Voip Fundamentals

Front 1

What is the role of gatekeeper in an IP network.

Back 1

Performs address translation and bandwidth management.

Front 2

VOIP requires: end-to-end QoS, low delay, min. jitter, echo cancellation, & low packet loss. Which network devices should be deployed to acheive the "best" voice performance?

Back 2

Layer 3 switches at network core with DiffServ enabled. Layer 2 switches at network edge with 802.1p CoS enabled.

Front 3

In H.323 network the gatekeeper and the gateway work together to manage the interface between the VOIP network and PSTN. Which two functions does the gatekeeper perform?

Back 3

1)Manages call routing
2)Handles the dialing plan

Front 4

Given:
*IP clients located on private network 10.168.254.0/24 using many to one NAT.
*H.323 gateway located on public network 207.46.197.102 at another location
*Customer network between IP clients and H.323 gateway is verified by ping

There is one-way speech path when calling from IP client to H.323 gateway. What is most likely cause?

Back 4

IP clients on the private network use private addresses mapped to a single public address. The media path from the public address cannot reach the IP clients with the private address.

Front 5

Which two network devices should be used to avoid network congestion when implementing VOIP on LAN.

Back 5

Layer 2 and 3 switches

Front 6

Routers and Switches prioritize VOIP packets ahead of IP data packets end-to-end. Why is this feature important?

Back 6

It controls jitter

Front 7

In a Layer 3 IP LAN/WAN, which method provides the best control for managing applications with different types of QoS behaviors?

Back 7

DiffServ (Differentiated Services)

Front 8

In LAN with multiple H.323 endpoints deployed, which component is responsible for call control, media access and bandwidth management between?

Back 8

The H.323 gateway

Front 9

What is the function of the H.323 gatekeeper when a call is placed from an H.323 endpoint to another H.323 endpoint?

Back 9

It maps the destination telephone number to the destination endpoint IP address.

Front 10

Which two CODECs have the least delay (processing or algorithmic)?

Back 10

G.711 & G.726

Front 11

Real-Time-Protocol (RTP) provides for support of applications like voice and video. Which three attributes of RTP support real-time applications?

Back 11

1)Timestamping
2)Packet sequencing
3)Payload identification

Front 12

Which CODEC is most tolerant to network delay?

Back 12

G.711

Front 13

What are two differences between IP, Asynchronous Transfer Mode (ATM) and Frame Relay (FR)?

Back 13

1)ATM has fixed cell size, but IP and FR do not.
2)IP is a Layer 3 protocol, but ATM and FR are Layer 2 protocols.

Front 14

A video conference is set up using UDP(User Datagram Protocol) between two gigabit Ethernet networked locations. Why should UDP be used instead of Transmission Control Protocol (TCP)?

Back 14

UDP serves as an efficient transport for handling real-time application traffic.

Front 15

Which transport protocol should be used to meet the real-time requirement for VOIP? Why?

Back 15

UDP
It ignores lost packets.

Front 16

How does Real-Time Control Protocol (RTCP), the control protocol of RTP, assist RTP in handling packetized voice in an IP telephone environment?

Back 16

Identifies sources and provides QoS feedback.

Front 17

Which CODEC delivers the greatest compression?

Back 17

G.723.1

Front 18

Assuming that a Permanent Virtual Circuit (PVC) is available to address converged voice and data.

To achieve the QoS necessary to deliver voice between two points on a Frame Relay network, Which two items are required to guarantee that voice quality is maintained?

Back 18

1)WAN Access device that shapes traffic
2)Committed Information Rte (CIR) large enough to address the total peak voice traffic plus a portion allocated for Best-Effort (BE) data

Front 19

What is the "ideal" Maximum Transmission Unit (MTU) for VOIP on a 64 kbps link?

Back 19

Set the MTU to 80 bytes

Front 20

An Ethernet switch in a path for VOIP traffic appears to be the cause of poor voice quality. Which two configuration parameters, know to cause problems for VOIP, should be checked on the Ethernet switch?

Back 20

1)Duplex setting
2)Autonegotiation

Front 21

A tech has identified that congestion appears to be cause of poor voice quality. Which two actions could be taken to address this issue?

Back 21

1)Increase bandwidth
2)Apply QoS methods

Front 22

A fixed 128 kbps microwave-wireless data connection links an offshore oil platform and land station on the coast. The land station links into a FR circuit back to headquarters.

You determine that 400 ms delay budget for all VOIP calls on the WAN is acceptable for this connection. Which area(s) should you consider for improving the overall delay?

Back 22

Decrease the WAN Maximum Transmission Unit (MTU), increase the wireless bandwidth, and change the FR circuit to Point-to-Point Protocol (PPP)

Front 23

Poor voice quality is being caused by excessive delay.
Given the following info:
*WAN is a 56 kbps link
*MTU on WAN is 1,024 bytes

What is the minimum WAN bandwidth required to get the delay introduced from the WAN to be dropped down to under 100 ms without changing the MTU?

Back 23

Increase WAN bandwidth to 128 kbps

Front 24

Excessive LAN collisions appear to be the cause of poor voice quality. Which action should be taken to correct the issue?

Back 24

Ensure that Layer 2 switching is applied at the endpoints.

Front 25

In Session Initiation Protocol (SIP), signaling allows call information to be carried across network boundaries. What does session management provide?

Back 25

The ability to control the attributes of an end-to-end call.

Front 26

Which two H.323 devices have CODECs which may be included in the speech path after a VOIP call setup?

Back 26

1)Terminal
2)Gateway

Front 27

When call setup is established over TCP in an IP based H.323 network, which protocol defines the call-signaling method (direct or gatekeeper-routed) that is decided by a gatekeeper during Registration, Admission, and Status (RAS) operations?

Back 27

H.225

Front 28

A network includes some equipment that uses Megaco signaling. There is need to ensure that the VOIP gateways are Megaco compliant. When configuring the VOIP gateways you see a signaling option with four choices, but Megaco is not one of them. Which one of the four options should you select to make the products inter operate?

Back 28

H.248

Front 29

Between the following two VOIP gateways:
*Gateway A uses H.323
*Gateway B uses H.248

Which statement(s) best describes the benefits of using H.248 over H.323?

Back 29

H.248 uses a stimulus-based signaling model getting intelligence and features from servers, not gateways. This allows for lower cost gateways with more features.

Front 30

H.323 and SIP standards can be extended to offer vendor-specific functionality, but differ significantly in method. In dealing with a multi-vendor network, what is the benefit of using SIP rather than H.323?

Back 30

Allows for standards-based extensions to perform specific functions.

Front 31

What are the three basic functions of SIP that enable it to support multimedia sessions for IP telephone without the assistance of other IP telephony protocols?

Back 31

1)Feature negotiation
2)Call-management changing
3)Feature changes of a session while it is in progress.

[Basic functions of SIP:*Name translation and user location *Feature negotiation *Call/Session participant mgmt *Call Feature Changes

Front 32

In comparison with the H.323 protocol, why is debugging an IP telephony call easier using SIP?

Back 32

Its signaling is modeled after the Hypertext Transfer Protocol (HTTP).

Front 33

When assessing a customer's data network for VOIP, which two factors are critical in determining consistent voice quality?

Back 33

1)Packet Loss
2)Peak delay times and jitter over a give link

Front 34

Before beginning a network assessment of a customer's data network for VOIP, why is it important to understand the customer's perception of good or acceptable voice quality?

Back 34

It will affect the CODEC selected for the network.

Front 35

The following VOIP network assesment has been completed:
*Estimation of VOIP traffic
*Assessment of LAN/WAN resources
*Capacity is available for VOIP

Which step should you perform next?

Back 35

Measure the network ability to provide QoS.

Front 36

During 30-day assessment of a network. Which tools are best suited for gathering the necessary data to ensure a successful VOIP deployment without requiring an engineer on-site for the duration of the testing?

Back 36

NetIQ Chariot

Front 37

Which tool is used to measure maximum jitter?

Back 37

NetIQ Chariot

Front 38

A geographically dispersed company with a mobile work force wants to quickly deploy Internet Telephones. Which device in the company LAN/WAN network can enable a simplified implementation of Internet Telephones?

Back 38

Dynamic Host Configuration Protocol (DHCP) Server

Front 39

During the first phase of a network assessment you discover the customer forces all switch ports to 100 Mbps full duplex. The customer plans to deploy an IP-enabled Private Branch Exchange (PBX) and Internet Telephones. Which network recommendation should you propose to the customer?

Back 39

Require all VOIP components to autonegotiate.

Front 40

LAN contains mix of Layer 2 and Layer 3 switches. To ensure the best QoS as VOIP is implemented, which two QoS methods should you recommend?

Back 40

1)802.1q
2)DiffServ (Differentiated Services)

Front 41

Customer wants to achieve the most efficient use of bandwidth over their 128 kbps WAN Links. The top priority is the best voice quality to meet call requirements (two concurrent calls) and network capacity. Which CODEC best meets the network requirements?

Back 41

G.729AB

Front 42

In a VOIP network, which three items are impacted by the customer's data traffic?

Back 42

1)echo
2)latency
3)packet loss

Front 43

Customer wants to implement packet fragmentation for VOIP on its WAN circuits that are less than 1 Mbps, to minimize jitter and reduce queuing delay. Which two methods could be used to implement packet fragmentation on the company's 256 kbps WAN?

Back 43

1)Frame Realy FRF.12
2)IP Fragmentation (Maximum Transmission Unit [MTU] size adjustment)

Front 44

Customer wants to implement VOIP in their LAN/WAN environment. What are the lower boundary, benchmark figures suited to determine good voice quality (satisfied) in the Mean Opinion Score (MOS) and G.107 E-Model (R Value)?

Back 44

MOS=4 and R=80

Front 45

Which queuing technique: 1)classifies and places packets into queues according to the information in the packets and 2)services packets from queues with the most important information first when congestion occurs?

Back 45

Priority Queuing

Front 46

Which two statements accurately describe the G.107 E-Model?

Back 46

1)The scale is typically from 50 to 94
2)It is the numerical average of the voice quality ratings (scores) from all listeners

Front 47

Which two factor into VOIP delay, jitter, and packet loss on a QoS-enabled IP network?

Back 47

1)CODEC selection
2)Packetization rate selection

Front 48

Customer has recently installed a PBX on their ATM WAN (AAL-5). They now want to establish voice connectivity between it and an existing PBX at another site across the WAN.
Requirements:
*There is a one-way delay of 135 ms
*The voice packet payload is 30 ms
*3 Mbps will be allocated for voice calls *Voice Activity Detection (VAD) is disabled.
*Fifty simultaneous voice calls must be handled.

Which CODEC should you use on the media gateway to provide the best possible voice quality?

Back 48

G.729

Front 49

Given the following customer network info:
*A fractional T1/E1 (384 kbps) is running between two sites *The data traffic peak usage is 80 kbps
*There are 20 Internet Telephones at each site with five call expected to be traversing the T1/E1 WAN link and any one time
*Test across the WAN T1/E1 link shows a one-way delay of 125 ms
*Due to the network architecture calls from the Internet Telephone through the VOIP gateways are incurring transcoding

When Internet Telephones communicate with the VOIP gateway for calls transversing the WAN router, what is the recommended CODEC for them to operate efficiently within the alloted WAN bandwidth and still provide good voice quality?

Back 49

G.711

Front 50

Which three voice packetization parameters should you use to determine bandwidth requirements for a VOIP solution?

Back 50

1)CODECS
2)Jitter buffer
3)Voice sample packet size

Front 51

Customer wants to connect a VOIP gateway from an IP-enabled PBX to its existing Ethernet Layer 2 switch. Internet Telephones will be directly attached to the Ethernet Layer 2 switch. In this situation using only Layer 2 technology, which two QoS methods should achieve the best voice quality using QoS prioritization?

Back 51

1)Port-based prioritization
2)Traffic separation using VLANs

Front 52

Customer with a low-speed, data-network link between two locations wants to pass two on current VOIP calls between locations. Their top priority is the best possible voice quality to meet call requirements and network capacity. Typically, which CODEC should best meet their network requirements?

Back 52

G.729A

Front 53

Which queuing technique services the queues using a round-robin approach to prevent any one packet source from overusing its share of network capacity?

Back 53

Fair Queuing

Front 54

Customer is running voice and data traffic between two sites over a WAN. They are planning to connect VOIP gateways in IP-enabled PBX to existing Ethernet Layer 3 switches at each site. In this situation, which QoS method should achieve the best voice quality using QoS prioritization?

Back 54

DiffServ (Differentiated Services)

Front 55

Which statement best describes the operation used by an echo canceler to reduce echo in digital circuits?

Back 55

It calculates an estimate of what the echo will be, and then subtract this from the actual returned signal.