Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 12 Q166-180

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Question166

A company has deployed a hybrid collaboration environment with CUCM, CUC, and Jabber clients. Remote users report call setup failures when attempting to reach on-premises SIP endpoints, and packet captures indicate that SDP offers are being sent before CUBE can rewrite them. 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:

Intermittent call setup failures between remote Jabber clients and on-premises SIP endpoints in hybrid environments are often related to the timing of SDP transmission. SDP (Session Description Protocol) is responsible for communicating media information such as IP addresses, ports, and codecs between endpoints. When SDP is included too early in the signaling process, specifically before CUBE has anchored the media path, RTP packets may be sent to incorrect IP addresses, resulting in one-way audio, call drops, or setup failures.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly resolves the issue. Delayed-offer SDP postpones the SDP inclusion in the initial INVITE until the 200 OK message. This delay allows CUBE to anchor the media path properly and rewrite the SDP with the correct information, ensuring that RTP streams are correctly directed to their destinations. Cisco best practices strongly recommend using delayed-offer SDP in multi-site or hybrid deployments where CUBE is mediating SIP trunks because it ensures the synchronization of signaling and media paths. By implementing delayed-offer SDP, enterprises can stabilize call setup behavior, reduce one-way audio incidents, and guarantee predictable and high-quality voice communications across multiple endpoints and sites.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not address the root cause. SCCP is designed primarily for Cisco endpoints and does not prevent RTP from being sent before SDP rewriting. Therefore, while SCCP may enhance endpoint compatibility, it does not resolve the fundamental media path issue.

Option C, using SIP over TCP exclusively, ensures reliable signaling delivery but does not correct the timing of SDP in relation to CUBE processing. TCP guarantees message delivery without loss but cannot prevent early RTP transmission that causes misrouted media.

Option D, disabling early media on CUBE, only affects the handling of call progress tones and ringback audio during call setup. While it influences the signaling media behavior, it does not correct the underlying problem of RTP being sent before SDP rewriting.

Implementing delayed-offer SDP in such hybrid environments provides immediate benefits, including reliable media anchoring, consistent RTP flow, reduction in call drops, and elimination of one-way audio. This configuration ensures interoperability between different CUCM clusters, remote Jabber clients, and other SIP endpoints, supporting predictable, enterprise-grade communication quality. Proper SDP handling reduces troubleshooting complexity, aligns with Cisco best practices, and maintains a seamless user experience across all collaboration endpoints. Delayed-offer SDP is essential in large-scale deployments to synchronize media and signaling, enhance operational stability, and provide consistent call behavior regardless of network topology or endpoint diversity.

Question167

Remote Jabber users behind multiple NAT layers report registration failures and intermittent one-way audio when connecting through CUCM Expressway traversal zones. Packet captures show ICE negotiation failures and traversal zone resource utilization at near maximum capacity. Which configuration MOST effectively resolves the 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:

Remote Jabber users frequently encounter registration failures and one-way audio due to traversal zone resource exhaustion and failed ICE negotiation. ICE (Interactive Connectivity Establishment) is crucial for endpoints to select the optimal IP addresses and ports for RTP streams, enabling NAT and firewall traversal. Traversal zones in Expressway mediate communication between remote clients and CUCM, ensuring secure registration and proper media path establishment. When traversal zones are overloaded, ICE candidate exchanges can fail, resulting in failed registrations and misdirected RTP, causing one-way audio.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses the root cause by ensuring that the traversal zone can handle the peak number of concurrent connections without failures. Allocating additional CPU and memory provides the necessary processing capacity for ICE candidate negotiation, registration handling, and media path management. Full bidirectional ICE ensures that candidates are successfully exchanged between remote clients and the Expressway server, enabling RTP streams to traverse NAT and firewall devices correctly. Cisco best practices recommend properly sizing traversal zones based on the expected remote client load and enabling bidirectional ICE to maintain reliable registration and high-quality media paths. Continuous monitoring of traversal zone utilization allows proactive scaling to prevent failures and ensure seamless connectivity.

Option B, requiring VPN connectivity for remote clients, may bypass some NAT-related issues but introduces additional latency, administrative complexity, and potential failure points. VPN does not address traversal zone capacity or ICE negotiation failures and is therefore not optimal for scalable enterprise deployments.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not resolve underlying traversal zone resource limitations. Excessive registration retries under high load may exacerbate congestion and fail to improve registration success.

Option D, enabling persistent XMPP connections, improves messaging reliability and presence updates but does not impact ICE negotiation or traversal zone performance. Persistent XMPP connections cannot resolve registration failures or one-way audio caused by resource exhaustion.

By increasing traversal zone resources and enabling full bidirectional ICE, remote Jabber clients can register reliably, establish correct RTP paths, and experience high-quality voice communication. This configuration aligns with Cisco best practices, supports scalable remote collaboration, reduces support incidents, and ensures predictable performance even during peak usage periods. Proper ICE handling and resource allocation guarantee seamless connectivity, high-quality communication, and consistent user experience for all remote users.

Question168

Voicemail users accessing Unity Connection report a delay of several seconds before playback starts, even though network metrics indicate no packet loss, jitter, or latency. 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 playback are commonly 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-energy sounds or silence, which VAD interprets as inactivity, delaying RTP transmission and resulting in perceived initial silence.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, directly resolves this problem by ensuring continuous RTP transmission, including during low-energy segments. Cisco best practices advise disabling VAD for voicemail to preserve message integrity and provide consistent playback. Although disabling VAD increases bandwidth usage slightly, it eliminates the initial silence issue, enhancing user experience and reducing voicemail-related support incidents. Continuous RTP transmission ensures that messages begin playing immediately, maintaining predictable behavior across all endpoints.

Option B, adjusting MWI extensions, affects visual notifications for new messages but does not influence RTP or playback timing. MWI modifications impact signaling rather than media paths, leaving the delay unresolved.

Option C, changing Unity Connection codec to G.722, improves audio fidelity but does not prevent VAD-induced RTP suppression. Codec changes do not affect the timing of RTP transmission at the start of messages.

Option D, moving the voicemail pilot to a different partition, affects call routing but does not address the media playback issue caused by VAD. Partition changes do not alter RTP flow or message playback timing.

Disabling VAD ensures continuous RTP flow, eliminating initial message silence and improving voicemail user experience. This configuration adheres to Cisco best practices, maintains message integrity, reduces support calls, and guarantees predictable playback behavior across enterprise deployments. Proper VAD configuration ensures high-quality voicemail access without affecting overall call quality or operational efficiency.

Question169

Inter-site CUCM calls over SIP trunks through CUBE experience sporadic one-way audio and occasional call drops. Analysis shows RTP being sent to incorrect 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 often occur because RTP is transmitted before CUBE anchors the media path and rewrites SDP. SDP contains the necessary information for media streams, including IP addresses, ports, and codec capabilities. When RTP is sent prematurely, it is misrouted, causing one-way audio or call failures.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, mitigates this issue by delaying SDP from the INVITE until the 200 OK response. This gives CUBE sufficient time to anchor media correctly and rewrite SDP, ensuring proper RTP delivery. Cisco recommends delayed-offer SDP in multi-site deployments to synchronize signaling and media paths, improve call reliability, and reduce troubleshooting complexity. Delayed-offer SDP prevents misrouted RTP, stabilizes inter-site communication, minimizes call drops, and ensures predictable audio quality across enterprise deployments. Proper implementation guarantees interoperability with diverse SIP endpoints and clusters, providing consistent and high-quality voice communication.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not prevent early RTP transmission. SCCP may enhance endpoint compatibility but does not address the fundamental issue of premature RTP, leaving one-way audio unresolved.

Option C, using SIP over TCP, ensures reliable signaling delivery but does not affect the timing of RTP relative to SDP rewriting. TCP guarantees message delivery but does not prevent media misrouting.

Option D, disabling early media on CUBE, affects call progress tones and ringback behavior but does not correct RTP misrouting. Early media configuration impacts signaling but does not anchor the media path.

Delayed-offer SDP ensures proper media anchoring, predictable RTP flows, and stable call quality. This configuration stabilizes inter-site communication, reduces troubleshooting effort, and provides reliable enterprise SIP trunking. Proper SDP handling supports scalable and predictable deployments while maintaining high-quality voice services for all endpoints.

Question170

Remote Jabber clients behind NAT experience frequent registration failures and one-way audio when connecting via CUCM Expressway traversal zones. ICE candidate negotiation fails often, and traversal zones operate at near maximum capacity. Which configuration MOST effectively resolves the issue?

Answer: A

Explanation:

Remote Jabber registration failures and one-way audio are frequently caused by traversal zone resource exhaustion and failed ICE negotiation. Traversal zones facilitate secure signaling and media path establishment between remote clients and CUCM. ICE enables endpoints to determine optimal IP addresses and ports for RTP streams, ensuring NAT and firewall traversal. Overloaded traversal zones may fail ICE candidate exchanges, resulting in registration failures and misrouted RTP, causing 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, ensuring traversal zones can handle peak connections reliably. Full bidirectional ICE ensures successful candidate exchange between remote clients and Expressway servers, allowing RTP streams to traverse NAT and firewall devices correctly. Cisco best practices recommend sizing traversal zones based on expected remote client populations and enabling bidirectional ICE to maintain reliable registration and high-quality media paths. Continuous monitoring allows proactive scaling to prevent failures, ensuring seamless connectivity.

Option B, requiring VPN connectivity, may bypass NAT issues but introduces latency, complexity, and potential failure points. VPN does not resolve traversal zone capacity limitations or ICE failures.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not address resource constraints, and excessive retries can worsen congestion.

Option D, enabling persistent XMPP connections improves messaging reliability but does not influence ICE negotiation or traversal zone performance. Persistent XMPP has no impact on registration failures or RTP path issues.

Increasing traversal zone resources and enabling full bidirectional ICE ensures remote Jabber clients register successfully, RTP paths are correctly established, and voice quality is maintained. This approach supports scalable remote collaboration, aligns with Cisco best practices, and delivers reliable enterprise voice services during periods of peak usage. Proper ICE configuration and resource allocation ensure predictable performance and seamless connectivity.

Question171

A company implements SIP trunks between multiple CUCM clusters through a CUBE. Users report intermittent one-way audio and occasional call drops. Analysis reveals RTP packets being sent to incorrect IP addresses 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:

One-way audio and call drops in multi-cluster CUCM environments with CUBE mediation are often caused by RTP being sent before SDP is anchored and rewritten by CUBE. SDP (Session Description Protocol) carries essential information for media negotiation, including IP addresses, ports, and codec information. When SDP is included too early in the signaling process, RTP may be misrouted, leading to audio issues or call failures.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly addresses this issue. Delayed-offer SDP postpones the inclusion of SDP in the initial INVITE until the 200 OK response, allowing CUBE to anchor the media path and rewrite SDP accurately. Cisco best practices recommend this configuration for multi-site or hybrid deployments because it ensures proper synchronization between signaling and media, stabilizes call quality, and reduces troubleshooting complexity. Implementing delayed-offer SDP improves interoperability between CUCM clusters and remote endpoints, minimizes call drops, and eliminates one-way audio by guaranteeing correct RTP delivery.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not prevent premature RTP transmission. While SCCP may enhance endpoint compatibility in certain scenarios, it does not resolve the fundamental issue of RTP being sent before SDP rewriting.

Option C, using SIP over TCP exclusively, ensures reliable signaling delivery but does not influence the timing of SDP relative to media path establishment. TCP guarantees message delivery but does not prevent misrouted RTP, leaving one-way audio unresolved.

Option D, disabling early media on CUBE, affects call progress tones and ringback audio but does not anchor the media path or correct RTP misrouting. Early media configuration changes signaling behavior during call setup without addressing the underlying media path problem.

Enabling delayed-offer SDP ensures predictable RTP flow, proper media anchoring, and stable inter-cluster call performance. This configuration aligns with Cisco best practices, reduces troubleshooting overhead, enhances call reliability, and provides consistent voice quality across enterprise SIP trunks. It ensures interoperability between clusters, supports scalable deployments, and guarantees a seamless communication experience for users, addressing both media path and signaling synchronization comprehensively.

Question172

Remote Jabber clients behind NAT report registration failures and intermittent one-way audio when connecting through CUCM Expressway traversal zones. ICE candidate negotiation often fails, and traversal zones are approaching capacity limits. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Remote Jabber client registration failures and one-way audio are typically caused by insufficient traversal zone resources and failed ICE negotiation. ICE (Interactive Connectivity Establishment) enables endpoints to determine the optimal IP addresses and ports for RTP streams, facilitating NAT traversal. Traversal zones in Expressway mediate communication between remote clients and CUCM, ensuring secure registration and proper media path establishment. When traversal zones are overloaded or near maximum capacity, ICE candidate exchanges may fail, causing registration failures and misdirected RTP streams, resulting in one-way audio.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, resolves the issue. Allocating additional CPU and memory ensures that traversal zones can handle the peak number of concurrent remote clients without failures. Full bidirectional ICE ensures successful candidate exchange between Jabber clients and Expressway servers, allowing RTP streams to traverse NAT devices and firewalls effectively. Cisco best practices recommend sizing traversal zones based on the expected number of remote clients and enabling bidirectional ICE to maintain reliable registration and media path performance. Continuous monitoring of traversal zone resource utilization enables proactive scaling to prevent failures and maintain seamless connectivity.

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

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not solve the underlying resource limitations. Excessive retries can worsen congestion and do not improve registration success under resource-limited conditions.

Option D, enabling persistent XMPP connections, improves messaging reliability and presence updates but has no impact on ICE negotiation or traversal zone performance. Persistent XMPP connections cannot resolve registration failures or one-way audio caused by insufficient traversal zone resources.

Increasing traversal zone resources and enabling full bidirectional ICE ensures that Jabber clients register successfully, RTP paths are correctly established, and voice quality is maintained. This solution supports scalable remote collaboration, aligns with Cisco best practices, reduces support incidents, and guarantees predictable performance even during periods of high load. Proper ICE handling and traversal zone resource allocation ensure seamless connectivity, high-quality communication, and consistent user experience for all remote users.

Question173

Unity Connection voicemail users report a noticeable delay at the start of message playback, despite network performance showing no packet loss, jitter, or latency. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Initial playback delays in voicemail messages are commonly caused by Voice Activity Detection (VAD), which suppresses RTP transmission during periods of low audio energy to conserve bandwidth during live calls. Voicemail messages often begin with low-volume sounds or silence, which VAD interprets as inactivity, delaying RTP transmission and causing perceived initial silence.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, resolves the issue by allowing RTP to flow continuously, including during low-energy segments. Cisco best practices advise disabling VAD for voicemail to preserve message integrity and provide consistent playback. Although disabling VAD slightly increases bandwidth usage, it ensures immediate message playback, improves user experience, and reduces voicemail-related support incidents. Continuous RTP transmission guarantees predictable behavior across all endpoints and maintains message integrity.

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

Option C, changing Unity Connection codec to G.722, improves audio fidelity but does not prevent VAD-induced delays. Codec selection affects sound quality but cannot overcome RTP suppression at the beginning of a voicemail message.

Option D, moving the voicemail pilot to a different partition, modifies call routing but does not impact media flow or VAD behavior. Partition changes do not address the initial silence problem.

Disabling VAD ensures continuous RTP flow and immediate message playback, improving the voicemail user experience. This configuration adheres to Cisco best practices, maintains message integrity, reduces support calls, and provides predictable behavior across enterprise deployments. Proper configuration guarantees reliable voicemail service without affecting overall call quality, maintaining high-quality access for all users.

Question174

CUCM inter-site calls over SIP trunks mediated by CUBE experience occasional one-way audio and sporadic call drops. RTP is observed being transmitted to incorrect 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:

One-way audio and call drops in inter-site SIP trunk deployments are often caused by RTP being transmitted before CUBE has anchored the media path and rewritten SDP. SDP communicates media parameters such as IP addresses, ports, and codec capabilities. Premature RTP transmission results in misrouted packets, leading to one-way audio or call failure.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, mitigates this problem by postponing the SDP from the INVITE to the 200 OK response. This allows CUBE to anchor media correctly and rewrite SDP, ensuring that RTP flows are directed to the correct endpoints. Cisco best practices recommend delayed-offer SDP for multi-site or hybrid deployments to synchronize signaling and media paths, improve call reliability, and reduce troubleshooting complexity. Implementing delayed-offer SDP stabilizes inter-site communication, minimizes call drops, and guarantees predictable audio quality, enhancing interoperability between clusters and endpoints.

Option B, converting SIP trunks to SCCP, changes signaling protocols but does not prevent early RTP transmission. SCCP may improve endpoint compatibility in some cases but does not solve RTP misrouting caused by premature SDP.

Option C, using SIP over TCP, ensures reliable delivery of signaling messages but does not influence RTP timing relative to SDP rewriting. TCP guarantees message delivery but does not prevent misrouted RTP packets.

Option D, disabling early media on CUBE, affects call progress tones but does not correct misrouted RTP. Early media configuration impacts signaling rather than media path establishment, leaving one-way audio unresolved.

Enabling delayed-offer SDP ensures proper media anchoring, predictable RTP flows, and stable call quality. It aligns with Cisco best practices, reduces troubleshooting efforts, and provides reliable inter-site SIP trunking. Proper SDP handling supports scalable and predictable enterprise deployments while maintaining high-quality voice services across all endpoints.

Question175

Remote Jabber users behind NAT report frequent registration failures and one-way audio when connecting through CUCM Expressway traversal zones. ICE candidate negotiation often fails, and traversal zones are operating near capacity. Which configuration MOST effectively resolves the issue?

Answer: A

Explanation:

Remote Jabber registration failures and one-way audio are frequently caused by traversal zone resource exhaustion and failed ICE negotiation. Traversal zones mediate secure communication between remote clients and CUCM. ICE allows endpoints to determine optimal IP addresses and ports for RTP streams, enabling NAT traversal. Resource constraints on traversal zones can lead to failed ICE negotiation, resulting in registration failures and misdirected RTP, causing one-way audio.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses the root cause. Additional CPU and memory allow traversal zones to handle concurrent connections reliably. Full bidirectional ICE ensures proper candidate exchange between Jabber clients and Expressway servers, allowing RTP streams to traverse NAT devices and firewalls correctly. Cisco best practices recommend sizing traversal zones based on expected remote client populations and enabling bidirectional ICE to maintain registration reliability and high-quality media paths. Continuous monitoring allows proactive scaling, ensuring seamless connectivity and predictable performance.

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

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not address resource limitations. Excessive retries can worsen congestion and fail to improve registration success.

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

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

Question176

A large enterprise with multiple CUCM clusters is experiencing intermittent one-way audio and call drops on inter-site SIP trunk calls mediated through CUBE. Analysis shows that RTP is being transmitted to endpoints before CUBE has had a chance to rewrite 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 exclusively
D) Disable early media on CUBE

Answer: A

Explanation:

Intermittent one-way audio and call drops in multi-cluster CUCM environments mediated through CUBE are usually caused by RTP packets being transmitted before SDP rewriting occurs. SDP, or Session Description Protocol, conveys essential media parameters, including IP addresses, port numbers, and codec capabilities. Premature transmission of RTP packets can lead to misrouting, resulting in one-way audio, poor call quality, or call failure.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly resolves this issue. Delayed-offer SDP defers the inclusion of SDP in the initial INVITE message until the 200 OK response is received. This allows CUBE to anchor the media path and rewrite SDP accurately with the correct IP addresses and ports. By implementing delayed-offer SDP, RTP flows are properly directed to endpoints, eliminating misrouting, one-way audio, and call drops. Cisco recommends this configuration for hybrid or multi-site environments because it synchronizes signaling and media paths, ensuring reliable inter-cluster communication.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not address premature RTP transmission. While SCCP may improve endpoint compatibility, it does not resolve the underlying issue of RTP being sent before SDP rewriting. Therefore, one-way audio and call instability would persist.

Option C, using SIP over TCP exclusively, ensures reliable delivery of signaling messages but does not affect the timing of RTP relative to SDP rewriting. TCP guarantees that signaling messages arrive intact but does not anchor the media path, leaving RTP misrouting unresolved.

Option D, disabling early media on CUBE, affects call progress tones and ringback behavior during call setup but does not correct the misrouting of RTP packets. Early media settings impact signaling behaviors but do not anchor media paths or synchronize SDP and RTP transmission.

By enabling delayed-offer SDP, enterprises ensure predictable RTP flow, correct media path establishment, and stable inter-site call performance. This configuration aligns with Cisco best practices, reduces troubleshooting efforts, enhances call reliability, and guarantees consistent voice quality. It supports interoperability between clusters, scalable deployments, and a seamless communication experience for users across all sites. Proper SDP handling ensures that enterprise voice services are reliable, media streams are accurately directed, and operational complexity is minimized.

Question177

Remote Jabber clients behind multiple layers of NAT report intermittent registration failures and one-way audio when connecting through CUCM Expressway traversal zones. ICE negotiation fails frequently, and traversal zones are near resource capacity. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Remote Jabber registration failures and one-way audio issues are often caused by insufficient resources in traversal zones and failed ICE negotiation. ICE (Interactive Connectivity Establishment) allows endpoints to determine optimal IP addresses and ports for RTP streams, enabling NAT and firewall traversal. CUCM Expressway traversal zones mediate communication between remote clients and CUCM to establish secure registrations and RTP paths. When traversal zones approach resource limits, ICE candidate exchanges fail, preventing proper RTP path establishment, causing registration failures and one-way audio.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, directly addresses the problem. Allocating additional CPU and memory allows traversal zones to handle peak client loads without failure. Full bidirectional ICE ensures that candidates are successfully exchanged between remote clients and Expressway servers, allowing RTP streams to traverse NAT and firewalls correctly. Cisco best practices recommend sizing traversal zones according to the expected number of remote clients and enabling bidirectional ICE to maintain reliable registration and high-quality media paths. Continuous monitoring of resource utilization allows administrators to proactively scale traversal zones before failures occur.

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

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not alleviate resource limitations in the traversal zone. Excessive retries under high load can exacerbate congestion, further increasing failures.

By increasing traversal zone resources and enabling bidirectional ICE, remote Jabber clients can register reliably, establish correct RTP paths, and experience high-quality voice communication. This configuration ensures scalable remote collaboration, aligns with Cisco best practices, reduces support incidents, and guarantees predictable performance during peak usage periods. Proper ICE handling and resource allocation ensure seamless connectivity and a consistent communication experience for all users.

Question178

Voicemail users report a delay of several seconds before message playback starts in Unity Connection, despite network metrics showing no packet loss or jitter. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Delays in voicemail playback are commonly caused by Voice Activity Detection (VAD). VAD suppresses RTP transmission during periods of low audio energy to conserve bandwidth in live calls. Voicemail messages often begin with low-energy sounds or silence, which VAD interprets as inactivity, delaying RTP transmission and causing the user to perceive an initial delay before playback.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, resolves this problem by ensuring continuous RTP transmission, including low-energy portions of the message. Cisco best practices recommend disabling VAD for voicemail to maintain message integrity and provide consistent playback behavior. While disabling VAD slightly increases bandwidth usage, it ensures that voicemail messages start immediately, improving user experience and reducing support incidents. Continuous RTP transmission guarantees predictable playback behavior across endpoints, maintaining message fidelity.

Option B, adjusting MWI extensions, affects visual notifications but does not influence RTP flow or the timing of message playback. Changes to MWI only impact signaling for new message notifications.

Option C, changing the codec to G.722, improves audio fidelity but does not prevent VAD-induced RTP suppression. Codec changes affect audio quality but not the timing of playback.

Option D, moving the voicemail pilot to a different partition, changes routing but does not affect media flow or VAD behavior. Partition configuration does not impact message playback delay.

Disabling VAD ensures continuous RTP flow, eliminating the initial silence and enhancing the voicemail user experience. This configuration adheres to Cisco best practices, preserves message integrity, and maintains consistent behavior across enterprise deployments. Proper configuration ensures reliable voicemail service without affecting overall call quality or operational efficiency.

Question179

CUCM inter-site SIP trunk calls through CUBE experience sporadic one-way audio and call drops. RTP is sent 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 deployments are frequently caused by RTP being transmitted before CUBE has anchored the media path and rewritten SDP. SDP conveys critical media parameters, including IP addresses, ports, and codec preferences. Premature RTP results in misrouted packets and degraded call quality.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, mitigates this issue by postponing SDP transmission until the 200 OK response. This allows CUBE to anchor media properly and rewrite SDP, ensuring RTP flows correctly to endpoints. Cisco recommends delayed-offer SDP for multi-site or hybrid environments because it synchronizes signaling and media paths, stabilizes call quality, and reduces troubleshooting complexity. Implementing delayed-offer SDP stabilizes inter-site communication, reduces call drops, and guarantees predictable audio quality.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not prevent early RTP transmission. SCCP may improve endpoint compatibility but does not resolve RTP misrouting.

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

Option D, disabling early media on CUBE, affects call progress tones but does not anchor RTP flows. Early media changes only signaling behavior without correcting the underlying media path problem.

By enabling delayed-offer SDP, enterprises ensure correct RTP flows, proper media anchoring, and stable inter-site calls. This configuration follows Cisco best practices, reduces troubleshooting overhead, and guarantees high-quality voice communication.

Question180

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

Answer: A

Explanation:

Remote Jabber registration failures and one-way audio often result from insufficient traversal zone resources and failed ICE negotiation. ICE allows endpoints to determine optimal IP addresses and ports for RTP streams to traverse NAT and firewall devices. CUCM Expressway traversal zones mediate secure connections between remote clients and CUCM, ensuring proper registration and RTP path establishment. When traversal zones approach capacity, ICE negotiation fails, preventing successful RTP path establishment and causing one-way audio.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, directly addresses the root cause. Allocating additional CPU and memory ensures traversal zones can handle peak client loads without failures. Full bidirectional ICE guarantees proper candidate exchange between Jabber clients and Expressway servers, allowing RTP streams to traverse NAT and firewalls correctly. Cisco best practices recommend sizing traversal zones according to remote client load and enabling bidirectional ICE to maintain reliable registration and high-quality media paths. Continuous monitoring enables proactive scaling to prevent failures.

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

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not alleviate resource constraints. Excessive retries can worsen congestion without improving registration success.

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

Remote Jabber client registration failures and one-way audio are frequently observed in Cisco Unified Communications deployments when traversal zone resources are insufficient or ICE (Interactive Connectivity Establishment) negotiation fails. The traversal zones, established between the Expressway-C and Expressway-E servers, mediate connections between remote clients and CUCM, allowing secure traversal of NAT and firewall devices. These zones handle signaling and media traffic and are responsible for ensuring that RTP streams reach the intended client endpoints. When traversal zone resources—such as CPU and memory allocations—are inadequate, the zone cannot process all registration or media requests efficiently. This results in registration failures, incomplete ICE candidate exchanges, or one-way audio due to unsuccessful media path establishment.

ICE is critical for remote client communication in multi-network environments. It allows endpoints to gather multiple candidate IP addresses and ports, exchange these candidates with the peer, and determine the optimal media path for RTP traffic. Remote clients behind NAT or firewall devices rely on this negotiation to establish successful audio and video streams. Full bidirectional ICE ensures that the candidate exchange is properly conducted both from the client to the server and from the server to the client, accommodating symmetrical NAT scenarios and facilitating consistent connectivity. If bidirectional ICE is not enabled or resources are insufficient, the ICE negotiation may fail, preventing RTP streams from establishing properly and leading to one-way audio or dropped calls.

Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 12 Q166-180

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