Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 2 Q16-30
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Question16
A company deploys Cisco Expressway for Mobile and Remote Access (MRA). Remote Jabber clients report that presence information for some users does not update in real-time, although internal users’ presence updates correctly. Logs indicate that XMPP traffic between Expressway-C and CUCM is delayed intermittently. Which configuration change MOST effectively resolves the issue?
A) Increase the polling interval for CUCM subscribers
B) Enable persistent XMPP connections between Expressway-C and CUCM
C) Configure firewall to block non-essential ports to reduce traffic
D) Disable traversal zones temporarily to isolate the problem
Answer: B
Explanation:
This scenario involves real-time presence updates in Cisco collaboration environments, specifically with MRA deployments where Jabber clients are external and traverse Expressway components. Presence information is carried over XMPP (Extensible Messaging and Presence Protocol) sessions. For remote users, Expressway-C acts as an intermediary between external Jabber clients and CUCM. The issue describes delayed presence updates, while internal users experience timely updates, indicating a problem in the communication path between Expressway-C and CUCM.
Option B is correct because enabling persistent XMPP connections ensures that sessions remain active and continuously available for sending and receiving presence notifications. Without persistent connections, XMPP connections might be established on-demand, leading to delays in updates or intermittent failures. Persistent connections also reduce the overhead of repeatedly establishing sessions, which is critical for real-time updates in a system supporting hundreds or thousands of remote users. Cisco best practices for MRA deployments specifically recommend persistent XMPP connections to maintain low-latency, real-time presence information for remote endpoints. By enabling persistence, CUCM can immediately propagate presence changes through Expressway-C to external Jabber clients without waiting for session establishment, addressing the observed delays.
Option A suggests increasing polling intervals on CUCM subscribers. Polling affects the frequency of data collection from endpoints or devices within CUCM. Increasing polling intervals would slow detection of status changes, worsening the problem rather than improving it. Since internal presence updates are fine, polling intervals are unlikely to be the root cause.
Option C recommends configuring the firewall to block non-essential ports. While securing ports is essential, indiscriminately blocking traffic may inadvertently block legitimate XMPP traffic, worsening the presence update problem. The issue is protocol timing and session persistence, not firewall misconfiguration.
Option D proposes disabling traversal zones temporarily. Traversal zones are required for Expressway-C and Expressway-E communication with external clients. Disabling them would prevent remote Jabber clients from connecting at all. This does not solve the intermittent XMPP delay and would disrupt service completely, making it an invalid solution.
Question17
A CUCM cluster handles a large number of SIP endpoints across multiple regions. During peak hours, some users report intermittent audio dropouts on calls traversing SIP trunks through CUBE. Packet captures show RTP packets being delivered to the wrong IP address before CUBE rewrites the SDP. Which configuration BEST resolves this issue?
A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Use SIP UDP instead of TCP to accelerate signaling
C) Disable early media on CUBE
D) Configure symmetric RTP on CUBE
Answer: A
Explanation:
This scenario highlights the interaction between CUCM, SIP endpoints, CUBE, and SDP negotiation. The problem described is one-way audio or intermittent audio dropouts caused by RTP packets being sent to incorrect IP addresses before CUBE can modify SDP. This is a common issue in multi-site SIP deployments when early media is sent before the signaling device has processed and rewritten the SDP.
Option A is correct because configuring delayed-offer SDP ensures that CUCM does not include SDP in the initial INVITE. Instead, SDP is offered in the 200 OK response after CUBE has processed the signaling. This timing allows CUBE to correctly anchor the media path, update the media IP and port addresses, and prevent RTP from being sent to an invalid destination. By aligning SDP delivery with media processing, delayed-offer resolves the root cause of one-way audio. Cisco best practices recommend using delayed-offer for SIP trunks traversing CUBE when early media can result in media path mismatches. This approach maintains call integrity across regions and ensures predictable RTP routing.
Option B suggests using UDP instead of TCP to accelerate signaling. While UDP may reduce signaling latency, it does not prevent premature media delivery or address incorrect SDP rewriting. The fundamental issue is timing of SDP negotiation relative to media flow, not transport protocol speed.
Option C recommends disabling early media on CUBE. Early media is often used for IVR prompts, ringback tones, and other call progress indications. Disabling early media can cause service disruptions and does not resolve the underlying problem of SDP timing. It addresses symptoms rather than the root cause.
Option D proposes symmetric RTP on CUBE. Symmetric RTP is useful for NAT traversal by ensuring that media is sent back to the source IP and port from which it was received. However, symmetric RTP does not correct SDP misalignment or prevent early media from being sent to an incorrect address. It is unrelated to the specific issue described.
Thus, Option A is the most effective solution, ensuring proper SDP negotiation timing, preventing one-way audio, and maintaining reliable media delivery across multiple regions.
Question18
A company deploys Cisco Unity Connection integrated with CUCM via SIP. Users report that when they access voicemail, the first few seconds of playback are silent. Network analysis shows RTP packets are flowing normally and no jitter or packet loss is present. Which configuration change MOST effectively resolves this problem?
A) Disable Voice Activity Detection (VAD) on the SIP trunk between CUCM and Unity Connection
B) Adjust Message Waiting Indicator (MWI) extensions
C) Change Unity Connection codec to G.722 to match endpoints
D) Move the voicemail pilot to a different partition
Answer: A
Explanation:
This scenario concerns RTP handling and media flow between CUCM and Unity Connection. The symptom is initial silence during voicemail playback, despite normal RTP flow. Voice Activity Detection (VAD) suppresses RTP packets when the system detects silence or low audio energy, which is common at the start of voicemail recordings.
Option A is correct because disabling VAD on the SIP trunk ensures continuous RTP transmission, including initial silence. Unity Connection relies on continuous media flow for smooth playback. VAD may delay the start of media if it misinterprets low-energy audio as silence, resulting in several seconds of perceived delay. Cisco best practices explicitly recommend disabling VAD for voicemail and automated messaging systems to prevent initial playback gaps. By disabling VAD, the system delivers the audio stream in real-time from the very beginning, resolving the silence issue.
Option B, adjusting MWI extensions, affects signaling for unread message indicators and does not influence RTP playback. Incorrect MWI configuration would affect lamp behavior but would not create silent gaps during playback.
Option C suggests changing the codec to G.722. While codec mismatches can affect audio quality, they do not cause the specific symptom of initial silence when RTP flow is normal. CUCM handles transcoding automatically if necessary.
Option D involves moving the voicemail pilot to another partition, which influences call routing, not media behavior. Playback delays are unrelated to partitions.
Disabling VAD ensures smooth and immediate voicemail playback, eliminating the initial silence while maintaining normal network performance.
Question19
During a CUCM deployment, users report that conference calls fail when adding SIP endpoints from remote sites. Conference attempts succeed internally. Analysis reveals that SIP trunks are configured with delayed offer, and mid-call re-INVITEs are failing for remote participants. Which configuration MOST reliably resolves the issue?
A) Enable early offer SIP on the trunks used by remote SIP endpoints
B) Assign hardware MTPs to all endpoints
C) Reduce the number of SIP trunks to simplify signaling
D) Change remote SIP endpoints to SCCP
Answer: A
Explanation:
This scenario involves conferencing behavior, SIP signaling, and SDP negotiation. Conference failures occur only for remote SIP endpoints, suggesting the issue is related to media path negotiation during mid-call re-INVITEs. Delayed-offer trunks cause SDP to be sent later in the call setup sequence. When adding participants, CUCM may lack proper SDP information to coordinate media paths, especially for remote endpoints traversing firewalls and NAT devices.
Option A is correct because enabling early offer SIP ensures that SDP is included in the initial INVITE. This allows CUCM to immediately understand the media capabilities of all participants and anchor media paths properly for conferences. Early offer reduces signaling ambiguity and prevents mid-call negotiation failures, which is critical for remote endpoints. Cisco best practices recommend early offer for SIP trunks when conferencing or advanced features are used, especially in geographically distributed environments.
Option B, assigning hardware MTPs, ensures transcoding but does not address SDP timing issues that lead to conference failures. While MTPs help with codec compatibility, the failure here is due to missing or delayed SDP during mid-call negotiation, not codec mismatch.
Option C, reducing the number of SIP trunks, simplifies routing but does not resolve SDP negotiation or re-INVITE failures. The issue is protocol timing, not trunk quantity.
Option D, changing remote SIP endpoints to SCCP, is impractical and does not address the underlying SDP and conferencing problem. Protocol change is not required to resolve SDP timing issues.
Enabling early offer SIP provides a reliable and scalable solution to ensure conference call reliability across remote SIP endpoints, resolving mid-call failures.
Question20
A CUCM administrator observes intermittent one-way audio for SIP calls routed through a CUBE to a remote site. The WAN link is stable, with no packet loss. Investigation shows that RTP packets are occasionally sent to the wrong IP address before CUBE rewrites SDP. Which configuration MOST effectively resolves the problem?
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 highlights one-way audio issues in multi-site SIP deployments traversing CUBE. One-way audio often occurs when RTP packets are sent to incorrect addresses due to early SDP offer before CUBE has processed and anchored media correctly.
Option A is correct because delayed-offer SDP ensures that SDP is included in the 200 OK response after CUBE has updated the media address. This prevents RTP from being sent to an incorrect destination, eliminating one-way audio. Cisco best practices recommend delayed-offer SIP for trunks traversing CUBE when early media could cause media path mismatches. Delayed-offer aligns signaling and media timing, ensuring predictable audio flows.
Option B, switching to UDP, does not correct SDP timing and cannot prevent early media from reaching incorrect IP addresses. Transport protocol change alone does not address the fundamental issue.
Option C, disabling early media, prevents initial audio delivery such as IVR prompts or ringback tones, which can disrupt service. It does not correct SDP path misalignment.
Option D, enabling symmetric RTP, assists in NAT traversal but does not correct incorrect SDP information or early media sending to invalid addresses. Symmetric RTP is unrelated to SDP timing issues.
Delayed-offer SDP ensures proper media anchoring through CUBE, resolving intermittent one-way audio while maintaining correct call flow across the WAN.
Question21
A company deploys a CUCM cluster with SIP endpoints across multiple regions. Users report that outbound calls to the PSTN fail intermittently with fast busy signals. Analysis shows that CUCM sequentially attempts multiple route groups, causing delays before finding an available gateway. Which configuration MOST effectively resolves this issue while maintaining high availability?
A) Remove all but one gateway from each route pattern
B) Implement Local Route Group (LRG) to dynamically select gateways based on device pool
C) Use time-of-day routing to direct calls to specific gateways
D) Configure weighted priorities within route lists to force gateway selection
Answer: B
Explanation:
This scenario focuses on optimizing outbound call routing in CUCM for multi-gateway deployments. The problem arises because CUCM attempts multiple route groups sequentially, introducing call setup delays that sometimes result in fast busy signals. High availability requires maintaining multiple gateways while reducing sequential processing delays. Option B, implementing Local Route Group, is the optimal solution. LRG allows CUCM to dynamically select the local gateway for a device based on the device pool or location. This ensures that the closest or most appropriate gateway is selected immediately without sequentially trying all configured route groups. As a result, call setup time is minimized, and high availability is maintained because alternative gateways remain available for failover.
Option A, removing all but one gateway, superficially reduces sequential searching but sacrifices redundancy. Without multiple gateways, there is no failover if the single gateway fails, violating high-availability principles.
Option C, using time-of-day routing, does not address sequential processing delays. Time-of-day routing is designed to route calls differently based on schedules, not to optimize gateway selection in real time.
Option D, configuring weighted priorities in route lists, may influence selection but does not eliminate sequential searching across multiple route groups. Sequential failover still occurs if the preferred gateway is unavailable, so delays may persist.
Thus, Option B directly resolves the problem by streamlining gateway selection, reducing call setup delays, and preserving high availability in line with Cisco best practices.
Question22
During a Mobile and Remote Access (MRA) deployment, remote Jabber clients report that presence updates for certain users do not occur in real time. Internal presence updates work as expected. Logs indicate intermittent XMPP delays between Expressway-C and CUCM. Which configuration MOST reliably resolves this issue?
A) Increase CUCM polling intervals for subscriber devices
B) Enable persistent XMPP connections between Expressway-C and CUCM
C) Restrict firewall rules to block non-essential ports to reduce traffic
D) Temporarily disable traversal zones to isolate the problem
Answer: B
Explanation:
This scenario addresses the critical aspect of real-time presence delivery in Cisco collaboration environments with MRA. Remote Jabber clients rely on XMPP traffic routed through Expressway-C to communicate with CUCM. Delayed presence updates indicate that XMPP sessions are not continuously maintained, causing intermittent notifications. Option B is correct because enabling persistent XMPP connections ensures that sessions remain open, allowing immediate propagation of presence changes. This reduces latency and prevents updates from being delayed while connections are re-established on demand. Cisco best practices for MRA deployments emphasize persistent XMPP connections to maintain timely presence updates for external users.
Option A suggests increasing polling intervals on CUCM. Polling controls data refresh rates but does not influence XMPP session persistence. Increasing polling would exacerbate delays rather than resolve them.
Option C recommends restricting firewall rules. While securing ports is important, indiscriminately blocking traffic could interrupt legitimate XMPP traffic and worsen the problem.
Option D proposes disabling traversal zones temporarily. Traversal zones are required for external Jabber clients to connect through Expressway-C/E. Disabling them would prevent remote clients from accessing CUCM entirely, making this option counterproductive.
Enabling persistent XMPP connections ensures that real-time presence updates propagate efficiently from CUCM through Expressway-C to remote Jabber clients, addressing the latency issue effectively.
Question23
Users report that conference calls fail intermittently when adding SIP endpoints from remote sites, while internal conferencing works without issue. Investigation reveals that SIP trunks are configured for delayed offer, and mid-call re-INVITEs for remote participants are failing. 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 focuses on SIP conferencing behavior and the impact of SDP negotiation on remote endpoints. Conference call failures occur during mid-call re-INVITEs when adding remote participants. Delayed-offer SIP trunks send SDP in the response rather than the initial INVITE, which can prevent CUCM from properly coordinating media paths for remote participants. Option A is correct because enabling early offer SIP ensures that SDP is included in the initial INVITE. This allows CUCM to understand endpoint media capabilities upfront, facilitating successful conference setup. Cisco best practices recommend early offer for SIP trunks used in distributed or remote conferencing scenarios, ensuring predictable call behavior and media negotiation.
Option B, assigning hardware MTPs, is primarily for codec transcoding. While MTPs ensure compatibility, the failures in this scenario are due to SDP negotiation and mid-call re-INVITE issues, not codec mismatch.
Option C, reducing the number of SIP trunks, may simplify routing but does not address the core issue of SDP timing.
Option D, converting endpoints to SCCP, is impractical and unnecessary, as the issue is protocol configuration and SDP handling, not the endpoint protocol itself.
Enabling early offer ensures consistent SDP negotiation, reliable media path establishment, and successful conference call additions for remote SIP endpoints.
Question24
A Cisco Unity Connection deployment experiences delayed voicemail playback, with several seconds of silence at the start of messages. RTP flow is normal, with no packet loss or jitter. Which configuration MOST effectively addresses this problem?
A) Disable Voice Activity Detection (VAD) on the SIP trunk between CUCM and Unity Connection
B) Adjust Message Waiting Indicator (MWI) extensions
C) Change Unity Connection codec to G.722
D) Move the voicemail pilot to a different partition
Answer: A
Explanation:
The scenario describes initial silence during voicemail playback despite normal RTP flow. This symptom is typically caused by Voice Activity Detection (VAD), which suppresses RTP packets when low-energy audio is detected, interpreting it as silence. Option A is correct because disabling VAD ensures continuous RTP delivery, including the initial portion of voicemail recordings. Cisco best practices for voicemail deployments emphasize disabling VAD on SIP trunks to prevent playback gaps. With VAD disabled, Unity Connection can transmit audio immediately, providing smooth and uninterrupted voicemail playback.
Option B, adjusting MWI extensions, affects only the lamp indicator on phones for unread messages and has no impact on audio playback.
Option C, changing the codec, does not address the root cause. Codec mismatches would cause audio quality issues, not silent gaps at the start of messages.
Option D, moving the voicemail pilot to a different partition, affects call routing rather than media playback. It does not address RTP flow or VAD behavior.
Disabling VAD ensures that voicemail playback starts immediately without silent gaps, resolving user complaints effectively.
Question25
Remote users report intermittent one-way audio when calling internal endpoints through a CUBE. RTP analysis shows that packets are occasionally sent to incorrect IP addresses before 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 involves one-way audio issues in multi-site SIP deployments with CUBE. The problem arises because RTP packets are sent to incorrect IP addresses before CUBE processes and rewrites the SDP. Option A is correct because configuring delayed-offer SDP ensures that SDP is included in the 200 OK response rather than the initial INVITE. This allows CUBE to anchor media correctly and rewrite the media addresses before RTP flow begins, preventing one-way audio. Delayed-offer SDP aligns signaling and media timing, ensuring predictable audio delivery. Cisco best practices recommend delayed-offer SIP trunks in multi-site deployments where early media may cause media path misalignment.
Option B, using UDP instead of TCP, does not solve SDP timing issues. Transport protocol choice does not influence when SDP is offered or how media is anchored.
Option C, disabling early media, may prevent initial audio but does not correct SDP misalignment or prevent RTP from being sent to an incorrect address. It addresses symptoms rather than the root cause.
Option D, enabling symmetric RTP, assists with NAT traversal but does not correct SDP timing or early media delivery to the wrong destination. Symmetric RTP is unrelated to this specific problem.
Delayed-offer SDP ensures proper media path establishment through CUBE, eliminating one-way audio while maintaining correct call flow across WAN links and multiple sites.
Question26
A CUCM administrator receives reports that calls placed to certain external PSTN numbers intermittently fail. Analysis shows that CUCM sequentially attempts multiple gateways, resulting in delayed call setup and occasional call failures. Which solution BEST resolves this issue while maintaining high availability and minimizing latency?
A) Remove all but one gateway from the route pattern
B) Implement Local Route Groups (LRG) to allow dynamic gateway selection 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:
The scenario highlights the challenges associated with managing outbound call routing in a multi-gateway CUCM deployment. Intermittent call failures when dialing external PSTN numbers often stem from inefficient route selection mechanisms. In CUCM, route patterns are used to define how calls are sent to available gateways. If multiple gateways are assigned to a route pattern without optimization, CUCM will attempt each gateway sequentially until one succeeds. This sequential search introduces latency, especially during peak traffic periods, and may cause intermittent fast busy signals when call attempts time out.
Option B is the correct solution because Local Route Groups (LRG) allow CUCM to dynamically select the appropriate gateway for a device based on its associated device pool. By associating devices with a local route group, CUCM ensures that the closest or most appropriate gateway is immediately selected, reducing sequential searching and call setup latency. LRG preserves high availability because alternate gateways remain available if the primary fails, ensuring that calls are not blocked if a gateway is down. This aligns with Cisco best practices for multi-gateway routing, where efficiency and reliability must be balanced.
Option A, removing all but one gateway, might reduce sequential searches superficially but compromises redundancy. Without multiple gateways, the system cannot fail over in the event of a gateway failure, violating high-availability requirements. While latency might improve slightly, this approach introduces significant operational risk and is therefore not advisable.
Option C, implementing time-of-day routing, is intended to redirect calls based on schedules, such as directing calls to specific gateways during off-peak hours. While useful in some scenarios, this does not address the root problem of sequential searching or call setup latency. Time-of-day routing modifies call paths based on time but does not optimize gateway selection dynamically for device pools.
Option D, configuring weighted priorities within route lists, allows CUCM to influence gateway selection by preference. However, sequential failover still occurs if the preferred gateway is unavailable, which does not fully eliminate delays. This approach only partially addresses the issue and does not leverage the dynamic selection capabilities provided by LRG.
Implementing Local Route Groups is the most effective solution because it streamlines call routing, maintains redundancy, and ensures that calls are routed through the most appropriate gateway immediately. By aligning device pools with local gateways, LRG enhances both performance and reliability, eliminating intermittent failures caused by sequential gateway searches.
Question27
Remote Jabber clients registered through Mobile and Remote Access (MRA) report that presence updates for certain internal users are delayed or missing, while internal users’ presence updates work correctly. Logs indicate 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 the firewall to block non-essential ports to reduce traffic
D) Temporarily disable traversal zones to isolate the issue
Answer: B
Explanation:
This scenario concerns the propagation of real-time presence information for remote users using Jabber through Mobile and Remote Access (MRA). The presence service relies on the Extensible Messaging and Presence Protocol (XMPP) to communicate user availability and status updates between clients and CUCM. In MRA deployments, Expressway-C acts as an intermediary between remote Jabber clients and CUCM. Delays in XMPP traffic can result in stale or missing presence information, particularly for remote users who are dependent on Expressway-C to relay these messages.
Option B is correct because enabling persistent XMPP connections ensures that sessions between Expressway-C and CUCM remain active and continuously available. Persistent connections eliminate the need to re-establish XMPP sessions for each update, reducing latency and ensuring that presence changes are immediately propagated. Cisco best practices for MRA deployments specifically recommend persistent XMPP connections to maintain real-time updates for remote endpoints. Persistent connections provide a reliable and low-latency communication path, allowing CUCM to instantly relay presence changes through Expressway-C to external Jabber clients.
Option A, increasing polling intervals for CUCM subscribers, would actually exacerbate the problem. Polling controls how frequently CUCM collects data from endpoints or devices. Increasing the interval slows the detection of status changes and would lead to further delays in presence updates rather than improving them.
Option C, configuring the firewall to block non-essential ports, may improve security, but indiscriminately blocking ports can disrupt legitimate XMPP traffic. The problem described is latency in real-time updates, not network congestion caused by extraneous traffic. Blocking ports would not solve the underlying issue and could make it worse.
Option D, temporarily disabling traversal zones, would prevent remote clients from connecting through Expressway-C/E entirely. This action is disruptive and does not address intermittent XMPP delays. It is not a viable solution for maintaining continuous presence updates.
Enabling persistent XMPP connections ensures that presence messages flow continuously and reliably from CUCM to Expressway-C and ultimately to remote Jabber clients. It addresses the root cause of delayed or missing updates, reduces latency, and aligns with Cisco best practices for high-performance MRA deployments.
Question28
During SIP-based conferencing in CUCM, internal participants can successfully join conferences, but remote SIP endpoints fail intermittently when being added to an ongoing conference. Analysis reveals that 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 the behavior of SIP conferencing and the critical role of Session Description Protocol (SDP) negotiation in media path establishment. Conference call failures specifically affecting remote SIP endpoints indicate an issue with how SDP is exchanged during mid-call re-INVITEs. Delayed offer trunks send SDP in the 200 OK response rather than in the initial INVITE. During mid-call operations such as adding new participants, CUCM may lack sufficient SDP information to coordinate media paths, particularly for remote endpoints traversing NAT or firewalls.
Option A is correct because enabling early offer SIP ensures that SDP is included in the initial INVITE. By providing SDP information upfront, CUCM can immediately understand the media capabilities of all participants, including remote endpoints, and establish the correct media path. Early offer is critical for reliable conferencing in distributed environments where participants may be behind NAT or remote. Cisco best practices recommend early offer SIP for trunks that handle conferencing and advanced call features to ensure predictable call behavior and successful addition of participants during active conferences.
Option B, assigning hardware MTPs, primarily addresses codec transcoding requirements. While MTPs are necessary for media compatibility between endpoints using different codecs, they do not resolve failures caused by delayed SDP negotiation during mid-call conference additions. The root cause is SDP timing, not codec mismatch.
Option C, reducing the number of SIP trunks, might simplify signaling but does not solve SDP negotiation issues. The number of trunks is unrelated to mid-call SDP failures.
Option D, converting remote endpoints to SCCP, is unnecessary and impractical. The issue arises from SDP negotiation and early offer timing, not endpoint protocol compatibility. Changing the protocol does not address the root cause.
Enabling early offer SIP ensures that SDP negotiation occurs at the correct time, allowing CUCM to successfully add remote SIP participants to conferences. This approach resolves mid-call failures, maintains reliable conferencing functionality, and follows Cisco best practices for distributed SIP deployments.
Question29
A Cisco Unity Connection deployment integrated with CUCM exhibits initial silence of several seconds at the beginning of voicemail playback. RTP flows normally, with no packet loss or jitter observed. 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 concerns media handling in voicemail playback. The symptom—initial silence at the start of messages—is commonly caused by Voice Activity Detection (VAD). VAD suppresses RTP packet transmission when low-energy audio is detected, which is frequently present at the beginning of recorded voicemail messages. While VAD is useful in normal telephony to reduce bandwidth usage during silence, it is detrimental for voicemail, where initial audio content is important.
Option A is correct because disabling VAD ensures that RTP is transmitted continuously, including the initial portion of voicemail recordings. Cisco best practices for voicemail integrations recommend disabling VAD on SIP trunks between CUCM and Unity Connection to prevent playback gaps. Disabling VAD ensures that users experience seamless message playback from the very start, eliminating several-second silence artifacts.
Option B, adjusting MWI extensions, affects signaling for message lamp indications and does not influence RTP flow or playback behavior.
Option C, changing the codec, affects audio quality but does not correct the issue caused by VAD suppression. Codec mismatches would degrade sound, not introduce initial silence.
Option D, moving the voicemail pilot to a different partition, changes call routing but has no effect on media delivery or RTP handling.
Disabling VAD guarantees that voicemail playback starts immediately and consistently, resolving user complaints and adhering to Cisco best practices for media handling in voicemail systems.
Question30
Remote users experience 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 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 involves one-way audio issues in multi-site SIP deployments where CUBE is responsible for anchoring media. The root cause is the timing of SDP negotiation relative to media flow. Early media sent before CUBE can rewrite SDP results in RTP being delivered to incorrect IP addresses, causing one-way audio.
Option A is correct because delayed-offer SDP ensures that SDP is included in the 200 OK response rather than the initial INVITE. This allows CUBE to process the signaling, anchor media correctly, and update media addresses before RTP flows. Cisco best practices recommend delayed-offer SIP for trunks traversing CUBE when early media can result in incorrect media delivery. By delaying SDP offer, CUCM ensures that the media path is fully established, RTP reaches the correct destination, and audio quality is maintained.
Option B, using SIP over UDP instead of TCP, does not address SDP timing issues. While UDP is faster, the problem lies in early SDP delivery, not transport protocol speed.
Option C, disabling early media, might prevent certain call progress tones but does not resolve RTP path misalignment caused by premature SDP. Disabling early media addresses symptoms, not the underlying cause.
Option D, enabling symmetric RTP, helps NAT traversal but does not correct SDP timing or prevent RTP from being sent to incorrect addresses. Symmetric RTP is unrelated to the root cause described.
Configuring delayed-offer SDP ensures correct media anchoring through CUBE, eliminates one-way audio, and maintains proper call flow across distributed sites, adhering to Cisco deployment best practices.
In complex multi-site Cisco Unified Communications deployments, one-way audio is a frequent and challenging problem, particularly when calls traverse a Cisco Unified Border Element (CUBE). The issue is typically rooted in the interaction between SIP signaling and RTP media delivery, especially in scenarios involving early media. Early media refers to audio streams such as ringback tones, announcements, or progress tones that are delivered before the call is formally answered, i.e., before a 200 OK response is sent. While early media can provide a seamless user experience in terms of audio feedback, it introduces a timing problem in SIP-based networks where the media path is anchored by a device like CUBE.
The primary cause of one-way audio in this scenario is that the SDP (Session Description Protocol) is being exchanged prematurely. SIP uses SDP to communicate the details of the media session, including codec types, IP addresses, and port numbers for RTP streams. If the SDP is included in the initial INVITE, CUBE may receive the media before it has had a chance to rewrite or anchor the media addresses. As a result, the RTP packets are sent to an incorrect destination, typically the original CUCM or the IP address listed in the early media SDP. Consequently, one leg of the audio path is disrupted, resulting in one-way audio.
Option A, enabling delayed-offer SDP on the CUCM SIP trunk toward CUBE, is the correct approach to resolve this issue. Delayed-offer SDP is a SIP trunk configuration where the initial INVITE does not include SDP. Instead, the SDP is included in the 200 OK response after the call has been processed by CUCM and CUBE. This delay allows CUBE to perform its media anchoring function before the endpoints negotiate media parameters. By deferring the SDP offer, CUBE can insert itself into the media path properly, rewriting the addresses and ports so that RTP flows through CUBE. This ensures that both audio streams can reach their intended destinations and eliminates one-way audio. Cisco best practices consistently recommend the use of delayed-offer SDP for SIP trunks traversing CUBE when early media is involved, because it provides deterministic media path establishment and maintains call quality and reliability.
Option B, changing the transport protocol from TCP to UDP, does not address the core problem. While TCP offers reliable transmission of SIP messages and UDP is typically faster due to its connectionless nature, the issue in this case is not related to transport reliability but to the timing of SDP negotiation relative to media flow. Changing the protocol may slightly alter message delivery speed but does not prevent RTP packets from being sent to incorrect IP addresses. Therefore, switching to UDP cannot resolve one-way audio issues caused by early SDP delivery.
Option C, disabling early media on CUBE, might superficially seem like a solution because it prevents the transmission of audio before the call is answered. However, disabling early media only suppresses ringback tones or other pre-answer audio and does not correct the fundamental issue: premature SDP delivery. Even with early media disabled, the call could still fail to establish a correct RTP path if the SDP is sent in the initial INVITE. This approach addresses the symptoms rather than the root cause, and it also reduces call experience quality because users may not hear expected call progress tones.
Option D, enabling symmetric RTP on CUBE, is a configuration typically used to handle NAT traversal issues. Symmetric RTP ensures that RTP packets are sent back to the source IP and port from which they were received, which can be helpful when endpoints are behind NAT or firewalls. While this is a useful feature in certain deployment scenarios, it does not resolve SDP timing issues. The one-way audio in this scenario is caused by SDP being delivered too early, not by asymmetric media paths or NAT. Therefore, symmetric RTP is unrelated to the problem at hand and would not prevent RTP from being sent to incorrect addresses when early SDP is involved.
The deployment of delayed-offer SDP has several advantages beyond resolving one-way audio. By ensuring that SDP is sent in the 200 OK response, CUCM and CUBE have complete visibility and control over the media path. This allows for proper media anchoring, application of any transcoding or media manipulation rules, and full compliance with enterprise voice policies. Calls become more predictable and easier to troubleshoot because the media path is fully determined before RTP streams begin. Moreover, delayed-offer SDP is consistent with Cisco’s recommended design for multi-site SIP deployments, ensuring that media flows correctly through all necessary network elements and gateways.