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CCNA Infrastructure And Design Questions

41 questions · Infrastructure And Design · All types, answers revealed

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1
Multi-Selectmedium

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

Select 2 answers
A.Configuring multiple TFTP servers in Option 150
B.IP address redundancy via DHCP
C.SRST deployment
D.Load balancing via round-robin
E.CUCM Group configuration
AnswersC, E

SRST lets a branch router assume call control for its local IP phones when WAN connectivity to the CUCM cluster is lost. Phones fall back to the gateway and retain basic calling, satisfying the branch survivability requirement.

Why this answer

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.

Exam trap

CLCOR often tests the confusion between redundancy for configuration files (TFTP) and redundancy for call control (CUCM Groups, SRST).

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2
MCQhard

A collaboration designer must ensure that inter-cluster calls between two CUCM clusters remain secure. Which mechanism must be implemented to ensure media remains encrypted from end to end?

A.Disable SIP inspection on the firewalls.
B.Configure a VPN between the two clusters.
C.Ensure SRTP is enabled on the SIP Trunk Security Profile.
D.Enable MTP (Media Termination Point) for all calls.
AnswerC

SRTP encrypts the media payload itself, so enabling it on the SIP Trunk Security Profile ensures calls traversing the inter-cluster trunk carry encrypted RTP end to end, satisfying the requirement that media stays protected between clusters.

Why this answer

To ensure media remains encrypted end-to-end between two CUCM clusters, SRTP must be enabled on the SIP Trunk Security Profile. This setting ensures that the media streams (RTP) are encrypted using SRTP, providing confidentiality and integrity for the call. Without SRTP, even if signaling is encrypted, the media could be sent in clear text.

Exam trap

CLCOR often tests the misconception that a VPN or firewall configuration alone can secure media, but the exam expects knowledge that SRTP must be explicitly enabled on the SIP trunk security profile for end-to-end media encryption.

How to eliminate wrong answers

Option A is wrong because disabling SIP inspection on firewalls does not encrypt media; it may allow SIP traffic to pass but does not provide encryption. Option B is wrong because a VPN between clusters encrypts all traffic at the network layer, but it does not ensure that the media itself is encrypted end-to-end; it only secures the tunnel, and media could still be unencrypted within the tunnel. Option D is wrong because enabling MTP for all calls does not encrypt media; MTPs are used for transcoding or DTMF conversion and can actually prevent encryption if not configured properly.

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3
MCQeasy

Which network condition is the most significant contributor to poor voice quality in a VoIP environment?

A.Low CPU usage on the router
B.Static IP addresses
C.Jitter
D.High bandwidth usage
AnswerC

Jitter, the variation in packet arrival delay, causes gaps and reordering that voice codecs cannot conceal, producing choppy, distorted audio. Consistent latency is tolerable, but variable delay is the dominant contributor to poor VoIP voice quality.

Why this answer

Jitter—variation in packet arrival delay—is the most significant contributor to poor voice quality because VoIP codecs rely on a steady packet stream. Excessive jitter causes the jitter buffer to underrun or overflow, resulting in clipped, robotic, or dropped audio that is far more noticeable than raw latency.

Exam trap

CLCOR often tests the misconception that bandwidth alone determines voice quality, leading candidates to pick 'high bandwidth usage' instead of the correct latency/jitter/loss triad.

How to eliminate wrong answers

Option A is wrong because low CPU usage on the router is a healthy condition and does not degrade voice quality; high CPU would be the concern. Option B is wrong because static IP addressing has no direct impact on voice quality; DHCP vs static is an addressing choice, not a QoS factor. Option D is wrong because high bandwidth usage alone does not necessarily degrade voice if QoS is properly configured to prioritize RTP; bandwidth is a capacity issue, whereas jitter directly disrupts playout.

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4
MCQeasy

In the context of collaboration design, what is the purpose of the 'Location' in CUCM?

A.Defining physical building layouts
B.Device registration grouping
C.Call Admission Control
D.TFTP file distribution
AnswerC

Location in CUCM defines a logical grouping of devices sharing bandwidth constraints, enabling Call Admission Control to deny or reroute new calls when WAN links are saturated. It protects voice quality across sites, rather than performing dial-plan translation or device registration.

Why this answer

In Cisco Unified Communications Manager (CUCM), a 'Location' is the configuration object used to implement Call Admission Control (CAC). It defines the total bandwidth available between groups of devices, typically representing a WAN link between sites. When a call is placed between devices in different Locations, CUCM checks whether sufficient bandwidth exists in both Locations before allowing the call to proceed.

Exam trap

CLCOR often tests the confusion between Locations (CAC/bandwidth) and Regions (codec selection), so candidates who pick 'device registration grouping' or 'physical layout' are conflating Locations with Device Pools or ERL.

How to eliminate wrong answers

Option A is wrong because physical building layouts are documented via separate constructs (like ERL/ELF in emergency responder location, or simply site documentation) — CUCM Locations do not model floor plans or building geometry. Option B is wrong because device registration grouping is handled by Device Pools, not Locations; Device Pools define region, location, softkey template, and other device-level defaults. Option D is wrong because TFTP file distribution is a function of the TFTP service and device configuration file requests, not Locations.

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5
MCQmedium

When designing for high availability of Cisco Unified Communications Manager (CUCM) services, which specific configuration ensures that a secondary node takes over call processing if the primary node fails?

A.Configure a CUCM Group in the Device Pool with multiple servers.
B.Set the 'Call Processing Load' value in the System settings.
C.Enable Auto-registration for all phones.
D.Configure the cluster-wide failover parameter in the Service Parameter page.
AnswerA

A CUCM Group lists an ordered set of servers, and assigning it to the Device Pool lets endpoints register with the primary node while retaining the secondary as a fallback. If the primary fails, call processing shifts to the next listed server, delivering the required high availability.

Why this answer

A CUCM Group is an ordered list of up to three CUCM servers (primary, secondary, tertiary) that a device uses for call processing. When you assign a CUCM Group to a Device Pool, every device referencing that pool will attempt registration with the first server in the list and, if it becomes unreachable, will fail over to the next server in the group. This is the standard, supported mechanism for call-processing redundancy in a CUCM cluster.

Exam trap

CLCOR often tests the misconception that failover is controlled by a global service parameter or a system-wide setting, when in fact it is enforced per-device through the CUCM Group assigned in the Device Pool.

How to eliminate wrong answers

Option B is wrong because 'Call Processing Load' is not a real CUCM System setting; there is no such parameter that controls failover behavior. Option C is wrong because Auto-registration is a feature that allows unprovisioned phones to automatically receive a DN from a defined range — it has nothing to do with server redundancy or failover. Option D is wrong because there is no cluster-wide 'failover parameter' on the Service Parameter page; failover is controlled per-device via the CUCM Group assigned through the Device Pool, not by a global service parameter.

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6
MCQmedium

A collaboration engineer is preparing a Cisco Unified Communications Manager cluster for a new site that requires the Cisco 8845 IP Phone to operate in a secure environment. The security team mandates that all signaling and media between the phone and CUCM be encrypted, and that the phone authenticate to CUCM using a locally significant certificate (LSC). Which action must the engineer take first to ensure the phone can obtain an LSC and register securely?

A.Upload a publicly signed certificate to the CUCM Tomcat trust store and restart the Cisco Tomcat service.
B.Enable 'Mixed Mode' on the CUCM cluster and restart the Cisco CallManager service.
C.Configure a Phone Security Profile with 'Device Security Mode' set to Encrypted and assign it to the phone.
D.Install a Certificate Authority Proxy Function (CAPF) certificate on the phone manually via the phone's web interface.
AnswerB

Enabling Mixed Mode activates the Cisco Certificate Trust List (CTL) and allows the cluster to issue LSCs to phones. Without Mixed Mode, the phone can only use a Manufacturer Installed Certificate (MIC) or no certificate, and secure signaling cannot be established. This is the foundational step before any secure registration can occur.

Why this answer

To enable secure signaling and media with LSCs, the CUCM cluster must first be switched to Mixed Mode. This activates the Cisco CTL and allows the CAPF service to issue LSCs to supported endpoints. Only after Mixed Mode is enabled can a Phone Security Profile with Encrypted mode be effective, and the phone can then register securely using its LSC.

Exam trap

The trap here is assuming that configuring a Phone Security Profile alone enables secure registration, when Mixed Mode must be enabled first to allow LSC issuance.

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7
MCQmedium

A company is deploying a centralized Cisco Unified Communications Manager (CUCM) cluster with a single publisher and two subscribers. The publisher is located in the data center, and the subscribers are in the same data center. Branch offices connect via a WAN. The design team wants to ensure that IP phones at the branches can register and make calls even if the WAN link fails. Which CUCM feature should be implemented to achieve this?

A.Cisco Expressway-C and Expressway-E for Mobile and Remote Access (MRA)
B.Cisco Unified Border Element (CUBE) at the branch
C.Cisco Unified Communications Manager Express (CME) on the branch router
D.Cisco Unified Survivable Remote Site Telephony (SRST) on the branch router
AnswerD

SRST runs on the branch router and provides call processing fallback when the WAN fails and IP phones lose registration to the centralized CUCM. It allows the phones to register to the router and make basic calls, including PSTN access, ensuring survivability. This directly addresses the requirement for branch offices to maintain service during WAN outages.

Why this answer

SRST is the correct solution because it enables branch routers to provide call processing when the centralized CUCM becomes unreachable due to WAN failure. It allows IP phones to fall back to the router for basic calling, including PSTN access. Other options either provide different functions (CME, CUBE) or are for remote access (Expressway) and do not address WAN survivability.

Exam trap

The trap here is confusing branch survivability with remote access solutions, such as Expressway MRA, which require WAN connectivity and do not provide local call processing during outages.

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8
MCQeasy

A company is deploying Cisco Unified Communications Manager in a centralized call processing model with remote branches connected via a WAN. The network team wants to ensure that if the WAN link fails, IP phones at the branch can still make calls to the PSTN. Which solution should be implemented at the branch?

A.Cisco Expressway-C and Expressway-E for Mobile and Remote Access (MRA)
B.A second CUCM subscriber at the branch office
C.Cisco Unified Survivable Remote Site Telephony (SRST) on the branch router
D.Cisco Unified Communications Manager Express (CME) on the branch router
AnswerC

SRST provides call control redundancy for remote branches. When the WAN link to the central CUCM fails, the branch router with SRST takes over call processing, allowing IP phones to register with it and make calls to the PSTN via local gateways. This ensures business continuity for voice services.

Why this answer

SRST is the designed solution for branch office survivability in a centralized CUCM deployment. When the WAN fails, the branch router with SRST assumes call control, enabling IP phones to register and make PSTN calls via local gateways. This maintains voice services during outages without requiring a full CUCM cluster at the branch.

Exam trap

The trap here is confusing SRST with CME; SRST is a fallback mechanism for CUCM, while CME is a standalone call agent that does not automatically provide redundancy.

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9
MCQmedium

Which design principle must be followed when implementing a centralized call processing deployment with remote branch offices connected via a WAN?

A.Enable SRST on the local branch gateway to handle call processing during a WAN outage.
B.Deploy a dedicated CUCM node in every branch office.
C.Configure H.323 gateways at each site to communicate directly with all remote phones.
D.Use RSVP to guarantee bandwidth across the WAN for signaling.
AnswerA

SRST lets the branch gateway register local phones and provide call processing if the WAN to the central cluster fails. This satisfies the resilience constraint of a centralised deployment, where remote sites lose call control when connectivity to the central Cisco Unified Communications Manager cluster drops.

Why this answer

In a centralized call processing deployment, branch offices rely on the central CUCM for call control. If the WAN link fails, branch phones lose registration and cannot make or receive calls. SRST on the local branch gateway provides fallback call processing, allowing basic PSTN calling during WAN outages.

This is a fundamental design requirement for centralized deployments with remote sites.

Exam trap

CLCOR often tests the misconception that QoS or additional CUCM nodes can replace SRST for branch survivability, but SRST is the specific required design element for WAN outage resilience.

How to eliminate wrong answers

Option B is wrong because deploying a dedicated CUCM node in every branch office defeats the purpose of centralized call processing and increases cost and complexity. Option C is wrong because H.323 gateways do not provide call processing for IP phones; they only provide PSTN gateway functionality. Option D is wrong because RSVP is a QoS mechanism for bandwidth reservation, not a call processing redundancy solution; it does not address WAN outage survivability.

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10
MCQhard

When designing a secure collaboration network, what is the best practice for handling certificates in a CUCM and Expressway cluster?

A.Use self-signed certificates on all nodes.
B.Disable certificate verification on all endpoints.
C.Use the Publisher as the CA for the entire network.
D.Use a trusted third-party CA for all servers.
AnswerD

A trusted third-party CA lets every CUCM and Expressway node present certificates that all peers already trust, eliminating self-signed trust bundles between clusters. This satisfies the secure collaboration requirement by enabling TLS verification across the cluster without manual certificate exchange.

Why this answer

The best practice is to use a trusted third-party Certificate Authority (CA) for all servers in a CUCM and Expressway cluster. Public CA-signed certificates are trusted by default on endpoints, browsers, and other servers, eliminating the need to distribute and manually trust self-signed or internal CA certificates. This is especially critical for Expressway-E, which faces external endpoints that will not trust an internal CA.

Exam trap

CLCOR often tests certificate trust models, and candidates mistakenly believe an internal CA or the Publisher can serve as a universal trust anchor, not realizing external endpoints will reject certificates from untrusted roots.

How to eliminate wrong answers

Option A is wrong because self-signed certificates are not trusted by external endpoints and browsers, causing security warnings and failed TLS negotiations in MRA and B2B scenarios. Option B is wrong because disabling certificate verification defeats the purpose of TLS, exposes the deployment to man-in-the-middle attacks, and is explicitly discouraged by Cisco. Option C is wrong because using the CUCM Publisher as a CA for the entire network creates a single point of trust that external endpoints will not recognize, and it is not a supported or scalable PKI design.

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11
MCQeasy

A collaboration administrator is deploying a new Cisco Unified Communications Manager (CUCM) cluster and needs to ensure that the publisher database is protected against corruption. The administrator plans to use the Disaster Recovery System (DRS) for backups. Which statement accurately describes a DRS backup requirement?

A.A DRS backup can only be restored to a cluster with the exact same IP addresses and hostnames.
B.DRS requires an external SFTP server to store the backup files, and the backup can include the publisher and all subscribers.
C.DRS backups must be scheduled on each node individually, and the backup files are stored locally on each node.
D.DRS backup is only supported for the publisher node and cannot include subscriber nodes.
AnswerB

DRS uses a configured SFTP server as the backup destination. The backup can be scheduled from the publisher and can include the publisher and any subscribers. This centralizes backup management and ensures that all cluster data is protected. The SFTP server must be reachable from all nodes being backed up.

Why this answer

The Disaster Recovery System (DRS) in CUCM uses an external SFTP server to store backups. Backups are scheduled from the publisher and can include the publisher and all subscribers. This ensures that the entire cluster's configuration and data are protected.

The SFTP server must be configured with sufficient storage and network access from all nodes.

Exam trap

The trap here is assuming that DRS backups are stored locally or that only the publisher can be backed up, when in fact an external SFTP server is required and subscribers can be included.

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12
MCQhard

You are designing a Cisco Unified Communications Manager (CUCM) dial plan for a multinational company. The company has offices in North America, Europe, and Asia. Each region has its own PSTN gateway and dial plan requirements. The customer wants to minimize dial plan complexity and ensure that users can dial local numbers as they are accustomed to. Which approach should you recommend?

A.Use a single dial plan with no partitions and rely on translation patterns to handle regional differences.
B.Implement separate CUCM clusters for each region to isolate dial plans.
C.Deploy a centralized dial plan with location-specific partitions and calling search spaces (CSS) to provide regional dialing habits.
D.Implement a single global dial plan with a uniform dialing pattern for all regions.
AnswerC

Using partitions and CSS, you can create a centralized dial plan that allows each region to have its own dialing patterns while sharing a common CUCM cluster. This minimizes complexity by centralizing administration, but still provides the flexibility for local dialing habits. Each region's phones are assigned a CSS that grants access to the appropriate partitions.

Why this answer

A centralized dial plan with partitions and calling search spaces allows each region to have its own dialing patterns while maintaining a single CUCM cluster. This reduces complexity by centralizing administration, but provides the flexibility for local dialing habits. The other options either do not meet the requirement or introduce unnecessary complexity.

Exam trap

The trap here is assuming that a single global dial plan or separate clusters are the only ways to handle multinational dialing, when partitions and CSS provide a more efficient solution.

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13
MCQhard

A collaboration engineer is designing a CUCM cluster with two nodes: a publisher and a subscriber. The engineer must ensure that the cluster can support 2500 IP phones and that call processing continues if the publisher fails. Which statement accurately describes the design requirement?

A.The subscriber can register up to 2500 phones, and if the publisher fails, the subscriber continues call processing but database changes are not possible until the publisher is restored.
B.The cluster can support up to 2500 phones per node, but only if the publisher and subscriber are in the same LAN.
C.The subscriber must be configured to handle all call processing, and the publisher should only be used for database and TFTP services.
D.The publisher must be dedicated solely to database replication and not perform call processing.
AnswerA

In a two-node cluster, the subscriber can handle call processing for up to 2500 phones (depending on the OVA). If the publisher fails, the subscriber continues to process calls, but database replication stops, and no configuration changes can be made until the publisher is restored. This is a key consideration for high availability.

Why this answer

In a two-node CUCM cluster, the subscriber can register up to 2500 phones and handle call processing. If the publisher fails, the subscriber continues to process calls, but database replication and configuration changes are halted until the publisher is back online. This design provides call processing redundancy but not database redundancy without additional nodes.

Exam trap

The trap here is assuming that the publisher cannot perform call processing or that a two-node cluster cannot support 2500 phones; the actual limitation is that database changes are unavailable during publisher outage.

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14
MCQhard

A network designer is planning a centralized call processing deployment with 20 remote sites. Each site has 50 IP phones and uses G.711 for internal calls. The designer must ensure that calls between sites use G.729 to conserve WAN bandwidth, but calls within a site use G.711. Which CUCM configuration element should be used to achieve this?

A.Location
B.Region
C.Device Pool
D.Media Resource Group List (MRGL)
AnswerB

Regions define the maximum codec for calls between devices assigned to different regions. By placing each site's devices in its own region and configuring inter-region links to use G.729, while intra-region calls default to G.711, the designer can enforce the required codec policy. This directly controls call bandwidth between sites.

Why this answer

Regions are the correct configuration element to enforce codec selection between sites. By defining regions and inter-region relationships with G.729 as the maximum codec, while allowing G.711 within a region, the designer ensures that inter-site calls use the lower-bandwidth codec. This directly meets the requirement without relying on call admission control.

Exam trap

The trap here is confusing Locations with Regions; Locations manage bandwidth and call admission, while Regions control which codecs are permitted between groups of devices.

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15
MCQhard

You are designing a high-availability solution for Cisco Expressway. Which element is required to load-balance traffic between multiple Expressway nodes?

A.Cisco CUBE redundancy
B.Cisco UCM Group
C.External Load Balancer
D.DNS Round Robin only
AnswerC

Cisco Expressway clusters do not natively distribute inbound signalling across nodes; an external load balancer is required to spread traffic between them. This satisfies the high-availability load-balancing requirement, since DNS round-robin alone cannot health-check or evenly balance the Expressway nodes.

Why this answer

To load-balance traffic across multiple Expressway nodes (e.g., Expressway-C or Expressway-E clusters), an external load balancer is required because Expressway does not provide built-in load balancing. The load balancer distributes incoming connections across the nodes for scalability and high availability.

Exam trap

CLCOR often tests the misconception that DNS Round Robin or built-in clustering provides load balancing for Expressway, when an external load balancer is mandatory for true HA and scalability.

How to eliminate wrong answers

Option A is wrong because Cisco CUBE redundancy is for voice gateways and SIP trunk high availability, not for Expressway clustering. Option B is wrong because a Cisco UCM Group is a CUCM concept for phone registration redundancy, not for Expressway load balancing. Option D is wrong because DNS Round Robin alone is not sufficient for Expressway high availability; it does not perform health checks or session persistence, and Expressway requires a proper load balancer for clustering.

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16
MCQmedium

You are designing a network for video collaboration. What is the recommended maximum one-way latency for high-quality video conferencing?

A.300ms
B.500ms
C.50ms
D.150ms
AnswerD

150ms is the industry-standard threshold for real-time media.

Why this answer

For high-quality, interactive video conferencing, the ITU-T G.114 recommendation specifies that one-way latency should not exceed 150 ms to maintain conversational quality. Beyond 150 ms, participants begin to notice talker overlap and difficulty with turn-taking, degrading the collaboration experience. 150 ms is the accepted threshold for real-time voice and video applications.

Exam trap

CLCOR often tests the confusion between one-way and round-trip latency; candidates who recall '300 ms' from RTT guidelines may incorrectly select 300 ms instead of the 150 ms one-way limit.

How to eliminate wrong answers

Option A (300 ms) is wrong because 300 ms one-way latency is well above the G.114 limit and causes noticeable degradation in interactivity, though it may still be tolerable for one-way streaming. Option B (500 ms) is wrong because 500 ms one-way latency is far too high for real-time video conferencing and would make conversation feel like a satellite call with awkward pauses. Option C (50 ms) is wrong because while 50 ms is excellent and desirable, it is not the recommended maximum — it is an ideal target, not the threshold; the question asks for the maximum acceptable one-way latency.

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17
MCQeasy

When deploying Cisco IP phones, what is the best practice for VLAN configuration?

A.Use the default VLAN 1 for all voice and data.
B.Use a dedicated Voice VLAN.
C.Use a separate physical switch for phones.
D.Do not use VLANs; use static routing.
AnswerB

A dedicated voice VLAN separates signalling and RTP traffic from data traffic, letting you apply QoS trust boundaries and voice-specific DHCP scopes. This satisfies the stem's best-practise requirement by isolating latency-sensitive voice from broadcast-heavy data, easing troubleshooting and security policy.

Why this answer

Best practice is to place voice traffic on a dedicated Voice VLAN, separate from the data VLAN, so that QoS policies, security, and broadcast domain boundaries can be applied specifically to voice. This also enables features like voice VLAN access (CDP/LLDP-MED) that let the phone tag its own traffic while the PC behind it uses the data VLAN untagged.

Exam trap

CLCOR often tests the misconception that VLAN 1 or physical separation is acceptable for voice — candidates underestimate why a dedicated Voice VLAN is the standard, especially for QoS and LLDP-MED-based phone VLAN assignment.

How to eliminate wrong answers

Option A is wrong because using the default VLAN 1 for both voice and data mixes traffic, prevents clean QoS classification, and is a security and broadcast-domain best-practice violation. Option C is wrong because using a separate physical switch for phones is not a best practice — it's costly, doesn't scale, and VLANs exist precisely to logically separate traffic on shared infrastructure. Option D is wrong because avoiding VLANs and using static routing is nonsensical for voice deployment — VLANs are the standard Layer 2 mechanism for traffic separation, and static routing doesn't address voice segmentation at all.

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18
MCQmedium

What is the primary function of the CUBE in a SIP trunking scenario?

A.Call routing for internal phones
B.Demarcation and SIP normalization
C.Voice messaging storage
D.DHCP Server functionality
AnswerB

CUBE sits at the network demarcation point, terminating the SIP trunk from the provider and normalising SIP signalling and media between disparate implementations. This satisfies the stem's requirement for the primary function: interworking vendor-specific SIP dialects, headers and codecs so the enterprise call control platform interoperates cleanly with the carrier.

Why this answer

The CUBE (Cisco Unified Border Element) is Cisco's session border controller function that sits at the edge between the enterprise VoIP network and the ITSP's SIP trunk. Its primary role is to provide a demarcation point and perform SIP normalization — translating between different SIP dialects, header formats, and SDP variations so that the enterprise and provider can interoperate. It also handles topology hiding, security, and call admission control at that boundary.

Exam trap

CLCOR often tests the confusion between CUBE's edge demarcation/normalization role and CUCM's internal call routing role — candidates mistakenly pick 'call routing' because CUBE does route calls, but only at the SIP trunk boundary.

How to eliminate wrong answers

Option A is wrong because call routing for internal phones is the function of Cisco Unified Communications Manager (CUCM) or the dial plan, not the CUBE — CUBE only routes at the SIP trunk edge. Option C is wrong because voice messaging storage is provided by Cisco Unity Connection, not CUBE. Option D is wrong because DHCP server functionality is provided by a router, switch, or dedicated DHCP server, not by CUBE.

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19
Multi-Selecthard

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

Select 3 answers
A.Use broadcast mode
B.Configure external Stratum 1 or 2 servers
C.Open UDP 123 on firewalls
D.Enable NTP on the local voice gateway
E.Consistent configuration across all nodes
AnswersB, C, E

CUCM has no internal authoritative clock, so it must sync to an external reliable time source. Stratum 1 or 2 servers provide accurate reference time, satisfying the stem's requirement. Without reachable external servers, nodes drift and cluster services fail.

Why this answer

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.

Exam trap

CLCOR often tests the misconception that broadcast mode or gateway NTP is required, when the focus should be on external servers, firewall ports, and consistent configuration.

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20
Multi-Selectmedium

Which TWO of the following are essential components of a Cisco network readiness assessment for collaboration?

Select 2 answers
A.Bandwidth capacity analysis
B.Analyzing NTP Stratum levels
C.QoS configuration audit
D.Verifying phone firmware versions
E.Checking IP address assignment
AnswersA, C

Bandwidth capacity analysis determines whether links and WAN circuits can carry projected voice, video and sharing traffic at peak load. Collaboration flows are bandwidth-intensive and delay-sensitive, so confirming sufficient capacity is essential to a readiness assessment.

Why this answer

Bandwidth capacity analysis (A) is essential because Cisco collaboration services such as voice, video, and conferencing have strict per-call and per-stream bandwidth requirements, and the readiness assessment must confirm the WAN/LAN can carry the projected concurrent media load with adequate headroom. A QoS configuration audit (C) is equally essential because real-time collaboration traffic (RTP media, signaling) must be properly classified, marked, and queued (e.g., EF for voice, AF41 for video) to guarantee low latency, jitter, and packet loss across the network. Analyzing NTP Stratum levels (B) is not a core readiness component, since NTP accuracy matters for call detail records and certificates but stratum level itself is not a standard readiness metric.

Verifying phone firmware versions (D) is a deployment/upgrade task rather than a network readiness assessment element. Checking IP address assignment (E) is a general design consideration, not one of the two essential readiness assessment components for collaboration.

Exam trap

CLCOR often tests whether candidates can separate core readiness pillars (bandwidth, QoS) from peripheral operational tasks (firmware, NTP, IP assignment) — the trap is picking device-level checks as 'essential components.'

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21
Multi-Selectmedium

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.)

Select 2 answers
A.Each IP phone can be configured with a prioritized list of up to three CUCM servers for registration using a CUCM Group.
B.The publisher must be included in the CUCM Group to ensure that phones can register if all subscribers fail.
C.The maximum number of CUCM servers in a CUCM Group is five, allowing for greater redundancy.
D.If a subscriber fails, phones registered to it will automatically re-register to another subscriber in the same CUCM Group based on priority order.
E.CUCM Groups are assigned to phones directly on each phone configuration page, not via Device Pool.
AnswersA, D

A CUCM Group can contain up to three CUCM servers in a prioritized order. When a phone boots, it attempts to register with the first server in its assigned CUCM Group. If that server is unavailable, it tries the next server in the list. This provides redundancy and ensures that phones can register even if one or more servers fail.

Why this answer

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.

Exam trap

The trap here is thinking that the publisher should be used for call-processing redundancy or that CUCM Groups can contain more than three servers, when in fact the limit is three and the publisher should be reserved for database and service management.

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22
MCQhard

A collaboration engineer is designing a SIP trunk between Cisco Unified Communications Manager (CUCM) and Cisco Unified Border Element (CUBE). The engineer must ensure that calls are routed correctly and that CUCM can send calls to CUBE. Which element should be configured on CUCM to point to CUBE as the next hop for calls destined to the ITSP?

A.SIP trunk configured on CUCM with destination address of CUBE
B.Translation pattern that rewrites the called number to CUBE's IP address
C.H.323 gateway configured on CUCM with destination address of CUBE
D.Route group with CUBE as the only member
AnswerA

A SIP trunk on CUCM is used to route calls to an external device such as CUBE. The trunk is configured with the destination address (IP address or DNS name) of CUBE's signaling interface. When a call matches a route pattern that points to this SIP trunk, CUCM sends the SIP INVITE to CUBE, which then forwards it to the ITSP. This is the correct way to integrate CUCM with CUBE.

Why this answer

To route calls from CUCM to CUBE, you configure a SIP trunk on CUCM with the destination address of CUBE. This trunk is then referenced by a route group, route list, and route pattern. When a call matches the route pattern, CUCM sends the call to CUBE via the SIP trunk.

The trunk itself is the element that points to CUBE as the next hop.

Exam trap

The trap here is confusing the role of a route group or translation pattern with the actual trunk that specifies the destination address.

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23
MCQeasy

When designing an NTP deployment for a Cisco Collaboration environment, what is the best practice for time synchronization across CUCM nodes?

A.Point all CUCM nodes to the same external NTP stratum 1 or 2 servers.
B.Configure NTP only on the publisher node.
C.Point each CUCM node to a different NTP server for redundancy.
D.Use the publisher as the NTP server for all subscriber nodes.
AnswerA

Pointing every CUCM node at the same external stratum 1 or 2 sources keeps cluster time consistent and avoids CUCM acting as an NTP server for itself. Uniform external reference prevents drift between nodes that would disrupt CDR timestamps and database replication.

Why this answer

Cisco Collaboration best practice requires every CUCM node (publisher and subscribers) to synchronize directly with the same external, reliable NTP source — typically stratum 1 or 2 servers. This ensures consistent, accurate time across all nodes, which is critical for database replication, CDR timestamps, certificate validation, and log correlation. Pointing all nodes to the same external source avoids a single point of failure and prevents time drift that could break cluster operations.

Exam trap

CLCOR often tests the misconception that the CUCM publisher should act as the NTP server for subscribers, confusing the publisher's role in database replication with time synchronization.

How to eliminate wrong answers

Option B is wrong because configuring NTP only on the publisher leaves subscribers unsynchronized, causing time drift that breaks database replication and certificate checks. Option C is wrong because pointing each node to a different NTP server introduces inconsistent time sources, leading to clock skew across the cluster. Option D is wrong because using the publisher as an NTP server for subscribers creates a single point of failure and is not a Cisco-supported best practice; if the publisher goes down, subscribers lose time synchronization.

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24
MCQeasy

Which component in a Cisco Collaboration design is responsible for providing the dial tone and call routing logic for IP phones?

A.Cisco Expressway
B.Cisco Unity Connection
C.Cisco CUBE
D.Cisco Unified Communications Manager
AnswerD

Cisco Unified Communications Manager provides call processing, generating dial tone and executing call routing logic for registered IP phones. It is the call-control core, distinct from gateways and endpoints that merely carry or originate media.

Why this answer

Cisco Unified Communications Manager (CUCM) is the call-processing component that provides dial tone and call routing logic for IP phones in a Cisco Collaboration design. It handles call setup, routing, and features for endpoints.

Exam trap

CLCOR often tests the functions of different collaboration components, and candidates may confuse CUCM with CUBE or Expressway, which serve different purposes in the architecture.

How to eliminate wrong answers

Option A is wrong because Cisco Expressway provides firewall traversal and mobile remote access, not dial tone or call routing for internal IP phones. Option B is wrong because Cisco Unity Connection is a voicemail and unified messaging system, not a call-processing server. Option C is wrong because Cisco CUBE (Cisco Unified Border Element) is a session border controller for SIP trunking and interconnecting networks, not for providing dial tone to IP phones.

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25
MCQmedium

You are designing a Cisco Collaboration deployment for a hospital where 911 calls must always be routed to the nearest Public Safety Answering Point (PSAP) based on the physical location of the calling device. The hospital has multiple buildings across a campus, and wireless IP phones (Cisco 8821) are used by nurses who roam between buildings. Which CUCM feature should you implement to ensure that emergency calls are routed with the correct location information?

A.Enhanced Location Call Admission Control (ELCAC) in CUCM
B.Cisco Unified Survivable Remote Site Telephony (SRST)
C.Cisco Emergency Responder (CER) integrated with CUCM
D.Device Mobility in Cisco Unified Communications Manager
AnswerC

Cisco Emergency Responder tracks the switch port or wireless access point to determine the exact physical location of the caller. For wireless phones, it uses the associated access point or switch information to update the emergency response location dynamically. This ensures that 911 calls are routed to the correct PSAP and that the location is transmitted to the emergency dispatcher.

Why this answer

Cisco Emergency Responder is the dedicated solution for E911 location tracking in Cisco Collaboration. It integrates with CUCM and network infrastructure to map devices to emergency response locations. Device Mobility, SRST, and ELCAC serve different purposes and do not provide the required location granularity for emergency calls.

Exam trap

The trap here is assuming that Device Mobility or ELCAC can provide location tracking for E911, but they are designed for call routing and bandwidth management, not emergency location.

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26
MCQmedium

Which service must be running for a Cisco IP Phone to download its configuration file via TFTP?

A.Cisco CallManager
B.Cisco CTIManager
C.Cisco TFTP
D.Cisco Database Layer Monitor
AnswerC

Cisco TFTP is the service that must be running on the server for a Cisco IP Phone to download its configuration file via TFTP. Without it, the phone cannot retrieve its config and registration fails.

Why this answer

Cisco IP Phones download their configuration files (XML) from the CUCM TFTP service, which runs as the 'Cisco TFTP' service on the CUCM server. Without this service running, phones cannot retrieve their config and will fail to register. The Cisco TFTP service is separate from CallManager and must be active on at least one node in the cluster.

Exam trap

CLCOR often tests the distinction between CUCM services: candidates confuse CallManager (registration) with TFTP (file download), so they pick CallManager when the question asks about configuration file download.

How to eliminate wrong answers

Option A is wrong because Cisco CallManager (Cisco CallManager service) handles call processing and registration, not file downloads. Option B is wrong because Cisco CTIManager provides computer telephony integration (CTI) services for applications like Jabber and contact centers, not TFTP. Option D is wrong because Cisco Database Layer Monitor monitors database replication and health, not configuration file delivery.

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27
MCQeasy

In a multisite WAN deployment, which codec should you choose to balance voice quality and bandwidth consumption across a constrained link?

A.G.711
B.G.729
C.AAC-LD
D.G.722
AnswerB

G.729 compresses voice to 8 kbps using conjugate-structure algebraic-code-excited linear prediction, satisfying the constrained-link bandwidth requirement while retaining toll-quality speech. Its lower bitrate than G.711 (64 kbps) reduces consumption across the WAN, and its modest processing delay remains acceptable for multisite voice, balancing quality against available capacity.

Why this answer

G.729 is the correct choice because it compresses voice into an 8 kbps stream using CS-ACELP, delivering toll-quality voice at roughly one-eighth the bandwidth of G.711 (64 kbps). In a multisite WAN with constrained links, this ~8x bandwidth savings per call is the standard trade-off Cisco recommends for WAN voice transport. It is the de facto codec for bandwidth-constrained voice deployments.

Exam trap

CLCOR often tests the misconception that higher-bandwidth codecs like G.722 or G.711 are 'better' choices for quality, when the question's constraint (limited WAN bandwidth) makes the compressed G.729 the only correct answer.

How to eliminate wrong answers

Option A is wrong because G.711 is an uncompressed 64 kbps PCM codec (G.711u/a-law) that consumes the most bandwidth per call and is intended for LAN/high-bandwidth environments, not constrained WAN links. Option C is wrong because AAC-LD is a wideband/low-delay codec used primarily in video conferencing and TelePresence endpoints, not the standard codec for WAN voice call bandwidth optimization. Option D is wrong because G.722 is a 7 kHz wideband codec operating at 48/56/64 kbps — it improves voice quality but does not reduce bandwidth consumption, so it fails the 'constrained link' requirement.

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28
MCQmedium

A network engineer is deploying Cisco Unified Border Element (CUBE) and must ensure that voice traffic is prioritized correctly across the enterprise network. Which command correctly maps the voice traffic DSCP value to the correct priority queue in a Cisco IOS-XE router?

A.class-map match-any VOICE; match protocol sip; policy-map VOICE-POLICY; class VOICE; queue-limit 100
B.class-map match-all VOICE; match dscp cs4; policy-map VOICE-POLICY; class VOICE; bandwidth 1000
C.class-map match-any VOICE; match dscp ef; policy-map VOICE-POLICY; class VOICE; fair-queue
D.class-map match-any VOICE; match dscp ef; policy-map VOICE-POLICY; class VOICE; priority 1000
AnswerD

The class-map matches DSCP EF (46), the standard marking for voice bearer traffic, then the policy-map's priority command places that class into the low-latency priority queue with a 1000 kbps guaranteed bandwidth. This satisfies the stem's requirement to map the voice DSCP value to the correct priority queue on IOS-XE.

Why this answer

The correct configuration is to match DSCP EF (46) in a class-map and apply the 'priority' command with a bandwidth value in the policy-map. This maps voice traffic to a low-latency priority queue, ensuring proper prioritization. The 'priority 1000' command creates a strict priority queue with a policed bandwidth of 1000 kbps, which is ideal for voice.

Exam trap

The trap is confusing DSCP values (EF vs. CS4) and the commands for bandwidth guarantee vs. priority; candidates often pick 'bandwidth' or 'fair-queue' because they sound like they provide prioritization, but only 'priority' creates a low-latency queue.

How to eliminate wrong answers

Option A is wrong because it matches protocol sip and uses 'queue-limit' instead of a priority queue; SIP is signaling, not voice media, and queue-limit does not provide priority. Option B is wrong because it matches DSCP CS4 (32), which is not the standard for voice (EF is 46), and uses 'bandwidth' instead of 'priority', which does not provide low-latency queuing. Option C is wrong because it matches DSCP EF correctly but uses 'fair-queue' instead of 'priority', which does not guarantee priority treatment.

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29
Multi-Selecthard

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

Select 3 answers
A.Install and trust CA certificates
B.Configure a SIP Trunk Security Profile
C.Disable SIP digest authentication
D.Enable TLS on the trunk
E.Enable H.323 Fast Start
AnswersA, B, D

Installing and trusting CA certificates lets CUCM and CUBE validate each other's identities during the TLS handshake. This satisfies the secure signalling path requirement by establishing the trust anchor needed before encrypted SIP signalling can be accepted.

Why this answer

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.

Exam trap

CLCOR often tests the inclusion of irrelevant options like H.323 Fast Start or disabling authentication, which are not part of SIP security best practices.

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30
MCQhard

A network engineer is deploying a Cisco Unified Border Element (CUBE) for a SIP trunk to an ITSP. The ITSP requires that all outbound calls present a specific calling number and that the SIP From header contain the same number. The engineer must ensure that the calling number is manipulated correctly on the outbound dial peer. Which CUBE configuration command should be used to modify the calling number in both the From header and the P-Asserted-Identity header?

A.voice translation-rule and voice translation-profile applied to the outbound dial peer
B.numbering-type on the outbound dial peer
C.SIP profile with header-passing and privacy settings
D.clid command on the outbound dial peer
AnswerA

Voice translation rules and profiles are the standard method to manipulate calling and called numbers on CUBE. When applied to the outbound dial peer, the translation profile can modify the calling number, which updates the SIP From header and, depending on the profile, the P-Asserted-Identity header. This satisfies the ITSP requirement to present a specific calling number consistently.

Why this answer

The correct method to modify the calling number for SIP calls on CUBE is to use voice translation rules and profiles. These allow granular manipulation of the calling number, which is then reflected in the SIP From and P-Asserted-Identity headers. Applying the translation profile to the outbound dial peer ensures that all outbound calls to the ITSP present the required calling number.

Exam trap

The trap here is assuming that dial-peer commands like clid or numbering-type can rewrite SIP headers, when in fact header manipulation requires translation rules and profiles.

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31
MCQhard

You are managing a Cisco Collaboration cluster and notice that database replication is stuck in a '3' (blocked) state. What is the standard troubleshooting step?

A.utils dbreplication reset
B.utils service restart Cisco Database Layer Monitor
C.utils system restart
D.utils network ping
AnswerA

A blocked replication state means the subscriber's data is inconsistent with the publisher, so the standard remedy is utils dbreplication reset, which drops and rebuilds replication between nodes, restoring a clean, consistent state across the cluster.

Why this answer

When database replication is stuck in a '3' (blocked) state in a Cisco Collaboration cluster, the standard troubleshooting step is to run 'utils dbreplication reset' on the publisher or affected subscriber. This command resets the replication and forces a full re-sync of the database, resolving blocked states caused by inconsistencies or network issues.

Exam trap

CLCOR often tests the confusion between restarting services and performing a full replication reset, leading candidates to choose less effective troubleshooting steps.

How to eliminate wrong answers

Option B is wrong because restarting the Cisco Database Layer Monitor service may temporarily alleviate issues but does not perform a full replication reset, which is necessary for a blocked state. Option C is wrong because restarting the entire system is disruptive and not the standard first step for replication issues. Option D is wrong because 'utils network ping' only tests network connectivity and does not address database replication problems.

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32
MCQmedium

A company's main site runs a Cisco Unified Communications Manager (CUCM) cluster. They are adding a new remote branch that will have 50 IP phones. The branch has limited bandwidth and must survive a WAN outage with basic call functionality between local phones. Which design approach should be used at the branch?

A.Deploy a CUCM subscriber at the branch to handle local call processing.
B.Deploy a Cisco Expressway-C and Expressway-E pair at the branch for call processing.
C.Use the existing CUCM cluster at the main site and rely on the WAN for all call processing.
D.Deploy a Cisco Unified Communications Manager Express (CME) router at the branch to provide local call processing.
AnswerD

CME provides call processing at the branch, allowing local calls to continue during a WAN outage. It supports 50 phones and integrates with CUCM using SIP or H.323 trunks for centralized dial plan and voicemail. This design is ideal for small branches with limited bandwidth, as it reduces WAN dependency for intra-branch calls.

Why this answer

For small remote branches with limited bandwidth and a need for local survivability, a Cisco Unified Communications Manager Express (CME) router is the appropriate choice. It provides local call processing and can be integrated with the central CUCM for centralized management and dial plan. This ensures that intra-branch calls continue even if the WAN link fails.

Exam trap

The trap here is confusing survivability solutions like CME with full CUCM nodes or Expressway, which serve different purposes such as MRA or centralized processing.

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33
MCQmedium

In a Cisco collaboration design, you are tasked with identifying the DHCP Option required for Cisco IP Phones to discover the TFTP server address automatically. Which option is standard for this purpose?

A.Option 66
B.Option 128
C.Option 150
D.Option 43
AnswerC

DHCP Option 150 carries a list of TFTP server IP addresses in a single field, which Cisco IP Phones use to locate their configuration files. It is the standard, vendor-specific alternative to Option 66's single hostname, matching the discovery requirement.

Why this answer

DHCP Option 150 is the Cisco-proprietary option that delivers a list of TFTP server IP addresses to Cisco IP Phones during DHCP scope configuration. Phones use this to locate the TFTP server, download their configuration XML and firmware, and register with CUCM. Option 66 is the standard (RFC 2132) TFTP server name option and can also work, but Option 150 is the Cisco-standard answer for IP phone TFTP discovery.

Exam trap

CLCOR often tests whether candidates pick Option 66 (the IETF standard) instead of Option 150 (the Cisco-specific standard) — for Cisco IP phones, Option 150 is the expected answer.

How to eliminate wrong answers

Option A is wrong because although Option 66 (TFTP server name, RFC 2132) can be used, the Cisco-standard and most commonly tested option for IP phone TFTP discovery is Option 150, which supports a list of IP addresses rather than a single hostname. Option B is wrong because Option 128 is not a standard or Cisco TFTP discovery option — it is sometimes used for PXE or vendor-specific purposes but not for Cisco IP phone TFTP. Option D is wrong because Option 43 is used for vendor-specific information (commonly for wireless AP controllers and some VoIP vendors), not for Cisco IP phone TFTP server discovery.

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34
Multi-Selecthard

Which THREE of the following items must be verified during a site-to-site VPN check for collaboration traffic?

Select 3 answers
A.DHCP relay configuration
B.MTU size
C.NTP server reachability
D.UDP port range availability
E.DSCP tag preservation
AnswersB, D, E

IPsec and GRE encapsulation add overhead, so oversized packets get dropped when the path MTU is exceeded. Verifying MTU size prevents fragmentation and black-holed collaboration traffic such as voice and video across the site-to-site tunnel.

Why this answer

Option B (MTU size) is correct because site-to-site VPN tunnels add encapsulation overhead (e.g., IPsec ESP/AH headers), and if the MTU is not adjusted or fragmentation/PMTUD is not handled properly, large collaboration packets such as voice and video streams will be dropped or fragmented, degrading real-time traffic. Option D (UDP port range availability) is correct because collaboration traffic relies heavily on UDP (for example, RTP media streams and SIP signaling), so the required UDP port ranges must be permitted through the VPN and any intervening firewalls/ACLs for the tunnel to carry that traffic. Option E (DSCP tag preservation) is correct because collaboration traffic depends on QoS markings; if DSCP values are not preserved across the VPN (e.g., due to tunnel encapsulation or a misconfigured QoS policy), real-time voice and video packets lose their prioritization and suffer latency, jitter, and packet loss.

Option A (DHCP relay configuration) is not required for a site-to-site VPN check because DHCP relay concerns local address assignment, not the transport of collaboration traffic across the tunnel. Option C (NTP server reachability) is not a mandatory item for this check, since NTP is important for general network health and certificate/logging accuracy but is not a direct requirement for forwarding collaboration traffic over a site-to-site VPN.

Exam trap

The trap here is selecting NTP or DHCP relay as VPN verification items because they are generally important in collaboration, but the question specifically targets VPN traffic handling: MTU, UDP ports, and DSCP.

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35
MCQmedium

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

A.IPsec
B.SRTP
C.TLS
D.SSH
AnswerC

TLS encrypts SIP signalling hop by hop, securing the signalling path between CUCM and endpoints or trunks. It satisfies the requirement for encrypted voice signalling, unlike plain SIP over TCP or UDP, which transmit headers and messages in cleartext.

Why this answer

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.

Exam trap

CLCOR often tests the distinction between signaling and media encryption; candidates may confuse SRTP (media) with TLS (signaling) or think IPsec is used for SIP encryption, but TLS is the correct protocol for SIP signaling.

How to eliminate wrong answers

Option A is wrong because IPsec is typically used for network-layer encryption (e.g., VPNs) and not specifically for SIP signaling encryption in CUCM. Option B is wrong because SRTP encrypts media (voice/video), not signaling. Option D is wrong because SSH is used for secure CLI access, not for SIP signaling.

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36
MCQhard

You are configuring DNS SRV records to support Jabber service discovery. Which domain format is correct for the _collab-edge._tls record?

A._tls._collab-edge.example.com
B.collab-edge.example.com._tls
C._collab-edge._tcp.example.com
D._collab-edge._tls.example.com
AnswerD

The _collab-edge._tls SRV record must sit under the organisation's own domain to let external clients resolve Expressway-E for Mobile and Remote Access. Placing it at example.com satisfies the stem's requirement for the correct service-discovery domain format, since _collab-edge is queried against the customer domain, not a subdomain such as _cisco-uds.

Why this answer

DNS SRV records use the format _service._protocol.domain. For Jabber service discovery, the service is 'collab-edge' and the protocol is 'tls', so the correct record is _collab-edge._tls.example.com. This record is used by clients to locate the Expressway-E for Mobile and Remote Access (MRA) and is queried during the initial service discovery process.

Exam trap

CLCOR often tests the exact ordering of service and protocol labels in SRV records, and candidates frequently reverse them or use _tcp instead of _tls for the collab-edge record.

How to eliminate wrong answers

Option A is wrong because it reverses the service and protocol labels, using _tls._collab-edge, which does not conform to the SRV record naming convention. Option B is wrong because it places the domain before the service and protocol labels, which is not valid DNS syntax. Option C is wrong because it uses _tcp instead of _tls; while _tcp is used for other services like _cisco-uds._tcp, the _collab-edge record specifically requires _tls.

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37
MCQmedium

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

A.Cisco CallManager Traces
B.System Health Reports
C.Cisco TFTP Logs
D.Cisco Tomcat Logs
AnswerA

Cisco CallManager Traces record SCCP, SIP and TFTP registration exchanges between IP phones and CUCM, capturing the exact signalling failures. This directly satisfies the stem's requirement for troubleshooting registration issues, unlike SDL or CDR logs, which cover call processing and billing records.

Why this answer

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.

Exam trap

CLCOR often tests the distinction between logs for different CUCM services, and candidates may incorrectly assume that TFTP logs (which handle configuration files) or Tomcat logs (which handle web services) are involved in registration, when in fact registration signaling is exclusively captured in CallManager traces.

How to eliminate wrong answers

Option B is wrong because System Health Reports provide high-level performance and status summaries, not the granular protocol-level details needed to troubleshoot a specific registration failure. Option C is wrong because Cisco TFTP Logs only record file transfers (e.g., configuration, firmware) and do not capture the registration signaling between the phone and CUCM. Option D is wrong because Cisco Tomcat Logs pertain to the Tomcat web service (used for CUCM web administration and services like Cisco Unified Reporting), not to the call-processing registration process.

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38
MCQmedium

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?

A.IP DSCP AF31 (26) and CoS 3.
B.IP DSCP EF (46) and CoS 5.
C.IP DSCP CS3 (24) and CoS 3.
D.IP DSCP AF41 (34) and CoS 4.
AnswerB

Voice bearer traffic should be marked with DSCP EF (Expedited Forwarding, 46) and CoS 5. This is the standard Cisco recommendation for voice media. EF provides low latency, low jitter, and low loss, which are critical for voice quality. CoS 5 is used at Layer 2 to ensure proper queuing on trunk links.

Why this answer

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.

Exam trap

The trap here is confusing the markings for voice with those for video or call signaling, such as using AF41 or CS3 for voice.

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39
MCQmedium

You are verifying the network readiness for a Cisco TelePresence endpoint. Which tool would you use to measure the packet loss, latency, and jitter on the path to the CUCM cluster?

A.NetFlow
B.SNMP
C.IP SLA
D.SPAN
AnswerC

IP SLA actively generates synthetic probes to measure packet loss, latency, and jitter along the path to the CUCM cluster, directly satisfying the endpoint readiness requirement. Its jitter operation reports round-trip delay and inter-packet variation, giving the precise metrics needed to validate voice-quality thresholds before deploying the TelePresence endpoint.

Why this answer

IP SLA is the correct tool to measure packet loss, latency, and jitter on the path to the CUCM cluster. It uses synthetic traffic to monitor network performance metrics continuously, providing accurate measurements for voice and video readiness. This is essential for verifying network readiness for TelePresence endpoints.

Exam trap

CLCOR often tests the confusion between monitoring tools like NetFlow and performance measurement tools like IP SLA, where candidates might choose NetFlow for latency measurement when it does not provide that.

How to eliminate wrong answers

Option A is wrong because NetFlow provides traffic flow data but does not measure latency and jitter directly; it's used for traffic analysis and accounting. Option B is wrong because SNMP is used for monitoring device health and interface statistics, not for measuring end-to-end network performance metrics like jitter. Option D is wrong because SPAN is used for port mirroring to capture packets for analysis, but it does not generate performance metrics; it requires manual analysis.

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40
MCQeasy

Which Cisco feature is used to ensure voice traffic is prioritized over web traffic on a Cisco access switch?

A.SNMP Traps
B.Quality of Service (QoS)
C.Spanning Tree Protocol
D.VLAN Trunking
AnswerB

Quality of Service (QoS) classifies and marks voice packets, then schedules them into a priority queue ahead of web traffic, satisfying the requirement to prioritise voice on the access switch. Cisco access switches use QoS mechanisms such as classification, marking, and queuing to guarantee voice gets strict priority treatment.

Why this answer

Quality of Service (QoS) is the Cisco feature set used to classify, mark, and prioritize voice traffic over less delay-sensitive traffic like web browsing on an access switch. QoS mechanisms such as classification, marking (DSCP EF for voice), queuing, and policing ensure voice packets are forwarded with low latency and jitter. This is the standard answer for prioritizing voice over data on a Cisco access switch.

Exam trap

CLCOR often tests the misconception that VLANs or STP provide prioritization, when only QoS mechanisms actually classify and prioritize voice traffic.

How to eliminate wrong answers

Option A is wrong because SNMP Traps are used for monitoring and alerting, not for traffic prioritization. Option C is wrong because Spanning Tree Protocol prevents Layer 2 loops and manages redundant paths, but does not prioritize traffic types. Option D is wrong because VLAN Trunking (802.1Q) carries multiple VLANs over a link but does not itself provide prioritization; QoS must still be applied to prioritize voice within or across VLANs.

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41
MCQeasy

In a Cisco CUCM cluster, which node is responsible for managing the database and performing configuration updates?

A.Publisher
B.Subscriber
C.TFTP node
D.Music on Hold (MoH) node
AnswerA

The Publisher is the sole node holding the master, read-write database. All configuration changes are written there first, then replicated to subscribers. This makes it the authoritative source for database management and configuration updates across the CUCM cluster.

Why this answer

The Publisher node in a CUCM cluster is the only node that owns the master database and is responsible for all configuration changes. Subscribers receive replicated copies of the database and are read-only for configuration purposes.

Exam trap

CLCOR often tests the assumption that any node can perform configuration changes, when only the Publisher has write access to the master database.

How to eliminate wrong answers

Option B is wrong because Subscribers only maintain replicated copies of the database and cannot perform configuration updates; they serve call processing and redundancy. Option C is wrong because the TFTP node (often colocated with the Publisher) only serves configuration files and firmware to endpoints, not database management. Option D is wrong because the MoH node only streams music on hold audio streams and has no database or configuration role.

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