Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 4 Q46-60

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Question46

During a CUCM deployment with multiple PSTN gateways, some outbound calls intermittently fail, showing fast busy signals despite available gateways. Logs indicate that CUCM attempts gateways sequentially, causing delays in call setup. Which configuration MOST effectively resolves this issue while maintaining redundancy?

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:

In multi-gateway CUCM environments, outbound call failures often occur due to sequential gateway attempts. When a route pattern is associated with multiple gateways, CUCM tries them in the order defined, and if the first gateways are busy, calls are delayed or fail, resulting in fast busy signals. Sequential attempts introduce latency in call setup and reduce call reliability.

Option B, implementing Local Route Groups (LRG), is the most effective solution. LRG dynamically selects the gateway based on the device pool of the calling endpoint, allowing CUCM to route calls immediately to the most appropriate gateway rather than sequentially testing all options. This reduces latency, prevents intermittent call failures, and maintains redundancy because alternate gateways remain available if the primary one fails. LRG implementation aligns with Cisco best practices for multi-gateway deployments and ensures efficient call routing.

Option A, removing all but one gateway, may reduce sequential attempts but sacrifices redundancy. If the single gateway fails, all outbound calls fail, which is unacceptable for high-availability environments.

Option C, time-of-day routing, directs calls based on schedules but does not resolve sequential gateway selection or reduce latency. It is suitable for load distribution or routing calls differently at off-peak times but does not address real-time call setup delays.

Option D, configuring weighted priorities in route lists, influences gateway preference but does not eliminate sequential search if the preferred gateway is busy. Weighted priorities partially optimize routing but do not solve the root cause of intermittent failures.

By implementing Local Route Groups, CUCM ensures efficient and reliable call routing, reduces latency, and maintains redundancy, directly addressing the issue of intermittent outbound call failures.

Question47

Remote Jabber clients connected through Mobile and Remote Access (MRA) report intermittent presence delays for internal users. Internal users’ presence updates are normal. XMPP logs between Expressway-C and CUCM show sporadic delays. 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:

Presence information in MRA relies on XMPP messages between Jabber clients and CUCM. Remote clients communicate through Expressway-C, and delays indicate that XMPP sessions are not persistent, causing intermittent propagation latency. Internal users do not experience issues because they communicate directly with CUCM.

Option B, enabling persistent XMPP connections, is the correct solution. Persistent connections maintain continuous XMPP sessions between Expressway-C and CUCM, ensuring real-time delivery of presence updates to remote clients. Persistent XMPP prevents the need to re-establish sessions for each update, reducing latency and missed presence notifications. Cisco best practices for MRA deployments emphasize persistent XMPP connections for reliable and immediate presence propagation.

Option A, increasing polling intervals, worsens the problem by delaying status updates. Longer intervals mean CUCM queries devices less frequently, increasing latency.

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

Option D, disabling traversal zones, would disconnect remote clients entirely, preventing MRA connectivity. Traversal zones are essential for remote users to communicate with CUCM through Expressway-C/E.

Persistent XMPP connections resolve delayed presence for remote users by maintaining continuous sessions, ensuring timely updates, improving reliability, and adhering to Cisco best practices for MRA.

Question48

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

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 and SDP negotiation. Mid-call re-INVITEs add participants to ongoing conferences. Delayed-offer trunks provide SDP in the 200 OK response rather than in the initial INVITE. Remote SIP endpoints may fail because CUCM lacks sufficient SDP information to establish media paths, particularly in distributed deployments with NAT or firewall traversal.

Option A, enabling early offer SIP, is the correct solution. Early offer includes SDP in the initial INVITE, providing CUCM with immediate media information for the remote participant. This ensures proper media path establishment, allowing reliable addition of remote endpoints to conferences. Cisco best practices recommend early offer SIP for conferencing trunks to maintain predictable behavior and prevent mid-call failures.

Option B, assigning hardware MTPs, addresses codec compatibility but does not solve SDP timing issues affecting remote participant addition.

Option C, reducing the number of SIP trunks, simplifies signaling but does not address the root cause, which is the timing of SDP negotiation.

Option D, converting remote endpoints to SCCP, is unnecessary. The issue relates to SDP timing and early offer requirements, not the endpoint protocol.

Enabling early offer SIP ensures proper media negotiation, allows remote SIP participants to join conferences successfully, and maintains reliable conferencing functionality in distributed environments.

Question49

A Cisco Unity Connection deployment integrated with CUCM experiences several seconds of silence at the start of voicemail playback. RTP flow shows no jitter or packet loss. 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 in voicemail playback is commonly caused by Voice Activity Detection (VAD), which suppresses RTP packets when it detects low-energy audio. VAD is intended to save bandwidth during calls but can misinterpret the beginning of voicemail messages as silence. This results in several seconds of no audio at message start.

Option A, disabling VAD on the SIP trunk, is the correct solution. This ensures continuous RTP transmission, including low-energy audio, eliminating initial silence. Cisco best practices recommend disabling VAD in voicemail deployments to guarantee smooth, uninterrupted playback. Users experience immediate audio at the start of messages, improving satisfaction. The bandwidth impact is minimal because voicemail RTP traffic is low compared to live calls.

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

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

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

Disabling VAD directly resolves the issue, ensuring immediate and uninterrupted voicemail playback, consistent with Cisco best practices.

Question50

Remote users report intermittent one-way audio when calling internal endpoints through a CUBE. RTP analysis shows 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 in multi-site SIP deployments with CUBE. The issue arises when RTP is sent before CUBE can rewrite SDP, causing packets to be sent to incorrect IP addresses. The root cause is early SDP negotiation before the media path is correctly anchored.

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

Option B, using UDP instead of TCP, changes transport but does not resolve SDP timing or media path misalignment.

Option C, disabling early media, prevents certain call progress tones but does not solve RTP misdelivery. It addresses symptoms rather than the root cause.

Option D, enabling symmetric RTP, assists with NAT traversal but does not correct early media being sent to incorrect addresses. Symmetric RTP is not a solution for SDP timing issues.

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

Question51

A CUCM administrator observes that outbound calls to a remote site intermittently fail despite multiple available gateways. The logs indicate that CUCM attempts each gateway sequentially, resulting in delayed call setup and occasional fast busy signals. Which configuration MOST effectively resolves this problem while maintaining high availability?

Answer: B

Explanation:

In CUCM deployments with multiple PSTN gateways, sequential gateway attempts can lead to intermittent call failures. When a route pattern is associated with several gateways, CUCM tries each gateway one by one until the call completes. If the first gateways are busy, call setup is delayed, and users may encounter fast busy signals. Sequential gateway selection introduces latency and reduces overall call reliability.

Option B, implementing Local Route Groups (LRG), is the most effective solution. LRG allows CUCM to dynamically select a gateway based on the device pool of the calling endpoint. This means the call is routed immediately to the most appropriate gateway without sequentially testing others, reducing latency and preventing intermittent failures. Redundancy is maintained because alternate gateways remain available if the primary gateway is down. This approach adheres to Cisco best practices for multi-gateway environments, providing efficient and reliable call routing.

Option A, removing all but one gateway, reduces sequential attempts but sacrifices redundancy. If the single gateway fails, all outbound calls fail, which is unacceptable for high-availability environments.

Option C, using time-of-day routing, allows calls to be routed differently based on schedules but does not address sequential gateway attempts or latency. While useful for traffic distribution, it does not resolve real-time call setup issues.

Option D, configuring weighted priorities in route lists, influences which gateway CUCM prefers, but sequential attempts still occur if the preferred gateway is busy. Weighted priorities partially optimize routing but do not eliminate the root cause of intermittent call failures.

By implementing LRG, CUCM ensures efficient routing, maintains redundancy, reduces call setup delays, and addresses the root cause of intermittent outbound call failures in multi-gateway deployments.

Question52

Remote Jabber clients connected via Mobile and Remote Access (MRA) report delayed presence updates for internal users. Internal users’ presence is functioning normally. XMPP logs between Expressway-C and CUCM show intermittent delays. Which configuration MOST effectively resolves this issue?

Answer: B

Explanation:

Presence information in MRA relies on XMPP messages exchanged between Jabber clients and CUCM. Remote clients communicate through Expressway-C, and delays suggest XMPP sessions are not persistent, causing intermittent latency. 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 maintain continuous XMPP sessions between Expressway-C and CUCM, ensuring that presence updates reach remote clients in real-time. Persistent sessions reduce latency, prevent missed updates, and eliminate the need to re-establish XMPP sessions for each update. Cisco best practices recommend persistent XMPP for MRA deployments to guarantee reliable presence propagation. With persistent connections, remote users experience timely presence updates, improving communication efficiency.

Option A, increasing polling intervals, would worsen delays. Polling determines how often CUCM queries devices for status; increasing intervals results in slower updates.

Option C, blocking non-essential firewall ports, risks preventing necessary XMPP traffic from reaching remote clients. While security is important, indiscriminate blocking can disrupt legitimate communications and exacerbate delays.

Option D, disabling traversal zones, would disconnect remote clients entirely. Traversal zones are necessary for MRA clients to reach CUCM through Expressway-C/E, so disabling them does not solve the underlying problem.

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

Question53

In a distributed CUCM SIP conferencing environment, internal participants join conferences successfully, but remote SIP endpoints fail intermittently when added mid-call. 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 involves SIP conferencing and SDP negotiation. 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 fail to join because CUCM does not have immediate SDP information, especially in distributed deployments where NAT or firewall traversal may complicate media paths.

Option A, enabling early offer SIP, is the correct solution. Early offer includes SDP in the initial INVITE, giving CUCM immediate knowledge of the remote participant’s media capabilities. This ensures proper media path negotiation and allows remote endpoints to join conferences reliably. Cisco best practices recommend early offer SIP for trunks handling conferencing traffic to prevent mid-call failures. Early offer improves media negotiation efficiency, reduces call setup delays, and ensures predictable behavior for remote participants.

Option B, assigning hardware MTPs, addresses codec compatibility but does not solve the timing of SDP negotiation, which is the root cause of failures.

Option C, reducing SIP trunks, simplifies signaling but does not address SDP timing. The number of trunks is irrelevant to mid-call participant addition failures caused by SDP misalignment.

Option D, converting remote endpoints to SCCP, is unnecessary. The issue is related to SDP timing, not endpoint protocol compatibility. Protocol conversion does not resolve the problem.

By enabling early offer SIP, CUCM ensures proper SDP negotiation, allows remote participants to join conferences successfully, and maintains reliable conferencing functionality in distributed deployments.

Question54

Answer: A

Explanation:

Initial silence at the start of voicemail messages is commonly caused by Voice Activity Detection (VAD). VAD suppresses RTP packets when it detects low-energy audio, which saves bandwidth during calls but can misinterpret the beginning of voicemail messages as silence. This results in several seconds of audio not being transmitted, creating the perception of delayed playback.

Option A, disabling VAD on the SIP trunk, is the correct solution. This ensures continuous RTP transmission, including low-energy audio at the start of messages. Cisco best practices for CUCM and Unity Connection integration recommend disabling VAD for voicemail to prevent playback gaps. Users hear voicemail messages immediately, improving satisfaction and eliminating complaints about delayed audio. The bandwidth impact is minimal since voicemail RTP traffic is limited compared to live calls.

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

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

Option D, moving the voicemail pilot to a different partition, changes call routing but does not impact media transmission or RTP flow.

Disabling VAD directly resolves initial silence in voicemail, ensuring smooth playback from the start of the message and aligning with Cisco best practices for CUCM and Unity Connection deployments.

Question55

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 issues in multi-site SIP deployments with CUBE often occur when RTP is transmitted before CUBE rewrites SDP, causing packets to be sent to incorrect IP addresses. This happens because early SDP negotiation occurs before media path anchoring is complete.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer ensures SDP is sent in the 200 OK response rather than the initial INVITE, allowing CUBE to anchor media correctly, rewrite 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 misalignment. This configuration ensures correct RTP delivery, eliminates one-way audio, and maintains reliable communication between remote and internal endpoints.

Option B, using UDP instead of TCP, changes the transport protocol but does not solve SDP timing or media path misalignment.

Option C, disabling early media, may prevent some call progress tones but does not fix RTP misdelivery. It addresses a symptom rather than the underlying SDP problem.

Option D, enabling symmetric RTP, helps with NAT traversal but does not correct early media being sent to incorrect addresses. Symmetric RTP does not address the SDP timing root cause.

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

Question56

A CUCM administrator notices that outbound calls intermittently fail, even though multiple gateways are available. Analysis shows that CUCM attempts gateways sequentially, causing fast busy signals and delayed call setup. Which configuration MOST effectively resolves this issue while ensuring high availability?

Answer: B

Explanation:

In multi-gateway CUCM deployments, sequential gateway attempts often result in intermittent outbound call failures. When a route pattern is assigned multiple gateways, CUCM tries each gateway in order until the call completes successfully. If the first gateways in the sequence are busy, call setup is delayed, leading to fast busy signals and reduced user satisfaction. Sequential attempts introduce latency, and frequent failures may occur during peak calling hours, impacting both operational efficiency and reliability.

Option B, implementing Local Route Groups (LRG), is the most effective solution. LRG dynamically selects the gateway based on the device pool of the calling endpoint. Instead of sequentially testing all gateways, CUCM immediately routes calls to the most appropriate gateway, reducing latency and improving call reliability. LRG preserves redundancy because alternate gateways remain available if the primary gateway is unavailable, ensuring high availability. This approach aligns with Cisco best practices for multi-gateway deployments, combining efficiency, reliability, and fault tolerance.

Option A, removing all but one gateway, reduces sequential attempts but eliminates redundancy. If the single remaining gateway fails, all outbound calls will be blocked, which is unacceptable for a high-availability environment.

Option C, time-of-day routing, is useful for distributing traffic based on schedules but does not address sequential gateway selection. It may optimize traffic flow during off-peak hours but does not reduce call setup latency or prevent intermittent failures in real-time.

Option D, configuring weighted priorities in route lists, influences which gateway CUCM prefers but does not eliminate sequential searching. If the preferred gateway is busy, CUCM will still attempt others sequentially, leading to the same intermittent failure scenario. Weighted priorities partially optimize routing but do not solve the fundamental problem.

By implementing LRG, CUCM ensures efficient call routing, maintains redundancy, minimizes latency, and addresses the root cause of intermittent outbound call failures. This solution is scalable, supports complex multi-gateway deployments, and adheres to Cisco best practices for high-availability enterprise telephony environments.

Question57

Remote Jabber clients connected via Mobile and Remote Access (MRA) report intermittent delays in presence updates for internal users. Internal users’ presence updates are normal. XMPP logs between Expressway-C and CUCM indicate sporadic delays. Which configuration MOST effectively resolves this problem?

Answer: B

Explanation:

Presence communication for remote Jabber clients in MRA relies on XMPP messages exchanged through Expressway-C to CUCM. Delays in presence updates suggest that XMPP sessions are not persistent, causing intermittent propagation issues. Internal users do not experience delays because their clients communicate directly with CUCM. Reliable presence information is critical for collaboration and user awareness, and delays can impact productivity, user experience, and real-time communication decisions.

Option B, enabling persistent XMPP connections, is the correct solution. Persistent connections maintain continuous XMPP sessions between Expressway-C and CUCM, ensuring that presence updates are delivered in real-time. Persistent XMPP eliminates the need to re-establish sessions for each update, reducing latency and preventing missed presence notifications. Cisco best practices for MRA deployments recommend persistent XMPP to guarantee reliable communication. By keeping sessions open, remote users receive immediate presence updates, improving responsiveness and collaboration efficiency.

Option A, increasing polling intervals, worsens the problem. Polling controls how frequently CUCM queries devices for status. Increasing intervals slows updates, making presence delays worse.

Option C, blocking non-essential firewall ports, could inadvertently prevent necessary XMPP traffic from reaching remote clients. Security measures should be carefully implemented to avoid disrupting essential communication.

Option D, disabling traversal zones, would disconnect remote clients entirely, preventing them from communicating with CUCM via Expressway-C/E. Traversal zones are critical for MRA connectivity.

Persistent XMPP connections directly address the root cause of delayed presence for remote users, ensure real-time updates, improve reliability, and follow Cisco best practices for MRA deployments.

Question58

In a distributed CUCM SIP conferencing environment, internal participants can join without issues, but remote SIP endpoints fail intermittently when added mid-call. 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 involves SIP conferencing and SDP negotiation. 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, which is critical in distributed deployments with NAT or firewall traversal.

Option A, enabling early offer SIP, is the correct solution. Early offer includes SDP in the initial INVITE, providing CUCM with immediate media capability information for the remote participant. This allows proper media path establishment, ensuring reliable addition of remote participants to conferences. Cisco best practices recommend early offer SIP for conferencing trunks to prevent mid-call failures and maintain predictable media behavior. Early offer ensures media negotiation occurs efficiently, reducing call setup delays and preventing intermittent failures for remote endpoints.

Option B, assigning hardware MTPs, ensures codec compatibility but does not resolve SDP timing issues, which are the primary cause of mid-call participant failures.

Option C, reducing SIP trunks, simplifies signaling but does not address SDP negotiation timing. The root cause is media path establishment, not the number of trunks.

Option D, converting remote endpoints to SCCP, is unnecessary. The issue is SDP-related, not protocol-related, and protocol conversion does not fix mid-call join failures.

By enabling early offer SIP, CUCM ensures proper media negotiation, allows remote participants to join conferences reliably, and maintains predictable conferencing behavior across distributed deployments.

Question59

A Cisco Unity Connection deployment integrated with CUCM experiences several seconds of silence at the start of voicemail playback. RTP analysis shows no packet loss or jitter. Which configuration MOST effectively resolves this issue?

Answer: A

Explanation:

Initial silence in voicemail playback is typically caused by Voice Activity Detection (VAD). VAD suppresses RTP packets when low-energy audio is detected, conserving bandwidth during calls. However, the beginning of voicemail messages often has low-energy audio, which VAD interprets as silence. This results in several seconds of audio not being transmitted, creating delayed 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 avoid playback gaps. Users experience immediate audio at message start, improving user satisfaction. The bandwidth impact is minimal because voicemail RTP traffic is limited compared to live calls.

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

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

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

Disabling VAD directly resolves the issue, ensuring uninterrupted voicemail playback from the start and aligning with Cisco best practices for CUCM and Unity Connection deployments.

Question60

Remote users report intermittent one-way audio when calling internal endpoints through a CUBE. RTP analysis shows packets 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:

One-way audio in multi-site SIP deployments using CUBE is often caused by early RTP transmission before CUBE rewrites SDP, sending packets to incorrect addresses. The issue originates from early SDP negotiation prior to media path anchoring.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer ensures that SDP is included in the 200 OK response rather than the initial INVITE. This allows CUBE to anchor media correctly, rewrite 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 may cause media misalignment. 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 or media misalignment issues.

Option C, disabling early media, prevents some call progress tones but does not fix RTP misdelivery. It addresses symptoms rather than the root cause.

Option D, enabling symmetric RTP, assists with NAT traversal but does not correct early media being sent to incorrect addresses. Symmetric RTP does not address the SDP timing issue.

Delayed-offer SDP ensures proper media anchoring through CUBE, eliminates one-way audio, and maintains reliable multi-site SIP communication, adhering to Cisco best practices.

One-way audio is a common and complex issue in multi-site Cisco Unified Communications deployments, particularly when SIP calls traverse a Cisco Unified Border Element (CUBE). This problem occurs when the RTP media path is not properly aligned with the signaling path. In a typical SIP session, the endpoints negotiate media using the Session Description Protocol (SDP), which is included in the SIP messages. The SDP specifies codec types, media formats, IP addresses, and port numbers for the RTP streams. In many cases, the initial INVITE from CUCM to CUBE contains the SDP, which can cause RTP streams to be sent before CUBE has had the opportunity to anchor and rewrite the media paths. Early RTP transmission to incorrect IP addresses leads to one-way audio conditions. The root cause of this problem is the timing of SDP exchange relative to media path establishment.

Delayed-offer SDP is a configuration that solves this issue by ensuring that CUCM does not include the SDP in the initial INVITE. Instead, CUCM sends SDP in the 200 OK response after the signaling path has been fully processed. This allows CUBE to rewrite the media addresses and port numbers, anchor the call correctly, and establish a proper media path before RTP flows. By deferring the SDP offer until after call processing, CUBE has full control of the media path, ensuring that both sides of the call receive RTP packets at the correct destinations. Cisco recommends delayed-offer SDP as a best practice for trunks traversing CUBE when early media can result in misaligned media paths. This approach directly addresses the root cause of one-way audio by synchronizing media path establishment with signaling, rather than merely addressing symptoms.

Implementing delayed-offer SDP provides several operational benefits. First, it improves the predictability of media flows across multi-site deployments. In distributed environments, RTP often traverses multiple networks, including private WANs, firewalls, and NAT devices. By anchoring media at CUBE after SDP negotiation, administrators can ensure consistent and reliable media routing. This predictability is critical for troubleshooting, performance monitoring, and maintaining call quality across sites. Packet captures and call detail records reflect accurate media paths, which significantly simplifies the process of diagnosing audio issues. Without delayed-offer SDP, RTP may be sent to addresses specified in early SDP, leading to inconsistent call behavior and complicating problem resolution.

Second, delayed-offer SDP enhances interoperability with other SIP entities, including service providers and third-party devices. Many SIP providers expect SDP to be offered at a particular stage of the call setup process. Early SDP included in the initial INVITE may conflict with provider expectations or with the media rewriting logic of CUBE. By using delayed-offer, CUCM ensures that the SDP is offered only after the signaling elements have processed the call, reducing the likelihood of misrouted RTP, codec mismatches, or unsupported media features. This is particularly important in environments where CUCM interconnects with multiple service providers or cloud-based SIP trunks, ensuring a seamless and reliable audio experience for users.

Option B, using SIP over UDP instead of TCP, does not address the root cause of the one-way audio issue. UDP and TCP are transport protocols with different characteristics: UDP is connectionless and faster, while TCP provides reliable, ordered delivery. However, the problem in this scenario is not related to transport reliability or connection behavior; it is related to the sequence of SDP negotiation relative to the media path. Switching the transport protocol may slightly alter message delivery times but does not prevent early RTP from being sent to an incorrect destination. Therefore, UDP cannot resolve the underlying issue of premature media delivery.

Option C, disabling early media on CUBE, may prevent users from hearing call progress tones, announcements, or ringback audio before the call is answered. While this might reduce some symptoms associated with one-way audio, it does not correct the SDP timing issue that causes RTP to be misrouted. Disabling early media addresses only the symptom, not the root cause, and may negatively affect the user experience because callers may no longer hear expected audio feedback. Early media suppression does not allow CUBE to anchor media correctly, and as a result, RTP packets may still fail to reach the correct endpoint. This approach does not guarantee a permanent resolution for one-way audio in multi-site SIP deployments.

Option D, enabling symmetric RTP on CUBE, is designed to handle NAT traversal scenarios by ensuring that RTP packets are sent back to the source IP address and port from which they were received. While this feature is useful for specific network topologies where endpoints reside behind NAT or firewalls, it does not resolve the timing issue associated with early SDP. One-way audio in this scenario is caused by early RTP delivery before media anchoring occurs, not by asymmetrical paths or NAT issues. Symmetric RTP does not modify SDP timing and therefore cannot prevent RTP from being sent to an incorrect address prior to CUBE processing. While symmetric RTP can be beneficial in certain deployments, it is not a solution for SDP-related one-way audio problems.

Implementing delayed-offer SDP also simplifies troubleshooting and operational management in enterprise environments. With a predictable sequence—INVITE without SDP, followed by SDP in the 200 OK—network administrators can more easily trace and verify media paths, ensuring alignment between signaling and media. This predictability reduces the complexity of diagnosing audio issues in multi-site deployments, where media may traverse multiple WAN links, firewalls, and NAT devices. Additionally, it ensures that any media policies, call routing rules, or codec manipulations applied at CUBE are fully enforced before RTP flows, maintaining quality of service and adherence to enterprise communication standards.

Delayed-offer SDP also enhances call reliability in scenarios with distributed sites or hybrid architectures. In large enterprise deployments, calls may traverse multiple locations, data centers, or cloud services. By delaying SDP, CUCM ensures that CUBE anchors the media path correctly, which is critical for maintaining consistent audio quality across geographically dispersed endpoints. This alignment also reduces the likelihood of intermittent audio issues or asymmetric audio problems that may arise when early SDP results in inconsistent media addresses.

Furthermore, delayed-offer SDP aligns with Cisco’s recommended best practices for SIP trunk deployments, particularly in multi-site environments. Cisco guidelines emphasize deterministic media path establishment, proper anchoring, and predictable signaling behavior. By configuring delayed-offer SDP, organizations adhere to these principles, ensuring robust call handling, high-quality audio, and compliance with recommended architectural patterns. This configuration is particularly important in enterprises that require guaranteed call quality and reliability across internal and remote sites, as well as with external SIP peers or service providers.

Delayed-offer SDP also contributes to overall network stability and performance. By controlling when media negotiation occurs, CUCM and CUBE can efficiently manage network resources, ensuring that RTP flows are correctly directed and that media streams do not traverse unintended paths. This prevents unnecessary load on WAN links, avoids dropped or misrouted packets, and ensures a consistent quality of experience for end users. In addition, it enables network teams to implement monitoring and alerting systems based on predictable media patterns, facilitating proactive maintenance and rapid resolution of potential issues.

It ensures proper media anchoring, prevents early RTP from being sent to incorrect endpoints, and maintains call quality and reliability. Other options, including changing transport protocols, disabling early media, or enabling symmetric RTP, do not address the underlying SDP timing issue and cannot reliably resolve one-way audio. Delayed-offer SDP provides a comprehensive, standards-aligned solution for ensuring correct media paths, seamless interoperability, and adherence to Cisco best practices in complex SIP deployments. This approach guarantees that audio flows correctly across all sites, that signaling and media are synchronized, and that enterprise communication systems operate with maximum reliability and efficiency.

Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 4 Q46-60

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