Cisco · Free Practice Questions · Last reviewed May 2026
36 real exam-style questions organised by domain, each with the correct answer highlighted and a plain-English explanation of why it's right — and why the others are wrong.
An endpoint sends a SIP REGISTER request. CUCM returns a 401 Unauthorized response. What is the next logical step in the SIP signaling flow for the endpoint?
Switch to SCCP
Resend REGISTER with Authorization header
A 401 Unauthorized response is a challenge demanding credentials, so the endpoint must resend REGISTER including an Authorization header carrying its digest credentials. This directly satisfies the challenge-response authentication constraint in the stem, completing the SIP registration handshake rather than retrying blindly or abandoning the attempt.
Terminate the TCP connection
Send a SIP INVITE
When configuring an IP phone for automatic provisioning, which DHCP option is most commonly used to provide the TFTP server address to the phone?
Option 150
Option 150 returns the TFTP server address as a single IP in the vendor-specific field, which Cisco IP phones read directly for provisioning. Option 66 carries it as a hostname string, which phones do not reliably parse.
Option 43
Option 66
Option 12
You are troubleshooting a scenario where G.711 is being used instead of the expected Opus codec between two endpoints in the same region. What is the most likely cause in the CUCM configuration?
Audio Codec Preference List
The Audio Codec Preference List defines the ordered codecs an endpoint offers during negotiation. If G.711 ranks above Opus on that list, the endpoints select G.711 despite Opus being available, directly explaining the unexpected codec in the same region.
Phone NTP settings
SIP Trunk Security Profile mismatch
Location-based bandwidth limit
Which protocol is primarily used for the transmission of real-time voice and video data between two endpoints?
SCCP
TFTP
RTP
RTP carries real-time voice and video media between endpoints, running over UDP to avoid retransmission delays. It provides the sequencing, timestamps and payload type identification needed for streaming audio and video, while signalling protocols such as SIP establish the session.
SIP
You are monitoring RTP traffic using a protocol analyzer. You observe that the Payload Type field in the RTP header is set to 0. Which codec is currently in use?
G.729
Opus
G.711 PCMU
Payload Type 0 is statically mapped to G.711 µ-law (PCMU) in the RTP audio/video profile, so the analyser reading confirms PCMU is in use. The PT field is the header's codec identifier, and value 0 is reserved for this 64 kbps narrowband codec.
G.711 PCMA
You are troubleshooting intermittent audio quality. You notice that RTCP packets are arriving with high variance. What does this indicate?
Latency
Codec Mismatch
Packet Loss
Jitter
RTCP receiver reports carry interarrival jitter statistics; high variance in packet arrival times relative to the RTP timestamp indicates jitter, the variation in packet spacing. This is distinct from loss or latency, and it explains the intermittent audio degradation being troubleshot.
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Practice this domainA 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?
class-map match-any VOICE; match protocol sip; policy-map VOICE-POLICY; class VOICE; queue-limit 100
class-map match-all VOICE; match dscp cs4; policy-map VOICE-POLICY; class VOICE; bandwidth 1000
class-map match-any VOICE; match dscp ef; policy-map VOICE-POLICY; class VOICE; fair-queue
class-map match-any VOICE; match dscp ef; policy-map VOICE-POLICY; class VOICE; priority 1000
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.
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?
Option 66
Option 128
Option 150
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.
Option 43
When designing an NTP deployment for a Cisco Collaboration environment, what is the best practice for time synchronization across CUCM nodes?
Point all CUCM nodes to the same external NTP stratum 1 or 2 servers.
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.
Configure NTP only on the publisher node.
Point each CUCM node to a different NTP server for redundancy.
Use the publisher as the NTP server for all subscriber nodes.
Which design principle must be followed when implementing a centralized call processing deployment with remote branch offices connected via a WAN?
Enable SRST on the local branch gateway to handle call processing during a WAN outage.
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.
Deploy a dedicated CUCM node in every branch office.
Configure H.323 gateways at each site to communicate directly with all remote phones.
Use RSVP to guarantee bandwidth across the WAN for signaling.
You are configuring DNS SRV records to support Jabber service discovery. Which domain format is correct for the _collab-edge._tls record?
_tls._collab-edge.example.com
collab-edge.example.com._tls
_collab-edge._tcp.example.com
_collab-edge._tls.example.com
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.
Which network condition is the most significant contributor to poor voice quality in a VoIP environment?
Low CPU usage on the router
Static IP addresses
Jitter
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.
High bandwidth usage
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Practice this domainYou are implementing a QoS policy on a WAN edge router. You have a requirement to prioritize video traffic. Which queueing mechanism is recommended by Cisco to minimize jitter and latency for video streams?
Weighted Fair Queuing (WFQ)
Priority Queuing (PQ) only
Random Early Detection (RED)
Low Latency Queuing (LLQ)
LLQ assigns video traffic to a strict-priority queue with a configured bandwidth guarantee, so latency-sensitive packets are drained ahead of other classes. This directly minimises jitter and latency, satisfying the stem's video prioritisation requirement, while the policed priority limit prevents starvation of remaining traffic classes.
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?
show class-map configuration
show ip rtp statistics
show qos interface
show policy-map interface
The service-policy is applied on the ingress interface, so show policy-map interface displays per-class matched and marked packet counters there. This confirms whether the class-map's marking (for example DSCP EF for voice) is actually being applied, revealing why one-way audio occurs.
What is the default behavior for DSCP markings on traffic originating from a Cisco IP Phone that has not been configured with specific QoS policies?
Marked as DSCP 34 (AF41)
Traffic is not marked until a policy-map is applied
Marked as DSCP 46 for voice and CS3 for signaling
Absent configured QoS policies, the phone applies its default markings: voice media as DSCP 46 (EF) for low-latency treatment and call signalling as CS3, allowing the network to prioritise voice traffic out of the box.
Marked as DSCP 0 (Best Effort)
You are configuring QoS for a Cisco Unified Communications Manager (CUCM) deployment. You need to ensure that audio media traffic between endpoints is marked correctly. Which DSCP value should be assigned to RTP audio traffic according to the Cisco SRND best practices?
DSCP 46 (EF)
DSCP 46 maps to Expedited Forwarding, the per-hop behaviour the Cisco SRND specifies for voice RTP media. EF provides low latency, low jitter and guaranteed bandwidth through priority queuing, matching the marking requirement for audio media between endpoints.
DSCP 34 (AF41)
DSCP 24 (CS3)
DSCP 0 (BE)
Which DSCP value is recommended for Call Signaling traffic in a Cisco Unified Communications network?
AF31 (26)
EF (46)
CS3 (24)
CS3 (24) is the standard DSCP marking for Call Signaling traffic in Cisco Unified Communications networks, matching the voice deployment model's per-hop behaviour classes. This satisfies the stem's constraint: the recommended DSCP value for Call Signaling.
AF41 (34)
When using 'RSVP' for Call Admission Control (CAC), what happens if the available bandwidth on the path is less than the requested amount?
The call is placed but with a lower quality codec
The call is put on hold until bandwidth is available
The call proceeds, but marks the packets with Best Effort
The call is blocked
RSVP signals the requested bandwidth along the path; when a router finds insufficient unreserved capacity, it returns a PathErr and the admission decision rejects the flow, so the call setup is blocked rather than rerouted or downgraded.
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Practice this domainYou are configuring a Webex Calling environment for a customer that requires integration with an existing on-premises PBX using a Cisco Unified Border Element (CUBE). Which gateway type should you register to Webex Control Hub to enable this connectivity?
Cloud Connected PSTN Gateway
Local Gateway
Registering a Local Gateway in Webex Control Hub provisions a premises-based gateway that connects Webex Calling to your on-premises PBX through the CUBE, satisfying the requirement for local PSTN or PBX integration rather than cloud-delivered connectivity.
Webex Edge Connect Gateway
Virtual Gateway
You are deploying Webex Calling with Local Gateway. Which component must be configured to allow PSTN connectivity for users who have migrated from on-premises CUCM to the cloud?
Webex Edge Connect
SIP Trunk via Local Gateway
The Local Gateway registers to Webex Calling and connects to the PSTN through a SIP trunk, providing the required voice path for migrated users. Without that trunk configuration, the gateway has no carrier connectivity, so cloud-registered endpoints cannot place or receive external calls.
Webex Calling Cloud Connected PSTN
Unified Border Element (CUBE) API
You are deploying Webex Hybrid Data Security (HDS). After deploying the virtual machines for the HDS nodes, you find they are not appearing in Control Hub. What is the first step you should take to troubleshoot the registration?
Disable the firewall on the HDS node.
Restart the Cisco Webex Hybrid Services host.
Check the local log files in the HDS node using the CLI.
HDS nodes log registration attempts locally, so the CLI log files on the node reveal why Control Hub registration fails, such as connectivity, certificate or configuration errors. Checking them first establishes the actual failure cause before altering network or platform settings.
Re-download the management certificate.
You need to ensure that the Webex Hybrid Calendar service can access an Exchange 2019 environment. Which authentication method is recommended by Cisco for this deployment?
Digest Authentication
OAuth 2.0
OAuth 2.0 satisfies Cisco's requirement for Webex Hybrid Calendar against Exchange 2019, since on-premises deployments must use the Modern Authentication path rather than basic authentication, which Microsoft has deprecated. OAuth 2.0 delegates access via tokens issued by Microsoft Entra ID, enabling the calendar connector to read and write mailboxes without storing user credentials.
Basic Authentication
Kerberos Constrained Delegation
Where in Webex Control Hub do you go to view the health status of all deployed Hybrid Services connectors?
Devices > Management
Analytics > Reports
Services > Hybrid
The health status of all deployed Hybrid Services connectors is viewed under Services > Hybrid in Webex Control Hub. That page lists each connector, its cluster and its current state, letting administrators confirm whether on-premises components are running or degraded.
Users > Settings
When configuring Hybrid Call Service for Cisco Unified Communications Manager, which specific requirement must be met for the Expressway-C and Expressway-E pair?
They must be configured with a Unified Communications traversal zone.
The 'Unified Communications' traversal zone is mandatory for Hybrid Call Service interworking.
They must use the same certificate authority as the CUCM nodes.
They must use a public IP address on the Expressway-C.
They must be in the same subnet.
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Practice this domainA user reports that their calls are failing when they roam to a different physical office site. You observe that the device is registered to the primary CUCM cluster but the Location-based CAC is blocking the call. What is the most likely cause?
The device is not assigned to a Device Pool with the correct Location configured for the new site.
Location-based CAC derives its bandwidth budget from the Location configured on the device pool. If the roaming device keeps its original site's device pool, CUCM applies the wrong Location's limits, blocking the call. Assigning a device pool with the new site's Location corrects the calculation.
The Region setting in the Device Pool has a lower bitrate than the required codec.
The device must be re-registered to a local subscriber instead of the primary cluster.
The Calling Search Space is not configured to allow inter-site dialing.
You are troubleshooting a scenario where internal calls between two offices are using the G.711 codec, but you want them to use G.729 to save bandwidth. Where should you change the codec setting?
Device > Device Settings > Common Device Configuration
System > Region Information > Region
System > Region Information > Region sets the codec for calls within a region, which matches the inter-office internal call constraint. Regions define bandwidth management and codec policy per site, so assigning both offices to one region and selecting G.729 there forces internal calls to negotiate G.729 instead of G.711.
System > Device Pool
System > Location
When designing a redundant CUCM deployment, which mechanism ensures that if a primary subscriber fails, phones automatically register to a secondary subscriber?
SIP Trunk Security Profile
Route List
CM Group
A CM Group defines an ordered list of subscriber nodes that phones use for registration. If the primary subscriber becomes unreachable, phones fall back to the next subscriber in the group, providing automatic registration redundancy.
Device Pool
You need to prevent certain internal users from making international calls while allowing them to dial local numbers. What is the most efficient way to implement this in CUCM?
Use an Application Dial Rule.
Modify the device pool's location.
Assign a CSS to the device that includes a Partition containing local routes but excludes a Partition containing international routes.
Class of Service controls which partitions a device may dial, so a CSS containing the local-route partition but omitting the international partition blocks international dialling while permitting local calls. This is applied once at device level rather than per-line or via translation patterns.
Configure a Route Filter on the Route Pattern.
If a Translation Pattern is configured in CUCM, at what stage of the call processing does the digit manipulation occur?
After the call is sent to the gateway.
At the device level during registration.
Only when the call fails to route.
Before the Route Pattern is matched.
CUCM applies Translation Pattern digit manipulation during digit analysis, before the transformed digits are matched against a Route Pattern. This ordering lets the translation rewrite the called number so the subsequent route lookup uses the altered digits.
You are implementing a centralized dial plan. You have multiple sites, each with local PSTN gateways. How can you ensure that an emergency call from a specific site is always routed to the local PSTN gateway of that site?
Use Transformation Masks on the device.
Configure the location in the device pool to match the site.
Create a separate CUCM cluster for each site.
Use site-specific partitions and assign the local gateway to a CSS that includes that partition.
Site-specific partitions isolate each site's emergency number, and assigning only that site's local gateway to the CSS forces calls in that partition out through the local PSTN gateway. This guarantees the emergency call reaches the correct local emergency services rather than a remote gateway.
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Practice this domainYou are deploying CUBE. Which command enables the CUBE functionality on the Cisco IOS XE router?
enable cube
mode border-element
Entering voice service configuration and issuing mode border-element switches the router into Cisco Unified Border Element mode, activating CUBE's session border controller functions such as SIP trunking, dial-peer handling and media interworking on the IOS XE platform.
voice-service enable
service cube
A Cisco IOS XE gateway is failing to route calls because the dial-peer is not matching the incoming call. Which command helps you identify the dial-peer that matched the incoming call?
show voice call status
show dial-peer voice summary
debug ccsip messages
debug voip dialpeer all
debug voip dialpeer all traces dial-peer matching, showing which peer is selected for the incoming call and why others were rejected. This directly reveals the mismatch causing the routing failure, unlike show commands that only display configuration.
When configuring CUBE for media flow-around, which command ensures the media goes directly between endpoints?
media flow-around
The media flow-around command configures the CUBE to pass media directly between endpoints rather than anchoring it through the gateway. This satisfies the requirement that RTP flows bypass the CUBE, leaving only signalling processed by the router.
media pass-through
media-bypass
media flow-through
What is the primary function of a dial-peer destination-pattern?
To define the codec
To specify the gateway IP
To match the dialed number
The destination-pattern matches the dialled number string so the router selects the correct outbound dial peer for that call. It functions as the matching criterion, while session target, codec and digit manipulation are configured separately on the same peer.
To enable SIP
When troubleshooting CUBE SIP normalization, which tool allows you to view the normalized SIP messages in real-time?
show voice class sip-profiles
debug ccsip messages
CUBE applies SIP normalization profiles on inbound and outbound dial peers, rewriting headers and SDP. The debug ccsip messages command traces these SIP messages as they are parsed and normalised in real time, exposing the before-and-after header transformations needed to verify the normalisation rules are firing correctly.
debug voip ccapi inout
debug sip-ua
show sip-profile
Which component in CUBE is responsible for ensuring that the call session is terminated correctly?
Dial-peer
Voice translation-profile
SIP Stack
The SIP Stack handles SIP signalling messages, including BYE requests that tear down sessions. It ensures the call session terminates correctly by processing the BYE transaction and confirming the dialog ends cleanly, satisfying the stem's requirement for correct session termination within CUBE.
Voice-class codec
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Practice this domainThe vendor does not publish an exact question count for the CLCOR; the exam lasts 120 minutes. Always confirm current details on the vendor's official exam page.
CLI output interpretation, network topology analysis, routing behaviour, switching concepts, troubleshooting, and configuration questions.
The exam covers 6 domains: Protocols And Endpoints, Infrastructure And Design, Media And Qos, Cloud And Hybrid Services, ON Premises Call Control, Voice Gateways And Session Border Controllers. Questions are weighted by domain — higher-weight domains appear more on your actual exam.
No. These are original exam-style practice questions written against the official Cisco CLCOR exam objectives. They are not copied from the real exam. Courseiva focuses on genuine understanding, not memorisation of braindumps.
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