SIP Essentials (SIP)

1. 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

2. SIP Architecture

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

3. Regular Expression

• Metacharacters
• Substitution
• REGEX Modifications

4. Routing the SIP INVITE

• Proxy Routing
• Via and Record-Route

5. The SIP Dialog

• SIP Dialog
• The reINVITE

6. SIP Entities

• SIP Topology
• SIP Proxy
• B2BUA
• Outbound Proxy

7. 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

8. SIP Call Routing

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

9. SIP Uniform Resource Indicators (URIs)

• URI vs. URL vs. URN
• SIP URI Examples
• URI Delimeters
• SIP and SIPs
• tel URI
• URI Escape Codes

10. 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

11. Session Description Protocol (SDP)

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

12. SIP and the DNS

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

13. ENUM

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

14. Legacy

• Early Media
• SIP-T and SIP-I

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

• RTP Headers
• RTP Dejitter
• Conferencing
• RTCP

16. DTMF Handling

• DTMF
• SIP INFO
• RFC 2833

17. Fax Handling

• T.30
• T.38
• SDP RFC 3407

18. Presence

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

19. SIP Timers

• Standard Timer Values
• Session-Expires

20. SIP Security

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

21. SIP NAT Traversal

• NAT
• NAT Types
• STUN & TURN

SIP Essentials Labs

• Understanding the Network
• Lab 1: Construct & 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 Softclient
• 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