Integrating SIP Communications
COURSE OUTLINE:
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, you�ll 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 � 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