14 September - 18 September 2009 (Mon - Fri)
A 5-day instructor led with hands-on labs using Cisco, Nortel, Pingtel, Microsoft and Snom SIP equipment with an Open Source PBX Asterisk for hands on learning, 0900 - 1730
Lagos, Nigeria

At the end of this VoIP with SIP course the student will be able to:
- describe the operations of SIP
- describe the benefits of SIP
- compare SIP to the PSTN and H.323
- plan a VoIP rollout
- describe the shortcomings of SIP
- interpret SIP protocol packets

An in depth seminar on how SIP works including protocol analysis. This course provides a detailed description of the Session Initiation Protocol to implement Voice over the Internet Protcol. The standards are covered and hands-on labs using Cisco, Linksys and Asterisk VoIP equipment to reinforce the material.

- What SIP is and the market opportunities
- What can be delivered using SIP technology
- What are the SIP deployment business models
- What are the SIP service bundles - voice, data, video

- Internet Telephony Service Providers (ITSPs)
- VoIP systems manufacturers, designers and integrators
- VoIP dealers, distributors, and channel specialists
- Telecommunication carriers
- Executives: CEO, CXO, CMO, CTO, CIOs
- Enterprise network managers, operations and planning staff
- Government Regulators
- Researchers, Analysts, Investors and others allied to the field
- Anyone who needs to learn the infrastructure of the IP Telephony

1. Technical Overview
- VoIP Benefits
- Circuit Switching
- Packet Switching
- VoIP Definition
- VoIP Performance Metrics
- Power Over Ethernet (PoE)
- VoIP over WiFi (VoFi)
- Voice over Broadband (VoBB)
- Analog Terminal Adapters (ATA)
- Embedded Multimedia
Terminal Adapter (EMTA)
- VoIP Implementation Examples
- Why Users Like VoIP
- Why Service Providers Like VoIP
- Internet Telephony Service Providers
- Next-Generation Carrier Networks

2. IP Protocol Format
- TCP/IP Architecture
- Best Effort Delivery
- IPv4
- Type of Service (TOS)
- IP Header
- TCP Header
- UDP Header
- VoIP Packet
- Class of Service (CoS)
- Quality of Service (QoS)
- Differentiated Service Code Point (DSCP)
- voice packet decode

3. SIP Standards
- Session Initiation Protocol (SIP)
- RFC3261
- SIP Requests
- SIP Responses
- SIP INVITE
- Session Description Protocol (SDP)
- SIP INVITE Components
- SIP Trapezoid
- SIP Uniform Resource Indicators
- SIP User Agents
- SIP Proxy Servers
- SIP Inbound Proxy Server
- SIP Outbound Proxy Server
- SIP Call Flow Scenarios
- SIP Call Attempt
- SIP Presence Subscription
- SIP Registration
- SIP Presence Notification
- SIP Call Setup
- SIP Call Hold
- SIP Call Transfer Attended
- SIP Call Transfer Unattended

4. Audio & Video Transport
- Real Time Transport Protocol (RTP)
- Payload Types (PT)
- Synchronizing Source Identifier (SSRC)
- RTP Packet Format
- Secure RTP (SRTP)
- Real Time Transport Control Protocol (RTCP)
- Sender Report
- Receiver Report
- RTCP Packet Format
- RTP Testing Strategies
- RTP Control Protocol Extended Reports (RTCP XR)
- Video Codecs

5. Quality of Service
- end to end delay
- calculating codec delays
- queuing delays
- packet jitter
- de-jitter buffer
- MOS Scores
- PESQ Scores
- Differentiated Service
- Expedited Forwarding (EF)
- Assured Forwarding (AF)

6. SIP Standards Reference
- SIP Method Cross Reference

7. SIP Call Flows
- Basic Call Flow Example
- SIP Registration
- SIP Register with Credentials
- SIP Request Methods
- SIP PUBLISH Method
- SIP UPDATE Method
- SIP REPLACES Header

8. SIP Session Timers
- Keep Alive
- Session Expires
- Mininum Session Expires

9. SIP Peering
- Open Settlement Protocol (OSP)
- Proxy Server Negotiation
- Network Address Translation (NAT) Issues
- RADIUS
- ENUM Translation
- Voice Peering Fabric
- IETF SPEERMINT

10. SIP Early Media
- In-band call progress
- SIP to SS7 Gateway
- PSTN to SIP
- SIP to PSTN
- SIP PRACK
- 183 Session Progress
- ISUP Tunneling
- REFER Method
- Stream Control Transmission Protocol (SCTP)