Training Course

VoIP and Unified Communications (UC)

  • 328
  • £2,085 +VAT
  • 3 days
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Why this VoIP course?

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.

VoIP Training Course Objectives

On completing this course, you will be able to:

  • 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.

Who should attend this VoIP course

  • 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.

VoIP Course Pre-requisites

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

Hands-on Practicals

This VoIP course incorporates extensive hands-on practical VoIP labs, with some 30 lab sections and demonstrations using a selection of current VoIP products including SIP and H.323 offerings. Hands-on practical voice over IP labs include:

  • Configuring TCP/IP on Microsoft Windows systems.
  • Testing IP operation.
  • Setting up a protocol analyser; performing packet capture and analysis.
  • Analysing IP, TCP, UDP.
  • Assessing voice quality from coded voice samples.
  • Conducting Voice over IP calls using different voice codecs.
  • Conducting voice calls under simulated adverse network conditions.
  • Identifying the VoIP protocols captured during VoIP conversations.
  • Observing the Layer 2, Layer 3, Layer 4 and Layer 7 interactions.
  • Assessing the effects of packet delay and packet loss on call quality.
  • Analysing RTP and encoded media streams.
  • Analysing RTCP reports.
  • Setting up a SIP Proxy; conducting SIP-Proxy controlled VoIP calls.
  • Implementing SIP telephony using SIP clients and SIP registrar, proxy, redirect and location servers.
  • Configuring SIP hardphones and SIP softphones.
  • Observing and capturing SIP interactions and call sequences.
  • Analysing SIP and SDP signalling.
  • Employing a SIP Media Gateway.
  • SIP – SIP and SIP – PSTN calls.
  • Setting up a market-leading SIP PBX.
  • Creating a dial plan; conducting PBX-based calls.
  • Working with Trixbox (a business class IP PBX system based on the Asterisk Open Source PBX software).
  • Classifying traffic with IP routers.
  • Configuring Voice over IP products to use the Diff-Serv Code Point.

Lifetime Post-Course Support

After completing this VoIP training course, delegates receive lifetime post-training support from LEVER Technology Group, to help them apply the technologies and skills they have learned with us, to provide career-long support, and to ensure they are better equipped for their future roles in IT and networking.

VoIP Training Course Content

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 versus 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, 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
  • 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
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