Introduction This 4-day course offers a practical introduction to 'hands on' VoIP engineering. Voice over IP promises to reduce your telephony costs and provides unique opportunities for integrating voice and data. The past and the future of telecommunications come together in VoIP, and to deploy this technology successfully you need to understand its benefits, risks and challenges. Objectives When you have completed this course you will be able to: Enter into discussion of products, services and technology that use VoIP Plan and size a VoIP solution Setup a PC VoIP Application Setup a SIP VoIP Application using analogue phones and SIP converters Describe in overview the use of layered protocols for packet services Capture and analyse VoIP traffic using a Protocol Analyser Analyse and troubleshoot Quality of Service over a VoIP Network Distinguish the important Internet protocols Compare and contrast the ITU H.323 and the IETF SIP approaches to VoIP Discuss the Cisco approach to VoIP Compare the business case for using VoIP and for Circuit Switching carriage of voice Prerequisites: The course will be aimed at technicians and engineers assuming a knowledge of Microsoft Windows and simple TCP/IP networking using PCs and elementary telecommunications. It will not assume any prior knowledge of VoIP or packet voice technology. Content Introduction and Background Evolution of Telecommunications - Circuit Switched voice - Packet Switching Data Motivation: Why use VOIP - Comparison between current voice and data networks One Integrated Network - Sharing resources - Migration Where VOIP can be deployed - Integration at the PBX - Integration at the PC - Integration at the desk with IP phones Which IP Network - Internet Telephony - VOIP over an Intranet - Internet Telephony Service Providers The Business Case - Cost per minute savings V3.1 2009 ProTel Solutions Page 1
- Improved Call Center Integration - e-commerce with Voice Enabled Websites Hands On Session 1: Here is Voice Over IP Working 1. Set up and use VoIP applications on each PC to place calls across the classroom 2. Set up a Protocol analyser on each PC and capture VoIP traffic 3. Experience Different IP environments and observe VoIP performance issues Internet Protocol Suite Fundamentals Sources for Protocols: ITU and IETF - ITU and its standards -Purpose of H, Q, G, I, E, X, V, T standards - IETF RFCs Protocol Structures - OSI Layers - Functions of Layer 2, 3 4 and 7 - Internet Application model - MTU and its impact on VoIP Layer 2 Frame level services - MAC level performance and capacity issues - IEEE 802.3 - IEEE 802.1P priority services - IEEE 802.1Q VLAN services What is IP - Datagrams - Routing - Routing Tables - Routing protocols; RIP, OSPF, BGP4 - Classification of Routers, Switches and Hosts Carriage of IP over LANs, Frame Relay, xdsl, Broadband Access, SONET/SDH - Ethernet and PPP - Ipv4 and Ipv6 - TCP and UDP - RTP and RTCP - Application Examples - Web Services - Streaming - FTP Hands-on Session 2: Taking apart VoIP 1. Identify the VoIP protocols captured during VoIP conversations 2. Observe the Layer 2, Layer 3 and Layer 4 Interactions 3. Observe Packet and frame count information available from routers and switches 4. Use protocol analysers to provide packet counts on LANs Telecommunications Fundamentals Principles of Circuit Switching - Digital voice circuits - CODECs - G.711 calls Switching Capacity V3.1 2009 ProTel Solutions Page 2
- Sizing a network or switch using Erlangs - Blocking and non-blocking services Connecting a Call In ISDN - Call Map Access signaling - Q.931 Signaling messages VoIP Architectures Source of VoIP standards - ITU and H323 - IETF and SIP Multimedia conferences over packet network - What counts as Multimedia - Voice - Video - Conference - Sources and mixes How does a normal phone call get connected - Call Map - Conversion to digital - Dialing and Signaling - Alerting and Call Progress Tones VoIP using H.323 How most Cisco VoIP networks function H.323 Components - Map of H323 Components - Gateway (GW) - MCU - Multi-point Controllers (MC) - Multi-point Processors (MP) - Gatekeepers (GK) and Call Managers - Address Mapping - H.225 - H.245 Capability Exchange - Negotiating codec - Negotiating Full Association Encoding the content - Codecs - RTP and RTCP Making an H323 Call Possible Configurations - Desktop to desktop - Desktop to phone - Phone to desktop - Phone to phone - Conference via a MC How does this fit into an organization - Carrier Bypass and Service Replacement Desktop integration - Attachment to the GSTN V3.1 2009 ProTel Solutions Page 3
- Carrier Network Replacement.Admission Control - Functions that VOIP signaling must perform - Address mapping Hands-on Session 3: H.323 Gatekeeper Managed Services 1. Setup a Gatekeeper managed service 2. Observe H.323 gatekeeper registrations and call connections 3. Observe Network Performance Using Netmeter 4. Setup and use an MCU and observe its performance VoIP using IETF Architecture SIP Why has SIP become important? SIP Components - SIP Addressing Connection signaling Capabilities exchange SIP Message Format Comparing SIP and H.323 Media Gateway Control Protocol (MGCP) and MEGACO H.248 - Protocol Architecture - Building Blocks - Topology Descriptors - Topology Changes - MEGACO Phone Profile H.323 and SIP compared Evolution of H.323 - Comparison with SIP Hands-on Session 4: SIP Proxy Controlled Services 1. Setup a SIP Proxy controlled VoIP service 2. Configure a SIP application 3. Observe and capture SIP Registrar interactions and call connections 4. Observe Network Performance Using Netmeter Quality of the Voice What Constitutes Quality - Delay - Availability - Understanding the speech - Recognizing the person speaking Quality Measures - Mean end to end delay - Mean up time - Mean Opinion Scores Codecs - Companded PCM - ADPCM - CELP - G.711,G.726, G.728, G.729, G.723.1 Hands-on Session 5: Observing Voice Quality Issues 1. Use different codecs to compare performance 2. Learn to identify the impact of delay and packet loss on VoIP services V3.1 2009 ProTel Solutions Page 4
VoIP and Circuit Switched Performance Issues Elements that affect voice quality: Delay and Loss - Packetization - Serialization - Propagation - Queuing Calculating Queue Delays - M/M/1 Queues Calculating Availability - Serial availability - Parallel availability Comparing VoIP with other solutions What Traditional Circuit Switching Brings to the Party - Reliability - Guaranteed Service Quality - Guaranteed Utilization - Relatively high bandwidth Demands What Packet Switching Promises - Lower bandwidth Demands - Statistical Service guarantees - Flexibility of service limits - Trade-offs between delay, loss, quality and price - Carriers and ITSPs Security Implications - Impact of VOIP on the firewall - Securing the VOIP signaling Delivering QOS with VoIP - Mixing Voice and Data - RSVP - Diffserv - Weighted Fair Queues Hands-on Session 6: Quality of service Planning 1. Size a VoIP service 2. Predict Delay and QOS performance 3. Mix voice and Data over a low speed Router-Router Link 4. Deliver QOS in practice Comparing VoIP against Circuit Switching - Bandwidth use - Flexibility of service - Integration with data services E-commerce - Voice enabled Web sites - Service guarantees - Capacity availability - Integration with telephony systems VoIP Over Broadband Access - Typical E-Commerce Solutions Commercial Alternatives and competitors to VoIP - VoFR - VoATM V3.1 2009 ProTel Solutions Page 5
Course Length 4 days V3.1 2009 ProTel Solutions Page 6