Cisco Implementing Cisco Collaboration Core Technologies (CLCOR, 350-801, v2.0 blueprint, CCNP Collaboration) (CLCOR) (CLCOR) — Questions 151–225

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.

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.

The Calling Search Space (CSS) determines which partitions a device can search for dialed numbers. If the CSS does not include the partitions where the route patterns or directory numbers reside, the phone cannot match any dialed digits, resulting in a fast busy tone. This is a common misconfiguration when a phone can register but cannot make calls.

A class map defines match criteria such as ACLs, DSCP values or protocol ports, identifying which packets belong to a traffic class. It performs classification only; the policy map then applies actions, so the class map's primary function is categorising and matching specific traffic flows.

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.

The SRST Reference in the Device Pool points phones to the local router acting as the survivable call-control fallback. When the CUCM WAN link fails, phones use this reference to register with the SRST gateway, satisfying the failover requirement.

A User Group in CUCM is the primary container for a specific set of users, grouping them for shared feature access, permissions and role assignments. It satisfies the stem's requirement by acting as the direct organisational container, unlike roles or partitions which serve different functions.

In Control Hub, the Monitoring section includes Hybrid Services, where you can view the health and status of each deployed hybrid service connector. This page shows whether connectors are up, down, or in an error state. It is the central place to monitor the operational status of Hybrid Calendar, Message, Calling, and other services. This is the correct answer.

When a Translation Pattern transforms the called number, CUCM performs a second lookup using the Called Party Transformation CSS configured on that Translation Pattern. This CSS determines which partitions are searched for the new digits, allowing the administrator to route the translated number differently from the original dialed digits.

The Connector status reflects the Expressway-C's connection to the Cisco cloud. If outbound TCP 443 is blocked, the connector cannot register and shows Down, making this the most likely cause of the failed Hybrid Calendar setup.

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.

The Cisco Directory Connector is a Windows service that synchronizes users from Active Directory to Webex. If the service is stopped or not registered correctly, the connector status in Control Hub will show 'Not Connected'. Ensuring the service is running and registered with the correct credentials is essential for synchronization.

A Cisco Unified CM Group defines an ordered list of nodes that devices use for registration failover, so if the primary node fails, endpoints re-register with the next listed node. This satisfies the stem's requirement by supplying the device-side redundancy mechanism within CUCM architecture.

A voice translation-profile with a translation rule can match the '+1' prefix and strip it, modifying the called number before the call is routed. This is the standard method for number manipulation on CUBE. Applying it to the inbound dial-peer from the provider ensures the '+' is removed before the call is sent to Unified CM.

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.

SIP 488 Not Acceptable Here signals that the offered SDP media attributes, typically the codec list, cannot be matched by the receiving endpoint. A codec mismatch between the CUBE and the far end therefore causes calls to be rejected after signalling succeeds.

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.

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.

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.

Local Gateway (LGW) allows Webex Calling to route PSTN calls to an on-premises CUBE registered to Webex. This is the recommended hybrid PSTN option when an enterprise wants to retain its existing voice gateway and PSTN circuits. The CUBE acts as the Local Gateway, providing a secure SIP trunk between Webex Calling and the on-premises PSTN. It is a supported and common deployment for hybrid calling.

The SIP ALG inspects the Session Description Protocol inside SIP messages and rewrites embedded connection and media addresses so the public address is signaled instead of the private one. This lets the provider send RTP to the NAT-translated address, restoring two-way audio for the internal phone behind the router.

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.

A Translation Pattern manipulates digits by matching a dialled string and transforming it into a different number before routing. It satisfies CUCM digit manipulation by rewriting called-party digits at a specific partition and CSS scope.

EF (Expedited Forwarding), which corresponds to DSCP 46, is the recommended marking for voice bearer traffic. It indicates that packets require low latency, low jitter, and low loss, and network devices should provide priority queueing for EF-marked traffic. This ensures that voice packets are transmitted first during congestion, maintaining voice quality. Cisco's QoS baseline recommends EF for voice bearer.

This SIP profile command copies the Diversion header value to a new History-Info header in the outbound INVITE. It effectively converts the Diversion header to History-Info, allowing CUCM to receive the redirecting number in the expected header. This is a common normalization technique on CUBE for interworking between different SIP implementations.

A route list groups gateways into an ordered priority sequence. When the primary gateway is busy or unreachable, Cisco Unified Communications Manager automatically reroutes the call to the next available gateway in the list, providing the required failover.

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.

Cisco Unified Serviceability hosts the Alarm and Trace Configuration interface, letting administrators define alarm events, trace levels and log outputs for troubleshooting. This is the tool's dedicated function, satisfying the requirement for a feature managed there rather than in Unified CM Administration.

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.

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.

Call Pickup works by dialling the pickup group's feature code or pressing the Pickup softkey, which redirects the ringing call from another group member's line to the answering phone. This satisfies the scenario of retrieving a call ringing on a colleague's phone within the same group.

DHCP (Dynamic Host Configuration Protocol) is used by Cisco IP Phones to obtain an IP address, subnet mask, default gateway, and TFTP server address. Without DHCP, the phone cannot communicate on the network. In this scenario, the phone is not obtaining an IP address, so DHCP is the protocol that should be providing it. This is a fundamental step in the phone boot process.

Softkey templates define which softkeys appear during call states, including Transfer. The template is assigned to the device, so checking Device > Device Settings > Softkey Template confirms whether a template containing Transfer is applied to that phone.

The debug dialpeer command shows dial-peer matching details for both inbound and outbound calls, including which dial-peer was selected and why. In this scenario, it will reveal whether the inbound call matched the intended POTS dial-peer or fell through to a default, causing the busy condition.

Moving a phone between CUCM nodes does not rewrite its Device Pool, so Region and Location values persist from the original configuration. The device registers successfully but keeps stale bandwidth and codec settings until the administrator reassigns the Device Pool.

Removing a node from a Hybrid Video Mesh cluster forces its hosted conferences to be redistributed or torn down. Because the node is actively serving media, those meetings drop or suffer disruption until participants reconnect via remaining nodes or the cloud.

Registering to Webex gives the Room device native access to cloud-only capabilities such as whiteboarding and AI-driven metrics, which on-premises CUCM registration cannot deliver. This satisfies the hybrid deployment's requirement for full cloud feature parity.

Common Device Configuration groups shared device-level settings, including softkey templates, phone button templates and softkey layout, so they can be applied across many phones rather than configured individually. This satisfies the stem's requirement by centralising reusable configuration, reducing administrative effort when deploying or updating large numbers of Cisco IP phones.

Calling capabilities are governed by the licence assigned to each user in Control Hub, not by which app interface they open. The Webex Calling app and Webex app both surface whatever telephony features that entitlement grants, so the licence is the determining factor.

EF (Expedited Forwarding) is the recommended DSCP for voice bearer traffic. It provides low loss, low latency, low jitter, and assured bandwidth. The service provider expects EF for voice, ensuring proper treatment across the network. Configuring IP phones to mark voice media as EF aligns with industry best practices.

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.

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.

Packet loss directly degrades video quality: dropped RTP packets cause macroblocking, freezing and audio artefacts, since real-time video cannot retransmit within its latency budget. It is one of the three factors, alongside latency and jitter, that determine call quality.

A Device Pool in Cisco Unified Communications Manager can include a Location, which enforces call admission control by limiting the bandwidth available to devices at that site. This satisfies the pool's CAC requirement, preventing oversubscription of WAN links.

Within sip-ua configuration mode, the timers expires command sets the SIP registration expiry timer in seconds. Setting it to 60 causes the gateway to send a new REGISTER refresh every 60 seconds, which matches the provider's requirement. This is the correct and standard way to adjust the registration expiry interval on a Cisco IOS XE voice gateway, ensuring the registration stays active according to the provider's policy.

Cisco CallManager (the CallManager service) is the SIP registrar and line-side call-control component in CUCM, so it owns device registration, authentication and keepalive for SIP endpoints. Other services handle TFTP configuration, database replication or media resources, not registration.

A Route Pattern in CUCM defines how dialed digits are routed to external destinations. It matches the dialed string and points to a Route List or Gateway. Without a Route Pattern, the call cannot be routed outside the cluster. For external dialing, the Route Pattern must be configured with the appropriate dial plan, such as 9.@ for outside access. This is essential for the Room Kit Pro to reach external numbers.

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.

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.

CUBE initiates an outbound TLS connection to the Webex Cloud, which is kept open and reused for signalling and media. Because the connection originates from on-premises, no inbound firewall ports are required, satisfying the stated constraint.

The calling search space (CSS) on the directory number determines which partitions the phone can reach when dialing. To allow PSTN calls, the CSS must include the partitions that contain the route patterns for local and international dialing. Without the correct CSS, the phone cannot match those route patterns, and calls will fail with a reorder tone.

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.

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.

SIP carries the call signalling — registration, invite and session setup — between Webex Room devices and CUCM. Registering on-premises requires this signalling channel, so SIP satisfies the stem's requirement, while media itself flows separately over RTP.

Cisco IP phones register with Cisco Unified Communications Manager using either SCCP, Cisco's proprietary signalling protocol, or SIP, the standards-based alternative. Both are supported registration protocols, so the option correctly names the pair the phones rely on.

To apply different DSCP markings to voice and signaling, the router must first identify each traffic type. A class-map with match statements such as 'match ip dscp' is not sufficient if traffic is unmarked; instead, matching on UDP port ranges (e.g., 16384-32767 for RTP) or on protocols like SCCP/SIP is needed. This class-map then feeds into the policy-map where 'set dscp ef' and 'set dscp cs3' are applied.

SIP over TLS is used for secure signaling between Cisco Webex Room devices and CUCM. When encryption is enabled, the device uses TLS to encrypt SIP messages, ensuring that signaling information such as call setup and teardown is protected. This is part of a secure collaboration environment and is supported by CUCM for endpoints that support TLS, including Webex Room devices.

The Key Management Service holds and manages the encryption keys used by the HDS nodes, keeping key custody within the enterprise rather than with Cisco. This satisfies the requirement that the organisation retains control over its encryption keys.

The IM and Presence Service on-premises is required to provide instant messaging and presence for CUCM users. The Hybrid Message Service connector integrates this on-premises service with Webex, allowing users to communicate with Webex users. Without it, there is no on-premises IM and presence to extend to the cloud.

The CUBE matches inbound calls against incoming dial peers; if none matches the provider's INVITE, or the peer lacks the correct destination pattern and voice translation rules, call admission fails and the CUBE returns 403 Forbidden rather than routing to the Local Gateway.

The Expressway-C connector is critical for Hybrid Call Service Connect because it handles call signaling between Webex and CUCM. If users can see presence but cannot make calls, the Expressway-C may not be properly registered or the call connector may be down. Checking its status and logs is the first step to isolate signaling issues.

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.

SIP profiles provide header manipulation: adding, removing, or modifying headers on inbound and outbound messages. They do not automatically derive the original called number and place it into a Diversion header for forwarded calls. That task generally requires additional logic such as a normalization script or specific CUBE features designed for diversion handling.

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.

Number porting transfers existing PSTN numbers from the current carrier to Webex Calling, preserving the digits users already know. This directly satisfies the requirement to keep existing numbers, unlike provisioning new numbers or forwarding, which would change them.

Used for header normalization.

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.

Hybrid Media Service logs record call routing decisions between on-premises and cloud media paths, showing whether media is delivered to the correct cloud node. They directly satisfy the need to verify correct cloud routing during troubleshooting.

Changing a device's Location in Cisco Unified Communications Manager reassigns its device pool, which governs the regional settings applied to that device. Location determines the maximum audio and video bandwidth permitted between devices, so video bandwidth limits shift accordingly. This satisfies the stem's constraint that Location changes affect bandwidth parameters.

Cisco IP Phones are designed to mark voice media as EF and signaling as CS3. When connected to a switch, the switch port can be configured to trust the DSCP markings from the phone. This simplifies QoS deployment and is a recommended practice in Cisco collaboration networks.

AF41 (Assured Forwarding Class 4, Drop Precedence 1) is the recommended DSCP for interactive video traffic in Cisco collaboration networks. It provides a higher priority than best effort but lower than voice, ensuring that video is prioritized over data while not competing with voice. For a Webex Room Kit sending 1080p video, marking with AF41 ensures appropriate queuing and bandwidth guarantees in the QoS policy.

The SIP Trunk Security Profile on the trunk controls TLS signalling encryption, determining whether the trunk uses a secure transport and which certificate trust settings apply. Without it, SIP signalling between CUCM and CUBE remains unencrypted regardless of other trunk parameters.

Webex Edge Connect requires a dedicated circuit provisioned through a certified Cisco Webex Edge Connect partner, who coordinates the physical cross-connect into Cisco's peering fabric. Cisco does not order the circuit directly, so partner involvement satisfies the dedicated-connection requirement.

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.

LLQ combines strict priority queuing with a policer. It places voice traffic in a priority queue that is serviced before other queues, ensuring low latency and jitter. The policer limits the amount of traffic that can enter the priority queue, preventing starvation of other queues. Configuring LLQ with a bandwidth limit of 1 Mbps meets the requirement exactly.