SIP, the Session Initiation Protocol, is a signaling protocol for conferencing, telephony, presence, events notification and instant messaging.
It is an application-layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination of these. Session Initiation Protocol (SIP) builds on the IP communications foundation by providing a standards-based approach to enabling IP communications with numerous devices and applications.
SIP invitations used to create sessions carry session descriptions which allow participants to agree on a set of compatible media types. SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can register their current location. SIP is not tied to any particular conference control protocol. SIP is designed to be independent of the lower-layer transport protocol and can be extended with additional capabilities. Session controllers promise to enable the same ubiquity, quality, and security for VoIP that the PSTN offers today, only in the more flexible, efficient, and economical manner that IP makes possible.
The SIP fundamentals course provides an overview of SIP, its components, and how it works. It covers data networking principles to telco engineers and signaling principles to IP engineers. It also outlines SIP implementations on the market in the form of single-line gateways, proxy servers, media gateways, Java toolkits, encoders/decoders and session authenticators.
Objectives
After successfully completing the course the attendees will:
Understand basics of VoIP
Explore Where, why, and how SIP is used
Comprehend the basics of SIP
Understand the architect and components of SIP
Understand the differences between SIP and H.323
Understand H.323-SIP-SS7 Interworking
Review SIP-T concept and architecture
Understand how to size up and choose from available SIP products
Course Outline
Content
Executive Summary
Circuit-switched network signaling
Introduction to SS7
SS7 signaling
Operation of voice and data networks
VoIP Basics
IP Signaling protocols
Initiating, managing and terminating voice and video sessions
Set up, modify, and tear down multimedia sessions over the Internet
The session initiation protocol (SIP)
A new signaling protocol
Key services through the use of SIP
SIP capabilities
SS7-SIP interworking requirements
SIP Overview
Fundamentals of how SIP works
SIP context and architectures
SIP sessions
SIP flows
Core SIP
Encapsulation
Translation
SIP content negotiation
Session description protocol (SDP)
Security considerations
HTTP and SMTP, SIP
SIP extended features and services
Call control services, mobility, interoperability with existing telephony systems
Standardization status
Supported services
Proprietary extension and negotiation mechanisms
Interoperability of services and features
Interworking with PSTN
Service creation issues
Basic call features
Quality of Service issues
Network services
Conferencing and addressing
SIP, H.323, MGCP and Megaco
Basic SIP Communication Services
Integrating SIP with PSTN
SIP in IPv4 and IPv6
SIP and 3G Wireless
Session Initiation Protocol for Telephones (SIP-T)
SIP and SIGTRAN
SIP System Operations
SIP Parameters
Protocols
User Agents
Call Processors
Customer Status
Address Tracking
Call Forwarding
SIP Protocol Operation
Client/Server transactions
Proxy servers
SIP messages
Transport layer
Extending SIP
Extension negotiation
Technical details of SIP extensions
SIP Extensions
Session Description Protocol (SDP)
SDP packets
SIP timer
SIP programming
JAIN API
SIP Lite
SIP servlets
SIP for J2ME
SIP and SOAP
SIP and VoiceXML
SIP Entities
Components of SIP
SIP Clients SIP as a peer-to-peer protocol
User Agents (UAs) as the peers in a session
User agent client (UAC)
User agent server (UAS)
SIP Servers
Using A Proxy Server
Using a Redirect Server
Proxy Server
Redirect Server
Registrar
SIP Messages
Message Types
Message Parts
Message Samples
Requests
Responses
Header Fields
Bodies
Framing SIP Messages
Status Code Definitions
Informational 1xx
Successful 2xx
Redirection 3xx
Request Failure 4xx
Server Failure 5xx
Global Failures 6xx
SIP Parameters
Header Fields
Option Tags
Warning Codes (warn-codes)
Methods and Response Codes
Reason Protocols
Security Mechanism Names
Compression Schemes
SIP vs. H.323
Robustness
Security
Legacy
Political Issues
Status Update
References
SIP-T
SIP-T for ISUP-SIP interconnections
SIP-T flows
SIP bridging (PSTN - IP - PSTN)
PSTN origination - IP termination
IP origination - PSTN termination
SIP-T roles and behavior
Components of the SIP-T protocol
Support for mid-call signaling
SIP Signaling Flows in 3G/UMTS
UMTS core network architecture
Call management in UMTS R4/R5
Soft handover
Hard handover
Registration
Session initiation
Session termination
Roaming scenarios
SIP Products and Trends
Application Servers
Applications
Session Border Controllers
SIP based Services
SIP Gateways
SIP Hardware Appliances
SIP Phones
SIP Presence and Messaging Servers
Software Development Kits
Testing and Simulation
Case Study: Cisco
SIP Messages and Compliance Information the Cisco VoIP Infrastructure Solution for SIP
SIP Messages and Methods
Requests
Responses
The Registration Process
The Invitation Process
SIP Compliance Information
PSTN Cause Code and SIP Event Mappings
Call Flow Scenarios for Successful Calls
SIP Gateway-to-SIP Gateway
Call Setup and Disconnect
Call via SIP Redirect Server
Call via SIP Proxy Server
Call Setup with Delayed Media via Third-Party Call Controller
Individuals who wish to develop a basic knowledge of SIP. Essential course for anyone involved in the development and operation of voice or data networks, wireless communications protocol, mobility technologies, and instant messaging.
Prerequisites
This is an introductory course with no prerequisites.
