Training Categories Integrating SIP Communications

Integrating SIP Communications

Integrating SIP Communications Course Details:

In this course, you will learn about Session Initiation Protocols (SIPs) and important protocols related to SIP implementations through a process of lecture and hands-on training. Gain insights into what SIP is, 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. Since more than half are hands-on, you can investigate and reinforce each lesson. In this course, you’ll examine how SIP interoperates into the current telecommunications network by going beyond the basics of the protocol and getting a big picture understanding of how it all fits together.

    No classes are currenty scheduled for this course.

    Call (919) 283-1674 to get a class scheduled online or in your area!

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. REGEX

  • Regular Expression

4. Routing the SIP INVITE

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

5. The SIP Dialog

  • The Purpose of the SIP Dialog
  • How to Begin and End a Dialog
  • The Dialog ID

6. SIP Entities

  • B2BUA
  • Proxy
  • SBC
  • Outbound Proxy
  • UA

7. SIP Call Flow Examples

  • The Following Call Flows Set Up and Examined Using Wireshark
  • REGISTER
  • Normal Call
  • Busy
  • Redirect
  • Transfer (REFER)

8. SIP Call Routing

  • How SIP Routing Is Used to Route Calls
  • Use of Record-Route in Stateless Routing Proxies
  • How SIP Is Used in the PSTN Migration to an All IP Network

9. SIP Uniform Resource Indicators (URIs)

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

10. SIP Message 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

11. Session Description Protocol (SDP)

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

12. SIP and the DNS

  • 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)

13. ENUM

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

14. SIP and DHCP

  • DHCP Protocol
  • SIP DHCP Options

15. 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

16. 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

17. DTMF Handling

  • Inband
  • RFC 2833
  • SIP INFO

18. Fax Handling

  • Inband
  • Fax Relay
  • T.38

19. 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

20. SIP Timers

  • T1, T2, T4
  • Timer A – K

21. SIP Security

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

22. NAT Traversal

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

23. SIPp: A SIP Testing Tool

  • SIPp
  • SIPp XML Examples

*Please Note: Course Outline is subject to change without notice. Exact course outline will be provided at time of registration.
  • Why SIP is a valuable protocol
  • SIP architecture
  • 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

Lab 1: Construct and Enable a VoIP Network 
Lab 2: SIP User Agent Configuration 
Lab 3: Direct UA to UA Routing with No Proxy 
Lab 4: Proxy Based SIP Routing 
Lab 5: Adding Authorized UAs to a Domain 
Lab 6: Intra Domain Routing (SIP in the Same Domain) 
Lab 7: SIP REGISTER – Registering a SIP UA 
Lab 8: Registering a SIP UA Soft Client
Lab 9: Registering a SIP UA Client to a Mobile Device 
Lab 10: Inter Domain Routing (SIP in Different Domains)
Lab 11: Strip off the Leading 9 
Lab 12: PDT Management 
Lab 13: Using Wireshark 
Lab 14: Capture a SIP Registration via Wireshark 
Lab 15: Capture a Normal SIP Call via Wireshark 
Lab 16: Capture a Call to a Vacant Number via Wireshark 
Lab 17: Capture a SIP Call to Busy Number via Wireshark
Lab 18: Capture a Call Forward via Wireshark
Lab 19: Via, Record Route, and Route Headers 
Lab 20: Examining Max Forwards 
Lab 21: INVITE with SDP – sendonly vs. sendrecv 
Lab 22: Silence Suppression 
Lab 23: DTMF RFC 2833 and SIP INFO 
Lab 24: SIP B2BUA Configuration Example 
Lab 25: Register Linksys SIP Phone with Asterisk PBX 
Lab 26: SIP Presence (NOTIFY) 
Lab 27: RTP Relay 
Lab 28: Direct RTP Flow between Two UAs – 3PCC 
Lab 29: ENUM Call Routing 
Lab 30: Testing SIP Connectivity Using SIP OPTIONS 
Lab 31: Advanced: SIP Testing with SIP-p

  • Individuals who want to learn more about SIP
  • Individuals responsible for installing SIP trunking or internetwork telephone systems using SIP

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