Font size:


This is a bundled training package. It contains training for each of the bundled items below:

Course Price
Cisco CVOICE 8.0: Cisco Unified Communications Manager Express Endpoints $74.95
Cisco CVOICE 8.0: Call Routing and Dial Plans $74.95
Cisco CVOICE 8.0: Path Selection and Calling Privileges $74.95
Cisco CVOICE 8.0: Gatekeeper and Cisco Unified Border Element $74.95
Cisco CVOICE 8.0: QoS Mechanisms $74.95
Cisco CVOICE 8.0: Congestion, Rate Limiting, and AutoQoS $74.95

Bundle Price: $219.00
Total Savings: $230.70

Cisco CVOICE 8.0: Cisco Unified Communications Manager Express Endpoints

This course describes how to configure the Skinny Client Control Protocol (SCCP) and Session Initiation Protocol (SIP) endpoints in the Cisco Unified Communications Manager Express. The SCCP endpoints are defined as the Ethernet phones (ephones) and have SCCP directory numbers (ephone-dns) associated with them. The SIP endpoints are defined as voice register pools and have SIP directory numbers (voice register directory numbers) associated with them. This course discusses the various types of directory numbers available for Cisco Unified IP phones using either SCCP or SIP.

Learning Objectives
  • Identify how directory numbers are implemented in Cisco Unified Communications Manager Express
  • Identify the commands used to configure directory numbers for SCCP phones
  • Identify the commands used to define an IP phone type by configuring an ephone-type template
  • Recognize how to configure major parameters of SCCP phones and assign directory numbers to the phones
  • recognize how to configure and assign directory numbers to the SIP phones
  • Identify how to enable Cisco IP Communicator to register with Cisco Unified Communications Manager Express
  • Identify how to generate configuration files for SCCP and SIP endpoints and reset and restart SCCP and SIP phones
  • Match the steps to verify all major aspects of Cisco Unified Communications Manager Express endpoint operation with their descriptions

Cisco CVOICE 8.0: Call Routing and Dial Plans

To integrate VoIP networks into existing voice networks, you must have the skills and knowledge to implement call routing and design an appropriate numbering plan. A scalable numbering plan establishes the baseline for a comprehensive, scalable, and logical dial plan. This course describes call routing principles, discusses attributes of numbering plans for voice networks, addresses the challenges of designing these plans, and identifies the methods of implementing numbering plans. A dial plan is the central part of any telephony solution and defines how calls are routed and interconnected. A dial plan consists of various components, which can be used in various combinations. This course describes the components of a dial plan and how they are used on Cisco IOS gateways. There are times when you might need to manipulate the digits of the telephone numbers that come into and go out of your voice gateway. You may need to remove site codes for intersite calls or add area codes and other digits for routing calls through the Public Switched Telephone Network (PSTN). This course covers digit manipulation and digit manipulation tools.

Learning Objectives
  • Identify the characteristics of a typical numbering plan
  • List the different types of numbering plans
  • Match the attributes of a scalable numbering plan to descriptions
  • Recognize how to address overlap numbering plans
  • Recognize how a gateway implements the numbering plan
  • Identify the principles of gateway call routing
  • Identify the characteristics and components of a typical dial plan
  • Recognize the concept of endpoint addressing, including overlapping directory numbers
  • Identify the characteristics of call routing and path selection
  • Match PSTN dial plan requirements to descriptions
  • Identify special ISDN dial plan requirements
  • Distinguish between calling privileges and call coverage on a voice gateway
  • Recognize how a gateway collects, processes, and consumes digits
  • Match the digit stripping, digit forwarding, digit prefixing, number expansion, and calling line ID (CLID) commands with their descriptions
  • Identify how to implement voice translation rules on a gateway
  • Contrast the dialplan-pattern command to the voice translation profiles
  • Recognize how to test and monitor digit manipulation on a gateway
  • Manipulate the calling number in outbound PSTN calls

Cisco CVOICE 8.0: Path Selection and Calling Privileges

Path selection is one of the most important aspects of a well-designed VoIP system. High availability is desirable so that there is usually more than one path for a call to take to its final destination. Multiple paths provide several benefits, including redundancy in case of a link failure or insufficient resources on that link and a reduction in toll costs of a call. This course introduces the path selection strategies and methods to implement them. Calling privileges on Cisco IOS gateways are dial plan components that define the types of calls that a phone, or group of phones, is able to place. This course describes the concept of calling privileges and how they can be implemented on Cisco IOS gateways using class of restriction (COR).

Learning Objectives
  • Identify how the voice gateways select the correct path when routing voice calls
  • Recognize the process of inbound and outbound dial-peer matching
  • Distinguish between the various path selection strategies
  • Recognize how to configure site-code dialing and toll bypass in a gateway
  • Recognize how to configure TEHO
  • Recognize how to implement calling privileges on Cisco IOS gateways
  • Identify how to implement calling privileges in Cisco Unified SRST and Cisco United Communications Manager Express
  • Recognize how to configure COR
  • Distinguish between the commands used to verify COR settings
  • Configure PSTN backup for BR2 outbound calls and configure call permissions for Site BR2

Cisco CVOICE 8.0: Gatekeeper and Cisco Unified Border Element

Gatekeepers play a major part in medium and large H.323 VoIP network solutions. Gatekeepers allow for dial-plan scalability and reduce the need to manage global dial plans locally. This course describes the functions of a gatekeeper and explains how to configure gatekeepers to interoperate with gateways. It also gives an overview of the Cisco Unified Border Element and describes how to implement a Cisco Unified Border Element within an enterprise network. A Cisco Unified Border Element has the ability to interconnect voice and VoIP networks, offering protocol interworking, address hiding, and security services.

Learning Objectives
  • Identify the functions of gatekeepers in an H.323 environment
  • Distinguish between the message types that are involved in gatekeeper-based H.323 signaling
  • Match concepts related to the gatekeeper call routing process with their descriptions
  • Identify how a gatekeeper supports CAC functions
  • Identify the steps necessary to configure a multizone gatekeeper for local and remote zone call routing
  • Recognize how to configure gatekeeper zones and prefixes
  • Distinguish between the commands used to adapt an H.323 gateway configuration to register with a gatekeeper
  • Recognize how to configure CAC functions on a gatekeeper
  • Recognize how to verify that H.323 endpoints are registered properly and calls are correctly routed across a gatekeeper
  • Describe the functionality of a Cisco Unified Border Element and its applications in enterprise VoIP environments
  • Identify the Cisco Unified Border Element protocol interworking capabilities
  • Recognize how media flows are managed by a Cisco Unified Border Element
  • Distinguish between the commands used to configure media flow-around, media flow-through, and transparent codec pass-through
  • Recognize how Cisco Unified Border Element can be used to perform RSVP-based CAC
  • Match the call flows in typical Cisco Unified Border Element deployments to their descriptions
  • Recognize how to configure H.323-to-H.323 interworking on a Cisco Unified Border Element
  • Identify how to implement H.323-to-SIP interworking on Cisco Unified Border Element
  • Recognize how to verify Cisco Unified Border Element operation
  • Configure a gatekeeper with basic parameters and to register with a gatekeeper

Cisco CVOICE 8.0: QoS Mechanisms

IP networks must provide a number of services to adequately support voice transmission using VoIP. These services include security, predictability, measurability, and some level of delivery guarantee. Network administrators and architects achieve this service level by managing delay, delay variation (jitter), bandwidth provisioning, and packet loss parameters with quality of service (QoS) techniques. This course introduces the concept of a converged network, identifies four problems that could lead to poor quality of service, and describes solutions to those problems. It also explains and evaluates the three generic models of implementing QoS. Differentiated services (DiffServ) is a multiple-service model for implementing quality of service (QoS) in the network. With DiffServ, the network tries to deliver a particular kind of service that is based on the QoS specified by each packet. This specification can occur in different ways, such as using the differentiated services code point (DSCP) in IP packets or source and destination addresses. The network uses the QoS specification of each packet to classify, shape, and police traffic and to perform intelligent queuing. This course focuses on the DiffServ model and explains the mechanisms that are used to implement DiffServ. The Modular quality of service (QoS) command-line interface (CLI), or MQC, provides a modular approach to the configuration of QoS mechanisms. MQC allows network administrators to introduce new QoS mechanisms and reuse available classification options. This course outlines how to implement QoS policies using MQC, and introduces the concepts of classification and marking. It explains the different markers that are available at the data-link and network layers, and identifies where classification and marking should be used in a network. The lesson also describes different approaches for improving the efficiency of WAN links.

Learning Objectives
  • Identify how the four key quality issues for voice traffic impact voice quality
  • Identify QoS goals for voice traffic
  • Match the four methods for implementing and managing a QoS policy – CLI, MQC, Cisco AutoQoS, and QPM – with their characteristics
  • Describe three models of QoS implementation
  • Identify the purpose and function of DiffServ
  • Match the different per-hop behaviors that are used in DSCP with their descriptions
  • Recognize the mechanisms that are implemented when deploying a DiffServ model
  • Identify the components of the Cisco QoS baseline model and its variants
  • Recognize how to implement QoS by using MQC
  • Recognize how to configure classification with MQC, input interface and RTP ports, and marking
  • Identify the Cisco IOS commands that are used to configure class-based marking and trust boundaries
  • Recognize how to configure mapping between the data link layer CoS and network layer QoS
  • Match link efficiency mechanisms to their functions
  • Describe VoIP susceptibility to increased latency when large packets traverse slow WAN links
  • Recognize how LFI operates and how it reduces the delay and jitter of VoIP packets
  • Distinguish between the commands used to configure MLP, FRF.12, and cRFP

Cisco CVOICE 8.0: Congestion, Rate Limiting, and AutoQoS

Queuing algorithms are one of the primary ways to manage congestion in a network. Network devices manage an overflow of arriving traffic by using a queuing algorithm to sort traffic and determine a method of prioritizing the traffic onto an output link. Traffic policing controls the maximum rate of traffic that is sent or received on an interface. Traffic policing is used on interfaces at the network edge to limit traffic into or out of the network. Traffic shaping controls outgoing traffic on an interface to match the transmission rate to the speed of the remote end, and ensures that the traffic conforms to administrative quality of service (QoS) policies. Class-based weighted fair queuing (CBWFQ) extends the standard weighted fair queuing (WFQ) functionality, providing support for user-defined traffic classes. A queue is reserved for each class, and traffic belonging to a class is directed to the queue for that class. Low latency queuing (LLQ) brings strict priority queuing to CBWFQ. Strict priority queuing allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic. This course describes the queuing architecture: traffic-policing, traffic-shaping, CBWFQ, and LLQ. Cisco AutoQoS represents two technologies, Cisco AutoQoS VoIP and Cisco AutoQoS for the Enterprise. These technologies simplify network administration challenges, reducing quality of service (QoS) complexity, deployment time, and cost in enterprise networks. Cisco AutoQoS VoIP incorporates value-added intelligence in Cisco IOS Software and Cisco Catalyst software to provision and manage large-scale QoS deployments. It provides QoS provisioning for individual routers and switches, simplifying deployment and reducing human error. Cisco AutoQoS VoIP offers straightforward capabilities to automate VoIP deployments for customers who want to deploy IP telephony but who lack the expertise and staffing to plan and deploy IP QoS and IP services. Cisco AutoQoS for the Enterprise is a process in which two intelligent mechanisms are deployed to detect voice, video, and data traffic in Cisco networks. The mechanisms generate best-practice QoS policies and apply those policies to WAN interfaces. This course explores the capabilities, requirements, and configuration of Cisco AutoQoS VoIP and Cisco AutoQoS for the Enterprise.

Learning Objectives
  • Identify the features of aggregation and queuing
  • Recognize the purposes of, and differences between, traffic shaping and traffic policing
  • Recognize how to configure and monitor class-based policing
  • Recognize how to configure and monitor class-based shaping
  • Recognize how to configure and calculate bandwidth for LLQ
  • Identify the features of Cisco AutoQoS VoIP
  • Recognize how to configure and monitor Cisco AutoQoS VoIP
  • Recognize how Cisco AutoQoS for the Enterprise is configured and monitored
  • Configure AutoQoS VoIP
Register Now
Implementing Cisco Voice Communications and QoS (CVOICE) Part 2 e-learning bundle
  • Course ID:
  • Duration:
    15 hours
  • Price: