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Question 106:
What is the primary purpose of SIP trunking in a Cisco Unified Communications environment?
A) To provide video conferencing capabilities between endpoints
B) To connect the enterprise telephony system to a service provider’s network using IP
C) To encrypt all voice communications within the internal network
D) To manage bandwidth allocation for quality of service
Answer: B
Explanation:
SIP trunking is a key technology in modern unified communications environments, providing IP-based connectivity between an organization’s private branch exchange (PBX) or unified communications platform and external service provider networks. Unlike traditional analog or digital circuits such as T1, E1, or PRI lines, SIP trunks leverage standard internet protocol connections to transmit signaling and media, offering organizations a more flexible, cost-efficient method of connecting to the Public Switched Telephone Network (PSTN) and other external endpoints. By replacing legacy voice circuits with IP-based connectivity, SIP trunking simplifies network architecture, reduces operational overhead, and supports the integration of voice, video, and messaging services over a converged infrastructure.
The primary advantage of SIP trunking lies in its ability to consolidate voice and data traffic onto a single network infrastructure. Enterprises no longer need to maintain separate physical circuits for voice, as voice calls can traverse the same network used for data traffic. This consolidation not only reduces telecommunications costs but also streamlines network management, monitoring, and troubleshooting. SIP trunks can support multiple concurrent calls over a single connection, with capacity determined by available bandwidth rather than fixed physical lines. This flexibility allows organizations to efficiently scale their telephony resources based on demand, avoiding the over-provisioning or underutilization that often occurs with traditional circuit-based voice services.
SIP trunking also provides enhanced scalability and business continuity capabilities. Enterprises can quickly add or remove channels to accommodate changing call volumes without the need for costly and time-consuming physical infrastructure changes. This scalability makes SIP trunks ideal for dynamic environments, including businesses experiencing seasonal fluctuations, mergers, or rapid growth. Geographic flexibility is another key benefit: organizations can maintain local phone numbers in multiple regions without requiring physical presence in each area. This capability supports remote workforce strategies, multi-site deployments, and global expansion initiatives while maintaining seamless access to telephony services for employees and customers alike.
Although SIP trunks can carry various types of media, including video or messaging traffic, their primary role is to provide voice connectivity to external networks. While additional functionalities such as encryption for secure signaling and media or quality of service (QoS) mechanisms to prioritize voice traffic can be implemented, these features serve as enhancements rather than the core purpose of SIP trunking. The protocol’s main value proposition is enabling reliable, cost-effective, and scalable IP-based telephony connectivity between enterprise communication systems and service provider networks. By separating signaling and media functions, SIP also facilitates interoperability with other standards-based systems, ensuring compatibility across heterogeneous network environments.
Implementation of SIP trunking in enterprise environments requires careful planning and configuration. Administrators must define proper dial plans, configure codec negotiation to optimize voice quality, and manage security policies to protect against unauthorized access or signaling attacks. Integration with the call control platform, such as Cisco Unified Communications Manager, allows organizations to apply consistent routing, calling privileges, and feature policies across internal and external calls. The ability to centrally manage SIP trunks and dynamically adjust resources ensures that enterprises can maintain high-quality voice services while controlling costs and optimizing network utilization.
Question 107:
Which protocol does Cisco Unified Communications Manager use for device registration and call control?
A)323
B) MGCP
C) SCCP
D) SDP
Answer: C
Explanation:
Skinny Client Control Protocol (SCCP), commonly referred to as Skinny, is a Cisco-proprietary signaling protocol specifically designed for communication between Cisco Unified Communications Manager (CUCM) and Cisco IP phones or other endpoints. Unlike general-purpose signaling protocols such as SIP or H.323, SCCP is optimized for Cisco’s ecosystem, providing lightweight and efficient call control while leaving media streams, typically carried via Real-time Transport Protocol (RTP), to flow directly between endpoints. By separating signaling from media, SCCP reduces endpoint processing requirements, simplifies network bandwidth usage, and allows centralized management of call control functions. This design philosophy aligns with Cisco’s approach to enterprise telephony, where intelligence and feature logic are concentrated in the call control server rather than distributed across individual phones.
The architecture of SCCP follows a classic client-server model, with IP phones or endpoints acting as clients and CUCM serving as the central server. When a Cisco phone powers on, it establishes a TCP connection to the CUCM on port 2000 to initiate registration. During this process, the phone downloads its configuration, which includes line appearances, speed dials, feature settings, and any device-specific policies. Once registered, all signaling for call setup, teardown, and supplementary services such as call transfer, call hold, and call forwarding is handled via SCCP messages exchanged between the phone and CUCM. The protocol also supports advanced features like conferencing, hunt groups, and voicemail integration, allowing these functions to be managed centrally without requiring sophisticated intelligence on the endpoint itself.
SCCP offers several operational advantages in Cisco-centric environments. One major benefit is simplified endpoint configuration and provisioning. Since the call control logic resides on CUCM, IP phones require minimal local processing, reducing complexity and allowing for easier mass deployments. Centralized control also enables consistent application of calling features and policies across the enterprise, improving maintainability and troubleshooting. Additionally, SCCP provides tight integration with CUCM’s broader feature set, ensuring that services such as device mobility, extension mobility, and Unified Messaging work seamlessly with endpoint devices. The lightweight nature of the protocol reduces signaling overhead on the network, contributing to improved performance and reliability in large deployments.
While SCCP remains widely used in Cisco deployments, alternative signaling protocols exist with broader industry adoption. H.323 was historically used for both endpoint registration and gateway communication, but its complexity and overhead led to limited use in modern deployments. Session Initiation Protocol (SIP) is a standards-based protocol that supports multivendor interoperability, voice, video, and messaging, but endpoints must implement additional logic to handle call control features that SCCP centralizes in CUCM. MGCP is primarily used for controlling media gateways rather than endpoint registration, and Session Description Protocol (SDP) is responsible for describing media streams rather than performing call signaling or endpoint control. Understanding these distinctions helps administrators design and troubleshoot enterprise networks effectively.
Knowledge of SCCP operation is critical for troubleshooting registration issues, call setup problems, and feature functionality in Cisco Unified Communications environments. Administrators must understand how phones interact with CUCM, including the registration process, configuration download, heartbeat signaling, and feature invocation. Proper configuration of device pools, regions, and call processing parameters ensures optimal performance and reliable service delivery. In large-scale deployments, SCCP’s efficiency and integration capabilities allow for centralized management of thousands of endpoints while providing consistent call quality, feature availability, and operational reliability across the enterprise.
Question 108:
What is the maximum number of directory numbers that can be configured on a single Cisco IP Phone 8851?
A) 2
B) 5
C) 8
D) 10
Answer: D
Explanation:
The Cisco IP Phone 8851 supports configuration of up to 10 directory numbers (lines) on a single device, providing flexibility for users who need to manage multiple phone lines or extensions from one physical endpoint. This capability is particularly valuable for administrative assistants, receptionists, and managers who must monitor and answer calls for multiple individuals or departments.
Each directory number configured on the phone represents a unique extension that can receive and place calls independently. Users can have multiple lines for different purposes such as a primary business line, a direct dial number, a shared line appearance with a colleague, and additional lines for specific functions or departments. The phone’s user interface displays these lines with visual indicators showing their status including idle, ringing, connected, or held.
Line configuration occurs within Cisco Unified Communications Manager administration interface where administrators assign directory numbers to the device. Each line can have unique settings for call forwarding, voicemail integration, calling search spaces, and partitions. Shared line appearances enable multiple phones to register the same directory number, allowing call pickup and presence monitoring across devices.
Physical limitations of the phone model determine the practical usability of multiple lines. The Cisco IP Phone 8851 features five programmable line keys that can be configured as line appearances or speed dials. When more than five directory numbers are configured, users access additional lines through soft keys or by scrolling through available lines. Organizations typically balance the number of configured lines against user requirements and interface complexity to ensure optimal user experience and productivity.
Question 109:
Which Cisco Unified Communications Manager service is responsible for providing TFTP services to IP phones?
A) Cisco CallManager
B) Cisco TFTP
C) Cisco DHCP Monitor Service
D) Cisco CTIManager
Answer: B
Explanation:
The Cisco TFTP service is a critical component within Cisco Unified Communications Manager that provides configuration files and firmware images to IP phones and other network devices during their boot and registration process. This service operates on UDP port 69 and delivers essential information that endpoints require to function properly within the unified communications environment.
When an IP phone powers on, it follows a specific boot sequence to obtain its operational parameters. First, the phone receives an IP address and TFTP server address through DHCP option 150 or option 66. The phone then contacts the TFTP server to download its configuration file, which contains device-specific settings including Unified Communications Manager server addresses, network parameters, line configurations, and feature settings. Subsequently, the phone may download firmware files if the version on the device differs from what the administrator has specified.
The TFTP service maintains various file types including device-specific configuration files with XML format, firmware loads for different phone models, ring tone files, localization files for multiple languages, and phone button templates. Administrators can monitor TFTP service health through Real-Time Monitoring Tool (RTMR) and analyze TFTP logs to troubleshoot download issues or configuration problems.
Other services mentioned serve different purposes in the unified communications infrastructure. Cisco CallManager handles call processing and signaling. Cisco DHCP Monitor Service tracks DHCP server availability but does not provide configuration files. Cisco CTIManager enables computer telephony integration applications to monitor and control devices. Understanding the role of TFTP service is essential for troubleshooting phone registration failures and ensuring proper device provisioning in Cisco collaboration environments.
Question 110:
What is the purpose of a route pattern in Cisco Unified Communications Manager?
A) To configure codec preferences for voice calls
B) To define which calls are routed to specific gateways or route lists
C) To establish quality of service markings for signaling traffic
D) To determine the maximum number of concurrent calls allowed
Answer: B
Explanation:
Route patterns in Cisco Unified Communications Manager serve as the fundamental mechanism for directing outbound calls to appropriate destinations through gateways, trunks, or route lists. These patterns define digit strings that, when dialed by users, trigger specific routing decisions to connect calls to the public switched telephone network, other sites, or external services.
The route pattern configuration includes a numerical or wildcard pattern that matches dialed digits, such as 9.011! for international calls or 9.[2-9]XXXXXX for seven-digit local calls. When a user dials a number, Unified Communications Manager compares the digits against configured route patterns using a closest match algorithm. The system selects the most specific matching pattern and routes the call according to the associated gateway or route list assignment.
Route patterns work in conjunction with route groups and route lists to provide flexible, fault-tolerant call routing. A route list contains an ordered list of route groups, and each route group contains one or more gateways or trunks. This hierarchical structure enables load distribution and automatic failover when primary paths become unavailable. Administrators can implement time-of-day routing, least-cost routing, and geographic routing strategies through careful route pattern design.
Additional route pattern features include digit manipulation capabilities such as prefix addition, digit stripping, and called party transformations. These modifications ensure dialed digits are properly formatted for carrier requirements or normalize numbering between sites. While codec selection, quality of service, and call admission control represent important unified communications functions, they are configured separately from route patterns. Mastering route pattern design and implementation is essential for establishing reliable, efficient call routing in enterprise collaboration environments.
Question 111:
Which two codecs provide the best voice quality in a Cisco collaboration environment? (Choose two)
A)711
B)729
C)722
D)723
E) iLBC
Answer: A, C
Explanation:
Voice codec selection significantly impacts audio quality, bandwidth consumption, and overall user experience in unified communications deployments. Among available codecs, G.711 and G.722 deliver superior voice quality compared to compressed alternatives, though they require greater bandwidth allocation.
G.711 operates at 64 kbps and provides toll-quality voice by sampling audio at 8 kHz with minimal compression. This codec uses either A-law (primarily in Europe) or μ-law (primarily in North America) companding algorithms. G.711 introduces negligible latency and computational overhead, making it ideal for high-quality communications over networks with adequate bandwidth. The codec serves as the baseline standard for voice quality comparison and is widely supported across all IP telephony equipment.
G.722 extends beyond traditional voice quality by implementing wideband audio at 7 kHz frequency response compared to narrowband codecs limited to 3.4 kHz. Operating at 64 kbps like G.711, this codec captures higher frequency components of human speech, resulting in noticeably clearer and more natural sounding conversations. Users can distinguish subtle vocal characteristics and experience reduced listening fatigue during extended calls. G.722 has become increasingly popular as organizations prioritize user experience and network bandwidth becomes more readily available.
Compressed codecs like G.729 (8 kbps) and G.723 (5.3 or 6.3 kbps) sacrifice audio quality to conserve bandwidth. These codecs introduce compression artifacts and increased processing delay but prove valuable for bandwidth-constrained environments or high-density deployments. Understanding codec characteristics enables administrators to balance quality requirements against available network resources when designing collaboration solutions.
Question 112:
What is the default SIP port used for unencrypted signaling traffic?
A) 5060
B) 5061
C) 2000
D) 8443
Answer: A
Explanation:
Session Initiation Protocol (SIP) uses port 5060 as its default for unencrypted signaling traffic, representing a fundamental aspect of IP telephony communications. This port handles all SIP messages including call setup (INVITE), call termination (BYE), registration (REGISTER), and various other signaling functions necessary for establishing and managing voice and video sessions.
SIP operates as an application-layer protocol that initiates, modifies, and terminates multimedia sessions between participants. The protocol uses a text-based message format similar to HTTP, making it human-readable and relatively straightforward to troubleshoot. When a SIP endpoint initiates a call, it sends an INVITE message to port 5060 of the destination or proxy server. The receiving entity responds with provisional responses like 100 Trying and 180 Ringing, followed by 200 OK when the call is answered. The calling party acknowledges with an ACK message, completing the three-way handshake.
Organizations deploying SIP infrastructure must ensure firewalls and access control lists permit traffic on port 5060 for proper call signaling. Network Address Translation (NAT) environments may require additional configuration or Session Border Controllers to handle SIP messages correctly, as the protocol embeds IP addresses within message bodies that require modification during NAT traversal.
Port 5061 serves as the standard for SIP over Transport Layer Security (TLS), providing encrypted signaling for enhanced security. Port 2000 is associated with Cisco SCCP protocol rather than SIP. Port 8443 commonly serves HTTPS administration interfaces but is not related to SIP signaling. Understanding these port assignments is essential for network configuration, firewall rule creation, and troubleshooting connectivity issues in collaboration deployments.
Question 113:
Which feature allows multiple Cisco Unified Communications Manager servers to share device registration load?
A) Device Mobility
B) Extension Mobility
C) Device Pool
D) Survivable Remote Site Telephony
Answer: C
Explanation:
Device pools in Cisco Unified Communications Manager serve as logical containers that group configuration settings applied to phones and other endpoints, including the assignment of primary and backup call processing servers. This architecture enables load distribution across multiple Unified Communications Manager servers within a cluster and provides redundancy for business continuity.
Each device pool specifies a prioritized list of Unified Communications Manager servers for device registration. When an IP phone boots, it attempts to register with the highest priority server assigned in its device pool. If that server is unavailable due to maintenance or failure, the phone automatically fails over to the next server in the priority list. This mechanism distributes registered devices across available servers, preventing any single server from becoming a bottleneck and ensuring continued service during server outages.
Device pools also encapsulate other important settings including region for codec selection, location for call admission control, date and time group for proper time display, and Survivable Remote Site Telephony (SRST) reference for WAN failure scenarios. By grouping these parameters into device pools, administrators can efficiently apply consistent configurations to multiple devices based on their physical location or functional requirements.
Device Mobility allows phones to adapt their settings when users move between locations. Extension Mobility enables users to log into any phone and receive their personal settings. SRST provides local call processing during WAN failures. While these features enhance unified communications functionality, device pools specifically address the requirement for distributing registration load and providing server redundancy. Proper device pool design is fundamental to creating resilient, scalable Cisco collaboration deployments.
Question 114:
What protocol does Cisco Unified Border Element use to communicate with Cisco Unified Communications Manager for call admission control?
A)323
B) RSVP
C) SIP
D)248
Answer: C
Explanation:
Cisco Unified Border Element (CUBE) communicates with Cisco Unified Communications Manager using Session Initiation Protocol (SIP) for call signaling and control, including call admission control functions. This integration enables enterprises to extend their unified communications capabilities to external networks while maintaining centralized policy enforcement and resource management.
When CUBE connects to Unified Communications Manager, it establishes a SIP trunk relationship that allows bidirectional call routing between the internal enterprise network and external service providers or remote locations. The SIP trunk carries all call signaling messages including INVITE requests for new calls, responses indicating call progress, and termination messages when calls complete. This SIP-based integration enables CUBE to query Unified Communications Manager for available resources and enforce bandwidth limitations configured in location-based call admission control policies.
Call admission control through SIP trunks between CUBE and Unified Communications Manager prevents network congestion by limiting concurrent calls based on available bandwidth. When a new call request arrives, Unified Communications Manager checks the configured location bandwidth against current usage. If sufficient bandwidth exists, the call proceeds normally. If the location has reached its configured limit, Unified Communications Manager can reject the call, reroute it through alternative paths, or queue it based on configured policies.
H.323 represents an alternative signaling protocol but has largely been superseded by SIP in modern deployments. RSVP handles resource reservation for quality of service but is not the primary control protocol between CUBE and Unified Communications Manager. H.248 (MEGACO) controls media gateways rather than border elements. Understanding SIP-based integration between CUBE and Unified Communications Manager is essential for designing secure, scalable enterprise collaboration solutions with proper admission control.
Question 115:
Which command is used to verify SIP trunk status on Cisco Unified Border Element?
A) show sip-ua status
B) show voice call summary
C) show dial-peer voice summary
D) show call active voice brief
Answer: C
Explanation:
The command “show dial-peer voice summary” provides administrators with a comprehensive overview of configured dial peers on Cisco Unified Border Element, including their administrative and operational status. This verification tool is essential for troubleshooting connectivity issues, validating configuration changes, and monitoring trunk health in voice gateway and CUBE deployments.
Dial peers represent the fundamental building blocks of voice routing in Cisco IOS-based voice platforms. Each dial peer defines either a source or destination endpoint for calls, with inbound dial peers matching incoming calls and outbound dial peers directing calls to destinations. The summary output displays dial peer tags, types (VOIP, POTS, or other), destination patterns, session targets for VOIP dial peers, and operational status indicators showing whether each dial peer is actively available for call routing.
When troubleshooting SIP trunk issues, administrators use this command to verify that dial peers are properly configured and operational. The status field indicates whether a dial peer is up or down, helping identify configuration errors or network connectivity problems. Additional details include the number of active calls on each dial peer, allowing capacity planning and load monitoring. For SIP-specific troubleshooting, administrators can combine this command with “show sip-ua calls” or “debug ccsip messages” for detailed protocol analysis.
The “show sip-ua status” command displays SIP user agent configuration and registration status but provides less comprehensive dial peer information. “show voice call summary” presents overall call statistics rather than trunk status. “show call active voice brief” displays currently active calls without comprehensive trunk configuration details. Mastering these verification commands enables effective management and troubleshooting of Cisco voice infrastructure components.
Question 116:
What is the purpose of Media Resource Groups in Cisco Unified Communications Manager?
A) To assign IP addresses to phones dynamically
B) To group media resources and control which devices can access them
C) To configure bandwidth limitations for video calls
D) To define codec preferences for transcoding
Answer: B
Explanation:
Media Resource Groups (MRGs) in Cisco Unified Communications Manager provide a logical grouping mechanism for media resources such as transcoders, conference bridges, media termination points, and music on hold servers. This architecture enables administrators to control which devices and users can access specific media resources based on geographical location, organizational hierarchy, or functional requirements.
The MRG framework works in conjunction with Media Resource Group Lists (MRGLs) to create a hierarchical resource allocation system. An MRG contains one or more media resources of various types, while an MRGL contains an ordered list of MRGs that defines the search order when a device requires a media resource. When a phone or gateway needs a resource like a conference bridge, Unified Communications Manager searches through the assigned MRGL, checking each MRG in sequence until it finds an available resource of the appropriate type.
This architecture provides several operational benefits including resource localization to minimize WAN bandwidth consumption, load distribution across multiple resource instances, and access control to prevent unauthorized resource usage. For example, administrators might create separate MRGs for headquarters and branch offices, ensuring that branch office users first attempt to use local conference bridges before consuming WAN bandwidth to access centralized resources. Similarly, premium resources can be restricted to executive users through selective MRGL assignments.
IP address assignment occurs through DHCP services rather than MRGs. Bandwidth management for video uses call admission control locations. Codec preferences are configured in regions and device configurations. Understanding Media Resource Groups and their integration with MRGLs is fundamental to designing efficient, scalable media resource deployments that optimize user experience while managing infrastructure costs and network resource consumption.
Question 117:
Which protocol is used to synchronize directory information between Cisco Unified Communications Manager and LDAP servers?
A) LDAP
B) SCCP
C) HTTPS
D) SMTP
Answer: A
Explanation:
Lightweight Directory Access Protocol (LDAP) serves as the standard protocol for directory services integration, enabling Cisco Unified Communications Manager to synchronize user information from enterprise directory servers such as Microsoft Active Directory or OpenLDAP. This integration centralizes user management and eliminates the need to maintain duplicate user databases across multiple systems.
LDAP synchronization allows Unified Communications Manager to import user attributes including names, telephone numbers, email addresses, and department information from corporate directories. Administrators configure LDAP directory servers within the Unified Communications Manager administration interface, specifying connection parameters like server addresses, authentication credentials, base distinguished names for search operations, and synchronization schedules. The system performs periodic synchronization to keep user information current as organizational changes occur.
The integration supports both user authentication and directory information lookup. For authentication, Unified Communications Manager can validate user credentials against the LDAP directory when users log into services like Extension Mobility or self-care portals. For directory lookup, the system imports user information that appears in phone directories and corporate directory searches from endpoint displays. This capability enhances user productivity by providing access to comprehensive contact information without manual data entry.
LDAP operates on TCP port 389 for unencrypted connections or port 636 for LDAP over SSL (LDAPS) when security requirements mandate encrypted communication. Administrators should implement LDAPS in production environments to protect sensitive credential information during authentication operations. SCCP handles phone signaling, HTTPS provides secure web access, and SMTP enables email functionality, but none of these protocols synchronize directory information. Proper LDAP integration reduces administrative overhead and ensures consistent user information across the collaboration environment.
Question 118:
What is the maximum number of Cisco Unified Communications Manager servers that can exist in a single cluster?
A) 4
B) 8
C) 16
D) 32
Answer: B
Explanation:
Cisco Unified Communications Manager clusters support a maximum of eight servers, providing a scalable architecture for enterprise-wide unified communications deployments. This cluster limitation balances system capacity, management complexity, and database replication overhead while supporting tens of thousands of registered devices and concurrent calls.
Within an eight-server cluster, each server can perform different roles to optimize system functionality and performance. Typical deployments include multiple call processing subscribers that handle device registration and call control, a publisher server that maintains the master database and provides administration access, and dedicated servers for specific services like Cisco Unity Connection voice messaging or IM and Presence Service. The publisher server replicates database changes to all subscriber servers, ensuring configuration consistency across the cluster.
Cluster scalability enables organizations to distribute registered devices across multiple subscriber servers for load balancing and redundancy. Each subscriber can support thousands of devices depending on server hardware specifications and feature complexity. When one subscriber fails, devices automatically re-register to backup servers configured in their device pool settings, maintaining service continuity. Inter-cluster trunks or SIP route patterns enable call routing between multiple clusters when organizational requirements exceed single-cluster capacity.
Database replication represents a critical cluster function where the publisher continuously synchronizes configuration data to all subscribers. This replication occurs in near real-time through the Informix database architecture. Hardware specifications and network connectivity between cluster members significantly impact cluster performance and reliability. Geographic dispersion of cluster members requires careful consideration of network latency, as excessive delays can affect database replication and call processing performance. Understanding cluster architecture and limitations is essential for designing resilient, high-capacity unified communications solutions.
Question 119:
Which feature provides users the ability to log into any Cisco IP phone and access their personal settings?
A) Device Mobility
B) Extension Mobility
C) Hot Desking
D) Unified Mobility
Answer: B
Explanation:
Extension Mobility is a Cisco Unified Communications Manager feature that allows users to temporarily transform any shared IP phone into their personal device by logging in with their credentials. When users authenticate through the phone’s Extension Mobility service, the device downloads their personal profile including directory numbers, speed dials, line appearances, and service subscriptions, providing a consistent experience regardless of physical location.
The Extension Mobility service operates through the Cisco Extension Mobility Service running on Unified Communications Manager servers. Users access the feature by pressing the Services button on an IP phone and selecting the Extension Mobility option. After entering their user ID and PIN, the phone communicates with the Extension Mobility service to download the user’s device profile. The phone then resets and registers with the user’s configuration, displaying their extension and personalized settings. When users finish their session, they log out, returning the phone to its default configuration.
This capability particularly benefits organizations with shared workspace environments, hotel desks, or remote workers who move between office locations. Sales teams, consultants, and employees in hot-desking arrangements can maintain their communications identity and access their contacts and features from any enabled phone. Extension Mobility also integrates with Active Directory for single sign-on capabilities, simplifying authentication without requiring separate PIN management.
Device Mobility automatically adjusts phone settings based on detected network location but does not transfer user identity. Hot Desking represents a generic workspace sharing concept rather than a specific Cisco feature. Unified Mobility integrates mobile devices with desk phones for seamless call handoff but does not enable logging into different phones. Extension Mobility remains the definitive solution for user profile portability across Cisco IP phone infrastructure.
Question 120:
What is the purpose of a Session Border Controller in a Cisco collaboration environment?
A) To provide DHCP services to IP phones
B) To secure and control SIP signaling and media traffic at network boundaries
C) To replicate databases between Unified Communications Manager servers
D) To transcode between different audio codecs
Answer: B
Explanation:
Session Border Controllers (SBCs) serve as critical security and interoperability components positioned at network boundaries where the enterprise unified communications infrastructure connects to external networks, service providers, or untrusted domains. These specialized devices control signaling and media traffic flows, enforcing security policies while ensuring protocol compatibility and reliable communications across network boundaries.
The security functions of SBCs include topology hiding where internal network architecture is concealed from external parties, preventing reconnaissance attacks. SBCs implement stateful firewall capabilities specifically designed for SIP and RTP protocols, permitting legitimate communications while blocking malicious traffic. Denial-of-service protection detects and mitigates attack patterns that could overwhelm infrastructure. Encryption and decryption of signaling and media streams protect sensitive communications from eavesdropping. Access control lists restrict communications to authorized sources and destinations.
Interoperability represents another crucial SBC function as different vendors and service providers may implement protocols with variations that cause compatibility issues. SBCs normalize protocol implementations, translate between different SIP profiles, and manipulate headers to ensure successful communications. They manage Network Address Translation traversal challenges by modifying embedded IP addresses within SIP message bodies and managing media pinholes through firewalls. Call admission control limits concurrent sessions to prevent network congestion and service degradation.
DHCP services are provided by network infrastructure rather than SBCs. Database replication occurs within Unified Communications Manager clusters through publisher-subscriber relationships. Transcoding converts between codecs using media resource devices. Cisco Unified Border Element (CUBE) implements SBC functionality within the Cisco collaboration portfolio, providing enterprises with robust security and connectivity for external communications. Understanding SBC capabilities is essential for securing modern unified communications deployments.