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VoIP
 
   
Course Name: VoIP and Unified Communications (UC) Training
   
Deployment Options: Onsite - Instructor-Led Training
   
Course Duration: 3-4 days depending on audience background and options
   
Related Courses
 
 
Introduction:
 

This unique VoIP course has been developed, improved and proven over a continuous twenty year period of training delivery.

  • Provides the best VoIP training available in the UK, covering the latest VoIP technologies.
  • Covers the three competing areas of standards development: SIP, H.323 and H.248/Megaco.
  • Examines in-depth voice coding, media coding, call signalling, with hands-on protocol analysis.
  • Provides extensive hands-on practical experience.
  • Covers IP Quality of Service issues.
  • Covers the markets, the key vendors and their products.
  • Examines the relative merits of a selection of current VoIP offerings
 
Audience:
 
The Voice over IP course is designed for staff who will evaluate, plan, install, configure, administrate or support VoIP products and networks
 
Prerequisites:
 
Delegates should ideally have a good understanding of the TCP/IP protocol suite and IP addressing prior to attending this course.
 
Customize it:
 
This 3-4-day VoIP and Unified Communications (UC) course will be customized to your needs and specifications. Eno.com will assist you in identifying those needs and specifications. A word to the wise, there are many vendors of telecommunication training. They will typically have a broad and general course, one size fits all, already developed and just put your organization?s name on the title slide. This minimizes their effort and time investment. At Eno.com, every course is made to your exact and exacting specifications. We help you ensure what you are getting is what you really need even if at the beginning you weren't too sure of what that was. We fit the class to your needs. We never fit you into our standard, one size fits all, class. Please call or e-mail to schedule a no-obligation conference call to help us understand your audience background and training objectives.
 
Objectives:
 
On completing this course, you will be able to:
 
Course Outline
 

Review of the TCP/IP protocol suite

  • The evolution of TCP/IP
  • Essentials of IP addressing
  • The US DoD / DARPA architectural model
  • 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
  • Understanding ICMP, ARP, DNS, HTTP

Overview of VoIP

  • Voice and data convergence
  • Status of VoIP as an emerging technology
  • Applications for VoIP
  • Markets for VoIP
  • Components of a VoIP system
  • H.323 Architectural overview: Terminals, Gateways, Gatekeeper, MCUs
  • Standards employed in current VoIP solutions
  • The role of Voice Processing

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
  • 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
  • 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
  • Cisco SCCP (Skinny)

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 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)

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
  • 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)
  • Resource ReserVation Protocol (RSVP)
  • 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

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
  • VoIP network and product testing
  • VoIP products from leading vendors, including Cisco, Nortel, 3Com, Avaya, Mitel, NEC, Siemens 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
  • In-class case study

Future developments in VoIP

  • Dedicated VoIP network segments
  • IPv6 and VoIP
  • VoiceXML
 
Other Expertise:
 
 

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