Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 3 Q31-45

Visit here for our full Cisco 350-801 exam dumps and practice test questions.

Question31

During a CUCM deployment, users report that outbound PSTN calls intermittently fail with fast busy signals. Analysis reveals that CUCM sequentially attempts multiple gateways, causing delayed call setup. Which solution MOST effectively resolves this issue while maintaining redundancy and minimizing latency?

A) Remove all but one gateway from the route pattern
B) Implement Local Route Groups (LRG) to dynamically select gateways based on device pool
C) Use time-of-day routing to direct calls through specific gateways
D) Configure route lists with weighted priorities to force gateway selection

Answer: B

Explanation:

This scenario highlights a common challenge in CUCM multi-gateway environments: outbound PSTN call failures due to sequential route group processing. CUCM uses route patterns to define outbound dialing behavior. When multiple gateways are associated with a route pattern, CUCM tries each gateway sequentially until it finds one that can successfully process the call. Sequential processing introduces latency during call setup, and if the first few gateways are busy or unresponsive, the caller may receive fast busy signals, resulting in failed calls.

Option B, implementing Local Route Groups (LRG), is the most effective solution. LRG allows CUCM to select the gateway dynamically based on the device pool associated with the calling endpoint. Instead of sequentially attempting all gateways, CUCM immediately routes the call to the gateway closest or most appropriate for the device, reducing setup delays. LRG also maintains redundancy: alternate gateways remain available for failover in case the primary gateway is unavailable. This approach aligns with Cisco best practices for optimizing multi-gateway deployments, ensuring efficient call routing and minimizing the chance of call failures.

Option A, removing all but one gateway, may reduce sequential search delays but sacrifices redundancy. If the single gateway fails, there is no failover, which compromises high availability. While latency may improve slightly, the operational risk outweighs the benefit.

Option C, using time-of-day routing, allows calls to be routed through specific gateways during scheduled periods but does not address the underlying sequential gateway search problem. Time-of-day routing controls routing paths based on schedules but does not improve real-time gateway selection efficiency.

Option D, configuring weighted priorities in route lists, influences which gateway CUCM prefers, but sequential searching still occurs if the preferred gateway is unavailable. Weighted priorities reduce the chance of choosing a less-preferred gateway first but do not eliminate delays caused by sequential attempts.

Implementing Local Route Groups is therefore the most reliable solution. It ensures immediate selection of the most appropriate gateway, preserves redundancy, and optimizes call setup times, fully addressing the issue described while following Cisco best practices for multi-gateway call routing in CUCM.

Question32

Remote Jabber users connected via Mobile and Remote Access (MRA) report delayed presence updates for certain internal users, while internal presence updates function normally. Logs show intermittent XMPP delays between Expressway-C and CUCM. Which configuration MOST effectively resolves this issue?

A) Increase polling intervals for CUCM subscriber devices
B) Enable persistent XMPP connections between Expressway-C and CUCM
C) Configure firewall to block non-essential ports to reduce traffic
D) Temporarily disable traversal zones to isolate the issue

Answer: B

Explanation:

This scenario centers on real-time presence communication in an MRA deployment. Presence information is communicated using XMPP between Jabber clients and CUCM. Remote users rely on Expressway-C to relay XMPP messages. Intermittent delays suggest that XMPP sessions are being established and torn down frequently or are not persistent, causing delayed or missed presence updates for remote clients. Internal users are unaffected because they communicate directly with CUCM without traversing Expressway-C.

Option B, enabling persistent XMPP connections, is the correct solution. Persistent connections keep XMPP sessions continuously open between Expressway-C and CUCM. This ensures that presence updates are immediately propagated to remote clients without waiting for session establishment. Persistent connections reduce latency and maintain a reliable communication path for presence notifications. Cisco best practices emphasize the use of persistent XMPP connections in MRA deployments to guarantee timely presence updates for remote users.

Option A, increasing polling intervals, would actually worsen the problem. Polling intervals control how often CUCM queries endpoints for status. Increasing the interval delays updates, exacerbating latency issues rather than improving real-time presence.

Option C, blocking non-essential firewall ports, may disrupt necessary XMPP traffic if misapplied. While security is important, indiscriminate port blocking can prevent legitimate presence updates from traversing the network.

Option D, disabling traversal zones, would disconnect remote clients entirely, which is counterproductive. Traversal zones are necessary for remote Jabber clients to communicate with CUCM via Expressway-C/E. Disabling them does not solve the intermittent XMPP delay problem.

Persistent XMPP connections are critical for low-latency, reliable presence delivery to remote Jabber clients. Enabling them addresses the root cause of delayed presence updates, improves user experience, and aligns with Cisco’s deployment guidelines for MRA.

Question33

During SIP-based conferencing in CUCM, internal participants successfully join calls, but remote SIP endpoints fail intermittently when added to ongoing conferences. SIP trunks are configured for delayed offer, and mid-call re-INVITEs fail for remote participants. Which configuration MOST effectively resolves the issue?

A) Enable early offer SIP on trunks used by remote endpoints
B) Assign hardware MTPs to all endpoints
C) Reduce the number of SIP trunks to simplify signaling
D) Convert remote SIP endpoints to SCCP protocol

Answer: A

Explanation:

This scenario involves SIP conferencing behavior and the role of SDP negotiation. Mid-call re-INVITEs are used to add participants to an existing conference. Delayed-offer SIP trunks send SDP in the 200 OK response rather than in the initial INVITE. When a remote SIP endpoint is added, CUCM may lack sufficient SDP information to negotiate media paths correctly, resulting in conference failures for remote participants.

Option A, enabling early offer SIP, is the correct solution. Early offer ensures SDP is included in the initial INVITE, providing CUCM with immediate information about the remote participant’s media capabilities. This allows CUCM to establish the correct media path during mid-call operations, preventing failures when adding remote endpoints. Early offer is particularly important in distributed environments where NAT traversal or firewall rules may affect media connectivity. Cisco best practices recommend early offer SIP for trunks used in conferencing, especially in geographically distributed or multi-site deployments.

Option B, assigning hardware MTPs, primarily addresses codec transcoding. While MTPs facilitate compatibility between endpoints with different codecs, they do not resolve mid-call SDP negotiation issues that prevent remote participants from joining conferences.

Option C, reducing the number of SIP trunks, may simplify signaling but does not address the fundamental SDP timing issue. The failures occur because SDP is delayed, not because of trunk quantity.

Option D, converting remote endpoints to SCCP, is unnecessary. The problem is related to SDP negotiation and early offer timing, not the protocol type. Converting endpoints would not resolve mid-call failures.

Enabling early offer SIP ensures that SDP negotiation occurs at the correct stage, allowing CUCM to successfully add remote SIP participants to conferences, maintaining reliable conferencing functionality across sites.

Question34

A Cisco Unity Connection deployment integrated with CUCM experiences several seconds of initial silence when playing back voicemail messages. RTP traffic is normal, with no jitter or packet loss. Which configuration MOST effectively resolves the 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:

This scenario involves voicemail playback issues, specifically initial silence despite normal RTP flow. Voice Activity Detection (VAD) is a mechanism that suppresses RTP transmission when low-energy audio is detected. While VAD conserves bandwidth in regular calls, it can inadvertently suppress the start of voicemail recordings where audio levels are initially low, causing several seconds of silence.

Option A, disabling VAD, is the correct solution. By disabling VAD on the SIP trunk between CUCM and Unity Connection, RTP is transmitted continuously, including low-energy audio at the start of messages. This ensures that voicemail playback begins immediately, eliminating the initial silence. Cisco best practices for voicemail systems explicitly recommend disabling VAD to prevent playback gaps and maintain a smooth user experience. Disabling VAD does not compromise system performance significantly because voicemail traffic is typically limited compared to live calls, and the benefit of uninterrupted playback outweighs any minor increase in bandwidth usage.

Option B, adjusting MWI extensions, only affects message lamp signaling and has no impact on RTP or voicemail playback behavior.

Option C, changing the codec to G.722, addresses audio quality rather than playback timing. Codec mismatch could affect sound fidelity but does not cause initial silence.

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

Disabling VAD ensures immediate and uninterrupted voicemail playback, resolving user complaints about delayed message start, and aligns with Cisco best practices for CUCM and Unity Connection integration.

Question35

Remote users report intermittent one-way audio when calling internal endpoints through a CUBE. RTP analysis reveals that packets are sometimes sent to incorrect IP addresses before the CUBE rewrites the 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:

This scenario addresses one-way audio issues in multi-site SIP deployments involving CUBE. The root cause is early RTP delivery before CUBE can rewrite SDP and correctly anchor the media path. If RTP packets are sent to an incorrect IP address, the receiving endpoint does not receive audio, resulting in one-way communication.

Option A, enabling delayed-offer SDP, is the correct solution. With delayed offer, CUCM sends SDP in the 200 OK response rather than the initial INVITE. This allows CUBE to anchor media correctly, rewrite IP addresses and ports, and establish the proper media path before RTP begins. Cisco best practices recommend delayed-offer SIP for trunks traversing CUBE when early media could cause media path misalignment. Delayed-offer SDP ensures predictable RTP delivery, resolves one-way audio, and maintains reliable communication across distributed sites.

Option B, using UDP instead of TCP, affects signaling transport speed but does not resolve early SDP or media path issues.

Option C, disabling early media, may prevent certain call progress tones but does not correct RTP misalignment. It addresses symptoms rather than the underlying SDP problem.

Option D, enabling symmetric RTP, assists with NAT traversal but does not fix early media delivery to the wrong destination. Symmetric RTP is unrelated to the root cause in this scenario.

Implementing delayed-offer SDP ensures correct media anchoring through CUBE, eliminates one-way audio, and maintains reliable call flow, fully aligning with Cisco best practices for multi-site SIP deployments.

Question36

A CUCM administrator notices that certain outbound calls fail intermittently, while others succeed, despite multiple gateways being available. Analysis shows that CUCM attempts gateways sequentially, and the first few attempts are often busy, causing fast busy signals. Which configuration MOST effectively resolves this issue while ensuring high availability?

Answer: B

Explanation:

This scenario illustrates the common problem of outbound call failures in multi-gateway CUCM environments due to sequential route processing. CUCM uses route patterns to define outbound dialing behavior. When multiple gateways are assigned to a route pattern without optimization, CUCM sequentially attempts each gateway until one successfully completes the call. If the first gateways in the sequence are busy or unreachable, call setup delays occur, resulting in fast busy signals.

Option B, implementing Local Route Groups (LRG), is the most effective solution. LRG allows CUCM to select a gateway dynamically based on the device pool associated with the calling endpoint. Instead of attempting gateways sequentially, CUCM immediately routes the call to the nearest or most appropriate gateway, reducing latency and minimizing the chance of call failure. LRG preserves high availability because alternate gateways remain available for failover if the primary gateway is unavailable. This dynamic selection aligns with Cisco best practices for multi-gateway deployments, ensuring both efficiency and redundancy.

Option A, removing all but one gateway, may reduce sequential search delays but sacrifices redundancy. In the event of a gateway failure, there is no failover, violating high-availability principles. While latency may improve slightly, the operational risk is significant.

Option C, time-of-day routing, allows calls to be routed based on schedules, such as redirecting traffic during off-peak hours. However, it does not address sequential gateway selection or reduce call setup latency. It is a scheduling tool, not a real-time routing optimization solution.

Option D, configuring weighted priorities in route lists, influences CUCM’s preference for certain gateways. However, sequential searching still occurs if the preferred gateway is busy or unavailable. Weighted priorities partially address the issue but do not eliminate the delays associated with sequential attempts.

By implementing Local Route Groups, CUCM can immediately select the optimal gateway for each device pool, ensuring efficient call routing, high availability, and reduced setup delays. This approach directly addresses the root cause of intermittent outbound call failures while adhering to Cisco deployment best practices.

Question37

Remote Jabber clients connected via Mobile and Remote Access (MRA) report delayed or missing presence updates for internal users. Internal users’ presence works correctly. Logs show intermittent XMPP delays between Expressway-C and CUCM. Which configuration MOST effectively resolves this problem?

Answer: B

Explanation:

This scenario involves delayed presence updates for remote Jabber clients in an MRA deployment. Presence information is transmitted using XMPP between Jabber clients and CUCM. Remote clients rely on Expressway-C to relay presence messages to and from CUCM. The delays suggest that XMPP sessions are not persistent, causing intermittent latency in the propagation of presence information. Internal users do not experience this issue because they communicate directly with CUCM without traversing Expressway-C.

Option B, enabling persistent XMPP connections, is the correct solution. Persistent connections maintain continuous XMPP sessions between Expressway-C and CUCM. This ensures that presence updates are transmitted in real-time, reducing latency and preventing missed updates. Persistent XMPP connections align with Cisco best practices for MRA deployments, guaranteeing reliable and immediate presence propagation for remote users. With persistent connections, CUCM does not need to re-establish XMPP sessions for each update, which eliminates intermittent delays and improves user experience.

Option A, increasing polling intervals, would exacerbate the problem. Polling intervals dictate how frequently CUCM queries devices for status updates. Increasing the interval slows updates, worsening latency rather than improving it.

Option C, blocking non-essential firewall ports, could inadvertently disrupt legitimate XMPP traffic. While network security is important, indiscriminate blocking of ports may prevent presence messages from reaching remote clients, compounding the problem rather than solving it.

Option D, disabling traversal zones, would disconnect remote clients entirely. Traversal zones are essential for remote Jabber clients to communicate with CUCM via Expressway-C/E. Disabling them does not address intermittent XMPP delays and would instead prevent connectivity.

Enabling persistent XMPP connections ensures that remote Jabber clients receive timely presence updates. It addresses the root cause of latency, improves reliability, and follows Cisco best practices for MRA and remote presence communication.

Question38

During a distributed CUCM SIP-based conferencing deployment, internal participants can join conferences without issues. Remote SIP endpoints, however, fail intermittently when being added to ongoing conferences. SIP trunks are configured for delayed offer, and mid-call re-INVITEs fail. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

This scenario focuses on SIP conferencing and SDP negotiation. Mid-call re-INVITEs are used to add new participants to ongoing conferences. Delayed-offer trunks send SDP in the 200 OK response rather than the initial INVITE. When adding remote SIP endpoints, CUCM may not have sufficient SDP information to establish media paths, resulting in failures. This is particularly common in distributed deployments where NAT and firewall traversal complicate media negotiation.

Option A, enabling early offer SIP, is the correct solution. Early offer includes SDP in the initial INVITE, giving CUCM immediate information about the remote participant’s media capabilities. This ensures proper media path establishment for all participants, including remote endpoints. Early offer is critical for reliable conferencing in multi-site or distributed deployments, preventing mid-call failures when adding remote participants. Cisco best practices recommend early offer SIP for trunks that handle conferencing to maintain predictable call behavior.

Option B, assigning hardware MTPs, primarily addresses codec compatibility. While MTPs ensure endpoints can communicate with different codecs, they do not resolve SDP timing issues that prevent remote participants from joining conferences.

Option C, reducing the number of SIP trunks, simplifies signaling but does not correct SDP negotiation problems. The number of trunks is irrelevant to mid-call SDP timing.

Option D, converting remote endpoints to SCCP, is unnecessary. The issue is SDP negotiation, not protocol compatibility. Converting endpoints does not solve mid-call failures caused by delayed-offer SDP.

Enabling early offer SIP ensures timely SDP negotiation, allowing CUCM to add remote SIP participants successfully. This approach resolves mid-call failures, maintains reliable conferencing functionality, and adheres to Cisco deployment best practices for distributed conferencing.

Question39

A Cisco Unity Connection system integrated with CUCM exhibits several seconds of silence at the start of voicemail messages, even though RTP flow is normal and there is no packet loss. Which configuration MOST effectively resolves this problem?

Answer: A

Explanation:

This scenario describes a voicemail playback issue. The initial silence is caused by Voice Activity Detection (VAD), which suppresses RTP packets when low-energy audio is detected. VAD is intended to conserve bandwidth during calls by not transmitting “silent” audio. However, in voicemail, the beginning of a message may be low-energy audio, which VAD mistakenly interprets as silence, resulting in several seconds of initial silence.

Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, is the correct solution. Disabling VAD ensures continuous RTP transmission, including the initial portion of the voicemail recording. Cisco best practices for voicemail deployments recommend disabling VAD to avoid playback gaps. This provides smooth and uninterrupted voicemail playback from the start of the message, improving user experience and reducing complaints about delayed audio.

Option B, adjusting MWI extensions, affects only the lamp on endpoints indicating message status. It does not influence RTP flow or message playback and therefore cannot resolve the issue.

Option C, changing the codec to G.722, improves audio fidelity but does not address initial silence caused by VAD. Codec selection affects quality, not timing or VAD behavior.

Option D, moving the voicemail pilot to a different partition, alters call routing but does not impact media delivery or RTP behavior.

Disabling VAD is the most effective solution because it directly addresses the suppression of initial RTP packets. It ensures that voicemail playback starts immediately and reliably, consistent with Cisco best practices for CUCM and Unity Connection integration.

Question40

Remote users report intermittent one-way audio when calling internal endpoints through a CUBE. RTP analysis shows that packets are sometimes sent to incorrect IP addresses before the 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:

This scenario concerns one-way audio issues in distributed SIP deployments using CUBE. Early RTP delivery before CUBE can rewrite SDP causes packets to be sent to incorrect addresses, resulting in one-way audio. The underlying problem is the timing of SDP negotiation relative to media flow.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP ensures that CUCM sends SDP in the 200 OK response rather than the initial INVITE. This allows CUBE to anchor media correctly, rewrite IP addresses and ports, and establish the proper media path before RTP begins. Cisco best practices recommend delayed-offer SIP for trunks traversing CUBE when early media could result in media misalignment. This configuration ensures predictable RTP delivery, eliminates one-way audio, and maintains reliable communication across sites.

Option B, using UDP instead of TCP, affects transport protocol but does not resolve early SDP delivery issues. Transport choice does not correct the misalignment of media paths.

Option C, disabling early media, might prevent certain call progress tones but does not solve the problem of RTP being sent to incorrect addresses. It addresses symptoms rather than the root cause.

Option D, enabling symmetric RTP, assists with NAT traversal but does not fix early media misdelivery. Symmetric RTP is unrelated to the root cause in this case.

Configuring delayed-offer SDP ensures correct media anchoring through CUBE, eliminates one-way audio, and maintains reliable call flow for multi-site SIP deployments, fully adhering to Cisco best practices.

Question41

A CUCM deployment with multiple gateways experiences intermittent outbound call failures. Logs show that CUCM sequentially attempts multiple gateways, causing delayed call setup and fast busy signals. Which configuration MOST effectively resolves this issue while ensuring high availability?

Answer: B

Explanation:

This scenario represents a classic issue in CUCM multi-gateway environments. Outbound call failures often result from sequential gateway attempts where CUCM tries each gateway in a route pattern one by one. Sequential searches can introduce latency, and if the first few gateways are busy or unresponsive, calls may fail, producing fast busy signals.

Option B, implementing Local Route Groups (LRG), is the optimal solution. LRG allows CUCM to dynamically select the gateway based on the device pool of the calling endpoint. Instead of sequential attempts, CUCM routes the call immediately to the most appropriate gateway. This reduces call setup time and prevents intermittent failures. LRG also maintains redundancy since alternate gateways remain available if the primary gateway is unavailable. This aligns with Cisco best practices, ensuring both efficiency and high availability in multi-gateway deployments.

Option A, removing all but one gateway, may reduce sequential attempts but eliminates redundancy. A single gateway failure would result in call blocking, which is unacceptable in production environments.

Option C, time-of-day routing, redirects calls based on schedules but does not address sequential gateway selection or latency issues. It is a scheduling tool rather than a solution for call setup delays.

Option D, configuring weighted priorities in route lists, influences gateway selection preference but does not eliminate sequential attempts if the preferred gateway is busy. Weighted priorities partially optimize routing but do not resolve the underlying issue.

Using LRG ensures that calls are routed efficiently, redundancy is maintained, and high availability is preserved, effectively solving the root cause of intermittent outbound call failures.

Question42

Remote Jabber users using Mobile and Remote Access (MRA) report delayed presence updates for internal users. Internal users’ presence updates are functioning normally. XMPP logs indicate intermittent delays between Expressway-C and CUCM. Which configuration MOST effectively resolves this problem?

Answer: B

Explanation:

This scenario involves real-time presence communication between remote Jabber clients and CUCM through MRA. Presence information relies on XMPP messages relayed by Expressway-C. Delayed or missing updates indicate that XMPP sessions may not be persistent, causing intermittent propagation delays. Internal users do not encounter this problem because their Jabber clients communicate directly with CUCM without traversing Expressway-C.

Option B, enabling persistent XMPP connections, is the correct solution. Persistent connections maintain continuous XMPP sessions between Expressway-C and CUCM, ensuring real-time propagation of presence updates to remote clients. This reduces latency and eliminates the risk of missed updates. Cisco best practices for MRA deployments recommend persistent XMPP connections to guarantee reliable presence updates. By keeping sessions open, CUCM does not need to re-establish XMPP connections for each update, ensuring consistent and immediate delivery of presence information.

Option A, increasing polling intervals, would worsen the issue. Polling intervals control how frequently CUCM queries endpoints for status. Longer intervals delay updates, making the problem more pronounced rather than resolving it.

Option C, blocking non-essential firewall ports, could inadvertently prevent necessary XMPP traffic from reaching remote clients. While security is important, indiscriminate blocking can interrupt legitimate communications, exacerbating delays.

Option D, disabling traversal zones, would disconnect remote Jabber clients completely, which is not a viable solution. Traversal zones are essential for enabling remote client connectivity through Expressway-C/E.

Enabling persistent XMPP connections ensures timely presence updates for remote users, addresses the root cause of delays, improves reliability, and follows Cisco best practices for MRA deployments.

Question43

In a SIP-based distributed CUCM conferencing deployment, internal participants join conferences without issues, but remote SIP endpoints fail intermittently when being added to ongoing conferences. SIP trunks are configured for delayed offer, and mid-call re-INVITEs fail. Which configuration MOST effectively resolves this problem?

Answer: A

Explanation:

This scenario focuses on conferencing and SDP negotiation in CUCM. Mid-call re-INVITEs are used to add participants to ongoing conferences. Delayed-offer trunks send SDP in the 200 OK response rather than in the initial INVITE. Remote SIP endpoints may fail to join because CUCM lacks immediate SDP information to establish media paths, especially in multi-site environments with NAT and firewall considerations.

Option A, enabling early offer SIP, is the correct solution. Early offer includes SDP in the initial INVITE, providing CUCM immediate knowledge of the remote participant’s media capabilities. This ensures proper media path negotiation, allowing remote endpoints to join conferences reliably. Early offer is essential in distributed or multi-site environments to prevent mid-call failures. Cisco best practices recommend early offer SIP for conferencing trunks to maintain predictable behavior and consistent media connectivity.

Option B, assigning hardware MTPs, addresses codec compatibility but does not solve the SDP timing issue that prevents remote endpoints from joining.

Option C, reducing the number of SIP trunks, simplifies signaling but does not correct SDP negotiation problems. SDP timing, not the number of trunks, causes mid-call failures.

Option D, converting remote endpoints to SCCP, is unnecessary. The issue is related to SDP negotiation, not endpoint protocol compatibility. Protocol conversion does not resolve mid-call conference failures.

Enabling early offer SIP ensures timely SDP negotiation, allowing CUCM to successfully add remote SIP participants. This resolves mid-call failures and maintains reliable conferencing functionality across sites.

Question44

A Cisco Unity Connection deployment integrated with CUCM exhibits several seconds of silence at the start of voicemail playback. RTP analysis shows normal flow without jitter or packet loss. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

The scenario describes initial silence at the beginning of voicemail messages. This is commonly caused by Voice Activity Detection (VAD), which suppresses RTP packets when it detects low-energy audio. VAD is designed to conserve bandwidth during active calls by not transmitting “silent” portions. However, voicemail recordings often start with low-energy audio, which VAD interprets as silence, resulting in several seconds of no sound at the beginning of playback.

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 at the start of messages. Cisco best practices recommend disabling VAD for voicemail deployments to prevent playback gaps. This guarantees that users hear the beginning of the message immediately, improving user experience and reducing complaints. Disabling VAD has minimal impact on bandwidth for voicemail traffic, which is limited compared to live calls.

Option B, adjusting MWI extensions, only affects the lamp indication for message waiting. It does not influence RTP or message playback.

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

Option D, moving the voicemail pilot to a different partition, changes call routing but does not affect media delivery or RTP handling.

Disabling VAD directly resolves the initial silence problem, ensuring uninterrupted voicemail playback and adherence to Cisco best practices for CUCM and Unity Connection integration.

Question45

Remote users report intermittent one-way audio when calling internal endpoints through a CUBE. RTP analysis indicates that packets are sometimes sent to incorrect IP addresses before the 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:

This scenario involves one-way audio issues in multi-site SIP deployments using CUBE. Early RTP delivery before CUBE can rewrite SDP results in packets being sent to incorrect IP addresses, causing one-way audio. The underlying problem is the timing of SDP negotiation relative to media path establishment.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP ensures that CUCM includes SDP in the 200 OK response rather than in the initial INVITE. This allows CUBE to anchor media correctly, rewrite IP addresses and ports, and establish the proper media path before RTP begins. Cisco best practices recommend delayed-offer SIP for trunks traversing CUBE when early media could cause misalignment of media paths. This configuration ensures correct RTP delivery, eliminates one-way audio, and maintains reliable communication between remote and internal endpoints.

Option B, using SIP over UDP instead of TCP, affects transport protocol but does not resolve SDP timing issues. The problem is related to media path misalignment, not transport.

Option C, disabling early media, may prevent certain call progress tones but does not correct RTP misdelivery. It addresses symptoms, not the underlying SDP problem.

Option D, enabling symmetric RTP, helps NAT traversal but does not resolve early media being sent to the wrong destination. Symmetric RTP does not fix SDP timing issues.

Configuring delayed-offer SDP ensures correct media anchoring through CUBE, eliminates one-way audio, and maintains reliable call flow across sites, fully following Cisco best practices for SIP deployments with CUBE.

One-way audio in multi-site Cisco Unified Communications environments often stems from the interaction between SIP signaling and RTP media paths when a Cisco Unified Border Element (CUBE) is involved. The underlying technical problem is related to the timing of SDP negotiation in relation to media path establishment. Early media, including ringback tones or announcements, may be transmitted before CUBE has the opportunity to anchor the call and rewrite the SDP. As a result, RTP packets are delivered to incorrect IP addresses or ports, causing one-way audio conditions.

Enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE addresses this issue effectively. In a delayed-offer configuration, CUCM does not include SDP in the initial SIP INVITE. Instead, SDP is offered in the 200 OK response after the signaling path has been fully processed by CUBE and CUCM. This approach allows CUBE to correctly anchor the call, rewriting the media IP addresses and port numbers in the SDP to ensure that RTP flows through the intended path. By establishing the media path only after SDP negotiation, delayed-offer SDP prevents early RTP from being sent to an incorrect endpoint. This mechanism is particularly critical in multi-site deployments where CUBE may serve as the media anchor between remote and internal sites. Correct SDP timing ensures that RTP flows correctly between endpoints, eliminating the one-way audio issue. Cisco deployment best practices explicitly recommend using delayed-offer SDP for trunks traversing CUBE, especially in scenarios where early media could result in misaligned media paths or incorrect RTP delivery.

Option B, using SIP over UDP instead of TCP, modifies the transport protocol but does not address the timing of SDP delivery. UDP and TCP differ in reliability and connection behavior, with UDP being connectionless and TCP providing guaranteed delivery. However, the core issue in this scenario is not the transport layer but the sequence in which SDP is exchanged and the subsequent media path establishment. Switching to UDP may change the delivery characteristics of SIP messages slightly, but it does not prevent RTP packets from being sent to an incorrect destination before CUBE rewrites the media addresses. Consequently, transport protocol selection alone cannot resolve one-way audio caused by premature SDP.

Option C, disabling early media on CUBE, might seem like a potential fix because it prevents RTP from being sent before the call is answered. While this could eliminate audio anomalies like premature ringback tones or announcements, it does not address the underlying SDP misalignment. The call may still fail to establish a correct RTP path if the SDP is offered too early in the signaling process. Disabling early media merely suppresses symptoms without correcting the root cause. Additionally, eliminating early media may negatively impact the user experience, as callers may not hear expected call progress signals. Therefore, this option does not provide a complete resolution to the one-way audio problem.

Option D, enabling symmetric RTP, is typically applied to resolve issues involving NAT traversal. Symmetric RTP forces media to be sent back to the source IP and port from which RTP packets are received, which can be useful when endpoints are behind NAT or firewalls. However, the one-way audio in this case is not caused by NAT or asymmetric media paths; it is caused by early SDP delivery that results in incorrect media addresses. While symmetric RTP can help in certain network topologies, it does not prevent early RTP from reaching an unintended destination before CUBE has anchored the call. Therefore, this configuration change does not address the primary problem in this scenario.

The deployment of delayed-offer SDP provides several advantages beyond resolving one-way audio. By sending SDP in the 200 OK rather than the INVITE, CUCM and CUBE can fully control media negotiation, ensuring correct anchoring and addressing of RTP packets. This configuration reduces troubleshooting complexity because the media path aligns with the signaling path, making monitoring, diagnostics, and packet captures more predictable and consistent. The predictable sequence—INVITE without SDP, followed by SDP in the 200 OK—ensures proper handling of codecs, media policies, and enterprise voice rules.

Moreover, delayed-offer SDP improves interoperability with external SIP peers or service providers. Many SIP trunks and devices expect the SDP to be offered at specific points in the signaling process. Early SDP offers may conflict with CUBE’s media rewriting or with service provider expectations, resulting in media delivery failures. Delayed-offer SDP guarantees that SDP exchange occurs after the signaling elements, including CUBE, have properly processed the call. This minimizes the risk of misrouted RTP, codec mismatches, or other media-related anomalies that could manifest as one-way audio.

Using delayed-offer SDP also supports consistency across distributed sites. In multi-site deployments, media paths can traverse multiple networks, firewalls, and NAT devices. Proper SDP timing ensures that the media is anchored at CUBE, with the correct IP addresses and ports communicated to each endpoint. This alignment is essential for calls involving remote sites, mobile endpoints, or external SIP peers, ensuring that both legs of the audio path are established before RTP flows. Without delayed-offer, premature SDP can result in media being sent to endpoints that have not yet been properly processed by CUBE, creating unpredictable one-way audio issues.

Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 3 Q31-45

Related posts: