Thoroughly 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.

What You'll Learn

• Why SIP is a Valuable Protocol
• SIP Architecture
• Understand SIP Uniform Resource Indicators (URIs)
• SIP Headers
• SIP-Related IP Services
• SIP for Instant Messaging and Presence Leveraging Extension (SIMPLE)
• How SIP Intelligently Routes Calls over Any Network
• SIP Security

Who Needs to Attend

• Those who want a technical explanation of how SIP works, how to make SIP work, and why SIP is important
• Those who are going to install SIP trunking or internetwork telephone systems using SIP

Prerequisites

• TCP/IP Networking
• Voice over IP Foundations

Follow-On Courses

There are no follow-ons for this course.

Course Outline

1. VoIP Introduction

• Circuit Switching
• VoIP Protocols
• VoIP Deployments: First Installations to Now
• SIP and the Softswitch

2. SIP Architecture

• The SIP Architecture
  • UA, Proxy, Redirect, Forking, and B2BUA
• Multimedia Architecture
  • RTP/RTCP
  • SDP
• Methods
  • REGISTER
  • INVITE and ACK
  • UPDATE
  • OPTIONS
  • REFER
  • CANCEL
  • SUBSCRIBE and NOTIFY
  • MESSAGE
  • BYE
• SIP Responses
• Via Path
• Record-Route

3. Call Routing Essentials

• The Via: path
• Creation of Response-Path
• Response Merging
• Record-Route: and Route:
• Forking
• Loops and Spirals

4. SIP Uniform Resource Indicators (URIs)

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

5. SIP Headers

• Via
• Branch
• Max-Forwards
• Dialog (To, From, and tag= fields, Call-ID)
• CSeq
• Proxy Authenticate
• Proxy-Authorize
• Contact
• Expires
• User-Agent
• Content-Length
• Allow
• Supported
• P-Access
• Network-Info
• P-Charging-Vector, P-Preferred-Identity, P-Asserted-Identity
• Authorization
• Security-Client
• Security-Server
• Content-Type

6. Session Description Protocol (SDP)

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

7. DNS Essentials

• Basic Resource Records (RR)
• A-Record, SOA, NS Record, MX Record
• The SRV Record (RFC 2782)
• How SIP Uses the SRV Record (RFC 3263 Locating SIP Servers)
• How to Configure a SRV Record
• The NAPTR Record (RFC 2915)

8. ENUM

• ENUM Protocol (RFC 3761)
• Dynamic Delegation Discovery System (RFC 3401, 3402, 3403, 3761, 3764)
• How SIP Uses ENUM

9. SIP and DHCP

• DHCP Protocol
• SIP DHCP Options

10. Interoperating SIP with Legacy PSTN Signaling

• Call Transfer (REFER)
• 183 Early Media
• Interworking SIP with Local Call Control (E&M or DID)
• SIP and the PSTN
• SIP-T

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

• 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
  • SDES
  • Sender/Receiver Reports
  • Bye Reports

12. DTMF Handling

• Inband
• RFC 2833
• SIP INFO

13. Fax Handling

• Inband
• Fax Relay
• T.38

14. 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. SIP Security

• Security for Call Setup
• Authentication
• S/MIME
• TLS

17. NAT Traversal

• How NAT Operates on SIP and SDP
• NAT Types
• STUN
• TURN
• ICE

Labs

Lab 1: Configure PCs and Build an IP Network

Lab 2: Learn How to Use Wireshark

Lab 3: Configure the SIPURA ATA

Lab 4: Configure X-Lite (Counterpath) SIP Soft Client

Lab 5: Configure the ONDO SIP Proxy

Lab 6: Configure the Asterisk-Based trixbox

Lab 7: Perform Call Trace Analysis

• SIP REGISTER without authentication
• SIP REGISTER with authentication
• Simple SIP Call without INVITE authentication
• SIP call with INVITE authentication
• Busy Call
• Vacant Number (Call a number that does not exist)
• Abandoned Call (Hang up on an unanswered call)
• Call Routing (Multiple Proxies)
• Via: routing
• Record-Route: and Route: "routing"
• 100rel (PRACK)
• Call routing and registration using the DNS SRV record
• Call routing and registration using the DNS NAPTR record
• RTP packet interval impact on QoS and Bandwidth
• RTP jitter buffer analysis and impact on quality (small, large, dynamic, etc.)
• RTP relay and the session border controller
• Bye message with RTCP call information
• SDP - CODEC Negotiation
• SDP - RTP session establishment
• DTMF - SIP INFO
• DTMF - RFC 2833
• DTMF - In Band
• Response 405 (example: X-lite phone, DTMF not supported)
• SIP NOTIFY (Voice Mail indication example)
• SIP SUBSCRIBE and NOTIFY (presence)
• SIP MESSAGE (Instant Messaging)
• Call Forward Immediate
• Call Forward No Answer
• Call Transfer (REFER)
• SIP Timers Effect on Call Processing
• Call Park and Retrieve
• Conference