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SIP and the IMS

Powerful 5-Day Hands on Course

Course Description

This course is a detailed review of the SIP protocol. The emphasis is on the protocol itself, not any specific application of SIP. Hands on training is used at least 50% of the course. This course is very popular and the course reviews are "raving".

Intended Audience

This course is intended for technical personnel that need a vendor neutral understanding of the SIP protocol.

 

Prerequisites

Student prerequisites are as follows: TCP/IP, VoIP, and Gatekeeper knowledge and understanding.

 

Course Outline

  1. Why does IMS use SIP?
    • Circuit vs. packet concepts
      • Circuit-based bearer channel vs. Packet-based bearer channel (DS0 vs. RTP)
      • SS7/ISDN vs. SIP, MEGACO
    • The IMS universal network design goal – build one horizontal network that can support all mobile services from one network
    • VoIP Protocols comparison
    • Why SIP?
    • A short history of SIP
  2. IP Routing and Switching Overview
    • IP Routing
    • Ethernet Essentials
  3. TCP and UDP Essentials
    • How VoIP uses TCP
    • How VoIP uses UDP
  4. IMS Architecture
    • IMS and SIP
    • RTP Introduction
    • 3GPP/3GPP2, IETF, OMA (Open Mobile Alliance)
    • Applications: Push to Talk, Instant Messaging, Presence, Telephony, Gaming
    • Proxy-CSCF
    • Serving-CSCF
    • Interrogating-CSCF
    • Policy Decision Function
    • HSS
  5. SIP Architecture
    • The SIP architecture
      • UA, Proxy, Redirect, Forking,
    • Multimedia Architecture
      • RTP/RTCP
      • SDP
    • Methods
      • REGISTER
      • INVITE and ACK
      • UPDATE
      • OPTIONS
      • CANCEL
      • REFER
      • SUBSCRIBE and NOTIFY
      • MESSAGE
      • BYE
    • SIP responses
      • 1xx   Informational
      • 2xx   Final
      • 3xx   Redirection
      • 4xx   Client Error
      • 5xx   Server Error
      • 6xx   Global Failure
  6. SIP Uniform Resource Indicators (URIs)
    • Generic URI information (RFC 2396)
    • Direct or Proxy
    • PSTN number (RFC 2808)
    • Instant messaging
    • Presence
    • In registrations
  7. SIP Headers
    • Via
    • Branch
    • Max-Forwards
    • Dialog (To, From, and tag= fields)
    • CSeq
    • Call-ID
    • Contact
    • SIP reliability
    • Expires
    • Authentication
    • IMS Signaling compression (SigComp architecture)
  8. Session Description Protocol (SDP)
    • Learn how SIP uses SDP to define technical parameters that support a voice over IP media channel.
    • Session parameters
    • SDP format
    • Extending SDP
    • SDPng
    • Media negotiation
    • Changing session parameters
  9. IMS Related IP Services
    • DHCP
    • ENUM
    • DNS NAPTR records
    • DNS SRV records
    • Regular Expressions
    • DDDS algorithm
  10. SIP Call Flow Examples
    • Call attempt – unsuccessful
    • Presence subscription
    • Registration
    • Registration with Authentication
    • Presence notification
    • Instant Message Exchange
    • Call setup – successful
    • Call hold
    • Call Forward
    • Call transfer
    • Unified messaging
  11. IMS Reference Points
    • Gm, Mw, Cx, Dx, Sh, Si, Dh, Mm, Mg, Mi, Mj, Mk, Ut, Mr, Mp, Go, Gq
  12. IMS Call Processing  
    • Registration
    • Service discovery
    • Identity modules
    • IMS Authentication and key agreement (User Identities)
    • Network domain security
    • Secure HTTP-based service
    • Controlling bearer traffic
    • Controlling the Media
    • Anonymous calling (Hide Caller-ID)
    • S-CSCF assignment processes
    • MEGACO and SIP controlled PSTN connectivity
    • Creation of via-path for response routing
    • Response merging
    • Record route
    • Control models
    • Third party call control
    • Conferencing (REFER)
    • Access and location information
      • P-Access
      • P-Visited
    • IMS Messaging (Immediate, Session-based, Deferred)
  13. RTP and RTCP (Real-Time Control Protocol)
    • Dealing Packet Loss, Latency, Jitter
    • How RTP defines the session
      • Session Description Protocol
      • The RTP profile
      • The RTP payload type field
      • RTP telephony events (RFC 2833)
    • How RTP removes jitter
    • How RTP handles packet loss
    • How RTP identifies the talking party
    • How RTP handles silence suppression
    • How RTP handles fixed length packets (padding)
    • How RTP is used to mix voice (conference calls)
    • The RTP header
    • RFC 2833 protocol
    • RTP Control Protocol (RTCP)
      • SDES
      • Sender/receiver reports
      • Bye reports
  14. IMS Presence
    • SIMPLE - SIP for Instant Messaging and Presence Leveraging Extensions
    •  Terminology
    •  Framework
    •  Resource List Manipulation Requirements
    •  Authorization Policy Manipulation
    •  Acceptance Policy Requirements
    •  Notification Requirements
    •  Content Requirements
    •  General Requirements
  15. SIP Timers
    • T1, T2, T4
    • Timer A – K
  16. IMS Security
    • Security for call setup
    • Authentication
    • S/MIME
    • TLS
    • Privacy and identity
    • Diameter

 

 

Course Labs

  • Configure PCs and Build an IP network
  • Learn how to use Ethereal
  • Configure the SIPURA ATA
  • Configure X-Lite SIP Client
  • Model the CSCF functions using the ONDO SIP Proxy
  • asterisk@home
  • Perform Call traces with ethereal and SIP UAs
    • SIP REGISTER without authentication
    • SIP REGISTER with authentication
    • Simple SIP Call without INVITE authentication
    • SIP call with INVITE authentication
    • 100rel  (PRACK)
    • Busy call
    • Vacant Number
    • Abandoned Call
    • DTMF - SIP INFO
    • DTMF - RFC 2833
    • DTMF – in band
    • SIP NOTIFY (voice mail indication example)
    • Call Forward Immediate
    • Call Forward No Answer
    • Call Transfer (REFER)
    • Bye message with RTCP call information
    • SIP Timers effect on call processing
    • Model the IMS using multiple layers of proxies

 


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