Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 13 Q181-195

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Question181

During deployment of CUCM clusters with inter-site SIP trunks, users report inconsistent call quality and one-way audio. Analysis indicates that RTP packets are sometimes delivered to endpoints before SDP information is properly rewritten by CUBE. 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 exclusively
D) Disable early media on CUBE

Answer: A

Explanation:

In multi-cluster CUCM deployments with CUBE mediating SIP trunks, one-way audio and inconsistent call quality are often caused by RTP being sent before SDP rewriting occurs. SDP carries critical information, such as IP addresses, ports, and codec capabilities, which are necessary for establishing correct media streams between endpoints. When RTP is transmitted prematurely, media may be misrouted, resulting in one-way audio or dropped calls.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, addresses this issue directly. Delayed-offer SDP defers the inclusion of SDP in the initial INVITE message until the 200 OK response, allowing CUBE to anchor media and rewrite SDP accurately. This ensures RTP flows are correctly directed to endpoints, stabilizing inter-cluster communication. Cisco recommends delayed-offer SDP for multi-site and hybrid deployments because it synchronizes signaling and media paths, reduces troubleshooting complexity, and improves call reliability. Implementing this configuration minimizes call drops, ensures predictable audio flow, and enhances the user experience by guaranteeing that RTP reaches the correct destination consistently.

Option C, using SIP over TCP exclusively, ensures reliable signaling message delivery but does not affect the timing of SDP relative to RTP transmission. TCP guarantees that messages arrive intact, but it does not anchor media or prevent misrouted RTP.

Option D, disabling early media on CUBE, influences call progress tones and ringback behavior but does not anchor the media path or prevent RTP from being sent prematurely. Early media adjustments do not correct the underlying SDP and RTP synchronization issue.

Enabling delayed-offer SDP ensures accurate media path establishment, predictable RTP flows, and consistent call quality. This configuration aligns with Cisco best practices, simplifies troubleshooting, and provides a reliable communication experience across multiple CUCM clusters. By properly synchronizing signaling and media, enterprises can maintain high-quality voice communication even in complex multi-site deployments, reducing operational overhead and increasing end-user satisfaction.

Question182

Remote Jabber clients behind NAT are experiencing intermittent registration failures and one-way audio when connecting through CUCM Expressway traversal zones. ICE candidate negotiation fails under high load, and traversal zones are near maximum resource utilization. Which configuration MOST effectively resolves this issue?

A) Increase traversal zone resources and enable full bidirectional ICE support
B) Require VPN connectivity for remote clients
C) Reduce CUCM SIP registration timers
D) Enable persistent XMPP connections between Expressway-C and CUCM

Answer: A

Explanation:

Registration failures and one-way audio in remote Jabber clients often stem from insufficient traversal zone resources and failed ICE (Interactive Connectivity Establishment) negotiation. ICE allows endpoints to determine optimal IP addresses and ports for RTP, facilitating NAT traversal and ensuring successful media path establishment. Traversal zones on CUCM Expressway mediate secure connections between remote clients and CUCM. When traversal zones are resource-constrained, ICE negotiation can fail, preventing RTP paths from being established and leading to registration failures and one-way audio.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses the root cause. By allocating additional CPU and memory to traversal zones, the system can handle peak loads, preventing ICE negotiation failures. Bidirectional ICE ensures proper candidate exchange between Jabber clients and Expressway servers, allowing RTP streams to traverse NAT and firewall devices correctly. Cisco best practices recommend sizing traversal zones based on the expected number of remote clients and enabling full bidirectional ICE for reliable registration and media path establishment. Continuous monitoring allows proactive resource management, preventing failures during peak usage periods and maintaining consistent call quality.

Option B, requiring VPN connectivity, may bypass some NAT issues but introduces additional complexity, latency, and potential points of failure. VPN does not address traversal zone resource constraints or ICE negotiation failures and is less suitable for large-scale deployments.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not resolve the root cause. Frequent retries under constrained resources can exacerbate congestion, leading to further registration failures.

Option D, enabling persistent XMPP connections, improves messaging and presence reliability but does not affect ICE negotiation or traversal zone resource availability. Persistent XMPP connections alone cannot resolve registration failures or one-way audio caused by insufficient resources.

Increasing traversal zone resources and enabling bidirectional ICE ensures reliable registration, proper RTP path establishment, and consistent voice quality. This solution aligns with Cisco best practices, supports scalable remote collaboration, reduces support incidents, and guarantees predictable performance for all remote users. Proper ICE configuration and resource allocation provide seamless connectivity and a consistent communication experience.

Question183

Unity Connection voicemail users report a delay of several seconds before message playback begins, although network metrics indicate 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:

Delays at the start of voicemail message playback are frequently caused by Voice Activity Detection (VAD), which suppresses RTP transmission during periods of low audio energy to conserve bandwidth in live calls. Voicemail messages often start with low-volume audio or silence, which VAD interprets as inactivity, delaying RTP transmission and creating an apparent delay in playback.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, resolves this issue by allowing continuous RTP transmission, including low-energy segments. Cisco best practices recommend disabling VAD for voicemail services to maintain message integrity and ensure consistent playback. Although disabling VAD slightly increases bandwidth consumption, it guarantees immediate message playback, improves the user experience, and reduces support calls related to voicemail playback delays. Continuous RTP flow ensures predictable message delivery and preserves the fidelity of audio recordings.

Option B, adjusting MWI extensions, affects visual notifications for new messages but does not impact RTP flow or playback timing. Changes to MWI do not resolve initial playback delays.

Option C, changing the codec to G.722, enhances audio quality but does not prevent VAD-induced delays. Codec selection affects sound fidelity but not RTP transmission timing.

Option D, moving the voicemail pilot to a different partition, changes call routing but does not influence media flow or VAD behavior. Partition modifications do not address initial playback delays.

Disabling VAD ensures immediate RTP transmission, eliminating the initial silence and improving voicemail usability. This configuration is consistent with Cisco best practices, maintains message integrity, and provides predictable behavior across all endpoints. Proper configuration guarantees reliable voicemail service without impacting overall call quality, enhancing end-user satisfaction and reducing operational support requirements.

Question184

CUCM inter-site SIP trunk calls mediated through CUBE experience sporadic one-way audio and call drops. RTP is sometimes transmitted to endpoints before SDP rewriting. 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 exclusively
D) Disable early media on CUBE

Answer: A

Explanation:

One-way audio and call drops in inter-site SIP trunk calls are commonly caused by RTP transmission before SDP rewriting by CUBE. SDP communicates essential media parameters, including IP addresses, ports, and codec choices. Premature RTP transmission can cause misrouted media, leading to audio problems and call failures.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, resolves this issue. Delayed-offer SDP postpones SDP inclusion until the 200 OK response, allowing CUBE to anchor media and rewrite SDP accurately. Cisco best practices recommend this approach in multi-site or hybrid deployments because it ensures correct synchronization between signaling and media, stabilizes call quality, and simplifies troubleshooting. Implementing delayed-offer SDP ensures predictable RTP flow, reduces call drops, and guarantees consistent audio quality across enterprise deployments.

Option B, converting SIP trunks to SCCP, changes signaling protocols but does not prevent early RTP transmission. While SCCP may improve endpoint compatibility, it does not solve misrouted RTP issues.

Option C, using SIP over TCP, ensures reliable signaling but does not affect SDP timing relative to RTP. TCP guarantees message delivery but cannot prevent misrouted RTP flows.

Option D, disabling early media on CUBE, affects call progress tones and ringback behavior but does not anchor RTP flows. Early media changes only signaling behavior, leaving the underlying media path problem unresolved.

Enabling delayed-offer SDP ensures proper media anchoring, correct RTP flow, and stable call quality. This configuration reduces operational issues, follows Cisco best practices, and provides a reliable enterprise voice environment.

Question185

Remote Jabber clients behind NAT encounter frequent registration failures and one-way audio when connecting through CUCM Expressway traversal zones. ICE candidate negotiation fails under high load, and traversal zones are near resource limits. Which configuration MOST effectively resolves the issue?

Answer: A

Explanation:

Remote Jabber registration failures and one-way audio are often caused by traversal zone resource exhaustion and failed ICE negotiation. ICE enables endpoints to determine optimal IP addresses and ports for RTP streams to traverse NAT and firewalls. CUCM Expressway traversal zones mediate secure client connections to CUCM, ensuring registration success and RTP path establishment. Resource-constrained traversal zones prevent successful ICE negotiation, causing registration failures and misrouted RTP.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses the root cause. Allocating additional CPU and memory ensures traversal zones can handle peak loads, preventing ICE negotiation failures. Bidirectional ICE guarantees proper candidate exchange between Jabber clients and Expressway servers, allowing RTP streams to traverse NAT and firewalls. Cisco best practices recommend sizing traversal zones based on expected remote client populations and enabling bidirectional ICE to maintain registration reliability and media quality. Monitoring resource usage allows proactive scaling, ensuring seamless performance during peak periods.

Option B, requiring VPN connectivity, may bypass some NAT issues but introduces additional complexity, latency, and points of failure. VPN does not solve resource constraints or ICE failures.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not resolve traversal zone limitations. Excessive retries can worsen congestion without improving registration success.

Option D, enabling persistent XMPP connections, improves messaging reliability but does not influence ICE negotiation or traversal zone performance. Persistent XMPP alone cannot fix registration failures or one-way audio due to insufficient resources.

Increasing traversal zone resources and enabling bidirectional ICE ensures reliable registration, proper RTP path establishment, and consistent voice quality. This aligns with Cisco best practices, supports scalable remote collaboration, reduces support incidents, and ensures predictable performance. Proper ICE configuration and resource allocation guarantee seamless connectivity and a consistent communication experience.

Question186

A large enterprise deploys Cisco DNA Center to automate network configuration and monitor device health. The network team notices that several devices show inconsistent telemetry data, and certain configuration changes fail to apply during scheduled pushes. Analysis shows that some devices are running older IOS XE versions incompatible with the latest DNA Center automation templates. Which action MOST effectively resolves these issues?

A) Upgrade all devices to the supported IOS XE versions and reapply templates
B) Disable telemetry collection for devices with older IOS XE versions
C) Remove devices from DNA Center and manage them manually
D) Use SNMP polling instead of telemetry for all devices

Answer: A

Explanation:

In Cisco DNA Center deployments, automation and telemetry rely heavily on device compatibility. DNA Center leverages modern protocols like NETCONF, REST APIs, and streaming telemetry to manage configurations, monitor device health, and provide advanced analytics. When devices run older IOS XE versions, they may lack full support for these protocols, causing telemetry inconsistencies and failed automation pushes. Upgrading all devices to supported IOS XE versions ensures that automation templates function as intended, telemetry data is accurate, and network health monitoring operates reliably. Option A directly addresses the root cause by aligning device software with DNA Center’s functional requirements. Older versions might support partial SNMP or CLI operations, but full automation features, including assurance, configuration compliance, and policy enforcement, require updated software. Without compatible software, features like software image management, configuration drift detection, and template-based deployments cannot be fully leveraged. This approach ensures network consistency, reduces operational errors, and enables scalable automation across the enterprise. It also simplifies troubleshooting, as devices now support standardized telemetry streams, enabling more accurate analytics and proactive alerts.

Option B, disabling telemetry collection for older devices, merely avoids symptom reporting rather than addressing the underlying incompatibility. While this might reduce error messages, it removes visibility into device performance and hinders proactive network management. Critical events or configuration drifts could go unnoticed, increasing operational risk. Telemetry provides granular data on interface statistics, CPU/memory usage, and anomaly detection, which is essential for performance assurance and security monitoring. Disabling it compromises these functions, leaving the network vulnerable to unnoticed degradation or misconfigurations.

Option C, removing devices from DNA Center and managing them manually, bypasses the automation and assurance capabilities entirely. Manual management is error-prone, less scalable, and increases operational overhead, especially in large enterprise environments. It prevents centralized policy enforcement, template-based configuration consistency, and assurance-driven analytics. Over time, this could lead to configuration drift, inconsistent network behavior, and difficulties in maintaining compliance with corporate or regulatory policies. Moreover, manual intervention does not provide real-time telemetry or performance metrics, significantly limiting the network team’s ability to proactively manage and troubleshoot devices.

Option D, using SNMP polling instead of telemetry, provides limited historical and snapshot data but cannot substitute for streaming telemetry’s real-time insights. SNMP lacks granularity, has polling delays, and may not expose all features needed for DNA Center automation. SNMP can give basic interface statistics or CPU/memory usage at intervals, but it cannot provide the continuous telemetry streams that DNA Center leverages for assurance, anomaly detection, and predictive analytics. Automation workflows may still fail if device software lacks support for NETCONF or RESTCONF APIs. This approach partially mitigates visibility issues but does not resolve configuration push failures or the lack of compatibility with advanced automation templates.

Upgrading devices ensures alignment with DNA Center capabilities, resolves telemetry inconsistencies, allows template-based automation to succeed, and provides full visibility into network health, performance, and compliance. It is a strategic, long-term solution that maximizes the benefits of Cisco DNA Center while reducing operational risk and manual intervention.

Question187

A company deploys Cisco SD-WAN to multiple branch sites. Some users report intermittent voice and video quality issues. Troubleshooting shows that packet loss occurs only on certain Internet paths during peak usage, and SLA metrics indicate latency and jitter exceed recommended thresholds. Which configuration MOST effectively resolves this issue?

A) Configure application-aware routing with SLA-based path selection
B) Force all traffic through a centralized data center tunnel
C) Increase tunnel MTU to maximum values
D) Disable dynamic path selection and use static routing

Answer: A

Explanation:

Cisco SD-WAN is designed to optimize application performance across multiple WAN links using intelligent path selection. When packet loss and jitter occur only on certain paths, and SLA metrics are exceeded, this indicates congestion, suboptimal path selection, or inefficient traffic steering. Option A, configuring application-aware routing with SLA-based path selection, leverages real-time monitoring to dynamically steer traffic based on latency, jitter, packet loss, and application type. Critical applications such as voice and video can be routed through optimal paths, while less critical traffic uses lower-priority links. Application-aware routing ensures that VoIP and video conferencing traffic meet performance thresholds, minimizing dropped calls and poor video quality. By continuously monitoring link performance, SD-WAN can reroute traffic when degradation is detected, improving user experience and operational reliability.

Option B, forcing all traffic through a centralized data center tunnel, may exacerbate the problem by introducing additional latency, consuming WAN bandwidth, and concentrating traffic, which can worsen packet loss and jitter during peak periods. Centralized tunnels reduce the efficiency and resilience of SD-WAN, as local Internet breakout for SaaS applications and latency-sensitive traffic is not leveraged.

Option C, increasing tunnel MTU, addresses potential fragmentation but does not resolve packet loss, latency, or jitter issues caused by congestion or suboptimal path selection. MTU adjustments alone cannot optimize traffic flows or guarantee SLA adherence.

Option D, disabling dynamic path selection and using static routing, eliminates SD-WAN intelligence. Static routing cannot adapt to real-time network conditions, leading to persistent performance issues on degraded links. Without dynamic path selection, voice and video traffic may traverse suboptimal routes, resulting in poor QoS and reduced application performance.

Application-aware routing with SLA-based path selection maximizes the benefits of SD-WAN by combining real-time monitoring, intelligent traffic steering, and QoS prioritization. This approach improves voice and video quality, reduces packet loss, and enhances overall network reliability, ensuring that enterprise-grade performance is maintained even during peak network usage periods.

Question188

An enterprise plans to deploy Cisco Webex Edge for Devices to integrate with its existing CUCM deployment. Remote users report failures in video meetings, and analysis indicates NAT traversal and ICE candidate failures. Which configuration MOST effectively resolves the issue?

A) Open UDP ports for TURN/STUN traffic on the firewall
B) Force all video traffic through VPN tunnels
C) Disable ICE on Webex Edge devices
D) Increase TCP timeout values for signaling

Answer: A

Explanation:

Webex Edge for Devices relies on TURN/STUN protocols to traverse NAT and firewall devices, establishing media paths for video and audio. Intermittent meeting failures and ICE candidate errors indicate that UDP traffic may be blocked or dropped by firewalls. Option A, opening UDP ports for TURN/STUN traffic, directly addresses the root cause by allowing media negotiation packets to pass unobstructed. TURN provides a relay mechanism when direct peer-to-peer connections fail, while STUN helps devices discover public IP addresses behind NAT. Correct firewall configuration ensures ICE can successfully exchange candidates, negotiate optimal media paths, and maintain high-quality video sessions for remote users.

Option B, forcing all video traffic through VPN tunnels, adds unnecessary latency, increases complexity, and may not address blocked UDP traffic effectively. VPN tunnels also concentrate traffic and may become a bottleneck during high usage.

Option C, disabling ICE, would break the NAT traversal mechanism entirely. ICE is fundamental for discovering viable media paths between endpoints and relays; disabling it results in call failures and poor media performance.

Option D, increasing TCP timeout values, affects signaling persistence but does not facilitate UDP media traffic or NAT traversal. It does not resolve ICE or TURN/STUN negotiation issues.

Opening the required UDP ports ensures ICE negotiation and TURN relays function properly, allowing remote users to reliably connect to Webex meetings. It aligns with Cisco best practices, supports scalable deployments, and provides predictable, high-quality collaboration experiences for users behind NAT or firewalls.

Question189

A network engineer is troubleshooting inter-cluster CUCM calls that experience one-way audio and call drops. SIP traces indicate that some endpoints send early media before SDP has been rewritten by the SIP trunk. Which configuration MOST effectively resolves the issue?

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

Answer: A

Explanation:

One-way audio and call drops in inter-cluster CUCM deployments are often caused by SDP timing mismatches. SDP communicates media parameters such as IP addresses, ports, and codecs. When early media is sent before SDP is rewritten by a SIP trunk or CUBE, RTP arrives prematurely, causing audio issues. Option A, enabling delayed-offer SDP, ensures that SDP is sent after the 200 OK response, allowing proper SDP rewriting and media anchoring. This aligns signaling and media negotiation, preventing one-way audio, call drops, and codec mismatches. Delayed-offer SDP is a best practice in multi-cluster environments, ensuring consistent media behavior across clusters and reducing troubleshooting complexity.

Option B, converting SIP endpoints to SCCP, changes signaling protocol but does not resolve SDP timing issues. Audio problems may persist, and interoperability with other SIP clusters remains a concern.

Option C, using SIP over TCP, ensures reliable message delivery but does not control early media timing. TCP guarantees signaling integrity but cannot prevent premature RTP delivery relative to SDP.

Option D, disabling early media globally, affects call progress tones but does not resolve SDP synchronization issues. Early media control alone does not anchor media correctly or prevent call drops caused by SDP mismatches.

Delayed-offer SDP provides predictable, reliable call behavior, anchors media flows correctly, and supports multi-cluster enterprise voice deployments. This configuration reduces one-way audio, prevents dropped calls, simplifies troubleshooting, and ensures consistent user experience across all endpoints. Proper SDP handling also ensures correct codec negotiation, media path establishment, and adherence to enterprise voice quality standards.

Question190

Remote Jabber clients behind NAT fail to establish calls with CUCM endpoints during peak usage. ICE negotiation fails intermittently, and RTP paths are not established. Which configuration MOST effectively resolves this issue?

A) Increase Expressway traversal zone resources and enable full bidirectional ICE support
B) Require all remote users to connect via VPN
C) Reduce CUCM SIP registration timers to force retries
D) Enable persistent XMPP connections between Expressway-C and CUCM

Answer: A

Explanation:

Jabber clients behind NAT depend on Expressway traversal zones to mediate signaling and media paths. ICE negotiation failure and RTP path establishment issues are often caused by resource constraints or insufficient ICE support. Option A, increasing Expressway traversal zone resources and enabling full bidirectional ICE support, addresses both root causes. Enhanced resources ensure traversal zones can handle high volumes of simultaneous registrations and calls, while bidirectional ICE guarantees proper candidate exchange and media path negotiation. This configuration aligns with Cisco best practices, enabling scalable remote collaboration, maintaining consistent call quality, and preventing registration or call failures. Properly configured ICE ensures optimal media routing, NAT traversal, and end-to-end audio/video quality.

Option B, requiring VPN, adds latency, complexity, and overhead, without addressing traversal zone capacity or ICE negotiation directly. VPN may mask NAT issues temporarily but does not scale well for large remote user populations.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not resolve ICE or resource limitations. It can exacerbate congestion and resource consumption during peak usage, worsening performance issues.

Option D, enabling persistent XMPP connections, improves messaging reliability but does not impact ICE negotiation or RTP path establishment. While it helps chat and presence services, it does not solve real-time media connectivity problems.

Increasing Expressway traversal zone resources and enabling bidirectional ICE ensures reliable Jabber registration, proper RTP path establishment, and predictable call performance. It supports large-scale remote deployments, reduces operational issues, and maintains high-quality audio and video communication for users behind NAT, fully adhering to Cisco best practices.

Question191

An enterprise has deployed a multi-site CUCM environment with SIP trunks routed through a CUBE. End users report sporadic one-way audio during peak hours, and analysis shows RTP packets are sometimes sent to endpoints before SDP rewriting occurs. 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 exclusively
D) Disable early media on CUBE

Answer: A

Explanation:

In multi-site CUCM deployments with CUBE mediating SIP trunks, one-way audio and sporadic call drops are typically caused by RTP being transmitted to endpoints before CUBE has rewritten the SDP. SDP, the Session Description Protocol, conveys essential parameters for establishing a media path, such as IP addresses, ports, and codec information. When RTP arrives prematurely, endpoints may not be ready to receive media, resulting in one-way audio, dropped calls, or poor voice quality.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, resolves this problem effectively. Delayed-offer SDP defers the inclusion of SDP until the 200 OK response from the destination endpoint is received. This ensures that CUBE anchors the media and can rewrite the SDP accurately, inserting the correct IP addresses, ports, and codecs for proper RTP delivery. Cisco best practices recommend delayed-offer SDP in multi-site or hybrid deployments because it ensures correct synchronization between signaling and media, stabilizes call quality, and simplifies troubleshooting. Implementing delayed-offer SDP guarantees predictable RTP flow, reduces call drops, and provides consistent high-quality audio across the enterprise. By deferring SDP, the media path is accurately anchored and coordinated with signaling, eliminating timing mismatches that lead to audio issues.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not resolve the underlying issue of premature RTP transmission. SCCP may improve endpoint compatibility or support additional features, but RTP timing remains uncoordinated if SDP is sent before media anchoring. Thus, one-way audio and call instability persist, and administrators may still face challenges troubleshooting media path issues.

Option C, using SIP over TCP exclusively, ensures that signaling messages are delivered reliably and in order, but does not influence the timing of RTP relative to SDP. TCP guarantees the integrity of SIP messages but cannot prevent RTP from being sent prematurely. Without delayed-offer SDP, misrouted RTP will continue, causing call quality issues and user dissatisfaction.

Option D, disabling early media on CUBE, affects ringback and call progress tones but does not anchor RTP or adjust SDP timing. Early media configuration changes signaling behavior, not the underlying media path establishment, and therefore cannot resolve one-way audio caused by premature RTP transmission.

By enabling delayed-offer SDP, enterprises ensure accurate media path establishment, predictable RTP flows, and stable call quality across all inter-site calls. This configuration aligns with Cisco best practices, reduces troubleshooting overhead, and enhances end-user satisfaction by providing reliable and consistent audio. Delayed-offer SDP also supports interoperability with diverse endpoints and enables scalable, high-quality enterprise voice deployments. Proper SDP management ensures that complex multi-site topologies function efficiently, mitigating operational risks associated with inter-cluster communication.

Question192

Remote Jabber clients behind NAT and firewalls are experiencing frequent registration failures and one-way audio when connecting through CUCM Expressway traversal zones. ICE candidate negotiation fails intermittently during high usage periods. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Remote Jabber registration failures and one-way audio are often caused by ICE (Interactive Connectivity Establishment) negotiation failures combined with resource limitations on Expressway traversal zones. ICE enables endpoints to determine optimal IP addresses and ports for RTP traversal through NAT and firewalls. Expressway traversal zones act as intermediaries, anchoring media and facilitating secure connections between remote clients and CUCM. When traversal zones are operating near maximum capacity, ICE candidate exchange can fail, preventing proper RTP path establishment and resulting in registration failures or audio issues.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses these problems at the root cause. Allocating additional CPU, memory, and other system resources ensures that traversal zones can handle peak loads without dropping ICE candidate exchanges. Full bidirectional ICE support guarantees that candidates are correctly exchanged between clients and Expressway servers, allowing RTP streams to traverse NAT and firewall devices reliably. Cisco best practices recommend sizing traversal zones according to anticipated remote client populations and enabling bidirectional ICE for consistent registration success and high-quality media paths. Continuous monitoring of traversal zone resource utilization allows proactive scaling, preventing failures during periods of high client activity and ensuring stable media performance.

Option B, requiring VPN connectivity, may bypass some NAT traversal issues but introduces operational complexity, additional latency, and potential points of failure. VPN does not address traversal zone resource limitations or ICE negotiation failures, making it an incomplete solution for large-scale deployments.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not resolve the underlying problem of resource exhaustion. Frequent retries under constrained resources can exacerbate congestion, further impacting registration reliability and user experience.

Option D, enabling persistent XMPP connections, improves messaging and presence reliability but does not affect ICE negotiation or traversal zone resource availability. Persistent XMPP alone cannot prevent registration failures or one-way audio caused by insufficient resources.

Increasing traversal zone resources and enabling bidirectional ICE ensures that remote Jabber clients can register reliably, establish proper RTP paths, and experience consistent voice quality. This approach follows Cisco best practices, supports scalable remote collaboration, reduces support incidents, and provides predictable performance under high-demand scenarios. Correct ICE configuration and resource allocation guarantee seamless connectivity, reliable registration, and optimal media quality for remote users.

Question193

Unity Connection voicemail users report a noticeable delay before message playback begins, despite network conditions showing no packet loss or jitter. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Delays at the start of voicemail playback are commonly caused by Voice Activity Detection (VAD), which suppresses RTP transmission during periods of low audio energy to conserve bandwidth. Voicemail messages often start with silence or low-volume audio, which VAD interprets as inactivity, delaying RTP transmission and causing the perceived delay.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, resolves this issue by allowing continuous RTP transmission, including low-energy segments. Cisco best practices recommend disabling VAD for voicemail services to maintain message integrity and consistent playback behavior. While bandwidth usage slightly increases, this configuration ensures immediate playback, improves user experience, and reduces support calls related to voicemail delays. Continuous RTP flow guarantees predictable message delivery and preserves the fidelity of recorded audio across endpoints.

Option B, adjusting MWI extensions, affects visual notifications for new messages but does not impact RTP flow or playback timing. MWI configuration changes are unrelated to the media delay problem.

Option C, changing the codec to G.722, improves audio quality but does not prevent VAD from suppressing initial RTP packets. Codec adjustments affect fidelity but not playback timing.

Option D, moving the voicemail pilot to a different partition, modifies routing but does not influence RTP flow or VAD behavior. Partition adjustments do not resolve playback delay issues.

Disabling VAD ensures continuous RTP transmission, eliminating initial silence and improving voicemail usability. This approach adheres to Cisco best practices, maintains message integrity, and provides consistent behavior across endpoints, enhancing user satisfaction and operational efficiency. Proper VAD configuration guarantees reliable voicemail service without impacting overall call quality, preventing user frustration and reducing support workload.

Question194

Inter-site SIP trunk calls mediated through CUBE experience intermittent one-way audio and call drops. RTP packets sometimes arrive at endpoints before SDP is properly rewritten. 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 exclusively
D) Disable early media on CUBE

Answer: A

Explanation:

One-way audio and call drops in multi-site SIP trunk deployments are commonly caused by RTP being transmitted to endpoints before CUBE anchors media and rewrites SDP. SDP conveys critical media parameters, including IP addresses, port numbers, and codecs. Premature RTP transmission can result in misrouted packets, causing audio issues and call instability.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, postpones SDP inclusion until the 200 OK response, allowing CUBE to rewrite the SDP accurately and anchor the media path. Cisco best practices recommend delayed-offer SDP for multi-site or hybrid environments to synchronize signaling and media, stabilize call quality, and reduce troubleshooting complexity. Implementing delayed-offer SDP ensures consistent RTP flow, minimizes call drops, and maintains high-quality audio across inter-site calls. This configuration supports interoperability with diverse endpoints, ensures proper media anchoring, and enhances overall operational reliability.

Option B, converting SIP trunks to SCCP, changes signaling protocols but does not prevent premature RTP transmission. One-way audio and call drops may persist, requiring continued troubleshooting of media flows.

Option C, using SIP over TCP, ensures reliable signaling delivery but does not affect RTP timing relative to SDP. TCP alone cannot prevent misrouted RTP.

Option D, disabling early media on CUBE, affects ringback behavior but does not anchor RTP flows or correct SDP timing. Early media adjustments alone cannot resolve the underlying media path issue.

Enabling delayed-offer SDP guarantees proper media anchoring, stable RTP flow, and reliable call quality. This configuration aligns with Cisco best practices, reduces operational complexity, and ensures predictable audio performance across inter-site deployments. Proper SDP management is critical in maintaining voice reliability in enterprise networks with complex topologies.

Question195

Remote Jabber clients behind NAT experience registration failures and one-way audio when connecting through CUCM Expressway traversal zones under peak load. ICE candidate negotiation occasionally fails. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Remote Jabber registration failures and one-way audio are typically caused by insufficient traversal zone resources and failed ICE negotiation. ICE allows endpoints to determine optimal IP addresses and ports for RTP traversal through NAT and firewalls. CUCM Expressway traversal zones mediate secure client connections to CUCM. When traversal zones operate near capacity, ICE candidate exchange may fail, preventing proper RTP path establishment and causing registration failures or audio issues.

Option A, increasing traversal zone resources and enabling bidirectional ICE, resolves these issues by ensuring the traversal zone can handle peak client loads without failure. Allocating additional CPU and memory guarantees successful ICE candidate exchange, allowing RTP streams to traverse NAT and firewall devices reliably. Bidirectional ICE ensures proper candidate negotiation between clients and Expressway servers. Cisco best practices recommend sizing traversal zones based on expected remote clients and enabling bidirectional ICE to maintain registration reliability and media quality. Monitoring resource utilization proactively prevents failures during high-demand periods.

Option B, requiring VPN connectivity, may bypass NAT-related issues but introduces complexity, latency, and additional points of failure. VPN does not resolve traversal zone resource limitations or ICE negotiation failures.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not address the underlying resource constraint. Excessive retries under constrained resources may worsen congestion.

Option D, enabling persistent XMPP connections, improves messaging reliability but does not affect ICE negotiation or traversal zone performance. Persistent XMPP alone cannot prevent registration failures or one-way audio caused by resource limitations.

Increasing traversal zone resources and enabling bidirectional ICE ensures successful registration, proper RTP path establishment, and consistent voice quality. This solution follows Cisco best practices, supports scalable remote collaboration, reduces support incidents, and ensures predictable performance for remote clients. Proper ICE configuration guarantees reliable connectivity and optimal communication quality.

Remote Jabber registration failures and one-way audio are often caused by traversal zones operating near capacity and unsuccessful ICE (Interactive Connectivity Establishment) negotiation. ICE is essential for determining optimal IP addresses and ports so that RTP media can traverse NAT and firewall devices. CUCM Expressway traversal zones mediate secure connections between remote clients and CUCM, managing both signaling and media flows. When these traversal zones have insufficient CPU or memory resources, ICE candidate exchange may fail, preventing proper RTP path establishment, which results in registration failures, one-way audio, or even dropped calls.

Option A, increasing traversal zone resources and enabling full bidirectional ICE, addresses the root cause of the problem. Allocating additional CPU and memory ensures the traversal zone can handle peak client loads without failures, and bidirectional ICE guarantees proper candidate exchange between Jabber clients and Expressway servers. This allows RTP streams to traverse NAT and firewalls reliably, ensuring media reaches its intended destination. Cisco best practices recommend sizing traversal zones according to expected remote client loads and enabling bidirectional ICE to maintain consistent registration reliability and high-quality audio. Continuous monitoring of resource utilization allows administrators to scale proactively, preventing failures during high-demand periods and maintaining predictable, reliable performance.

Option B, requiring VPN connectivity, may resolve some NAT-related connectivity issues but introduces latency, configuration complexity, and additional points of potential failure. VPNs do not address the fundamental limitations of traversal zone resources or ICE negotiation failures, making them a less scalable and less efficient solution.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not alleviate resource constraints or ICE negotiation failures. Excessive retries under constrained resources may worsen congestion and further compromise registration success.

Option D, enabling persistent XMPP connections between Expressway-C and CUCM, improves messaging reliability but does not impact ICE negotiation or traversal zone resource limitations. Persistent XMPP alone cannot prevent registration failures or one-way audio caused by inadequate traversal zone capacity.

Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 13 Q181-195

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