日本のVoIPトレーニング

The 3 day practical course covers the messages and call flows of the Session Initiation Protocol (SIP) and its use in voice networks. This course is a mixture of theory and practice (utilising protocol analyser traces where appropriate for explanation and troubleshooting) with practical VoIP configured using IP telephones, softphones, voice capable Cisco routers and SIP IP PBX,s (e.g. Trixbox).

Audience:

Network Planners, Designers, and Engineers requiring an understanding of SIP.

The course is around 40% practical.

The course consists of two complementary parts – a theoretical and a practical one. The first is a one day introduction covering motivation, philosophy, fundamentals and rules of operation of the SIP protocol and ways it is used to implement telecom services with focus on IP telephony and VoIP. The second two-day part enables participants to learn practical aspects of service operation within a framework of hand-on laboratory exercises giving detailed insight into configuration of components of SIP telephony architecture, SIP signalling at both message sequence chart and internal message structure levels, and assists in understanding of typical problems and troubleshooting including security and telecom fraud aspects. The trainers will share their experience in launching, operation and management of SIP telephony covering also virtualization and cloud based solutions. Practical part is presented using both SIP hardphones and softphones and IP telephony servers (Asterisk and Freeswitch). Participants can take advantage of the fact trainers have got rich technical and business experience in IP telephony and submit their own problems and questions. They will be included in the agenda at wrap-up as a supplement to the training to meet current urgent needs of clients. Training is addressed to participants with basic knowledge and experience in telecom services – specifically in VoIP and IP networking.

VoIP (Voice over Internet Protocol) is a technology that makes it possible for users to make voice calls over an Internet connection instead of through a regular phone line.

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

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

• Understand fundamental VoIP concepts
• Learn how to use VoIP equipment
• Configure CISCO routers for VoIP
• Implement VoIP and CallManager Express into their network
• Configure CME features for SIP, SCCP, and soft phones
• Work with advanced configuration settings and features of CME

Audience

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

Format of the course

• Part lecture, part discussion, exercises and heavy hands-on practice

The 3 day practical course covers the fundamentals of the various voice protocols used to carry voice (and video) over a packet switched network. This course is a mixture of theory and practice (utilising Wireshark where appropriate for explanation and troubleshooting) with practical VoIP configured using IP telephones, softphones and voice capable Cisco routers. This is primarily a 'generic' VoIP course – but uses Cisco equipment to provide an IP backbone and voice gateways. We can also provide Cisco specific voice training courses. The course also covers call setup between circuit switched and packet switched networks.

The course is approximately 40% practical.