SIP Essentials

Any company or individual who wants to advance their comprehension of VoIP and SIP

After completing this course you should understand:

• SIP Requests and Responses
• Live call capture
• Wireshark Analysis (pcaps & ng-pcap)
• RTP Voice and Video
• Session Description Protocol (SDP) negotiation
• DTMF transmission
• SIP Routing and Dialplan construction (regular expression)
• Call flow analysis
• Testing with SIP-p
• Troubleshooting (failed calls, 1-way or no way voice)
• STUN / TURN / ICE

SIP Introduction

• SIP Message Format
• Legacy Call Control
• Compare SIP
• Packetizing Voice
• SIP Call Flow
• How SIP Routes Media
• SIP Call Control
• SIP in 4G

SIP Architecture

• SIP UA
• SIP Requests
• SIP Response
• SIP URI
• SIP Architecture
• SIP Domain
• SIP Registration
• SIP Call Routing
• Loose Routing

Regular Expression

• Metacharacters
• Substitution
• REGEX Modifications

Routing the SIP INVITE

• Proxy Routing
• Via and Record-Route

The SIP Dialog

• SIP Dialog
• The reINVITE

SIP Entities

• SIP Topology
• SIP Proxy
• B2BUA
• Outbound Proxy

SIP Call Flow Examples

• Wireshark Colors
• Wireshark Preferences
• SIP Stack
• REGISTER with Authentication
• Wireshark Analysis of SIP Dialog
• SIP Redirect
• CFNA
• REFER and Call Transfer

SIP Call Routing

• PRACK 100-rel
• Call Forking
• Loop and Spiral
• Third Party Call Control
• Path Minimization
• SIP in the PLMN
• OPTIONS Method

SIP Uniform Resource Indicators (URIs)

• URI vs. URL vs. URN
• SIP URI Examples
• URI Delimiters
• SIP and SIPs
• Tel URI
• URI Escape Codes

SIP and the DNS

• Zone File
• SOA and NS Records
• A-Record
• SRV Record
• NAPTR Record
• Locating SIP Servers

ENUM

• ENUM Database Example
• ENUM Query and Response
• ENUM REGEX
• Post ENUM Routing

SIP and the PSTN

• Early Media
• SIP-T and SIP-I

SIP

• SIP QA testing
• SIP DOS Testing

SIP Message Headers

• SIP Header Overview
• Dialog ID Headers
• User-Agent
• SIPp Header Modification
• Proxy-Authenticate
• Allow and Supported
• History Info
• Join
• Session Expires
• PPI and PIA
• Establish Service Path
• IMS Hosted
• Content-Type

Session Description Protocol (SDP)

• SDP Background
• SDP Format
• SIP = one way?
• SDP Lines
• SDP Offer/Answer
• Call Hold

RTP and Real-Time Control Protocol (RTCP)

• RTP Headers
• RTP Dejitter
• Conferencing
• RTCP

DTMF Handling

• DTMF
• SIP INFO
• RFC 2833

Fax Handling

• T.30
• T.38
• SDP RFC 3407

Presence

• Presence Overview
• PIDF XML Example
• Rich Presence
• Presence Message Flow
• Instant Messaging

SIP Timers

• Standard Timer Values
• Session-Expires

SIP Security

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

SIP NAT Traversal

• NAT
• NAT Types
• STUN & TURN

Labs

Summary - A short overview on how to navigate the lab environment

• Welcome to Alta3 Labs
• Linux Fundamentals
• Using vim

Packet Captures - Learn how to use wireshark to test and debug SIP calls

• Wireshark
• Making pcaps with tcpdump
• Making pcaps with tshark

SIP Registrars - Analyze the SIP REGISTRATION

• REGISTER a SIP UA to SipGate
• Successful REGISTER by a User Agent
• REGISTER Fails Auth
• Live capture of SIP REGISTER with tcpdump

SIP Calls - Analyze SIP call flows via B2BUA

• The SIP INVITE
• SIP INVITE Packet Analysis with Wireshark
• Troubleshooting Common SIP Failures with Wireshark
• Live capture of SIP INVITE with tcpdump

SIP Proxies - Analyze call flow through a proxy

• INVITE Relay by SIP Proxies
• Canceled SIP call
• No Record Routes

SIP Tools - Use various SIP testing tools to view special call flows.

• SIPp SIP Tester
• SIP Swiss Army Knife

Exploring Media

• Methods for Transport of DTMF

Attendees should meet the following prerequisites:

• Networking Experience

 

 

Recommended as preparation for the following exams:

• There is no exam aligned to this course