Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 5 Q61-75

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

Question61

A CUCM administrator observes intermittent failures for outbound calls to remote sites, even though multiple PSTN gateways are available. Logs show that CUCM sequentially attempts each gateway, leading to delayed call setup and occasional fast busy signals. Which configuration MOST effectively resolves this issue while ensuring high availability?

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 deployments, sequential gateway attempts are a common cause of intermittent outbound call failures. When a route pattern includes multiple gateways, CUCM evaluates each gateway in the order configured until the call is successfully routed. If the initial gateways are busy or temporarily unavailable, call setup is delayed, and users may experience fast busy signals. This behavior results in an inefficient call routing mechanism that reduces both reliability and user satisfaction. The root cause is the sequential search, which introduces latency and does not optimally leverage the available gateways.

Option B, implementing Local Route Groups (LRG), is the most effective solution. LRG allows CUCM to dynamically select the most appropriate gateway based on the device pool of the calling endpoint. Instead of sequentially attempting all gateways, CUCM routes calls directly to the optimal gateway, minimizing latency and eliminating intermittent failures. Redundancy is preserved because alternate gateways remain available if the primary gateway becomes unavailable, ensuring high availability. LRG aligns with Cisco best practices for multi-gateway deployments, providing a scalable and efficient method to optimize call routing, improve fault tolerance, and maintain predictable call setup times across complex telephony environments.

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, creating a single point of failure that is unacceptable in enterprise environments requiring high availability and fault tolerance.

Option C, time-of-day routing, allows calls to be routed differently based on schedules but does not address sequential gateway attempts or latency. While time-of-day routing can optimize traffic distribution, it does not solve real-time call setup delays or prevent intermittent call failures caused by sequential attempts.

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 provide partial optimization but do not address the fundamental problem of sequential evaluation.

Implementing LRG ensures efficient call routing, maintains redundancy, reduces call setup delays, and addresses the underlying cause of intermittent outbound call failures, providing a scalable, fault-tolerant solution aligned with Cisco best practices for high-availability enterprise telephony.

Question62

Remote Jabber clients using Mobile and Remote Access (MRA) report intermittent delays in receiving presence updates for internal users. Internal users’ presence updates function normally. XMPP logs between Expressway-C and CUCM show inconsistent message delivery timing. 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 for remote Jabber clients in MRA relies on XMPP messages exchanged via Expressway-C with CUCM. Delays in presence updates indicate that XMPP sessions are not persistent, leading to intermittent propagation latency. Internal users are unaffected because their clients communicate directly with CUCM, bypassing Expressway-C. Reliable presence is critical for collaboration, decision-making, and user awareness, and delays can negatively impact productivity and real-time communication.

Option B, enabling persistent XMPP connections, is the correct solution. Persistent connections maintain continuous XMPP sessions between Expressway-C and CUCM, ensuring timely delivery of presence updates to remote clients. By keeping sessions open, CUCM does not need to re-establish connections for each update, reducing latency and preventing missed notifications. Cisco best practices for MRA deployments emphasize persistent XMPP for reliable presence propagation. With persistent connections, remote users receive immediate updates, improving communication efficiency and user experience.

Option A, increasing polling intervals, worsens delays. Polling determines how frequently CUCM queries devices for status, and longer intervals result in slower updates, increasing latency for remote users.

Option C, blocking non-essential firewall ports, risks preventing necessary XMPP traffic from reaching remote clients. Proper security configurations are important, but indiscriminate blocking may disrupt essential communication and exacerbate delays.

Option D, disabling traversal zones, would disconnect remote clients entirely, preventing them from communicating with CUCM via Expressway-C/E. Traversal zones are essential for enabling secure remote access for Jabber clients.

Persistent XMPP connections directly resolve delayed presence issues for remote users, ensuring real-time updates, reducing latency, and improving collaboration reliability while following Cisco best practices for MRA deployments.

Question63

In a distributed CUCM SIP conferencing environment, internal participants successfully join conferences, but remote SIP endpoints intermittently fail when added mid-call. 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 are used to add participants to ongoing conferences. Delayed-offer trunks provide SDP in the 200 OK response rather than the initial INVITE. Remote SIP endpoints may fail to join because CUCM does not have immediate SDP information to establish the media path, 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, allowing CUCM to immediately know 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 conferencing trunks to maintain predictable media behavior, prevent mid-call failures, and reduce latency during participant addition. Early offer ensures efficient media negotiation, minimizes call setup delays, and eliminates intermittent join failures for remote participants, maintaining conference integrity and quality.

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

Option C, reducing SIP trunks, simplifies signaling but does not address SDP negotiation timing. The number of trunks does not impact the root cause of failed media path establishment during mid-call participant additions.

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

Enabling early offer SIP guarantees proper media negotiation, reliable addition of remote participants, and predictable conference behavior across distributed deployments, fully aligning with Cisco best practices for SIP conferencing.

Question64

A Cisco Unity Connection deployment integrated with CUCM experiences several seconds of silence at the beginning of voicemail playback. RTP analysis shows normal flow without 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 at the start of voicemail playback is commonly caused by Voice Activity Detection (VAD). VAD suppresses RTP packets when low-energy audio is detected, conserving bandwidth during live calls. However, voicemail messages often start with low-energy audio, which VAD interprets as silence, causing several seconds of no audio at playback start. This impacts user experience and may lead to complaints about delayed message 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, eliminating initial playback silence. Cisco best practices recommend disabling VAD for voicemail deployments to provide seamless and uninterrupted message playback. Users hear messages from the very beginning, improving satisfaction and avoiding confusion caused by missing initial content. The bandwidth impact is minimal because voicemail traffic is lower in volume compared to live call traffic.

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

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

Option D, moving the voicemail pilot to a different partition, changes call routing but has no effect on media delivery or RTP flow.

Disabling VAD directly addresses the initial silence problem, ensuring smooth and uninterrupted voicemail playback, enhancing user experience, and complying with Cisco best practices for CUCM and Unity Connection integration.

Question65

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:

One-way audio in multi-site SIP deployments with CUBE often results from early RTP transmission before CUBE rewrites SDP. Packets may be sent to incorrect addresses, causing audio issues. This is primarily caused by early SDP negotiation before proper media path anchoring.

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

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

Option C, disabling early media, prevents some call progress tones but does not resolve RTP misdelivery. It treats 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 destinations. Symmetric RTP does not solve the SDP timing issue causing one-way audio.

By implementing delayed-offer SDP, media is anchored correctly through CUBE, one-way audio is eliminated, and communication remains reliable across multi-site SIP deployments, fully adhering to Cisco best practices.

Question66

A CUCM administrator notices that outbound calls to multiple remote sites intermittently fail despite several available gateways. Analysis shows CUCM sequentially attempts gateways, leading to delayed call setup and occasional fast busy signals. Which configuration MOST effectively resolves this problem while maintaining high availability?

Answer: B

Explanation:

In environments where multiple PSTN gateways are available, CUCM uses route patterns to determine which gateway should be used for outbound calls. When multiple gateways are associated with a route pattern, CUCM by default attempts each gateway sequentially until the call completes successfully. This sequential attempt process can result in delayed call setup and, in some cases, fast busy signals if the initial gateways are busy or temporarily unavailable. This problem becomes more pronounced during peak calling periods, leading to inconsistent user experiences and operational inefficiencies. The underlying issue is the sequential evaluation of gateways rather than a lack of available gateways.

Option B, implementing Local Route Groups (LRG), is the optimal solution. LRG allows CUCM to dynamically select gateways based on the device pool of the calling endpoint. With LRG, calls are immediately routed to the appropriate gateway without the need to sequentially test each option, reducing latency and ensuring reliable call delivery. High availability is preserved because alternate gateways remain available if the primary gateway fails. LRG is a scalable, best-practice solution for large enterprises with multiple gateways, as it optimizes resource utilization and reduces the likelihood of call setup failures while maintaining redundancy and fault tolerance.

Option A, removing all but one gateway, would eliminate sequential attempts but introduces a single point of failure. If the lone gateway fails, all outbound calls would be blocked, compromising high availability and operational resilience.

Option C, implementing time-of-day routing, redistributes calls based on schedules but does not address sequential gateway attempts. Time-of-day routing may optimize traffic flow during different periods but does not reduce call setup latency or prevent intermittent call failures in real-time.

Option D, configuring weighted priorities within route lists, influences which gateway CUCM prefers but does not eliminate sequential attempts. If the preferred gateway is busy, CUCM still attempts other gateways sequentially, so the root cause of delays and intermittent failures remains unresolved.

Using LRG ensures efficient call routing, reduces latency, maintains redundancy, and aligns with Cisco best practices for multi-gateway deployments, providing a robust and fault-tolerant solution for enterprise telephony environments.

Question67

Remote Jabber clients connected through Mobile and Remote Access (MRA) report delayed presence updates for internal users. Internal users’ presence functions normally. XMPP logs between Expressway-C and CUCM indicate inconsistent delivery. Which configuration MOST effectively resolves this problem?

Answer: B

Explanation:

Presence information in MRA environments relies on XMPP messages exchanged between Jabber clients and CUCM through Expressway-C. When presence updates are delayed for remote users but not for internal users, this typically indicates that XMPP sessions are not persistent, causing intermittent propagation issues. Internal users communicate directly with CUCM without traversing Expressway-C, which is why their presence information updates immediately. Presence delays for remote users can negatively impact real-time collaboration, decision-making, and overall user satisfaction.

Option B, enabling persistent XMPP connections between Expressway-C and CUCM, is the correct solution. Persistent connections maintain continuous XMPP sessions, ensuring that presence updates are delivered promptly to remote clients. This eliminates the need to repeatedly establish sessions for each update, reducing latency and preventing missed notifications. Cisco best practices recommend persistent XMPP connections for MRA deployments to guarantee reliable presence propagation. By keeping XMPP sessions open, CUCM ensures that remote users receive timely presence information, enhancing collaboration, communication efficiency, and the overall user experience.

Option A, increasing polling intervals, would worsen delays. Polling frequency determines how often CUCM queries devices for status. Increasing intervals slows updates, exacerbating latency for remote users.

Option C, blocking non-essential firewall ports, may inadvertently prevent necessary XMPP traffic from reaching remote clients. Proper security measures must be implemented carefully to avoid disrupting essential communication.

Option D, disabling traversal zones, would disconnect remote clients entirely. Traversal zones are essential for secure remote access, allowing Jabber clients to communicate with CUCM through Expressway-C/E.

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

Question68

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 where mid-call re-INVITEs are used to add participants to ongoing conferences. Delayed-offer trunks send SDP information in the 200 OK response rather than in the initial INVITE. Remote SIP endpoints may fail to join because CUCM does not have immediate SDP information, particularly in distributed deployments with NAT or firewall traversal. Immediate SDP information is essential for establishing the correct media path and ensuring successful conference participation.

Option A, enabling early offer SIP, is the correct solution. Early offer SIP includes SDP in the initial INVITE, allowing CUCM to know the remote participant’s media capabilities immediately. 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, maintain predictable media behavior, and reduce latency during participant addition. Early offer ensures efficient media negotiation, minimizes call setup delays, and eliminates intermittent join failures for remote endpoints, preserving conference integrity and quality.

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

Option C, reducing SIP trunks, simplifies signaling but does not solve SDP negotiation timing. The number of trunks does not impact the core issue of media path establishment for mid-call participants.

Option D, converting remote endpoints to SCCP, is unnecessary. The issue is SDP-related, not protocol-related, so protocol conversion does not address the failure.

Question69

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

Answer: A

Explanation:

Initial silence at the start of voicemail playback is commonly caused by Voice Activity Detection (VAD). VAD suppresses RTP packets during periods of low-energy audio, which conserves bandwidth during live calls. However, the beginning of voicemail messages typically contains low-energy audio, which VAD interprets as silence, resulting in several seconds of missing audio. This creates a poor user experience and may lead to complaints regarding delayed message 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, eliminating the initial playback silence. Cisco best practices recommend disabling VAD for voicemail deployments to ensure seamless playback from the very start of the message. Users immediately hear the voicemail, improving satisfaction and avoiding confusion caused by missing content. The bandwidth impact is minimal because voicemail RTP traffic is small compared to live call traffic.

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

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

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

Disabling VAD directly resolves the issue, ensuring smooth voicemail playback, enhancing the user experience, and following Cisco best practices for CUCM and Unity Connection integration.

Question70

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:

One-way audio issues in multi-site SIP deployments using CUBE often arise when RTP is transmitted before CUBE rewrites SDP, causing packets to be sent to incorrect IP addresses. Early SDP negotiation without proper media anchoring results in misdirected RTP, which prevents audio from being heard by one side. The root cause is the timing of SDP delivery relative to CUBE’s media anchoring process.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP ensures that SDP is included in the 200 OK response rather than the initial INVITE. This allows CUBE to properly anchor media, rewrite addresses and ports, and establish a correct 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 guarantees accurate 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 resolve SDP timing or media misalignment.

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

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 SDP timing, which is the root cause of the one-way audio.

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

Question71

A CUCM administrator observes intermittent outbound call failures for remote sites, despite having multiple gateways. CUCM logs indicate sequential attempts on all available gateways, causing call setup delays and fast busy signals during peak hours. Which configuration MOST effectively resolves this issue while ensuring high availability?

Answer: B

Explanation:

In enterprise telephony environments, CUCM is responsible for routing outbound calls to various gateways based on route patterns, partitions, and device pools. When multiple gateways are associated with a route pattern, CUCM attempts them sequentially by default. This sequential evaluation introduces latency and can result in fast busy signals when the first few gateways are busy or temporarily unavailable. Sequential attempts, particularly during high-traffic periods, can cause inconsistent call behavior, delayed call setup, and user frustration. Sequential evaluation is the root cause of these intermittent failures, as it does not leverage the redundancy provided by multiple gateways efficiently.

Option B, implementing Local Route Groups (LRG), is the most effective solution. LRG dynamically selects gateways based on the device pool of the calling endpoint, allowing CUCM to route calls directly to the optimal gateway without sequentially testing each one. By immediately selecting the appropriate gateway, LRG reduces call setup time, eliminates fast busy signals, and ensures consistent call success. High availability is preserved because alternate gateways remain available if the primary gateway is unavailable. LRG also scales efficiently in large deployments, allowing administrators to manage multi-gateway environments without introducing unnecessary complexity. Cisco best practices recommend LRG for multi-gateway deployments to optimize routing, maintain redundancy, and reduce call setup delays.

Option A, removing all but one gateway, reduces sequential attempts but creates a single point of failure. If the remaining gateway fails, all outbound calls are blocked, compromising high availability and operational reliability.

Option C, using time-of-day routing, redistributes calls based on schedules but does not address sequential gateway attempts. Time-of-day routing optimizes traffic flow during specific periods but does not prevent call setup delays or intermittent failures caused by sequential evaluation.

Option D, configuring weighted priorities in route lists, influences gateway selection preference but does not eliminate sequential attempts. If the preferred gateway is busy, CUCM still attempts other gateways sequentially, so the underlying problem remains unresolved. Weighted priorities are useful for traffic distribution but are insufficient to prevent intermittent call failures in real-time.

Implementing LRG ensures efficient call routing, reduces latency, maintains redundancy, and adheres to Cisco best practices for high-availability multi-gateway deployments. This approach addresses the root cause of intermittent outbound call failures and ensures predictable and reliable call performance, which is essential for enterprise telephony environments.

Question72

Remote Jabber clients connected through Mobile and Remote Access (MRA) report delayed presence updates for internal users, whereas internal users’ presence updates are immediate. XMPP logs between Expressway-C and CUCM show inconsistent delivery timing. Which configuration MOST effectively resolves this problem?

Answer: B

Explanation:

Presence information is a fundamental aspect of unified communications, enabling users to see availability and status of colleagues in real-time. In MRA deployments, remote Jabber clients rely on XMPP messages exchanged via Expressway-C to CUCM for presence updates. When remote clients experience delays but internal users do not, it indicates that XMPP sessions may not be persistent, leading to intermittent delivery issues. Internal users communicate directly with CUCM, so they are unaffected by traversal or session persistence problems. Delays in presence updates can affect collaboration efficiency, decision-making, and overall user satisfaction.

Option B, enabling persistent XMPP connections between Expressway-C and CUCM, is the correct solution. Persistent XMPP connections maintain continuous sessions, ensuring immediate delivery of presence updates to remote clients. This approach eliminates the need to repeatedly establish sessions, which introduces latency and causes delayed updates. Persistent connections are recommended by Cisco best practices for MRA environments because they provide reliable, real-time presence information. By keeping XMPP sessions open, remote users receive timely presence notifications, improving collaboration, communication efficiency, and user experience.

Option A, increasing polling intervals, worsens delays. Polling determines how frequently CUCM queries devices for status; increasing the interval reduces update frequency, exacerbating latency for remote users.

Option C, blocking non-essential firewall ports, may inadvertently prevent necessary XMPP traffic from reaching remote clients. While security is important, indiscriminate blocking can disrupt essential communication and worsen presence delays.

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 secure MRA connectivity, and disabling them would resolve nothing while creating new connectivity issues.

Enabling persistent XMPP connections directly addresses the root cause of delayed presence for remote users, ensures real-time updates, improves communication reliability, and aligns with Cisco best practices for MRA deployments, creating a seamless and efficient collaboration experience.

Question73

In a distributed CUCM SIP conferencing environment, internal participants join conferences without issues, but remote SIP endpoints intermittently fail 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:

Distributed SIP conferencing environments rely on precise Session Description Protocol (SDP) negotiation to establish media paths for all participants. Mid-call re-INVITEs allow new participants to join ongoing conferences. When delayed-offer trunks are used, SDP is sent in the 200 OK response rather than the initial INVITE. Remote endpoints may fail to join if CUCM does not have SDP information available at the start, particularly when NAT, firewall traversal, or multi-site configurations are involved. The lack of immediate SDP prevents proper media path establishment, causing intermittent failures for remote participants.

Option A, enabling early offer SIP on trunks used by remote endpoints, is the correct solution. Early offer includes SDP in the initial INVITE, providing CUCM with immediate media information about the remote endpoint. This allows the system to negotiate media paths proactively, ensuring reliable addition of remote participants to conferences. Cisco best practices recommend early offer for SIP trunks handling conferencing traffic to maintain predictable media behavior, reduce latency, and prevent mid-call failures. Early offer ensures efficient media negotiation, minimizes delays, and eliminates intermittent join failures for remote participants, preserving conference quality and integrity.

Option B, assigning hardware MTPs, can ensure codec compatibility and handle certain SDP-related limitations but does not address the timing of SDP delivery, which is the root cause of mid-call participant failures.

Option C, reducing SIP trunks, simplifies signaling but does not solve SDP negotiation issues. The core problem is the timing of media capability information, not the number of trunks.

Option D, converting remote endpoints to SCCP, is unnecessary because the issue is SDP timing, not protocol incompatibility. Converting endpoints would not resolve mid-call join failures and would introduce unnecessary complexity.

By enabling early offer SIP, CUCM ensures proper media negotiation, reliable addition of remote participants, and consistent conferencing behavior across distributed environments, fully aligning with Cisco best practices.

Question74

A Cisco Unity Connection deployment integrated with CUCM exhibits several seconds of silence at the beginning of voicemail playback. RTP analysis indicates 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 to conserve bandwidth during live calls. However, the beginning of voicemail messages often contains low-energy audio, which VAD interprets as silence, resulting in several seconds of no audio being transmitted. This negatively impacts user experience and can lead to complaints about delayed message 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, eliminating initial silence. Cisco best practices recommend disabling VAD for voicemail deployments to provide uninterrupted playback from the beginning of messages. Users hear voicemails immediately, improving satisfaction and avoiding confusion caused by missing content. The bandwidth impact is minimal because voicemail RTP traffic is small compared to live call traffic.

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

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

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

Disabling VAD directly resolves the issue, ensuring smooth voicemail playback, enhancing user experience, and aligning with Cisco best practices for CUCM and Unity Connection integration.

Question75

Remote users experience intermittent one-way audio when calling internal endpoints through a CUBE. RTP analysis shows packets sometimes sent to incorrect IP 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 multi-site SIP deployments using CUBE often occurs when RTP is transmitted before CUBE has rewritten SDP information. Packets may be sent to incorrect IP addresses or ports, resulting in audio being heard on only one side. This is a common issue in scenarios with early media, NAT, or multi-site SIP trunks where media anchoring is required before RTP transmission. Proper media anchoring ensures that all participants receive audio and that RTP paths are correctly established.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct solution. Delayed-offer SDP ensures that SDP is included in the 200 OK response instead of the initial INVITE, allowing CUBE to anchor media properly, rewrite IP addresses and ports, and establish correct RTP paths before media flows. Cisco best practices recommend delayed-offer SDP for trunks through CUBE when early media could cause misalignment of media paths. This configuration guarantees accurate 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 address SDP timing or media misalignment.

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

Option D, enabling symmetric RTP, assists with NAT traversal but does not fix early media being sent to incorrect addresses. Symmetric RTP does not address SDP timing, which is the root cause of one-way audio issues.

Implementing delayed-offer SDP ensures correct media anchoring through CUBE, eliminates one-way audio, and provides reliable multi-site SIP communication, fully complying with Cisco best practices.

One-way audio in multi-site Cisco Unified Communications deployments is a significant operational challenge, especially when SIP calls traverse a Cisco Unified Border Element (CUBE). This problem typically arises due to premature RTP transmission before CUBE has had an opportunity to anchor the media session and rewrite the Session Description Protocol (SDP) information. SDP is critical in SIP communication because it defines media characteristics such as codec types, IP addresses, and port numbers. When SDP is sent too early in the call setup, such as within the initial INVITE message, RTP streams may be sent to incorrect IP addresses or ports. This misalignment causes one-way audio, where only one participant can hear audio while the other cannot.

Early media scenarios exacerbate this issue. Early media refers to audio signals transmitted before the call is formally answered, such as ringback tones, announcements, or call progress indicators. While early media improves user experience by providing immediate feedback, it creates a timing problem for media path establishment. If RTP begins flowing before CUBE has anchored the call and rewritten the SDP, packets may be delivered to endpoints that are not prepared to receive them. In multi-site deployments, this problem is compounded because calls often traverse WAN links, multiple firewalls, and NAT devices. Proper anchoring of media at CUBE ensures that RTP is directed through the correct path, allowing audio to flow correctly to both sides of the call.

Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct and recommended solution. Delayed-offer SDP changes the timing of SDP delivery so that CUCM does not include SDP in the initial INVITE but instead includes it in the 200 OK response after the signaling has been processed. This approach allows CUBE to rewrite the media information, establish correct RTP paths, and apply any necessary transcoding or media policies before media begins flowing. By delaying the SDP offer, the network ensures that RTP reaches the intended destinations and that audio is fully bi-directional. Cisco best practices advocate delayed-offer SDP for trunks traversing CUBE, particularly in scenarios where early media or multi-site connectivity could result in media path misalignment. This configuration directly addresses the root cause of one-way audio, rather than merely mitigating symptoms.

The benefits of delayed-offer SDP extend beyond resolving one-way audio. In complex, multi-site environments, consistent and predictable media flows are essential for operational efficiency. Without delayed-offer, RTP may be sent to an address that has not yet been processed by CUBE, creating intermittent audio issues that are difficult to troubleshoot. By implementing delayed-offer SDP, administrators can ensure that all RTP streams are correctly anchored, which simplifies troubleshooting and makes packet captures and call detail records more reliable. The media path aligns with the signaling path, providing a clear and deterministic flow for both voice and video traffic.

Interoperability is another significant advantage of delayed-offer SDP. In many enterprise environments, CUCM must interconnect with service providers, cloud-based SIP trunks, or third-party SIP devices. Each of these entities may have specific expectations regarding the timing of SDP. Early SDP offers may conflict with CUBE’s media rewriting logic or with the service provider’s call handling policies, resulting in one-way audio, call setup failures, or codec mismatches. Delayed-offer SDP ensures that SDP is only transmitted after signaling elements have completed processing the call, providing consistent interoperability across different SIP endpoints and network segments.

Option B, using SIP over UDP instead of TCP, does not resolve the one-way audio problem. While TCP provides reliable message delivery and UDP offers a faster, connectionless transport method, the core issue in this scenario is not transport reliability but SDP timing relative to media path anchoring. Changing the transport protocol does not prevent RTP from being sent to incorrect addresses before CUBE processes the call. Therefore, switching to UDP may slightly alter message delivery timing but does not address the underlying cause of one-way audio.

Option C, disabling early media on CUBE, can prevent callers from hearing pre-answer audio such as ringback tones or announcements. However, it does not correct the misdelivery of RTP caused by early SDP. Disabling early media addresses only the symptom of hearing audio too early, not the actual problem of SDP timing and media misalignment. Furthermore, this approach negatively impacts user experience because callers may not receive immediate feedback during call setup. It also does not ensure correct RTP path establishment, meaning one-way audio could still occur after the call is answered.

Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 5 Q61-75

Related posts: