Info: 2 day course

When: in curand

Where: Bucharest, Romania

Course Overview:
In this course, you will discover the explosive dynamics of bringing voice and data together on a single network. You will learn How SIP Phone has evolved; review the typology, architecture, SIP, SDP and RTP protocols including different implementations.

The course participants will setup IP Phone and register to a Softswitch, define different services and applications which are widely used by SMB and large Enterprise.

Each one of the participants will access sophisticated Softswitch which will be on the same network for generating end-to-end IP telephony and Video calls. Using the Wireshark each participant will experience the usage of the Wireshark to analyse the various protocols.

Prerequisites:
Basic knowledge of:
– IP Networks
– Basic telephony

Course Content:

Day 1 – 9.00-16.00

1. SIP Introduction
• What is Session Initiation Protocol
• The Incentive
• SIP Components
• SIP Servers (Proxy, Registrar, Redirect, Location)

2. SIP architecture
• Protocol Stack
• SIP Transactions and response codes
• Addressing format
• DTMF and VoIP (In-Band and Out of Band methods)

3. RTP, CRTP and RTCP
• Internet Multimedia Protocol Stack
• RTP Structure
• RTCP Profile and structure
• CRTP and Bandwidth consumption

4. Hands-on Session: Real Time Registration capture analysis (using Wireshark)
• Setting and registration process
• CA List Configuration
• Network parameters and Account setting
• Compression setting (Codecs) and QoS setting
• Real Time Call setup capture analysis (using Wireshark)
• Digest Authentication process analysis

5. QoS and QoE challenges for SIP Telephony
• Definitions and terms
• The need for QoS
• Solutions to provide QoS

6. SIP and Media Compression methods
• Coders types and Bandwidth utilization
• Side effects due to compression

Day 2 – 9.00-16.00

7. SIP Protocol structure
• SIP Timers for reliability
• Forking Methods
• Provisional Response Acknowledge method

8. Session Description Protocol
• SDP Main tasks
• SDP Messages
• Mandatory fields and optional fields

9. SIP based Softswitch and Class 5 services
• REFER: Call Transfer using REFER
• SIP new methods and Supplementary Services
• Presence and Instant Messaging
• Conference – SIP Call flow analysis

10. Hands-on Session – Real Live CaptureAnalysis
• Session Description Protocol Analysis
• RTP Capture analysis
• Payload type analysis
• RTCP Capture analysis
• Secured and Non secured conference
• DTMF over IP – Real Live capture analysis

11. SIP Trunking Overview
• Introduction
• What is SIP Trunk
• SIP trunk Challenges
• References and relevant RFC

12. SIP and Telephony networks
• SCTP- Stream Control Transmission Protocol
• SCTP and adaptation Layers
• SIP-T and SIP-I (ISUP encapsulation and mapping)

Cost:

Early Bird

cost total per participant

Full Price

cost total per participant

Groups larger than 10 participants

10%

discount

Pentru a va inscrie sau a solicita informatii suplimentare, va rugam sa folositi formularul de mai jos sau trimiteti un mail la adresa office@basmtel.ro

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