CLCOR Practice Test 2 — 25 Questions

Question 1

An administrator needs to enforce a bandwidth limit for video calls at a specific site in CUCM. Which configuration object should be used?

Accepted Answer

Location. In Cisco Unified CM, the Location object is the configuration element that enforces call admission control (CAC) by defining the maximum audio and video bandwidth available between locations. When a video call is placed, CUCM subtracts the call's bandwidth from the Location's video bandwidth pool, and if the limit is exceeded, the call is denied or rerouted. This makes Location the correct object for enforcing a per-site bandwidth limit for video calls.

Answer

Region

Answer

Partition

Answer

Device Pool

Question 2

What is the primary function of a 'Class Map' in a Cisco IOS QoS configuration?

Accepted Answer

To categorize and match specific traffic flows. A Class Map in Cisco IOS QoS is used to identify and classify traffic by matching criteria such as ACLs, DSCP, CoS, or NBAR protocols. It defines 'what' traffic to match, and is later referenced by a Policy Map that defines the action. This separation of classification (class-map) from action (policy-map) is the foundation of Modular QoS CLI (MQC).

Answer

To define the action taken on traffic

Answer

To set the DSCP value for a packet

Answer

To assign a specific interface for traffic

Question 3

An administrator is troubleshooting a failed Hybrid Calendar setup. The 'Connector' shows as 'Down' in Control Hub. What is the most likely reason?

Accepted Answer

The Expressway-C host cannot reach the cloud over port 443.. The Hybrid Calendar Connector in Control Hub relies on the on-premises Expressway-C (or Cisco Webex Hybrid Services connector) maintaining an outbound TLS connection to Webex cloud endpoints over TCP 443. When that outbound path is blocked by a firewall, proxy, or DNS failure, Control Hub reports the connector status as 'Down' because the heartbeat to the cloud cannot be established. License, user password, and AD group issues would not cause the connector itself to show Down — they would affect calendar processing for individual users, not the connector health state.

Answer

The calendar integration license is invalid.

Answer

The user's password expired.

Answer

The user is not in the correct group in Active Directory.

Question 4

Which log file is most useful for troubleshooting registration issues between an IP phone and CUCM?

Accepted Answer

Cisco CallManager Traces. Cisco CallManager Traces (often called SDI/SDL traces) capture the detailed signaling exchange between the IP phone and CUCM during registration, including SCCP/SIP messages, device state changes, and error codes. These traces are the primary tool for diagnosing why a phone fails to register, as they show the exact point of failure in the registration handshake. Other logs either provide aggregated health metrics or focus on different services (TFTP, Tomcat) that are not directly involved in the registration signaling path.

Answer

System Health Reports

Answer

Cisco TFTP Logs

Answer

Cisco Tomcat Logs

Question 5

A CUBE is receiving calls but rejecting them due to '488 Not Acceptable Here'. What is the most likely reason?

Accepted Answer

Codec mismatch. SIP 488 'Not Acceptable Here' is returned when the called party cannot support the media parameters in the offer — most commonly a codec mismatch where the offered codec (e.g., G.729) is not configured on any matching dial-peer on the CUBE. CUBE rejects the INVITE because it cannot negotiate a compatible codec between the inbound and outbound legs.

Answer

Authentication failure

Answer

SIP registration expired

Answer

Dial-peer inactive

Question 6

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 7

You are designing Quality of Service (QoS) for a Cisco Collaboration deployment. The customer wants to ensure that voice traffic is prioritized over data traffic. Which marking should be applied to voice bearer traffic on Cisco IP Phones and switches?

Accepted Answer

IP DSCP EF (46) and CoS 5.. Voice bearer traffic must be marked with DSCP EF (46) and CoS 5 to ensure it receives priority treatment across the network. This is the Cisco best practice for QoS. The other markings are used for different traffic types, such as video or signaling, and would not provide the required priority for voice.

Answer

IP DSCP AF31 (26) and CoS 3.

Answer

IP DSCP CS3 (24) and CoS 3.

Answer

IP DSCP AF41 (34) and CoS 4.

Question 8

You are deploying a Webex Room device in a hybrid environment. What is the benefit of registering it to Webex instead of on-premises CUCM?

Accepted Answer

It enables full access to Webex cloud features like whiteboarding and AI-driven metrics.. Registering a Webex Room device to Webex (cloud) rather than on-premises CUCM unlocks the full Webex cloud feature set, including Webex Assistant, whiteboarding, AI-driven meeting metrics, and cloud-based management through Control Hub. Cloud registration also simplifies updates and provides native Webex Meetings join experience. On-premises CUCM registration limits the device to CUCM call control features and requires expressway/Collaboration Edge for cloud service access.

Answer

It allows the device to function without an internet connection.

Answer

It reduces the bandwidth required for video calls.

Answer

It forces the device to use on-premises media routing.

Question 9

You are observing that video traffic is marked as DSCP 0. How can you fix this at the network level without changing the endpoint configuration?

Accepted Answer

Use an ACL to match the traffic and a policy-map to re-mark it. When video endpoints send DSCP 0 (best-effort), the marking must be corrected in the network path. A class-map matches the traffic (e.g., by ACL identifying RTP/Video over the correct port range), and a policy-map with a set dscp command re-marks it to the desired value such as AF41 for interactive video. Applying this via MQC (Modular QoS CLI) on the ingress interface fixes the marking without touching the endpoint configuration.

Answer

Disable traffic shaping

Answer

Change the interface speed

Answer

Enable auto-qos on the interface

Question 10

Which deployment model should be chosen for Cloud Connected PSTN (CCP) if the organization wants to leverage their existing carrier contracts?

Accepted Answer

Cloud Connected PSTN (CCP). Cloud Connected PSTN (CCP) lets an organization bring its own PSTN carrier into Webex Calling by peering the carrier's network with the Webex Calling cloud, so existing carrier contracts, rates, and DIDs are preserved. This is the deployment model explicitly designed for 'use my existing carrier' scenarios, as opposed to Cisco-provided PSTN or on-premises gateways. The carrier must be a Webex Calling CCP partner.

Answer

Local Gateway PSTN

Answer

Webex Calling PSTN via Cisco

Answer

Webex Calling Dedicated Instance

Question 11

An engineer needs to verify the current active calls on a Cisco IOS XE voice gateway and see details such as call legs, codecs, and durations. Which command should be used?

Accepted Answer

show call active voice brief. The 'show call active voice brief' command is the correct tool to verify active calls on a Cisco IOS XE voice gateway. It lists each call leg with critical details such as calling/called numbers, codec, and duration, allowing an engineer to quickly assess current call activity. Other commands either focus on configuration or are protocol-specific.

Answer

show sip-ua calls

Answer

show voice call status

Answer

show dial-peer voice summary

Question 12

A collaboration engineer is deploying a new Cisco Webex Room Kit in a conference room. The endpoint will send 1080p video at 3 Mbps. The network administrator needs to ensure that the video traffic is marked with the appropriate DSCP value and receives proper treatment in the QoS policy. Which DSCP value should be used for this interactive video traffic?

Accepted Answer

AF41 (34). The correct DSCP for interactive video is AF41 (34). This marking ensures that video traffic is treated with assured forwarding and low drop precedence, providing a balance between priority and bandwidth guarantees. Voice uses EF, signaling uses CS3, and interactive video uses AF41. This hierarchy ensures that voice gets the highest priority, followed by video, then signaling, then data. Marking video as AF41 allows network devices to apply appropriate queuing and congestion management.

Answer

CS4 (32)

Answer

AF31 (26)

Answer

EF (46)

Question 13

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 14

A network administrator is troubleshooting one-way audio issues between a Cisco IP Phone and a Cisco TelePresence endpoint. The path traverses a router with a configured class-map. Which command would you use to verify the packet markings on the ingress interface?

Accepted Answer

show policy-map interface. The command `show policy-map interface` displays the actual packet markings and QoS statistics applied to an interface by a policy-map. Since the class-map is referenced within a policy-map, this command shows the number of packets matching each class and the marking actions (e.g., DSCP, CoS) that were applied. This is the only option that directly verifies the markings on the ingress interface.

Answer

show class-map configuration

Answer

show ip rtp statistics

Answer

show qos interface

Question 15

A company is implementing Webex Edge Connect. They require a dedicated circuit. Who must they coordinate with to set up the physical connection?

Accepted Answer

A certified Cisco Webex Edge Connect partner.. Webex Edge Connect is a dedicated, private connectivity option between a customer's network and the Webex cloud, delivered via a Cisco-certified partner that has direct peering into Cisco's Edge Connect locations. The physical cross-connect, BGP peering, and circuit turn-up must be coordinated with that certified partner because they own the interconnection into Cisco's data centers. Cisco does not sell the circuit directly, and the customer's local ISP alone cannot provision the private handoff into Webex.

Answer

The local Internet Service Provider only.

Answer

The Webex Control Hub automated provisioning tool.

Answer

Cisco Technical Support

Question 16

You have a cluster with multiple nodes. You want to ensure that phones in the 'New York' office prioritize registration to a 'New York' subscriber. How do you implement this?

Accepted Answer

Create a CM Group with the local subscribers and assign it to the Device Pool.. A Cisco CallManager Group (CM Group) defines an ordered list of subscribers that devices in an associated Device Pool will register to. By creating a CM Group containing the New York subscribers and assigning it to the Device Pool used by New York phones, those phones will prefer the local subscriber for registration, keeping signaling local and reducing WAN dependency. This is the standard CUCM mechanism for call processing redundancy and locality.

Answer

Use a Route Group to point to the subscriber.

Answer

Assign the phones to a specific Region.

Answer

Configure the TFTP server address in the phone.

Question 17

Which component in the Cisco Collaboration architecture provides the initial configuration files (XML) to an IP phone upon registration?

Accepted Answer

TFTP Service. The Cisco TFTP service, running on Cisco Unified Communications Manager, is responsible for serving XML configuration files, phone firmware, and other provisioning files to IP phones during registration. When a phone boots, it requests its SEP<MAC>.cnf.xml file from the TFTP server, which contains its directory number, device pool, and cluster settings. Without TFTP, phones cannot complete registration.

Answer

CallManager Service

Answer

CTI Manager

Answer

Webex Control Hub

Question 18

You notice high latency on calls placed through a Local Gateway. Which metric should you inspect in the Webex Control Hub to confirm the issue is with the network path?

Accepted Answer

Network latency and jitter.. Network latency and jitter are key metrics that measure the quality of the network path. High latency indicates delay in packet delivery, while jitter measures variation in delay, both of which directly affect voice quality. Inspecting these metrics in Webex Control Hub can confirm if the issue is with the network path.

Answer

Calling license usage.

Answer

User status.

Answer

Device firmware version.

Question 19

A Cisco Unified Border Element (CUBE) is deployed between Cisco Unified Communications Manager and a SIP provider. The provider sends an INVITE with a Diversion header containing the original called number. You need to strip the Diversion header before sending the INVITE to Unified CM. Which configuration element should you use?

Accepted Answer

voice class sip-profiles with a sip-header Diversion remove rule applied on the outbound dial peer. To remove a SIP header such as Diversion, you must use a voice class sip-profiles with a sip-header command that specifies the header name and the action. The profile is then applied to the outbound dial peer so that the header is stripped from the INVITE before it is sent to the next hop. This is the standard method on CUBE.

Answer

dial-peer voice with the diversion-header disable command

Answer

voice class uri with a diversion pattern to match and block the header

Answer

voice service voip with the sip-header Diversion remove command

Question 20

You have configured a Cisco Unified CM Route List with two Route Groups: one containing a SIP trunk to Provider A and another containing a SIP trunk to Provider B. You want calls to be attempted to Provider A first, and only if that fails, to Provider B. Which Route List configuration ensures this behavior?

Accepted Answer

Set the Route List's distribution algorithm to Top Down and ensure the Route Group for Provider A is listed first.. To route calls to Provider A first and then Provider B only on failure, the Route List must use the Top Down distribution algorithm and list the Route Group for Provider A before Provider B. Top Down tries each Route Group in order, providing sequential failover. Other algorithms like Circular or Longest Idle do not provide this strict primary/secondary behavior.

Answer

Set the Route List's distribution algorithm to Broadcast and ensure the Route Group for Provider A is listed first.

Answer

Set the Route List's distribution algorithm to Circular and ensure the Route Group for Provider A is listed first.

Answer

Set the Route List's distribution algorithm to Longest Idle and ensure the Route Group for Provider A is listed first.