Integrating SIP Communications

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

• 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