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BCOC 1-13 > D02X > Flashcards

D02X Flashcards

0
Q

D12

Define convergence as it applies to communications networks in accordance with Newton’s Telecom dictionary

A

Convergence is the combination of voice, video, and data services across the same network infrastructure.

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1
Q

D12

Define Voice over Internet Protocol (VoIP)

A

Voice over Internet Protocol (VoIP) is a technology which allows Internet Protocol-based networks to be utilized for real- time voice applications.

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4
Q

D12

Identify the benefits of Voice over Internet Protocol (VOIP)

A

Cost
Savings
Bandwidth
Rich Applications and Features

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5
Q

D12

Identify the four basic components of a Voice over Internet Protocol (VoIP) network

A

The IP network (Infrastructure)
Call processor/controllers
Media/signaling gateways
Subscriber terminals/Endpoints

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6
Q

D12

Identify the two types of call processing used within a Voice over Internet Protocol (VoIP) network

A

Centralized - a Cisco Call Manager is established at one central location.
Distributed - either a Cisco Call Manager or Voice-enabled router running Cisco Call Manager Express is established at each separate site.

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7
Q

D12

Given illustrated models of physical network topologies, match a Voice over Internet Protocol (VoIP) employment method (physical topology) to its corresponding definition

A

Know these pictures:

  1. IP Trunking
  2. Integration of a POTS long local on an IP network
  3. IP-enabled PBX
  4. IP Telephony
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8
Q

D13

Identify in writing the functions of a call processor

A

The call processor/controller is the device which runs the call processing software within a VoIP network. This includes call processing, device control, directory service, computer telephony integration (CTI), trunking, and gateway support.

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9
Q

D13

Identify the purpose of a Cisco Unified Communications Manger (CUCM) in relation to Voice over Internet Protocol (VoIP) networks

A

Cisco Call Manager is an application which runs on a dedicated server providing call processing services for up to 10,000 users.

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10
Q

D13

Define Cluster in relation to Voice over Internet Protocol (VoIP) networks

A

Group of 8 call managers - max 30,000 users/cluster

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11
Q

D13

Identify the function of a Publisher Server in relation to Voice over Internet Protocol (VoIP) networks

A

The Publisher server is the only server whose database can be altered.

  • No matter how many Cisco Call Managers are in a cluster (max of 8), the first server is always the Publisher server.
  • Changes to a cluster can only be made when the Publisher is online.
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12
Q

D13

Identify the function of a Subscriber Server in relation to Voice over Internet Protocol (VoIP) networks

A

Subscriber servers are the primary call controllers responsible for registering phones and setting up calls.

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13
Q

D13

Identify the purpose of an Inter-Cluster Trunk in relation to Voice over Internet Protocol (VoIP) networks

A

Communication between clusters is accomplished through programming Inter-Cluster Trunks (ICT).

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14
Q

D13

Identify the four main components of Call-Routing in a Cisco Unified Communications Manager (CUCM) network

A

Route pattern - matches a string of dialed digits.

Route list - an ordered list of route groups.

Route Group - a list of devices (gateways or trunks) that are configured to support circuits to other networks, either the PSTN or another CUCM cluster.

Gateways and Trunks - devices that physically terminate and support circuits to the PSTN, to digital or analog PBXs, and to IP WAN circuits leading to remote clusters.

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15
Q

D13

Identify in writing the purpose of a Cisco Unified Communications Manger Express (CME) in relation to Voice over Internet Protocol (VoIP) networks

A

CME is a VoIP management application which runs on an Integrated Services Router’s (ISR) Internetworking Operating System (IOS) IOT provide call processing in a VoIP network.

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16
Q

D13

Identify the purpose of Dial Peers

A

Dial peers are a software structure that binds a dialed digit string to a voice port or IP address of the destination network.

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17
Q

D13

Identify the power options associated with Cisco IP phones

A

Power over Ethernet

Power over Ethernet (PoE) is codified by the IEEE as standard 802.3AF. PoE switches provide 15.4 Watts over all four pairs on Cat 5 and Cat 6 cable. This electricity is used to power the device which is connected to the switch; in this case, a VoIP telephone set.

*PoE Switches are required in our networks because VoIP telephones are not fielded to Marine Corps units with external power cords.

18
Q

D13

Identify the characteristics of the Voice VLAN

A

Increased performance - By reducing the size of the broadcast domain, network devices run more efficiently.

Improved manageability - The division of the network into logical groups of users, applications, or servers allows you to understand and manage the network better

Physical topology independence - VLANs allow you to group users regardless of their physical location in a network.

Increased security - A VLAN boundary marks the end of a logical subnet.

19
Q

D13

Define the role of the TFTP Server in IP networking

A

A TFTP server is a simple file store that serves files to unauthenticated clients on demand.

In the case of VoIP, the TFTP server serves as a central repository for all required phone configurations in the network.

20
Q

D13

Identify the Voice over Internet Protocol (VoIP) Telephone Initialization Process

A
  1. The VoIP telephone is connected to the switch and receives Power over Ethernet.
  2. Using Cisco Discovery Protocol (CDP), the network switch identifies which VLAN the telephone should be on and delivers the Voice VLAN.
  3. The VoIP Phone requests an IP from the DHCP Server (If not set as a static IP. If statically assigned, skip to step #4.
  4. The VoIP Phone contacts the TFTP server for the configuration file and loads it.
  5. The VoIP Phone registers with the CME or Cisco Call Manager using Skinny Client Control Protocol (SCCP) and is now able to place and receive calls.
21
Q

D14

Identify the purpose of Signaling Protocols in relation to Voice over Internet Protocol (VoIP) networks

A

VoIP signaling protocols exist to setup and maintain a bi- directional real-time media stream between VoIP subscriber terminals. They also provide the necessary signaling for VoIP networks to interface with the PSTN.

22
Q

D14

Define the H.323 protocol as it pertains to VoIP networks

A

H.323 is a suite of protocols (e.g. call signaling, audio/video codecs etc) that provide IP networks with traditional telephony functionality.

23
Q

D14

Define the Media Gateway Control Protocol (MGCP) as it pertains to VOIP networks

A

MGCP is a client-server call control protocol used by call- control devices such as Cisco Call Manager to manage and control IP gateways.

Packet switched network routers that have a voice port or Voice WAN Interface Card (VWIC) add on are called voice enabled routers or gateways. They have the capability to receive a PRI T-1 from a CSN switchboard and convert it to packet switching format. In other words, a voice port on a PSN router provides a physical interface for a PRI T-1 on a packet switched network.

*Call Manager controlling the T1 - most common way to connect a T-1 from a circuit switching network to a VoIP network in the Marine Corps.

24
Q

D14

Identify the purpose of a Cisco Router Voice Port

A

Packet switched network routers that have a voice port or Voice WAN Interface Card (VWIC) add on are called voice enabled routers or gateways. They have the capability to receive a PRI T-1 from a CSN switchboard and convert it to packet switching format. In other words, a voice port on a PSN router provides a physical interface for a PRI T-1 on a packet switched network.

*Call manager controlling the T-1 - most common way to connect a T-1 from a circuit switching network to a VoIP network in the Marine Corps.

25
Q

D14

Define the Session Initiation Protocol (SIP) as it pertains to VoIP networks

A

SIP is an ASCII-based peer-to-peer protocol in which user agents initiate sessions similar to the method utilized by H.323. It defines end to end call signaling between devices by using a text based HTTP-like model of communication.

*Cisco Call Manager does not control devices running SIP, and those devices will not register with Cisco Call Manager.

26
Q

D14

Define the Skinny Station Control Protocol (SCCP) as it pertains to VoIP networks

A

SCCP is a Cisco proprietary protocol designed to be utilized between the call manager device and the subscriber terminals.

27
Q

D14

Define Real-time Transport Protocol (RTP) as it pertains to VoIP networks

A

RTP is based off of the User Datagram Protocol (UDP) which was created to provide end-to-end transmission of streaming media traffic.

*RTP is used to communicate directly between the voice endpoints themselves.

28
Q

D14

Describe in writing the Basic Phone-to-Phone Call Process within a Cisco Unified Communications Manager (CUCM) administered network

A
  1. The caller dials the distant end telephone number and the digits are received by the Call Manager.
  2. The Call Manager performs a look-up of the distant end telephone’s IP address.
  3. The Call Manager then checks the status of the distant end telephone to verify that it is registered and is able to receive calls.
  4. If available to receive a call, the Call Manager rings the telephone.
  5. Alerted by the ringing, the distant end user takes their phone off-hook.
  6. The Call Manager then connects the telephone sets using Realtime Transport Protocol (RTP). The telephones communicate with one another for the duration of the call without the use of the Call Manager.
29
Q

D15

Define transcoding

A

Transcoding is the direct digital-to-digital conversion from one codec to another.

Transcoding compresses and decompresses voice streams to match endpoint device capabilities. Transcoding is commonly used when an incoming voice stream is digitized and compressed (by means of a codec) to save bandwidth, but the local site does not support the codec.

30
Q

D15

Define Quality of Service (QoS) as it relates to VOIP networks

A

Quality of Service is the ability of the network to provide better or special services to a set of users and applications at the expense of other users and applications.

31
Q

D15

Identify the components of QoS within VOIP networks

A
  1. Classification and Marking
  2. Queuing - the congestion management function of QoS.
    2a. Weighted Fair Queuing (WFQ) - tries to balance available bandwidth among all senders. This is often the default method applied to serial interfaces.
    2b. Class-Based Weighted Fair Queuing (CBWFQ) - allows you to specify guaranteed amounts of traffic for the various classes of traffic. For example, you could specify 20% for Voice, 20% for Video, and 30% for all other traffic.
    2c. Low Latency Queuing (LLQ) - often referred to as PQ-CBWFQ, because it is the same exact thing as CBWFQ, but adds a priority queuing (PQ) component.
32
Q

D15

Define Call Admission Control (CAC) as it relates to IP networks

A

Call Admission Control (CAC) prevents IP calls from being extended across the WAN link, if the additional bandwidth required would exceed the CAC-allocated bandwidth for concurrent calls, thus preventing oversubscription.

33
Q

D16

Define Class of Control as it applies to VoIP networks

A

Class of Control (CoC), otherwise known as Class of Service, (CoS) or Class of Restriction (CoR), is defined as the ability to apply calling restrictions to devices.

34
Q

D16

Identify the purpose of Survivable Remote Site Telephony (SRST)

A

Fault tolerance within a Cisco Call Manager administered VoIP network can be provided through the use of Survivable Remote Site Telephony (SRST). SRST is a capability which can be enabled throughout a VoIP network to allow calls to be processed and routed between sites in the event that connectivity to the Cisco Call Manager is lost.

35
Q

D16

Identify the purpose of the Disaster Recovery System (DRS)

A

The Disaster Recovery System (DRS) provides a backup (with scheduler) and restore capability.

Backups can be written to a local DLT tape drive or to an SFTP server (most common in the Marine Corps). A scheduler is provided for automated backups, or an immediate start to the backup can be selected. Individual server or full cluster backups may be performed.

36
Q

D16

Identify the purpose of the Real Time Monitoring Tool (RTMT)

A

The Cisco Unified Real-Time Monitoring Tool (RTMT) allows administrators to collect, view, interpret, and monitor the various counters, trace files, and logs generated by CUCM.

37
Q

D17

Match the correct VoIP Network symbol to its definition

A

Cisco Unified Communications Manager (CUCM) or Call Manager. Denote whether the Call Manager is a Publisher or Subscriber. Ensure the IP address is included. The IP address will come from your server VLAN Publisher 192.168.1.37

Voice-Enabled Router. This symbol is commonly used to show a router that is performing the function of a media gateway.

IP Telephony-Enabled Router. This symbol is used to show Call Manager Express.

38
Q

D17

5 More FEX ELOs…

A

TL; DR

BCOC 1-13 (9 decks)

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