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Question 16:
What is the primary purpose of a Session Border Controller (SBC) in a VoIP network?
A) To provide DHCP services for IP phones
B) To secure and control SIP signaling and media streams at network borders
C) To convert analog signals to digital signals
D) To provide power over Ethernet to endpoint devices
Answer: B
Explanation:
A Session Border Controller is a critical network element deployed at the perimeter of modern VoIP and Unified Communications networks to manage, secure, and optimize the flow of real-time communication sessions across network boundaries. Positioned between enterprise networks and external service provider networks—or in some cases between different segments of large internal networks—an SBC serves as a gatekeeper for SIP signaling and the associated RTP media streams. Its fundamental purpose is to ensure that voice, video, and other SIP-based applications can traverse otherwise restrictive network environments safely, consistently, and with high performance. As VoIP environments frequently involve communication with untrusted or semi-trusted external entities, the SBC assumes an essential role in maintaining both security and interoperability, ensuring that calls connect properly while preventing external threats from compromising internal systems.
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One of the most important functions of an SBC is topology hiding, which involves masking all internal IP addressing and network structure from outside networks. Without topology hiding, external endpoints or potential attackers could glean sensitive information about the internal VoIP architecture, including device IPs, signaling paths, or domain details. SBCs prevent this by rewriting SIP headers and SDP parameters, ensuring that only SBC-managed addresses are visible externally. This enhances security while also simplifying firewall policies, as all inbound and outbound SIP and RTP traffic is consistently anchored to the SBC’s interfaces. Closely related to topology hiding is protocol normalization, or protocol interworking, in which the SBC resolves interoperability problems caused by differing SIP implementations. Because SIP is a flexible and loosely structured protocol, variations in header usage, message formatting, timer behavior, and SDP attributes often create compatibility issues. The SBC inspects, modifies, and normalizes these messages, allowing systems from different vendors or carriers to interoperate seamlessly without requiring administrators to modify internal UC infrastructure.
Another key responsibility of the SBC is enabling NAT and firewall traversal. Most enterprise networks use private IP addressing and strict firewall policies to prevent unauthorized access, but these protections can break SIP and RTP because they rely on embedded IP addresses, dynamic port allocations, and bidirectional traffic flows. An SBC intelligently manages these challenges by anchoring media, rewriting addresses in SIP and SDP, and maintaining firewall pinholes as needed. This ensures that call signaling and media streams traverse security boundaries without being blocked or misrouted. In doing so, the SBC becomes the authoritative source for routing RTP traffic, minimizing one-way audio, call failures, or media path inconsistencies that commonly arise in NAT environments.
Security remains one of the SBC’s most indispensable functions. SBCs defend the network against denial-of-service attacks that attempt to overwhelm VoIP services, block malformed packets that could exploit SIP parsing vulnerabilities, and enforce authentication and authorization policies that help prevent toll fraud or unauthorized use of trunk resources. Through rate limiting, access control lists, IP reputation filtering, and deep SIP inspection, the SBC ensures that only legitimate and properly formed signaling traffic reaches internal systems. Enhancing security further, SBCs support encryption for both signaling and media through TLS for SIP messages and SRTP for voice and video streams. This encryption protects communications from interception, tampering, or eavesdropping as data travels over public or untrusted networks.
In addition to security and interoperability, SBCs contribute significantly to maintaining service quality. They perform QoS enforcement by prioritizing latency-sensitive RTP traffic, managing bandwidth allocations, and implementing call admission control to prevent oversubscription of network resources. By ensuring that the network does not exceed its real-time traffic capacity, SBCs help preserve audio and video quality even under heavy load. They may also perform transcoding or codec negotiation, enabling endpoints with different codec support to communicate effectively while optimizing bandwidth use. While DHCP services are provided by dedicated DHCP servers, analog-to-digital conversion is performed by voice gateways, and Power over Ethernet delivery is handled by network switches, the SBC specializes in securing, managing, and controlling SIP sessions and RTP media. This makes it an indispensable component in enterprise VoIP and cloud telephony deployments where secure, reliable, and high-quality communication is essential.
Question 17:
Which codec provides the best voice quality but requires the highest bandwidth?
A)729
B)711
C)722
D) GSM
Answer: B
Explanation:
G.711 is a pulse code modulation codec designed to deliver toll-quality voice by sampling audio at a rate of 8 kHz and encoding each sample with 8 bits, resulting in a total bandwidth requirement of 64 kbps per call. As one of the foundational codecs in traditional telephony, G.711 has long been regarded as the benchmark for high-quality voice transmission due to its extremely low compression and near-transparent audio reproduction. Because it performs only minimal signal manipulation, the codec introduces negligible distortion and maintains a natural, clear representation of human speech. This simplicity also means very low computational overhead, allowing encoding and decoding operations to take place with latencies typically below one millisecond. For this reason, G.711 remains a preferred codec in enterprise collaboration environments where voice quality and minimal delay are top priorities and bandwidth availability is not a significant limiting factor.
The operation of G.711 is conceptually straightforward. The analog waveform of human speech is sampled 8000 times per second in accordance with the Nyquist theorem, which ensures accurate reconstruction of the frequency range relevant to voice communication. Each sample is then encoded using 8 bits through a companding algorithm that reduces the dynamic range of the signal while preserving intelligibility. This fixed-bit-rate encoding process consistently produces a 64 kbps stream, which is well within the capability of most local area networks and traditional TDM circuits. The codec comes in two standardized variants: mu-law, used primarily in North America and Japan, and A-law, used across Europe and most other parts of the world. While the underlying companding curves differ, both versions deliver effectively identical perceptual voice quality. The existence of these two variants ensures compatibility with regional telephony systems while maintaining consistent audio characteristics across global deployments.
One of the greatest advantages of G.711 lies in its balance of quality and simplicity. Because it avoids heavy compression, the codec does not suffer from the artifacts or voice distortion that can occur with low-bit-rate codecs. This makes it ideal for environments where voice clarity is essential, such as contact centers, executive communications, and conferencing systems. Additionally, G.711 requires very little processing power from endpoints, allowing even older IP phones, analog gateways, and software clients to handle calls efficiently. In LAN environments, where bandwidth is generally plentiful and predictable, the 64 kbps per call requirement is easy to accommodate, and the absence of compression delays contributes to smooth, natural call flow and minimal jitter.
Despite its strengths, G.711’s high bandwidth consumption becomes a disadvantage in WAN deployments or bandwidth-constrained networks. When multiple simultaneous calls must traverse remote links, VPN tunnels, MPLS circuits, or internet-based SIP trunks, using 64 kbps per stream can quickly saturate available resources. For this reason, organizations often deploy compressed codecs for WAN traversal while using G.711 internally. G.729, for example, compresses audio to 8 kbps, making it far more efficient for low-bandwidth links. However, the significant compression introduces noticeable degradation in audio quality, and processing delays are considerably higher than those associated with G.711. GSM, at approximately 13 kbps, offers moderate quality and is widely used in mobile networks, but it does not match the clarity of G.711 in enterprise environments. G.722, on the other hand, is a wideband codec that also operates at 64 kbps but samples audio at 16 kHz, delivering enhanced clarity and more natural sound. Its requirement for the same bandwidth as G.711 makes it more suitable for LAN use rather than WAN use, but its improved frequency response provides superior voice quality when supported end-to-end.
For environments where bandwidth is not a limitation and where maximum audio fidelity and minimal latency are desired, G.711 remains the preferred codec. Its reliability, broad compatibility, long-established standardization, and consistently excellent sound quality make it the default choice in many enterprise VoIP and unified communications deployments.
Question 18:
What is the maximum number of Active Unified CM servers recommended in a single cluster?
A) 4
B) 6
C) 8
D) 12
Answer: C
Explanation:
Cisco Unified Communications Manager clusters can support up to 8 active call processing servers in a single cluster configuration. This architecture provides significant scalability and redundancy for enterprise voice deployments, allowing organizations to distribute call processing load across multiple servers while maintaining centralized administration and dial plan management.
The cluster architecture operates with a distributed processing model where each server can handle call control for registered endpoints. All servers in the cluster share a common database that is replicated across the nodes, ensuring consistency of configuration data including users, phones, route patterns, and dial plan information. The database replication occurs automatically using a publisher-subscriber model.
Within the 8-server limit, one server functions as the publisher, which maintains the master database and is the only node where configuration changes can be made. The remaining servers operate as subscribers that receive replicated database updates from the publisher. Each subscriber can independently process calls for its registered endpoints, providing both load distribution and redundancy.
The 8-server recommendation represents a balance between scalability and manageability. Exceeding this limit can introduce database replication delays, increased network traffic for inter-cluster communication, and potential performance degradation. Organizations requiring more than 8 call processing servers should consider multi-cluster deployments connected through SAF or GDPR mechanisms.
Options A, B, and D represent either insufficient capacity for large enterprises or exceed the supported architectural limits. The 8-server maximum ensures optimal performance while supporting tens of thousands of users and devices in a properly designed deployment with appropriate hardware resources and network infrastructure.
Question 19:
Which protocol is used for presence and instant messaging in Cisco Jabber?
A) SIP
B) XMPP
C)323
D) MGCP
Answer: B
Explanation:
Extensible Messaging and Presence Protocol (XMPP) is the standard protocol used by Cisco Jabber for instant messaging and presence functionality. XMPP is an open-standard, XML-based protocol originally developed for real-time communication and has become the foundation for enterprise instant messaging systems.
XMPP operates on a client-server architecture where Jabber clients connect to Cisco Unified Presence or IM and Presence Service servers. The protocol supports real-time message delivery, presence status updates, contact list management, and multi-user chat capabilities. Users can see availability status of colleagues, send instant messages, and participate in persistent chat rooms all through XMPP communication.
The protocol uses XML stanzas to transmit information between clients and servers. Three primary stanza types exist: message stanzas for instant messaging content, presence stanzas for availability status updates, and IQ (info/query) stanzas for request-response interactions. XMPP connections typically use TCP port 5222 for client-to-server communication and can be secured using TLS encryption.
While SIP (option A) is used by Jabber for voice and video calling functions, H.323 (option C) is a legacy telephony protocol not commonly used in modern Jabber deployments, and MGCP (option D) is a gateway control protocol unrelated to messaging. XMPP provides the robust, extensible framework necessary for enterprise messaging with features including offline message storage, message archiving for compliance, and federation capabilities that allow communication between different organizations. The separation of messaging (XMPP) and calling (SIP) protocols allows Jabber to optimize each communication modality independently.
Question 20:
What is the purpose of Media Resource Groups (MRG) in Cisco Unified Communications Manager?
A) To group IP phones by location
B) To organize media resources and control their allocation to devices
C) To configure trunk groups for PSTN connectivity
D) To manage bandwidth allocation across WAN links
Answer: B
Explanation:
Media Resource Groups are logical containers in Cisco Unified Communications Manager that organize media resources such as transcoders, conference bridges, music on hold servers, and media termination points. The primary purpose of MRGs is to provide granular control over which media resources are available to specific devices or groups of devices within the collaboration environment.
MRGs work in conjunction with Media Resource Group Lists (MRGL) to create a hierarchical resource allocation system. An MRG contains one or more media resources of the same or different 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 media resource for a specific function, Unified CM searches through the assigned MRGL in order until it finds an available resource.
This architecture provides several benefits including resource optimization by ensuring devices use geographically nearby resources to minimize latency and WAN bandwidth consumption. It also enables resource redundancy where primary and backup resources can be configured in different MRGs within the same MRGL. Administrative control is enhanced through the ability to dedicate premium resources to specific user groups while providing basic resources to others.
While grouping phones by location (option A) is accomplished through device pools and locations, trunk groups (option C) are managed through route lists and route groups, and bandwidth management (option D) uses locations and call admission control mechanisms. MRGs specifically address the organization and allocation of media processing resources, making them essential for efficient media handling in distributed enterprise deployments with multiple sites and varying quality requirements.
Question 21:
Which feature allows Cisco Unified Communications Manager to route calls based on the time of day?
A) Partitions and Calling Search Spaces
B) Time-of-Day Routing
C) Call Forward All
D) Hunt Groups
Answer: B
Explanation:
Time-of-Day Routing is a feature in Cisco Unified Communications Manager that enables dynamic call routing decisions based on time schedules and date ranges. This functionality allows organizations to automatically route calls differently depending on business hours, holidays, weekends, or other time-specific criteria without manual intervention.
The feature operates through time periods and time schedules configured in Unified CM administration. Time periods define specific clock times and days of the week, while time schedules group multiple time periods together to create complex timing rules. These schedules are then applied to partitions, making certain route patterns accessible or inaccessible based on the current time.
When a call is placed, Unified CM evaluates the calling device’s Calling Search Space against available partitions while considering active time schedules. If a partition has an associated time schedule that is currently inactive, the routes within that partition are not considered for the call, causing Unified CM to search subsequent partitions in the CSS for alternative routing paths.
Common implementations include routing calls to voicemail or auto-attendant after business hours, directing emergency calls differently during nights and weekends, providing alternate routing during lunch breaks, and implementing holiday routing patterns. The feature can also work with hunt pilots to control when hunt groups are active.
While Partitions and Calling Search Spaces (option A) provide the mechanism for route filtering, Call Forward All (option C) is a static forwarding feature, and Hunt Groups (option D) distribute calls among multiple destinations, none of these specifically provide time-based routing logic. Time-of-Day Routing specifically addresses temporal call routing requirements essential for business continuity and customer service optimization.
Question 22:
What is the function of a Cisco Unity Connection voicemail port?
A) To provide network connectivity for the Unity Connection server
B) To handle concurrent voicemail sessions for recording and playback
C) To connect analog phones to the voicemail system
D) To provide administrative access to the system
Answer: B
Explanation:
Voicemail ports in Cisco Unity Connection represent licensed sessions that determine how many concurrent voicemail operations the system can handle simultaneously. Each port allows one user to interact with the voicemail system at any given time, whether for recording messages, retrieving messages, changing greetings, or performing other voicemail functions.
Unity Connection integrates with Unified Communications Manager through SIP or SCCP protocols, with each voicemail port appearing as a virtual device that can accept calls. When a call is forwarded to voicemail due to no answer, busy status, or direct voicemail deposit, it consumes one port for the duration of the session. The port is released when the caller finishes leaving a message or when a subscriber completes checking their messages.
Port licensing directly impacts system capacity and user experience. Organizations must size their port count based on expected peak concurrent usage, taking into account factors such as average message duration, call volume patterns, and the number of users who might simultaneously access voicemail. Insufficient ports result in callers receiving busy signals when attempting to leave messages during peak periods.
Advanced port features include overflow handling where calls queue when all ports are busy, and port grouping for different classes of service. Some deployments separate ports for message deposit operations from retrieval operations to ensure message checking always remains available even during high incoming call volumes.
Network connectivity (option A) is provided through standard Ethernet interfaces, analog phone connectivity (option C) is not applicable to Unity Connection’s IP-based architecture, and administrative access (option D) uses web interfaces and SSH rather than voicemail ports. Ports specifically govern user-facing voicemail session capacity.
Question 23:
Which QoS marking value is recommended for voice bearer traffic according to Cisco best practices?
A) AF41
B) CS3
C) EF
D) CS5
Answer: C
Explanation:
Expedited Forwarding (EF) with a DSCP value of 46 is the Cisco recommended marking for voice bearer traffic, which consists of the actual RTP media streams carrying voice conversations. This marking ensures that voice packets receive the highest priority treatment as they traverse the network, minimizing delay, jitter, and packet loss that would degrade call quality.
The EF per-hop behavior is defined in RFC 3246 and guarantees low loss, low latency, low jitter, and assured bandwidth for marked traffic. Voice traffic is particularly sensitive to network conditions, with quality degrading noticeably when one-way delay exceeds 150 milliseconds, jitter exceeds 30 milliseconds, or packet loss exceeds 1 percent. The EF marking instructs network devices to place these packets in priority queues with strict scheduling.
Implementation of EF marking occurs at the network edge, typically on IP phones, voice gateways, or first-hop switches using trusted boundaries. Once marked, the EF value is preserved across the network infrastructure, assuming all intermediate devices are configured to recognize and honor the marking. Network devices allocate dedicated bandwidth to EF queues and use sophisticated queue scheduling algorithms to service these packets preferentially.
AF41 (option A) is recommended for video conferencing traffic, CS3 (option B) is used for call signaling protocols like SIP and SCCP, and CS5 (option D) is designated for video signaling. The distinct marking values allow networks to apply differentiated treatment to different traffic types, but voice bearer traffic specifically requires the superior handling provided by EF marking to maintain toll-quality communications across IP networks.
Question 24:
What protocol does Cisco Unified Border Element (CUBE) use for transcoding services?
A)323
B) SCCP
C) SIP
D) MGCP
Answer: C
Explanation:
Cisco Unified Border Element uses Session Initiation Protocol (SIP) as the primary signaling protocol for collaboration services including transcoding. CUBE functions as a sophisticated SIP back-to-back user agent that can manipulate SIP signaling messages while providing media services such as codec transcoding, DTMF relay, and protocol normalization.
When CUBE performs transcoding, it terminates the media stream from one call leg, converts the audio codec to a different format, and then sends the transcoded stream on the other call leg. This process is transparent to the endpoints, which maintain their SIP signaling sessions with CUBE rather than directly with each other. The SIP signaling coordinates codec negotiation through SDP (Session Description Protocol) carried within SIP messages.
CUBE’s transcoding capability is particularly valuable when connecting networks that support different codec sets. For example, when an internal enterprise network using G.711 connects to a service provider network requiring G.729 to conserve bandwidth, CUBE can perform the necessary codec conversion. The device negotiates appropriate codecs with each side of the call using SIP INVITE, 200 OK, and ACK messages containing SDP codec offerings.
Modern CUBE deployments prefer SIP over legacy protocols due to its flexibility, extensibility, and interoperability advantages. While H.323 (option A) is a legacy signaling protocol that CUBE can also support, SCCP (option B) is used for device control within Unified CM clusters, and MGCP (option D) is a gateway control protocol. For border element functionality including transcoding, SIP provides the robust feature set and industry-wide compatibility required for service provider interconnections and enterprise SIP trunk deployments.
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Question 25:
Which component is responsible for call routing decisions in Cisco Unified Communications Manager?
A) Media Gateway Control Protocol
B) Digit Analysis Engine
C) Session Border Controller
D) Phone Load Server
Answer: B
Explanation:
The Digit Analysis Engine is the core component within Cisco Unified Communications Manager responsible for making all call routing decisions. This engine analyzes dialed digits against configured route patterns, translation patterns, and directory numbers to determine how each call should be processed and where it should be routed.
When a user dials a number, the digits are sent to Unified CM where the Digit Analysis Engine performs pattern matching against the dial plan database. The engine evaluates the dialed string against all possible matches considering the calling device’s Calling Search Space, which determines which partitions and their contained patterns are accessible. The analysis follows a longest-match-first algorithm, selecting the most specific pattern that matches the dialed digits.
The engine performs several sophisticated functions including digit manipulation using translation patterns, route filtering to select specific PSTN routes based on area codes or number ranges, and route selection among multiple available paths using route lists and route groups. It also handles special prefixes like urgent priority codes, blocks restricted numbers through route filters, and implements class of service restrictions.
Once digit analysis determines the destination, the engine selects appropriate gateways or trunks for PSTN calls, routes internal calls directly to destination devices, or forwards calls to voicemail systems. The engine also considers factors such as device availability, alternate routing for failed paths, and time-of-day restrictions during the routing decision process.
While MGCP (option A) is a gateway control protocol, SBC (option C) provides border security services, and Phone Load Server (option D) distributes firmware files, the Digit Analysis Engine specifically handles the critical function of interpreting dialed digits and determining call routing paths throughout the Unified CM cluster.
Question 26:
What is the primary purpose of Cisco Expressway-C in a collaboration deployment?
A) To provide external access for remote users
B) To handle internal firewall traversal and registration of endpoints
C) To transcode video calls between different codecs
D) To provide PSTN connectivity
Answer: B
Explanation:
Cisco Expressway-C (Core) is the internal component of the Expressway solution that resides within the enterprise network and handles internal firewall traversal, endpoint registration, and call control services. It works as the trusted internal gateway for collaboration traffic, managing communications between internal endpoints and providing a secure communication path to the external Expressway-E (Edge) server.
The primary responsibilities of Expressway-C include registering internal and remote video endpoints, SIP devices, and Jabber clients. It maintains these registrations and provides routing services for calls between registered endpoints. When deployed for Mobile and Remote Access, Expressway-C receives authenticated connections from Expressway-E and proxies them to appropriate internal resources such as Unified CM, Unity Connection, and IM and Presence servers.
Expressway-C implements zone-based architecture where different network segments are defined as zones with specific traversal and security policies. It performs protocol normalization, ensuring that various SIP and H.323 implementations can communicate effectively. The server also provides bandwidth management through zone-based and link-based call admission control, preventing network congestion.
Firewall traversal is accomplished through the Expressway-C to Expressway-E connection, which typically uses a single authenticated traversal connection that eliminates the need for inbound firewall port openings. This secure tunnel carries all remote access traffic between the DMZ and internal network.
While external access (option A) is provided by Expressway-E, transcoding (option C) requires separate transcoding resources, and PSTN connectivity (option D) is handled by voice gateways, the Expressway-C specifically focuses on internal network services and secure connectivity to the edge component for comprehensive remote access solutions.
Question 27:
Which dial plan component defines a collection of reachable route patterns in Cisco Unified CM?
A) Route Group
B) Route List
C) Partition
D) Gateway
Answer: C
Explanation:
A Partition in Cisco Unified Communications Manager is a logical container that holds dialable route patterns, directory numbers, translation patterns, and other dial plan elements. Partitions provide the fundamental building block for implementing class of service and toll fraud prevention by creating isolated collections of reachable destinations that can be selectively made available to different users or devices.
The partition system works in conjunction with Calling Search Spaces to create a powerful access control mechanism. Each dial plan element is assigned to a specific partition, and partitions are then grouped into Calling Search Spaces that determine which collections of patterns any given device can reach. This two-tier architecture allows administrators to create complex but manageable dial plans with granular control over calling privileges.
For example, an organization might create partitions named Internal, Local, Long Distance, and International, each containing the appropriate route patterns for those destinations. Executive phones might have a CSS that includes all four partitions, while lobby phones might only access Internal and Local partitions, effectively restricting them from making toll calls.
Partitions also enable duplicate patterns to exist in the dial plan for different purposes. The same number pattern can exist in multiple partitions with different routing destinations, allowing different treatment based on who is calling. This is particularly useful for implementing after-hours routing or providing different calling capabilities to various user groups without complex pattern manipulation.
Route Groups (option A) contain lists of gateways, Route Lists (option B) define the order to try route groups, and Gateways (option D) are physical devices providing PSTN connectivity. However, Partitions specifically define collections of reachable destinations that form the foundation of dial plan organization and access control.
Question 28:
What type of trunk is used to connect Cisco Unified CM to a SIP service provider?
A)323 Gateway
B) MGCP Gateway
C) SIP Trunk
D) SCCP Trunk
Answer: C
Explanation:
A SIP Trunk is the standard method for connecting Cisco Unified Communications Manager to Session Initiation Protocol-based service providers for PSTN connectivity, SIP-to-SIP calling, or inter-cluster communications. SIP Trunks provide a flexible, scalable, and cost-effective alternative to traditional TDM circuits by carrying voice calls over IP networks.
SIP Trunks in Unified CM are configured as logical connections that define the signaling and media parameters for communications with external SIP entities. The trunk configuration includes the destination IP address or fully qualified domain name of the service provider’s session border controller, transport protocol preferences (TCP, UDP, or TLS), codec preferences for media negotiation, and DTMF signaling methods.
The trunk operates as a peer-to-peer SIP relationship where both sides can initiate calls. Outbound calls from Unified CM to the PSTN are routed through route patterns that point to the SIP trunk, while inbound calls from the service provider arrive at the trunk and are routed based on called party number to appropriate internal destinations. SIP trunks support multiple concurrent calls over the same logical connection, with capacity limited only by licensing and bandwidth.
Advanced features available with SIP Trunks include header manipulation for calling party information, SIP OPTIONS ping for trunk health monitoring, codec filtering to control which audio formats are offered, and normalization scripts to address interoperability issues with different SIP implementations.
H.323 Gateway (option A) is a legacy protocol for voice over IP, MGCP Gateway (option B) is used for controlled gateway deployments where Unified CM maintains call control, and SCCP Trunk (option D) does not exist as a trunk type. SIP Trunks provide the modern, standards-based approach for service provider connectivity.
Question 29:
Which feature provides automatic failover for IP phones when the primary Unified CM server becomes unavailable?
A) Device Mobility
B) SRST (Survivable Remote Site Telephony)
C) Call Forward Unregistered
D) Device Pool with backup servers
Answer: D
Explanation:
Device Pool configuration with backup Unified CM servers provides automatic failover capability for IP phones when their primary call processing server becomes unavailable. Each phone in Unified CM is assigned to a device pool, which specifies up to three Unified CM servers in priority order: primary, secondary, and tertiary.
When a phone boots or loses connectivity to its registered Unified CM server, it automatically attempts to register with the servers in the order defined in its device pool. If the primary server is unreachable due to network failure, server outage, or maintenance, the phone seamlessly registers to the secondary server and continues operating with full call control functionality. This failover process typically completes within 30-60 seconds depending on network conditions and phone model.
The device pool approach provides within-cluster redundancy where all servers share the same dial plan and configuration database. Users maintain their directory numbers, speed dials, and calling privileges regardless of which server is actively controlling their phone. Call features including hold, transfer, conferencing, and voicemail integration continue to function normally during failover conditions.
Device Mobility (option A) provides location-aware services when phones roam between sites, SRST (option B) provides limited call control during WAN failures when phones cannot reach any Unified CM server in the cluster, and Call Forward Unregistered (option C) redirects calls to phones that have lost registration. While these features address specific scenarios, the Device Pool with configured backup servers specifically provides the automatic high availability mechanism for normal server redundancy within a Unified CM cluster, making it essential for resilient enterprise voice deployments.
Question 30:
What is the purpose of a SIP Normalization Script in Cisco Unified Communications Manager?
A) To modify bandwidth allocation for SIP calls
B) To manipulate SIP headers and messages for interoperability
C) To compress SIP signaling for bandwidth efficiency
D) To encrypt SIP messages for security
Answer: B
Explanation:
SIP Normalization Scripts in Cisco Unified Communications Manager are Lua-based scripts that allow administrators to modify SIP headers, message bodies, and signaling flows to achieve interoperability with diverse SIP implementations. These scripts provide deep customization capabilities for handling non-standard SIP behaviors from service providers, third-party systems, or legacy equipment.
The scripts execute at specific trigger points during SIP message processing, allowing inspection and modification of INVITE, 200 OK, BYE, and other SIP messages both inbound and outbound. Administrators can add, remove, or modify SIP headers such as From, To, Contact, or custom proprietary headers. They can also manipulate Session Description Protocol content within message bodies to adjust codec offerings, media port numbers, or connection information.
Common use cases include normalizing calling party information when service providers require specific header formats, working around SIP implementation bugs in third-party equipment, implementing custom call features through proprietary headers, and adjusting SDP parameters for media negotiation compatibility. Scripts can also perform conditional logic based on called or calling numbers, trunk identity, or header content.
Unified CM provides a built-in Lua scripting engine with SIP-specific libraries and functions. The platform includes script debugging capabilities and pre-built script examples for common scenarios. Multiple scripts can be chained together with execution order controls, and scripts can be applied globally or to specific trunks for targeted customization.
While bandwidth allocation (option A) is handled through regions and locations, compression (option C) is not a standard SIP feature, and encryption (option D) uses TLS transport, SIP Normalization Scripts specifically address the interoperability challenges inherent in connecting heterogeneous SIP environments through flexible message manipulation capabilities.