Cisco 350-801 Implementing Cisco Collaboration Core Technologies (CLCOR) Exam Dumps and Practice Test Questions Set 11 Q151-165
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Question151
A network engineer is designing a multi-site SIP trunk deployment with CUCM clusters connected via CUBE. Users report intermittent call failures and one-way audio when calling between sites. RTP packet traces indicate media being sent to incorrect IP addresses before SDP rewriting occurs. Which configuration MOST effectively resolves this issue?
A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Force SIP over UDP exclusively
C) Disable early media on CUBE
D) Enable symmetric RTP on CUBE
Answer: A
Explanation:
In multi-site SIP trunk deployments where CUCM communicates through a CUBE, the presence of one-way audio and intermittent call failures is commonly caused by RTP being transmitted before the CUBE has completed SDP rewriting. SDP, which is part of SIP signaling, specifies media attributes, including IP addresses, ports, and codec parameters. If RTP packets are sent prior to proper SDP handling, the packets may be misrouted, leading to one-way audio or dropped calls.
Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly mitigates this problem by delaying the inclusion of SDP in the INVITE message until the 200 OK response. This ensures that the CUBE can anchor the media path and modify SDP information correctly, resulting in properly aligned RTP flows. Cisco best practices recommend delayed-offer SDP in multi-site SIP deployments because it synchronizes signaling and media paths, stabilizes call quality, and reduces operational troubleshooting complexity. By implementing delayed-offer SDP, administrators prevent misrouted RTP packets and call drops, maintain interoperability across SIP endpoints and clusters, and ensure predictable media flow. Delayed-offer SDP also aligns with enterprise requirements for high-quality inter-site voice communication, supporting scalable deployments and reducing support incidents.
Option B, forcing SIP over UDP exclusively, provides fast signaling but does not prevent early RTP transmission. UDP transport reduces latency but does not control when RTP is transmitted relative to SDP rewriting. Therefore, one-way audio and dropped calls would still occur because the core problem is media misalignment, not transport method.
Option C, disabling early media on CUBE, affects call progress tones and ringback behavior but does not solve RTP misrouting. Early media configuration influences how audio is played during call setup but does not correct the underlying SDP timing issue that leads to one-way audio.
Option D, enabling symmetric RTP on CUBE, ensures that RTP packets are received and transmitted from the same IP and port pair, which is useful for NAT traversal. However, symmetric RTP alone cannot prevent early RTP transmission prior to SDP rewriting. While it complements proper SDP handling, it does not address the root cause of one-way audio or call failures.
Implementing delayed-offer SDP ensures that RTP flows only after proper media anchoring, leading to predictable and high-quality inter-site communications. This solution aligns with Cisco recommendations for SIP trunking in multi-site environments, stabilizes call performance, reduces support tickets, and ensures that enterprise users experience consistent audio quality. Proper SDP handling simplifies troubleshooting, improves reliability, and guarantees that media paths are correctly established, even in complex multi-cluster or NATed environments. Delayed-offer SDP also improves interoperability with various SIP endpoints and third-party SIP providers, making it the most effective configuration to address one-way audio and call drop issues in inter-site SIP trunk deployments.
Question152
Remote Jabber clients behind NAT report intermittent registration failures and occasional one-way audio when connecting to CUCM via Expressway traversal zones. ICE candidate exchanges fail intermittently, and traversal zones are frequently near maximum capacity. Which configuration MOST effectively mitigates this issue?
A) Increase traversal zone resources and enable full bidirectional ICE support
B) Require VPN connectivity for remote clients
C) Reduce CUCM SIP registration timers
D) Enable persistent XMPP connections between Expressway-C and CUCM
Answer: A
Explanation:
Remote Jabber clients utilize Expressway traversal zones to register securely and establish media paths with CUCM. Intermittent registration failures and one-way audio often occur when traversal zone resources are insufficient and ICE candidate negotiation fails. ICE enables endpoints to determine the optimal IP addresses and ports for RTP streams, facilitating NAT traversal and firewall traversal. When traversal zones are overloaded, ICE candidate exchanges can fail, leading to registration delays and misrouted media, causing one-way audio and dropped calls.
Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses the root cause. Allocating additional CPU and memory ensures that traversal zones can handle peak concurrent connections, preventing registration failures. Full bidirectional ICE support guarantees successful candidate exchange between remote clients and Expressway servers, allowing RTP streams to traverse NAT and firewall environments effectively. Cisco best practices recommend scaling traversal zones based on the number of expected remote endpoints and enabling full ICE support to maintain reliable registration and media path establishment. Monitoring traversal zone performance continuously ensures administrators can proactively allocate resources, maintaining seamless connectivity and high-quality voice services for remote users.
Option B, requiring VPN connectivity, bypasses some NAT-related challenges but introduces latency, complexity, and potential points of failure. VPNs do not address traversal zone resource limitations or ICE negotiation failures and are therefore not considered optimal for large-scale deployments.
Option C, reducing CUCM SIP registration timers, increases retry frequency but does not resolve resource constraints. Frequent retries under high load can exacerbate performance issues rather than improve registration success.
Option D, enabling persistent XMPP connections, improves messaging reliability and presence updates but does not affect ICE negotiation or traversal zone performance. Persistent XMPP connections do not influence registration or RTP path establishment, leaving one-way audio and intermittent registration issues unresolved.
Implementing Option A ensures remote Jabber clients can register successfully and establish correct RTP paths, providing high-quality voice service. This approach supports scalable remote collaboration, reduces call drops, and aligns with Cisco best practices for Mobile and Remote Access deployments. Proper traversal zone sizing combined with bidirectional ICE guarantees predictable performance, minimizes operational support incidents, and ensures seamless communication even during periods of high load, delivering reliable enterprise voice services to all remote users.
Question153
Unity Connection voicemail users report initial silence of several seconds when playing messages. RTP analysis indicates no packet loss, jitter, or network delay. Which configuration MOST effectively resolves this issue?
A) Disable Voice Activity Detection (VAD) on the SIP trunk between CUCM and Unity Connection
B) Adjust Message Waiting Indicator (MWI) extensions
C) Change Unity Connection codec to G.722
D) Move the voicemail pilot to a different partition
Answer: A
Explanation:
Initial silence during voicemail playback is commonly caused by Voice Activity Detection (VAD), which suppresses RTP transmission during periods of low audio energy to conserve bandwidth during live calls. Voicemail messages often begin with low-energy segments, such as soft sounds or silence, which VAD interprets as inactivity. This delay in RTP transmission creates the perception of initial silence at the start of messages.
Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, allows RTP to flow continuously, including low-energy audio, eliminating the initial silence. Cisco best practices recommend disabling VAD for voicemail traffic to maintain message integrity and deliver a consistent user experience. The slight increase in bandwidth usage is negligible compared to the improvement in message playback reliability. By removing the VAD-related delay, users experience immediate playback of messages, reducing support calls and improving satisfaction. This configuration ensures predictable behavior across endpoints and provides uninterrupted access to voicemail messages.
Option B, adjusting MWI extensions, affects visual notifications for new messages but does not influence RTP or playback timing. MWI changes impact lamp signaling and notifications, leaving the initial silence issue unresolved.
Option C, changing the codec to G.722, improves audio fidelity but does not address VAD-induced delays. Codec selection affects sound quality but cannot correct the suppression of RTP during low-energy audio.
Option D, moving the voicemail pilot to a different partition, alters call routing but does not impact RTP or VAD behavior. Partition changes have no effect on message playback timing.
Disabling VAD ensures immediate, continuous playback of voicemail messages, improving user experience and preserving message integrity. This aligns with Cisco best practices, reduces support incidents, and guarantees predictable behavior for enterprise voicemail users. Proper configuration of VAD provides reliable service without compromising overall call quality or operational efficiency, ensuring consistent and high-quality voicemail access.
Question154
During inter-site CUCM calls over SIP trunks through CUBE, users report one-way audio and occasional call drops. RTP packets are sometimes sent to incorrect IP addresses prior to SDP rewriting. Which configuration MOST effectively resolves this issue?
A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Convert SIP trunks to SCCP
C) Use SIP over TCP exclusively
D) Disable early media on CUBE
Answer: A
Explanation:
In inter-site SIP trunk deployments using CUBE, one-way audio and call drops often occur when RTP packets are transmitted before CUBE has anchored the media path and rewritten SDP. SDP defines media parameters, including IP addresses, ports, and codecs. Premature RTP transmission results in misrouted media, causing audio issues and dropped calls.
Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, addresses this problem. Delayed-offer SDP postpones SDP transmission from the INVITE to the 200 OK response, giving CUBE the opportunity to anchor media and modify SDP correctly. Cisco best practices recommend this approach in multi-site deployments to synchronize signaling and media paths, improve call stability, and ensure interoperability. Delayed-offer SDP prevents misdirected RTP flows, reduces troubleshooting complexity, and provides consistent audio quality for end users. It also supports scalability and reliability in complex, multi-cluster enterprise networks, maintaining predictable behavior for inter-site SIP communications.
Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not prevent early RTP transmission. While SCCP may improve endpoint interoperability, it does not address the underlying media path misalignment caused by premature RTP.
Option C, using SIP over TCP, guarantees reliable delivery of signaling messages but does not influence the timing of RTP transmission. TCP ensures signaling reliability but does not prevent one-way audio caused by early RTP.
Option D, disabling early media on CUBE, affects call progress tones but does not correct the misaligned RTP flows. Early media configuration only influences signaling behavior, not media path establishment.
By enabling delayed-offer SDP, administrators ensure correct media anchoring and predictable RTP flows, leading to consistent call quality and reduced support incidents. This configuration aligns with Cisco best practices for SIP trunking in multi-site environments and stabilizes inter-site communication. Proper SDP handling ensures interoperability, reduces troubleshooting overhead, and provides a reliable user experience across complex enterprise networks.
Question155
Remote Jabber users report intermittent registration failures and one-way audio when connecting to CUCM via Expressway traversal zones. ICE candidate exchanges frequently fail, and traversal zones are near capacity. Which configuration MOST effectively resolves this issue?
A) Increase traversal zone capacity and enable full bidirectional ICE support
B) Require VPN connectivity for remote clients
C) Reduce CUCM SIP registration timers
D) Enable persistent XMPP connections between Expressway-C and CUCM
Answer: A
Explanation:
Remote Jabber client issues, including registration failures and one-way audio, are commonly caused by insufficient traversal zone resources and failed ICE candidate negotiation. Traversal zones facilitate secure registration, signaling, and media path establishment between remote clients and CUCM. ICE allows endpoints to select optimal IP addresses and ports for RTP streams, ensuring successful NAT and firewall traversal. High traversal zone load can lead to failed candidate exchanges, causing registration delays and misrouted media.
Option A, increasing traversal zone capacity and enabling full bidirectional ICE support, directly mitigates the problem. Allocating additional CPU and memory allows traversal zones to handle peak concurrent connections without failure. Full bidirectional ICE support ensures successful candidate exchange between remote clients and the Expressway server, allowing RTP streams to traverse NAT and firewall environments effectively. Cisco best practices recommend proper sizing of traversal zones based on expected remote endpoints and enabling full ICE support to maintain reliable registration and media paths. Continuous monitoring and proactive scaling prevent overload during peak usage, ensuring seamless registration and media delivery.
Option B, requiring VPN connectivity, introduces latency, complexity, and potential failure points without addressing traversal zone resource limitations or ICE negotiation failures.
Option C, reducing CUCM SIP registration timers, increases retry frequency but does not solve resource constraints, and can worsen congestion.
Option D, enabling persistent XMPP connections, improves messaging reliability but does not impact ICE negotiation or traversal zone performance. Persistent XMPP does not resolve media path issues or registration failures.
Implementing Option A ensures successful registration, proper RTP path establishment, and high-quality voice for remote Jabber users. This solution supports scalable remote collaboration, reduces call drops, and aligns with Cisco best practices. Correct traversal zone sizing and full ICE support provide seamless connectivity even during high load periods, delivering predictable, high-quality enterprise voice services to remote users.
Question156
During a CUCM multi-site deployment, users report intermittent call failures and one-way audio on SIP trunks routed through CUBE. Packet captures indicate RTP packets being sent before CUBE has rewritten SDP. Which configuration MOST effectively resolves this issue?
A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Convert SIP trunks to SCCP
C) Use SIP over TCP exclusively
D) Disable early media on CUBE
Answer: A
Explanation:
Intermittent call failures and one-way audio in multi-site CUCM deployments using SIP trunks through CUBE are frequently the result of RTP packets being transmitted prior to SDP rewriting. SDP, or Session Description Protocol, carries essential information about media, including IP addresses, ports, and codec parameters. If RTP is sent before the CUBE has the chance to anchor the media path and rewrite SDP, packets may be misdirected, resulting in one-way audio or dropped calls.
Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly addresses this issue by postponing the inclusion of SDP in the INVITE until the 200 OK message. This delay ensures that CUBE can properly anchor media and modify SDP information so that all RTP packets are directed to the correct endpoints. Cisco recommends delayed-offer SDP for multi-site deployments because it synchronizes signaling and media, stabilizes call quality, and reduces troubleshooting complexity. It ensures interoperability across different SIP endpoints and clusters, maintains consistent voice quality, and aligns with enterprise requirements for reliable inter-site communication. Delayed-offer SDP also simplifies operational support, reduces call drops, and ensures predictable media flows across complex network topologies.
Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not resolve early RTP transmission. SCCP may improve endpoint interoperability in some cases, but it does not prevent RTP from being sent before SDP rewriting, leaving the root cause unresolved.
Option C, using SIP over TCP exclusively, ensures reliable signaling delivery but does not address the timing of RTP relative to SDP rewriting. TCP guarantees that SIP messages are delivered without loss, but RTP misdirection still occurs if SDP has not been properly anchored.
Option D, disabling early media on CUBE, affects call progress tones and ringback behavior but does not prevent misrouted RTP packets. Early media configuration influences how audio is handled during call setup, not the proper anchoring of RTP streams.
Implementing delayed-offer SDP ensures that media paths are correctly established before RTP transmission begins. This configuration improves call stability, reduces one-way audio, and aligns with Cisco best practices for multi-site SIP trunk deployments. By enabling delayed-offer SDP, enterprises can achieve predictable call behavior, maintain high-quality voice communication, and support complex inter-site deployments without operational disruptions. Proper SDP handling is a critical component of scalable, reliable SIP trunking, providing consistent media flows across clusters and ensuring that enterprise users experience seamless voice communication.
Question157
Remote Jabber clients behind NAT report intermittent registration failures and one-way audio when connecting to CUCM through Expressway traversal zones. ICE candidate exchanges frequently fail, and traversal zones are near maximum capacity. Which configuration MOST effectively mitigates this issue?
A) Increase traversal zone resources and enable full bidirectional ICE support
B) Require VPN connectivity for remote clients
C) Reduce CUCM SIP registration timers
D) Enable persistent XMPP connections between Expressway-C and CUCM
Answer: A
Explanation:
Intermittent registration failures and one-way audio for remote Jabber clients are often caused by insufficient resources on traversal zones and failed ICE candidate negotiation. Traversal zones are essential for secure registration, signaling, and media path establishment between remote clients and CUCM. ICE, or Interactive Connectivity Establishment, allows endpoints to select optimal IP addresses and ports for RTP streams, facilitating NAT and firewall traversal. When traversal zones operate near capacity, ICE candidate exchanges may fail, causing registration failures and misdirected media that results in one-way audio.
Option A, increasing traversal zone resources and enabling full bidirectional ICE support, resolves the issue by ensuring that traversal zones have sufficient CPU and memory to handle peak concurrent connections. Full bidirectional ICE support guarantees that candidates are exchanged successfully between remote clients and the Expressway server, enabling RTP streams to traverse NAT and firewall devices effectively. Cisco best practices recommend scaling traversal zones based on expected remote client numbers and enabling bidirectional ICE to maintain reliable registration and media path establishment. Continuous monitoring ensures proactive resource allocation, maintaining seamless connectivity and high-quality voice services even during peak usage periods.
Option B, requiring VPN connectivity, can bypass some NAT-related problems but introduces latency, additional complexity, and potential points of failure. VPN does not address traversal zone resource constraints or ICE negotiation failures and is therefore suboptimal for large-scale deployments.
Option C, reducing CUCM SIP registration timers, increases registration retry frequency but does not resolve resource limitations. Excessive retries under high load may worsen congestion and fail to improve registration success.
Option D, enabling persistent XMPP connections improves messaging reliability and presence updates but does not impact ICE negotiation or traversal zone performance. Persistent XMPP connections do not influence registration success or RTP path establishment, leaving one-way audio and intermittent registration issues unresolved.
By increasing traversal zone resources and enabling full bidirectional ICE support, remote Jabber clients can reliably register and establish correct RTP paths, providing high-quality voice communication. This approach supports scalable remote collaboration, reduces call drops, aligns with Cisco best practices, and ensures predictable performance for enterprise users. Proper sizing and ICE configuration prevent operational disruptions, deliver seamless connectivity, and maintain high-quality communication for all remote users regardless of network conditions.
Question158
Voicemail users accessing Unity Connection report initial silence of several seconds during message playback. RTP analysis shows no packet loss, jitter, or network delay. 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:
The initial silence during voicemail playback is commonly caused by Voice Activity Detection (VAD), which suppresses RTP transmission during periods of low audio energy to conserve bandwidth during live calls. Voicemail messages often begin with soft sounds or silence that VAD interprets as inactivity, delaying RTP transmission and causing users to perceive a gap at the start of messages.
Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, resolves the issue by ensuring continuous RTP transmission, including during low-energy segments. Cisco best practices recommend disabling VAD for voicemail services to maintain message integrity and provide consistent playback. While disabling VAD slightly increases bandwidth usage, it eliminates the initial silence problem, improving user experience and reducing voicemail-related support calls. Continuous RTP transmission ensures immediate playback of messages, preserving message integrity and maintaining consistent behavior across all endpoints.
Option B, adjusting MWI extensions, impacts visual notifications for new messages but does not affect RTP or playback timing. Changes to MWI influence lamp signaling rather than media transmission, leaving the initial silence problem unresolved.
Option C, changing Unity Connection codec to G.722, improves audio fidelity but does not correct VAD-induced delays. Codec changes affect sound quality but cannot overcome the suppression of RTP during low-energy segments.
Option D, moving the voicemail pilot to a different partition, affects call routing but does not impact RTP or VAD behavior. Partition changes have no effect on message playback timing.
Disabling VAD ensures immediate and continuous playback of voicemail messages, improving user experience and maintaining message integrity. This configuration aligns with Cisco best practices, reduces support incidents, and guarantees predictable behavior for enterprise voicemail users. Proper VAD configuration ensures reliable service without compromising overall call quality or operational efficiency, maintaining high-quality voicemail access.
Question159
During multi-site CUCM calls over SIP trunks via CUBE, users experience occasional call drops and one-way audio. RTP packets are sometimes sent to incorrect IP addresses before SDP rewriting occurs. Which configuration MOST effectively resolves this issue?
A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Convert SIP trunks to SCCP
C) Use SIP over TCP exclusively
D) Disable early media on CUBE
Answer: A
Explanation:
One-way audio and call drops in multi-site SIP trunk deployments are typically caused by RTP being sent before CUBE has anchored the media and rewritten SDP. SDP contains media parameters, including IP addresses, ports, and codecs, that must be accurate for proper media delivery. Premature RTP transmission leads to misrouted packets, causing audio issues and dropped calls.
Option A, enabling delayed-offer SDP, directly addresses this problem by delaying SDP from the INVITE to the 200 OK response. This gives CUBE the opportunity to anchor media correctly and modify SDP so that RTP flows are correctly routed. Cisco best practices recommend delayed-offer SDP for multi-site deployments to align media and signaling paths, improve call stability, and reduce troubleshooting complexity. Delayed-offer SDP prevents misdirected RTP, stabilizes inter-site communications, reduces call drops, and ensures predictable audio quality. It supports interoperability with different SIP endpoints and clusters and aligns with enterprise communication requirements for scalable, reliable multi-site voice services.
Option B, converting SIP trunks to SCCP, changes signaling but does not prevent early RTP transmission. SCCP may improve endpoint compatibility but does not resolve RTP misdirection.
Option C, using SIP over TCP, ensures reliable signaling delivery but does not influence RTP timing relative to SDP rewriting. TCP guarantees message delivery but does not prevent one-way audio caused by early RTP.
Option D, disabling early media on CUBE, affects call progress tones but does not correct RTP misrouting. Early media configuration influences signaling behavior rather than the media path.
Delayed-offer SDP ensures correct media anchoring, predictable RTP flows, and consistent call quality. This configuration stabilizes inter-site communications, aligns with Cisco best practices, reduces troubleshooting overhead, and guarantees high-quality voice communication across multi-site deployments. Proper SDP handling is critical for enterprise SIP trunking and supports scalable, reliable inter-site voice services.
Question160
Remote Jabber clients behind NAT report intermittent registration failures and one-way audio when connecting through CUCM Expressway traversal zones. ICE candidate negotiation fails frequently, and traversal zones operate near capacity. Which configuration MOST effectively resolves this issue?
A) Increase traversal zone resources and enable full bidirectional ICE support
B) Require VPN connectivity for remote clients
C) Reduce CUCM SIP registration timers
D) Enable persistent XMPP connections between Expressway-C and CUCM
Answer: A
Explanation:
Remote Jabber clients rely on Expressway traversal zones to establish secure registration and media paths with CUCM. Intermittent registration failures and one-way audio commonly occur due to insufficient traversal zone resources and failed ICE negotiation. ICE enables endpoints to determine optimal IP addresses and ports for RTP streams, facilitating NAT and firewall traversal. Overloaded traversal zones may fail ICE candidate exchanges, resulting in registration failures and misdirected media, producing one-way audio.
Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses the root cause. Allocating additional CPU and memory ensures traversal zones can handle peak concurrent connections without failures. Full bidirectional ICE ensures successful candidate exchange between remote clients and Expressway, enabling RTP streams to traverse NAT and firewall environments effectively. Cisco best practices recommend proper sizing of traversal zones and enabling full ICE support to maintain reliable registration and media paths. Continuous monitoring ensures proactive resource allocation, maintaining seamless connectivity and high-quality voice services during peak usage periods.
Option B, requiring VPN connectivity, bypasses some NAT-related issues but introduces latency, complexity, and potential failure points. VPN does not resolve traversal zone capacity issues or ICE negotiation failures.
Option C, reducing CUCM SIP registration timers, increases retry frequency but does not solve resource constraints and may worsen congestion.
Option D, enabling persistent XMPP connections, improves messaging reliability but does not influence ICE negotiation or traversal zone performance. Persistent XMPP does not resolve registration or media path failures.
Increasing traversal zone resources and enabling bidirectional ICE ensures successful registration, proper RTP path establishment, and high-quality voice for remote Jabber clients. This solution supports scalable remote collaboration, aligns with Cisco best practices, and delivers predictable, reliable enterprise voice services for remote users, even during periods of peak load. Proper sizing and ICE configuration maintain seamless connectivity and high-quality communication.
Question161
A network administrator observes that SIP trunk calls between two CUCM clusters through CUBE intermittently fail and experience one-way audio. Analysis shows RTP packets being sent to incorrect IP addresses before SDP rewriting occurs. Which configuration MOST effectively resolves this issue?
A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Convert SIP trunks to SCCP
C) Use SIP over TCP exclusively
D) Disable early media on CUBE
Answer: A
Explanation:
Intermittent call failures and one-way audio in SIP trunk deployments are typically due to RTP being transmitted before the CUBE has anchored the media path and rewritten SDP. SDP, or Session Description Protocol, conveys crucial media information such as IP addresses, ports, and codec capabilities. When RTP packets are sent prematurely, before the SDP is anchored and rewritten by the CUBE, the media may be misrouted, resulting in one-way audio or call failures.
Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, directly mitigates this issue by postponing the inclusion of SDP in the INVITE until the 200 OK response. Delayed-offer SDP ensures that CUBE has sufficient time to anchor the media path and properly modify SDP information so that RTP flows are directed correctly. Cisco recommends this configuration in multi-site deployments because it synchronizes signaling and media, stabilizes call quality, and reduces troubleshooting complexity. Delayed-offer SDP guarantees interoperability between SIP endpoints and clusters, maintains consistent audio quality, and aligns with enterprise requirements for reliable inter-site communication. Implementing delayed-offer SDP also simplifies operational support, reduces call drops, and ensures predictable media flows across complex topologies. This configuration allows enterprises to achieve high-quality voice communication across multiple sites while preventing the misrouting of RTP packets.
Option B, converting SIP trunks to SCCP, changes the signaling protocol but does not prevent early RTP transmission. While SCCP can improve endpoint compatibility in certain deployments, it does not address the fundamental issue of RTP being sent prior to SDP rewriting. The underlying cause of one-way audio and call failures remains unresolved.
Option C, using SIP over TCP exclusively, ensures reliable delivery of signaling messages but does not influence RTP timing relative to SDP rewriting. TCP guarantees that signaling messages are delivered without loss, but premature RTP transmission continues to cause misrouted media and one-way audio issues.
Option D, disabling early media on CUBE, impacts call progress tones and ringback behavior but does not prevent RTP misrouting. Early media configuration affects how audio is played during call setup but does not correct media path establishment.
Implementing delayed-offer SDP ensures that RTP flows only after the media path has been anchored correctly, providing predictable and high-quality inter-site communications. This configuration aligns with Cisco best practices for SIP trunking, stabilizes call performance, reduces troubleshooting overhead, and guarantees seamless voice communication across enterprise networks. Proper SDP handling improves interoperability, ensures consistent user experience, and supports scalable, reliable multi-site deployments.
Question162
Remote Jabber clients behind NAT report frequent registration failures and one-way audio when connecting through Expressway traversal zones. ICE candidate negotiation often fails, and traversal zones are approaching maximum capacity. Which configuration MOST effectively resolves this issue?
A) Increase traversal zone resources and enable full bidirectional ICE support
B) Require VPN connectivity for remote clients
C) Reduce CUCM SIP registration timers
D) Enable persistent XMPP connections between Expressway-C and CUCM
Answer: A
Explanation:
Remote Jabber client issues such as registration failures and one-way audio are typically caused by resource constraints on traversal zones and failed ICE negotiation. Traversal zones facilitate secure registration, signaling, and media path establishment between remote clients and CUCM. ICE allows endpoints to select the optimal IP addresses and ports for RTP streams, enabling NAT and firewall traversal. When traversal zones are overloaded or near capacity, ICE candidate exchanges may fail, causing registration failures and misdirected RTP, resulting in one-way audio.
Option A, increasing traversal zone resources and enabling full bidirectional ICE support, addresses the root cause. Allocating additional CPU and memory ensures that traversal zones can handle peak concurrent connections without failures. Enabling full bidirectional ICE guarantees successful candidate exchange between remote clients and the Expressway server, ensuring that RTP streams traverse NAT and firewall devices effectively. Cisco best practices recommend sizing traversal zones based on expected remote client populations and enabling full ICE support to maintain reliable registration and media path establishment. Continuous monitoring allows administrators to proactively scale resources, ensuring seamless connectivity and high-quality voice services even during peak usage periods.
Option B, requiring VPN connectivity, can circumvent some NAT-related issues but introduces latency, complexity, and potential points of failure. VPN solutions do not address traversal zone resource limitations or ICE negotiation failures, making them suboptimal for large-scale deployments.
Option C, reducing CUCM SIP registration timers, increases registration retry frequency but does not resolve underlying resource constraints. Excessive retries can worsen congestion and fail to improve registration success.
Option D, enabling persistent XMPP connections improves messaging reliability and presence updates but does not affect ICE negotiation or traversal zone performance. Persistent XMPP connections have no impact on registration success or RTP path establishment, leaving one-way audio and intermittent registration unresolved.
Increasing traversal zone resources and enabling full bidirectional ICE ensures successful registration, proper RTP path establishment, and high-quality voice for remote Jabber users. This configuration supports scalable remote collaboration, aligns with Cisco best practices, and provides predictable and reliable enterprise voice services. Proper sizing and ICE configuration ensure seamless connectivity, high-quality communication, and reduced support incidents.
Question163
Unity Connection voicemail users report several seconds of silence at the start of message playback. RTP analysis shows no packet loss, jitter, or network delay. 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 transmission during periods of low audio energy to conserve bandwidth during live calls. Voicemail messages often start with soft sounds or silence, which VAD interprets as inactivity, delaying RTP transmission and causing initial playback silence.
Option A, disabling VAD on the SIP trunk between CUCM and Unity Connection, resolves this issue by allowing RTP to flow continuously, including during low-energy segments. Cisco best practices recommend disabling VAD for voicemail traffic to preserve message integrity and provide consistent playback. Although disabling VAD slightly increases bandwidth usage, it eliminates the initial silence problem, improving user experience and reducing voicemail-related support calls. Continuous RTP transmission ensures immediate message playback, maintaining predictable behavior across all endpoints and enhancing overall user satisfaction.
Option B, adjusting MWI extensions, affects visual notifications for new messages but does not influence RTP or playback timing. MWI changes impact lamp signaling rather than the media path, leaving the initial silence problem unresolved.
Option C, changing Unity Connection codec to G.722, improves audio fidelity but does not address VAD-induced delays. Codec selection affects sound quality but cannot overcome RTP suppression during low-energy segments.
Option D, moving the voicemail pilot to a different partition, modifies call routing but does not affect RTP or VAD behavior. Partition adjustments have no impact on initial message playback timing.
Disabling VAD ensures continuous RTP flow and immediate playback, improving user experience and maintaining message integrity. This approach aligns with Cisco best practices, reduces support incidents, and ensures predictable voicemail behavior across the enterprise. Proper configuration guarantees reliable service without compromising overall call quality, maintaining consistent and high-quality voicemail access for all users.
Question164
During inter-site CUCM calls over SIP trunks routed through CUBE, users report occasional call drops and one-way audio. RTP is sometimes transmitted to incorrect IP addresses prior to SDP rewriting. Which configuration MOST effectively resolves this issue?
A) Enable delayed-offer SDP on CUCM SIP trunks toward CUBE
B) Convert SIP trunks to SCCP
C) Use SIP over TCP exclusively
D) Disable early media on CUBE
Answer: A
Explanation:
One-way audio and call drops in inter-site SIP trunk deployments often occur because RTP is sent before CUBE anchors the media path and rewrites SDP. SDP contains media parameters, including IP addresses, ports, and codecs, which must be accurate for proper RTP delivery. Premature RTP transmission results in misrouted packets, leading to one-way audio and dropped calls.
Option A, enabling delayed-offer SDP on CUCM SIP trunks toward CUBE, addresses this problem by delaying SDP transmission from the INVITE to the 200 OK response. This allows CUBE to anchor the media path and correctly modify SDP, ensuring that RTP is delivered to the correct endpoints. Cisco best practices recommend delayed-offer SDP for multi-site deployments to synchronize media and signaling paths, enhance call stability, and reduce troubleshooting complexity. This configuration prevents misdirected RTP, stabilizes inter-site communication, reduces call drops, and ensures predictable audio quality.
Option B, converting SIP trunks to SCCP, changes signaling but does not prevent early RTP transmission. While SCCP may improve endpoint interoperability in some scenarios, it does not address RTP misrouting caused by early SDP.
Option C, using SIP over TCP, guarantees reliable signaling delivery but does not influence RTP timing relative to SDP rewriting. TCP ensures message delivery but does not resolve one-way audio caused by early RTP.
Option D, disabling early media on CUBE, affects call progress tones but does not correct misrouted RTP packets. Early media impacts signaling rather than media path establishment.
Delayed-offer SDP ensures proper media anchoring, predictable RTP flows, and consistent call quality. It stabilizes inter-site communications, reduces troubleshooting overhead, and guarantees reliable voice communication in multi-site deployments. Proper SDP handling supports scalable and predictable enterprise SIP trunking.
Question165
Remote Jabber users behind NAT experience intermittent registration failures and one-way audio when connecting through CUCM Expressway traversal zones. ICE negotiation frequently fails, and traversal zones are operating near capacity. Which configuration MOST effectively resolves this issue?
A) Increase traversal zone resources and enable full bidirectional ICE support
B) Require VPN connectivity for remote clients
C) Reduce CUCM SIP registration timers
D) Enable persistent XMPP connections between Expressway-C and CUCM
Answer: A
Explanation:
Remote Jabber clients depend on Expressway traversal zones for secure registration and media path establishment with CUCM. Registration failures and one-way audio often occur due to limited traversal zone resources and failed ICE negotiation. ICE allows endpoints to determine optimal IP addresses and ports for RTP streams, enabling NAT and firewall traversal. When traversal zones operate near capacity, ICE candidate exchanges may fail, resulting in registration errors and misrouted RTP, causing one-way audio.
Option A, increasing traversal zone resources and enabling full bidirectional ICE support, directly addresses this problem. Allocating additional CPU and memory ensures traversal zones can handle peak concurrent connections reliably. Enabling bidirectional ICE guarantees successful candidate exchange between remote clients and Expressway servers, allowing RTP streams to traverse NAT and firewall devices effectively. Cisco best practices recommend proper sizing of traversal zones based on expected remote client populations and enabling bidirectional ICE to maintain reliable registration and media paths. Continuous monitoring allows administrators to scale resources proactively, ensuring seamless connectivity and high-quality voice services during peak periods.
Option B, requiring VPN connectivity, circumvents some NAT-related issues but adds latency, complexity, and potential points of failure. VPN does not resolve traversal zone capacity or ICE negotiation issues, making it suboptimal for large deployments.
Option C, reducing CUCM SIP registration timers, increases retry frequency but does not address resource limitations, and excessive retries can exacerbate congestion.
Option D, enabling persistent XMPP connections improves messaging reliability but does not influence ICE negotiation or traversal zone performance. Persistent XMPP has no impact on registration or RTP path issues, leaving one-way audio and intermittent registration unresolved.
By increasing traversal zone resources and enabling bidirectional ICE, remote Jabber clients can register successfully, establish proper RTP paths, and experience high-quality voice communication. This solution supports scalable remote collaboration, aligns with Cisco best practices, and delivers reliable enterprise voice services even during periods of high load. Proper sizing and ICE configuration maintain seamless connectivity, high-quality communication, and predictable performance.