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Question 1:
An engineer is configuring a Cisco Unified Communications Manager cluster for high availability. Which service must be activated on all nodes in the cluster to enable database replication?
A) Cisco Database Layer Monitor
B) Cisco DRF Local
C) Cisco CallManager
D) Cisco Unified Serviceability
Answer: A
Explanation:
The Cisco Database Layer Monitor service is essential for maintaining database replication across all nodes within a Cisco Unified Communications Manager cluster, and its role becomes even more significant as deployments grow in size and complexity. By constantly monitoring the health, connectivity, and synchronization status of the Informix-based database layer, this service ensures that all configuration, feature, and operational data are accurately distributed between the publisher and all subscriber servers. Because CUCM relies on a distributed database architecture, every node must have access to consistent and updated information in order to perform tasks such as device registration, call routing, feature activation, and user provisioning. When the Database Layer Monitor service is activated on all nodes in the cluster, it enables continuous verification of replication status and immediate detection of issues that could lead to data divergence. This proactive monitoring is a critical component of CUCM’s high availability strategy, as it ensures that any modifications made on the publisher—whether they involve new endpoints, dial-plan updates, or security and mobility settings—are reliably replicated to subscriber nodes without delay.
The Database Layer Monitor goes beyond simply verifying that replication is active; it also evaluates replication queues, database table status, connectivity between nodes, and the health of key processes required for maintaining synchronization. If any irregularity is detected, such as a node falling out of sync, a replication queue buildup, or a communication problem between the publisher and a subscriber, the service generates alerts that appear in Cisco Unified Serviceability and the Real-Time Monitoring Tool. These alerts equip administrators with actionable diagnostic information that can help resolve replication issues before they escalate into outages or inconsistencies. Without this service running properly, database tables could begin to drift out of alignment, causing subscriber nodes to base their operations on outdated information. Such inconsistencies can lead to unpredictable behavior, including registration failures, mismatched dial plans, incorrect call routing decisions, and feature malfunctions. Over time, the lack of replication integrity can severely degrade cluster reliability and compromise overall system stability.
It is important to distinguish the Database Layer Monitor from other CUCM services that might appear related but serve entirely different purposes. For example, Cisco DRF Local is involved specifically in disaster recovery operations, such as managing backup and restore files, but it does not participate in ongoing database synchronization. Unlike the Database Layer Monitor, it does not ensure that updated data is continuously delivered to all subscriber servers. Similarly, Cisco CallManager, while arguably the most critical service for call processing, device control, and signaling, plays no role in managing or validating database replication. Its functionality depends heavily on the assumption that the underlying database is synchronized and reliable; however, it does not handle the replication mechanism itself. Cisco Unified Serviceability, on the other hand, provides a framework for monitoring, troubleshooting, and managing the various services within CUCM but does not directly maintain or coordinate database replication across the cluster.
Proper activation and configuration of the Database Layer Monitor ensures that the entire CUCM environment operates as a unified and coherent system. Features such as device mobility, corporate directory access, hunt groups, call admission control, and routing patterns all rely on consistent database updates to perform reliably across multiple nodes. When replication is functioning correctly, administrators can make changes on the publisher with confidence that these updates will be propagated to every subscriber node, enabling seamless failover and redundancy. This consistency is especially important in large environments where multiple call processing nodes handle significant volumes of registration and signaling traffic. By maintaining database uniformity, the Database Layer Monitor contributes directly to fault tolerance, operational continuity, and the overall resilience of the Unified Communications deployment.
Question 2:
Which protocol does Cisco Jabber use for presence and instant messaging when integrating with Cisco Unified Communications Manager IM and Presence Service?
A) SIMPLE
B) XMPP
C) SIP
D) SCCP
Answer: B
Explanation:
Cisco Jabber utilizes the Extensible Messaging and Presence Protocol for all presence and instant messaging functions when integrating with the Cisco Unified Communications Manager IM and Presence Service, and this reliance on XMPP is fundamental to how the collaboration ecosystem maintains real-time awareness and communication across users and devices. XMPP, being an open and highly extensible standard, was designed from the ground up to support lightweight, real-time exchanges, making it ideal for presence updates, chat messages, and subscription-based notifications. In the Jabber environment, every user’s presence state—whether they are available, away, busy, or offline—is communicated through XMPP mechanisms that efficiently distribute these updates only to subscribed contacts. When a user changes devices, logs in from multiple endpoints, or updates their status manually, XMPP ensures these changes propagate immediately. This behavior is crucial in modern enterprise environments where users often switch between desktop clients, mobile devices, and virtual desktops while expecting a consistent and synchronized communication experience.
Within this architecture, the Cisco IM and Presence Service operates as the central XMPP server responsible for managing presence clusters, routing messages, and handling subscription requests. The service maintains rosters, contact lists, and presence information while ensuring that each user’s state is accurately reflected across the network. It also manages multi-user chat rooms, message archiving when configured, and various XMPP extensions that support richer communication features. Because XMPP is based on XML streams and supports namespaces, Cisco can incorporate extended functionality—such as enhanced status states, custom presence attributes, and internal policy controls—while remaining interoperable with standard XMPP tools. This extensibility also supports gateway functions, enabling federation with external XMPP domains so organizations can communicate securely with partners, customers, or other business entities without sacrificing consistency or policy enforcement.
Cisco Jabber connects to the IM and Presence Service using XMPP over secure TLS channels, ensuring that all presence subscriptions, user authentication processes, and instant message payloads are encrypted during transit. This security is essential in enterprise environments that require strict compliance with data protection and information governance standards. TLS not only protects message content but also secures credentials and prevents unauthorized access or impersonation attempts. As users increasingly rely on mobile networks, Wi-Fi, and public connections, encrypted XMPP sessions help maintain privacy and integrity regardless of the underlying transport medium. The architecture also supports redundancy, allowing Jabber clients to reconnect seamlessly to alternate IM and Presence nodes if a failure occurs, ensuring high availability and resilient communication paths.
In modern Cisco deployments, SIMPLE—the SIP-based approach to presence and messaging—is not used for Jabber’s IM and presence functions. While SIMPLE once provided an option for SIP-centric environments, Cisco’s implementation standardized on XMPP due to its scalability, extensibility, and superior feature set for enterprise messaging. SIP still plays a major role in Jabber, but specifically for voice and video calling, signaling, and session control through CUCM. These SIP capabilities are entirely separate from the presence and messaging layer. Likewise, SCCP remains relevant for IP phone communication with CUCM but has no interaction with Jabber’s messaging services and is not used by soft clients for presence.
The decision to rely on XMPP ensures that Cisco Jabber can scale effectively across large organizations with thousands of users while maintaining rapid presence updates and dependable message routing. It also supports federation, rich presence, multi-device awareness, and flexible integrations—capabilities that are increasingly important as enterprises shift toward hybrid work models and distributed communication systems.
Question 3:
An administrator needs to configure a SIP trunk between Cisco Unified Communications Manager and a service provider. Which SIP profile setting controls whether the CUCM will accept calls with Replaces headers?
A) Early Offer support for voice and video calls
B) Handle Replaces header
C) Enable OPTIONS ping
D) SIP Rel1XX Options
Answer: B
Explanation:
The Handle Replaces header setting in the SIP profile directly controls whether Cisco Unified Communications Manager will process and accept SIP INVITE messages that include the Replaces header, and its role is fundamental to several advanced SIP call control functions. Defined in RFC 3891, the Replaces header enables one SIP dialog to replace another, providing the mechanism required for standards-based attended call transfers, call pickup, and other scenarios where an existing call leg must be seamlessly taken over by a new signaling session. When this setting is enabled within the SIP profile applied to a trunk or device, CUCM will correctly interpret incoming SIP INVITE messages containing the Replaces header and will allow the new dialog to replace the existing one. This behavior is essential for ensuring interoperability with SIP carriers, PBXs, and endpoints that rely on RFC-compliant call transfer procedures. Many third-party systems, softphones, and Session Border Controllers depend heavily on Replaces support to complete attended transfers, especially in environments where call control must remain consistent across heterogeneous SIP deployments.
When Handle Replaces header is disabled, CUCM will reject SIP INVITE messages that contain a Replaces header, leading to immediate failures in attended transfer attempts. In such situations, users may experience symptoms like failed transfers, dropped calls, incomplete handoffs, or fallback to blind transfer behavior. These issues arise because CUCM cannot match the new SIP dialog to the existing call leg referenced in the Replaces header, effectively breaking the transfer workflow expected by the external SIP device or service provider. This limitation becomes even more pronounced in interworking scenarios involving cloud telephony services, enterprise carriers, or third-party collaboration platforms, all of which commonly use the Replaces mechanism to maintain standards-based interoperability. Therefore, properly enabling the Handle Replaces header option is crucial for seamless communication between CUCM and any SIP entity that uses RFC 3891 to manage call transitions.
Although the Handle Replaces header setting directly affects attended transfer and call replacement behavior, other SIP profile settings may appear related but serve completely different purposes. Early Offer support, for example, determines whether CUCM includes SDP in the initial INVITE message during call setup. This impacts media negotiation, call setup timing, and interworking with carriers that require SDP upfront, but it has no influence on whether CUCM accepts or processes Replaces headers. Likewise, Enable OPTIONS ping configures periodic SIP OPTIONS messages used for monitoring the availability of SIP trunks and endpoints. This keepalive mechanism ensures that CUCM can detect when a trunk or device has become unreachable, but it does nothing to facilitate or restrict call transfer behavior. Another setting, SIP Rel1XX Options, controls the reliability of provisional responses—specifically whether CUCM requires reliable 1xx messages using PRACK. This affects early media and session establishment timing but does not participate in any aspect of dialog replacement or attended transfer negotiation.
Ensuring the correct configuration of the Handle Replaces header setting is therefore critical for maintaining a functional and standards-compliant SIP environment within CUCM. When properly enabled, it allows CUCM to fully participate in SIP attended transfers, coordinated call control, and seamless dialog replacement with both internal and external SIP entities. This capability preserves the expected user experience by supporting smooth call transitions and avoiding transfer failures during interworking with service-provider trunks or third-party SIP devices. By correctly handling the Replaces header, CUCM maintains compatibility with RFC 3891 and upholds the interoperability required in modern, mixed-vendor SIP networks.
Question 4:
Which Cisco Unified Communications Manager feature allows administrators to automatically assign directory numbers and device profiles to users based on LDAP attributes?
A) Bulk Administration Tool
B) Extension Mobility
C) Self-Provisioning
D) User Data Services
Answer: C
Explanation:
The Self-Provisioning feature in Cisco Unified Communications Manager enables automatic assignment of directory numbers and device configurations to users based on their LDAP directory attributes, eliminating the need for administrators to manually configure every user and device during onboarding. This automation is particularly valuable in large or distributed enterprises where thousands of users must be provisioned consistently and efficiently. When Self-Provisioning is enabled and properly configured, CUCM integrates tightly with the organization’s LDAP directory—often Active Directory—by reading specific attributes that define a user’s location, device entitlements, service profiles, and other criteria that determine how the system should assign numbers and apply configuration templates. These LDAP-driven rules support a true zero-touch deployment model: when a user first authenticates or connects their device, CUCM queries the directory, retrieves the relevant user attributes, and assigns directory numbers dynamically from administrator-defined number pools. These number pools can differ by region, site, department, or any logical grouping, enabling highly flexible provisioning even in complex multi-site environments with unique dial plans, calling privileges, and service requirements.
During the initial registration or login process—typically through the Self-Provisioning IVR or an endpoint that supports secure user authentication—CUCM automatically binds the device to the user, applies the correct device profile, and assigns the proper directory numbers and calling permissions. This process substantially decreases provisioning time, which is especially valuable during large organizational expansions, mergers, or onboarding cycles. By eliminating repetitive manual tasks, Self-Provisioning reduces the likelihood of configuration errors that could affect call routing, voicemail access, or device functionality. It also ensures consistent application of organizational standards, as every user receives configurations derived from authoritative HR or identity-management systems rather than relying on administrators to interpret or manually apply individualized settings. Because the system is rules-based and centrally managed, updates to numbering plans or provisioning rules immediately apply to future users without requiring bulk reconfiguration or spreadsheet-driven workflows.
Although the Bulk Administration Tool (BAT) remains useful for mass operations such as uploading device MAC addresses, updating configurations in batches, or modifying large sets of user records, it requires administrators to prepare CSV files manually. BAT does not integrate with LDAP attributes for dynamic or rules-based provisioning, meaning it cannot offer the same automated or scalable approach that Self-Provisioning provides. Similarly, Extension Mobility allows users to log into various physical phones and temporarily apply their personal settings, but it does not perform initial provisioning or assign directory numbers. Extension Mobility operates only after a user and device profile have already been provisioned through standard methods, including Self-Provisioning. User Data Services, which provides corporate directory information and resolution services to endpoints, is also separate from the provisioning process. While it enhances user experience by enabling directory searches and contact lookups, it does not determine how users or devices are created, assigned numbers, or configured within CUCM.
Self-Provisioning represents a structural shift in how organizations deploy collaboration infrastructure, emphasizing automation, consistency, and alignment with identity and HR systems as sources of truth. Instead of relying on administrators to configure every device or assign every directory number, CUCM reads attributes already maintained in the corporate directory, reducing redundant administrative workloads and synchronizing communication services with existing enterprise data. For organizations with distributed campuses, international numbering schemes, or role-based service entitlements, Self-Provisioning greatly improves operational scalability. It supports rapid deployment of new users, enables seamless onboarding during organizational changes, and ensures that device configurations reflect corporate policies automatically. As enterprises move toward zero-touch and identity-driven IT models, Self-Provisioning functions as a foundational capability that aligns communications infrastructure with modern provisioning and automation frameworks.
Question 5:
What is the maximum number of Cisco Unified Communications Manager subscribers that can be configured in a single cluster?
A) 4
B) 8
C) 16
D) 21
Answer: B
Explanation:
Cisco Unified Communications Manager supports a maximum of eight subscriber nodes in addition to the publisher node within a single cluster configuration, resulting in a total of nine nodes when the publisher is included. This limitation reflects a deliberate architectural balance designed to ensure that the cluster maintains high performance, predictable behavior, and reliable database replication under varying operational loads. The core reason behind this constraint relates to the way CUCM manages and distributes its Informix-based database across all nodes. Because the publisher hosts the primary read–write database and all subscribers maintain read-only replicas, any configuration change entered on the publisher must be replicated efficiently to each subscriber. As the number of subscribers increases, replication fan-out and acknowledgment traffic increase proportionally. Allowing more than eight subscriber nodes would significantly elevate replication latency, strain available network bandwidth, and increase the risk of database inconsistency across the cluster. Cisco’s design ensures that all nodes receive timely, accurate updates without overload, preventing scenarios in which subscribers would fall behind and operate with outdated dial plan entries, device configurations, or service data.
The eight-subscriber architecture also preserves cluster stability in large enterprise environments where thousands or even tens of thousands of endpoints may register simultaneously. Each subscriber node functions as a call processing engine capable of handling device registration, call routing decisions, signaling exchanges, and feature control. By distributing this workload across multiple subscribers, CUCM ensures that device capacity scales effectively while maintaining redundancy in case of node failures. This approach supports high availability by allowing endpoints to fail over to alternate subscribers if their primary call processing node becomes unavailable. In addition, it keeps call processing traffic insulated from administrative tasks, since the publisher focuses on configuration and replication duties while subscribers handle the operational signaling load. The division of labor supports predictable and resilient behavior during peak usage times, maintenance windows, or failover events. Maintaining the limit at eight subscribers prevents scenarios where the cluster’s signaling or replication subsystems might become saturated by excessive node counts, which could compromise call reliability or introduce delays in configuration propagation.
The publisher node maintains a crucial role because it hosts the authoritative database from which all configuration information originates. Administrative changes—whether related to dial plans, device pools, calling search spaces, user configurations, or security profiles—always occur on the publisher, which then initiates replication to all subscriber nodes. The subscriber nodes never modify the database; instead, they rely completely on the replicated data that ensures consistent call processing logic across the cluster. If additional subscriber nodes were permitted, the system would be forced to manage increased replication topology complexity, higher replication queue loads, and expanded synchronization intervals. This could lead to situations where some nodes become unsynchronized or only partially updated, undermining the reliability of critical services such as call routing, emergency calling mechanisms, mobility features, and inter-node failover. By enforcing an eight-subscriber limit, CUCM preserves database uniformity and establishes a predictable, manageable replication environment that maintains accuracy even in large deployments.
When organizations exceed the capacity limits of a single nine-node cluster, Cisco recommends deploying multiple CUCM clusters rather than attempting to enlarge a single one beyond supported boundaries. These multi-cluster environments are commonly connected using technologies such as Global Dial Plan Replication (GDPR), the Intercluster Lookup Service (ILS), or the Service Advertisement Framework (SAF). These mechanisms allow otherwise independent clusters to share directory information, route patterns, user data, and dial plans. By architecting networks using multiple interconnected clusters instead of a single oversized one, enterprises maintain operational resilience, logical segmentation, and consistent performance without placing undue stress on replication processes. The eight-subscriber limitation has remained an intentional and stable design principle across numerous CUCM releases, reflecting a mature balance between scalability, efficiency, and reliability within distributed call processing environments.
Question 6:
Which codec provides the best voice quality while using approximately 64 kbps of bandwidth?
A)711
B)729
C)722
D) Opus
Answer: A
Explanation:
The G.711 codec delivers the highest voice quality among standard telephony codecs by using pulse code modulation to sample audio at 8 kHz with 8-bit resolution, resulting in a 64 kbps bit rate. This codec provides toll-quality voice that is essentially indistinguishable from traditional PSTN circuits because it uses minimal compression, preserving the full frequency range of narrowband voice communications. G.711 is the mandatory codec in SIP and H.323 standards, ensuring universal interoperability across all VoIP systems. The codec exists in two variants, mu-law used primarily in North America and Japan, and A-law used in Europe and most other regions, both providing equivalent quality with slightly different companding algorithms. Because G.711 applies minimal processing to the audio signal, it introduces negligible delay and requires minimal computational resources, making it ideal for implementations where bandwidth is sufficient and voice quality is paramount. The codec’s simplicity also means it can be implemented in hardware with very low cost and power consumption. G.729 uses only 8 kbps but achieves this through aggressive compression that reduces voice quality compared to G.711. G.722 provides wideband quality with better frequency response than G.711 but uses similar bandwidth. Opus is a modern codec with variable bit rates but is not as universally supported in legacy systems as G.711.
Question 7:
An engineer is troubleshooting one-way audio issues on SIP calls. Which tool in Cisco Unified Communications Manager can capture and analyze RTP streams?
A) Real-Time Monitoring Tool
B) Dialed Number Analyzer
C) SIP Station Message Display
D) CDR Analysis and Reporting
Answer: A
Explanation:
The Real-Time Monitoring Tool in Cisco Unified Communications Manager provides comprehensive capabilities for capturing and analyzing RTP media streams during active calls, making it the primary diagnostic tool for troubleshooting audio quality and one-way audio issues. RTMT can monitor specific devices, hunt groups, or gateway channels in real time, displaying detailed statistics about packet loss, jitter, latency, and codec usage for both audio and video streams. When investigating one-way audio problems, RTMT allows administrators to verify that RTP packets are flowing in both directions, identify which direction is experiencing issues, and determine whether the problem is related to firewall rules, NAT configurations, or routing issues. The tool provides voice quality metrics including Mean Opinion Score calculations based on actual network performance, helping administrators quantify the user experience. RTMT can also display the IP addresses and port numbers used for RTP streams, which is crucial for troubleshooting firewall and routing problems that often cause one-way audio. The Dialed Number Analyzer is used for call routing analysis and does not handle media stream inspection. SIP Station Message Display shows signaling messages but does not capture or analyze the actual RTP media streams. CDR Analysis and Reporting provides call detail records for historical analysis but cannot monitor active RTP streams in real time or diagnose ongoing audio quality issues.
Question 8:
Which CUCM service parameter must be configured to enable Mobile and Remote Access for Cisco Jabber clients?
A) Cluster ID
B) SIP Station Keepalive
C) Certificate Trust List
D) SRST Reference
Answer: A
Explanation:
The Cluster ID service parameter is a critical prerequisite for enabling Mobile and Remote Access functionality in Cisco Unified Communications Manager because it uniquely identifies the CUCM cluster to the Cisco Expressway infrastructure that provides secure external connectivity. When Jabber clients connect from outside the corporate network, they authenticate through Expressway-E which then proxies the connection to Expressway-C and ultimately to CUCM. The Cluster ID serves as a routing identifier that allows Expressway to determine which CUCM cluster should handle requests from external clients, which is essential in environments where multiple CUCM clusters might be served by the same Expressway infrastructure. This parameter must be configured identically on all nodes in the CUCM cluster and must match the configuration on the corresponding Expressway-C traversal zone. The Cluster ID is included in various SIP headers and messages, enabling Expressway to correctly route registration requests and call signaling to the appropriate cluster. Without a properly configured Cluster ID, external Jabber clients cannot register because Expressway cannot determine the destination CUCM cluster. SIP Station Keepalive controls how frequently endpoints send keepalive messages but is not specific to MRA configuration. Certificate Trust List manages trusted certificates but does not enable the MRA feature itself. SRST Reference configures survivability for branch offices but has no relationship to MRA functionality for mobile clients connecting through Expressway.
Question 9:
What is the default SIP port used for secure communications between Cisco Unified Communications Manager and SIP endpoints?
A) 5060
B) 5061
C) 8443
D) 2443
Answer: B
Explanation:
Port 5061 is the standard TCP port designated for secure SIP communications using Transport Layer Security, and it is the default port that Cisco Unified Communications Manager uses when establishing encrypted signaling connections with SIP endpoints. This port is defined in RFC 3261 and subsequent SIP-related standards as the well-known port for SIP over TLS, ensuring consistent implementation across different vendors and platforms. When CUCM is configured to use encrypted SIP signaling, all SIP phones, trunks, and applications establish TLS connections on port 5061 to protect call signaling information from eavesdropping and tampering. The use of TLS provides authentication of both endpoints, confidentiality through encryption, and message integrity verification, which are essential security requirements for enterprise communications. Endpoints present certificates to CUCM during the TLS handshake, and CUCM validates these certificates against its Certificate Trust List before allowing registration. This mutual authentication prevents rogue devices from registering with the system. Port 5060 is the standard port for unencrypted SIP communications and is used when security is not required, but most modern deployments mandate encryption. Port 8443 is commonly used for HTTPS administration interfaces on various Cisco applications but is not used for SIP signaling. Port 2443 is used for some Cisco proprietary secure communications but is not the standard SIP TLS port. Proper firewall configuration must allow bidirectional communication on port 5061 between endpoints and CUCM for secure SIP registration and call control.
Question 10:
Which feature in Cisco Unity Connection allows users to access their voicemail through a web browser?
A) ViewMail for Outlook
B) Cisco Unity Connection Web Inbox
C) Personal Call Transfer Rules
D) Cisco Unified Messaging Gateway
Answer: B
Explanation:
Cisco Unity Connection Web Inbox provides users with browser-based access to their voicemail messages, allowing them to manage messages, listen to voicemails, and configure voicemail settings without requiring desktop applications or phone access. This web interface is accessible through HTTPS from any device with a modern browser, providing flexibility for users who are traveling or working remotely. The Web Inbox displays a list of all messages with details including caller information, timestamp, duration, and message priority, allowing users to quickly identify important messages. Users can play messages directly through the browser using embedded media players, delete unwanted messages, forward voicemails to other users, and save important messages for future reference. The interface also provides access to personal settings including greetings, message notification rules, and PIN management. Web Inbox supports message organization through folders and search functionality, helping users manage large numbers of voicemails efficiently. The browser-based architecture means no client software installation is required, reducing IT support overhead and ensuring compatibility across different operating systems. ViewMail for Outlook integrates voicemail into the Outlook client but is a separate add-in rather than a web-based interface. Personal Call Transfer Rules configure call forwarding behavior but do not provide message access. Cisco Unified Messaging Gateway handles protocol conversion between different messaging systems but does not provide user interfaces for message management.
Question 11:
An administrator needs to configure quality of service for voice traffic. Which DSCP value is recommended for voice bearer traffic?
A) CS3
B) AF41
C) EF
D) CS5
Answer: C
Explanation:
Expedited Forwarding is the recommended Differentiated Services Code Point value for voice bearer traffic because it provides the lowest latency and highest priority treatment required for real-time voice communications. EF corresponds to a DSCP decimal value of 46 or binary value of 101110 and is specifically designed for traffic that requires minimal delay, jitter, and packet loss. Network devices that implement quality of service examine the DSCP field in IP packet headers and place EF-marked packets in priority queues, ensuring voice packets are transmitted before lower-priority traffic. This prioritization is critical for maintaining voice quality because human speech is highly sensitive to delay and packet loss. Industry standards and Cisco best practices consistently recommend EF for voice RTP streams to ensure consistent treatment across multi-vendor networks. The RFC 4594 configuration guidelines explicitly designate EF as the appropriate marking for telephony services requiring strict latency and jitter requirements. Voice signaling traffic typically uses CS3 to differentiate it from the higher-priority voice bearer streams while still ensuring reliable delivery. AF41 is used for video conferencing traffic which has different latency and bandwidth requirements than voice. CS5 is typically reserved for broadcast video or high-priority data applications rather than voice traffic. Proper end-to-end DSCP marking from endpoints through the entire network path is essential for QoS effectiveness, and CUCM can be configured to mark voice traffic appropriately at the source.
Question 12:
Which protocol does Cisco Unified Border Element use to interact with Cisco Unified Communications Manager for call admission control?
A) RSVP
B)323
C) SIP
D) SCCP
Answer: C
Explanation:
Cisco Unified Border Element uses the Session Initiation Protocol when interfacing with Cisco Unified Communications Manager for call routing and call admission control functions in modern collaboration architectures. SIP has become the standard signaling protocol for enterprise voice communications, replacing older protocols like H.323 and MGCP in contemporary deployments. When CUBE receives incoming calls from service providers or remote sites, it communicates with CUCM using SIP trunk configurations to request call admission, exchange media capabilities, and coordinate call setup. The SIP signaling carries all necessary information about the calling and called parties, requested media types, and codec preferences. CUCM analyzes this information against configured dial plans, route patterns, and call admission control policies before accepting or rejecting the call. CUBE can also implement local call admission control policies based on bandwidth availability, maximum call limits, or other criteria before forwarding calls to CUCM. The SIP interface between CUBE and CUCM supports features like early offer and delayed offer for media negotiation, REFER for call transfers, and various SIP headers for passing custom call information. RSVP is a resource reservation protocol that can be used for call admission control in some scenarios but is not the signaling protocol between CUBE and CUCM. H.323 is a legacy protocol that has been largely superseded by SIP in modern networks. SCCP is used between Cisco IP phones and CUCM but is not suitable for gateway or border element communications.
Question 13:
What is the purpose of the Device Mobility feature in Cisco Unified Communications Manager?
A) Enable Extension Mobility for users
B) Allow devices to roam between locations and automatically update network settings
C) Provide device redundancy across clusters
D) Support SRST fallback for remote sites
Answer: B
Explanation:
Device Mobility is a sophisticated feature in Cisco Unified Communications Manager that enables phones and other devices to automatically detect their physical location based on IP subnet information and dynamically adjust their configuration parameters to match the local site settings. When a device roams from one location to another, Device Mobility compares the device’s current IP address against configured subnets to determine which physical location the device is operating in. Based on this determination, CUCM can automatically update settings such as the local date and time format, the emergency services calling number, the SRST router for failover, the local media resources for conferencing and transcoding, and the applicable region for codec selection and bandwidth management. This automation is particularly valuable for mobile workers who travel between offices with laptops running Jabber or for devices that are physically relocated. Without Device Mobility, devices would retain their home location settings even when operating at a different site, potentially causing issues with emergency dialing, incorrect local gateway selection, or inefficient media routing. The feature requires careful planning to define device pools, regions, locations, and physical location entities that map to IP subnets. Extension Mobility is a separate feature that allows users to log into different physical phones and is not related to automatic location detection. Device redundancy is handled through subscriber servers in a cluster rather than Device Mobility. SRST configuration can be influenced by Device Mobility but is a separate survivability feature.
Question 14:
Which media resource in CUCM is responsible for converting media streams between different codecs?
A) Conference Bridge
B) Media Termination Point
C) Transcoder
D) Music on Hold Server
Answer: C
Explanation:
The Transcoder media resource in Cisco Unified Communications Manager performs real-time conversion of audio and video streams between different codec formats, enabling interoperability between endpoints that do not share common codec capabilities. When two devices in a call support different codecs and cannot negotiate a common format, CUCM automatically inserts a transcoder into the media path to translate between the codecs. For example, if a G.711 endpoint needs to communicate with a G.729 endpoint due to bandwidth constraints on one leg of the call, the transcoder receives the G.711 stream, decodes it to raw audio, re-encodes it to G.729, and forwards it to the destination. This process requires significant computational resources because it involves complete decode and encode operations rather than simple stream manipulation. Transcoders can be implemented as software resources on CUCM servers, as dedicated hardware DSP resources in Cisco ISR routers, or as dedicated transcoding appliances. The availability and allocation of transcoder resources can significantly impact call capacity and quality in networks where codec mismatches are common. CUCM’s region configuration and codec preference lists work together to minimize transcoding requirements by negotiating common codecs when possible. Conference Bridge resources mix multiple audio streams for conference calls but do not perform codec conversion. Media Termination Point resources anchor media streams for certain call control scenarios but do not change codec formats. Music on Hold Server provides audio for callers on hold but is not involved in codec conversion between endpoints.
Question 15:
What authentication method does Cisco Jabber use when integrating with Cisco Unified Communications Manager?
A) LDAP bind
B) OAuth 2.0
C) Digest authentication
D) Certificate-based authentication
Answer: C
Explanation:
Digest authentication is the primary authentication mechanism used by Cisco Jabber when registering with Cisco Unified Communications Manager for voice and video services. This challenge-response authentication method allows CUCM to verify the user’s credentials without transmitting passwords in clear text across the network. When Jabber attempts to register, CUCM responds with a challenge containing a nonce value and other parameters. Jabber then creates a hash of the username, password, and challenge parameters using the MD5 algorithm and sends this digest response back to CUCM. CUCM performs the same hash calculation using the stored password and compares the results to authenticate the user. This method provides reasonable security for credentials while maintaining compatibility with SIP and other protocols that support digest authentication schemes. The authentication credentials used are typically the end user’s CUCM username and password, which can be synchronized from corporate directory services. When combined with TLS encryption for the signaling channel, digest authentication provides adequate security for most enterprise deployments. LDAP bind authentication is used when Jabber queries directory services for contact information but not for CUCM registration. OAuth 2.0 is used in some cloud-based collaboration scenarios but is not the standard authentication method between Jabber and on-premises CUCM. Certificate-based authentication can be used in some deployment scenarios for device authentication but digest authentication remains the standard for user credential verification during Jabber registration with CUCM for call control purposes.