNS/ENUM
SIP and ENUM

The SIP Redirect Server
SIP REDIRECT methods
Using redirection to route calls
Using redirection to implement mobility
SIP redirection, proxies, registrars, and location services to provide mobility and roaming

Problems of SIP NAT Traversal
Common solutions to SIP NAT Traversal
Simple Traversal of UDP through NAT’s, STUN
Traversal using Relay NAT, TURN
Universal Plug and Play, UPnP
Tunnelling/VPN
Session Border Controller, SBC

SIP In the Cloud
SIP Gateways
SIP integration with the PSTN/ISDN
SIPI and SS7 ISUP
SIP and H.248/Megaco
SIP and H.225/H245/H.323
SIP and MPLS/QOS
SIP Trunking
Hosted SIP PBX
Hosted SIP Conferencing
Hosted SIP based contact centre

Potential threats to SIP
Registration Hijacking
Impersonating a Server
Tampering with Message Bodies
Tearing Down Sessions
Denial of Service and Amplification

Securing SIP
Transport and Network Layer Security
SIPS URI Scheme
SIP Method Authentication
Registration
Interdomain Requests
Peer-to-Peer Requests
DoS Protection
HTTP Digest
S/MIME
TLS

SIP Solutions and Capacity Planning
Design considerations when deploying SIP solutions
Feature requirements
– Matching customer requirement to cloud offering
– Identifying the Customer Premises Equipment (CPE) requirements
– Server implementation
Security and resilience of SIP servers
Security to the Cloud
Interconnection and communication with other severs such as DNS, and RADIUS/DIAMETER

Connecting to the Cloud
SIP based VOIP Key Performance Indicators (KPIs)
Sizing VOIP voice channel capacity
Impact of VOIP on data applications
Translating Erlangs and Grade of Service (GOS) into VOIP channel capacity
Impact of VOIP QOS on Grade of Service, GOS
Concept of Connection Admissions Control, CAC/VCAC
SIP and QOS
– Possible QOS signalling within SIP
– Enforcing the GOS from the SIP servers
Determining the bandwidth requirements for SIP signalling