CLCOR Infrastructure And Design Practice Test 3 — 15 Questions

Question 1

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

Accepted Answer

Enable SRST on the local branch gateway to handle call processing during a WAN outage.. 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.

Answer

Deploy a dedicated CUCM node in every branch office.

Answer

Configure H.323 gateways at each site to communicate directly with all remote phones.

Answer

Use RSVP to guarantee bandwidth across the WAN for signaling.

Question 2

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

Accepted Answer

_collab-edge._tls.example.com. 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.

Answer

_tls._collab-edge.example.com

Answer

collab-edge.example.com._tls

Answer

_collab-edge._tcp.example.com

Question 3

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

Accepted Answer

Jitter. 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.

Answer

Low CPU usage on the router

Answer

Static IP addresses

Answer

High bandwidth usage

Question 4

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?

Accepted Answer

IP SLA. 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.

Answer

NetFlow

Answer

SNMP

Answer

SPAN

Question 5

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 6

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?

Accepted Answer

Ensure SRTP is enabled on the SIP Trunk Security Profile.. 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.

Answer

Disable SIP inspection on the firewalls.

Answer

Configure a VPN between the two clusters.

Answer

Enable MTP (Media Termination Point) for all calls.

Question 7

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

Accepted Answer

G.729. 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.

Answer

G.711

Answer

AAC-LD

Answer

G.722

Question 8

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?

Accepted Answer

Configure a CUCM Group in the Device Pool with multiple servers.. 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.

Answer

Set the 'Call Processing Load' value in the System settings.

Answer

Enable Auto-registration for all phones.

Answer

Configure the cluster-wide failover parameter in the Service Parameter page.

Question 9

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

Accepted Answer

Cisco Unified Communications Manager. 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.

Answer

Cisco Expressway

Answer

Cisco Unity Connection

Answer

Cisco CUBE

Question 10

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

Accepted Answer

150ms. 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.

Answer

300ms

Answer

500ms

Answer

50ms

Question 11

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

Accepted Answer

Publisher. 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.

Answer

Subscriber

Answer

TFTP node

Answer

Music on Hold (MoH) node

Question 12

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

Accepted Answer

Use a trusted third-party CA for all servers.. 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.

Answer

Use self-signed certificates on all nodes.

Answer

Disable certificate verification on all endpoints.

Answer

Use the Publisher as the CA for the entire network.

Question 13

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

Accepted Answer

Use a dedicated Voice VLAN.. 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.

Answer

Use the default VLAN 1 for all voice and data.

Answer

Use a separate physical switch for phones.

Answer

Do not use VLANs; use static routing.

Question 14

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 15

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

Accepted Answer

External Load Balancer. 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.

Answer

Cisco CUBE redundancy

Answer

Cisco UCM Group

Answer

DNS Round Robin only