Introduction:

Session Initiation Protocol (SIP) is the signalling protocol that forms the basis of future Voice over IP (VoIP) and converged networks, both fixed and mobile. This two day course describes SIP and its operation in detail. Attendees will gain and understanding of the architecture, components and functions of the SIP suite. This SIP course explains SIP's strengths, reliability, performance, separation, extensibility, modularity and simplicity. Delegates will also understand the role of SIP within the wider context of signalling, content and communications protocols, and in Fixed and Mobile Convergence (FMC).

Audience:

SIP courses are designed for technical planners, developers, network managers and everyone involved in IP Multimedia, VoIP and Convergence from Service Providers, Enterprise networks and Telecommunications, who will find this course invaluable. Engineers and technical professionals from fixed and mobile telecoms operators, ISPs, system vendors and integrators, and pre- / post-sales and support engineers in vendor organisations should also attend. End-users developing or deploying Voice over IP (VoIP), or other real-time services over IP networks, will also find this course relevant to their projects.

Objectives:

This two day course describes SIP and its operation in detail. Attendees will gain and understanding of the architecture, components and functions of the SIP suite.

Course in a Nutshell:

Introduction to SIP What is SIP and why use it? SIP attributes and efficient call signalling protocols SIP Development Proposals, development and emerging standards Properties and Operational Overview Key features and extensions Components and relative functions Call Flows based on methods Addresses and URL flexibility Syntax and Message Programming Text based message structure Servers, clients, requests and responses 3-digit HTTP Code and SIP Extensions Session Description Protocol (SDP) Protocols overview SDP syntax and message structure Relationship to SIP SIP Server Functions Registration Home and away scenarios Location and Directory Services Stateful and stateless servers Call Establishment Connection properties Session initiation and server location Outbound Proxy INVITE method Tagging Key header fields Call Control Redirect options Transfers Conferences SIP services Security Considerations Encryption and Keying Access Control and Authentication RTP and RTCP (Optional module) RTP header, features and packet types Encapsulation, timing, and header compression RTCP functions and packet types RTP performance statistics SIP and Applications SIP in 3rd Generation (3G) mobile networks Characteristics of SIP and H.323 Scalability and services Comparison of SIP and MGCP/MEGACO (H.248) **Fundamental benefits of SIP