07 February - 09 February 2007 (Wed - Fri)
3-day Instructor-Led, 0900 - 1730
Singapore City, Singapore

After years of research, development and implementation of several approaches, the Internet is becoming the only network providing universal connectivity and offering all kind of services. Among the newest services that users are expecting from the Internet we can find all multimedia based applications, mainly IP Telephony.

SIP is the IETF proposal for signaling telephony and other multimedia calls, in order to establish, manage and release real-time sessions in IP based networks. Including signaling not only for basic services, but for supplementary services as well.

In this seminar you will learn the basics of SIP and mostly related to IP Telephony. You will get an understanding of the architecture, components and functions of the SIP suite and at the same time you will confirm its simplicity, modularity and performance by comparing it to other existing solutions.

Attendees must have a good understanding of IP and related protocols and a basic knowledge on telephony networks.

i.Network Engineers, Network Operators, Network Managers or Network Designers who are responsible for designing, maintaining and operating IP services on public or private IP networks

ii.Managers evaluating telecommunications related investments and the application of SIP for their end-user organizations

iii.Pre- and post sales and support engineers in vendor organizations

i.Introduction i.PSTN vs IP Networks ii.What is IP Telephony iii.VOIP principles iv.VOIP evolution v.VOIP Benefits vi.SIP principles vii.Standardization bodies viii.IETF’s and other organizations standards ii.Media Transport i.Real Time Transport Protocol (RTP) ii.Header format and functions iii.Real Time Control Protocol (RTCP) iv.Functions and packet types v.Compressed RTP vi.DTMF in RTP iii.SIP architecture i.Definition ii.SIP Components: iii.User Agent Clients iv.Servers i.Proxy server ii.Registrar iii.Redirect server iv.Location server iv.SDP i.Protocol overview ii.Syntax and message structure v.SIP basic methods i.SIP methods ii.SIP messages and headers iii.SIP status and responses iv.SIP URL vi.Basic call flows using SIP i.SIP modes ii.Direct end to end mode iii.Proxy mode iv.Redirect mode v.Call establishment vi.Call teardown vii.SIP new methods and services i.SUBSCRIBE ii.NOTIFY iii.REFER iv.Presence and Instant Messaging v.Supplementary services (Call Park, Call Forward) vi.Call Transfer using REFER vii.Voice Mail viii.Unified Messaging viii.SIP conference call i.SIP Conference server ii.Call establishment and tear down iii.Subscriber joins a conference iv.Subscriber leaves a conference ix.SIP interworking i.SIP – PSTN i.SIP-T ii.Sigtran ii.SIP and other IP based solutions i.SIP vs. H.323 ii.SIP vs. MGCP/Megaco x.QOS and SIP i.Reliability – problems and solutions ii.The PRACK method iii.The COMET method xi.Security in SIP i.Authentication and authorizations methods in SIP ii.SIP support for Firewall iii.Application Layer Gateway iv.SIP and NAT v.The problem vi.STUN (RFC 3489) – the solution vii.TURN and other solutions xii.3G Mobile and SIP i.SIP in 3GPP ii.SIP in IMS architecture iii.SIP in MMD architecture