Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 6 Q76-90

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Question76

A CUCM administrator observes intermittent outbound call failures to remote sites, even though multiple gateways are available. Logs indicate CUCM sequentially attempts all gateways, causing call setup delays and occasional fast busy signals. Which configuration MOST effectively resolves this issue while ensuring high availability?

A) Remove all but one gateway from the route pattern
B) Implement Local Route Groups (LRG) to dynamically select gateways based on device pool
C) Use time-of-day routing to direct calls through specific gateways
D) Configure route lists with weighted priorities to force gateway selection

Answer: B

Explanation:

Enterprise telephony environments require predictable call setup and high availability for outbound calls. When multiple gateways are associated with a route pattern, CUCM attempts them sequentially by default. Sequential attempts can introduce latency and occasionally lead to fast busy signals if the first gateways are busy or unreachable. This behavior is particularly problematic during peak call periods or in multi-site deployments where call routing efficiency directly affects user experience. Sequential attempts do not leverage available redundancy efficiently, and while multiple gateways exist, CUCM does not immediately select the optimal gateway.

Option B, implementing Local Route Groups (LRG), is the optimal solution. LRG dynamically selects gateways based on the device pool of the calling endpoint, allowing CUCM to route calls directly to the most appropriate gateway without sequentially testing each one. This reduces call setup time, eliminates fast busy signals, and provides predictable call success rates. High availability is maintained because alternate gateways are still available if the primary gateway fails. LRG also provides scalability for large deployments and simplifies configuration management, aligning with Cisco best practices for multi-gateway environments.

Option A, removing all but one gateway, would reduce sequential attempts but introduces a single point of failure. If the lone gateway becomes unavailable, all outbound calls fail, compromising operational reliability.

Option C, time-of-day routing, optimizes call distribution based on schedules but does not address sequential attempts. It cannot prevent call setup delays or intermittent failures caused by sequential evaluation during real-time call attempts.

Option D, configuring weighted priorities in route lists, influences gateway preference but does not eliminate sequential attempts. If the preferred gateway is busy, CUCM still sequentially attempts other gateways, so the underlying problem persists. Weighted priorities help distribute traffic but do not address real-time call setup failures effectively.

Implementing LRG ensures efficient call routing, reduced latency, high availability, and predictable call performance, fully aligning with Cisco best practices for multi-gateway CUCM deployments. This approach addresses the root cause of intermittent outbound call failures and maintains reliable telephony services for enterprise users.

Question77

Remote Jabber clients connected via Mobile and Remote Access (MRA) experience delays in receiving presence updates for internal users. Internal users update normally. XMPP logs between Expressway-C and CUCM indicate inconsistent message delivery timing. Which configuration MOST effectively resolves this issue?

A) Increase polling intervals for CUCM subscriber devices
B) Enable persistent XMPP connections between Expressway-C and CUCM
C) Configure firewall to block non-essential ports to reduce traffic
D) Temporarily disable traversal zones to isolate the issue

Answer: B

Explanation:

Presence is critical in unified communications for real-time collaboration. Remote Jabber clients rely on XMPP messages relayed via Expressway-C to CUCM for presence updates. When delays occur for remote clients but not internal users, it indicates that XMPP sessions may not be persistent, leading to intermittent propagation. Internal users communicate directly with CUCM and therefore do not experience delays. Presence delays reduce collaboration efficiency, create confusion about user availability, and can negatively affect decision-making and productivity in enterprise environments.

Option B, enabling persistent XMPP connections between Expressway-C and CUCM, is the correct solution. Persistent XMPP connections maintain continuous sessions, ensuring immediate delivery of presence updates to remote clients. This eliminates the need to repeatedly establish sessions for each update, reducing latency and preventing missed notifications. Cisco best practices for MRA recommend persistent XMPP connections to guarantee reliable, real-time presence information. By maintaining persistent sessions, remote users receive timely updates, improving collaboration, communication efficiency, and overall user experience.

Option A, increasing polling intervals, would exacerbate delays. Polling determines how often CUCM queries devices for status; longer intervals reduce update frequency, worsening latency for remote clients.

Option C, blocking non-essential firewall ports, risks inadvertently preventing necessary XMPP traffic. While security is important, improper firewall configurations can disrupt communication and increase presence latency.

Option D, disabling traversal zones, would disconnect remote clients completely. Traversal zones are essential for secure MRA connectivity, and disabling them does not solve the underlying session persistence problem.

Persistent XMPP connections directly address delayed presence for remote users, ensuring real-time updates, reliability, and alignment with Cisco best practices for MRA deployments. This results in seamless and efficient communication, essential for enterprise productivity and user satisfaction.

Question78

In a distributed CUCM SIP conferencing environment, internal participants successfully join conferences, but remote SIP endpoints fail intermittently when added mid-call. SIP trunks are configured for delayed offer, and mid-call re-INVITEs fail. Which configuration MOST effectively resolves this problem?

A) Enable early offer SIP on trunks used by remote endpoints
B) Assign hardware MTPs to all endpoints
C) Reduce the number of SIP trunks to simplify signaling
D) Convert remote SIP endpoints to SCCP protocol

Answer: A

Explanation:

SIP conferencing relies on accurate Session Description Protocol (SDP) negotiation to establish media paths. Mid-call re-INVITEs allow new participants to join ongoing conferences. When delayed-offer trunks are used, SDP is included in the 200 OK response instead of the initial INVITE. Remote endpoints may fail to join because CUCM lacks SDP information at the outset, especially in NAT, firewall, or multi-site deployments. This lack of immediate media capability information prevents proper media path establishment, causing intermittent conference join failures for remote endpoints.

Option A, enabling early offer SIP on trunks used by remote endpoints, is the correct solution. Early offer includes SDP in the initial INVITE, providing CUCM with immediate media information about the remote endpoint. This allows media negotiation to occur proactively, ensuring that remote participants can join reliably. Cisco best practices recommend early offer for SIP trunks handling conferencing traffic to maintain predictable media behavior, reduce latency, and prevent mid-call failures. Early offer ensures efficient media negotiation, minimizes call setup delays, and eliminates intermittent join failures, maintaining conference integrity and quality.

Option B, assigning hardware MTPs, ensures codec compatibility but does not resolve SDP timing issues. While MTPs can assist with transcoding and support certain signaling scenarios, they do not prevent mid-call re-INVITE failures due to delayed SDP.

Option C, reducing SIP trunks, simplifies signaling but does not address SDP negotiation problems. The core issue is timing of SDP delivery, not the number of trunks.

Option D, converting remote endpoints to SCCP, is unnecessary. The issue is related to SDP timing, not protocol compatibility. Converting endpoints does not resolve mid-call join failures and adds unnecessary complexity.

Enabling early offer SIP guarantees proper media negotiation, reliable addition of remote participants, and predictable conferencing behavior, aligning with Cisco best practices for distributed SIP conferencing environments.

Question79

A Cisco Unity Connection deployment integrated with CUCM shows several seconds of silence at the beginning of voicemail playback. RTP flow analysis indicates no packet loss or jitter. Which configuration MOST effectively resolves this issue?

A) Disable Voice Activity Detection (VAD) on the SIP trunk between CUCM and Unity Connection
B) Adjust Message Waiting Indicator (MWI) extensions
C) Change Unity Connection codec to G.722
D) Move the voicemail pilot to a different partition

Answer: A

Explanation:

Initial silence during voicemail playback is typically caused by Voice Activity Detection (VAD). VAD suppresses RTP packets when low-energy audio is detected, conserving bandwidth during live calls. Voicemail messages often start with low-energy audio, which VAD interprets as silence, causing a noticeable delay at the beginning of playback. This negatively impacts user experience, causing frustration and confusion.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, is the correct solution. Disabling VAD ensures continuous RTP transmission, including low-energy audio, eliminating the initial silence. Cisco best practices for voicemail deployments recommend disabling VAD to provide seamless playback from the beginning of the message. Users hear messages immediately, improving satisfaction and avoiding confusion caused by missing content. The bandwidth impact is minimal because voicemail RTP traffic is relatively low compared to live call traffic.

Option B, adjusting MWI extensions, only affects lamp signaling and does not influence RTP flow or message playback.

Option C, changing the codec to G.722, may improve audio fidelity but does not resolve VAD-induced silence.

Option D, moving the voicemail pilot to a different partition, affects call routing but does not alter media delivery.

Disabling VAD directly addresses initial silence issues, ensuring uninterrupted voicemail playback, enhancing user experience, and following Cisco best practices for CUCM and Unity Connection integration.

Question80

Remote users report intermittent one-way audio when calling internal endpoints through a CUBE. RTP analysis shows that packets are sometimes sent to incorrect IP addresses before the CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

A) Enable delayed-offer SDP on the CUCM SIP trunk toward CUBE
B) Use SIP over UDP instead of TCP
C) Disable early media on CUBE
D) Enable symmetric RTP on CUBE

Answer: A

Explanation:

One-way audio in multi-site SIP deployments with CUBE often occurs when RTP is transmitted before CUBE has rewritten SDP, causing packets to be sent to incorrect IP addresses or ports. This issue is common in scenarios with early media, NAT, or multi-site SIP trunks. Proper media anchoring ensures correct RTP delivery to both endpoints and is essential for reliable communication.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP includes SDP in the 200 OK response rather than the initial INVITE, allowing CUBE to anchor media, rewrite addresses and ports, and establish a correct media path before RTP begins. Cisco best practices recommend delayed-offer SDP for trunks through CUBE when early media may misalign media paths. This guarantees accurate RTP delivery, eliminates one-way audio, and maintains reliable communication between remote and internal endpoints.

Option B, using UDP instead of TCP, changes transport but does not correct SDP timing or media misalignment.

Option C, disabling early media, prevents call progress tones but does not resolve RTP misdelivery. It addresses symptoms rather than the root cause.

Option D, enabling symmetric RTP, assists with NAT traversal but does not fix early media being sent to incorrect addresses. Symmetric RTP does not correct SDP timing, which is the primary cause of one-way audio issues.

Implementing delayed-offer SDP ensures proper media anchoring through CUBE, eliminates one-way audio, and provides reliable multi-site SIP communication, fully following Cisco best practices.

Question81

A CUCM administrator notices that certain remote endpoints consistently fail to register during peak hours, while internal endpoints register successfully. RTP and SIP signaling to the affected endpoints are intact. The administrator suspects that the Expressway traversal zones are not optimally configured. Which configuration MOST effectively resolves the registration issue?

A) Increase the number of Expressway traversal zone connections
B) Disable SIP normalization on Expressway
C) Use Expressway-E only without Expressway-C
D) Adjust bandwidth allocation for traversal zones

Answer: A

Explanation:

CUCM deployments with Mobile and Remote Access (MRA) rely on Expressway-C and Expressway-E to facilitate secure registration of remote endpoints. During peak hours, if remote endpoints fail to register while internal endpoints work normally, it indicates that the traversal zones connecting Expressway-C and Expressway-E may be reaching their maximum connection limit. Each traversal zone connection allows a specific number of simultaneous SIP sessions, and exceeding this limit results in failed registrations, delayed call setup, or intermittent service issues. Maintaining consistent, high availability registration services for remote endpoints is critical, as delayed or failed registration impacts user productivity, messaging, call routing, and collaboration.

Option A, increasing the number of Expressway traversal zone connections, is the correct solution. By increasing the number of simultaneous allowed connections, remote endpoints can register reliably even during peak usage periods. Cisco best practices for MRA recommend sizing traversal zones based on expected concurrent endpoints, ensuring that resources are sufficient to handle peak traffic. Properly sized traversal zones prevent registration failures, reduce latency, and maintain consistent presence and call capabilities for remote users. This approach ensures high availability and optimal performance for remote endpoints while maintaining secure traversal across enterprise firewalls.

Option B, disabling SIP normalization, addresses header manipulation issues but does not solve capacity limitations within traversal zones. Normalization is generally used to correct signaling inconsistencies between endpoints and CUCM. While disabling it may resolve specific signaling errors, it does not prevent registration failures caused by traversal zone resource exhaustion.

Option C, using Expressway-E only, would remove Expressway-C from the deployment. Expressway-C is essential for internal signaling and device authentication. Removing it would break the MRA topology, resulting in even greater registration failures and security issues. This does not address the root cause of the problem and would reduce overall system functionality.

Option D, adjusting bandwidth allocation for traversal zones, may improve RTP media flow but does not affect SIP session capacity or registration attempts. Registration failures during peak hours are caused by the number of simultaneous SIP connections, not by media bandwidth allocation.

By increasing the number of traversal zone connections, administrators ensure reliable registration for all remote endpoints, maintain optimal performance under peak loads, and adhere to Cisco best practices for MRA deployments. This resolves the root cause of intermittent registration failures while supporting enterprise-wide unified communications efficiently.

Question82

During a CUCM deployment, administrators notice that when remote Jabber clients initiate video calls to internal endpoints, the video often fails to establish or is significantly delayed. Internal-to-internal calls work perfectly. Analysis shows that NAT traversal and ICE candidates are inconsistently applied. Which configuration MOST effectively resolves this issue?

A) Enable persistent XMPP connections between Expressway-C and CUCM
B) Ensure traversal zones allow bidirectional UDP and TCP signaling with proper ICE support
C) Disable media encryption to simplify ICE negotiation
D) Use SIP over TCP exclusively for remote clients

Answer: B

Explanation:

Video calls from remote Jabber clients require correct NAT traversal and signaling negotiation to establish video sessions with internal endpoints. Internal-to-internal calls bypass Expressway traversal, resulting in smooth operation. Remote-to-internal video often fails or experiences latency due to misconfigured traversal zones, inconsistent ICE candidate exchange, or blocked ports. ICE (Interactive Connectivity Establishment) is critical in NAT environments for determining the optimal media path, and it relies on bidirectional UDP and TCP signaling through Expressway traversal zones. Without correct ICE negotiation, remote video sessions may fail or experience significant delays.

Option B, ensuring traversal zones allow bidirectional UDP and TCP signaling with proper ICE support, is the correct solution. This configuration enables Expressway-C and Expressway-E to fully support ICE candidates for media path establishment. Cisco best practices for remote video deployments recommend full ICE support to handle NAT, firewall, and multi-site traversal scenarios. Proper bidirectional signaling ensures remote endpoints can negotiate media paths quickly, reducing video setup latency and preventing call failures. This improves user experience, ensures reliable video communication, and maintains high-quality media sessions across the enterprise network.

Option A, enabling persistent XMPP connections, ensures timely presence updates but does not directly resolve video media path negotiation. While persistent XMPP is beneficial for presence and messaging, video media setup issues are unrelated to XMPP session persistence.

Option C, disabling media encryption, may simplify ICE negotiation slightly but compromises security. Media encryption is required for secure communications, and removing it introduces unacceptable security risks. ICE negotiation can be performed efficiently while maintaining encryption.

Option D, using SIP over TCP exclusively, ensures reliable signaling transport but does not address UDP-dependent ICE negotiation required for media traversal. Many remote endpoints rely on UDP for efficient RTP transport, and restricting SIP to TCP does not resolve media path failures.

Ensuring proper bidirectional UDP and TCP signaling with full ICE support allows remote Jabber clients to establish reliable video sessions with internal endpoints, reduces call setup delays, prevents media failures, and aligns with Cisco best practices for remote video deployments in MRA topologies.

Question83

A distributed CUCM environment uses SIP trunks for inter-site calls. Remote sites occasionally report one-way audio and dropped calls. RTP inspection shows that packets arrive at incorrect IP addresses before the CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Convert SIP trunks to SCCP for inter-site calls
C) Use SIP over TCP to improve transport reliability
D) Disable early media on CUBE

Answer: A

Explanation:

One-way audio and dropped calls in multi-site SIP deployments often stem from improper SDP handling during call setup. When RTP packets are sent to incorrect IP addresses before the CUBE rewrites SDP, media cannot flow correctly, resulting in one-way audio or dropped calls. Early media scenarios exacerbate the issue, as endpoints may begin sending RTP before the CUBE completes SDP manipulation. Accurate SDP negotiation and proper media anchoring are essential for reliable communication between sites.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP ensures SDP is included in the 200 OK response rather than the initial INVITE. This allows CUBE to anchor media, rewrite addresses and ports correctly, and establish proper RTP paths before media transmission begins. Cisco best practices recommend delayed-offer SDP in trunks traversing CUBE when early media could misalign RTP streams. This configuration guarantees reliable audio, prevents one-way media issues, and supports predictable inter-site communications.

Option B, converting SIP trunks to SCCP, does not address the underlying media path issue. SCCP may have different signaling behavior, but the problem here is the timing of SDP negotiation and media path establishment, not protocol incompatibility. Converting trunks adds complexity without resolving one-way audio issues.

Option C, using SIP over TCP, changes signaling transport but does not resolve SDP timing or media misalignment. TCP ensures reliable delivery of signaling messages but does not affect RTP path correctness or address rewriting.

Option D, disabling early media, prevents call progress tones but does not resolve the root cause of RTP packets being sent to incorrect addresses. It treats symptoms rather than addressing SDP timing issues that cause one-way audio.

Enabling delayed-offer SDP ensures correct media anchoring through the CUBE, resolves one-way audio and dropped calls, maintains consistent inter-site SIP communication, and adheres to Cisco best practices for multi-site deployments.

Question84

During CUCM to Unity Connection voicemail integration, users report an initial several-second silence when listening to messages. RTP analysis shows no packet loss or jitter. Which configuration MOST effectively resolves the problem?

Answer: A

Explanation:

Silence at the beginning of voicemail playback is a classic symptom of Voice Activity Detection (VAD) interference. VAD suppresses RTP packets during low-energy audio to conserve bandwidth during live calls. Voicemail messages, especially at the beginning, often contain low-energy audio. VAD interprets this as silence, resulting in several seconds of missing audio. This creates a poor user experience, leading to complaints and decreased productivity, particularly in environments where timely message retrieval is critical.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, is the correct solution. Disabling VAD allows continuous RTP transmission, including low-energy audio segments, eliminating the initial silence. Cisco best practices recommend disabling VAD for voicemail environments to ensure uninterrupted playback from the start of messages. Users immediately hear voicemail content, improving satisfaction and reducing confusion caused by missing audio. The bandwidth impact is negligible because voicemail RTP traffic is minimal compared to live call traffic.

Option B, adjusting MWI extensions, affects lamp signaling only and does not influence RTP or message playback.

Option C, changing the codec to G.722, may improve audio fidelity but does not resolve VAD-induced silence. Codec selection is unrelated to VAD behavior in RTP streams.

Option D, moving the voicemail pilot to a different partition, impacts routing but not media delivery. The initial silence is caused by RTP suppression, not routing issues.

Disabling VAD directly addresses initial playback silence, ensures smooth voicemail experience, and adheres to Cisco best practices for CUCM and Unity Connection integration.

Question85

Remote users experience intermittent one-way audio when calling internal endpoints through a CUBE. RTP is sometimes sent to incorrect addresses before the CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

A) Enable delayed-offer SDP on the CUCM SIP trunk toward CUBE
B) Use SIP over UDP instead of TCP
C) Disable early media on CUBE
D) Enable symmetric RTP on CUBE

Answer: A

Explanation:

One-way audio occurs in multi-site SIP deployments through CUBE when RTP packets are transmitted to incorrect IP addresses before SDP rewriting occurs. This misalignment prevents proper media flow and often causes one-way communication or dropped calls. Proper SDP negotiation and media anchoring are critical in multi-site deployments, especially with early media scenarios or NAT traversal. Without correct SDP handling, remote endpoints cannot establish the proper media path, leading to intermittent one-way audio.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP includes SDP in the 200 OK response instead of the initial INVITE. This gives CUBE sufficient time to rewrite SDP, anchor media, and establish the correct RTP path before packets flow. Cisco best practices for multi-site SIP deployments recommend delayed-offer SDP to avoid one-way audio, ensure predictable media delivery, and maintain consistent call quality.

Option B, using UDP instead of TCP, does not correct SDP timing or address rewriting. Transport protocol change ensures reliable signaling but does not resolve RTP misalignment issues.

Option C, disabling early media, prevents call progress tones but does not fix RTP being sent to incorrect addresses. The underlying problem is SDP timing and media anchoring, not early media signaling.

Option D, enabling symmetric RTP, assists with NAT traversal but does not resolve SDP timing or media misalignment. Symmetric RTP alone cannot fix one-way audio caused by premature RTP transmission before SDP rewriting.

Implementing delayed-offer SDP ensures correct media anchoring through CUBE, eliminates one-way audio, supports reliable multi-site SIP communication, and adheres to Cisco best practices.

Question86

During a CUCM deployment, some remote Jabber clients connected through MRA report delayed call setup and intermittent call drops, while internal endpoints function normally. RTP and signaling analysis show that ICE candidates from remote clients are not consistently applied, causing NAT traversal issues. Which configuration MOST effectively resolves this problem?

A) Enable persistent XMPP connections between Expressway-C and CUCM
B) Ensure traversal zones allow bidirectional UDP and TCP signaling with ICE support
C) Disable SIP normalization on Expressway
D) Use SIP over TCP exclusively for remote clients

Answer: B

Explanation:

Remote Jabber clients rely on Expressway-C and Expressway-E traversal zones to connect securely to CUCM via MRA. In scenarios where call setup is delayed or calls drop intermittently, the root cause is often improper NAT traversal or inconsistent ICE candidate negotiation. ICE (Interactive Connectivity Establishment) allows endpoints behind NAT to discover the optimal path for media. If traversal zones do not support bidirectional UDP and TCP signaling, ICE candidates cannot be exchanged correctly, leading to media failures, one-way audio, or dropped calls.

Option B, ensuring traversal zones allow bidirectional UDP and TCP signaling with proper ICE support, is the correct solution. This configuration allows Expressway to handle signaling and media negotiation effectively, enabling remote clients to establish reliable RTP paths to internal endpoints. Cisco best practices for MRA deployments emphasize correct ICE handling and bidirectional connectivity to support NAT traversal, ensure consistent call setup, and prevent call drops. Proper ICE support allows remote Jabber clients to participate in real-time communications without interruption, improving user experience and reducing support incidents.

Option A, enabling persistent XMPP connections, ensures timely presence updates but does not directly affect media negotiation or ICE candidate handling. While beneficial for presence and messaging, it does not solve the underlying NAT traversal issue affecting call reliability.

Option C, disabling SIP normalization, addresses header inconsistencies but does not resolve ICE or media path issues. Normalization primarily fixes signaling syntax mismatches and does not impact RTP path determination.

Option D, using SIP over TCP exclusively, ensures reliable signaling transport but does not address the need for UDP in media path negotiation for ICE. TCP-only configurations cannot fully support ICE, leading to continued media path failures.

By configuring traversal zones for bidirectional UDP and TCP signaling with ICE support, remote Jabber clients can successfully establish media paths through NAT, call setup times improve, call drops are reduced, and the deployment aligns with Cisco best practices for MRA.

Question87

A CUCM administrator observes that during peak hours, remote endpoints fail to register intermittently. Internal endpoints register without issue. Logs indicate that traversal zones between Expressway-C and Expressway-E are saturated. Which configuration MOST effectively resolves this problem?

A) Increase the number of allowed traversal zone connections
B) Disable traversal zones and use VPN for remote access
C) Reduce SIP timers on CUCM to force faster registration retries
D) Enable persistent XMPP connections between Expressway-C and CUCM

Answer: A

Explanation:

Mobile and Remote Access (MRA) relies on Expressway-C and Expressway-E traversal zones for secure registration and communication of remote endpoints. When traversal zones are saturated during peak usage, CUCM cannot process additional registration requests, resulting in intermittent failures. Internal endpoints bypass traversal zones, explaining why they register normally. Ensuring high availability and reliability for remote endpoints is critical to maintaining consistent collaboration capabilities across the enterprise.

Option A, increasing the number of allowed traversal zone connections, is the correct solution. Cisco best practices recommend sizing traversal zones based on the number of expected simultaneous remote registrations. By allowing more concurrent connections, the system can accommodate peak usage without failures. This approach provides scalability, maintains registration reliability, and ensures consistent availability of services such as presence, instant messaging, and voice/video communication for remote users.

Option B, disabling traversal zones and using VPN, introduces complexity, latency, and potential security issues. MRA is designed to provide secure access without VPN; removing traversal zones negates the purpose of MRA and does not address peak-hour registration saturation.

Option C, reducing SIP timers, forces faster retries but does not solve capacity limitations. Faster retries may increase signaling load, exacerbating congestion in traversal zones, and could create additional registration failures.

Option D, enabling persistent XMPP connections, supports presence updates and messaging but does not increase traversal zone registration capacity. It improves session continuity but does not address peak-hour registration failures.

Increasing traversal zone connections ensures reliable registration, reduces user impact during peak hours, maintains high availability, and aligns with Cisco best practices for MRA deployments, resolving the root cause of intermittent registration failures.

Question88

During a multi-site CUCM deployment with CUBE, remote sites report one-way audio and intermittent call drops during SIP trunk calls. RTP analysis shows packets are sent to incorrect IP addresses before CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Convert SIP trunks to SCCP
C) Use SIP over TCP instead of UDP
D) Disable early media on CUBE

Answer: A

Explanation:

One-way audio and call drops in multi-site SIP deployments often occur when RTP packets are transmitted before CUBE completes SDP rewriting. Early media scenarios exacerbate the issue as endpoints may begin sending RTP prematurely. SDP negotiation and media anchoring are critical to ensure RTP reaches the correct destination. Without proper handling, audio fails to flow to one side, resulting in one-way communication or dropped calls.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP ensures SDP is included in the 200 OK response rather than the initial INVITE. This allows CUBE to anchor media, rewrite IP addresses and ports correctly, and establish accurate RTP paths before media transmission begins. Cisco best practices recommend delayed-offer SDP for trunks traversing CUBE to prevent one-way audio and maintain predictable inter-site communications. Delayed-offer SDP guarantees reliable audio delivery, reduces call drops, and ensures consistent quality of experience across sites.

Option B, converting SIP trunks to SCCP, does not address the SDP timing issue. SCCP may offer different signaling, but the root cause—premature RTP transmission before SDP rewrite—remains unresolved.

Option C, using SIP over TCP, ensures reliable signaling but does not correct RTP path misalignment. Transport reliability alone cannot resolve media misrouting caused by SDP timing issues.

Option D, disabling early media, prevents call progress tones but does not fix RTP being sent to incorrect addresses. Early media is a symptom rather than the root cause.

Enabling delayed-offer SDP allows proper media anchoring, corrects one-way audio, prevents call drops, and ensures reliable multi-site SIP communication, fully adhering to Cisco best practices for deployments using CUBE.

Question89

During voicemail playback from Unity Connection integrated with CUCM, users report several seconds of initial silence. RTP monitoring shows no packet loss or jitter. Which configuration MOST effectively resolves the problem?

Answer: A

Explanation:

Initial silence in voicemail playback is caused by Voice Activity Detection (VAD). VAD suppresses RTP during low-energy audio to conserve bandwidth. Voicemail messages typically start with low-energy audio, which VAD interprets as silence, delaying playback. This negatively affects user experience and can cause confusion or frustration.

Option A, disabling VAD on the SIP trunk, is the correct solution. Disabling VAD ensures continuous RTP transmission, including low-energy audio, eliminating the initial silence. Cisco best practices for CUCM and Unity Connection integration recommend disabling VAD for voicemail deployments to ensure uninterrupted message playback. Users hear the message immediately, enhancing satisfaction and usability. Bandwidth impact is minimal because voicemail RTP traffic is small compared to live call traffic.

Option B, adjusting MWI extensions, affects lamp signaling only and does not influence RTP or playback.

Option C, changing the codec to G.722, may improve audio quality but does not resolve VAD-induced silence.

Option D, moving the voicemail pilot to a different partition, affects call routing but does not influence media delivery.

Disabling VAD directly addresses initial silence, ensures smooth voicemail playback, and follows Cisco best practices for CUCM and Unity Connection deployments.

Question90

Remote users report intermittent one-way audio when calling internal endpoints through CUBE. RTP is sometimes sent to incorrect addresses before CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

A) Enable delayed-offer SDP on the CUCM SIP trunk toward CUBE
B) Use SIP over UDP instead of TCP
C) Disable early media on CUBE
D) Enable symmetric RTP on CUBE

Answer: A

Explanation:

One-way audio occurs when RTP packets are sent to incorrect addresses before SDP rewriting. Early media or NAT traversal scenarios exacerbate this problem. Correct SDP negotiation and media anchoring are critical for consistent RTP delivery. Without proper handling, one side may not receive audio, leading to dropped or incomplete calls.

Option A, enabling delayed-offer SDP, is the correct solution. SDP is included in the 200 OK response rather than the initial INVITE, allowing CUBE to anchor media, rewrite IP addresses and ports, and establish proper RTP paths before media flow. Cisco best practices for multi-site SIP deployments using CUBE recommend delayed-offer SDP to prevent one-way audio, ensure reliable media delivery, and maintain call quality.

Option B, using SIP over UDP, does not address SDP timing or address rewriting issues.

Option C, disabling early media, prevents call progress tones but does not fix RTP misrouting.

Option D, enabling symmetric RTP, assists with NAT traversal but does not resolve SDP timing issues causing one-way audio.

Delayed-offer SDP ensures proper media anchoring, eliminates one-way audio, and provides reliable communication in multi-site deployments, aligning with Cisco best practices.

One-way audio in multi-site Cisco Unified Communications deployments is a frequent challenge, particularly in environments where Cisco Unified Border Element (CUBE) is responsible for media anchoring. The problem arises when RTP packets are sent to endpoints before SDP negotiation has been properly completed and rewritten by CUBE. The Session Description Protocol (SDP) contains critical information, including IP addresses, port numbers, and codec selections for media streams. If SDP is included in the initial SIP INVITE and media begins flowing immediately, CUBE may not yet have modified the addresses to route the RTP correctly. This misalignment results in RTP arriving at an unintended location, causing one-way audio where only one side hears the call.

Early media, including ringback tones, announcements, or progress tones, can exacerbate this problem. While early media is intended to improve the caller experience by providing immediate audio feedback, it introduces the possibility that RTP will flow before CUBE can anchor the media path. In multi-site deployments, where calls may traverse WAN links, firewalls, or NAT devices, incorrect media delivery can occur even more frequently. Without proper media anchoring, calls are prone to one-way audio, incomplete audio, or dropped RTP packets, impacting both the quality of the call and the user experience.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the recommended solution. Delayed-offer SDP postpones the transmission of SDP from the initial INVITE and instead includes it in the 200 OK response. This approach ensures that CUBE receives the signaling information before any media is transmitted, allowing it to anchor the call properly and rewrite IP addresses and port numbers for RTP streams. By establishing the media path after SDP is offered in the 200 OK, RTP is correctly directed to the endpoints, eliminating the possibility of one-way audio. Cisco best practices for multi-site SIP deployments emphasize the use of delayed-offer SDP for trunks traversing CUBE to ensure correct media path alignment and reliable communication.

The benefits of delayed-offer SDP extend beyond resolving one-way audio. It provides consistent media flows, simplifies troubleshooting, and enhances interoperability with other SIP devices or service providers. In multi-site deployments, calls may traverse multiple network segments, firewalls, or NAT devices. Correctly anchored media ensures that RTP is sent to the proper endpoints across all segments, improving call quality and reliability. Administrators benefit from predictable media behavior, allowing them to analyze packet captures, monitor media streams, and verify call quality with confidence.

Option B, using SIP over UDP instead of TCP, changes the transport protocol but does not address the root cause of one-way audio. UDP and TCP differ in terms of connection reliability and delivery guarantees, but the issue in this scenario is the timing of SDP and the misalignment of RTP. Changing transport protocols does not prevent early RTP from reaching incorrect endpoints, and therefore does not resolve one-way audio.

Option C, disabling early media on CUBE, prevents the transmission of pre-answer tones and announcements. While this may reduce the perception of premature audio, it does not correct the underlying SDP timing problem. RTP could still be sent to incorrect addresses after call setup, leaving one-way audio unresolved. Disabling early media only addresses a symptom of the problem rather than the root cause and may reduce the quality of user experience by removing expected call progress tones.

Option D, enabling symmetric RTP on CUBE, is primarily used to address NAT traversal or asymmetrical routing issues. Symmetric RTP ensures that RTP is sent back to the source IP and port from which it was received, which can be helpful in networks where endpoints are behind NAT devices. However, symmetric RTP does not solve the early SDP issue that causes RTP to be sent to the wrong destination. While it can assist in certain network topologies, it does not replace the need for proper media anchoring and correct SDP timing.

Implementing delayed-offer SDP ensures that media is anchored at the correct point in the network and that all participants receive audio reliably. In multi-site deployments, delayed-offer SDP maintains a predictable sequence of signaling and media, facilitating easier troubleshooting, performance monitoring, and operational management. It aligns with Cisco’s recommended architecture for SIP trunking and CUBE deployments, providing a robust, standards-aligned solution for preventing one-way audio and ensuring high-quality voice communications.

Overall, delayed-offer SDP directly addresses the underlying cause of one-way audio by ensuring that SDP is transmitted only after signaling has been fully processed, allowing CUBE to rewrite media addresses and establish the correct RTP path. Other options, including changing the transport protocol, disabling early media, or enabling symmetric RTP, do not resolve the root cause and cannot reliably eliminate one-way audio in multi-site deployments. By enabling delayed-offer SDP, organizations ensure correct media anchoring, predictable RTP flows, and reliable communication across all sites, fully adhering to Cisco best practices for SIP trunk deployments using CUBE.

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