Advanced SIP and IP Telephony - Hands On

Description
This course reviews and practices the usage and architecture of SIP and extends the theory into practice with practical labs which are an important part for gaining a firm knowledge base for engineering/support personal. As SIP is the dominant protocol for Voice/ Multimedia Over IP (VoIP / MoIP) and IP Telephony. Many organizations consolidate as many services possible under their existing IP infrastructure as VoIP/MoIP/IPT messaging and presence services are top market requirements in today's developing IP network environments. This course will deliver a hands-on, expert level experience in the SIP architecture and thorough review of important subjects. During the course labs are interweaved in the syllabus to provide a fuller understanding of issues learned in class.

Objectives
By the end of the course, the participant will be able to:
Implement SIP telephony over IP networks
Describe the IETF approach to IPT – SIP, RTP and RTCP, RDP and other protocols in the architecture.
Describe the SIP's integration into the telecommunications infrastructure.
Perform a Thorough analysis of SIP call flows for troubleshooting purpose.
Describe RTP and RTCP mechanisms.
Describe and analyze interface between IPT and PSTN and cellular networks
Understand QoS and Security in IPT networks
Understand the role of MGCP/H.248 protocol and gateways

Topics
Introduction to telephony
SIP architecture and components
SIP header structure and message types
SDP – Session Description Protocol
SIP call flows and call routing
RTP and RTCP – protocols and operation

Target Audience
R&D, Engineering and Technical Support

Prerequisites
Strong technical background in data communications

Duration
5 Days

Outline

• Introduction to Telephony
  • Introduction to the PSTN network
  • Digitizing voice
  • Voice packetizing
  • Signaling in the PSTN and SS#7
  • H.323 and prior protocols
• SIP Architecture
  • SIP Architecture
    • SIP entities and functions
  • Request – response model
    • Requests types – methods: INVITE and ACK , REGISTER , UPDATE , OPTIONS ,REFER , CANCEL , BYE
    • Responses classes: 1xx Informational, 2xx Final, 3xx Redirection, 4xx Client Error, 5xx Server Error, 6xx Global Failure.
  • SIP message structure
    • SIP addressing
    • Message Parts
    • SIP Header Fields
    • Option Tags and extensions
    • Message body and SDP
  • SIP Timers
    • T1, T2, T4 …
  • LAB - SIP components and basic call flows
    • In the lab, we will familiarize ourselves with sip basic entities and practice simple scenario call flows.
• SIP Uniform Resource Indicators (URIs)
  • Generic URI information (RFC 3986)
  • PSTN Number
  • Instant Messaging
  • Presence
  • LAB - Addressing and Numbering
  • o In the lab we will evaluate the different addressing methods and perform
• SIP Header Structure
  • Standard fields
    • SIP headers structure and syntax
    • Via header
    • Branch header
    • Max-Forwards header
    • Dialog context (To, From, and tag= fields)
    • CSeq header
    • Call-ID header
    • Contact header
  • SIP extensions
    • SIP Reliability
    • INFO
    • EARLY CONNECT
  • Registration
    • Expires header
    • Authentication
  • LAB - SIP message formats
    • In this lab the SIP message format will be thoroughly analyzed with an in depth review of each field and its derivatives. Example call flows will be analyzed to review the issues discussed in class.
• Session Description Protocol (SDP)
  • Session Parameters and RFC 4566
  • SDP Format – fields structure and types
  • Extending SDP
  • Media Negotiation and failure responses
  • Changing Session Parameters
  • LAB - SDP usage
    • In this lab we will examine the procedure for choosing media types and test different compatibility scenarios.
• Call Flow Examples and SIP Extensions
  • Call setup
    • Call Attempt – Successful/ Unsuccessful
    • Incoming/Outgoing
    • Registration with/without credentials
  • Presence
    • Presence Subscription
    • Presence Notification
    • Instant Message Exchange
  • Call Hold / Transfer
  • LAB No.5 Call flow analysis
    • In this lab we will analyze the call flows discussed in class, reviewing the different requests and responses of SIP extensions.
• Advanced SIP and Call Routing
  • Call routing issues
    • SIP triangle
    • SIP trapezoid
    • SIP Peer to peer
    • Direct / Proxied Call
  • Forking to multiple locations and HERFP
  • Response Path
  • Via-Path Creation
  • ENUM Architecture
  • Top hierarchy DNS operation
  • LAB - SIP routing lab
    • In this lab we will simulate and analyze the different architectures, fork a call and regard the via, route record and DNS mechanisms
• RTP and RTCP operation
  • Basics
    • Packet Loss, Latency, Jitter – resolution ,methods
    • Defining the session with RTP
  • RTP
    • RTP Payload - Type Field
    • RTP Telephony Events (RFC 4734)
    • Jitter removal using RTP
    • Handling Packet Loss
  • Silence Suppression/Fixed Length Packets (Padding)
  • Conferencing – Mixing different Voices with RTP
  • The RTP Header
  • LAB – RTP and RTCP
    • In this lab we will examine the effects of jitter and packet loss on the session and investigate the mechanisms involved in the solution.
• SIMPLE - SIP for Instant Messaging and Presence Leveraging Extensions
  • SIMPLE Terminology
  • The SIMPLE Framework
  • Resource List Manipulation Requirements
  • Manipulation of Authorization Policy
  • Acceptance Policy Requirements
  • Notification Requirements
  • Content Requirements
• SIP Security and advanced features
  • Security
    • Security for Calls
    • Authentication
    • S/MIME
    • TLS
    • Privacy and Identity
  • NAT Issues
    • Firewall Traversal
    • SIP Traversal
    • RTP Traversal
    • NAT Function
    • Full / Restricted / Symmetric Cone NATs
    • STUN - Simple Traversal of UDP through NATs
    • TURN - Traversal Using Relay NAT
• SIP Signaling in 3G/UMTS/3G+ architecture (entities call session control function, PTT implementation, who do you register with (PTT over IMS architecture)
  • Architecture
    • Registration
    • Billing
    • SIP - IMS Architecture: P-CSCF, I-CSCF, and S-CSCF
  • IMS Sessions
    • Session initiation / termination
    • Roaming scenarios
    • SIP from terminal to network
    • AIN call trigger points
  • Deployments in UMTS
    • 3GPP IMS
    • Security QoS and Signaling issues
    • Media conversion ( PCM to IP )
    • Instant Messaging & Presence
    • Conferencing
• Advanced SIP Communication Services
  • Architecture
    • Session Border Controllers
    • SIP based Services
    • SIP Gateways
    • Media Gateway Control Protocol (MGCP)
    • Megaco/H248
    • Session Border Controllers
    • Softswitches
  • Implementation
    • Unified messaging
    • Pre/Post paid
    • SIP phones
    • One number access
    • Virtual PBX
    • Vendors