CLCOR Practice Test 8 — 10 Questions

Question 1

You are designing a solution for secure voice signaling. Which protocol provides encryption for SIP signaling messages in a CUCM environment?

Accepted Answer

TLS. TLS (Transport Layer Security) is the protocol used to encrypt SIP signaling messages in a CUCM environment. SIP over TLS (SIPS) provides encryption and authentication for signaling, ensuring that messages like INVITE, BYE, and REGISTER are secure. CUCM supports TLS for SIP trunking and line-side signaling.

Answer

IPsec

Answer

SRTP

Answer

SSH

Question 2

Which protocol is used by IP phones to register with the Cisco Unified Communications Manager?

Accepted Answer

SCCP or SIP. Cisco IP phones register with Cisco Unified Communications Manager using either SCCP (Skinny Client Control Protocol) or SIP (Session Initiation Protocol), depending on the phone model and firmware configuration. Both are supported signaling protocols for phone registration and call control in CUCM.

Answer

RTCP

Answer

MGCP

Answer

H.323

Question 3

When configuring Hybrid Call Service for Cisco Unified Communications Manager, which specific requirement must be met for the Expressway-C and Expressway-E pair?

Accepted Answer

They must be configured with a Unified Communications traversal zone.. For Hybrid Call Service with Cisco Unified Communications Manager, the Expressway-C and Expressway-E pair must be configured with a Unified Communications traversal zone. This zone enables Mobile and Remote Access (MRA) by allowing signaling and media to traverse the firewall securely between the internal CUCM and external endpoints.

Answer

They must use the same certificate authority as the CUCM nodes.

Answer

They must use a public IP address on the Expressway-C.

Answer

They must be in the same subnet.

Question 4

Which option allows a customer to keep their existing PSTN numbers while moving to Webex Calling?

Accepted Answer

Number porting.. Number porting is the regulated process (governed by FCC/local authorities) that transfers an existing PSTN number from one carrier to another while preserving the number itself. When migrating to Webex Calling, the customer submits a port request with a Letter of Authorization (LOA) and recent bill so Cisco's carrier partner can pull the DIDs from the incumbent provider. This is the only option that actually moves ownership of the number rather than just redirecting or labeling it.

Answer

Number aliasing.

Answer

Number forwarding.

Answer

Number provisioning.

Question 5

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 6

A user's Cisco IP Phone 8841 is registered to Cisco Unified Communications Manager but cannot place calls to any internal or external numbers. The phone displays 'Registration Complete' but every call attempt results in a fast busy tone. Which configuration element is most likely missing or misconfigured for this phone?

Accepted Answer

The phone's Calling Search Space does not include the partitions containing the dialed numbers.. A registered phone that cannot place any calls typically has a Calling Search Space that lacks access to the partitions containing the dialed numbers. The CSS must include the partitions where directory numbers and route patterns reside. Without this, digit analysis fails and the phone receives a fast busy tone. Other settings like Region or security profile do not block call placement in this manner.

Answer

The phone's Device Security Profile is set to Non-Secure.

Answer

The phone's Network Locale is set to United States.

Answer

The phone's Device Pool is not assigned to a Region.

Question 7

When configuring CUBE for media flow-around, which command ensures the media goes directly between endpoints?

Accepted Answer

media flow-around. The 'media flow-around' command in CUBE configuration tells the Cisco Unified Border Element to leave the RTP media streams out of the signaling path, allowing endpoints to exchange media directly. This is the correct syntax for enabling media flow-around on CUBE. It is configured under the voice service voip or dial-peer context to control whether media is anchored or bypassed.

Answer

media pass-through

Answer

media-bypass

Answer

media flow-through

Question 8

What is the function of 'Cisco Unified Reporting' in a CUCM environment?

Accepted Answer

To generate cluster and system reports.. Cisco Unified Reporting is a CUCM administration interface that generates on-demand reports about the cluster, such as device counts, license usage, dial plan summaries, and system health snapshots. It is designed for reporting and troubleshooting, not real-time monitoring or configuration. It pulls data from the CUCM database and presents it in downloadable formats like PDF or CSV.

Answer

To monitor network bandwidth.

Answer

To manage real-time call logs.

Answer

To perform bulk configuration.

Question 9

You are troubleshooting a scenario where G.711 is being used instead of the expected Opus codec between two endpoints in the same region. What is the most likely cause in the CUCM configuration?

Accepted Answer

Audio Codec Preference List. The Audio Codec Preference List in CUCM defines the ordered list of codecs an endpoint will offer and accept during SIP/SDP negotiation. If Opus is not placed above G.711 (or is missing entirely) in the assigned codec preference list, the endpoints will fall back to G.711 even when both support Opus. Since both endpoints are in the same region, no location-based bandwidth constraint would force G.711, so the codec preference list is the most likely culprit.

Answer

Phone NTP settings

Answer

SIP Trunk Security Profile mismatch

Answer

Location-based bandwidth limit

Question 10

When using 'RSVP' for Call Admission Control (CAC), what happens if the available bandwidth on the path is less than the requested amount?

Accepted Answer

The call is blocked. With RSVP-based Call Admission Control, when a call's RSVP PATH/RESV signaling requests more bandwidth than is available on any link along the path, the router rejects the reservation and the call is blocked (denied). RSVP CAC is a deterministic admission control mechanism: if the requested bandwidth cannot be reserved, the call setup fails rather than proceeding with degraded quality. This ensures that admitted calls receive their guaranteed quality of service.

Answer

The call is placed but with a lower quality codec

Answer

The call is put on hold until bandwidth is available

Answer

The call proceeds, but marks the packets with Best Effort