Learn About Implementing Cisco IOS Unified Communications

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These questions are based on 640-460: Implementing Cisco IOS Unified Communications
A Self Test Software Practice Test

Objective: Describe the components of the Cisco Unified Communications Architecture.
Sub-objective: Describe the function of endpoints in a UC environment.

Single answer, multiple-choice

Which protocol is supported by Cisco IP phones and Cisco gateways operating at the endpoints layer of the Cisco Unified Communications system?

A.    RTP
B.    MGCP
C.    SCCP
D.    G.711

Answer:
C

Tutorial:
Skinny Call Control Protocol (SCCP) is a Cisco protocol used for communication between IP phones or gateways and Cisco Unified Communications Manager. It follows the server/client model, in which the main control and processing is done by the Unified Communications Manager, which is the central call-control unit. Any device, software or application that connects to the network and directly interacts with the end user will come under the endpoints layer. Other products that function as endpoints are wireless phones, video phones, analog phones, cell phones, PCs and instant messenger clients.

There are four layers to the Cisco Unified Communications system: the infrastructure layer, the endpoints layer, the applications layer and the call processing layer. All of these layers and their functionalities complete a VoIP structure.

Real-Time Transport Protocol (RTP) is a standardized packet format for delivering real-time traffic such as audio and video.

Media Gateway Control Protocol (MGCP) is used for communication with gateways. It is not used for communicating with IP phones.

G.711 is a codec used in Cisco Unified Communications. It is not a protocol.

References:

Cisco > Support > Unified Communications Architecture Basics

CCNA Voice Official Exam Certification Guide (Cisco Press, ISBN 1-58720-207-7), Chapter 7: Gateway and Trunk Concepts – Trunking CME to Other VoIP Systems

Objective: Describe PSTN components and technologies.
Sub-objective: Describe time division and statistical multiplexing.

Single answer, multiple-choice

Which technique is used by the Public Switched Telephone Network (PSTN) for transmitting multiple digitized data, voice and video signals simultaneously over one communications media?

A.    Frequency Division Multiplexing (FDM).
B.    Time Division Multiplexing (TDM).
C.    Code Division Multiple Access (CDMA).
D.    Wavelength Division Multiplex (WDM).

Answer:
B

Tutorial:
Time Division Multiplexing (TDM) is the correct option. In TDM, the bandwidth of a link is divided into separate channels with fixed time slots. Multiple bit streams are transmitted through these channels with time slots and are recollected at the destination. TDM can be defined as total available time divided into several users. The best example of TDM is the telephone system.

Frequency Division Multiplexing (FDM) is an incorrect option because in FDM, carrier bandwidth is divided into sub-channels of different frequency widths, each carrying a signal at the same time in parallel. FDM is used in analog transmissions such as twisted pair telephone line, cable access, cellular, radio and TV communications.

Code Division Multiple Access (CDMA) is an incorrect option because this technique is used in wireless technology and digital communication. CDMA uses code division in which the transmitter is given a code to transmit multiple data simultaneously.

Wavelength Division Multiplex (WDM) is an incorrect option. WDM works on the same principles as FDM, but WDM applies to digitized wavelengths of light in optical fiber.

References:

Cisco > Introduction to DWDM Technology > Introducing DWDM > Time-Division Multiplexing

CCNA Voice Official Exam Certification Guide (Cisco Press, ISBN 1-58720-207-7), Chapter 1: Perspectives on Voice Before Convergence – Sending Multiple Calls over a Single Line

Objective: Describe VoIP components and technologies.
Sub-objective: Describe the process of voice packetization.

Single answer, multiple-choice

What are the two components of a Real-Time Transport Protocol (RTP) media stream?

A.    TCP and UDP.
B.    UDP and Data.
C.    RTCP and Data.
D.    RTCP and UDP.

Answer:
C

Tutorial:
Real-Time Transport Control Protocol (RTCP) and data are the two components of an RTP media stream. Real-Time Transport Protocol (RTP) is a standardized packet format for delivering real-time traffic, such as audio and video. RTCP provides control information for RTP streams in packaging and delivering multimedia but does not transfer any data itself.

While sending data or completing a call, there are two parts of a RTP stream:

•    Data: Delivers real-time data and real-time properties for applications. It also helps in loss detection and identification of stream contents.
•    RTCP: Provides media control and quality of service (QoS) feedback. It also helps in identifying the participants in the RTP session.

Since phone conversations of any type are bidirectional, two streams are required to complete a call.

TCP and UDP are incorrect. Transmission Control Protocol (TCP) is one of the two transport protocols of the IP suite. TCP ensures sequential, reliable delivery of data from one program running on a computer to another program on another computer. User Datagram Protocol (UDP) is the other transport protocol of the IP suite. UDP is used to send short messages or datagrams from one computer to another in a networked environment on a connectionless (non-guaranteed) basis. It is the underlying transport protocol that encapsulates RTP, but RTP runs on top of UDP. Neither is part of the RTP stream itself.

References:

Cisco > Articles > Network Technology > General Networking > VoIP: An In-Depth Analysis

Cisco > Internetworking Technology Handbook > Voice/Data Integration Technologies > VoIP Network Design Constraints

Cisco > Products & Services > Cisco IOS Software > Cisco IOS Technologies > Quality of Service (QoS) > Network Based Application Recognition (NBAR) > Product Literature > White Papers > nBAR RTP Payload Classification > RTP Overview

CCNA Voice Official Exam Certification Guide (Cisco Press, ISBN 1-58720-207-7), Chapter 7: Gateway and Trunk Concepts – Understanding RTP and RTCP

 
Objective: Describe and configure gateways, voice ports and dial peers to connect to the PSTN and service provider networks.
Sub-objective: Describe and configure voice dial peers.

Single answer, multiple-choice

Which is NOT an option of the dtmf-relay command?

A.    sip-notify.
B.    rtp-nte.
C.    info-notify.
D.    sub-notify.

Answer:
C

Tutorial:
The info-notify option is not a dtmf-relay command option. Dual-Tone Multifrequency (DTMF) dialing consists of simultaneous voice-band tones generated when a button is pressed on a telephone. The use of DTMF signaling enables support for advanced telephony services. For devices that utilize these features but may not support media connections, the DTMF Events through SIP Signaling feature provides this signaling capability.

The Media Termination Point (MTP) or transcoding module on a gateway detects RFC 2833 (DTMF) packets from an IP endpoint and can be instructed to generate and send out-of-band signal events to the call manager or to pass the packets through to the other IP endpoint, which is the default behavior. This is accomplished with the dtmf-relay command.

There are four options for the dtmf-relay command:

•    info
•    rtp-nte
•    sip-notify
•    sub-notify

The dtmf-relay info command sets the Session Initiation Protocol (SIP) Dual Tone Multifrequency (DTMF) relay mechanism by using Info messages to relay outgoing DTMF signals from Cisco Unity Express to the Cisco IOS SIP gateway.

The dtmf-relay rtp-nte command sets the SIP DTMF relay mechanism to use the media path to relay incoming and outgoing DTMF signals to Cisco Unity Express.

The dtmf-relay sip-notify command sets the SIP DTMF relay mechanism to use Unsolicited-Notify messages to relay incoming and outgoing DTMF signals.

The dtmf-relay sub-notify command sets the SIP DTMF relay mechanism to use Subscribe and Notify messages to relay incoming DTMF signals to Cisco Unity Express.

The other options are incorrect because rtp-nte, sip-notify and sub-notify are all valid options of the dtmf-relay command.

Reference:

Cisco > Cisco Unity Express 3.1 Command Reference – D [Cisco Unity Express] > dmtf-relay

Objective: Implement Cisco Unified Communications Manager Express to support endpoints using CLI.
Sub-objective: Describe the requirements and correct settings for DHCP, NTP and TFTP.

Single answer, multiple-choice

What is the function of the configuration file stored on the Trivial File Transfer Protocol (TFTP) server in a Cisco Unified Communications Manager Express (CME) environment?

A.    It provides information about voice VLAN to the IP phone.
B.    It provides an IP address to the IP phone upon request.
C.    It provides firmware images while booting the IP phone.
D.    It provides parameters for connecting the IP phone to Cisco Unified Communications Manager Express.

Answer:
D

Tutorial:
The configuration file provides parameters for connecting the IP phones to Cisco Unified Communications Manager Express. Each Cisco IP phone contacts the TFTP server and downloads a configuration file after receiving the server's address via option 150, which is included in the Dynamic Host Configuration Protocol (DHCP) server's scope options. The configuration file stored on the TFTP server provides parameters to establish connectivity with CME. After receiving the file, the IP phone tries to make a TCP connection to the Cisco Unified CME router.

The configuration file does not provide information about voice VLAN to the IP phone. The switch provides VLAN information to the IP phones, such as the VLAN number.

The configuration file does not provide an IP address to the IP phone upon request. IP addresses are provided by the DHCP server.

The configuration file does not provide firmware images while booting the IP phone. The firmware images are stored inside the non-volatile flash memory.

References:

Cisco > Cisco IP Phone Model 7960, 7940, and 7910 Administration Guide for Cisco CallManager Release 3.0 and 3.1 > Preparing to Install the Cisco IP Phone on Your Network > Requesting the Configuration File

CCNA Voice Official Exam Certification Guide (Cisco Press, ISBN 1-58720-207-7), Chapter 4: Installing Cisco Unified Communications Manager Express – Generated Configuration Files

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