• 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
  

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

Required Prerequisites:

• TCP/IP Networking
  
• Voice over IP Foundations
  

Outline

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
  

Labs

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