Course benefits
Voice over IP (VoIP) is now a major growth sector within IT and telecommunications. VoIP is revolutionising both industries, and is driving organisations to unify their voice and data networks.
This unique, authoritative and highly practical VoIP training course from the UK's leading telecoms training company, provides an in-depth coverage of H.323, SIP, SIP-T, SIP-I, H.248, IMS and related protocols, codecs and technologies.
VoIP and convergence training courses can be delivered at our training centres World-wide. Hands-on, on-site VoIP courses can be delivered in a fully-tailored format at your location.
Why this course?
This unique course from LEVER Technology Group PLC has been developed and proven over a seven year period to provide what we believe is the best training available in the UK on the latest VoIP technologies, the three latest competing areas of standards development, the markets, the key vendors and their products. The course also covers IP Quality of Service issues and provides an awareness of the weaknesses inherent in current VoIP offerings.
Since 1999, LEVER has delivered industry-leading Voice over IP training for the UK's leading Telcos, mobile network operators, Times 100 companies and the Military.
Our VoIP course features substantial hands-on practical with products from Cisco, Avaya, Microsoft and other leading VoIP vendors.
In-class labs
This course incorporates extensive hands-on practical labs and demonstrations using a selection of current VoIP products, including H.323, SIP and MGCP offerings.
Hands-on practical labs include:
- Configuring TCP/IP on Microsoft Windows systems
- Testing IP operation
- Packet capture and analysis
- Analysing IP, ARP, TCP, UDP
- Assessing voice quality from coded voice samples
- Conducting Voice over IP calls using different voice codecs
- Comparing voice codecs
- Conducting voice calls under simulated adverse network conditions
- Assessing the effects of packet delay and packet loss
- Analysing RTP and encoded media streams
- Analysing RTCP reports
- Implementing H.323 telephony using H.323 terminals, and H.323 Media Gateways
- Configuring H.323 terminals and Media Gateways
- Analysing H.323 signalling protocols through complete call sequences
- H.323 Gatekeeper operation
- Configuring H.323 Gatekeeper routed calls
- Analysing RAS protocol exchanges
- Employing an H.323 MCU
- Implementing SIP telephony using SIP clients and SIP servers
- SIP - SIP and SIP - PSTN calls
- Analysing SIP and SDP signalling
- Employing a SIP Media Gateway
- Classifying traffic with IP routers
- Configuring Voice over IP products to use the Diff-Serv Code Point
- Implementing RSVP on IP routers
- Configuring Voice over IP products to use RSVP
- Analysing RSVP traffic
- Implementing packet scheduling
- Implementing CBQ
The course also incorporates a number of in-class IP Telephony solutions design exercises.
Who should attend?
The Voice over IP course is designed for staff who will evaluate, plan, install, configure, administrate or support VoIP products and networks.
Pre-requisites
Delegates must have a good understanding of the TCP/IP protocol suite and of IP addressing prior to attending. Course 313: TCP/IP: Protocols, Implementation, Analysis, Troubleshooting and Support, is recommended.
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Course 328: Content
Review of the TCP/IP protocol suite
What is TCP/IP?
The evolution of TCP/IP
Essentials of IP addressing
The US DoD / DARPA architectural model
The Microsoft TCP/IP architecture
Application layer protocols
Internet Protocol (IP)
IP addressing
Address classes A, B, C, D
IP Multicasting
IP Datagram format
Transmission Control Protocol (TCP)
TCP Segment format
User Datagram Protocol (UDP)
UDP Datagram format
Ports and sockets
ICMP, ARP, DNS, HTTP
Overview of VoIP
Why VoIP?
Markets for VoIP
Who uses VoIP?
Evolution of the PC Phone
Components of a VoIP system
Architectural overview: Terminals, Gateways, Gatekeeper, MCUs
Status of VoIP as an emerging technology
Overview of VoIP
Why operate voice over data networks?
Voice and data convergence
Why Voice over IP?
Applications for VoIP
Markets for VoIP
Who uses VoIP?
Evolution of the PC Phone
Components of a VoIP system
Architectural overview: Terminals, Gateways, Gatekeeper, MCUs
Standards employed in current VoIP solutions
The role of Voice Processing.
Status of VoIP as an emerging technology
Review of Voice Telephony
Telephone system components
Telephone signaling
The Local loop
Voice switches
Echo in telephone networks
Analogue and Digital signals
Traditional digital voice transmission and switching sytems
Multiplexing techniques
The speech encoding process
Sampling, Quantisation, Coding, Framing
Silence suppression
Voice coding and compression standards
Adaptive encoding techniques
Coding fax signals
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
PSQM, PAMS, PESQ
Example voice quality testers for VoIP networks
Overview of H.26x video codecs
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)
Implementing centralised number and dial plans
Mapping E.164 addresses to IP addresses
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 / SDP
MGCP and Megaco/H.248
ITU H.32x series standards
ITU H.323 and related standards
H.320 / H.324 / H.323
H.323 Voice over IP protocol stack
H.225 (Q.931) in detail
Registration, Admission and Status (RAS)
Endpoint registration
H.225 and RAS signals in detail
H.245 in detail
Selecting capabilities with H.245 Call Control
H.323 call setup
Call termination and clearing
H.323 v2, v3 and v4
Fast Connect Procedure
H.235 security extensions
Related H.2xx standards
Related T.xxx series standards
The IETF Framework for VoIP
IETF initiatives and working groups for Voice over IP
The relevant IETF standards
Session Initiation Protocol (SIP)
SIP-T and SIP-I
SIP server roles
SIP call setup procedures in detail
SIP protocol header
Session Description Protocol (SDP)
Understanding SDP coding
Transport of telephony signalling over IP networks (sigtran)
Interworking SIP and H.323
Other Internet multimedia conferencing over IP protocols:
Session Announcement Protocol (SAP)
Simple Conference Control Protocol (SCCP)
Real-Time Streaming Protocol (RTSP)
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ITU / IETF Megaco / H.248
MGCP and Megaco
Media Gateway Control Protocol (MGCP)
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
The role of Call Processing Language (CPL)
Current status: H.323 vs SIP vs Megaco
Understanding 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 Integrated Services (Int-Serv)
ISSLL WG
Implementing Int-Serv on IP routers
The problems with Int-Serv
IP Differentiated Services (Diff-Serv)
The Diff-Serv framework
Implementing Diff-Serv
Packet classification for Diff-Serv
Interworking Int-Serv and Diff-Serv
Common Open Policy Service (COPS)
Resource ReserVation Protocol (RSVP)
Resource ReserVation Protocol (RSVP)
Characteristics of RSVP
RSVP simplex operation
RSVP in IP multicast environments
RSVP message propagation
RSVP traffic control modules
Reviewing RSVP operation in detail
Combining RSVP with Weighted Fair Queuing
Implementing IP QoS for Voice over IP
Providing Quality of Service (QoS) over an IP network
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
The Label concept
MPLS terminology
How MPLS works
MPLS over ATM, SDH and Ethernet networks
Employing MPLS for IP network engineering
Evaluation of Leading Voice over IP Solutions
Classifying Voice over IP products
Mainstream VoIP manufacturers
Mainstream VoIP Gateways and PBXs
Mainstream IP-based PBXs
H.323 Gatekeepers
Admission control
IP Telephony Call Agents and Softswitches
IP Soft Phones
PC IP Telephony Cards
Enhancements for security firewalls
VoIP test equipment
VoIP product interoperability
IMTC iNOW! Compatibility Program
VoIP network and product testing
VoIP products from leading vendors, including Cisco, Nortel, 3Com, Avaya, Mitel, NEC, Siemens, Clarent and others
Planning and design of a VoIP solution
Cost-benefits analysis for Voice over IP
Trunking voice over IP
Replacing existing PABXs
New installations
Advanced and Call Centre installations
Conducting a six-phase feasibility and cost-benefits analysis for Voice over IP
Six-phase planing and design methodology
Determining bandwidth requirements for Voice over IP
Understanding traffic engineering
Combining voice and data traffic effectively
Calculating queuing delays
Sizing link capacity needs and required trunk speeds
Calculating routing delays
Designing for service availability
Prediction of voice quality
Operating VoIP through security firewalls
Operating VoIP with Network Address Translation (NAT)
Interoperating with external networks
VoIP clearinghouses
Open Settlement Protocol (OSP)
In-class case study
Future developments in VoIP
Dedicated VoIP network segments
IPv6 and VoIP
ATM and Frame Relay versus VoIP
VoiceXML
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