This course first explores the operation of traditional telephone systems and the public switched telephone network (PSTN). Topics include analog to digital conversion, the North American digital hierarchy, local loop operation, and call routing considerations. Next the course moves to the fundamental of Voice over IP (VoIP) and how voice information is packetized. An examination of the infrastructure required to support VoIP along with the protocols used allows the students to deploy IP based phones within the lab. With basic operation in place more advanced voice features are explored and the students configure, customize, and troubleshoot the infrastructure and its functionality. Topics include how to setup IP phones, configure users, how to configure phones and users for Class of Service, configuration of user features such as Do Not Disturb, Conferencing, Shared Lines, and Barge. Multiple site operation and connectivity is discussed along with integration to the PSTN. Topics include dial plans, bandwidth management, and call admission control. With reliability a major concern, students explore options to provide redundant and/or backup connectivity within the VoIP infrastructure. Students must have access to a Windows based multi-media computer.
CITX 2060 - Cisco CCNA Level 2, or a Valid CCENT Certification.
No class on February 10 (Family Day Weekend) and March 31 (Easter Weekend). Students MUST have successfully completed CITX 2060 - Cisco CCNA Level 2, or have a Valid CCENT Certification in order to register for CITX 1186 – Cisco CCNA Voice. For more information on this course, please contact Camille Crowther at firstname.lastname@example.org or Randy James at Randy_James@bcit.ca. Note: BCIT reserves the right to cancel courses. In the event of a part-time studies course cancellation, you will be notified at least two business days prior to the course start. Please ensure that your contact information is current.
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Upon successful completion, the student will be able to:
Describe the operation of traditional telephony systems
Describe legacy signalling including FXS, FXO, Loop-start, Ground-start, and E&M
Describe Time Division Multiplexing (TDM) including the Digital Hierarchy (Digital Signal Levels 1,2,3)
Describe numbering plans including E.164 standards
Describe analog to digital conversion of voice signals
Describe the building blocks required to deploy unified communications systems
Describe the functionality and purpose of VoIP
List VoIP protocols and describe their usage within the unified communications framework
Describe the function of a Digital Signal Processor (DSP) and how a DSP packetizes voice streams
Describe the functions of a CODEC and list some differences between models of CODECs
Describe how voice is transmitted in RTP packets
Describe the purpose of a voice VLAN, and network services used to support VoIP
Configure voice VLANs, DHCP service options, DHCP Relay Server, NTP
Describe and verify power over Ethernet (IEEE 802.3af)
Configure IP phones for use in Unified Communications network
Configure and deploy a call manager to enable VoIP operation
Update IP Phone firmware files and XML configuration files
Deploy key features and functionality of call managers
Configure analog voice interfaces, digital voice interfaces, and dial peers to set up VoIP communications
Configure additional GUI features and phone features
Configure networking functionality to improve VoIP performance with attention to quality of service (QoS)
Effective as of Fall 2012
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