Indepth Voice over IP and SIP Networking Course

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1 Introduction SIP is fast becoming the Voice over IP protocol of choice. During this 3-day course delegates will examine SIP technology and architecture and learn how a functioning VoIP service can be established. Delegates will be able to take part in demonstrations of working SIP calls and see captured SIP traffic over a classroom network. They will also be able to connect their own Laptop or classroom PCs to the network and learn to capture traffic and examine sizing networks using spreadsheets. Prerequisites The course assumes a basic understanding of PCs, Windows, simple TCP/IP networking using PCs and elementary telecommunications. It does not assume any prior knowledge of VoIP or packet voice technology 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 SIP based VoIP solution Describe in overview the use of layered protocols for packet services Distinguish the important Internet protocols Compare and contrast IETF SIP with other approaches to VoIP Compare the business case for using VoIP and for Circuit Switching carriage of voice Who Should Attend IP engineers and project managers involved in VoIP projects. V ProTel Solutions Page 1

2 Course Content Voice over IP VoIP - How it works VoIP Encapsulation VoIP - Associated Internet Protocols VoIP - Service Providers Networks The Internet IP Networks Why VoIP What does VoIP Offer? Internet Telephony Product Classes Carrier Class Enterprise Class Single Clients Internet Telephony IPT Class 1 IPT Class 2 IPT Class 3 VoIP Regulatory Bodies Voice Encoding Schemes Waveform Encoding Digital Recording and Playback PAM - Pulse Amplitude Modulation Quantization Vocoders MOS - Mean Opinion Score Voice Quality Measurement Voice Quality Issues Latency Jitter Silence Suppression Packet Loss Out of Sequence Packets Delayed Packets Duplicate Packets Echo Problems in VoIP Acoustic Echo Hybrid Echo Echo Suppression Echo Cancellation Protocols - Transport and Network Layers Existing Transport and Network Layer Protocols Internet Protocol Transmission Control Protocol User Datagram Protocol V ProTel Solutions Page 2

3 The Impact of Routing Protocols Real-time Protocols ITU-T H.323 Protocol Stack RTP Real-Time Transport Protocol Encapsulation Overhead RTP Translators Audio Conferencing (Mixers) RTCP Real-Time Control Protocol RTCP Bandwidth Control RTCP Packet Header 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 VoIP Architectures Source of VoIP standards ITU H323 and IETF SIP evolution compared Multimedia conference 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 Carrying voice over IP Encoding voice using codecs Preserving timing Impact of Jitter Removing Jitter RTP V ProTel Solutions Page 3

4 RTCP How does A VoIP call get connected Separation of signalling and media streams TCP/UDP Port numbers Impact of packet loss on signalling VoIP using IETF Architecture SIP Why has SIP become important? SIP Components SIP Addressing the URI Connection signaling Signaling when the phone rings Capabilities exchange Closing calls SIP Message Format SIP Registration Services Identification of users and phones SIP Registrar Mapping Identities SIP Proxy Services Redirection Impact of NAT SIP Location Services Forwarding registrations Canceling Forwards Security in SIP exchanges SIPS Authentication Encryption Quality of the Voice What Constitutes Quality Delay and Availability Understanding the speech and recognizing the person speaking Quality Measures Mean end to end delay Mean Opinion Scores Codecs Companded PCM ADPCM CELP G.711,G.726, G.728, G.729, G Delivering QOS with VoIP Mixing Voice and Data Congestion Control Leaky Bucket Algorithm Token Bucket Algorithm Quality of Service Models Integrated Services (IntServ) Differentiated Services (DiffServ) V ProTel Solutions Page 4

5 Congestion Management FIFO - First In First Out PQ - Priority Queuing CQ - Custom Queuing WFQ - Weighted Fair Queuing LLQ - Low Latency Queuing MPLS - Multi Protocol Labelled Switching RSVP Diffserv Weighted Fair Queues Sizing VoIP Services The following practical exercises will be completed during the course: 1. Make a SIP Call using SIP Softphone software provided and observe results using a protocol analyzer. 2. Capture the sigalling from SIP calls using a protocol analyzer. 3. Setup a VoIP service using a SIP Proxy Server 4. Setup a SIP application 5. Size a VoIP service Mix voice and data over a low-speed Router link and adjust QoS parameters. Course Length 4 days V ProTel Solutions Page 5