Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 8 Q106-120

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Question106

A CUCM administrator observes that remote Jabber clients experience delayed call setup and intermittent audio issues through Mobile and Remote Access (MRA). Internal endpoints operate normally. RTP and signaling logs indicate inconsistent ICE candidate negotiation. Which configuration MOST effectively resolves this issue?

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

Answer: B

Explanation:

Remote Jabber clients using MRA rely on traversal zones between Expressway-C and Expressway-E to establish secure connectivity to CUCM. Delays in call setup and intermittent audio issues often stem from ICE (Interactive Connectivity Establishment) failures, which are critical for establishing optimal media paths through NAT or firewall boundaries. ICE ensures that endpoints discover the best candidate IP and port combinations for RTP traffic. Inconsistent ICE candidate negotiation can result in delayed call establishment, one-way audio, dropped calls, and poor overall quality.

Option B, ensuring traversal zones allow bidirectional UDP and TCP signaling with full ICE support, is the correct solution. Bidirectional signaling is essential for the exchange of ICE candidates, enabling endpoints to negotiate the most effective media path. Cisco best practices for MRA deployments stress the importance of enabling full ICE support for traversal zones. Full ICE support ensures that both signaling and media negotiation occur reliably, even in environments with complex NAT or firewall setups. By implementing this configuration, administrators can eliminate call setup delays, reduce one-way audio occurrences, and provide a consistent and high-quality collaboration experience for remote users.

Option A, enabling persistent XMPP connections, enhances messaging and presence continuity but does not influence ICE candidate negotiation or media path establishment. While persistent XMPP helps maintain session state, it does not resolve delayed call setup caused by NAT traversal issues.

Option C, disabling SIP normalization on Expressway, addresses potential SIP header inconsistencies but does not impact ICE candidate exchange or RTP path setup. SIP normalization primarily corrects protocol syntax issues, not media path reliability.

Option D, using SIP over TCP exclusively, ensures reliable signaling delivery but does not fully address ICE negotiation, which relies on UDP for effective media candidate exchange. TCP-only configurations cannot guarantee consistent media path selection, leaving call setup delays and one-way audio unresolved.

By configuring traversal zones for bidirectional UDP and TCP signaling with full ICE support, CUCM administrators can ensure consistent remote endpoint connectivity, reliable media flow, and a high-quality collaboration experience, fully adhering to Cisco MRA best practices.

Question107

During high utilization periods, remote endpoints fail to register intermittently through Mobile and Remote Access, while internal endpoints register successfully. Analysis indicates that traversal zones between Expressway-C and Expressway-E have reached maximum connection capacity. Which configuration MOST effectively resolves this problem?

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

Answer: A

Explanation:

Mobile and Remote Access relies on traversal zones to provide secure registration and media connectivity for remote endpoints. During periods of high utilization, if the traversal zone reaches its maximum connection limit, additional remote endpoints cannot register. Internal endpoints are unaffected because they do not rely on traversal zones for registration. Ensuring that traversal zones are sized appropriately to accommodate peak concurrent users is critical for maintaining high availability, user satisfaction, and uninterrupted collaboration services.

Option A, increasing the number of allowed traversal zone connections, is the correct solution. Cisco best practices recommend sizing traversal zones based on expected concurrent remote endpoints, including peak periods. By increasing the allowed number of connections, remote endpoints can register without failures, maintaining voice, video, and messaging functionality. Properly sized traversal zones also reduce call setup delays and ensure reliable presence and collaboration services. Implementing this change allows administrators to maintain consistent service levels, prevent registration failures, and optimize remote user experience during peak usage.

Option B, disabling traversal zones and requiring VPN access, introduces additional complexity and latency, counteracting the benefits of MRA. VPN infrastructure adds administrative overhead and does not address the underlying issue of traversal zone saturation.

Option C, reducing SIP timers, forces faster registration retries but does not solve the root cause of maximum connection limits. Rapid retries may increase signaling load, potentially exacerbating congestion within the traversal zones.

Option D, enabling persistent XMPP connections, improves session continuity for messaging and presence but does not increase traversal zone capacity or prevent registration failures during peak periods.

Increasing traversal zone connections ensures high availability, reliable registration, and uninterrupted remote collaboration, fully adhering to Cisco best practices for Mobile and Remote Access deployments.

Question108

In a multi-site CUCM deployment utilizing CUBE, users report one-way audio and dropped calls when using inter-site SIP trunks. RTP packets are sometimes sent to incorrect IP addresses before CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

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

Answer: A

Explanation:

One-way audio and dropped calls in multi-site deployments often result from RTP being transmitted before CUBE completes SDP rewriting. When RTP packets are sent prematurely, they may reach the wrong IP address or port, causing call quality degradation or call failure. Early media scenarios and NAT traversal issues exacerbate these problems. Proper SDP negotiation and media anchoring are essential for ensuring RTP reaches the correct endpoint and maintaining reliable communication.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP places SDP in the 200 OK response instead of the initial INVITE, allowing CUBE to anchor media, rewrite IP addresses and ports, and establish correct RTP paths before media transmission begins. Cisco best practices recommend delayed-offer SDP for multi-site SIP trunks that traverse CUBE. Implementing this ensures predictable RTP delivery, eliminates one-way audio, reduces call drops, and maintains consistent voice quality across sites.

Option B, converting SIP trunks to SCCP, changes signaling protocols but does not address the timing of SDP delivery or early RTP transmission. SCCP does not inherently resolve RTP misalignment or media path issues caused by early SDP.

Option C, using SIP over TCP, ensures signaling reliability but does not correct RTP misalignment caused by early SDP transmission. TCP alone cannot prevent one-way audio or dropped calls when SDP rewriting is necessary.

Option D, disabling early media, prevents call progress tones but does not solve the fundamental issue of RTP being sent to incorrect addresses. Early media affects signaling behavior but does not correct SDP timing or media anchoring issues.

Enabling delayed-offer SDP ensures proper media anchoring through CUBE, eliminates one-way audio, prevents call drops, and maintains high-quality inter-site SIP communication in line with Cisco best practices.

Question109

Users report several seconds of initial silence when playing voicemail from Unity Connection integrated with CUCM. RTP monitoring shows no packet loss or jitter. Which configuration MOST effectively resolves this issue?

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

Answer: A

Explanation:

Initial silence during voicemail playback is usually caused by Voice Activity Detection (VAD). VAD suppresses RTP during low-energy audio segments to conserve bandwidth during live calls. Voicemail messages often start with low-energy audio, which VAD interprets as silence, resulting in an initial delay in message playback. This creates a negative user experience, as users may perceive messages as incomplete or delayed.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, is the correct solution. Disabling VAD ensures continuous RTP transmission, including low-energy audio, eliminating the initial silence. Cisco best practices recommend disabling VAD for voicemail deployments to ensure uninterrupted message playback. This configuration allows users to hear messages immediately, improving usability and satisfaction. Bandwidth impact is minimal since voicemail RTP traffic is lower compared to live call traffic.

Option B, adjusting MWI extensions, only affects indicator lamp signaling and does not impact media playback or RTP flow.

Option C, changing the codec to G.722, may improve audio quality but does not address the silence caused by VAD.

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

Disabling VAD ensures smooth voicemail playback from the beginning of messages, improves user experience, and aligns with Cisco best practices for CUCM and Unity Connection integration.

Question110

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

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

Answer: A

Explanation:

One-way audio occurs when RTP packets are transmitted before CUBE completes SDP rewriting, causing them to be sent to incorrect IP addresses or ports. This problem is compounded in NAT and multi-site environments. Accurate SDP negotiation and media anchoring are critical to ensure RTP reaches the correct endpoint, preventing one-way communication and dropped calls.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP places SDP in the 200 OK response rather than the initial INVITE. This allows CUBE to anchor media, rewrite addresses and ports, and establish correct RTP paths before media flow begins. Cisco best practices recommend delayed-offer SDP for multi-site SIP deployments with CUBE to prevent one-way audio, maintain call quality, and ensure reliable communication.

Option B, using SIP over UDP, does not address SDP timing or media path misalignment. While UDP is necessary for RTP transport, changing transport does not resolve early SDP or address rewriting issues.

Option C, disabling early media, prevents call progress tones but does not correct RTP misrouting. Early media affects signaling behavior but does not solve SDP timing issues.

Option D, enabling symmetric RTP, helps with NAT traversal but does not correct SDP timing or address misalignment causing one-way audio. Symmetric RTP alone cannot correct media delivery issues without proper SDP anchoring.

Implementing delayed-offer SDP ensures proper media anchoring through CUBE, eliminates one-way audio, prevents dropped calls, and maintains reliable multi-site SIP communication consistent with Cisco best practices.

Question111

Remote Jabber users report intermittent call drops and one-way audio during peak hours through Mobile and Remote Access (MRA). Internal endpoints function normally. Logs indicate traversal zone capacity saturation and inconsistent ICE candidate negotiation. Which configuration MOST effectively resolves this issue?

A) Increase the number of allowed traversal zone connections and ensure bidirectional UDP/TCP signaling with full ICE support
B) Enable persistent XMPP connections between Expressway-C and CUCM
C) Disable SIP normalization on Expressway
D) Use SIP over TCP exclusively for remote clients

Answer: A

Explanation:

Mobile and Remote Access relies on Expressway-C and Expressway-E traversal zones to provide secure registration and media connectivity for remote Jabber clients. Peak-hour call drops and one-way audio often result from two primary factors: traversal zone capacity saturation and failed ICE (Interactive Connectivity Establishment) negotiation. Traversal zone saturation occurs when the number of concurrent remote endpoints exceeds the configured limit, preventing new registrations and affecting ongoing calls. ICE negotiation is critical for establishing RTP media paths through NAT or firewall boundaries. If ICE candidates are not exchanged correctly due to bidirectional signaling limitations, call setup may be delayed and media flow disrupted.

Option A, increasing the number of allowed traversal zone connections while ensuring bidirectional UDP/TCP signaling with full ICE support, is the correct solution. Increasing traversal zone capacity directly addresses the registration failures by allowing more concurrent endpoints to connect during peak usage. Ensuring bidirectional UDP/TCP signaling enables ICE candidates to be exchanged properly, establishing optimal RTP paths. Full ICE support ensures that endpoints behind NAT or firewall restrictions can determine the best IP address and port combination for media transmission, eliminating one-way audio issues. Cisco best practices for MRA deployments emphasize the importance of both proper sizing of traversal zones and complete ICE support to maintain high availability and reliable media flow. This configuration ensures that remote endpoints experience seamless voice and video communication, reduces call setup delays, and prevents intermittent call drops.

Option B, enabling persistent XMPP connections, improves messaging and presence continuity but does not directly address ICE negotiation or traversal zone saturation. While beneficial for session persistence, it cannot prevent media path failures or registration issues during peak periods.

Option C, disabling SIP normalization, addresses potential SIP header mismatches but does not influence ICE negotiation or traversal zone capacity. Normalization primarily corrects signaling syntax discrepancies, which are unrelated to RTP path establishment.

Option D, using SIP over TCP exclusively, guarantees reliable signaling but cannot substitute for ICE candidate negotiation, which requires UDP for proper media path determination. TCP-only configurations may inadvertently introduce additional latency without resolving one-way audio or call drops caused by traversal zone or ICE limitations.

By implementing option A, administrators ensure robust remote client connectivity, reliable call setup, consistent media flow, and adherence to Cisco best practices for Mobile and Remote Access deployments.

Question112

During high-load periods, remote endpoints fail to register through MRA while internal endpoints register successfully. Analysis indicates that traversal zones between Expressway-C and Expressway-E have reached full utilization. Which configuration MOST effectively resolves this problem?

Answer: A

Explanation:

MRA traversal zones provide the secure connection path for remote endpoints to register and communicate with CUCM. When traversal zones are fully utilized, additional remote endpoints are unable to register, leading to service disruptions. Internal endpoints do not rely on traversal zones, which explains why they continue to register normally. Ensuring adequate traversal zone capacity is essential for high availability and consistent service delivery during peak periods.

Option A, increasing the number of allowed traversal zone connections, is the correct solution. Cisco best practices recommend sizing traversal zones according to expected concurrent remote endpoints, including anticipated peak periods. Increasing capacity allows more remote endpoints to register without failure, maintaining voice, video, and messaging functionality. Proper sizing reduces the risk of registration delays, call setup failures, and user dissatisfaction. Administrators can monitor traversal zone utilization trends to adjust capacity dynamically as the number of remote users changes over time.

Option B, disabling traversal zones and requiring VPN access, adds complexity and latency and negates the benefits of MRA. VPN infrastructure introduces administrative overhead and does not address the root cause of traversal zone saturation.

Option C, reducing SIP timers, forces endpoints to retry registrations more frequently but does not alleviate capacity limitations. Increased retry traffic may exacerbate the problem rather than resolve it.

Option D, enabling persistent XMPP connections, improves messaging and presence continuity but does not increase traversal zone capacity or prevent registration failures during peak periods.

Increasing traversal zone connections ensures high availability, reliable registration, and uninterrupted remote collaboration services, aligning with Cisco best practices for MRA deployments.

Question113

A multi-site CUCM deployment using CUBE reports one-way audio and intermittent call drops for inter-site SIP trunk calls. RTP packets are sometimes sent to incorrect addresses before CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

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

Answer: A

Explanation:

One-way audio and dropped calls are often caused by RTP being transmitted before SDP is correctly rewritten by CUBE. Early RTP transmission can send media packets to incorrect IP addresses or ports, causing audio failures. Proper SDP negotiation and media anchoring are critical to ensuring RTP reaches the correct endpoint and maintaining reliable communication.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP places the SDP in the 200 OK response instead of the initial INVITE, allowing CUBE to anchor media, rewrite IP addresses and ports, and establish correct RTP paths before media transmission. Cisco best practices recommend delayed-offer SDP for multi-site SIP trunks traversing CUBE. Implementing this configuration ensures predictable RTP delivery, eliminates one-way audio, reduces dropped calls, and maintains consistent voice quality across sites. Proper SDP handling through CUBE is critical in multi-site deployments, especially where NAT traversal or firewall restrictions exist.

Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not address the timing of SDP delivery or early RTP transmission. SCCP cannot inherently correct RTP misalignment caused by early SDP.

Option C, using SIP over TCP, ensures reliable signaling but does not resolve misaligned RTP packets caused by early SDP transmission. TCP does not correct SDP timing issues.

Option D, disabling early media, prevents call progress tones but does not correct RTP being sent to incorrect addresses. Early media configuration affects signaling behavior but does not address SDP or media anchoring problems.

Question114

Users report several seconds of initial silence when listening to voicemail from Unity Connection integrated with CUCM. RTP monitoring shows no packet loss or jitter. 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 segments to conserve bandwidth. Voicemail messages often start with low-energy audio that VAD interprets as silence, resulting in delayed playback. This negatively impacts user experience, as messages appear to start late or partially, reducing usability.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, is the correct solution. Disabling VAD ensures continuous RTP transmission, including low-energy audio, eliminating initial silence. Cisco best practices recommend disabling VAD for voicemail deployments to ensure uninterrupted playback. Users experience immediate message delivery, improving usability and satisfaction. The impact on bandwidth is minimal since voicemail RTP traffic is significantly lower than live call traffic.

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

Option C, changing the codec to G.722, may improve audio fidelity but does not address silence caused by VAD.

Option D, moving the voicemail pilot to a different partition, affects routing but does not influence RTP delivery or message playback.

Disabling VAD ensures smooth voicemail playback, improves user experience, and aligns with Cisco best practices for CUCM and Unity Connection integration.

Question115

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

Answer: A

Explanation:

One-way audio occurs when RTP packets are transmitted before CUBE completes SDP rewriting, causing packets to be sent to incorrect IP addresses or ports. This issue is common in multi-site and NAT traversal scenarios. Accurate SDP negotiation and media anchoring are essential to ensure RTP reaches the intended endpoint, preventing one-way audio and dropped calls.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP includes SDP in the 200 OK response instead of the initial INVITE. This allows CUBE to anchor media, rewrite addresses and ports, and establish correct RTP paths before media transmission begins. Cisco best practices recommend delayed-offer SDP for multi-site SIP deployments using CUBE to prevent one-way audio, maintain call quality, and ensure reliable communication.

Option B, using SIP over UDP, does not address SDP timing or media path misalignment. While UDP is necessary for RTP transport, changing transport does not resolve early SDP issues or incorrect media paths.

Option C, disabling early media, prevents call progress tones but does not correct RTP path misalignment. Early media affects signaling but does not solve SDP rewriting or media anchoring problems.

Option D, enabling symmetric RTP, assists with NAT traversal but does not resolve SDP timing or address misalignment causing one-way audio. Symmetric RTP alone cannot correct media delivery issues without proper SDP anchoring.

Question116

Remote Jabber clients report call setup delays and intermittent audio issues through Mobile and Remote Access (MRA), while internal endpoints work normally. RTP and signaling logs indicate ICE candidate exchange failures and traversal zone saturation. Which configuration MOST effectively resolves this problem?

A) Increase traversal zone capacity and ensure bidirectional UDP/TCP signaling with full ICE support
B) Enable persistent XMPP connections between Expressway-C and CUCM
C) Disable SIP normalization on Expressway
D) Use SIP over TCP exclusively for remote clients

Answer: A

Explanation:

Mobile and Remote Access relies on Expressway-C and Expressway-E traversal zones for secure remote client connectivity. Call setup delays and intermittent audio often result from traversal zone saturation and failed ICE (Interactive Connectivity Establishment) negotiation. ICE enables endpoints to determine the optimal IP and port for RTP media, which is essential when clients are behind NAT or firewalls. Traversal zone saturation occurs when concurrent connections exceed the configured limit, preventing additional endpoints from registering or successfully negotiating media paths.

Option A, increasing traversal zone capacity and ensuring bidirectional UDP/TCP signaling with full ICE support, directly addresses these issues. By increasing traversal zone capacity, more concurrent remote endpoints can register, eliminating registration failures during peak periods. Bidirectional UDP/TCP signaling ensures that ICE candidates can be exchanged correctly, allowing endpoints to establish the optimal media path. Full ICE support guarantees reliable connectivity even in complex NAT and firewall scenarios. Cisco best practices emphasize proper sizing of traversal zones and enabling full ICE support to prevent call setup delays, one-way audio, and dropped calls.

Option B, enabling persistent XMPP connections, improves messaging and presence continuity but does not resolve ICE failures or traversal zone saturation. While beneficial for session persistence, it cannot prevent media path failures or call setup delays caused by capacity limitations.

Option C, disabling SIP normalization, addresses potential SIP header issues but does not impact ICE negotiation or traversal zone capacity. SIP normalization corrects syntax discrepancies but does not ensure RTP delivery to the correct destination.

Option D, using SIP over TCP exclusively, ensures reliable signaling but does not address ICE negotiation or media path optimization, which relies on UDP. TCP-only configurations may introduce latency without resolving one-way audio or call setup issues.

Implementing option A ensures reliable remote client registration, proper media path establishment, and adherence to Cisco MRA best practices, delivering a seamless collaboration experience.

Question117

During high-utilization periods, remote endpoints fail to register through MRA, while internal endpoints register successfully. Traversal zone logs indicate maximum connection capacity is reached. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Mobile and Remote Access relies on traversal zones for secure remote endpoint registration. When traversal zones reach maximum capacity, additional remote endpoints cannot register, causing intermittent connectivity issues. Internal endpoints bypass traversal zones, explaining why they continue to register normally. Ensuring traversal zones are sized appropriately is essential for high availability and reliable service delivery.

Option A, increasing the number of allowed traversal zone connections, is the correct solution. Cisco best practices recommend sizing traversal zones based on anticipated peak concurrent remote endpoints. Increasing capacity allows more remote endpoints to register successfully, maintaining voice, video, and messaging functionality. Proper sizing reduces call setup failures, registration delays, and user frustration. Administrators can monitor traversal zone usage trends and adjust capacity dynamically to accommodate fluctuating remote endpoint loads.

Option B, disabling traversal zones and requiring VPN access, adds complexity, latency, and administrative overhead while negating the benefits of MRA. VPN infrastructure does not address traversal zone saturation and may introduce additional points of failure.

Option C, reducing SIP timers, forces faster retries but does not resolve the root cause of capacity limits. Excessive retries may increase signaling load, potentially worsening congestion within traversal zones.

Option D, enabling persistent XMPP connections, improves messaging and presence continuity but does not increase traversal zone capacity or prevent registration failures during high utilization.

Increasing traversal zone connections ensures high availability, reliable registration, and uninterrupted collaboration services, fully aligning with Cisco best practices for MRA deployments.

Question118

In a multi-site CUCM deployment utilizing CUBE, inter-site SIP trunk calls experience one-way audio and dropped calls. RTP packets are transmitted to incorrect addresses before CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

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

Answer: A

Explanation:

One-way audio and call drops in multi-site deployments are often caused by RTP being sent before CUBE can rewrite SDP. Early RTP transmission to incorrect IP addresses or ports results in audio failure and dropped calls. Correct SDP negotiation and media anchoring are essential for reliable communication.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP moves SDP information into the 200 OK response instead of the initial INVITE. This allows CUBE to anchor media, rewrite IP addresses and ports, and establish correct RTP paths before media transmission begins. Cisco best practices for multi-site SIP trunks traversing CUBE recommend delayed-offer SDP to prevent one-way audio and dropped calls. Proper SDP handling ensures predictable RTP delivery, consistent audio quality, and reliable inter-site communication.

Option B, converting SIP trunks to SCCP, changes signaling protocol but does not address early RTP transmission or SDP rewriting issues. SCCP does not inherently correct media misalignment caused by early SDP.

Option C, using SIP over TCP, improves signaling reliability but does not correct RTP misalignment caused by early SDP. TCP alone cannot resolve call drops or one-way audio resulting from improper media anchoring.

Option D, disabling early media, prevents call progress tones but does not correct RTP being sent to incorrect addresses. Early media configuration affects signaling but does not resolve SDP timing issues or media path alignment.

Enabling delayed-offer SDP ensures proper media anchoring through CUBE, eliminates one-way audio, prevents dropped calls, and maintains high-quality inter-site SIP communication per Cisco best practices.

Question119

Users experience several seconds of initial silence when playing voicemail from Unity Connection integrated with CUCM. RTP analysis indicates no packet loss or jitter. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Initial silence in voicemail playback is frequently caused by Voice Activity Detection (VAD). VAD suppresses RTP transmission during low-energy audio segments to conserve bandwidth during live calls. Voicemail messages often begin with low-energy audio that VAD interprets as silence, causing a delay at the start of message playback. This affects user experience as messages appear to start late or partially.

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

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

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

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

Disabling VAD ensures smooth and immediate voicemail playback, enhancing user experience and aligning with Cisco best practices for CUCM and Unity Connection integration.

Question120

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

Answer: A

Explanation:

One-way audio occurs when RTP packets are transmitted before CUBE completes SDP rewriting, causing packets to be sent to incorrect IP addresses or ports. Multi-site deployments and NAT traversal exacerbate the problem. Accurate SDP negotiation and media anchoring are crucial to ensure RTP reaches the intended destination, preventing one-way communication and dropped calls.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP places SDP in the 200 OK response instead of the initial INVITE, allowing CUBE to anchor media, rewrite addresses and ports, and establish correct RTP paths before media transmission begins. Cisco best practices recommend delayed-offer SDP for multi-site SIP deployments using CUBE to prevent one-way audio, maintain call quality, and ensure reliable communication.

Option B, using SIP over UDP, does not address SDP timing or media path misalignment. While UDP is required for RTP, changing transport does not correct early SDP or incorrect media paths.

Option C, disabling early media, prevents call progress tones but does not resolve RTP misalignment or address rewriting issues.

Option D, enabling symmetric RTP, assists with NAT traversal but does not correct SDP timing or media path misalignment causing one-way audio. Symmetric RTP alone cannot resolve the underlying media delivery problem without proper SDP anchoring.

One-way audio in multi-site Cisco Unified Communications deployments is a common issue that arises when RTP media streams are sent before the Session Description Protocol (SDP) has been properly rewritten by the Cisco Unified Border Element (CUBE). The SDP carries critical information for media path establishment, including IP addresses, port numbers, and codec details. If RTP begins flowing before CUBE has anchored the media and updated the SDP, packets may be sent to incorrect endpoints, causing audio to be heard on only one side of the call. Multi-site deployments and NAT traversal scenarios make this problem even more pronounced because RTP may traverse multiple networks, firewalls, or NAT devices before reaching the destination, increasing the likelihood of misdelivery. Accurate SDP negotiation and proper media anchoring are therefore essential to ensure bi-directional audio and prevent dropped or incomplete calls.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the recommended solution. In a delayed-offer configuration, CUCM does not include SDP in the initial SIP INVITE but instead sends it in the 200 OK response after the signaling path has been fully processed. This ensures that CUBE can anchor the media correctly, rewrite IP addresses and port numbers, and establish proper RTP paths before any audio is transmitted. By controlling the timing of SDP, delayed-offer SDP prevents early RTP from being sent to unintended destinations, eliminating one-way audio. Cisco best practices for multi-site SIP deployments consistently recommend delayed-offer SDP when CUBE is deployed and early media or complex routing scenarios could result in media misalignment.

The use of delayed-offer SDP also ensures predictable and deterministic media flows, which is particularly important in multi-site deployments. When calls traverse multiple locations, each with separate WAN links, firewalls, and potentially NAT devices, the media path must be fully controlled to avoid errors in RTP delivery. By delaying the SDP until the 200 OK response, administrators ensure that the media path aligns with the signaling path, simplifying troubleshooting and improving the reliability of call flows. This alignment reduces operational complexity and helps IT teams identify and resolve issues quickly because the media flows are consistent and match the signaling path.

Option B, using SIP over UDP instead of TCP, does not address the root cause of one-way audio. While UDP is typically used for RTP because it provides low-latency, connectionless transport, changing the SIP signaling transport protocol does not resolve issues with SDP timing or media path misalignment. Early RTP could still be sent to incorrect IP addresses or ports if SDP is included in the initial INVITE, making UDP versus TCP irrelevant to the actual problem. Therefore, modifying the transport protocol does not prevent one-way audio caused by premature media transmission.

Option C, disabling early media on CUBE, prevents call progress tones, announcements, or ringback signals from being transmitted before the call is answered. While this may eliminate audible artifacts or pre-answer audio, it does not address the underlying SDP timing issue that causes RTP to be sent to incorrect addresses. Disabling early media only addresses a symptom of the problem and may negatively impact the caller experience by removing expected audio feedback. Additionally, even after early media is disabled, RTP could still flow incorrectly if the SDP is offered too early in the call setup process. Therefore, disabling early media is not a comprehensive solution.

Option D, enabling symmetric RTP on CUBE, is primarily used to address NAT traversal or asymmetrical routing issues by ensuring RTP is sent back to the IP address and port from which it was received. While symmetric RTP can help in certain topologies, it does not correct the early SDP problem that causes RTP to be transmitted to the wrong destination initially. One-way audio occurs because media is flowing before the media path has been anchored and the SDP has been rewritten; symmetric RTP does not prevent this initial misdelivery. Symmetric RTP is useful for complementary NAT solutions but cannot substitute for correct SDP timing.

Implementing delayed-offer SDP ensures that media anchoring occurs correctly and that RTP flows only after the call setup and SDP negotiation have been processed. This configuration guarantees bi-directional audio, prevents one-way audio issues, and ensures that calls are delivered with high quality across all sites. It also supports interoperability with service providers, cloud SIP trunks, and third-party SIP devices by ensuring that SDP is transmitted only after signaling has been completed, reducing the likelihood of codec mismatches, unsupported media, or misrouted RTP.

From an operational perspective, delayed-offer SDP simplifies troubleshooting and monitoring. With a predictable signaling and media sequence, network administrators can easily trace RTP flows, capture packets for analysis, and verify media alignment. This predictable behavior reduces the effort required to diagnose audio problems and allows proactive management of call quality. Delayed-offer SDP also supports proper application of media policies, QoS mechanisms, and call routing rules, ensuring that all voice traffic complies with enterprise standards and operates efficiently.

It ensures correct media anchoring, proper RTP path establishment, and reliable bi-directional audio. Other options, including changing the SIP transport protocol, disabling early media, or enabling symmetric RTP, do not address the SDP timing issue and cannot reliably eliminate one-way audio. Delayed-offer SDP provides a robust, standards-aligned solution that guarantees call quality, operational predictability, and adherence to Cisco best practices for multi-site SIP deployments.

Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 8 Q106-120

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