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Question 166:
Which QoS model provides the highest level of service guarantee for voice and video traffic?
A) Best Effort
B) Differentiated Services
C) Integrated Services
D) Class of Service
Answer: C
Explanation:
Integrated Services (IntServ) is a network architecture designed to provide the highest level of quality-of-service guarantees by implementing per-flow resource reservations across an IP network. IntServ relies on the Resource Reservation Protocol (RSVP) to signal and reserve resources for individual communication flows, ensuring that each session receives the bandwidth, delay bounds, and packet loss characteristics required for high-quality performance. Before a session can begin, RSVP messages are exchanged between the endpoints and every router along the path to reserve the necessary resources. If any router along the path cannot allocate the requested bandwidth or meet the session’s requirements, the reservation is rejected, preventing the session from proceeding under substandard conditions. This admission control mechanism ensures that all admitted sessions receive their promised service quality without interference from other traffic.
Each RSVP reservation maintains a “soft state” in routers, which requires periodic refresh messages to keep the reservation active. This per-flow state management allows very precise control of network resources for each session, which is critical for applications such as real-time voice, video conferencing, or other delay-sensitive services. However, the necessity to track individual flows on every router introduces significant scalability challenges. Large networks with thousands of concurrent flows can quickly exhaust router memory and processing capabilities, making IntServ less practical for enterprise-wide deployment. Consequently, it is typically deployed in environments with predictable traffic patterns or limited scale, such as dedicated video conference links or specific collaboration circuits where end-to-end guarantees are critical.
By contrast, the Best Effort model offers no quality guarantees and treats all traffic equally, meaning voice or video traffic may compete with bulk data traffic and experience delays, jitter, or packet loss during congestion. Differentiated Services (DiffServ) provides an alternative approach by classifying and prioritizing traffic into a limited number of classes rather than reserving resources per flow. While DiffServ does not provide strict per-flow guarantees like IntServ, it is far more scalable and can deliver acceptable voice and video quality when combined with proper queuing, scheduling, and bandwidth provisioning. Class of Service (CoS) operates at Layer 2, prioritizing traffic within local segments such as VLANs but does not provide end-to-end guarantees across the network.
IntServ’s precise resource control makes it ideal for scenarios where guaranteed quality is essential, but the operational complexity and scalability limitations have led modern enterprise networks to favor DiffServ for most large-scale deployments. When implemented effectively, DiffServ can support high-quality real-time communication by marking voice and video traffic with higher priority, ensuring timely delivery even under congestion, and still allowing flexible use of network resources for other applications. Enterprises may also combine elements of IntServ and DiffServ selectively, using IntServ-style reservations on critical links or for specialized applications while leveraging DiffServ across the broader network to balance performance with scalability.
Ultimately, IntServ demonstrates the principle of strict per-flow service guarantees through RSVP-based reservations, providing predictable performance for delay-sensitive traffic. While not widely adopted across large-scale enterprise networks due to complexity, it remains a valuable tool for scenarios where the absolute assurance of network resources is required. Understanding its role relative to Best Effort, DiffServ, and CoS allows network designers to choose the appropriate QoS strategy for each application, ensuring that voice, video, and other real-time services operate reliably and meet user expectations.
Question 167:
What is the recommended DSCP value for marking voice bearer traffic in a Cisco collaboration deployment?
A) AF41
B) CS3
C) EF
D) CS5
Answer: C
Explanation:
Expedited Forwarding (EF), marked with a DSCP value of 46, is the industry-standard recommendation for marking voice bearer traffic in Cisco collaboration deployments. This marking ensures that real-time voice media receives the highest priority across the network, helping to minimize latency, jitter, and packet loss, all of which are critical for maintaining natural and reliable voice communications. By assigning EF to voice traffic, network devices place these packets into a strict priority queue, meaning they are transmitted ahead of other traffic types whenever the queue contains packets. This behavior is essential because even short delays or variations in packet delivery can noticeably degrade the quality of a voice call.
Voice bearer traffic has stringent performance requirements, including one-way delay under 150 milliseconds, jitter under 30 milliseconds, and packet loss below 1 percent. The combination of EF marking and priority queuing helps meet these metrics by reducing queuing delays and preventing drops, particularly during congestion events. For this reason, consistent end-to-end implementation of EF is critical; any network device along the path that fails to recognize or prioritize EF-marked packets can introduce latency or jitter, undermining the quality of the call. Proper configuration extends across access switches, distribution and core switches, WAN routers, and even service provider links to maintain seamless prioritization.
Different traffic types within the Cisco collaboration environment receive alternative DSCP markings based on their performance requirements. For example, video conferencing bearer traffic is typically marked as AF41 (DSCP 34), providing high priority but slightly lower than voice. Call signaling traffic, such as SIP or SCCP messages, is marked as CS3 (DSCP 24), which ensures reliable delivery without needing the extreme prioritization of voice bearer traffic. Some video signaling or broadcast video streams may use CS5 (DSCP 40), which is suitable for high-priority video signaling but not appropriate for interactive voice media. This hierarchical approach to marking ensures that critical real-time traffic receives the priority it requires while preventing unnecessary preemption of less time-sensitive traffic.
Network administrators must also implement policing mechanisms to limit the amount of EF-marked traffic entering the priority queue. Without such controls, excessive EF traffic could monopolize the strict priority queue, starving other traffic classes and potentially causing network instability. Proper traffic shaping, queuing policies, and monitoring are therefore necessary to maintain balanced and predictable network behavior. When configured correctly, EF marking for voice traffic, combined with complementary DSCP markings for video and signaling, ensures that Cisco collaboration environments deliver high-quality, consistent real-time communication across both LAN and WAN infrastructure.
Question 168:
Which component provides centralized dial plan management and call routing in a multi-cluster Cisco Unified Communications Manager deployment?
A) Cisco Unified Communications Manager IM and Presence
B) Cisco Unified Communications Manager Session Management Edition
C) Cisco Unity Connection
D) Cisco Expressway
Answer: B
Explanation:
Cisco Unified Communications Manager Session Management Edition (SME) is designed to simplify and centralize call routing in large, multi-cluster Cisco Unified Communications deployments. Its primary function is to act as a centralized routing hub, interconnecting multiple CUCM clusters and abstracting the complexities of managing inter-cluster dial plans.
In traditional multi-cluster deployments, administrators must configure route patterns, route lists, and route groups on every cluster to reach every other cluster. As the number of clusters grows or directory numbers change, maintaining this full-mesh configuration becomes increasingly complex and error-prone. SME addresses this challenge by maintaining a global dial plan and routing intelligence, allowing clusters to register to SME via SIP trunks rather than maintaining full-mesh connectivity. Each leaf cluster advertises its locally controlled directory numbers to SME, which then makes centralized call routing decisions. When a call originates in one cluster destined for another, SME determines the appropriate destination and forwards the call accordingly.
Beyond basic inter-cluster routing, SME supports advanced features like time-of-day routing, least-cost routing based on multiple criteria, and geographic redundancy for high availability. Administrators configure routing policies centrally through SME, reducing operational complexity and ensuring consistent routing behavior across the enterprise.
Other Cisco components serve complementary but distinct functions:
IM and Presence provides instant messaging and presence services, not call routing.
Unity Connection handles voicemail integration.
Expressway facilitates Mobile and Remote Access (MRA) and business-to-business SIP federation but does not manage inter-cluster call routing.
In essence, SME is the recommended solution for enterprises that require scalable, manageable inter-cluster call routing while minimizing manual configuration and reducing the risk of routing inconsistencies.
Question 169:
What is the maximum recommended one-way latency for voice traffic in a Cisco collaboration network?
A) 50 milliseconds
B) 100 milliseconds
C) 150 milliseconds
D) 200 milliseconds
Answer: C
Explanation:
The maximum recommended one-way latency for voice traffic is 150 milliseconds, as outlined by ITU-T G.114 recommendations and supported by Cisco best practices. This threshold represents the point at which most users begin to notice a degradation in voice quality, with the conversation starting to feel less natural due to the presence of noticeable delay. One-way latency, in this context, refers to the time it takes for a voice packet to travel from the speaker to the listener. This delay is cumulative, arising from a combination of factors that include serialization delay on network links, propagation delay determined by the physical distance between endpoints, queuing delay at network devices, and processing delay within endpoints and infrastructure equipment. Each of these factors contributes incrementally to the overall latency experienced in a voice communication session, and even small increases in any one area can push the total delay closer to or beyond the 150-millisecond threshold, negatively impacting the perceived quality of the conversation. Voice communications remain interactive and natural when the one-way delay stays below 150 milliseconds, allowing users to speak and respond without noticeable pauses or interruptions. Delays under this threshold are generally imperceptible, enabling normal conversational flow and minimizing the risk of participants speaking over each other. However, as one-way latency increases beyond this point, conversations become increasingly difficult to manage. Users may encounter situations where they unintentionally interrupt one another due to overlapping speech, or they may experience awkward silence as each participant waits for the other to respond, creating a frustrating and unnatural communication experience. The 150-millisecond recommendation provides a practical margin that accounts for normal variations in network conditions while still maintaining voice quality. Network designers and engineers are encouraged to target even lower latencies, typically under 100 milliseconds, to ensure that voice quality remains acceptable during transient congestion events or when additional delays are introduced by processes such as codec encoding and decoding, packet loss concealment, or jitter buffering. Effective latency management requires careful planning and monitoring of all potential sources of delay across the entire communication path. WAN links, in particular, can introduce significant latency depending on the physical distance and the available bandwidth, with serialization delay becoming more pronounced on slower links where each packet takes longer to transmit. Queuing delay also becomes a critical factor during periods of network congestion, when packets must wait in buffers before being forwarded. Similarly, processing delays at routers, switches, gateways, and endpoints, including IP phones and soft clients, contribute to the overall latency and must be accounted for in network design. Proper network design, including the allocation of sufficient bandwidth, implementation of quality of service (QoS) mechanisms, and optimization of routing paths, is essential to keeping latency within acceptable bounds for voice communications. QoS, for example, allows voice traffic to be prioritized over less time-sensitive data, minimizing the likelihood that packets will be delayed or dropped. Additionally, techniques such as traffic shaping, link aggregation, and careful topology planning can further reduce the impact of network congestion and propagation delays. Overall, maintaining one-way latency below 150 milliseconds is critical for ensuring that voice communications remain clear, responsive, and natural, while targeting lower latencies under 100 milliseconds provides an additional safety margin that helps preserve voice quality even under less-than-ideal network conditions. Through careful planning, monitoring, and optimization, network operators can create an environment where real-time voice communications are consistently reliable and user-friendly, supporting effective collaboration and minimizing frustration caused by delayed responses or conversational disruption
Question 170:
Which protocol does Cisco Unified Communications Manager use to download configuration files to IP phones?
A) HTTP
B) TFTP
C) FTP
D) HTTPS
Answer: B
Explanation:
Cisco Unified Communications Manager uses TFTP (Trivial File Transfer Protocol) to deliver configuration files to IP phones during the boot process. When a phone starts, it obtains the TFTP server address through DHCP option 150 or 66, then contacts the TFTP service to download its specific configuration file.
The TFTP service runs on Unified Communications Manager servers and maintains configuration files for each registered device. These files contain parameters including server addresses, feature settings, softkey templates, line appearances, speed dials, and other device-specific configurations. The phone parses this configuration and adjusts its behavior accordingly.
TFTP uses UDP port 69 and provides simple, lightweight file transfer without authentication or encryption. While this simplicity benefits phone boot performance, it does not provide security for configuration data. For enhanced security, administrators can configure encrypted TFTP where configuration files are encrypted before transfer, protecting sensitive information from network eavesdropping.
The phone registration and boot process follows a specific sequence: obtain IP address via DHCP, receive TFTP server address, download configuration file via TFTP, register to Unified Communications Manager using SCCP or SIP, and download firmware if version mismatch exists. The TFTP service is critical because phones cannot complete initialization without their configuration files.
HTTP and HTTPS are used for various web-based communications with the phone but not primary configuration delivery. FTP is not used in the standard phone boot process. Modern deployments may use HTTP for some file transfers but TFTP remains the standard protocol for delivering initial configuration files to Cisco IP phones during boot and registration processes.
Question 171:
What is the primary purpose of a Device Mobility feature in Cisco Unified Communications Manager?
A) Enable phones to roam between wireless access points
B) Automatically adjust device settings based on location
C) Provide mobile application connectivity
D) Support remote worker VPN connections
Answer: B
Explanation:
Device Mobility in Cisco Unified Communications Manager automatically adjusts device settings based on the device’s current network location. This feature enables phones to move between sites while maintaining appropriate configuration for their current location including region settings, location-based call admission control, and PSTN access codes.
When a phone registers, Device Mobility compares the phone’s IP subnet against configured physical location definitions. If the phone is at its home location, it uses the device pool configuration assigned to the device. If the phone is roaming at a different location, Device Mobility dynamically overrides certain settings including region, location, and network locale to match the roaming location.
This automatic adjustment ensures proper call routing and feature behavior regardless of where the phone physically connects. For example, an executive’s phone normally at headquarters might temporarily connect at a branch office. Device Mobility ensures the phone uses the branch office’s PSTN gateway for external calls rather than routing through headquarters, providing proper local emergency services access.
The feature also maintains appropriate call admission control by placing the phone in the correct location bandwidth pool. This prevents the phone from consuming WAN bandwidth as if it were still at its home location. Device Mobility updates the location association dynamically based on subnet detection without requiring administrator intervention.
Wireless roaming between access points is handled by the wireless infrastructure, not Device Mobility. Mobile applications connect through Expressway Mobile and Remote Access. VPN connectivity is provided by network security infrastructure. Device Mobility specifically addresses the automatic configuration adjustment challenge when physical IP phones move between enterprise locations.
Question 172:
Which Cisco Unity Connection feature allows callers to skip the greeting and go directly to leaving a message?
A) Caller Input
B) Skip Greeting
C) Rapid Input
D) Direct Messaging
Answer: A
Explanation:
Caller Input in Cisco Unity Connection allows callers to press keys during greeting playback to perform actions including skipping the greeting and immediately recording a message. This feature improves caller efficiency by enabling them to bypass greetings they have heard previously or when they are in a hurry.
The most common caller input configuration is pressing pound (#) to skip the greeting and proceed directly to the beep for message recording. Other keys can be configured for different actions such as transferring to an operator, accessing a directory, or selecting alternate extensions. These caller input options function like a simple interactive voice response during greeting playback.
Unity Connection administrators configure caller input options per user through the subscriber configuration settings. The feature can be enabled or disabled based on organizational preferences and user requirements. When enabled, the system monitors for DTMF tones during greeting playback and takes the configured action when recognized keys are pressed.
Callers appreciate this feature especially when calling the same individuals repeatedly. Business users who frequently exchange voicemail messages can skip greetings and leave messages quickly without waiting through complete greeting playback. The time savings become significant for users who leave numerous voicemail messages throughout their workday.
Skip Greeting, Rapid Input, and Direct Messaging are not actual Cisco Unity Connection feature names but describe the behavior enabled by caller input. The caller input feature provides the mechanism for implementing greeting skip and other mid-greeting navigation options that enhance the voicemail user experience for both message senders and recipients in collaboration deployments.
Question 173:
What is the function of the Cisco CTI Manager service in Unified Communications Manager?
A) Provides database replication between cluster nodes
B) Enables third-party application integration and call control
C) Manages TFTP file distribution to devices
D) Handles user authentication and authorization
Answer: B
Explanation:
Cisco CTI Manager service provides computer telephony integration capabilities that enable third-party applications to monitor and control calls in Unified Communications Manager. This service acts as an intermediary between external applications and the CallManager service, translating application requests into call control commands.
CTI Manager uses the JTAPI (Java Telephony API) or TAPI (Telephony API) interfaces to communicate with applications. These APIs allow applications to perform functions including making calls, answering calls, transferring calls, holding calls, and monitoring line and device status. Applications can also receive real-time events about call state changes, enabling features like screen pop in contact centers.
Common applications using CTI Manager include contact center platforms, unified communications clients like Jabber, attendance consoles, and business intelligence tools that track call metrics. For example, Jabber uses CTI Manager to enable click-to-dial, call control from the application interface, and visual voicemail integration.
The CTI Manager service supports both first-party call control where the application controls a device it is associated with, and third-party call control where the application controls devices on behalf of users. Multiple redundant CTI Manager services can run in a cluster providing high availability for CTI applications.
Database replication is handled by the Database Layer Monitor service. TFTP services are provided by the separate Cisco TFTP service. User authentication integrates with LDAP directories through the Cisco DirSync service. CTI Manager specifically enables the programmable call control capabilities that third-party applications require for integration with Unified Communications Manager.
Question 174:
Which feature allows multiple Cisco Unified Communications Manager clusters to share a common dial plan?
A) Call Routing
B) Global Dial Plan Replication
C) Intercluster Lookup Service
D) Unified Dial Plan
Answer: C
Explanation:
Intercluster Lookup Service (ILS) allows multiple Cisco Unified Communications Manager clusters to share dial plan information and enable transparent inter-cluster calling without requiring extensive manual dial plan configuration on each cluster. ILS creates a federated dial plan across the enterprise by automatically exchanging directory number information.
Each participating cluster runs the ILS service and joins an ILS network by configuring mesh relationships with other clusters. Clusters advertise their locally controlled directory numbers to the ILS network, and receive advertisements from other clusters. This automatic exchange eliminates the need to manually configure route patterns for every remote cluster.
When a user dials a number not locally registered, Unified Communications Manager queries ILS to locate the owning cluster. If found, the call routes through the inter-cluster trunk to the remote cluster. From the user perspective, calling between clusters becomes seamless without requiring knowledge of special access codes or routing prefixes.
ILS also supports advanced features including learned pattern propagation where clusters share not only their local directory numbers but also their external route patterns. This enables features like extension mobility across clusters where users can log into phones at any site and maintain their directory number and calling privileges.
Call Routing is a general function not a specific inter-cluster service. Global Dial Plan Replication is not a Cisco feature name. Unified Dial Plan is a concept but ILS is the specific implementation providing automated dial plan federation. ILS significantly reduces administrative overhead in multi-cluster deployments while improving user experience through simplified inter-site calling.
Question 175:
What is the recommended maximum Round-Trip Time (RTT) for voice traffic?
A) 100 milliseconds
B) 200 milliseconds
C) 300 milliseconds
D) 400 milliseconds
Answer: C
Explanation:
The recommended maximum Round-Trip Time for voice traffic is 300 milliseconds, which corresponds to the 150 millisecond one-way delay recommendation. RTT measures the time for a packet to travel from source to destination and back, effectively doubling the one-way latency measurement.
RTT is commonly measured using ping commands or network monitoring tools and provides a practical measurement for assessing network suitability for voice communications. Since voice is a two-way interactive conversation, understanding the round-trip characteristics helps predict whether both parties will experience acceptable quality during their conversation.
When RTT approaches or exceeds 300 milliseconds, conversations become noticeably impacted by delay. Users may begin talking over each other because of the lag between speaking and hearing responses. The conversation loses its natural flow requiring conscious adjustment to communication patterns including longer pauses and explicit turn-taking cues.
Network designs must account for RTT when planning voice deployments across geographically distributed sites. Satellite links inherently introduce high RTT due to propagation delay to geostationary orbit and back. Long-distance international WAN links may approach the RTT threshold requiring careful network design and optimization to maintain acceptable voice quality.
Measuring RTT during network assessment helps identify potential voice quality issues before deployment. Sustained RTT above 300 milliseconds indicates the path may not provide adequate quality for real-time voice. Additional delay from queuing, jitter buffers, and codec processing compounds the base network RTT, making it important to maintain significant margin below the 300 millisecond maximum for reliable voice communications.
Question 176:
Which Cisco Unity Connection feature provides automated attendant functionality?
A) Call Handler
B) Interview Handler
C) Directory Handler
D) System Call Handler
Answer: A
Explanation:
Call Handlers in Cisco Unity Connection provide automated attendant functionality by answering calls, playing greetings, and routing callers based on their input. Call handlers are the fundamental building blocks for creating auto-attendant applications that present menus and route calls without human operator intervention.
A call handler can play customized greetings that guide callers through available options, accept DTMF input from caller keypresses, and route calls to extensions, other call handlers, or user mailboxes based on the input received. Organizations typically create a main call handler that serves as the primary automated attendant greeting callers who dial the main company number.
Call handlers support multiple greetings for different scenarios including business hours, after hours, holidays, and alternate schedules. Administrators record professional greetings that maintain brand image while efficiently directing callers. The routing rules can be sophisticated, including time-based routing, transfer to live operators, access to dial-by-name directories, or escalation paths.
Complex auto-attendant applications use multiple call handlers chained together to create hierarchical menu systems. For example, a main menu might offer options for sales, support, or directory access, with each option transferring to another specialized call handler. This structure enables sophisticated call routing while maintaining manageable configuration.
Interview Handler is used for surveys and information gathering rather than automated attendant services. Directory Handler enables dial-by-name lookups but is typically accessed from a call handler. System Call Handler is a specific type of call handler used for internal purposes. Call handlers provide the flexible framework for implementing custom automated attendant solutions meeting diverse organizational requirements.
Question 177:
What protocol does Cisco Jabber use for presence and instant messaging services?
A) SIP
B) XMPP
C) HTTP
D) BFCP
Answer: B
Explanation:
Cisco Jabber uses XMPP (Extensible Messaging and Presence Protocol) for presence and instant messaging services when connecting to Cisco Unified Communications Manager IM and Presence Service. XMPP is an open standard protocol specifically designed for real-time communication including presence updates and instant messaging.
XMPP establishes persistent connections between Jabber clients and the IM and Presence servers, enabling immediate delivery of presence changes and instant messages. When a user’s status changes from available to busy or in a meeting, the XMPP protocol propagates that change to all watchers who have subscribed to that user’s presence information.
The protocol uses XML streams for message encoding, making it extensible and human-readable for troubleshooting. XMPP supports both one-to-one instant messaging and group chat functionality. The persistent nature of XMPP connections ensures that messages and presence updates arrive with minimal delay, providing a responsive user experience.
Jabber’s XMPP implementation includes security features like TLS encryption for client-to-server connections and SASL for authentication. The IM and Presence Service manages XMPP connections, maintains presence state, stores offline messages, and handles message routing between users.
SIP is used by Jabber for voice and video calling capabilities but not presence and messaging. HTTP is used for some service discovery and configuration but not the core messaging protocol. BFCP (Binary Floor Control Protocol) is used for application sharing control. XMPP specifically provides the real-time, persistent connection framework required for responsive presence and instant messaging in Cisco collaboration deployments.
Question 178:
Which Cisco Unified Communications Manager feature prevents toll fraud by restricting calling privileges?
A) Partitions and Calling Search Spaces
B) Route Filters
C) Translation Patterns
D) Route Groups
Answer: A
Explanation:
Partitions and Calling Search Spaces in Cisco Unified Communications Manager provide the primary mechanism for preventing toll fraud by implementing class of service restrictions that control which destinations users can call. This feature creates logical boundaries around directory numbers and route patterns, allowing granular control over calling privileges.
Partitions are logical groupings of directory numbers and route patterns. Calling Search Spaces are ordered lists of partitions that define what a device can reach. By assigning appropriate calling search spaces to devices based on user roles and requirements, administrators control whether users can make local calls only, domestic long distance, international calls, or premium rate numbers.
The partition and calling search space model provides flexible security by enabling administrators to create different privilege levels. For example, lobby phones might only reach internal extensions and emergency services, while executive phones might have unrestricted dialing capabilities. This segmentation prevents unauthorized users from making expensive international or premium rate calls.
Toll fraud occurs when attackers exploit voice systems to make unauthorized calls, often routing expensive international calls through the system. Proper partition and calling search space implementation prevents this by ensuring devices cannot reach international route patterns unless explicitly authorized. Combined with strong voicemail passwords and secure remote access, this reduces toll fraud risk significantly.
Route Filters match digit patterns but do not provide access control. Translation Patterns manipulate digits but do not restrict access. Route Groups organize gateway resources. Partitions and Calling Search Spaces specifically implement the class of service and access control framework essential for toll fraud prevention in Unified Communications Manager deployments.
Question 179:
What is the purpose of Device Defaults in Cisco Unified Communications Manager?
A) Define backup servers for device registration
B) Provide global configuration settings applied to new devices
C) Specify firmware versions for different phone models
D) Configure network time protocol settings
Answer: B
Explanation:
Device Defaults in Cisco Unified Communications Manager provide global configuration settings that are automatically applied to new devices when they are added to the system. This feature streamlines device provisioning by establishing consistent baseline configurations across the deployment without requiring administrators to manually configure every parameter.
Device defaults include settings like audio codecs, video capabilities, DND softkey, feature buttons, device security mode, and other common parameters. When an administrator adds a new phone through auto-registration or manual configuration, the system applies these default values, ensuring consistency while allowing customization of specific devices as needed.
Using device defaults reduces configuration errors and speeds deployment, especially in large environments adding many similar devices. Administrators define appropriate defaults based on organizational standards and security policies. For example, setting encrypted device security mode as the default ensures new phones automatically use secure communications without manual intervention.
Changes to device defaults only affect subsequently added devices; existing devices retain their current configurations unless explicitly modified. This behavior allows administrators to update defaults for future deployments while maintaining stability of the existing environment. Organizations should carefully plan device defaults to match their standard deployment requirements.
Device pools define backup server assignments rather than device defaults. Load server settings specify firmware locations but device defaults do not directly control firmware versions. NTP configuration occurs through operating system and service parameters. Device defaults specifically provide the initial configuration template applied when creating new device records in Unified Communications Manager.
Question 180:
Which Cisco Expressway feature enables external access for mobile clients without VPN?
A) Business to Business
B) Mobile and Remote Access
C) Remote Gateway
D) External Federation
Answer: B
Explanation:
Mobile and Remote Access (MRA) in Cisco Expressway enables mobile clients like Jabber to access enterprise collaboration services from external networks without requiring VPN connections. MRA provides secure, clientless remote access that maintains full collaboration functionality including calling, messaging, presence, and voicemail regardless of user location.
The MRA solution uses Expressway-C (Control) and Expressway-E (Edge) in a dual-server deployment. Expressway-E resides in the DMZ with public IP addressing, accepting connections from external clients. Expressway-C sits inside the trusted network with access to Unified Communications Manager and other collaboration services. The two Expressways establish secure tunnels through which client traffic flows.
When a Jabber client on an external network starts, it discovers Expressway-E through DNS queries and establishes a TLS connection. Expressway-E authenticates the client using credentials validated against the internal directory. After authentication, client traffic proxies through the Expressway pair to reach internal collaboration servers, maintaining end-to-end security without VPN complexity.
MRA supports full feature parity with internal users including making and receiving calls, accessing voicemail, instant messaging, presence, desktop sharing, and directory services. The transparent experience means remote users work exactly as if they were in the office, improving productivity without compromising security.
Business to Business provides federation with external organizations but not mobile client access. Remote Gateway is not a standard Expressway feature name. External Federation generally refers to inter-organization connectivity rather than mobile remote access. MRA specifically solves the mobile worker access challenge without VPN in Cisco collaboration deployments.