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Question 181:
An administrator is configuring presence functionality in Cisco Unified Communications Manager. Which service must be activated to support presence features?
A) Cisco Presence Engine
B) Cisco SIP Proxy
C) Cisco Unified Presence
D) Cisco AXL Web Service
Answer: A
Explanation:
The Cisco Presence Engine service must be activated on Cisco Unified Communications Manager to support presence functionality. This service is responsible for collecting and distributing presence information for users within the collaboration environment. Presence allows users to see the real-time availability status of their colleagues, including whether they are available, busy, on a call, in a meeting, or away from their desk, which improves communication efficiency and reduces unnecessary interruptions.
The Cisco Presence Engine works by monitoring various presence indicators including phone line status, calendar information when integrated with Microsoft Exchange or other calendaring systems, and manual status settings configured by users. The service aggregates this information and makes it available to presence-aware applications such as Cisco Jabber, Webex, and other unified communications clients. When a user status changes, such as when they answer a phone call, the Presence Engine updates their status and notifies subscribed clients in near real-time.
The Presence Engine must be activated through Cisco Unified Serviceability on one or more CUCM nodes depending on the deployment size and redundancy requirements. In larger deployments, multiple presence servers may be configured to distribute the processing load and provide high availability. The service integrates with the CUCM database to access user information and directory numbers, and it uses SIP SUBSCRIBE and NOTIFY messages to communicate presence updates between servers and clients.
Cisco SIP Proxy handles SIP message routing but does not provide presence aggregation. Cisco AXL Web Service provides API access for administrative functions. While Cisco Unified Presence is a related product, the specific CUCM service for presence is the Cisco Presence Engine, making this the correct answer for enabling presence functionality within CUCM.
Question 182:
Which codec provides the best voice quality but requires the most bandwidth for a single G.711 call?
A) 64 kbps
B) 32 kbps
C) 16 kbps
D) 8 kbps
Answer: A
Explanation:
G.711 codec requires 64 kbps of bandwidth for the payload portion of a single voice call and provides the best voice quality among commonly deployed voice codecs. G.711 uses Pulse Code Modulation (PCM) to digitize analog voice signals by sampling the audio 8,000 times per second with 8 bits per sample, resulting in a 64 kbps data stream. This high sampling rate and bit depth preserve voice quality with minimal compression artifacts, making G.711 the gold standard for voice quality in VoIP deployments.
The 64 kbps bandwidth requirement represents only the voice payload data. When calculating total bandwidth consumption for capacity planning, administrators must also account for packet overhead including IP headers (20 bytes), UDP headers (8 bytes), and RTP headers (12 bytes). With standard 20-millisecond packetization, a G.711 call typically consumes approximately 80-90 kbps of total bandwidth including all headers. This bandwidth requirement is significantly higher than compressed codecs but ensures toll-quality voice with mean opinion scores (MOS) consistently above 4.0.
G.711 comes in two variants: G.711u (mu-law) used primarily in North America and Japan, and G.711a (a-law) used in Europe and most other regions. Both variants provide identical quality and bandwidth characteristics. The codec is computationally simple, requiring minimal processing power for encoding and decoding, which makes it ideal for situations where bandwidth is abundant and voice quality is the primary concern, such as LAN environments or high-bandwidth WAN connections.
Organizations deploying voice over bandwidth-constrained links often use compressed codecs like G.729 (8 kbps) or G.722 (64 kbps for wideband) as alternatives. Understanding codec bandwidth requirements is essential for proper network capacity planning, QoS design, and ensuring adequate voice quality for end users throughout the collaboration infrastructure.
Question 183:
An engineer is configuring a Cisco Expressway-C and Expressway-E for Mobile and Remote Access. Which protocol is used for firewall traversal between the two Expressways?
A)323
B) SIP
C) HTTPS
D)460
Answer: C
Explanation:
HTTPS (HTTP Secure) is the protocol used for firewall traversal between Cisco Expressway-C (Core) and Expressway-E (Edge) when deploying Mobile and Remote Access (MRA) functionality. The communication between these two Expressway servers uses HTTPS tunnels to securely traverse the corporate firewall, allowing remote users to access internal Cisco Unified Communications Manager services from outside the enterprise network without requiring a VPN connection.
The Expressway-E is deployed in the DMZ or external network and accepts incoming connections from remote endpoints using HTTPS on TCP port 8443. The Expressway-C is deployed inside the corporate network and maintains a persistent HTTPS connection to the Expressway-E. This architecture requires only outbound connections from the internal network to the DMZ, which is more secure and firewall-friendly than allowing inbound connections from the internet directly to internal resources. The HTTPS tunnels carry SIP signaling and media traffic encapsulated within the secure HTTP connections.
When a remote Cisco Jabber client connects to corporate services, it establishes an HTTPS connection to the Expressway-E, which then proxies the traffic through the existing HTTPS tunnel to the Expressway-C. The Expressway-C then forwards the traffic to the appropriate internal services such as CUCM, Cisco Unity Connection, or presence servers. This elegant design provides secure remote access without exposing internal collaboration infrastructure to the internet or requiring complex VPN configurations for end users.
H.323 and H.460 are protocols used for video and firewall traversal in different contexts but are not the primary protocols for Expressway-C to Expressway-E communication in MRA deployments. SIP is the signaling protocol carried within the HTTPS tunnels but is not the tunnel protocol itself, making HTTPS the correct answer.
Question 184:
Which feature in Cisco Unified Communications Manager allows administrators to apply different calling privileges based on the physical location where a phone is connected?
A) Device Mobility
B) Extension Mobility
C) Automated Alternate Routing
D) Call Admission Control
Answer: A
Explanation:
Device Mobility in Cisco Unified Communications Manager enables administrators to automatically apply different calling privileges and configurations based on the physical network location where an IP phone is connected. This feature is particularly valuable for supporting roaming users who move their phones between different office locations, work from home, or travel to branch offices, as it ensures phones maintain appropriate configuration for their current location without manual administrative intervention.
Device Mobility works by comparing the phone current subnet (detected from its IP address) against configured Device Mobility Information. When CUCM detects that a phone has connected from a subnet associated with a different location than its home location, it can automatically switch the phone to use a roaming Device Pool. This roaming Device Pool applies location-appropriate settings including different calling search spaces for external dialing, region settings for codec selection, SRST references for local survivability, and location-based call admission control parameters.
The calling privilege adjustment is particularly important for controlling costs and ensuring proper emergency services routing. For example, a phone configured for a US office might have unrestricted international dialing when connected at its home location. When the same phone travels to a branch office in another country, Device Mobility can automatically apply a more restrictive calling search space that blocks expensive international calls or routes them differently. Similarly, the phone emergency calls will route to the appropriate local emergency services based on detected location.
Extension Mobility allows users to log into different phones with their profiles but does not automatically adjust settings based on network location. Automated Alternate Routing provides call rerouting during WAN failures. Call Admission Control manages bandwidth but is not the feature that changes calling privileges based on physical location, making Device Mobility the correct answer for this location-aware configuration capability.
Question 185:
An administrator configures a Route Group in Cisco Unified Communications Manager. What is the primary purpose of a Route Group?
A) To group devices that share the same location
B) To group gateways or trunks for call routing
C) To group phones into hunt groups
D) To group media resources together
Answer: B
Explanation:
The primary purpose of a Route Group in Cisco Unified Communications Manager is to group gateways or trunks together for call routing purposes. Route Groups are logical containers that hold one or more devices capable of routing calls to external destinations, such as PSTN gateways, SIP trunks, H.323 gateways, or MGCP gateways. By organizing these devices into Route Groups, administrators can implement sophisticated routing strategies including load balancing, redundancy, and preference-based routing for outbound calls.
Within a Route Group, administrators define the order in which devices should be attempted when routing calls. The distribution algorithm determines how calls are allocated across the available devices in the group. Options include Top Down (always try devices in order from first to last), Circular (distribute calls evenly across all devices), and Priority-based (prefer certain devices but use others as backup). This flexibility allows organizations to optimize gateway utilization, implement least-cost routing strategies, and ensure call completion even when primary routes are unavailable or at capacity.
Route Groups become even more powerful when combined with Route Lists, which contain multiple Route Groups in priority order. This two-tier architecture enables complex routing scenarios such as attempting local gateways first, then regional gateways, and finally centralized gateways as a last resort. The Route Group also supports digit manipulation and calling party transformations, allowing administrators to modify called and calling numbers appropriately for each gateway or trunk in the group.
Device Pools group devices by location but serve a different purpose. Hunt Groups are configured using Line Groups and Hunt Lists for distributing incoming calls. Media Resource Groups organize conferencing and transcoding resources. Only Route Groups specifically organize gateways and trunks for outbound call routing, making this their primary and distinguishing function within CUCM architecture.
Question 186:
Which command displays the current registration status of SIP endpoints on Cisco Unified Border Element?
A) show sip-ua status registrar
B) show voice register pool
C) show sip-ua register status
D) show voice register statistics
Answer: C
Explanation:
The show sip-ua register status command displays the current registration status of SIP endpoints on a Cisco Unified Border Element (CUBE). This command provides comprehensive information about SIP registrations that CUBE is maintaining, including which endpoints are currently registered, their registration state, the registrar server they registered with, registration expiration times, and associated contact information. This visibility is essential for troubleshooting registration issues and verifying that endpoints can successfully register through CUBE.
When CUBE is configured as a SIP registrar proxy or back-to-back user agent, it maintains registration state for SIP endpoints that register through it. The show sip-ua register status command output includes details such as the registered user address-of-record (AOR), the contact address where the endpoint can be reached, the registration expiration time showing how long until the registration must be refreshed, and the current state of the registration (active, expired, or pending). This information helps administrators verify that remote endpoints are properly registered and can receive incoming calls.
The command is particularly useful in troubleshooting scenarios where calls fail to reach registered endpoints. By examining the registration status, engineers can determine if registrations are timing out prematurely, if endpoints are registering with incorrect contact information, or if registration refresh messages are failing. The output also helps identify registration conflicts or duplicate registrations that could cause call routing problems.
The show sip-ua status registrar command shows registrar configuration but not current registrations. The show voice register pool command is used on Cisco Unified Communications Manager Express, not CUBE. The show voice register statistics command does not exist as a standard IOS command. Only show sip-ua register status provides the specific registration status information needed for troubleshooting SIP endpoint registrations on CUBE.
Question 187:
An engineer needs to configure Cisco Unity Connection to deliver voicemail notifications via SMS. Which feature must be configured?
A) SMTP Smart Host
B) Message Notification
C) External Message Store
D) Unified Messaging
Answer: B
Explanation:
The Message Notification feature in Cisco Unity Connection must be configured to deliver voicemail notifications via SMS text messages to user mobile phones. Message Notification is a user-configurable feature that allows Unity Connection subscribers to receive alerts when new voicemail messages arrive in their mailbox. These notifications can be delivered through multiple methods including SMS text messages, email, pager, and phone calls to external numbers, providing users with immediate awareness of new messages even when away from their desk phones.
When configuring SMS notifications, administrators must first establish connectivity between Unity Connection and an SMS gateway or service provider that can deliver text messages to mobile carriers. This typically involves configuring an SMTP connection to an SMS-to-email gateway service where email messages sent to specially formatted addresses (such as phonenumber@carrier-sms-gateway.com) are converted to SMS messages. Users then configure their individual notification preferences through either the Unity Connection web interface or telephone user interface, specifying their mobile phone number and when they want to receive notifications.
The Message Notification feature provides granular control options including scheduling when notifications should be sent, setting up different notification rules based on caller identity, configuring escalation where multiple notification methods are attempted in sequence, and specifying whether notifications should include message details like caller name and callback number. This flexibility ensures users stay connected to important communications while minimizing notification fatigue from less urgent messages.
SMTP Smart Host is used for outbound email delivery but does not specifically enable notification features. External Message Store relates to accessing voicemail in Exchange mailboxes. Unified Messaging provides Exchange integration but is not the feature for configuring SMS notifications. Message Notification is the specific Unity Connection feature that enables and controls SMS delivery of voicemail alerts.
Question 188:
Which three QoS mechanisms are recommended for voice traffic in a converged network? (Choose three)
A) Low Latency Queuing (LLQ)
B) Weighted Fair Queuing (WFQ)
C) Classification and Marking
D) Traffic Shaping
E) Policing
F) Call Admission Control
Answer: A, C, F
Explanation:
For voice traffic in converged networks, three QoS mechanisms are particularly recommended: Low Latency Queuing (LLQ), Classification and Marking, and Call Admission Control. These mechanisms work together to ensure voice packets receive appropriate prioritization and that network resources are not oversubscribed, maintaining consistent voice quality across the infrastructure.
Low Latency Queuing (LLQ) is essential for voice because it provides strict priority queuing for time-sensitive traffic. LLQ ensures voice packets are always serviced first before any other traffic types, regardless of other queue depths. This strict prioritization keeps voice latency consistently low (typically under 150ms end-to-end) and jitter minimal (under 30ms), which are critical requirements for toll-quality voice. LLQ is implemented at every network hop where queuing might occur, particularly on WAN links where congestion is more likely.
Classification and Marking establish the foundation of any QoS strategy by identifying voice packets and marking them with appropriate DSCP values (typically EF/46 for voice bearer traffic). Classification can be performed at various points including the endpoint, access switch, or router. Proper marking ensures that every network device along the path can identify voice traffic and apply appropriate QoS policies. Without accurate classification and marking, other QoS mechanisms cannot distinguish voice from other traffic types.
Call Admission Control (CAC) prevents network oversubscription by limiting the number of concurrent calls allowed over bandwidth-constrained links. CAC mechanisms like locations-based CAC in CUCM or RSVP calculate available bandwidth and reject new call attempts when insufficient bandwidth remains, ensuring existing calls maintain quality rather than allowing all calls to degrade. This proactive approach prevents the choppy audio, dropped packets, and poor quality that occur when too many voice calls compete for limited bandwidth.
While useful in some contexts, WFQ, Traffic Shaping, and Policing are not among the top three recommended mechanisms specifically for voice traffic prioritization.
Question 189:
An administrator is troubleshooting DTMF issues with a SIP trunk. Which method should be configured to ensure DTMF tones are transmitted reliably across the trunk?
A) In-band audio
B) SIP INFO
C) RFC 2833
D) SIP NOTIFY
Answer: C
Explanation:
RFC 2833 (also known as RTP Telephone Events) is the most reliable method for transmitting DTMF tones across SIP trunks and should be configured when troubleshooting DTMF issues. RFC 2833 transmits DTMF digits as distinct RTP packets separate from the voice stream, using a special payload type to indicate telephone events. This out-of-band method is codec-independent and reliably delivers DTMF across networks even when voice codecs compress or distort the audio frequencies used for in-band DTMF tones.
The RFC 2833 method works by detecting DTMF tones at the originating gateway or endpoint and converting them into named telephone events (such as digit 5 or digit pound) that are sent in special RTP packets. These packets are sent multiple times for redundancy to ensure reliable delivery even if some packets are lost. The receiving gateway or endpoint then regenerates the DTMF tone locally or passes the digit information to the connected system. This approach is superior to transmitting DTMF as audio because compressed codecs like G.729 can distort the precise frequencies required for DTMF recognition, causing digit recognition failures on IVR systems, voicemail platforms, or remote PBX systems.
When configuring SIP trunks between CUCM and service providers or between CUCM and CUBE, administrators must ensure both ends are configured for RFC 2833. This is typically configured in the SIP Profile assigned to the trunk, with the DTMF method set to RFC 2833 or “out-of-band.” Mismatched DTMF settings between endpoints are a common cause of DTMF failures where users report that IVR systems do not respond to their keypad entries.
In-band audio transmission of DTMF is unreliable with compressed codecs. SIP INFO is an alternative out-of-band method but is less commonly supported and reliable. SIP NOTIFY is used for other purposes like message waiting indication, not DTMF transmission, making RFC 2833 the preferred standard method.
Question 190:
Which Cisco Unified Communications Manager feature allows administrators to automatically configure phones based on predefined templates during the initial installation?
A) Extension Mobility
B) Auto-registration
C) Bulk Administration Tool (BAT)
D) Device Defaults
Answer: C
Explanation:
The Bulk Administration Tool (BAT) in Cisco Unified Communications Manager allows administrators to automatically configure large numbers of phones based on predefined templates during initial installation or during bulk phone additions to the system. BAT is a web-based tool integrated into CUCM that dramatically reduces the time and effort required to configure multiple devices by allowing administrators to create configuration templates, generate device records in bulk, and apply consistent settings across many phones simultaneously.
BAT works through a template-based approach where administrators first create phone templates containing common configuration parameters such as device pool, location, calling search space, media resource group list, and button layouts. These templates capture the standard configuration that should apply to groups of similar phones. Administrators then use BAT to create CSV files or use the BAT spreadsheet tools to define device-specific information like MAC addresses and directory numbers. BAT processes these files and automatically creates all the phone records in the CUCM database with the appropriate template settings applied.
The tool is particularly valuable during initial installations of large phone systems where hundreds or thousands of phones need to be configured quickly and consistently. BAT supports bulk operations for phones, users, directory numbers, and other CUCM objects. It also supports bulk modifications, allowing administrators to update settings across many existing devices simultaneously. BAT can integrate with Cisco Tool for Auto-Registered Phones Support (TAPS) to streamline phone deployment where phones auto-register first and then are converted to specific configurations based on BAT-prepared data.
Extension Mobility allows users to access their profiles on different phones but does not provide bulk configuration capabilities. Auto-registration allows phones to register without pre-configuration but does not apply detailed templates. Device Defaults provide baseline settings but lack the bulk configuration power of BAT for large-scale phone deployments.
Question 191:
An engineer is configuring a SIP trunk on Cisco Unified Border Element. Which command is used to configure the dial-peer for the SIP trunk?
A) dial-peer voice 1 voip
B) voice class sip-options-keepalive
C) sip-ua
D) voice service voip
Answer: A
Explanation:
The dial-peer voice 1 voip command is used to configure a VoIP dial-peer for SIP trunks on Cisco Unified Border Element. Dial-peers are fundamental configuration elements in Cisco IOS voice gateways and CUBE that define how calls should be routed and what attributes should be applied to call legs. For SIP trunks, VoIP dial-peers specify the destination addresses, codec preferences, DTMF relay methods, and various SIP-specific parameters needed to establish and maintain calls with remote SIP endpoints or service providers.
When configuring a SIP trunk dial-peer, administrators specify critical parameters including the destination pattern that matches dialed digits, the session protocol (SIP), the target destination using either session target commands specifying IP addresses or DNS names, codec preferences, and voice class associations that define SIP profiles, URI handling, and other protocol behaviors. A typical SIP trunk dial-peer configuration includes commands for inbound and outbound call matching, digit manipulation, and SIP header modifications necessary for proper interoperability with the connected SIP system.
The dial-peer configuration serves different purposes depending on whether it is an inbound or outbound dial-peer. Outbound dial-peers match dialed digits and determine where to send calls, while inbound dial-peers match incoming calls based on called number, calling number, or SIP headers and determine what call treatment to apply. CUBE typically uses multiple dial-peers for each trunk direction to implement sophisticated routing, header manipulation, and call policy enforcement. The dial-peer architecture provides the flexibility needed for complex SIP trunk deployments with multiple service providers or interconnected UC systems.
Voice class sip-options-keepalive configures SIP OPTIONS keepalive behavior but is not the command for creating dial-peers. The sip-ua command configures global SIP user agent parameters. Voice service voip sets global VoIP parameters. Only dial-peer voice creates the actual trunk configuration for routing calls.
Question 192:
Which protocol does Cisco Unified Communications Manager use to control MGCP gateways?
A)323
B) SIP
C) MGCP
D) SCCP
Answer: C
Explanation:
Cisco Unified Communications Manager uses MGCP (Media Gateway Control Protocol) to control MGCP gateways, as the protocol name directly indicates. MGCP is a master-slave protocol where CUCM acts as the call agent or controller and the gateway acts as a slave device that follows commands from CUCM. This centralized control model allows CUCM to maintain complete control over call routing, digit manipulation, codec selection, and other call processing decisions while the gateway simply performs the instructed media conversion and connectivity functions.
MGCP architecture separates call control intelligence from media gateway functions. CUCM sends commands to the MGCP gateway instructing it to create connections, modify connections, notify about events (like digits dialed or phones going off-hook), delete connections, and audit endpoint states. The gateway responds to these commands and sends notifications back to CUCM when events occur. This command-response model makes MGCP gateways simpler to manage than autonomous protocols like H.323 because all configuration and call routing intelligence resides centrally in CUCM rather than being distributed to each gateway.
MGCP gateways are commonly deployed for analog phone connections (FXS ports), analog trunk connections to traditional PBX or central office lines (FXO ports), T1/E1 PRI connections, and BRI connections. The protocol uses UDP port 2427 for gateway communication and 2727 for backhaul connections. MGCP provides reliable gateway control with features like connection auditing to recover from failures and transaction identifiers to match commands with responses, ensuring robust gateway operation even in network disruption scenarios.
H.323 and SIP are peer-to-peer signaling protocols that operate differently from MGCP master-slave model. SCCP is used for controlling Cisco IP phones, not gateways. MGCP is the specific protocol designed for centralized gateway control, making it the correct answer for how CUCM controls MGCP gateways in Cisco collaboration deployments.
Question 193:
An administrator needs to configure time-of-day routing in Cisco Unified Communications Manager. Which two components are required? (Choose two)
A) Time Period
B) Time Schedule
C) Partitions
D) Translation Patterns
E) Route Filters
Answer: A, C
Explanation:
Configuring time-of-day routing in Cisco Unified Communications Manager requires two essential components: Time Periods and Partitions. These work together to control when specific dial plan elements are accessible to users, enabling scenarios like routing calls to different destinations during business hours versus after hours, or restricting expensive international calling to specific time windows. This time-based call routing capability is fundamental for implementing business policies and optimizing resource usage.
Time Periods define specific time windows including days of the week and hours of the day when certain routing behaviors should be active. Administrators create Time Period objects that specify schedules like “Monday through Friday 8:00 AM to 5:00 PM” for business hours or “Daily 5:00 PM to 8:00 AM plus weekends” for after hours. These Time Period objects can be as simple or complex as needed, supporting multiple time ranges and even recurring schedules for holidays. Time Periods are reusable objects that can be associated with multiple partitions, making schedule management efficient.
Partitions, which are fundamental to CUCM call routing architecture, can have Time Periods associated with them. When a Time Period is applied to a Partition, the dial plan elements within that Partition (such as directory numbers, translation patterns, or route patterns) are only accessible during the times defined in the Time Period. By creating multiple Partitions for the same resources but with different Time Periods, administrators implement time-based routing. For example, a Translation Pattern for routing to an automated attendant might exist in a “Business Hours” Partition that is only accessible during work hours, while a different Translation Pattern routing to an after-hours message exists in an “After Hours” Partition.
Time Schedule is not a valid CUCM configuration object. Translation Patterns and Route Filters can be used with time-of-day routing but are not required components. Only Time Periods and Partitions are the essential elements needed to implement time-based call routing in CUCM.
Question 194:
Which Cisco Jabber feature allows users to escalate an audio call to a video call without disconnecting?
A) Mid-call feature
B) Call upgrade
C) Call escalation
D) Video promotion
Answer: A
Explanation:
The mid-call feature in Cisco Jabber allows users to escalate an audio call to include video without disconnecting and re-establishing the call. This capability provides a seamless user experience where participants can start with a simple audio conversation and then add video when needed, such as when visual collaboration becomes necessary for discussing documents, demonstrating products, or enhancing personal connection. The transition happens smoothly without requiring users to hang up and initiate a new call with different media parameters.
Mid-call feature functionality works through SIP re-INVITE messages that renegotiate the media capabilities of an existing call session. When a user clicks the video button during an active audio call in Jabber, the client sends a SIP re-INVITE to the remote endpoint proposing to add video streams to the existing audio-only call. If the remote endpoint supports video and the user accepts, both endpoints negotiate video codec parameters and establish video RTP streams while maintaining the existing audio streams. This protocol-level capability is supported by Cisco Unified Communications Manager and requires that both endpoints have video capabilities and that network bandwidth is sufficient for video.
The mid-call feature extends beyond just audio-to-video escalation. Users can also add or remove additional audio or video streams, start or stop screen sharing, and modify other media parameters during active calls without disconnection. This flexibility makes Jabber conversations more natural and adaptive to changing communication needs without the disruption of ending and restarting calls. The feature respects privacy settings and permissions, allowing users to decline video additions or control camera activation.
Call upgrade, call escalation, and video promotion are not standard Cisco terminology for this feature. The official term used in Cisco documentation and Jabber interfaces is mid-call feature or mid-call capabilities, making this the correct answer for describing the functionality that allows seamless media modification during active calls.
Question 195:
An engineer is configuring call recording on Cisco Unified Communications Manager. Which component is responsible for capturing and storing the media streams?
A) Cisco Unified Serviceability
B) Media Resource Manager
C) Recording Server
D) Built-in Bridge
Answer: C
Explanation:
The Recording Server is the component responsible for capturing and storing media streams when call recording is configured in Cisco Unified Communications Manager environments. A Recording Server is a dedicated application, typically a third-party solution that integrates with CUCM, designed to receive, process, store, and manage recorded voice and video conversations. These systems are essential for organizations requiring call recording for compliance, quality assurance, training, dispute resolution, or customer service improvement purposes.
Recording Servers integrate with CUCM using either SPAN/monitor port technology where network traffic is mirrored to the recording system, or through built-in bridge (BIB) technology where the recording server is explicitly inserted into the media path of calls that need to be recorded. The built-in bridge method is more selective and efficient because it allows administrators to specify exactly which calls should be recorded based on criteria like user identity, line appearance, or call direction. When a call meeting the recording criteria is established, CUCM signals the Recording Server, which then receives media streams directly and stores them according to configured retention policies.
Modern recording solutions provide extensive capabilities beyond simple media capture including metadata tagging with caller information, screen capture integration, quality monitoring analytics, searchable recording archives, role-based access controls for playback, and integration with workforce management systems. Recording Servers must provide secure storage with encryption, audit trails showing who accessed recordings, and retention management to comply with legal and regulatory requirements like PCI-DSS for payment card data or FINRA for financial services.
Cisco Unified Serviceability is used for monitoring and troubleshooting services but does not record calls. Media Resource Manager allocates resources but does not capture media. Built-in Bridge is a CUCM media resource that enables recording by providing media forking capability, but the actual capture and storage is performed by the Recording Server, making it the correct answer.