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 must have a good understanding of the TCP/IP protocol suite and of IP addressing prior to attending. Course 315: Internetworking with TCP/IP, is recommended. DURATION:   3 days. OBJECTIVES:   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. Internet service providers are now competing head-on with traditional Telcos by offering Internet telephony and video conferencing services. Voice over IP not only offers substantial cost savings, but also allows the development of new applications that can dramatically enhance customer service and organisational efficiency. COURSE 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 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 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