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Cisco CVOICE 8.0: Voice Port Implementation, Codecs, and DSPs
Overview/Description
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.
Target Audience
The target audience for this course is established IT professionals with a solid, existing background in Cisco and networking technologies.
Expected Duration (hours)
3.5
Lesson Objectivesrecognize 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
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.
Target Audience
The target audience for this course is established IT professionals with a solid, existing background in Cisco and networking technologies.
Expected Duration (hours)
3.5
Lesson Objectives
Cisco CVOICE 8.0: Voice Port Implementation, Codecs, and DSPs
Trajanje:
3,5 h
Šifra:
cc_voic_a02_it_enus
Katalog: