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VoIP Protocol Essentials: SIP

  • Código del Curso GK3251
  • Duración 5 días

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eur1,200.00

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Este curso está disponible en los siguientes formatos:

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Study the SIP protocol through a process of lecture and hands-on training. Learn what SIP is and how it works, and get a practical guide on how to use it. The lessons in this course are clear, very technical, and always practical, and since at least 60% are hands on, you can investigate and reinforce each lesson. In this course, you'll examine how SIP weaves into the current telecommunications network, going beyond the basics of the protocol and getting a big picture understanding of how it all fits together.

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This course is for people who want a technical explanation of how SIP works, how to make SIP work, and why SIP is important and for those who are going to internetwork telephone systems using SIP or install SIP trunking.

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Objetivos del Curso

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  • Why SIP is a Valuable Protocol
  • How Voice over IP uses TCP and UDP
  • SIP Architecture
  • Understand SIP Uniform Resource Indicators (URIs)
  • SIP Call Flow Examples
  • SIP-Related IP Services
  • SIP for Instant Messaging and Presence Leveraging Extension (SIMPLE)
  • How SIP Intelligently Routes Calls over Any Network
  • SIP Security
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Contenido

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1. Why SIP?

  • Circuit Switching
  • VoIP Protocols
  • SIP and the Softswitch
  • History of SIP Ethernet II

2. IP Routing and Switching Overview

  • IP Routing
  • Ethernet Essentials

3. TCP and UDP Essentials

  • How Voice over IP uses TCP
  • How Voice over IP uses UDP

4. SIP Architecture

  • The SIP architecture
  • UA, Proxy, Redirect, and Forking
  • Multimedia Architecture
  • RTP/RTCP
  • SDP
  • Methods
  • REGISTER
  • INVITE and ACK
  • UPDATE
  • OPTIONS
  • REFER
  • CANCEL
  • BYE
  • SIP Responses
  • 1xx Informational
  • 2xx Final
  • 3xx Redirection
  • 4xx Client Error
  • 5xx Server Error
  • 6xx Global Failure

5. SIP Uniform Resource Indicators (URIs)

  • Generic URI information (RFC 2396)
  • Direct or Proxy
  • PSTN Number (RFC 2808)
  • Instant Messaging
  • Presence

6. SIP Headers

  • Via
  • Branch
  • Max-Forwards
  • Dialog (To, From, and tag= fields)
  • CSeq
  • Call-ID
  • Contact
  • SIP Reliability
  • Expires
  • Authentication

7. Session Description Protocol (SDP)

  • Session Parameters
  • SDP Format
  • Extending SDP
  • SDPng
  • Media Negotiation
  • Changing Session Parameters

8. SIP-Related IP Services

  • DHCP
  • ENUM
  • DNS NAPTR Records
  • DNS SRV Records
  • Regular Expressions
  • DDDS Algorithm

9. Call Flow Examples

  • Call Attempt - Unsuccessful
  • Presence Subscription
  • Registration
  • Presence Notification
  • Instant Message Exchange
  • Call Setup - Successful
  • Call Hold
  • Call Transfer

10. Call Routing

  • Direct Call
  • Proxied Call
  • Forking
  • Loops and Spirals
  • Response Path
  • Creation of Via-Path
  • Response Merging
  • Record Route
  • Heterogeneous Error Response Forking Problem (HERFP)
  • Control Models
  • Third-Party
  • Multi-Party
  • ENUM
  • ENUM Architecture
  • How ENUM will change things
  • How DNS works from the very top of the hierarchy

11. RTP and Real-Time Control Protocol (RTCP)

  • Dealing Packet Loss, Latency, Jitter
  • How RTP Defines the Session
  • Session Description Protocol
  • H.245 Terminal Capabilities
  • 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
  • SDES
  • Sender/Receiver Reports
  • Bye Reports

12. 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

13. SIP Timers

  • T1, T2, T4
  • Timer A - K

14. SIP Security

  • Security for Call Setup
  • Authentication
  • S/MIME
  • TLS
  • Privacy and Identity
  • Firewall Traversal
  • SIP Traversal
  • RTP Traversal
  • SIP Application Level (layer) Gateway (ALG)
  • Network Address Translation Function
  • Full and Restricted Cone NATs
  • Symmetric Cone NATs
  • Simple Traversal of UDP through NATs (STUN)
  • Traversal Using Relay NAT (TURN)
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Pre-requisitos

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  • GK9025 - TCP/IP Networking
  • GK3277 - Voice over IP Foundations
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