350-801 practice questions 231–240

1. Question 231 · 450 in bank

   During a conference call, the CTO of a company adds an external user to the call. The CTO leaves at the end of the call, but the external user stays on and dials out to a premium rate number. Several weeks later, the CTO receives an unexpectedly nigh phone bill. The CTO decides to implement an Ad Hoc conference restriction. Which parameters must be adjusted?

   • ADrop Ad Hoc Conference
   • BLimit Ad Hoc Conference
   • CRestrict Ad Hoc Conference
   • DMonitor Ad Hoc Conference

   Show answer

   Answer: A

   Explanation

   Ad hoc conference calls can be dropped when the originator hangs up. This is achieved by setting the Drop Ad Hoc Conference service parameter to When Conference Controller Leaves under Clusterwide Parameters > Feature > Conference. This ensures that the other parties (such as external users) cannot initiate a call to another external number.

   Source: Adapted from community discussion — a3c4c84

   References

   • Cisco documentation

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2. Question 232 · 450 in bank

   

   Refer to the exhibit. An engineer deploys this Cisco Unified Border Element configuration, but the phone at extension 2221 fails to ring. These configurations were performed already in Cisco UCM:

   

   • phone configuration and registration • SIP trunk configuration • translation pattern configuration Which action resolves the issue?

   • AApply voice translation-rule 3 to dial-peer 11 voip.
   • BChange the destination pattern for dial-peer voice 10 to 2... $.
   • CApply voice translation-rule 3 to dial-peer 10 voip.
   • DReplace voice translation-rule 3 with voice translation-rule 1.

   Show answer

   Answer: A

   Explanation

   A is presumably the correct answer, but you can't actually apply a translation-rule to a dial-peer. You have to apply a translation-PROFILE, which then references the translation-rule. i.e. it needs "translation-profile incoming profile1" added to dial-peer 11. Presumably that's what they mean with A here.

   Source: Adapted from community discussion — sneff91

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3. Question 233 · 450 in bank

   How does an administrator ensure compliance with ITU-T G.114 latency requirements when deploying voice and video applications over routes that include a geostationary satellite?

   • AFor voice and video applications over satellite, Cisco recommends using WAN optimization compression, which will save space or time of the transmission. The latency will then be below the recommended latency threshold for voice.
   • BUsing G.729 voice codec Instead of G.711 voice codec will reduce the packet size and the latency to stay within the recommended threshold.
   • CThe satellite is 35,786 km (22,236 miles) above sea level (from the equator), so an engineer will not get below 150 ms latency in one direction and will need to manage the expectation of the user.
   • DThe application will work because it will be below the recommenced latency requirement If an engineer uses compressed headers to speed up the voice packets.

   Show answer

   Answer: C

   Explanation

   " For satellite circuits, the expected latency can be in the range of 250 to 900 ms. For example, signals being relayed via geostationary satellites will need to be sent to an altitude of 35,786 km (22,236 miles) above sea level (from the equator) out into space and then back to Earth again. There is nothing an administrator can do to decrease latency in such scenarios because they can do nothing about increasing the speed of light or radio waves. All that can be done to address the effect of latency in these scenarios is to educate the user-base so that realistic performance expectations are set. " C

   Source: Adapted from community discussion — pizsi

   References

   • Cisco documentation

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4. Question 234 · 450 in bank

   How does WRED prevent congestion?

   • AIt marks which packets to drop randomly.
   • BIt drops packets selectively based on the IP precedence value.
   • CIt marks which packets to drop based on how much traffic the client sent.
   • DIt drops packets that have a higher DSCP value last.

   Show answer

   Answer: B

   Explanation

   Answer B About Weighted Random Early Detection Random Early Detection (RED) is a congestion avoidance mechanism that takes advantage of the congestion control mechanism of TCP. By randomly dropping packets prior to periods of high congestion, RED tells the packet source to decrease its transmission rate. WRED drops packets selectively based on IP precedence. Edge routers assign IP precedences to packets as they enter the network. (WRED is useful on any output interface where you expect to have congestion. However, WRED is usually used in the core routers of a network, rather than at the edge.) WRED uses these precedences to determine how it treats different types of traffic. https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/qos_conavd/configuration/xe-16/qos-conavd-xe-16-book/qos-conavd-cfg-wred.html

   Source: Adapted from community discussion — gorzad

   References

   • Cisco documentation
   • Cisco documentation
   • Cisco documentation
   • Cisco documentation

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5. Question 235 · 450 in bank

   Select 2

   Which two media encryptions should be used when configuring SRTP between Cisco UCM and Cisco VCS Expressway? (Choose two.)

   • Aoptional
   • Bpass-through
   • Cpriority
   • Dmandatory
   • Enone

   Show answer

   Answer: DE

   Explanation

   Wording stuff: question looks not asking to do SRTP, but to configure.In the related link https://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-vcs/118877-config-vcs-00.html#anc12 , there is only None, Mandatory and Best Effort. So only apply what matched the answers. Typical Cisco questions ;-)

   Source: Adapted from community discussion — Gabcac

   References

   • Cisco documentation

   Next question

6. Question 236 · 450 in bank

   

   Refer to the exhibit. An engineer configured NTP on Cisco UCM and runs the utils ntp status command from Cisco UCM CLI and receives an error message. How is this issue mitigated and fixed?

   • AUse an external NTP server and allow communication on port 53.
   • BEnsure that Cisco UCM can communicate with the NTP server on port 53 and have a Windows NTP server configured with NTPv3.
   • CConfigure the NTP server to answer requests on port 443.
   • DEnsure that Cisco UCM can communicate with the NTP server on port 123 and have a Linux NTP server configured with NTPv4.

   Show answer

   Answer: D

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7. Question 237 · 450 in bank

   When designing location-based CAC, how much overprovisioning is calculated for video bandwidth?

   • A10%
   • B15%
   • C20%
   • D25%

   Show answer

   Answer: C

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8. Question 238 · 450 in bank

   Location-based call admission control is used in Cisco UCM to control the amount of bandwidth allowed between predefined locations. What must be considered when calculating the number of calls that can be made?

   • AA limit of 900 kb is the total amount of bandwidth used for the sum of the media and signaling in the calls. The system allows for an actual of 20% overhead of IP packet headers. This is to compare the use of ISDN lines with no IP overhead.
   • BA limit of 900 kb Is the total amount of bandwidth used for the sum of calls of the defined codec including headers and media traffic.
   • CA limit of 900 kb is the total amount of bandwidth used for the sum of calls. The use of compression does not impact the amount of calls because only the codec definition is considered when calculating the CAC in Cisco UCM.
   • DA limit of 900 kb is the total amount of actual used RTP traffic so the oversubscription can be done for multiple calls adding Its maximum RTP stream requirements.

   Show answer

   Answer: B

   Explanation

   The correct answer is B. When calculating the number of calls that can be made using location-based call admission control (CAC) in Cisco UCM, it is important to consider the amount of bandwidth used by the calls. The amount of bandwidth used by a call is the sum of the media traffic and signaling traffic, including any IP packet headers. The limit of bandwidth allowed is typically defined as a total amount for all calls, and the number of calls that can be made is calculated by dividing the total amount of available bandwidth by the amount of bandwidth used by each call. Therefore, option B is the correct answer

   Source: Adapted from community discussion — Testme1235

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9. Question 239 · 450 in bank

   Select 2

   What are two functions of a dial-plan? (Choose two.)

   • Acall recording
   • Bendpoint features
   • Ccalling privileges
   • Ddigit manipulation
   • Ering cadence

   Show answer

   Answer: CD

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10. Question 240 · 450 in bank

    Which DSCP class selector is necessary to mark scavenger traffic?

    • AAF21
    • BCS1
    • CAF11
    • DCS2

    Show answer

    Answer: B

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350-801 practice, question 231. 450 questions available.