Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 9 Q121-135
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Question121
An enterprise is experiencing intermittent audio issues and registration failures for remote Jabber clients through Mobile and Remote Access (MRA). Internal endpoints function correctly. Logs show traversal zone congestion, dropped ICE candidate exchanges, and occasional NAT traversal failures. Which configuration MOST effectively resolves this issue?
A) Increase traversal zone capacity and enable full bidirectional UDP/TCP signaling with 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 (MRA) is a critical component in modern collaboration environments, providing secure connectivity for remote Jabber clients without requiring VPN access. Remote endpoint connectivity relies on Expressway-C and Expressway-E traversal zones for registration, signaling, and media path establishment. Intermittent audio issues, registration failures, and NAT traversal problems often manifest under peak load conditions due to two primary factors: traversal zone saturation and incomplete ICE (Interactive Connectivity Establishment) candidate exchange. ICE is fundamental in NAT scenarios, allowing endpoints to determine the most efficient IP addresses and ports for RTP media. When traversal zones are saturated, additional remote endpoints cannot register, and ongoing calls may experience one-way audio or drop entirely.
Option A, increasing traversal zone capacity while enabling full bidirectional UDP/TCP signaling with ICE support, is the most effective approach. By increasing traversal zone capacity, administrators ensure that a sufficient number of concurrent connections can be accommodated during peak usage, eliminating registration failures and call setup delays. Bidirectional signaling guarantees that ICE candidates are exchanged correctly, allowing endpoints to select optimal media paths. Full ICE support ensures that endpoints behind complex NAT or firewall configurations can establish RTP connectivity reliably, preventing one-way audio, call drops, and media disruption. Cisco best practices for MRA emphasize proper sizing of traversal zones and full ICE support, highlighting that network topology, firewall policies, and endpoint behavior must be considered together to maintain high availability and media reliability.
Option B, enabling persistent XMPP connections, primarily benefits messaging and presence continuity but does not directly resolve ICE failures or traversal zone congestion. While XMPP persistence can improve user experience during session drops, it does not affect media path negotiation or RTP delivery, which are the primary causes of one-way audio and registration failures.
Option C, disabling SIP normalization, addresses potential signaling inconsistencies but has no impact on ICE candidate negotiation or traversal zone capacity. SIP normalization primarily corrects header discrepancies in SIP signaling; while it may improve interoperability with certain endpoints, it does not prevent media path failures or registration issues under high load conditions.
Option D, using SIP over TCP exclusively, ensures reliable signaling transport but does not resolve ICE negotiation failures, which depend on UDP for proper media path discovery. TCP-only configurations may introduce signaling delays and can inadvertently exacerbate media path issues in NAT environments.
In conclusion, implementing option A provides a holistic solution that addresses the root causes of audio issues and registration failures for remote endpoints. Proper traversal zone sizing ensures that the system can handle peak load conditions without service degradation. Full bidirectional UDP/TCP signaling with ICE support guarantees optimal media path selection, overcoming NAT and firewall traversal challenges. This configuration aligns with Cisco best practices, ensuring remote users experience seamless voice and video communication, reducing operational support burdens, and maintaining enterprise collaboration reliability. By combining these measures, administrators can maintain high-quality user experiences and ensure consistent availability of MRA services. Continuous monitoring and dynamic adjustment of traversal zone capacity, along with periodic verification of ICE performance, further enhance network resiliency. The approach ensures that as remote endpoint populations fluctuate, the system remains robust, scalable, and compliant with enterprise collaboration policies.
Question122
During peak usage periods, remote endpoints cannot register through MRA, while internal endpoints register successfully. Expressway traversal zones report full connection capacity. Which configuration MOST effectively resolves this problem?
A) Increase traversal zone connections to accommodate peak load
B) Require VPN access and disable traversal zones
C) Reduce CUCM SIP registration timers to force rapid retries
D) Enable persistent XMPP connections between Expressway-C and CUCM
Answer: A
Explanation:
Traversal zones between Expressway-C and Expressway-E form the backbone of Mobile and Remote Access, enabling remote endpoint registration and secure communication. When traversal zones are at maximum capacity, additional remote endpoints cannot register, leading to service degradation. Internal endpoints bypass traversal zones, which explains why they register without issues. High availability in MRA deployments is heavily dependent on appropriately sized traversal zones, especially during peak usage periods when concurrent remote endpoint connections surge.
Option A, increasing traversal zone connections, is the optimal solution. By raising the allowed concurrent connections, the enterprise ensures that remote endpoints can register even under high load conditions. Cisco best practices recommend sizing traversal zones with headroom to accommodate anticipated peak loads, including unplanned spikes. Correctly sized traversal zones prevent call setup delays, registration failures, and one-way audio issues. Administrators can monitor utilization trends to anticipate demand and dynamically adjust traversal zone configurations as remote endpoint populations fluctuate, maintaining reliability. The configuration supports seamless remote collaboration, mitigates support tickets, and maintains enterprise communication standards.
Option B, requiring VPN access and disabling traversal zones, introduces additional complexity and latency, negating the benefits of MRA. VPN infrastructure creates administrative overhead and potential single points of failure, failing to address the root cause of traversal zone saturation.
Option C, reducing CUCM SIP registration timers, accelerates retry attempts but does not alleviate capacity constraints. Increased retry traffic could exacerbate traversal zone saturation, leading to further registration failures rather than resolving them.
Option D, enabling persistent XMPP connections, improves messaging and presence continuity but does not expand traversal zone capacity or prevent registration failures during peak periods. XMPP persistence has no direct impact on media path establishment or call setup for remote endpoints.
Increasing traversal zone connections provides a scalable, reliable solution that directly addresses registration failures during peak usage. By ensuring sufficient capacity and monitoring utilization trends, enterprises can maintain consistent service quality, avoid one-way audio or dropped calls, and align with Cisco best practices for Mobile and Remote Access deployments. This proactive approach minimizes operational disruption and supports an optimal remote collaboration experience.
Question123
In a multi-site CUCM deployment using CUBE for inter-site SIP trunking, calls experience one-way audio and occasional call drops. RTP packets are sometimes delivered 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 are commonly caused by RTP being transmitted before CUBE has completed SDP rewriting. Early RTP delivery to incorrect addresses or ports results in media path failures. Proper SDP negotiation and media anchoring are critical for predictable RTP delivery, especially in multi-site or NAT traversal scenarios.
Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, is the correct solution. Delayed-offer SDP includes the SDP payload 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. Cisco best practices recommend delayed-offer SDP for multi-site SIP trunks traversing CUBE, as it prevents one-way audio, call drops, and poor call quality. By anchoring media, CUBE ensures predictable packet delivery and alignment of signaling and media streams, addressing the root cause of these audio issues. Proper SDP handling improves audio reliability, reduces troubleshooting overhead, and aligns with enterprise quality standards.
Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not resolve early RTP transmission or media misalignment caused by SDP timing. SCCP cannot inherently correct media path errors introduced by premature RTP transmission.
Option C, using SIP over TCP, enhances signaling reliability but does not address RTP misalignment resulting from early SDP. TCP alone does not guarantee proper media path establishment or prevent one-way audio.
Option D, disabling early media, prevents call progress tones but does not fix RTP packets being sent to incorrect addresses. Early media configuration affects signaling but does not address SDP rewriting and media anchoring.
Implementing delayed-offer SDP ensures proper media anchoring through CUBE, eliminates one-way audio, reduces dropped calls, and maintains consistent call quality across multi-site SIP trunk deployments. This aligns with Cisco best practices and optimizes voice quality in enterprise collaboration environments.
Question124
Users report a few seconds of silence at the start of voicemail playback from Unity Connection integrated with CUCM. RTP analysis 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 often caused by Voice Activity Detection (VAD), which suppresses RTP during low-energy audio segments to conserve bandwidth during live calls. Voicemail messages frequently begin with low-energy audio that VAD interprets as silence, causing delayed playback and poor user experience.
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 for voicemail deployments recommend disabling VAD to ensure uninterrupted message playback. This approach guarantees that users hear messages immediately, improving usability and satisfaction. Bandwidth impact is minimal since voicemail traffic is low compared to live calls.
Option B, adjusting MWI extensions, affects lamp signaling but does not influence RTP delivery or message playback.
Option C, changing the codec to G.722, may enhance audio fidelity but does not resolve silence caused by VAD suppression.
Option D, moving the voicemail pilot to a different partition, affects call routing but does not impact RTP delivery or playback.
Disabling VAD ensures smooth voicemail playback, enhances user experience, and aligns with Cisco best practices for CUCM and Unity Connection integration, reducing support calls and improving satisfaction for enterprise users.
Question125
Remote users report intermittent one-way audio when calling internal endpoints through CUBE. RTP is sometimes sent to incorrect addresses before SDP rewriting. 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 completes SDP rewriting, causing packets to be sent to incorrect IP addresses or ports. Accurate SDP negotiation and media anchoring are essential for reliable RTP delivery and consistent call quality, particularly in multi-site deployments and NAT traversal scenarios.
Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP places the SDP in the 200 OK response rather than 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 trunk deployments using CUBE to prevent one-way audio, maintain call quality, and ensure reliable communications.
Option B, using SIP over UDP, does not address SDP timing or media path misalignment. UDP is required for RTP transport, but transport protocol alone does not correct early SDP transmission issues.
Option C, disabling early media, affects call progress tones but does not resolve RTP misalignment or address rewriting problems.
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 cannot replace 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, following Cisco best practices and ensuring consistent enterprise collaboration quality.
Question126
An organization deploys a multi-site CUCM and Expressway infrastructure to support remote Jabber clients. During peak hours, remote clients experience registration delays and intermittent one-way audio. Traversal zone logs indicate high CPU and memory utilization. Which configuration MOST effectively addresses this issue?
A) Increase traversal zone resources and enable bidirectional ICE for remote endpoints
B) Enable persistent XMPP connections between Expressway-C and CUCM
C) Disable SIP normalization to reduce CPU load
D) Use SIP over TCP exclusively for remote client signaling
Answer: A
Explanation:
Remote Jabber client connectivity relies on traversal zones established between Expressway-C and Expressway-E to handle registration, signaling, and media negotiation. Intermittent one-way audio and registration delays are often indicative of traversal zone resource saturation. High CPU and memory utilization within the traversal zone suggests that the infrastructure is under-provisioned for the current volume of remote clients. Proper capacity planning and configuration are essential to ensure seamless service delivery.
Option A, increasing traversal zone resources and enabling bidirectional ICE for remote endpoints, directly addresses the root cause. Increasing CPU and memory allocations allows the traversal zone to manage a higher number of simultaneous registrations and call sessions. ICE (Interactive Connectivity Establishment) enables endpoints behind NAT and firewalls to negotiate the optimal IP addresses and ports for RTP media. Bidirectional ICE ensures that both client and server can exchange candidates effectively, reducing the likelihood of one-way audio. Cisco best practices emphasize careful sizing of traversal zones based on anticipated peak load, with attention to both signaling and media resources. Additionally, enabling bidirectional ICE ensures proper media path selection even in complex network topologies with multiple NAT layers, which is a common scenario in enterprises supporting remote users.
Option B, enabling persistent XMPP connections, can improve the continuity of messaging and presence, particularly in cases of intermittent network drops. However, this configuration does not address the underlying traversal zone resource limitations or ICE-related media issues. Persistent XMPP is beneficial for user experience but insufficient for resolving registration delays or one-way audio caused by resource exhaustion.
Option C, disabling SIP normalization, may reduce some CPU processing overhead, particularly in environments where headers are heavily modified, but it does not resolve ICE negotiation issues or traversal zone saturation. SIP normalization primarily ensures interoperability between endpoints, and while it may have minor impact on CPU utilization, it is not a comprehensive solution for high CPU/memory scenarios affecting remote registration and media.
Option D, using SIP over TCP exclusively, ensures reliable signaling transport but does not solve media path problems caused by ICE failures or traversal zone saturation. TCP transport may increase signaling latency slightly and is not a substitute for properly sized traversal zones or ICE-enabled configurations.
By implementing Option A, administrators provide sufficient resources to handle peak registrations, ensure reliable ICE candidate negotiation, and maintain a high-quality collaboration experience for remote Jabber clients. Continuous monitoring of traversal zone utilization is recommended to preemptively scale resources before reaching critical thresholds. This proactive approach aligns with Cisco best practices for Mobile and Remote Access, ensuring scalability, reliability, and consistent call quality across multi-site deployments. Proper sizing, monitoring, and ICE enablement guarantee that remote users experience seamless voice, video, and presence services, even during peak traffic periods, while also reducing administrative overhead and support incidents.
Question127
During SIP trunk deployment between two CUCM clusters via CUBE, users report intermittent one-way audio, and some calls fail with early media not being established correctly. RTP flows show misaligned endpoints 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 to improve media negotiation
C) Use SIP over TCP instead of UDP
D) Disable early media on CUBE
Answer: A
Explanation:
One-way audio and early media issues in multi-cluster SIP trunking environments typically result from RTP being sent before CUBE completes SDP rewriting. In such scenarios, the RTP packets are directed to incorrect IP addresses or ports, leading to audio failures and call drops. Delayed-offer SDP ensures that the SDP information is only sent after CUBE has anchored the media path and rewritten the relevant addresses and ports.
Option A, enabling delayed-offer SDP, is the correct approach. Delayed-offer SDP shifts the SDP payload from the initial INVITE to the 200 OK response. This delay allows CUBE to inspect, modify, and anchor the media path appropriately. Cisco best practices recommend this configuration in multi-site SIP trunk deployments to prevent misaligned RTP delivery, one-way audio, and early media failures. Proper media anchoring ensures consistent call quality, predictable RTP paths, and robust inter-cluster communication. This solution also supports NAT traversal scenarios, as CUBE can rewrite SDP in a way that accommodates different network topologies, avoiding misrouted media streams.
Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not address the underlying media path misalignment caused by premature RTP transmission. SCCP does not inherently solve SDP timing or media anchoring issues and may introduce additional complexity without resolving the root cause.
Option C, using SIP over TCP, provides reliable signaling transport but does not solve misrouted RTP or early media problems. While TCP ensures delivery of SIP messages, it does not prevent media from being sent before SDP is properly processed by CUBE.
Option D, disabling early media on CUBE, affects call progress tone delivery but does not resolve the misalignment of RTP streams caused by early SDP transmission. Early media configuration influences signaling behavior but does not correct the actual RTP path issues causing one-way audio.
Enabling delayed-offer SDP provides a comprehensive solution, ensuring that RTP packets are correctly routed, media anchoring is properly established, and audio issues are resolved. It supports robust multi-site communications, reduces call failures, and aligns with Cisco best practices for SIP trunking via CUBE, delivering consistent voice quality across enterprise networks.
Question128
Users report several seconds of initial silence when accessing voicemail from Unity Connection via CUCM. RTP analysis shows no jitter or packet loss. 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 the 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 a classic symptom of Voice Activity Detection (VAD). VAD suppresses RTP during low-energy segments in live calls to conserve bandwidth. However, voicemail messages often start with low-energy audio that VAD interprets as silence, leading to delayed playback for end users.
Option A, disabling VAD on the SIP trunk, is the most effective solution. Disabling VAD ensures continuous RTP transmission, including low-energy segments, eliminating the initial silence during voicemail playback. Cisco best practices recommend VAD be disabled for voicemail services to maintain message integrity and improve user experience. While bandwidth usage slightly increases, the impact is minimal because voicemail RTP traffic is considerably lower than live call traffic. Ensuring uninterrupted RTP flow guarantees that users hear messages immediately and avoids misinterpretation of initial silence as a system error.
Option B, adjusting MWI extensions, impacts lamp signaling to indicate new messages but has no effect on RTP delivery or playback.
Option C, changing the codec to G.722, may enhance audio fidelity but does not address the problem caused by VAD. The initial silence will persist even with higher-quality codecs if VAD is active.
Option D, moving the voicemail pilot to a different partition, affects call routing but does not influence RTP delivery or playback.
By disabling VAD, administrators ensure smooth voicemail playback, improve end-user satisfaction, and align with Cisco best practices. This change reduces support calls related to perceived playback delays and ensures a predictable, high-quality user experience for voicemail across the enterprise.
Question129
An enterprise using CUCM and CUBE for inter-site SIP trunking reports one-way audio for remote users behind NAT. RTP is 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 in SIP trunk deployments often stems from premature RTP delivery before CUBE completes SDP rewriting. Without proper media anchoring, RTP packets are directed to incorrect IP addresses or ports, leading to failed audio transmission. Proper SDP negotiation is critical to ensure accurate RTP routing and high-quality voice communications.
Option A, enabling delayed-offer SDP, is the correct approach. By placing SDP in the 200 OK response rather than the initial INVITE, CUBE has the opportunity to anchor media, rewrite addresses, and ensure correct RTP paths before media transmission. Cisco best practices emphasize delayed-offer SDP for multi-site SIP trunk deployments, particularly in NAT or firewall traversal environments. This ensures predictable RTP flow, eliminates one-way audio, and reduces dropped calls.
Option B, using SIP over UDP, does not resolve misaligned RTP or early SDP delivery. While UDP is required for RTP, the transport protocol alone cannot address media path errors caused by premature SDP negotiation.
Option C, disabling early media, affects call progress tones but does not correct RTP misalignment caused by early SDP transmission.
Option D, enabling symmetric RTP, assists with NAT traversal but does not fix SDP timing issues or media path misalignment. Symmetric RTP can complement delayed-offer SDP but cannot replace it.
Implementing delayed-offer SDP guarantees proper media anchoring, reliable audio, and consistent communication quality, following Cisco best practices for multi-site SIP trunking. It ensures remote users behind NAT can communicate seamlessly with internal endpoints.
Question130
Remote Jabber clients behind enterprise firewalls report registration failures and intermittent call drops. ICE candidate negotiation frequently fails, and traversal zones are under high load. Which configuration MOST effectively mitigates this issue?
A) Increase traversal zone capacity 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 for secure registration, signaling, and media negotiation. Intermittent registration failures and call drops during peak hours typically result from traversal zone resource exhaustion and failed ICE candidate negotiation. ICE is critical in NAT environments to establish optimal RTP paths.
Option A, increasing traversal zone capacity and enabling full bidirectional ICE support, addresses the root cause. Proper resource allocation ensures traversal zones can handle peak connections, preventing registration delays. Bidirectional ICE allows efficient candidate exchange, ensuring endpoints select optimal media paths, eliminating one-way audio. Cisco best practices emphasize traversal zone sizing and ICE enablement for reliable MRA deployment.
Option B, requiring VPN connectivity, increases complexity and latency without addressing traversal zone saturation. VPNs may introduce single points of failure and do not solve ICE negotiation failures.
Option C, reducing SIP registration timers, increases retry frequency but does not alleviate resource limitations. Rapid retries may worsen congestion and further degrade performance.
Option D, enabling persistent XMPP connections, improves messaging continuity but does not resolve traversal zone overload or ICE failures.
Increasing traversal zone capacity with full ICE support ensures reliable registration, media path establishment, and high-quality remote collaboration, aligning with Cisco best practices and minimizing user-impacting call drops or registration failures.
Question131
During a multi-site CUCM deployment with CUBE trunks for SIP connectivity, some inter-site calls fail intermittently, and users report one-way audio. RTP analysis shows packets sent to incorrect addresses before CUBE SDP rewriting. Which configuration MOST effectively resolves this issue?
A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Switch SIP trunks to SCCP for inter-site calls
C) Use SIP over TCP exclusively
D) Disable early media on CUBE
Answer: A
Explanation:
One-way audio and inter-site call failures often occur when RTP is transmitted before CUBE can rewrite SDP addresses, causing media packets to be sent to incorrect destinations. In multi-site environments, proper SDP negotiation is critical to establish reliable media paths. Delayed-offer SDP ensures that the SDP payload is not included in the initial INVITE but in the 200 OK response, allowing CUBE to anchor the media path, rewrite addresses, and confirm the correct endpoints for RTP traffic.
Option A, enabling delayed-offer SDP, addresses the root cause by ensuring that CUBE processes signaling and media sequentially. This prevents premature RTP delivery and aligns the media path with the signaling path, maintaining consistent audio quality and eliminating one-way audio issues. Cisco best practices emphasize this configuration in multi-site deployments to support complex topologies, NAT traversal, and inter-cluster SIP connectivity. Delayed-offer SDP ensures interoperability between different sites while reducing call drops and user impact.
Option B, switching SIP trunks to SCCP, changes the signaling protocol but does not address early RTP issues. SCCP may improve endpoint compatibility but does not resolve media misalignment caused by premature SDP transmission.
Option C, using SIP over TCP exclusively, ensures reliable signaling delivery but does not prevent RTP from being sent before SDP is anchored. TCP helps with message delivery but does not influence media path establishment, so one-way audio can still occur.
Option D, disabling early media on CUBE, affects call progress tones but does not prevent RTP from reaching incorrect destinations before SDP rewriting. Early media configuration impacts signaling rather than media path alignment.
Enabling delayed-offer SDP provides a comprehensive solution to one-way audio and inter-site call failures. By properly anchoring media, CUBE ensures correct RTP routing, improves voice quality, and aligns with Cisco best practices for multi-site SIP trunk deployments. This configuration supports predictable call behavior, reduces troubleshooting complexity, and maintains enterprise-grade voice communication reliability across all sites.
Question132
Remote Jabber users behind NAT report intermittent call drops and registration failures. Traversal zones are near full capacity, and ICE candidate exchanges occasionally fail. Which configuration MOST effectively addresses 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:
Remote Jabber clients rely on traversal zones between Expressway-C and Expressway-E to handle secure registration, signaling, and media path establishment. Intermittent call drops and registration failures typically indicate traversal zone saturation and failed ICE negotiation. ICE (Interactive Connectivity Establishment) allows endpoints behind NAT and firewalls to determine the best IP and port combination for RTP media.
Option A, increasing traversal zone capacity and enabling full bidirectional ICE support, is the most effective solution. By allocating sufficient CPU and memory resources, traversal zones can handle peak connection loads, preventing registration delays and dropped calls. Bidirectional ICE ensures that both the client and server exchange candidates properly, enabling successful NAT traversal and reliable RTP path establishment. Cisco best practices emphasize properly sizing traversal zones based on expected concurrent remote endpoints and enabling full ICE support to prevent one-way audio or registration failures. This approach minimizes user impact, supports remote collaboration, and reduces operational support demands.
Option B, requiring VPN connectivity, increases infrastructure complexity and latency without addressing traversal zone overload. VPN access introduces a single point of failure and does not resolve ICE negotiation failures.
Option C, reducing CUCM SIP registration timers, increases retry frequency but does not alleviate capacity constraints. Frequent retries can worsen traversal zone saturation, potentially leading to more failures.
Option D, enabling persistent XMPP connections, improves messaging continuity but does not resolve registration or media path failures caused by traversal zone overload and ICE issues.
Increasing traversal zone capacity with full bidirectional ICE support ensures reliable registration and high-quality media for remote Jabber users. This configuration supports scalable remote access, reduces call drops, and aligns with Cisco best practices for Mobile and Remote Access deployments.
Question133
Users report several seconds of silence at the start of voicemail playback from Unity Connection via CUCM. RTP analysis 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 commonly caused by Voice Activity Detection (VAD), which suppresses RTP transmission during low-energy audio segments to conserve bandwidth during live calls. Voicemail messages often start with soft or silent segments that VAD interprets as silence, resulting in delayed playback.
Option A, disabling VAD on the SIP trunk, ensures continuous RTP transmission, including low-energy audio, eliminating the initial silence. Cisco best practices recommend disabling VAD for voicemail services to maintain message integrity and provide a seamless end-user experience. The increase in bandwidth is minimal due to the low volume of voicemail RTP traffic compared to live calls. Properly configured, disabling VAD improves user satisfaction, reduces support calls, and ensures consistent message playback across the enterprise.
Option B, adjusting MWI extensions, affects lamp signaling to indicate new messages but does not influence RTP delivery or playback.
Option C, changing the codec to G.722, improves audio fidelity but does not address the issue caused by VAD suppressing low-energy audio segments.
Option D, moving the voicemail pilot to a different partition, impacts call routing but does not influence RTP flow or initial silence.
By disabling VAD, administrators ensure immediate and consistent voicemail playback, improving the end-user experience and aligning with Cisco best practices for CUCM and Unity Connection integration.
Question134
During high traffic periods, remote Jabber clients experience one-way audio and intermittent registration failures. Traversal zones report near-full CPU and memory utilization, and ICE candidate exchanges fail. Which configuration MOST effectively resolves these issues?
A) Increase traversal zone resources and enable full bidirectional ICE support
B) Require VPN connectivity for remote clients
C) Reduce CUCM SIP registration timers to increase retries
D) Enable persistent XMPP connections between Expressway-C and CUCM
Answer: A
Explanation:
One-way audio and registration failures for remote Jabber clients are symptomatic of traversal zone resource exhaustion combined with failed ICE candidate negotiation. Traversal zones manage secure signaling, registration, and media path establishment between remote endpoints and CUCM. ICE enables endpoints behind NAT or firewalls to establish optimal RTP paths, preventing one-way audio.
Option A, increasing traversal zone resources and enabling full bidirectional ICE, addresses both root causes. By allocating sufficient CPU and memory, traversal zones can handle peak connection loads, reducing registration failures. Full bidirectional ICE ensures proper candidate exchange for reliable NAT traversal, establishing correct RTP paths and preventing one-way audio. Cisco best practices emphasize properly sized traversal zones with ICE support to maintain remote user service quality, scalability, and high availability. Continuous monitoring and capacity planning are essential to preemptively adjust resources in anticipation of peak usage periods.
Option B, requiring VPN connectivity, introduces additional complexity and latency without addressing traversal zone overload or ICE failures.
Option C, reducing SIP registration timers, increases retry frequency but does not solve resource limitations and may worsen congestion during peak periods.
Option D, enabling persistent XMPP connections, enhances messaging continuity but does not resolve traversal zone overload or ICE failures affecting voice quality.
Implementing Option A ensures reliable registration, correct media path establishment, and high-quality voice for remote Jabber users. This configuration supports scalable remote collaboration, reduces call drops, and aligns with Cisco best practices for Mobile and Remote Access deployments.
Question135
An enterprise using CUCM and CUBE reports intermittent call drops and one-way audio during inter-site SIP trunk calls. RTP packets are sometimes sent to incorrect destinations 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 UDP instead of TCP for signaling
C) Disable early media on CUBE
D) Enable symmetric RTP on CUBE
Answer: A
Explanation:
One-way audio and call drops in multi-site SIP trunk deployments are typically caused by RTP transmission before CUBE can rewrite SDP, resulting in media being sent to incorrect destinations. Correct SDP negotiation is crucial to align signaling and media paths, ensuring reliable call setup and media delivery.
Option A, enabling delayed-offer SDP, resolves the issue by sending the SDP in the 200 OK response rather than in 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 highlight delayed-offer SDP as essential for multi-site SIP trunk deployments to prevent one-way audio, early media failures, and dropped calls. Proper SDP handling ensures predictable RTP flow, improves call quality, and reduces troubleshooting overhead.
Option B, using SIP over UDP, ensures faster signaling but does not address early RTP delivery or misaligned media paths.
Option C, disabling early media on CUBE, prevents call progress tones from being sent but does not resolve RTP misalignment issues.
Option D, enabling symmetric RTP, assists NAT traversal but does not correct SDP timing or media anchoring issues. Symmetric RTP is complementary but not a substitute for delayed-offer SDP.
Enabling delayed-offer SDP ensures correct media anchoring, predictable RTP delivery, and consistent call quality. This approach follows Cisco best practices, minimizes call drops, and provides a reliable user experience for multi-site SIP trunking.
One-way audio and call drops in multi-site SIP trunk deployments are primarily caused by RTP packets being sent before the Cisco Unified Border Element (CUBE) has had the opportunity to rewrite Session Description Protocol (SDP) information. SDP carries critical media parameters, including IP addresses, port numbers, and codec selections, which are necessary for proper media path alignment. If RTP flows before CUBE has anchored the media and rewritten the SDP, audio may be delivered to the wrong endpoint, resulting in one-way audio or even dropped calls. This problem is compounded in multi-site deployments, where calls traverse multiple WAN links, firewalls, and sometimes NAT devices, making proper media anchoring even more crucial for ensuring reliable call setup and media delivery.
Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly addresses the root cause of the problem. Delayed-offer SDP postpones the transmission of SDP from the initial INVITE to the 200 OK response. This allows CUBE to receive the signaling, perform media anchoring, and rewrite IP addresses and ports as needed before RTP is transmitted. By controlling when SDP is offered, CUBE ensures that RTP packets flow to the correct destinations, eliminating one-way audio and reducing the likelihood of dropped calls. Cisco best practices for multi-site SIP trunk deployments consistently highlight delayed-offer SDP as a critical configuration for ensuring predictable media paths, high call quality, and reliable operation. Implementing delayed-offer SDP guarantees that signaling and media are synchronized and aligned, which is particularly important in distributed enterprise environments where media may traverse complex networks.
Using delayed-offer SDP also provides operational benefits beyond immediate issue resolution. By establishing predictable and deterministic media paths, IT teams can more efficiently monitor and troubleshoot call quality. Packet captures and call detail records reflect accurate media paths, allowing administrators to identify and resolve issues quickly. This predictability is particularly important for large, geographically dispersed deployments, where media flows may cross multiple network segments and infrastructure components. By ensuring that RTP is transmitted only after CUBE has anchored the call, delayed-offer SDP prevents media misrouting and reduces the operational burden of diagnosing intermittent audio problems.
Option B, using SIP over UDP instead of TCP for signaling, does not solve the underlying issue. While UDP provides low-latency, connectionless signaling, the problem in this scenario is the timing of SDP negotiation relative to media flow, not the transport protocol. Early RTP delivery occurs because SDP is included in the initial INVITE rather than because of transport method. Switching to UDP may slightly affect message timing but does not prevent one-way audio or media misalignment, so it is not a sufficient solution.
Option C, disabling early media on CUBE, only affects pre-answer audio, such as ringback tones, announcements, or call progress indicators. While this may prevent certain early audio artifacts, it does not correct the fundamental problem of RTP being sent to incorrect addresses before media anchoring occurs. Disabling early media treats only the symptoms of the problem rather than the underlying cause. Furthermore, disabling early media may negatively impact user experience, as callers no longer hear expected audio feedback during call setup. The root cause—premature SDP transmission—remains unaddressed, meaning one-way audio could still occur once the call is answered.
Option D, enabling symmetric RTP on CUBE, is designed to handle NAT traversal or asymmetric media routing scenarios. Symmetric RTP ensures that media is sent back to the IP address and port from which it originated, which is useful in some network topologies. However, symmetric RTP does not address the early SDP problem or the need for proper media anchoring. One-way audio caused by RTP being sent to incorrect addresses before SDP rewriting cannot be resolved solely through symmetric RTP. While this feature is complementary and may be useful in conjunction with delayed-offer SDP, it cannot replace proper SDP timing as a solution.