Course 624: SIP Training Course - Session Initiation Protocol (SIP) - TONEX.com

TONEX SIP training 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.

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.

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

Content

Executive Summary

• 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 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

• Differences and Similarities
• Robustness
• Security
• Legacy
• Political Issues
• Status Update
• References

SIP Signaling Flows in HSPA/HSPA+/LTE

• UMTS/LTE core network architecture
• Call management in IMS
• 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

Product overview: Cisco and Avaya

• 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
• Call Setup using a FQDN and Delayed Media
• Redirection with CC-Diversion
• SIP 3xx Redirection Response
• Successful Call Setup and Disconnect
• SIP Gateway-to-SIP IP Phone
• Successful Call Setup and Call Hold
• Successful Call Setup and Transfer
• Call Setup with Voice Mail
• Automatic Route Selection
• SIP IP Phone-to-SIP Gateway
• Call Setup and Call Hold with Delayed Media
• SIP IP Phone-to-SIP IP Phone
• Simple Call Hold
• Call Hold with Consultation
• Call Waiting
• Call Transfer without Consultation
• Call Transfer with Consultation
• Network Call Forwarding (Unconditional)
• Network Call Forwarding (Busy)