Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 10 Q136-150

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Question136

An organization experiences intermittent one-way audio and call drops on inter-site SIP trunk calls using CUCM and CUBE. RTP analysis shows packets occasionally 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) Use SIP over TCP exclusively
C) Disable early media on CUBE
D) Enable symmetric RTP on CUBE

Answer: A

Explanation:

One-way audio and intermittent call drops in multi-site SIP trunk deployments are commonly caused by RTP packets being sent before CUBE completes SDP rewriting, which results in media being directed to incorrect IP addresses or ports. SDP (Session Description Protocol) provides the necessary media parameters such as codecs, IP addresses, and ports that endpoints use to establish the media path. If RTP is transmitted before CUBE anchors the media path and rewrites SDP, endpoints may attempt to send audio to wrong destinations, causing one-way audio or call failures.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly addresses this issue. Delayed-offer SDP moves the SDP payload from the initial INVITE message to the 200 OK response, allowing CUBE to process signaling first, anchor the media, and rewrite IP addresses and ports. This ensures that RTP flows are correctly aligned with signaling paths. Cisco best practices recommend delayed-offer SDP for multi-site SIP trunk deployments, especially in complex networks with NAT traversal, multiple clusters, and CUBE mediation. Delayed-offer SDP prevents early RTP transmission issues, stabilizes call quality, and reduces troubleshooting complexity. Implementing delayed-offer SDP also ensures interoperability with a variety of SIP endpoints and service providers.

Option B, using SIP over TCP exclusively, provides reliable signaling transport but does not prevent early RTP transmission. TCP ensures delivery of SIP messages but does not influence when or how RTP packets are sent. Consequently, one-way audio and media misalignment issues would persist if delayed-offer SDP is not implemented. TCP may improve call setup reliability but does not directly resolve media path issues caused by premature RTP.

Option C, disabling early media on CUBE, affects call progress tones and ringback behavior, but it does not prevent RTP from being transmitted to incorrect destinations prior to SDP rewriting. Early media configuration is relevant for signaling and user experience but does not address the fundamental cause of one-way audio due to early RTP transmission. Disabling early media might mask the symptom temporarily but does not provide a sustainable solution to media path alignment issues.

Option D, enabling symmetric RTP on CUBE, assists NAT traversal by forcing RTP to be sent and received from the same source, which can help with endpoints behind NAT or firewalls. While symmetric RTP may improve RTP flow in certain scenarios, it does not replace the need for proper SDP handling and delayed-offer SDP. Without delayed-offer SDP, RTP packets may still be misaligned with signaling paths, leading to intermittent audio issues.

By enabling delayed-offer SDP, administrators ensure that the signaling and media paths are correctly synchronized, preventing one-way audio and call drops. This approach provides predictable call behavior, consistent voice quality, and simplified troubleshooting. Furthermore, delayed-offer SDP ensures that complex multi-site deployments, including those using CUBE for SIP trunking, maintain high-quality voice services even during peak traffic or under high load. Proper SDP handling aligns with Cisco’s best practices for enterprise collaboration deployments, reduces support incidents, and delivers a seamless user experience across multiple sites.

Question137

Remote Jabber clients report intermittent registration failures and one-way audio when accessing CUCM via Expressway traversal zones. RTP packet analysis indicates ICE candidate negotiation failures. Which configuration MOST effectively resolves this issue?

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

Answer: A

Explanation:

Remote Jabber clients rely on Expressway traversal zones to facilitate secure registration, signaling, and media path establishment to CUCM. Intermittent registration failures and one-way audio are usually caused by traversal zone resource limitations and failed ICE (Interactive Connectivity Establishment) candidate negotiation. ICE enables endpoints behind NAT and firewalls to identify the best IP addresses and ports for RTP media. When traversal zones are overloaded or ICE negotiation fails, clients may not register successfully, and media paths may not establish correctly, resulting in call quality issues.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, is the most effective solution. Allocating additional CPU and memory to traversal zones allows them to handle peak registration and call loads, reducing the likelihood of registration failures. Full bidirectional ICE ensures that both the Jabber client and the Expressway server can exchange candidates effectively, allowing RTP streams to traverse NAT or firewall barriers correctly. Cisco best practices emphasize properly sizing traversal zones based on the expected number of remote clients and enabling bidirectional ICE to maintain high-quality voice services. Continuous monitoring of traversal zone utilization is recommended to anticipate peak loads and prevent resource exhaustion.

Option B, requiring VPN connectivity for remote clients, increases administrative complexity and latency while introducing a potential single point of failure. While VPNs can bypass NAT traversal challenges, they do not directly solve the underlying traversal zone resource limitations or ICE negotiation failures and may reduce user experience due to additional overhead.

Option C, reducing CUCM SIP registration timers, increases the frequency of retry attempts but does not alleviate the core issue of traversal zone resource constraints. Frequent retries under high load can exacerbate congestion and lead to more registration failures.

Option D, enabling persistent XMPP connections, may improve messaging reliability and presence updates but does not resolve registration or media path establishment failures caused by overloaded traversal zones or ICE candidate negotiation issues. XMPP persistence does not directly influence RTP path selection or registration success.

By implementing Option A, the organization ensures that traversal zones can manage peak registration loads, bidirectional ICE negotiation occurs reliably, and RTP paths are established correctly. This solution provides consistent registration, eliminates one-way audio for remote Jabber users, supports scalable remote collaboration, and reduces the operational burden on IT staff. Aligning traversal zone capacity with ICE support follows Cisco best practices for Mobile and Remote Access, ensuring robust and predictable voice and video performance across enterprise deployments.

Question138

Users accessing Unity Connection via CUCM report initial silence of several seconds when playing voicemail messages. RTP analysis indicates no packet loss or jitter. Which configuration MOST effectively addresses 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:

The initial silence in voicemail playback is typically caused by Voice Activity Detection (VAD). VAD suppresses RTP transmission during periods of low energy to conserve bandwidth during live calls. However, voicemail messages often begin with low-energy segments, such as silence or soft voice, which VAD interprets as inactivity. As a result, the start of the voicemail message is delayed until VAD allows RTP transmission to resume, creating perceived initial silence.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, is the most effective solution. Disabling VAD ensures continuous RTP transmission for all segments of voicemail messages, including low-energy portions, preventing delayed playback. Cisco best practices recommend disabling VAD for voicemail to preserve message integrity and ensure a high-quality user experience. While this slightly increases bandwidth usage, the impact is minimal compared to the improvement in playback reliability. Eliminating the initial silence reduces user frustration and minimizes support calls related to perceived message delays.

Option B, adjusting MWI extensions, affects lamp signaling to indicate new messages but has no impact on RTP delivery or message playback. This configuration does not address the root cause of initial silence in voicemail playback.

Option C, changing Unity Connection codec to G.722, improves audio fidelity but does not solve the issue caused by VAD. The initial silence will persist if VAD is still enabled, regardless of codec selection.

Option D, moving the voicemail pilot to a different partition, changes call routing but does not influence RTP transmission or VAD behavior. Voicemail playback delays are unrelated to partition assignments.

By disabling VAD, administrators ensure that voicemail messages play immediately and continuously, enhancing user experience and maintaining message integrity. This approach aligns with Cisco best practices, reduces the number of support incidents related to voicemail, and ensures predictable performance for enterprise users across all endpoints.

Question139

Inter-site calls between two CUCM clusters using SIP trunks through CUBE occasionally experience one-way audio and call drops. RTP packets are misdirected before SDP rewriting 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:

One-way audio and intermittent call drops in multi-site SIP trunk deployments are frequently caused by early RTP transmission before CUBE has rewritten the SDP payload. The SDP contains essential information about codecs, IP addresses, and ports that endpoints use to send RTP. When RTP packets are sent prior to SDP rewriting, audio can be misrouted, resulting in one-way audio and call failures.

Option A, enabling delayed-offer SDP, is the correct solution. Delayed-offer SDP postpones the inclusion of SDP information from the INVITE to the 200 OK response. This allows CUBE to anchor media, rewrite IP addresses and ports correctly, and ensure that RTP flows align with signaling paths. Cisco best practices recommend delayed-offer SDP in multi-site SIP trunk deployments to ensure proper media path alignment, reduce troubleshooting complexity, and prevent one-way audio. Delayed-offer SDP is particularly critical when NAT, firewall traversal, or multi-cluster scenarios are involved.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not address early RTP transmission. While SCCP may improve endpoint interoperability, it does not solve media misalignment issues caused by premature RTP flows.

Option C, using SIP over TCP, provides reliable delivery of signaling messages but does not prevent RTP from being transmitted before SDP processing. TCP transport ensures message reliability but does not influence the timing of media path establishment.

Option D, disabling early media on CUBE, affects call progress tones but does not correct the misaligned RTP paths. Early media settings influence signaling but cannot resolve the root cause of one-way audio.

Implementing delayed-offer SDP ensures proper media anchoring, eliminates one-way audio, and stabilizes inter-site call quality. This configuration adheres to Cisco best practices, minimizes user-impacting call issues, and provides predictable voice quality across multi-site SIP trunk deployments.

Question140

Remote Jabber clients report intermittent registration failures and one-way audio when connecting to CUCM via Expressway. ICE candidate exchanges frequently fail, and traversal zones are under high load. Which configuration MOST effectively mitigates these issues?

A) Increase traversal zone capacity 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:

Intermittent registration failures and one-way audio for remote Jabber clients typically indicate traversal zone resource exhaustion and failed ICE negotiation. Traversal zones facilitate secure signaling, registration, and media path establishment between remote endpoints and CUCM. ICE ensures endpoints behind NAT or firewalls can establish optimal RTP paths. When traversal zones are under high load, candidate exchanges fail, registration is delayed, and audio may be one-way.

Option A, increasing traversal zone capacity and enabling full bidirectional ICE support, addresses the root cause. Proper CPU and memory allocation allows traversal zones to handle peak connections, preventing registration failures. Bidirectional ICE ensures both clients and servers exchange candidates successfully, allowing RTP streams to traverse NAT and firewalls correctly. Cisco best practices emphasize sizing traversal zones according to anticipated concurrent remote endpoints and enabling full ICE support to maintain reliable remote collaboration. Continuous monitoring of traversal zone utilization helps anticipate peaks and adjust resources proactively.

Option B, requiring VPN connectivity, increases complexity and latency while introducing potential single points of failure. VPNs do not solve traversal zone overload or ICE failures.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not alleviate resource limitations and may worsen congestion.

Option D, enabling persistent XMPP connections, improves messaging reliability but does not impact traversal zone overload or ICE failures that affect registration and media path quality.

By implementing Option A, the organization ensures reliable registration, correct media path establishment, and high-quality voice for remote Jabber users. This solution supports scalable remote collaboration, reduces call drops, and aligns with Cisco best practices for Mobile and Remote Access deployments, ensuring predictable and high-quality voice and video performance across enterprise networks.

Question141

During an inter-site SIP trunk deployment using CUCM and CUBE, users report occasional one-way audio and dropped calls. RTP analysis shows packets sent to incorrect IP addresses prior to SDP rewriting. Which configuration MOST effectively resolves this issue?

A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Use SIP over UDP exclusively
C) Disable early media on CUBE
D) Enable symmetric RTP on CUBE

Answer: A

Explanation:

In multi-site SIP trunk environments with CUCM and CUBE, one-way audio and dropped calls commonly result from RTP packets being sent before CUBE can rewrite the SDP payload. SDP defines key media parameters such as codec, IP address, and port information. When RTP is transmitted before CUBE anchors the media path and modifies SDP, the media can be directed incorrectly, resulting in one-way audio or call failures.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, addresses the root cause. Delayed-offer SDP postpones sending the SDP from the initial INVITE to the 200 OK response, allowing CUBE to process signaling, anchor media, and rewrite SDP properly. This ensures that RTP packets are transmitted only after the media path is correctly aligned with signaling. Cisco best practices emphasize delayed-offer SDP in multi-site SIP deployments to prevent early RTP issues, improve call stability, and enhance overall voice quality. Implementing delayed-offer SDP allows predictable call behavior, simplifies troubleshooting, and maintains interoperability across multiple SIP endpoints and clusters.

Option B, using SIP over UDP exclusively, ensures faster transmission but does not prevent early RTP delivery. While UDP is efficient for signaling, it does not influence the timing of RTP transmission or SDP rewriting. Consequently, one-way audio issues would persist.

Option C, disabling early media on CUBE, impacts call progress tones but does not address RTP misalignment. Early media affects signaling behavior rather than media path establishment, so the underlying cause of one-way audio remains unaddressed.

Option D, enabling symmetric RTP, assists NAT traversal by ensuring RTP is received and transmitted from the same source IP and port. While helpful for NAT-related issues, symmetric RTP does not replace proper SDP handling, and early RTP transmission can still occur without delayed-offer SDP.

Implementing delayed-offer SDP ensures synchronized signaling and media paths, stabilizes call quality, and reduces operational troubleshooting. Proper SDP handling in multi-site environments supports complex deployments, mitigates one-way audio, prevents call drops, and aligns with Cisco’s recommended practices for enterprise voice networks. This solution maintains consistent audio quality, ensures reliable inter-site SIP communication, and provides a seamless experience for end users.

Question142

Remote Jabber users behind NAT report intermittent registration failures and one-way audio when accessing CUCM via Expressway traversal zones. ICE candidate exchanges are failing intermittently. Which configuration MOST effectively addresses this issue?

Answer: A

Explanation:

Remote Jabber clients depend on Expressway traversal zones to manage secure registration, signaling, and media path establishment with CUCM. Intermittent registration failures and one-way audio typically result from traversal zone resource limitations and failed ICE (Interactive Connectivity Establishment) candidate negotiation. ICE allows endpoints behind NAT or firewalls to determine optimal IP addresses and ports for RTP media streams. When traversal zones are overloaded, ICE candidate exchanges may fail, leading to registration failures and one-way audio.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, directly addresses these issues. Allocating additional CPU and memory ensures traversal zones can accommodate peak concurrent connections, preventing registration failures due to capacity limitations. Full bidirectional ICE guarantees proper candidate exchange, enabling RTP streams to traverse NAT and firewall barriers successfully. Cisco best practices recommend scaling traversal zones according to the number of remote endpoints and enabling full ICE support to ensure reliable voice and video communication. Continuous monitoring of traversal zone performance is crucial to preemptively adjust resources, particularly during periods of high usage.

Option B, requiring VPN connectivity for remote endpoints, increases administrative complexity and latency while introducing a potential single point of failure. VPNs may circumvent NAT traversal challenges, but they do not resolve resource constraints or ICE negotiation failures within traversal zones.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not alleviate the fundamental issue of traversal zone overload. Excessive retries under high load can exacerbate resource utilization and worsen registration failures.

Option D, enabling persistent XMPP connections between Expressway-C and CUCM, improves messaging reliability but does not resolve registration or media path failures caused by traversal zone overload or ICE failures. Persistent XMPP connections do not influence RTP path establishment or registration success.

By implementing Option A, administrators ensure reliable registration, consistent media path establishment, and high-quality voice for remote Jabber clients. This solution supports scalable remote collaboration, reduces call drops, and aligns with Cisco best practices for Mobile and Remote Access deployments, delivering predictable and high-quality performance across enterprise networks. Proper traversal zone sizing and ICE support ensure seamless communication even during periods of peak load.

Question143

Users accessing Unity Connection through CUCM report several seconds of silence at the start of voicemail playback. RTP analysis shows no loss, jitter, or packet delay. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Initial silence during voicemail playback is frequently caused by Voice Activity Detection (VAD), which suppresses RTP transmission during low-energy audio periods to conserve bandwidth during live calls. Voicemail messages often start with low-energy segments such as silence or soft audio, which VAD interprets as inactivity, resulting in a delayed start to playback.

Option A, disabling VAD on the SIP trunk, is the most effective solution. Disabling VAD allows continuous RTP transmission, including low-energy audio, eliminating initial playback silence. Cisco best practices recommend disabling VAD for voicemail services to maintain message integrity and provide a consistent user experience. Although bandwidth utilization increases slightly, the trade-off ensures immediate and uninterrupted message playback. Removing VAD-related delays improves end-user satisfaction, reduces support calls, and ensures predictable behavior across different endpoints.

Option B, adjusting MWI extensions, affects lamp signaling to indicate new messages but does not impact RTP transmission or message playback. Adjusting MWI addresses notification behavior rather than media delivery, so the initial silence remains unaddressed.

Option C, changing Unity Connection codec to G.722, improves audio fidelity but does not eliminate silence caused by VAD. The delayed playback issue is independent of codec selection and persists unless VAD is disabled.

Option D, moving the voicemail pilot to a different partition, modifies call routing but has no effect on RTP transmission or VAD behavior. Partition changes do not influence message playback timing or initial silence.

By disabling VAD, voicemail messages play immediately and continuously, preserving message integrity and improving the end-user experience. This configuration aligns with Cisco best practices, reduces voicemail-related support incidents, and ensures consistent performance for enterprise users accessing Unity Connection. Administrators can combine this approach with monitoring and capacity planning to maintain high-quality voice services while minimizing operational impact.

Question144

During inter-site CUCM calls via SIP trunks through CUBE, users experience occasional call drops and one-way audio. RTP packets sometimes go to incorrect IP addresses 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 intermittent call drops in multi-site SIP trunk deployments are frequently caused by RTP transmission occurring before CUBE completes SDP rewriting. SDP contains the necessary media parameters, including codecs, IP addresses, and ports. If RTP is transmitted prematurely, packets are misdirected, resulting in one-way audio and dropped calls.

Option A, enabling delayed-offer SDP, addresses the root cause. Delayed-offer SDP shifts the SDP from the initial INVITE to the 200 OK response, allowing CUBE to anchor media, rewrite IP addresses and ports, and ensure that RTP flows align with signaling paths. Cisco best practices emphasize the use of delayed-offer SDP in multi-site SIP trunk deployments to prevent early RTP transmission, improve call stability, and enhance voice quality. Proper SDP handling simplifies troubleshooting and ensures interoperability across various SIP endpoints and clusters.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not prevent early RTP transmission. SCCP may improve endpoint interoperability but does not correct media path alignment issues caused by premature RTP.

Option C, using SIP over TCP, provides reliable signaling delivery but does not influence the timing of RTP transmission. TCP ensures message delivery but does not prevent one-way audio due to early RTP flows.

Option D, disabling early media on CUBE, affects call progress tones but does not correct the misaligned RTP path. Early media configuration impacts signaling rather than media path establishment, so one-way audio persists without delayed-offer SDP.

Enabling delayed-offer SDP ensures correct media anchoring, predictable RTP delivery, and consistent call quality. This solution aligns with Cisco best practices, reduces call drops, and provides a reliable user experience for inter-site SIP trunking in complex enterprise deployments.

Question145

Remote Jabber clients report intermittent registration failures and one-way audio when connecting to CUCM via Expressway. ICE candidate exchanges frequently fail, and traversal zones are near capacity. Which configuration MOST effectively mitigates these issues?

Answer: A

Explanation:

Intermittent registration failures and one-way audio for remote Jabber clients typically indicate traversal zone resource constraints and failed ICE candidate negotiation. Traversal zones facilitate secure signaling, registration, and media path establishment between remote endpoints and CUCM. ICE allows endpoints behind NAT or firewalls to establish the optimal IP address and port combination for RTP media streams. When traversal zones are under heavy load, candidate exchanges can fail, leading to registration failures and media path issues.

Option A, increasing traversal zone capacity and enabling full bidirectional ICE support, directly addresses the root causes. Allocating sufficient CPU and memory allows traversal zones to handle peak concurrent connections without overloading, preventing registration failures. Bidirectional ICE ensures successful exchange of candidates, allowing RTP streams to traverse NAT and firewall barriers reliably. Cisco best practices recommend proper sizing of traversal zones based on expected remote endpoints and enabling full ICE support to maintain consistent registration and media path establishment. Continuous monitoring of traversal zone utilization helps anticipate peak loads and adjust resources proactively, ensuring uninterrupted service.

Option B, requiring VPN connectivity for remote clients, introduces complexity, latency, and a potential single point of failure. While VPNs can bypass NAT traversal, they do not resolve traversal zone overload or ICE negotiation failures.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not alleviate resource limitations, potentially worsening congestion.

Option D, enabling persistent XMPP connections, improves messaging reliability but does not address traversal zone overload or ICE failures affecting registration and RTP paths. XMPP persistence does not influence media path establishment.

Implementing Option A ensures reliable registration, correct RTP paths, and high-quality voice for remote Jabber clients. This approach supports scalable remote collaboration, reduces call drops, and aligns with Cisco best practices for Mobile and Remote Access deployments, ensuring consistent and predictable performance for remote enterprise users.

Question146

During SIP trunk deployment across two CUCM clusters with a CUBE in between, users report intermittent one-way audio and occasional call drops. RTP analysis shows that packets are sent to incorrect IP addresses before CUBE has rewritten SDP. Which configuration MOST effectively resolves this issue?

A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Use SIP over UDP exclusively
C) Disable early media on CUBE
D) Enable symmetric RTP on CUBE

Answer: A

Explanation:

In multi-site SIP trunk deployments, particularly when CUCM communicates through a CUBE, intermittent one-way audio and call drops often stem from premature RTP transmission before CUBE has the opportunity to rewrite SDP. The SDP portion of SIP signaling carries critical media information, including codec selections, media port assignments, and IP address details that guide RTP packet routing. When RTP is sent prior to SDP rewriting, endpoints may attempt to transmit media to incorrect destinations, resulting in one-way audio or dropped calls.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly resolves this problem by delaying the transmission of the SDP from the INVITE to the 200 OK response. This delay ensures that CUBE anchors the media path and modifies the SDP information to reflect the correct media endpoints. Cisco best practices advocate the use of delayed-offer SDP for multi-site SIP trunk deployments because it guarantees correct alignment of signaling and media paths. By implementing delayed-offer SDP, administrators can prevent misrouted RTP packets, stabilize call quality, and reduce operational troubleshooting. The approach also ensures interoperability with a range of SIP endpoints and service providers, maintaining consistent voice quality across complex network topologies. Delayed-offer SDP provides predictable call behavior and aligns with enterprise requirements for reliable inter-site communication.

Option B, using SIP over UDP exclusively, ensures low-latency signaling but does not address the root cause of early RTP transmission. UDP guarantees fast delivery but does not influence the timing of RTP in relation to SDP rewriting. Consequently, one-way audio and dropped calls would persist even with SIP over UDP, as the issue stems from media path misalignment rather than signaling transport.

Option C, disabling early media on CUBE, affects call progress tones and ringback behavior but does not correct RTP packets being sent to incorrect IP addresses. Early media configuration only determines how audio is handled during call setup, so while it can improve perceived ringing or signaling tones, it does not resolve the underlying SDP timing problem.

Option D, enabling symmetric RTP on CUBE, is useful for NAT traversal and ensures that RTP streams are received and transmitted from the same source. However, symmetric RTP alone does not prevent early RTP transmission before SDP rewriting, and one-way audio issues may continue without delayed-offer SDP. Symmetric RTP complements proper SDP handling but is not a substitute.

By enabling delayed-offer SDP, the media and signaling paths are synchronized, preventing one-way audio and ensuring consistent call quality across multi-site SIP deployments. This configuration reduces operational complexity, improves reliability, and aligns with Cisco enterprise best practices. Proper SDP handling is critical in multi-cluster and NATed environments, as it anchors media paths effectively and ensures that all endpoints receive RTP correctly. Overall, delayed-offer SDP provides the most robust solution to address media misalignment, eliminates intermittent call drops, and supports scalable, high-quality inter-site voice communication.

Question147

Remote Jabber clients report intermittent registration failures and one-way audio when accessing CUCM via Expressway traversal zones. ICE candidate exchanges are failing intermittently, and traversal zones are operating near capacity. Which configuration MOST effectively mitigates this issue?

Answer: A

Explanation:

Remote Jabber clients rely on Expressway traversal zones to establish secure registration, signaling, and media paths with CUCM. Intermittent registration failures and one-way audio often occur when traversal zone resources are constrained and ICE (Interactive Connectivity Establishment) candidate negotiation fails. ICE enables endpoints behind NAT or firewall devices to select the best IP addresses and ports for RTP streams. When traversal zones are overloaded, ICE negotiation may fail, causing registration delays and misrouted media, which manifests as one-way audio.

Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses the core problem. Allocating additional CPU and memory ensures traversal zones can handle peak concurrent connections, reducing registration failures and media path disruptions. Full bidirectional ICE guarantees successful exchange of candidates between clients and the Expressway server, allowing RTP streams to traverse NAT and firewalls efficiently. Cisco best practices recommend sizing traversal zones according to the anticipated number of remote endpoints and enabling full ICE support to maintain reliable registration and media delivery. Continuous monitoring of traversal zone utilization ensures that administrators can scale resources proactively to accommodate peak loads.

Option B, requiring VPN connectivity, can bypass some NAT-related issues but introduces additional latency, complexity, and potential points of failure. VPNs do not address resource limitations or ICE negotiation failures within traversal zones and are not considered an optimal solution for scalable deployments.

Option C, reducing CUCM SIP registration timers, increases the frequency of registration retries but does not resolve the underlying resource constraint. Excessive retries under high load may worsen traversal zone performance rather than improving reliability.

Option D, enabling persistent XMPP connections, can improve messaging reliability and presence updates but does not impact ICE negotiation or traversal zone performance. Persistent XMPP connections do not influence media path establishment or registration success, so one-way audio issues persist without proper ICE support.

Implementing Option A ensures that remote Jabber clients can reliably register and establish correct RTP paths, providing high-quality voice services. This approach supports scalable remote collaboration, reduces call drops, and aligns with Cisco best practices for Mobile and Remote Access deployments. Correct traversal zone sizing combined with full ICE support delivers predictable performance, minimizes operational support incidents, and ensures that remote users experience seamless voice and video communication regardless of network conditions.

Question148

Voicemail users accessing Unity Connection via CUCM report several seconds of initial silence when playing messages. RTP analysis shows no packet loss, jitter, or network delay. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

The initial silence experienced during voicemail playback is most often caused by Voice Activity Detection (VAD), which suppresses RTP transmission during low-energy audio periods to conserve bandwidth in live calls. Voicemail messages frequently begin with soft audio or silence that VAD interprets as inactivity, delaying RTP transmission and causing perceived pauses at the start of messages.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, resolves this issue by allowing continuous RTP transmission regardless of audio energy levels. This ensures that all audio, including soft starts, is transmitted immediately, eliminating initial silence. Cisco best practices recommend disabling VAD for voicemail services to preserve message integrity and provide a consistent, high-quality user experience. While bandwidth utilization increases slightly, the trade-off is negligible compared to the improvement in message playback reliability. Removing VAD delays improves user satisfaction, reduces voicemail-related support calls, and ensures consistent behavior across endpoints and devices.

Option B, adjusting MWI extensions, affects visual indicators for new messages but has no impact on RTP transmission or playback timing. MWI modifications influence lamp behavior, not media delivery, so the initial silence remains unaddressed.

Option C, changing the Unity Connection codec to G.722, enhances audio fidelity but does not resolve silence caused by VAD. Codec selection affects sound quality but cannot correct delayed RTP transmission.

Option D, moving the voicemail pilot to a different partition, alters call routing but does not affect RTP or VAD behavior. Partition changes do not influence playback timing, so initial silence persists.

Disabling VAD ensures uninterrupted, immediate playback of voicemail messages, enhancing user experience and maintaining message integrity. This aligns with Cisco best practices, reduces support incidents, and ensures predictable performance for enterprise voicemail users. Proper configuration allows for reliable service without compromising call quality or operational efficiency.

Question149

During inter-site CUCM calls over SIP trunks through CUBE, users experience occasional call drops and one-way audio. RTP flows are observed being misdirected prior to SDP rewriting 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:

One-way audio and call drops in multi-site SIP trunk deployments are often caused by RTP packets being sent before CUBE completes SDP rewriting. SDP contains critical media information including codecs, IP addresses, and ports. If RTP is transmitted too early, it may be misdirected, causing one-way audio and call failures.

Option A, enabling delayed-offer SDP on CUCM SIP trunks, directly addresses the root cause. By delaying SDP from the INVITE to the 200 OK, CUBE can anchor media and modify SDP correctly before RTP flows are established. Cisco best practices emphasize delayed-offer SDP for multi-site deployments to align media and signaling paths, prevent early RTP transmission, and improve call stability. This configuration ensures interoperability across clusters, reduces troubleshooting complexity, and enhances user experience.

Option B, converting SIP trunks to SCCP, changes signaling protocol but does not prevent early RTP transmission. SCCP may improve endpoint interoperability but does not address media path misalignment.

Option C, using SIP over TCP, ensures reliable delivery of signaling but does not influence RTP timing relative to SDP rewriting. TCP alone cannot correct one-way audio caused by premature RTP.

Option D, disabling early media on CUBE, affects call progress tones but does not resolve misaligned RTP flows. Early media configuration impacts signaling rather than media path establishment.

Enabling delayed-offer SDP ensures proper media anchoring, predictable RTP delivery, and consistent call quality. This aligns with Cisco best practices, stabilizes inter-site communication, and provides a reliable user experience. Proper SDP handling reduces operational support requirements and supports complex, multi-cluster deployments.

Question150

Remote Jabber clients behind NAT report intermittent registration failures and one-way audio when connecting to CUCM via Expressway. ICE candidate exchanges often fail due to high traversal zone load. Which configuration MOST effectively mitigates these issues?

Answer: A

Explanation:

Intermittent registration failures and one-way audio in remote Jabber deployments are commonly caused by traversal zone resource limitations and failed ICE candidate negotiation. Traversal zones are responsible for secure registration, signaling, and media path establishment. ICE allows endpoints to determine optimal IP addresses and ports for RTP streams. Under high load, candidate exchanges can fail, resulting in registration failures and misrouted media.

Option A, increasing traversal zone capacity and enabling full bidirectional ICE support, directly addresses these problems. Providing sufficient CPU and memory allows traversal zones to handle peak concurrent connections, preventing registration failures. Bidirectional ICE ensures proper candidate exchange, enabling RTP streams to traverse NAT and firewalls successfully. Cisco best practices recommend scaling traversal zones according to the number of remote clients and enabling full ICE support to maintain reliable registration and media path establishment. Continuous monitoring ensures administrators can preemptively adjust resources, maintaining seamless connectivity and high-quality audio for remote users.

Option B, requiring VPN connectivity, introduces latency and complexity without resolving traversal zone resource constraints or ICE negotiation failures.

Option C, reducing CUCM SIP registration timers, increases retry frequency but does not resolve resource limitations and may worsen congestion.

Option D, enabling persistent XMPP connections, improves messaging reliability but does not affect ICE negotiation or traversal zone performance, leaving one-way audio issues unresolved.

Implementing Option A ensures reliable registration, correct RTP paths, and high-quality voice for remote Jabber clients. This solution supports scalable remote collaboration, reduces call drops, and aligns with Cisco best practices, ensuring consistent and predictable performance across enterprise deployments. Proper traversal zone sizing and ICE support provide seamless communication for remote users even during periods of peak load.

Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 10 Q136-150

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