Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 7 Q91-105

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Question91

A CUCM administrator is troubleshooting delayed call setup and intermittent call failures for remote Jabber clients using Mobile and Remote Access (MRA). Internal endpoints work without issues. Analysis shows ICE candidates are inconsistently applied, causing NAT traversal problems. 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 rely on Expressway-C and Expressway-E traversal zones to securely connect to CUCM. When call setup is delayed or calls fail intermittently while internal endpoints work normally, the issue often lies in ICE (Interactive Connectivity Establishment) candidate negotiation. ICE is essential for establishing the optimal media path for endpoints behind NAT. Without proper ICE negotiation, RTP may not flow correctly, leading to one-way audio, dropped calls, or call setup delays.

Option B, ensuring traversal zones allow bidirectional UDP and TCP signaling with full ICE support, is the correct solution. Bidirectional signaling ensures that both CUCM and remote clients can exchange ICE candidates, enabling proper media path establishment. Cisco best practices for MRA deployments emphasize full ICE support to handle NAT traversal, firewall constraints, and remote media negotiation. This configuration allows remote Jabber clients to establish reliable RTP paths to internal endpoints, reduces call setup time, prevents call drops, and ensures consistent user experience. Proper ICE handling also supports video calls, content sharing, and other real-time collaboration features.

Option A, enabling persistent XMPP connections, helps maintain real-time presence and messaging updates but does not solve media path establishment or NAT traversal issues. While persistent XMPP improves session continuity, it does not address delayed media negotiation or call setup failures.

Option C, disabling SIP normalization, corrects signaling header inconsistencies but does not affect ICE negotiation or media path establishment. Normalization primarily fixes protocol syntax mismatches and has minimal impact on RTP flow or NAT traversal.

Option D, using SIP over TCP exclusively, ensures reliable signaling but does not solve UDP-dependent ICE candidate exchange required for RTP media flow. SIP over TCP alone cannot establish proper media paths through NAT, so one-way audio or call setup delays may persist.

By configuring traversal zones for bidirectional UDP and TCP signaling with full ICE support, remote Jabber clients can reliably establish media sessions, call setup delays are minimized, call drops are eliminated, and the deployment aligns with Cisco best practices for Mobile and Remote Access environments.

Question92

During peak hours, a CUCM administrator observes that remote endpoints fail to register intermittently while internal endpoints register successfully. Logs indicate that traversal zones between Expressway-C and Expressway-E are reaching maximum 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 (MRA) relies on Expressway-C and Expressway-E traversal zones to provide secure registration and connectivity for remote endpoints. During peak hours, if traversal zones reach their maximum session capacity, additional remote endpoints cannot register, leading to intermittent registration failures. Internal endpoints bypass traversal zones, explaining why they register without issues. Ensuring high availability for remote endpoints is critical to maintain reliable communication, presence, and collaboration services across the enterprise.

Option A, increasing the number of allowed traversal zone connections, is the correct solution. Cisco best practices recommend sizing traversal zones based on the expected number of concurrent remote endpoints. By allowing more simultaneous connections, the system can handle peak loads without registration failures. This configuration maintains high availability, ensures consistent remote endpoint connectivity, and reduces support tickets related to failed registrations. Properly sized traversal zones also improve call setup times and maintain user satisfaction during high-demand periods.

Option B, disabling traversal zones and requiring VPN, adds unnecessary complexity and latency while defeating the purpose of MRA. MRA is designed to eliminate the need for VPN for secure remote access. Removing traversal zones does not resolve capacity limitations and introduces new connectivity challenges.

Option C, reducing SIP timers, forces faster registration retries but does not address the fundamental issue of traversal zone capacity. Faster retries may increase signaling load and exacerbate congestion, potentially causing more registration failures.

Option D, enabling persistent XMPP connections, supports presence and messaging continuity but does not increase traversal zone registration capacity. Persistent XMPP helps maintain session state but does not solve peak-hour registration issues.

Increasing the allowed traversal zone connections ensures remote endpoints can register reliably, maintains high availability, reduces user impact during peak usage, and aligns with Cisco best practices for MRA deployments, addressing the root cause of intermittent registration failures effectively.

Question93

A multi-site CUCM deployment uses SIP trunks to CUBE for inter-site communications. Remote sites report one-way audio and occasional call drops. RTP packets are sent to incorrect IP addresses before CUBE rewrites SDP. Which configuration MOST effectively resolves this issue?

A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Convert SIP trunks to SCCP 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 SIP deployments often result from improper SDP handling and early RTP transmission. When RTP packets are sent before CUBE completes SDP rewriting, the media path is incorrect, causing one side to receive no audio. Early media scenarios, NAT traversal, and multi-site deployments exacerbate the problem. Accurate SDP negotiation and media anchoring are crucial to ensure reliable RTP delivery and maintain call quality.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP includes SDP in the 200 OK response instead of the initial INVITE, allowing CUBE to anchor media and rewrite IP addresses and ports correctly before RTP transmission begins. Cisco best practices for multi-site deployments recommend delayed-offer SDP for trunks traversing CUBE to prevent one-way audio and maintain predictable media flow. Implementing delayed-offer SDP ensures reliable audio, reduces dropped calls, and provides consistent inter-site communications.

Option B, converting SIP trunks to SCCP, does not address the SDP timing issue. SCCP may have different signaling behavior, but premature RTP transmission before SDP rewriting remains unresolved. Converting trunks adds complexity without resolving the underlying media problem.

Option C, using SIP over TCP, ensures reliable signaling delivery but does not correct RTP path misalignment caused by early SDP transmission. TCP alone cannot address the root cause of one-way audio.

Option D, disabling early media, prevents call progress tones but does not resolve RTP being sent to incorrect addresses. Early media settings affect signaling behavior but not SDP timing and media anchoring.

By enabling delayed-offer SDP, administrators ensure proper media anchoring through CUBE, eliminate one-way audio, prevent call drops, and maintain high-quality inter-site SIP communication consistent with Cisco best practices.

Question94

Users report several seconds of initial silence when listening to voicemail messages from Unity Connection integrated with CUCM. RTP monitoring indicates no packet loss or jitter. Which configuration MOST effectively resolves this problem?

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 in voicemail playback is commonly caused by Voice Activity Detection (VAD). VAD suppresses RTP during low-energy audio to conserve bandwidth during live calls. Voicemail messages often start with low-energy segments, which VAD interprets as silence, resulting in a delay at the beginning of playback. This negatively impacts user experience, as users may perceive messages as incomplete or delayed, reducing satisfaction and efficiency.

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

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

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

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 voicemail playback from the start of messages, eliminates confusion caused by initial silence, and adheres to Cisco best practices for CUCM and Unity Connection deployments.

Question95

Remote users report intermittent one-way audio when calling internal endpoints through a 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 is transmitted before CUBE rewrites SDP, causing packets to be sent to incorrect IP addresses or ports. Early media scenarios, NAT traversal, and multi-site deployments exacerbate the problem. Proper SDP negotiation and media anchoring are essential to ensure that RTP reaches the correct endpoints, 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 ensures SDP is included in the 200 OK response rather than the initial INVITE. This allows CUBE to anchor media, rewrite IP addresses and ports, and establish the correct RTP path before media flows. 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 transport, switching protocols does not solve early media or SDP rewrite issues.

Option C, disabling early media, prevents call progress tones but does not resolve RTP being sent to incorrect addresses. It treats symptoms rather than addressing the underlying media path problem.

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

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

Question96

A CUCM administrator observes that during peak hours, multiple remote endpoints fail to register via Mobile and Remote Access (MRA), while internal endpoints register without issues. Logs indicate that traversal zones between Expressway-C and Expressway-E are fully utilized. Which configuration MOST effectively resolves this problem?

A) Increase the number of allowed traversal zone connections
B) Disable traversal zones and require VPN 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 Expressway-C and Expressway-E traversal zones to provide secure registration and connectivity for remote endpoints. During peak usage, traversal zones may reach their maximum connection limit, preventing additional remote endpoints from registering. Internal endpoints bypass traversal zones, explaining why they register normally. Ensuring high availability and reliability for remote endpoints is essential to maintain uninterrupted collaboration services, including voice, video, messaging, and presence.

Option A, increasing the number of allowed traversal zone connections, is the correct solution. Cisco best practices recommend sizing traversal zones based on expected simultaneous remote endpoints. By increasing the number of concurrent connections, the system can handle peak-hour demand without registration failures. This approach maintains high availability, ensures reliable access for remote users, and reduces support incidents related to failed registrations. Proper traversal zone sizing also improves call setup times and ensures consistent user experience during peak usage periods.

Option B, disabling traversal zones and requiring VPN access, adds complexity, latency, and security considerations. Mobile and Remote Access is designed to provide secure connectivity without the need for VPN; removing traversal zones negates MRA benefits and does not address the underlying capacity problem.

Option C, reducing SIP timers, forces faster registration retries but does not solve the root cause of capacity exhaustion. Rapid retries may increase signaling traffic, exacerbating traversal zone congestion and potentially causing more registration failures.

Option D, enabling persistent XMPP connections, maintains presence and messaging continuity but does not increase traversal zone registration capacity. Persistent XMPP helps keep sessions alive but does not resolve peak-hour registration failures.

Increasing traversal zone connections ensures reliable registration, maintains high availability during peak usage, and aligns with Cisco best practices for MRA, directly addressing the root cause of intermittent remote registration failures.

Question97

A CUCM administrator notices that remote Jabber clients experience delayed call setup and intermittent media failures, whereas internal endpoints function normally. Analysis indicates inconsistent ICE candidate application, leading to NAT traversal issues. Which configuration MOST effectively resolves the problem?

Answer: B

Explanation:

Remote Jabber clients use Mobile and Remote Access to securely communicate through Expressway-C and Expressway-E. Delayed call setup and intermittent media failures often result from incomplete ICE (Interactive Connectivity Establishment) negotiation. ICE determines the optimal path for RTP media between endpoints behind NAT. If traversal zones do not support bidirectional UDP and TCP signaling, ICE candidates may not be exchanged properly, leading to call setup delays, one-way audio, or dropped calls.

Option B, ensuring traversal zones allow bidirectional UDP and TCP signaling with full ICE support, is the correct solution. This configuration guarantees that ICE candidates are correctly exchanged, enabling remote endpoints to establish optimal media paths. Cisco best practices emphasize full ICE support for MRA deployments to handle NAT traversal, firewall constraints, and remote media negotiation. With proper ICE handling, remote Jabber clients experience faster call setup, reliable media flow, and improved user experience. Video calls and content sharing also benefit from consistent ICE negotiation, ensuring a high-quality collaboration environment.

Option A, enabling persistent XMPP connections, improves presence and messaging updates but does not directly impact media path establishment or NAT traversal. While persistent XMPP maintains session continuity, it does not address delayed call setup or media failures.

Option C, disabling SIP normalization, corrects header inconsistencies but does not affect ICE negotiation or RTP flow. Normalization primarily addresses protocol syntax mismatches, not media path establishment.

Option D, using SIP over TCP exclusively, ensures reliable signaling delivery but does not address UDP-dependent ICE candidate exchange needed for RTP. TCP-only signaling cannot fully support ICE negotiation, leaving media path issues unresolved.

By configuring traversal zones for bidirectional UDP and TCP signaling with full ICE support, administrators can resolve delayed call setup and intermittent media failures, provide reliable remote access, and adhere to Cisco best practices for Mobile and Remote Access deployments.

Question98

During a multi-site CUCM deployment with CUBE, users report one-way audio and occasional 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 in multi-site deployments are frequently caused by RTP packets being sent before CUBE completes SDP rewriting. Early media scenarios, NAT traversal, and multi-site configurations exacerbate these issues. Accurate SDP negotiation and media anchoring are crucial for ensuring correct RTP delivery and maintaining call quality. When RTP is sent prematurely, the media path may be misaligned, causing one-way audio and dropped calls.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP includes the SDP in the 200 OK response instead of the initial INVITE. This allows CUBE to anchor media, rewrite IP addresses and ports, and establish proper RTP paths before media transmission begins. Cisco best practices recommend delayed-offer SDP for SIP trunks traversing CUBE in multi-site deployments to prevent one-way audio, reduce dropped calls, and ensure predictable media flow.

Option B, converting SIP trunks to SCCP, does not address the SDP timing issue. SCCP changes the signaling protocol but does not correct the problem of early RTP transmission before SDP rewriting. Converting trunks introduces complexity without resolving the underlying media problem.

Option C, using SIP over TCP, ensures reliable signaling but does not correct RTP misalignment caused by premature SDP transmission. TCP alone cannot prevent one-way audio or dropped calls resulting from early media issues.

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

Enabling delayed-offer SDP ensures proper media anchoring through CUBE, resolves one-way audio, prevents call drops, and maintains high-quality inter-site SIP communication, fully adhering to Cisco best practices.

Question99

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

Answer: A

Explanation:

Initial silence in voicemail playback is typically caused by Voice Activity Detection (VAD). VAD suppresses RTP during low-energy audio segments to conserve bandwidth during live calls. Voicemail messages often begin with low-energy audio, which VAD interprets as silence, delaying playback. This negatively impacts user experience, as users may perceive messages as incomplete or delayed, reducing efficiency and satisfaction.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, is the correct solution. Disabling VAD allows continuous RTP transmission, including low-energy audio, eliminating the initial silence. Cisco best practices for CUCM and Unity Connection integration recommend disabling VAD for voicemail deployments to ensure uninterrupted message playback. Users immediately hear the content of voicemail messages, improving usability and reducing confusion. Bandwidth impact is minimal, as voicemail RTP traffic is small relative to live call traffic.

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

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

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

Disabling VAD ensures smooth voicemail playback from the start of messages, enhances user satisfaction, and aligns with Cisco best practices for CUCM and Unity Connection.

Question100

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 has completed SDP rewriting, causing them to be sent to incorrect IP addresses or ports. Early media scenarios, NAT traversal, and multi-site deployments exacerbate this issue. Accurate SDP negotiation and media anchoring are crucial to ensure that RTP reaches the correct endpoint, preventing one-way communication and call drops.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP includes the 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 flows. Cisco best practices for multi-site SIP deployments using CUBE recommend delayed-offer SDP to prevent one-way audio, ensure call quality, and maintain reliable communication.

Option B, using SIP over UDP, does not address SDP timing or media path misalignment. While UDP is required for RTP transport, simply switching transport protocol does not solve early media or SDP rewrite issues.

Option C, disabling early media, prevents call progress tones but does not resolve RTP being sent to incorrect addresses. It addresses a symptom rather than the root cause.

Implementing delayed-offer SDP ensures correct media anchoring through CUBE, eliminates one-way audio, prevents call drops, and maintains reliable multi-site SIP communication, fully adhering to Cisco best practices.

Question101

During a CUCM deployment with Mobile and Remote Access (MRA), remote Jabber clients report call setup delays and intermittent audio issues. Internal endpoints operate without problems. RTP and signaling logs show inconsistent ICE candidate exchange. Which configuration MOST effectively resolves this problem?

Answer: B

Explanation:

Mobile and Remote Access relies on Expressway-C and Expressway-E traversal zones to allow secure remote connectivity for Jabber clients. When remote clients experience delayed call setup or intermittent media issues, ICE (Interactive Connectivity Establishment) candidate exchange is often the culprit. ICE enables endpoints behind NAT to discover the most efficient media path. If traversal zones do not support bidirectional UDP and TCP signaling with full ICE, ICE candidates cannot be exchanged reliably, resulting in call setup delays, one-way audio, or dropped calls.

Option B, ensuring traversal zones allow bidirectional UDP and TCP signaling with full ICE support, is the correct solution. This configuration guarantees proper ICE candidate exchange, enabling optimal media path selection. Cisco best practices emphasize bidirectional signaling and ICE support for MRA deployments, particularly when NAT or firewall traversal is involved. Proper ICE handling ensures remote clients can establish RTP paths efficiently, minimizing call setup delays, preventing media failures, and maintaining consistent voice and video quality. Full ICE support also improves interoperability with various network conditions, including different NAT types and firewalls, enhancing overall user experience for remote collaboration.

Option A, enabling persistent XMPP connections, improves presence and messaging continuity but does not address media path issues or NAT traversal. While beneficial for session persistence, it cannot resolve delayed call setup caused by ICE failures.

Option C, disabling SIP normalization, resolves header inconsistencies but does not impact ICE negotiation or RTP flow. Normalization primarily addresses signaling syntax mismatches rather than media path establishment.

Option D, using SIP over TCP exclusively, ensures reliable signaling but cannot fully support ICE, which requires UDP for proper media negotiation. TCP-only configurations do not prevent one-way audio or delayed call setup in NAT scenarios.

By ensuring traversal zones allow bidirectional UDP and TCP with full ICE support, administrators can provide consistent remote client connectivity, optimize media paths, reduce call setup delays, and adhere to Cisco best practices for Mobile and Remote Access deployments.

Question102

During peak periods, remote endpoints fail to register intermittently through Mobile and Remote Access, while internal endpoints register normally. Logs indicate that traversal zones between Expressway-C and Expressway-E are saturated. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Mobile and Remote Access relies on Expressway-C and Expressway-E traversal zones to provide secure registration and connectivity for remote endpoints. During peak hours, if traversal zones reach maximum capacity, remote endpoints cannot register, while internal endpoints that bypass traversal zones remain unaffected. Ensuring sufficient capacity for traversal zones is crucial for high availability, reliable collaboration services, and seamless user experience.

Option A, increasing the number of allowed traversal zone connections, is the correct solution. Cisco best practices recommend sizing traversal zones according to expected peak concurrent remote endpoints. By increasing allowed connections, remote endpoints can register successfully even during high-demand periods. Proper sizing ensures consistent registration, reduces user disruption, and maintains presence, messaging, and call functionality. Increasing capacity also reduces the need for workaround solutions such as VPN, minimizing complexity and latency.

Option B, disabling traversal zones and requiring VPN, adds complexity and latency and negates the benefits of MRA. VPN introduces additional infrastructure requirements and does not address the root cause of capacity exhaustion.

Option C, reducing SIP timers, increases retry frequency but does not solve traversal zone congestion. Faster retries may increase signaling load, potentially worsening the problem rather than resolving it.

Option D, enabling persistent XMPP connections, improves session persistence for messaging and presence but does not increase traversal zone capacity. It does not resolve intermittent registration failures during peak usage.

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

Question103

A multi-site CUCM deployment with CUBE is experiencing one-way audio and occasional call drops for inter-site SIP trunk calls. RTP packets are being 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 typically result from RTP being transmitted before SDP is rewritten by CUBE. Early media scenarios, NAT traversal, and multi-site configurations worsen the problem. Correct SDP negotiation and media anchoring are crucial to ensure RTP is delivered to the correct destination, preventing audio failures and call drops.

Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP sends SDP in the 200 OK response rather than the initial INVITE, allowing CUBE to anchor media, rewrite IP addresses and ports, and establish the correct RTP path before media transmission begins. Cisco best practices recommend delayed-offer SDP for trunks traversing CUBE in multi-site deployments to prevent one-way audio and ensure reliable inter-site communications. This approach guarantees predictable RTP delivery, reduces dropped calls, and maintains voice quality across sites.

Option B, converting SIP trunks to SCCP, does not address the timing of SDP delivery. SCCP may change signaling behavior, but early RTP transmission before SDP rewriting remains unresolved. Converting protocols adds complexity without solving the root cause.

Option C, using SIP over TCP, ensures signaling reliability but does not correct misaligned RTP paths due to early SDP transmission. TCP alone cannot prevent one-way audio or dropped calls caused by SDP timing issues.

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

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

Question104

Answer: A

Explanation:

Initial silence in voicemail playback is commonly 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, which VAD interprets as silence, causing several seconds of perceived silence at the start of playback. This affects user experience, leading to confusion or the perception of missing message content.

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 playback silence. Cisco best practices for CUCM and Unity Connection integration recommend disabling VAD for voicemail deployments to ensure uninterrupted message playback. Users can hear messages immediately, improving usability and satisfaction. The impact on bandwidth is minimal because voicemail RTP traffic is small relative to live call traffic.

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

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

Option D, moving the voicemail pilot to a different partition, affects routing but has no impact on RTP delivery or playback.

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

Question105

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, resulting in packets being sent to incorrect IP addresses or ports. NAT traversal and multi-site deployment scenarios exacerbate the problem. Accurate SDP negotiation and media anchoring are critical 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 ensures SDP is included 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 transmission begins. Cisco best practices for multi-site SIP deployments using CUBE recommend delayed-offer SDP to prevent one-way audio, maintain call quality, and provide reliable communication.

Option B, using SIP over UDP, does not address SDP timing or media path misalignment. While UDP is required for RTP, simply 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 path errors. This addresses a symptom rather than the root cause.

Option D, enabling symmetric RTP, helps with NAT traversal but does not resolve SDP timing or address misalignment causing one-way audio. Symmetric RTP alone cannot correct RTP 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.

One-way audio in multi-site Cisco Unified Communications deployments is a frequent and complex problem, particularly when calls traverse a Cisco Unified Border Element (CUBE). The root cause of this issue is typically related to the timing of SDP (Session Description Protocol) negotiation in relation to media path establishment. RTP (Real-time Transport Protocol) packets carry the audio media, and these packets rely on the IP addresses and port numbers communicated through SDP. If RTP begins flowing before CUBE has rewritten the SDP information, the media may be sent to incorrect IP addresses or ports, resulting in audio being delivered to only one side of the call. In multi-site or NAT traversal scenarios, this misalignment becomes even more pronounced because calls often cross multiple networks, firewalls, and possibly NAT devices, which further complicates proper RTP delivery.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP postpones the inclusion of SDP until the 200 OK response is sent, rather than including it in the initial INVITE. This delay allows CUBE to properly anchor media, rewrite IP addresses and port numbers, and establish correct RTP paths before media begins flowing. By ensuring that SDP is offered only after signaling processing, delayed-offer SDP prevents early RTP from being sent to incorrect destinations. Cisco best practices recommend this configuration in multi-site SIP deployments using CUBE, especially in environments where early media could cause RTP misalignment or one-way audio. Implementing delayed-offer SDP guarantees accurate RTP delivery, maintains high call quality, and ensures reliable communication between endpoints across distributed sites.

The benefits of delayed-offer SDP extend beyond resolving immediate audio issues. By controlling when SDP is offered, administrators can ensure predictable and consistent media flows across all sites. This predictability simplifies troubleshooting and operational monitoring because the media path aligns directly with the signaling path. In environments without delayed-offer, RTP streams might be transmitted to intermediate devices or incorrect endpoints before CUBE has anchored the media. Such scenarios lead to inconsistent call behavior and complicate packet capture analysis, call detail record interpretation, and overall call quality management. Delayed-offer SDP eliminates these complications by providing a deterministic flow for both signaling and media, which is crucial for maintaining a high-quality user experience.

Option B, using SIP over UDP instead of TCP, does not address the underlying cause of one-way audio. While UDP is a connectionless protocol often used for RTP due to its low latency, the timing of SDP delivery is independent of the transport protocol used for SIP signaling. Changing the transport protocol does not prevent early RTP from being sent to incorrect endpoints before CUBE has anchored the call. Therefore, while UDP is required for the media itself, simply changing the SIP transport does not correct misaligned media flows or one-way audio.

Option C, disabling early media, prevents pre-answer audio such as call progress tones or ringback signals. While this might eliminate audible artifacts or early audio streams, it does not resolve the fundamental problem of RTP being sent to the wrong destination. Disabling early media addresses only a symptom of the issue rather than the root cause, which is the premature transmission of SDP and unanchored RTP. Furthermore, removing early media may negatively affect the caller experience by preventing them from hearing expected progress tones during call setup.

Option D, enabling symmetric RTP on CUBE, is primarily used for handling NAT traversal or asymmetrical routing issues. Symmetric RTP ensures that media is sent back to the IP address and port from which it originated. While this feature can assist in scenarios with NAT devices or firewalls, it does not resolve the underlying issue of early SDP delivery or media path misalignment. One-way audio caused by premature SDP cannot be fully mitigated by symmetric RTP alone because the media may still be initially directed to an incorrect address before the symmetric mechanism takes effect. Symmetric RTP is a complementary feature but is insufficient on its own to solve SDP timing issues in multi-site deployments.

Implementing delayed-offer SDP ensures that CUBE has full control over the media path before RTP flows. This guarantees that audio is delivered reliably to both endpoints, eliminating one-way audio issues. In multi-site deployments, where calls may traverse multiple networks and devices, delayed-offer SDP provides a predictable and robust solution for media path management. It also enhances interoperability with service providers, cloud SIP trunks, and third-party SIP devices, as SDP is offered only after the signaling path has been fully processed, reducing the likelihood of codec mismatches, unsupported media formats, or misrouted RTP.

From a Cisco best practices standpoint, delayed-offer SDP aligns with recommended SIP trunk design in multi-site deployments. Cisco emphasizes deterministic media path establishment, proper anchoring, and predictable call setup behavior. By implementing delayed-offer SDP, organizations maintain compliance with these principles, ensuring that calls are reliable, media is anchored correctly, and audio quality remains high across all sites. This configuration also minimizes operational risks associated with misrouted RTP and supports consistent performance for voice and video communications.

Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 7 Q91-105

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