Online VoIP Training in the UAE

This three-day hands-on course delves into the message flows and call sequences of the Session Initiation Protocol (SIP) within voice networks. It combines theoretical knowledge with practical application, using protocol analyzer traces for illustrative and troubleshooting purposes. Participants will configure VoIP systems using IP phones, softphones, SIP-capable Cisco routers, and SIP-based PBXs such as Trixbox.

Audience:

The course is designed for Network Planners, Designers, and Engineers who need to understand SIP. Approximately 40% of the course involves practical work.

The course comprises two interconnected segments—a theoretical component and a practical one. The initial segment spans one day, providing an overview of motivation, philosophy, foundational principles, and operational rules of the SIP protocol, with a particular emphasis on its application in IP telephony and VoIP services. The subsequent two-day portion allows participants to delve into practical aspects through hands-on laboratory exercises, offering comprehensive insights into configuring components of the SIP telephony architecture, analyzing SIP signaling at both message sequence chart and internal structure levels, and addressing typical issues and troubleshooting, including security and telecom fraud concerns. Trainers will share their expertise in deploying, operating, and managing SIP telephony, encompassing virtualization and cloud-based solutions. The practical segment utilizes both SIP hardphones and softphones along with IP telephony servers (Asterisk and Freeswitch). Participants can leverage the trainers' extensive technical and business experience in IP telephony to discuss their specific challenges and queries. These will be incorporated into the agenda during the wrap-up session as an extension of the training, addressing current critical needs of participants. This course is designed for individuals with foundational knowledge and experience in telecom services, particularly VoIP and IP networking.

Voice over Internet Protocol (VoIP) is a technology that enables users to make voice calls through an internet connection rather than traditional telephone lines.

In this instructor-led live training session, participants will learn how to establish a VoIP system using Cisco CallManager Express.

By the end of this training, participants will be able to:

• Grasp essential VoIP concepts
• Operate VoIP equipment effectively
• Set up CISCO routers for VoIP usage
• Integrate VoIP and CallManager Express into their network infrastructure
• Configure CME features for SIP, SCCP, and soft phones
• Navigate advanced configuration settings and features of CME

Audience

• Managers
• System administrators
• Software and support system developers
• Technicians
• Network engineers

Course Format

• A blend of lecture, discussion, exercises, and extensive hands-on practice

This three-day hands-on course delves into the core principles of various voice protocols that enable voice (and video) transmission over a packet-switched network. The curriculum blends theoretical knowledge with practical application, utilizing Wireshark for demonstrations and troubleshooting. Participants will configure VoIP using IP phones, softphones, and Cisco routers equipped for voice services. While this course is generally focused on generic VoIP concepts, it employs Cisco equipment to establish the IP backbone and voice gateways. We also offer specialized training courses tailored specifically to Cisco voice solutions. Additionally, the course addresses call setup between circuit-switched and packet-switched networks.

Approximately 40% of the course is dedicated to practical exercises.