Integrating SIP Communications

Start Date	Class Duration	Session(s)	Price
Global Knowledge - Online
Apr 01, 2019	5 Days	Apr 01, 2019	$ 2,995.00	Add to Cart
Jun 03, 2019	5 Days	Jun 03, 2019	$ 2,995.00	Add to Cart
Aug 12, 2019	5 Days	Aug 12, 2019	$ 2,995.00	Add to Cart
Sep 30, 2019	5 Days	Sep 30, 2019	$ 2,995.00	Add to Cart
Dec 02, 2019	5 Days	Dec 02, 2019	$ 2,995.00	Add to Cart

COURSE OUTLINE:

Description

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, youï¿½ll 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.

Audience

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

Learning Objectives

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

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 ï¿½ 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 ï¿½ 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

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