CIPTV1 Implementing Cisco IP Telephony and Video, Part 1 (E-Learning)

Course Duration: 5 days

Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 is a five-day course that prepares the learner for implementing a Cisco Collaboration solution at a single-site environment. This course focuses primarily on Cisco Unified Communications Manager Version 10.x, which is the call-routing and signaling component for the Cisco Collaboration solution. Lab exercises included in the course help learners to perform post installation tasks, configure Cisco Unified Communications Manager, implement MGCP and H.323 and, SIP trunks, and build dial plans to place single site on-cluster and off-cluster calling for voice and video. Learners will also implement media resources, audio and video conferencing, and describe QoS. In addition, you will use the Cisco Unified Border Element (CUBE) with SIP Trunks. With the increasing importance of video and video conferencing, you will also learn the configuration tasks for audio and video conference bridges, Cisco TelePresence conferencing products (including TelePresence Server, TelePresence MCU, and TelePresence Conductor).

• How the CUCM administrative and service GUIs work
• Activate, start, and stop CUCM services
• Configure base CUCM components, such as date time groups, device pools, Call Manager groups, and other common elements
• Add and delete phones manually and using auto registration
• Add users, assign them capabilities, and associate them with phones
• LDAP Integration including LDAP synchronization and LDAP authentication
• LDAP attribute mapping and filters
• Deploying IP Phone services
• Configure phone features: Music on Hold (MOH) and phone services
• Set up media resources to use for MOH and conferencing
• Build a dial plan including route patterns, route lists, and route groups supporting both the NANP and variable-length dial plans
• Deploy line/device Class of Service using partitions and calling search spaces for call blocking
• Call hunting (hunt lists) and call queuing configuration
• PSTN access methods, gateway vs. Cisco Unified Border Element (CUBE), and codec selection
• PSTN access using MGCP gateways, including route lists, route groups, and digit manipulation
• PSTN access using H.323 gateways including inbound and outbound dial peer selection
• 323 gateway digit manipulation, codec selection, and class of restriction
• PSTN access using the CUBE and SIP trunks CUBE and URI dialing
• Media Resources including MOH, annunciators, and Media Termination Points (MTPs)
• Hardware and software audio and video conference bridges
• TelePresence MSE 800, TelePresence server, and TelePresence Conductor conferencing
• Quality of Service (QoS) and bandwidth calculations
• Best-Effort, IntServ, and DiffServ QoS models
• QoS classification and marking
• QoS policing and shaping