{"id":21270,"date":"2026-09-24T11:49:37","date_gmt":"2026-09-24T11:49:37","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=21270"},"modified":"2026-09-24T11:49:37","modified_gmt":"2026-09-24T11:49:37","slug":"cisco-ccnp-collaboration-350-801-practice-test-questions-and-exam-dumps-part8-q141-160","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-350-801-practice-test-questions-and-exam-dumps-part8-q141-160\/","title":{"rendered":"Cisco CCNP Collaboration 350-801 Practice Test Questions and Exam Dumps Part8 Q141-160"},"content":{"rendered":"<h2><b>View Full <\/b><a href=\"https:\/\/www.examlabs.com\/350-801-exam-dumps\"><b>Cisco CCNP Collaboration 350-801 Exam Dumps<\/b><\/a><b> and Practice Test Dumps.<\/b><\/h2>\n<p>&nbsp;<\/p>\n<h3><b>Question 141<\/b><\/h3>\n<p><b>Which CUCM feature determines which partitions a user or device can access when placing a call?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Region<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Calling Search Space<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Device Pool<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Route Group<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 2<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">A Calling Search Space determines which partitions a device, line, or user can search when processing a call in Cisco Unified Communications Manager. This provides an effective way to control access to internal extensions, external routes, services, and other dial-plan destinations. Partitions contain the destinations, while the CSS determines which of those partitions are visible to the caller. Regions control codec and bandwidth relationships, device pools provide shared configuration, and route groups organize gateways. When a user receives a message indicating that a destination cannot be reached, administrators should verify the caller&#8217;s CSS and the partition assigned to the target destination.<\/span><\/p>\n<h3><b>Question 142<\/b><\/h3>\n<p><b>Which SIP message is used to initiate a new SIP session?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">BYE<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">REGISTER<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">INVITE<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">OPTIONS<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 3<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">The SIP INVITE method is used to initiate a new session or modify an existing session. In a voice or video call, an INVITE typically contains session information, often including SDP, that describes the proposed media characteristics. BYE terminates an established session, REGISTER associates an endpoint with a registrar, and OPTIONS can be used to query capabilities or availability. When troubleshooting call setup, administrators commonly inspect the INVITE and subsequent responses to determine whether the destination was reached and whether media negotiation succeeded. Understanding the purpose of SIP methods makes signaling traces easier to interpret and helps identify where a call setup process fails.<\/span><\/p>\n<h3><b>Question 143<\/b><\/h3>\n<p><b>Which CUCM component groups gateways for selection by a route list?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Route Group<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Route Pattern<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Translation Pattern<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Device Pool<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 1<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">A Route Group logically groups one or more gateways for use by CUCM call-routing functions. Route lists can reference route groups and provide an ordered sequence for selecting routing resources. This structure is useful for implementing primary and backup gateways and simplifying enterprise dial-plan administration. Route patterns identify digit patterns, translation patterns modify numbers, and device pools provide common endpoint settings. If outbound calls are not using the expected gateway, administrators should verify the route pattern, route list, route group membership, gateway status, and any digit manipulation. Proper route-group configuration provides flexibility and redundancy for external call routing.<\/span><\/p>\n<h3><b>Question 144<\/b><\/h3>\n<p><b>Which technology protects real-time voice and video media from unauthorized interception?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">SIP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">TLS<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">SRTP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">LDAP<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 3<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">Secure Real-time Transport Protocol, or SRTP, is designed to protect real-time voice and video media. It provides encryption and integrity protection for RTP traffic, helping prevent unauthorized parties from reading or modifying media streams. SIP is used for signaling, TLS commonly protects signaling connections, and LDAP provides directory services. Secure media depends on compatible endpoint capabilities and successful security negotiation. When troubleshooting encrypted calls, administrators should check whether both endpoints support SRTP, whether security profiles are correctly configured, and whether the media path permits the negotiated traffic. Understanding the separate roles of TLS and SRTP is essential when implementing end-to-end security.<\/span><\/p>\n<h3><b>Question 145<\/b><\/h3>\n<p><b>Which CUCM configuration provides common settings to a group of devices at a particular site or location?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Device Pool<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Partition<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Route Group<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Translation Pattern<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 1<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">A Device Pool provides common configuration settings to a group of devices. It can include settings such as CUCM Group, region, date\/time group, SRST reference, and other parameters depending on the deployment. Assigning devices to an appropriate device pool simplifies administration and ensures consistent behavior across endpoints with similar requirements. Partitions organize dial-plan destinations, route groups organize gateways, and translation patterns transform numbers. When deploying phones across different sites, administrators can create separate device pools according to location, network characteristics, and call-processing requirements. Reviewing device-pool assignments is also useful when troubleshooting registration, codec, time, or failover behavior.<\/span><\/p>\n<h3><b>Question 146<\/b><\/h3>\n<p><b>Which protocol is responsible for carrying media packets after a SIP session is established?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">LDAP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">RTP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">NTP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">DNS<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 2<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">Real-time Transport Protocol, or RTP, carries real-time audio and video media after signaling establishes a session. RTP uses sequence numbers and timestamps to support proper media processing and synchronization. SIP is responsible for signaling rather than carrying the actual audio or video stream. LDAP supports directory services, NTP synchronizes system time, and DNS provides name resolution. RTP traffic requires careful network design because packet loss, jitter, and delay can directly affect media quality. If signaling succeeds but users experience one-way audio, no audio, or degraded video, administrators should investigate the RTP path, firewall policies, NAT behavior, routing, codec negotiation, and endpoint media configuration.<\/span><\/p>\n<h3><b>Question 147<\/b><\/h3>\n<p><b>Which Cisco collaboration component provides centralized voicemail functionality?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco CUBE<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco Expressway<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco Unity Connection<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco IM and Presence<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 3<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">Cisco Unity Connection provides centralized voicemail functionality for supported Cisco collaboration environments. It manages voicemail boxes, greetings, message storage, call handlers, message notifications, and user access to voicemail. CUCM can redirect unanswered or busy calls to Unity Connection through the configured voicemail integration. CUBE provides SIP border functions, Expressway provides edge connectivity, and IM and Presence supports messaging and presence. If voicemail is not working, administrators should verify CUCM voicemail configuration, ports, call routing, mailbox settings, integration status, and network connectivity. Proper voicemail integration ensures that users receive messages reliably and can access them through supported clients and telephony interfaces.<\/span><\/p>\n<h3><b>Question 148<\/b><\/h3>\n<p><b>Which DNS record type can identify the hostname and port for a specific service?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">A<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">CNAME<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">PTR<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">SRV<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 4<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">An SRV record identifies the hostname and port associated with a particular network service. Cisco collaboration environments can use SRV records to support service discovery and help endpoints locate appropriate services. An A record maps a hostname to an IPv4 address, a CNAME provides an alias, and a PTR record is used for reverse DNS lookups. When collaboration endpoints fail to discover a required service, administrators should inspect both SRV and host records, verify DNS server reachability, and confirm that records contain the correct values. Accurate DNS service discovery is particularly important for distributed collaboration systems and remote-access deployments.<\/span><\/p>\n<h3><b>Question 149<\/b><\/h3>\n<p><b>Which CUCM feature is used to transform a number before routing it to another destination?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Translation Pattern<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Region<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Media Resource Group<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Device Pool<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 1<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">A Translation Pattern can transform a dialed number before CUCM continues processing the call. Depending on the configuration, it can add digits, remove digits, or otherwise change the number to match a required numbering format. This is useful when internal dialing conventions differ from the format required by gateways or external service providers. Regions control codec and bandwidth behavior, media resource groups organize media resources, and device pools provide common endpoint settings. When troubleshooting unexpected called-number transformations, administrators should inspect translation patterns, partitions, calling search spaces, and other digit-manipulation settings. Clear transformation rules make complex dial plans easier to maintain and troubleshoot.<\/span><\/p>\n<h3><b>Question 150<\/b><\/h3>\n<p><b>Which protocol is used to synchronize time across Cisco collaboration infrastructure?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">NTP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">RTP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">SIP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">DHCP<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 1<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">Network Time Protocol, or NTP, synchronizes clocks across Cisco collaboration servers and network devices. Accurate time is essential for certificate validation, authentication, logging, troubleshooting, and other security-related operations. RTP transports real-time media, SIP provides session signaling, and DHCP supplies network configuration. Collaboration systems should use reliable NTP sources and administrators should verify actual synchronization rather than assuming that configuration alone is sufficient. Significant clock differences can lead to certificate failures, authentication issues, and confusing event timestamps. When troubleshooting security or synchronization problems, checking NTP status and system time on all relevant components should be part of the initial investigation.<\/span><\/p>\n<h3><b>Question 151<\/b><\/h3>\n<p><b>Which CUCM feature can control whether a user is allowed to place calls to specific classes of destinations?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Device Pool<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Calling Search Space<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Region<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Media Resource Group<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 2<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">Calling Search Spaces are used to control which dial-plan destinations a user or device can access. They contain partitions that determine which directory numbers, route patterns, translation patterns, and other destinations can be searched. This allows administrators to create different calling privileges for different users or devices. Device pools provide shared configuration, regions manage codec and bandwidth relationships, and media resource groups organize media resources. If one user can reach a destination while another cannot, administrators should compare their Calling Search Spaces and the partitions assigned to the destination. Proper CSS configuration provides an organized method for enforcing calling restrictions within CUCM.<\/span><\/p>\n<h3><b>Question 152<\/b><\/h3>\n<p><b>Which Cisco technology provides SIP border functions between an enterprise and an external provider?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco IM and Presence<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco Unified Border Element<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco TFTP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco Unity Connection<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 2<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">Cisco Unified Border Element, or CUBE, provides SIP border functions between enterprise collaboration systems and external SIP networks. It can support SIP normalization, interoperability, topology hiding, media handling, and security controls. CUBE is commonly deployed between CUCM and a SIP service provider, creating a controlled boundary between the internal and external environments. IM and Presence provides messaging and presence, TFTP supports provisioning, and Unity Connection provides voicemail. When troubleshooting a SIP provider connection, administrators should examine CUBE dial peers, SIP signaling, codec negotiation, number manipulation, routing, NAT, and firewall policies to identify connectivity or interoperability problems.<\/span><\/p>\n<h3><b>Question 153<\/b><\/h3>\n<p><b>Which CUCM object identifies the pattern of digits that should be matched for routing?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Route Pattern<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Route Group<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Device Pool<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Media Resource Group<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 1<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">A Route Pattern defines the digit pattern CUCM uses to match a dialed number for routing. Once a number matches the pattern, CUCM can direct the call toward the configured routing destination, such as a route list or gateway. Route groups organize gateways, device pools provide common endpoint configuration, and media resource groups organize resources such as conference bridges and transcoders. Route patterns are central to CUCM dial-plan design and can support internal, external, emergency, and service numbers. During troubleshooting, administrators should inspect the dialed digits, pattern, partition, Calling Search Space, route list, route group, and gateway to identify routing problems.<\/span><\/p>\n<h3><b>Question 154<\/b><\/h3>\n<p><b>Which SIP response indicates that a service is temporarily unavailable?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">404<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">503<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">200<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">486<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 2<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">SIP response code 503 indicates that the service is currently unavailable. It can occur when a SIP server or intermediary cannot provide the requested service at that time. A 404 response generally indicates that the requested resource was not found, 200 indicates successful processing, and 486 commonly indicates that the destination is busy. When a 503 response occurs, administrators should investigate server availability, resource utilization, routing, SIP trunk status, DNS resolution, and intermediate devices such as CUBE. Reviewing the complete SIP signaling exchange can help determine which component generated the response and whether the condition is temporary or caused by a configuration issue.<\/span><\/p>\n<h3><b>Question 155<\/b><\/h3>\n<p><b>Which Cisco collaboration service provides instant messaging and presence information?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco Expressway<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco Unity Connection<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco IM and Presence<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Cisco CUBE<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 3<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">Cisco IM and Presence provides instant messaging and presence capabilities in supported Cisco collaboration deployments. Presence allows users and applications to see availability information, while instant messaging supports real-time text communication. Expressway provides edge connectivity, Unity Connection provides voicemail, and CUBE provides SIP border functions. IM and Presence commonly integrates with CUCM and directory services to obtain user and availability information. When users cannot see presence states or exchange messages, administrators should verify service status, CUCM integration, user configuration, directory synchronization, authentication, and network connectivity. Correct integration is necessary for reliable presence publication, subscription, and messaging.<\/span><\/p>\n<h3><b>Question 156<\/b><\/h3>\n<p><b>Which CUCM configuration is used to group media resources such as transcoders and conference bridges?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Route Group<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Media Resource Group<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Calling Search Space<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Partition<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 2<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">A Media Resource Group, or MRG, groups media resources such as conference bridges, transcoders, and music-on-hold resources. MRGs can then be placed into Media Resource Group Lists, which determine the media resources available to devices. Route groups organize gateways, calling search spaces control dial-plan access, and partitions organize destinations. If an endpoint cannot access a required media service, administrators should verify its assigned MRGL, the associated MRG, and the status of the underlying resource. Proper media-resource grouping helps administrators organize collaboration resources and control which endpoints can use particular conference, transcoding, or media services.<\/span><\/p>\n<h3><b>Question 157<\/b><\/h3>\n<p><b>Which codec is generally associated with wideband speech in Cisco collaboration systems?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">G.729<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">G.711<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">G.722<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">G.726<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 3<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">G.722 is a wideband audio codec commonly used to provide enhanced speech quality compared with traditional narrowband codecs. When supported by the endpoints and network path, it can provide greater speech clarity. G.711 provides high-quality narrowband audio, while G.729 is a compressed codec designed for lower bandwidth usage. Codec selection depends on endpoint capabilities, region configuration, trunks, and intermediate media resources. When the negotiated codec differs from the expected one, administrators should inspect the region relationships, endpoint capabilities, and signaling details. The selected codec must be supported throughout the required media path for the call to operate correctly.<\/span><\/p>\n<h3><b>Question 158<\/b><\/h3>\n<p><b>Which CUCM component specifies the prioritized CUCM servers used by registered devices?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">CUCM Group<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Route List<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Partition<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">Translation Pattern<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 1<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">A CUCM Group specifies an ordered list of CUCM servers that devices can use for registration and call processing. Device pools commonly reference CUCM Groups, allowing administrators to assign registration behavior to groups of endpoints. The ordered server list can provide redundancy if a preferred CUCM server becomes unavailable. Route lists control gateway routing, partitions organize dial-plan destinations, and translation patterns transform numbers. When a phone cannot register or does not fail over as expected, administrators should inspect the assigned device pool, CUCM Group, server availability, network connectivity, and TFTP configuration. Proper CUCM Group design supports availability and predictable endpoint registration behavior.<\/span><\/p>\n<h3><b>Question 159<\/b><\/h3>\n<p><b>Which technology can provide basic telephony service at a branch during loss of connectivity to centralized CUCM?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">DNS<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">SRST<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">LDAP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">CDP<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 2<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">Survivable Remote Site Telephony, or SRST, provides backup call-processing functionality for supported Cisco IP phones when communication with centralized CUCM servers is unavailable. It is typically implemented on a Cisco router at the remote site. During normal operation, endpoints register with CUCM, while SRST provides a fallback mode during WAN or centralized-service failures. DNS provides name resolution, LDAP provides directory integration, and CDP provides neighbor discovery. Administrators should configure the appropriate SRST reference in CUCM and verify the router&#8217;s SRST settings. Testing failover ensures that phones can transition correctly and that essential branch telephony functions remain available during an outage.<\/span><\/p>\n<h3><b>Question 160<\/b><\/h3>\n<p><b>Which protocol provides secure signaling encryption for supported SIP connections?<\/b><\/p>\n<ol>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">RTP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">SRTP<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">TLS<\/span><\/li>\n<li style=\"font-weight: 400;\" aria-level=\"1\"><span style=\"font-weight: 400;\">NTP<\/span><\/li>\n<\/ol>\n<p><b>Correct Answer: 3<\/b><\/p>\n<p><b>Explanation<\/b><\/p>\n<p><span style=\"font-weight: 400;\">TLS provides encryption and authentication for supported SIP signaling connections. It protects signaling messages from unauthorized interception and modification and commonly uses digital certificates to establish trust. SRTP protects real-time media, RTP carries media without the additional SRTP security layer, and NTP provides time synchronization. Secure SIP deployments require correct security profiles, valid certificates, appropriate trust relationships, and accurate system time. When TLS signaling fails, administrators should check certificate validity, hostname matching, trust chains, expiration dates, and connectivity. It is important to troubleshoot TLS and SRTP separately because signaling security and media security are different parts of a secure collaboration architecture.<\/span><\/p>\n<p>&nbsp;<\/p>\n","protected":false},"excerpt":{"rendered":"<p>View Full Cisco CCNP Collaboration 350-801 Exam Dumps and Practice Test Dumps. &nbsp; Question 141 Which CUCM feature determines which partitions a user or device can access when placing a call? Region Calling Search Space Device Pool Route Group Correct Answer: 2 Explanation A Calling Search Space determines which partitions a device, line, or user [&hellip;]<\/p>\n","protected":false},"author":1,"featured_media":0,"comment_status":"closed","ping_status":"closed","sticky":false,"template":"","format":"standard","meta":[],"categories":[1648,1647],"tags":[],"_links":{"self":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21270"}],"collection":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts"}],"about":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/types\/post"}],"author":[{"embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/users\/1"}],"replies":[{"embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/comments?post=21270"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21270\/revisions"}],"predecessor-version":[{"id":21271,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21270\/revisions\/21271"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=21270"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=21270"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=21270"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}