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Voice over IP Essentials *** Duration: 2 days
Course Content
An in-depth examination of the methods and techniques used to
transport voice in real-time over networks. Technology issues at all the
protocol stack layers are dealt with including voice compression techniques,
network limitations, QoS transport protocols and call set-up and control. The
course concludes with a brief look at H.323, SIP and MEGACO and some examples of
VoIP implementations
Course Objectives
Voice over IP standards, regulations and structures
Understand how the requirements of voice impact on networks -
interoperability and integration
Recognise the potential of VoIP and the role played by SIP, H.323 and MEGACO
Appreciate the operation of all layers of the protocol stack in the
transmission of voice across data networks
Who Should Attend
Technical development and support professionals from carriers, ISPs, system
vendors, integrators, end-users and corporates
Software and hardware development engineers, technical marketing, developers
and planning professionals in communications applications and systems vendors
and carriers
Datacoms and telecoms managers
Prerequisites
Delegates should have knowledge of communications or standard data networking
with some knowledge of telephony concepts and TCP/IP
Course Outline
Introduction
Telephony Service and the Internet
Voice/data Convergence Infrastructure
Bandwidth Consolidation
VoIP protocols, standards and integration
Telecoms Review
- Voice and Telecom Circuits
- Switching
- Coders/Decoders
- Mean Opinion Scores
IP Attributes for VoIP Review
- IP Protocols
- TCP or UDP Transport
- Port Significance and application Interface
- Fragmentation and header compression
- Timing issues and Packet Switching
RTP/RTCP
- Features of RTP and IP Framework
- Encapsulation and Payload Mapping
- Header Fields
- Source Identifiers (SSRC/CSRC)
- RTCP Functions and Packet Types
QOS and Performance
- Grade of Service and Capacity Planning
- Queuing and buffering methods
- Differentiated Services (Forwarding)
- MPLS, IEEE 802.1p/Q, RSVP
VoIP Developments
- IETF - SIP, MEGACO;ITU-T - H.323
VoIP Call Establishment
- Establishing the requirements
- Registration, Capability, Call Setup, Control
H.323 v 3 with Version 4 Updates
- ITU-T H.323 Family and G Series
- Terminals as H.323 endstations
- Gatekeepers and Call Agent functionality
- Gateways and Protocol translation
- MCU and Conference capability
- H.225.0 RAS and Q.931
- H.245 and capability exchange
- Fast Connect
- Call/Session Flows Expounded
Session Initiation Protocol
- Server and Proxy Server
- Session Description Protocol
- Messages, Call Methods and Headers
- Status Codes and HTTP Compatibility
MGCP, MEGACO/H248
- Concepts and Gateway Decompression
Call Agents and Residential gateways
VoIP Application
Remote Access and Internet Telephony
PC Client to PSTN Telephone
PBX Call Routing and Branch Extension
Route Replacement
ISP Internet telephony Model
Convergence of Technologies
Copyright © 2001 Voice-Technologies.com
Voice-Technologies.com reserves the right to alter the
contents of this course as necessary.