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Description:

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

Course Price
Cisco CVOICE 8.0: Introduction to Voice Gateways $74.95
Cisco CVOICE 8.0: Voice Port Implementation, Codecs, and DSPs $74.95
Cisco CVOICE 8.0: Transporting Voice over IP networks $74.95
Cisco CVOICE 8.0: SIP and MGCP Signaling Protocols $74.95
Cisco CVOICE 8.0: VoIP Call Legs $74.95
Cisco CVOICE 8.0: Cisco Unified Communications Manager Express $74.95

Bundle Price: $219.00
Total Savings: $230.70


Cisco CVOICE 8.0: Introduction to Voice Gateways

Cisco voice gateway relays high quality voice and fax traffic across an IP network. This course describes the operational modes of a voice gateway and how it fits in the Cisco Unified Communications architecture. It explains the voice gateway functions in each Cisco Unified Communications deployment model and the call legs that are associated with each operational mode. A primary function of the Cisco Unified Communications gateways is to route calls. The process of call routing includes the processing of incoming and outgoing call legs. This course also describes how call legs are created when inbound and outbound dial peers are matched. It provides details about the dial-peer matching process and explains the direct inward dialing (DID) feature.
  • recognize the components of Cisco Unified Communications architecture
  • identify the function of voice gateways
  • recognize the major roles in Cisco Unified Communications networks
  • identify the role of gateways in four supported Cisco Unified Communications deployment models
  • recognize the different Cisco voice gateway platforms
  • identify the call legs that are created by a voice gateway in each operational mode
  • recognize how gateways route calls end to end
  • recognize how to configure POTS dial peers
  • match dial peer matching commands with their descriptions
  • recognize how routers work with inbound, outbound, and default dial peers
  • identify the steps in the process of two-stage dialing
  • identify the steps in the process of one-stage dialing

Cisco CVOICE 8.0: Voice Port Implementation, Codecs, and DSPs

Connecting voice devices to a network infrastructure requires an in-depth understanding of the signaling and characteristics that are specific to each type of interface. Digital trunks are used to connect to the public switched telephone network (PSTN), to a PBX, or to the WAN, and are widely available worldwide. This course maps out analog and digital interfaces; examines analog voice ports, analog signaling, and configuration parameters for analog voice ports; and explains how to implement and verify digital trunks. The course also explains the compression schemes that you can use to transport voice using various coder-decoders (codecs), and the implications of these compression schemes on bandwidth usage. How to calculate the amount of bandwidth that a VoIP call will consume is also explained. Finally, the course discusses the digital signal processors (DSPs) that convert analog and digital voice signals into VoIP traffic.
  • recognize how the various types of analog and digital voice port interfaces are used in enterprise scenarios
  • identify the characteristics of analog voice ports
  • recognize how to configure analog voice ports
  • recognize the features of T1 CAS
  • identify the features of ISDN
  • recognize how to configure T1 and E1 trunks to the PSTN
  • identify the steps in configuring ISDN PRI and BRI trunks
  • identify how to fine-tune the analog and digital voice ports
  • identify how echo is generated in a telephone conversation and how the echo cancellation feature works
  • identify the commands used to verify analog and digital voice port configuration
  • configure voice ports
  • match the major voice codecs with their features
  • recognize how voice quality evaluation methods are applied with voice codecs
  • recognize how the packet rate and protocol overhead impacts the total per-call bandwidth
  • identify the functions of digital signal processors
  • distinguish between DSP modules
  • recognize the recommended codec choice in the various gateway deployment models
  • recognize how to configure a DSP for voice termination at a voice gateway
  • identify the commands used to verify DSPs

Cisco CVOICE 8.0: Transporting Voice over IP networks

H.323 gateways are among the most common Cisco IOS voice gateways within Cisco Unified Communications Manager environments. H.323 gateways are the endpoints on a LAN that provide real-time, two-way communications between H.323 terminals on the LAN and other ITU-T terminals on the network. H.323 gateways can also communicate with other H.323 gateways. Gateways enable H.323 terminals to communicate with terminals that are not H.323 terminals by converting protocols. Gateways are the point where a circuit-switched call is encoded and repackaged into IP packets. Because gateways function as H.323 endpoints, they provide admission control, address lookup and translation, and accounting services. The inherent characteristics of a converged voice and data IP network present certain challenges to network engineers and administrators in delivering voice traffic correctly. This course describes the challenges of integrating a voice and data network and explains the technologies that enable voice media transmission.
  • recognize how voice is transported over IP networks end-to-end
  • recognize the steps in the process of analog-to-digital voice conversion using PCM
  • recognize the considerations for VoIP packetization
  • identify the characteristics of the four protocols used for media transmission in an IP network
  • identify conditions for transporting VoIP through firewalls
  • recognize the process of suppressing silence to conserve per-call bandwidth
  • identify the characteristics of the H.323 signaling protocol architecture
  • identify how H.323 establishes and terminates calls
  • recognize the steps in the process of codec negotiation in an H.323 environment
  • recognize how to configure H.323 gateways
  • recognize how to customize H.323 gateways

Cisco CVOICE 8.0: SIP and MGCP Signaling Protocols

Session Initiation Protocol (SIP) is one of the most important voice signaling protocols within service provider VoIP networks and is supported by most IP telephony system vendors. As such, it is an ideal protocol for interconnecting different VoIP systems and networks. An understanding of the features and functions of SIP components, and the relationships that the components establish with each other, is important in implementing a scalable, resilient, and secure SIP environment. This course describes how to configure SIP. It explores the features and functions of the SIP environment, including its components, how these components interact, and how to accommodate scalability and survivability. The Media Gateway Control Protocol (MGCP) enables the remote control and management of voice and data communications devices at the edge of multiservice IP packet networks. Because of its centralized architecture, MGCP overcomes the distributed configuration and administration problems inherent in the use of protocols such as H.323. This course describes how to configure MGCP on a gateway, and the features and functions of the MGCP environment.
  • recognize the features and functions of the Session Initiation Protocol (SIP)
  • recognize the steps involved in common SIP call flows and addressing
  • recognize the process of codec negotiation in SIP
  • recognize how to configure basic SIP functionality on voice gateways
  • identify the commands used to configure the SIP ISDN calling name display feature
  • identify the commands used to block or substitute the caller ID in SIP ISDN
  • match the SIP commands used to configure secure signaling and secure media with their descriptions
  • recognize how to tune SIP options
  • distinguish between the commands used to monitor and verify a SIP gateway operation
  • recognize the components and features of the MGCP architecture
  • identify the steps involved in setting up and tearing down calls in the MCGP call process
  • recognize the process of codec negotiation and digit collection in MGCP
  • match the MGCP commands used for basic MGCP configuration on the voice gateways with their descriptions
  • recognize how to customize and verify MGCP settings on a voice gateway

Cisco CVOICE 8.0: VoIP Call Legs

The inherent characteristics of a converged voice and data IP network create challenges for network engineers and administrators in delivering voice traffic. This course describes the challenges of integrating a voice and data network and offers solutions for designing a VoIP network for optimal voice quality, fax, and modem transmission. Successful implementation of a VoIP network depends upon the correct application of dial peers, the digits that the dial peers match, and the services that they specify. This course provides you with a knowledge of dial-peer configuration options and their uses.
  • match factors that affect audio clarity in IP networks to descriptions
  • identify the requirements for QoS to ensure proper VoIP transmission
  • recognize the challenges of transporting fax and modem calls over IP networks
  • identify the key features of pass-through and relay techniques for fax and modem transport
  • identify how T.38 and pass-through are supported by H.323, SIP, and MGCP
  • recognize how DTMF relay is supported in MGCP, H.323, and SIP environments
  • describe how to configure VoIP dial peers
  • recognize how to configure DTMF relay
  • recognize how to configure fax/modem pass-through and relay
  • recognize how to configure a single codec or codec negotiation on an SIP and H.323 gateway
  • identify how to limit the number of concurrent calls on a VoIP dial peer
  • configure VoIP Dial peers and select codecs

Cisco CVOICE 8.0: Cisco Unified Communications Manager Express

Cisco Unified Communications Manager Express provides call processing for Cisco Unified IP phones for small-office or branch-office environments. It enables the large portfolio of Cisco integrated services routers to deliver unified communications features that are commonly used by business users to meet the voice and video communications requirements of the small or medium-sized office. Cisco Unified Communications Manager Express allows the deployment of a cost-effective, highly-reliable communications system using a single device with Cisco IOS Software. This course introduces the key features and functionality of Cisco Unified Communications Manager Express and explains what is required to deploy it on Cisco IOS routers. It is important to be able to distinguish between various Cisco Unified Communications enduser devices that you may encounter during the course of deploying and administering a Cisco Unified Communications network. In addition, understanding the boot and registration communication between a Cisco Unified IP phone and the Cisco Unified Communications Manager Express is critical for understanding normal voice network operations and for troubleshooting. To this end, this course also introduces the endpoints that are supported by Cisco Unified Communications Manager Express and describes their features.
  • recognize the functions of Cisco Unified Communications Manager Express
  • categorize examples of Cisco Unified Communications Manager Express features by type
  • recognize the supported platforms and the required memory, licensing, and software needed to deploy Cisco Unified Communications Manager Express
  • recognize how Cisco Unified Communications Manager Express operates with calls to and from PSTN and through an IP network
  • identify the capabilities of the Cisco Unified Communications Manager Express 8.0 SCCP and SIP endpoints
  • recognize the steps in the startup process of Cisco Unified IP phones
  • match options to power endpoints with descriptions
  • recognize how to implement VLANs for separation of voice from data traffic
  • identify the steps to configure access and trunk ports for voice VLANs using Cisco IOS
  • identify how to assign addresses to Cisco Unified IP phones
  • recognize how to configure NTP
  • recognize how the Cisco Unified IP phones obtain their configuration and firmware image
  • recognize how to configure system-level parameters in an SCCP environment
  • identify the steps to set up system-level parameters in a SIP environment
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Implementing Cisco Voice Communications and QoS (CVOICE) Part 1 e-learning bundle
  • Course ID:
    252726
  • Duration:
    16 hours
  • Price:
    $219