CLCOR Multiple Select — 20 Questions

Question 1

Which THREE of the following factors impact the quality of video calls in a network?

Accepted Answer

Packet loss. Packet loss (A) is correct because dropped RTP/UDP packets in a video stream cause visible artifacts, freezing, and audio gaps, directly degrading call quality. Jitter (C) is correct because variation in packet inter-arrival times disrupts the playout buffer, causing choppy audio and stuttering video even when average delay is acceptable. Latency (E) is correct because high end-to-end delay (typically above 150 ms one-way) impairs interactivity and, combined with jitter and loss, degrades the real-time experience. Interface description (B) is only a text label on an interface and has no effect on traffic handling, and router uptime (D) is an availability metric that does not by itself determine video quality as long as the device is forwarding packets.

Answer

Interface description

Answer

Router uptime

Question 2

Which THREE actions are possible using SIP profiles on a CUBE?

Accepted Answer

Modify SIP headers. SIP profiles on Cisco Unified Border Element (CUBE) are used to manipulate SIP signaling by inspecting and altering SIP messages, so option A (Modify SIP headers) is correct because rules within a SIP profile can rewrite existing header values such as From, To, or Diversion. Option B (Add SIP headers) is correct because SIP profiles support the 'request' and 'response' header-insertion rules that inject new headers into inbound or outbound SIP messages. Option C (Remove SIP headers) is correct because SIP profiles can strip unwanted headers via header-removal rules, which is common for privacy or interop scenarios. Option D (Change voice codec) is not a SIP-profile function; codec negotiation is handled by dial peers (codec preference lists, 'codec' commands) and SDP, not by SIP header manipulation. Option E (Route calls to PSTN) is also not a SIP-profile action; call routing is determined by dial-peer matching and translation rules, while SIP profiles only affect SIP message content.

Answer

Change voice codec

Answer

Route calls to PSTN

Question 3

A collaboration engineer is deploying Cisco Webex Hybrid Message Service with Cisco Unified Communications Manager (CUCM) and Expressway-C. The requirement is to enable secure instant messaging and presence between on-premises CUCM users and Webex. Which two components are required for this deployment? (Choose two.)

Accepted Answer

Cisco Unified Communications Manager IM and Presence Service. Hybrid Message Service requires the Expressway-C with the Hybrid Message Service connector and the on-premises IM and Presence Service. The connector bridges the on-premises IM and Presence with Webex, enabling secure instant messaging and presence federation. Other components like Webex Meetings Server, VCS, or CUBE are not part of this architecture.

Answer

Cisco Webex Meetings Server

Answer

Cisco Unified Border Element (CUBE)

Answer

Cisco TelePresence Video Communication Server (VCS)

Question 4

Which THREE network conditions contribute to poor VoIP call quality in an RTP stream?

Accepted Answer

Latency. Latency (A) is correct because one-way delay above roughly 150 ms degrades RTP voice interactivity, causing talk-over and echo that the jitter buffer cannot fix. Jitter (C) is correct because variable inter-packet arrival times force the jitter buffer to either underrun (clipping) or grow (added delay), directly harming voice quality. Packet Loss (D) is correct because dropped RTP packets cannot be retransmitted in real time, so codecs conceal the gaps, producing robotic or choppy audio. High MTU (B) is not inherently harmful—fragmentation is the real risk—and Fast switching (E) is a LAN forwarding behavior, not a WAN impairment affecting RTP quality.

Answer

High MTU

Answer

Fast switching

Question 5

A network designer is implementing QoS for a Cisco Collaboration deployment. Which two statements about the trust boundary in a Cisco QoS design are true? (Choose two.)

Accepted Answer

Cisco IP Phones are typically trusted to mark voice traffic correctly, so the switch port should trust DSCP.. The trust boundary should be as close to the source as possible, and Cisco IP Phones can be trusted to mark voice and signaling traffic correctly. This ensures that QoS markings are preserved end-to-end and that other devices trust them. Placing the boundary at the core or trusting all PC traffic is not recommended.

Answer

All traffic from a PC connected to an IP Phone should be trusted by default.

Answer

The trust boundary should be placed at the core layer to centralize QoS marking.

Answer

The trust boundary is only necessary for voice media, not for call signaling.

Question 6

Which TWO tasks must be performed when setting up Webex Hybrid Calendar service with Microsoft Exchange?

Accepted Answer

Grant the service account ApplicationImpersonation rights. Option D is correct because the Webex Hybrid Calendar service connects to Microsoft Exchange on-premises via Exchange Web Services (EWS) using a dedicated service account, and that account must be granted the ApplicationImpersonation management role so it can act on behalf of users to read and manage their calendar data. Option E is correct because the Hybrid Calendar service must be activated and configured in Control Hub (the Webex administrative portal), where you enable the service and supply the Exchange environment details and service account credentials that tie the on-premises deployment to the cloud service. Options A, B, and C are incorrect: no dedicated DNS zone for Exchange is required, Hybrid Calendar does not use a SIP trunk between Exchange and Webex, and no H.323 gateway in the DMZ is involved since the integration relies on HTTPS/EWS rather than telephony signaling.

Answer

Create a dedicated DNS zone for Exchange

Answer

Configure a SIP trunk between Exchange and Webex

Answer

Install an H.323 gateway in the DMZ

Question 7

Which TWO of the following are functions performed by the CUBE as a Session Border Controller?

Accepted Answer

Media interworking. Option B (Media interworking) is correct because a CUBE acting as a Session Border Controller terminates the signaling and media on both legs, allowing it to transcode between different codecs (e.g., G.711 to G.729) and perform media flow-around or flow-through, which is a core SBC function. Option E (Topology hiding) is correct because CUBE uses back-to-back user agent (B2BUA) behavior to strip and rewrite SIP headers such as Via, Record-Route, and Contact, concealing the internal CUCM/ITSP topology from the far-end network. Option A (Call routing in CUCM) is not an SBC function; dial plan and call routing decisions are performed by Cisco Unified Communications Manager, not CUBE. Option C (IP phone registration) is handled by CUCM's TFTP and SCCP/SIP registration services, not by CUBE. Option D (Voice mail storage) is provided by Cisco Unity Connection, not by CUBE.

Answer

Call routing in CUCM

Answer

IP phone registration

Answer

Voice mail storage

Question 8

A collaboration engineer is troubleshooting poor video quality on a Cisco Collaboration deployment. The engineer suspects that video traffic is not receiving appropriate QoS treatment and wants to verify the classification and marking on a Cisco IOS router. Which two actions should the engineer take to confirm that video traffic is being handled correctly? (Choose two.)

Accepted Answer

Use the show class-map command to confirm the match criteria and verify the video class is referencing the correct DSCP or ACL.. Verifying QoS for video requires checking both the classification logic and the applied policy. The show class-map command confirms the match criteria used to identify video, while show policy-map interface reveals per-class match counts and drops on the interface. Together they confirm that video is classified correctly and treated as intended.

Answer

Use the show call active voice brief command to inspect the codec negotiated for video calls.

Answer

Run the show interfaces command to check the interface error counters for video drops.

Answer

Run the debug ip rsvp command to observe reservation messages for video flows.

Question 9

Which TWO of the following are valid methods to handle digit manipulation in CUCM?

Accepted Answer

Translation Pattern. Translation Pattern (C) is a valid digit-manipulation method in CUCM because it matches a dialed digit string in a partition and applies transformations such as discarding digits (Discard Digits), prefixing digits, or rewriting the called/calling number before the call is routed. Calling Party Transformation Pattern (E) is also valid because it specifically manipulates the calling-party number (and related presentation settings) on inbound or outbound calls by matching the calling number and applying prefix, strip, or transformation masks. Route Filter (A) is not a digit-manipulation tool; it is used with route lists and time schedules to filter which route groups or dial peers can be selected based on dial-peer tags, not to rewrite digits. Region Matrix (B) governs codec selection between regions for calls, not digit manipulation. Location Configuration (D) is used for call admission control and bandwidth allocation between locations, not for altering dialed digits.

Answer

Route Filter

Answer

Region Matrix

Answer

Location Configuration

Question 10

Which TWO of the following are valid methods for providing high availability for IP phones in a CUCM design?

Accepted Answer

SRST deployment. Option C (SRST deployment) is correct because Survivable Remote Site Telephony lets remote-site IP phones register with a local router/gateway when WAN connectivity to the CUCM cluster is lost, keeping basic call processing available and thus providing high availability for those phones. Option E (CUCM Group configuration) is correct because a CUCM Group defines an ordered list of CUCM nodes that a device pool's phones use for registration; if the primary subscriber fails, phones fail over to the next CUCM node in the group, which is the standard CUCM redundancy mechanism. Option A is not a high-availability method for phone registration: multiple TFTP servers in Option 150 only provide redundancy for configuration file download, not for call processing/registration. Option B is invalid because DHCP cannot provide IP address redundancy for phones; DHCP only leases addresses and does not offer a failover address mechanism for the phone itself. Option D is not a CUCM phone high-availability method; CUCM uses a primary/secondary/tertiary node failover model via CUCM Groups and device pools, not round-robin load balancing of phone registrations.

Answer

Configuring multiple TFTP servers in Option 150

Answer

IP address redundancy via DHCP

Answer

Load balancing via round-robin

Question 11

Which TWO features are provided by the Cisco Webex App when registered as a Softphone?

Accepted Answer

SIP signaling. The Cisco Webex App, when registered as a softphone, functions as a SIP endpoint, so it performs SIP signaling (Option B) to set up, manage, and tear down calls with the call control platform. It also handles RTP media transmission (Option D), sending and receiving the actual audio/video packets for the call. These two functions are core to any softphone client, which must both signal and carry media. Options A, C, and E are incorrect because a softphone does not host TFTP configuration files, act as a DNS root server, or provide DHCP address assignment; those are infrastructure services typically provided by dedicated servers or network devices.

Answer

TFTP service hosting

Answer

DNS root server

Answer

DHCP server functionality

Question 12

Which THREE of the following are valid reasons to use a Media Termination Point (MTP)?

Accepted Answer

Bridging different signaling protocols. Option B is correct because an MTP can bridge different signaling protocols by terminating the signaling and media on each side, allowing calls to interconnect between otherwise incompatible protocol variants. Option C is correct because MTPs enable media anchoring, which is required for features such as call recording, lawful intercept, and certain supplementary services that need the media stream to pass through a controlled point. Option E is correct because MTPs provide DTMF interworking, converting out-of-band DTMF (for example, SIP INFO or H.323) into in-band or RFC 2833 events so that endpoints using different DTMF transport methods can communicate. Option A is not correct because transcoding between G.711 and G.729 is the function of a transcoder (hardware or software DSP resource), not an MTP. Option D is not correct because an MTP does not increase voice bandwidth; it terminates and relays media, and any bandwidth impact would typically be neutral or negative, not an increase.

Answer

Providing transcoding for G.711 to G.729

Answer

Increasing voice bandwidth

Question 13

Which THREE items are required to successfully register a room device to Webex?

Accepted Answer

A valid Webex Device license. Option A is correct because registering a room device to Webex requires a valid Webex Device license assigned to the device in Control Hub, which authorizes the device to use Webex cloud services. Option D is correct because the device must be onboarded using a valid activation code generated from Control Hub, which the device uses to authenticate and complete registration. Option E is correct because the device needs an active internet connection to reach Webex cloud endpoints and complete the cloud registration process. Options B and C are not required: a local SIP trunk and an existing CUCM registration are on-premises telephony concepts and are not prerequisites for Webex cloud device registration.

Answer

A local SIP trunk

Answer

An existing CUCM registration

Question 14

Which THREE of the following are necessary to ensure successful NTP deployment in a CUCM cluster?

Accepted Answer

Configure external Stratum 1 or 2 servers. Option B is correct because CUCM requires an external NTP source of good quality, and Stratum 1 or 2 servers provide the accurate, reliable time reference needed to keep all cluster nodes synchronized. Option C is correct because NTP communicates over UDP port 123, so any firewall between CUCM and the NTP servers must permit UDP 123 traffic or synchronization will fail. Option E is correct because every node in the CUCM cluster must point to the same consistent NTP configuration so that all servers share a common time source and avoid clock drift that could break database replication and certificate validation. Option A is not required because CUCM uses unicast NTP to reach specified servers, not broadcast mode. Option D is not required because enabling NTP on a local voice gateway is unrelated to CUCM cluster time synchronization.

Answer

Use broadcast mode

Answer

Enable NTP on the local voice gateway

Question 15

A collaboration engineer is deploying a new Cisco Webex Room Kit Pro in a conference room. The device is registered to Cisco Unified Communications Manager (CUCM) using SIP. The engineer needs to ensure that the device can place calls to external numbers and that the dialed digits are correctly processed. Which two configuration elements must be defined on CUCM to enable external dialing from this endpoint? (Choose two.)

Accepted Answer

Route Pattern. To enable external dialing from a Cisco Webex Room Kit Pro registered to CUCM, you must configure a Route Pattern to match the dialed digits and route the call, and a Gateway or Route List to provide the physical connection to the external network. The Route Pattern directs the call to the Route List, which then selects a Gateway. Without these, the call cannot leave the CUCM cluster.

Answer

SIP Profile with Outbound Proxy

Answer

Device Pool with Region settings

Answer

Translation Pattern

Question 16

Which TWO of the following are common audio codecs used in Cisco collaboration?

Accepted Answer

G.711. G.711 (option B) is a correct answer because it is a standard ITU-T narrowband audio codec (64 kbps PCM, using either A-law or mu-law companding) that is universally supported in Cisco collaboration endpoints and gateways for toll-quality voice. G.729 (option C) is also correct because it is a widely used ITU-T compressed audio codec (8 kbps CS-ACELP) commonly deployed in Cisco voice and collaboration environments to conserve bandwidth on WAN links. Option A (JPEG) is incorrect because it is a still-image compression standard, not an audio codec. Option D (H.264) is incorrect because it is a video compression standard used for video streams, not audio. Option E (MP3) is incorrect because, although it is an audio codec, it is a consumer music format and is not a standard codec used in Cisco collaboration call signaling/media negotiation.

Answer

JPEG

Answer

H.264

Answer

MP3

Question 17

Which THREE of the following are recommended steps when designing a secure signaling path between CUCM and a CUBE gateway?

Accepted Answer

Install and trust CA certificates. Option A is correct because securing the signaling path with TLS requires that both CUCM and CUBE trust each other's certificates, which means installing and trusting the CA certificates (or the specific device certificates) that signed the endpoints' certificates. Option B is correct because a SIP Trunk Security Profile on CUCM defines the security-related settings for the trunk, including device security mode (encrypted/authenticated), incoming/outgoing transport (TLS), and digest authentication, and it must be applied to the SIP trunk toward CUBE. Option D is correct because enabling TLS on the trunk encrypts SIP signaling between CUCM and CUBE, protecting the signaling path from eavesdropping and tampering. Option C is not a recommended step for securing the signaling path; digest authentication is a separate authentication mechanism and disabling it does not secure signaling, and in many designs it is left enabled or configured per policy. Option E is incorrect because H.323 Fast Start is an H.323 feature and is irrelevant to a SIP-based signaling path between CUCM and CUBE.

Answer

Disable SIP digest authentication

Answer

Enable H.323 Fast Start

Question 18

Which THREE types of media resources can be managed by an MRGL?

Accepted Answer

Transcoder. A Media Resource Group List (MRGL) in Cisco Unified Communications Manager is used to prioritize and group media resources, and the three media resource types it can manage are transcoders (B), conference bridges (D), and Music on Hold (MoH) servers (E). Transcoder (B) is correct because MRGLs assign transcoders to devices for codec conversion between different codecs (e.g., G.711 and G.729). Conference Bridge (D) is correct because MRGLs control which conference bridge resources are used for ad hoc and Meet-Me conferences. Music on Hold (MoH) Server (E) is correct because MRGLs determine which MoH server(s) provide audio streams to callers on hold. Cisco Unity Connection (A) is a voicemail system, not a media resource managed by MRGLs, and SIP Trunk (C) is a signaling/transport interface, not a media resource type in an MRGL.

Answer

Cisco Unity Connection.

Answer

SIP Trunk.

Question 19

A collaboration engineer is designing a Cisco Unified Communications Manager (CUCM) cluster for a company with 5000 users. The cluster will have one publisher and three subscribers. The engineer must ensure that the design supports high availability and that IP phones can register and make calls even if one subscriber fails. Which two statements are true regarding CUCM redundancy and device registration? (Choose two.)

Accepted Answer

Each IP phone can be configured with a prioritized list of up to three CUCM servers for registration using a CUCM Group.. CUCM Groups provide a prioritized list of up to three CUCM servers for device registration. When a subscriber fails, phones automatically re-register to the next available server in their CUCM Group. The publisher should not be included in CUCM Groups for call processing to avoid overloading it. CUCM Groups are assigned via Device Pools, not directly to phones.

Answer

The publisher must be included in the CUCM Group to ensure that phones can register if all subscribers fail.

Answer

The maximum number of CUCM servers in a CUCM Group is five, allowing for greater redundancy.

Answer

CUCM Groups are assigned to phones directly on each phone configuration page, not via Device Pool.

Question 20

Which TWO of the following fields are required when defining a Route Pattern?

Accepted Answer

Gateway/Route List. In Cisco Unified Communications Manager, a Route Pattern must specify the destination to match and where to send the matched calls, so option A (Gateway/Route List) is required to define the egress path for the pattern, and option E (Route Pattern String) is required to define the dialed digits or wildcard pattern that the route pattern matches. Options B (Codec), C (Bandwidth Limit), and D (Softkey Template) are not mandatory fields when creating a route pattern; codec and bandwidth are typically associated with regions/locations or device pools, and softkey templates apply to devices such as phones, not to route patterns themselves.

Answer

Codec

Answer

Bandwidth Limit

Answer

Softkey Template