{"id":21266,"date":"2026-09-24T11:49:05","date_gmt":"2026-09-24T11:49:05","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=21266"},"modified":"2026-09-24T11:49:05","modified_gmt":"2026-09-24T11:49:05","slug":"cisco-ccnp-collaboration-350-801-practice-test-questions-and-exam-dumps-part6-q101-120","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-350-801-practice-test-questions-and-exam-dumps-part6-q101-120\/","title":{"rendered":"Cisco CCNP Collaboration 350-801 Practice Test Questions and Exam Dumps Part6 Q101-120"},"content":{"rendered":"

View Full <\/b>Cisco CCNP Collaboration 350-801 Exam Dumps<\/b><\/a> and Practice Test Dumps.<\/b><\/h2>\n

 <\/p>\n

Question 101<\/b><\/h3>\n

Which CUCM configuration allows administrators to control access to specific dial-plan destinations?<\/b><\/p>\n

    \n
  1. Region<\/span><\/li>\n
  2. Calling Search Space<\/span><\/li>\n
  3. Device Pool<\/span><\/li>\n
  4. Route Group<\/span><\/li>\n<\/ol>\n

    Correct Answer: 2<\/b><\/p>\n

    Explanation<\/b><\/p>\n

    Calling Search Spaces are used in Cisco Unified Communications Manager to control which partitions a device, line, or user can access. Because directory numbers, route patterns, translation patterns, and other destinations can be assigned to partitions, the CSS determines which of those destinations are searchable. This mechanism is commonly used to implement calling restrictions and different classes of service. When troubleshooting a call that is rejected because a destination cannot be reached, administrators should verify the calling device’s CSS and confirm that the target partition is included. Proper CSS and partition design provides predictable and controlled call routing throughout the collaboration environment.<\/span><\/p>\n

    Question 102<\/b><\/h3>\n

    Which SIP response code indicates that a requested resource was not found?<\/b><\/p>\n

      \n
    1. 486<\/span><\/li>\n
    2. 503<\/span><\/li>\n
    3. 404<\/span><\/li>\n
    4. 200<\/span><\/li>\n<\/ol>\n

      Correct Answer: 3<\/b><\/p>\n

      Explanation<\/b><\/p>\n

      SIP response code 404 indicates that the requested resource or destination was not found. It is commonly encountered when a SIP server cannot locate the requested user or destination. Response 200 indicates successful processing, 486 generally indicates that the destination is busy, and 503 indicates that the service is unavailable. When troubleshooting a 404 response, administrators should verify the Request-URI, SIP routing, dial peers, DNS configuration, and destination registration. Understanding SIP response codes helps identify whether a failure is related to addressing, availability, authentication, or another signaling condition. Packet captures and SIP traces can provide additional details during troubleshooting.<\/span><\/p>\n

      Question 103<\/b><\/h3>\n

      Which CUCM feature can remove or add digits to a number during call routing?<\/b><\/p>\n

        \n
      1. Digit manipulation<\/span><\/li>\n
      2. Device pool<\/span><\/li>\n
      3. Media Resource Group<\/span><\/li>\n
      4. Region<\/span><\/li>\n<\/ol>\n

        Correct Answer: 1<\/b><\/p>\n

        Explanation<\/b><\/p>\n

        Digit manipulation allows CUCM to modify called or calling numbers during call routing. Depending on the dial-plan design, administrators can add prefixes, remove access codes, or transform numbers into formats required by gateways and service providers. Digit manipulation can be configured through several CUCM routing mechanisms and should be planned carefully to avoid unexpected call behavior. Device pools provide shared endpoint settings, media resource groups organize media resources, and regions influence codec and bandwidth relationships. When a call reaches the correct gateway but the provider rejects the number, administrators should inspect the translated digits and verify that the expected numbering format is being sent.<\/span><\/p>\n

        Question 104<\/b><\/h3>\n

        Which protocol is commonly used to discover directly connected Cisco network devices?<\/b><\/p>\n

          \n
        1. NTP<\/span><\/li>\n
        2. SIP<\/span><\/li>\n
        3. LDAP<\/span><\/li>\n
        4. CDP<\/span><\/li>\n<\/ol>\n

          Correct Answer: 4<\/b><\/p>\n

          Explanation<\/b><\/p>\n

          Cisco Discovery Protocol, or CDP, is used by Cisco devices to discover information about directly connected Cisco neighbors. It can provide details such as device identification, platform, interface information, and management addresses. CDP is useful during network deployment and troubleshooting because administrators can determine which Cisco device is connected to a particular interface. NTP synchronizes time, SIP provides collaboration signaling, and LDAP supports directory integration. In a collaboration environment, CDP can also assist with identifying network connectivity for Cisco endpoints. Administrators should remember that CDP is primarily a neighbor-discovery mechanism and does not provide call signaling or media transport.<\/span><\/p>\n

          Question 105<\/b><\/h3>\n

          Which CUCM object provides an ordered list of route groups for call routing?<\/b><\/p>\n

            \n
          1. Route List<\/span><\/li>\n
          2. Partition<\/span><\/li>\n
          3. Device Pool<\/span><\/li>\n
          4. Media Resource Group<\/span><\/li>\n<\/ol>\n

            Correct Answer: 1<\/b><\/p>\n

            Explanation<\/b><\/p>\n

            A route list provides an ordered collection of route groups that CUCM can use when routing calls. This design allows administrators to define primary and alternate gateway paths and provides flexibility for redundancy. Each route group can contain one or more gateways. Partitions control dial-plan visibility, device pools provide common endpoint configuration, and media resource groups organize media resources. When implementing an enterprise dial plan, route patterns can point toward route lists, which then determine the available route groups. If the primary gateway becomes unavailable, CUCM can use another configured route group according to the route-list design.<\/span><\/p>\n

            Question 106<\/b><\/h3>\n

            Which technology provides secure media encryption for RTP-based voice and video streams?<\/b><\/p>\n

              \n
            1. TLS<\/span><\/li>\n
            2. SRTP<\/span><\/li>\n
            3. LDAP<\/span><\/li>\n
            4. DNS<\/span><\/li>\n<\/ol>\n

              Correct Answer: 2<\/b><\/p>\n

              Explanation<\/b><\/p>\n

              Secure Real-time Transport Protocol, or SRTP, protects RTP-based audio and video streams by providing encryption and integrity protection. It is designed specifically for real-time media and is negotiated separately from signaling security. TLS is commonly used to secure signaling connections, LDAP provides directory services, and DNS resolves hostnames and service records. When deploying secure collaboration services, administrators should verify that endpoints support compatible security settings and that the required key-management process succeeds. If signaling works but encrypted media fails, administrators should inspect SRTP negotiation, endpoint capabilities, security profiles, and the media path rather than focusing only on SIP signaling.<\/span><\/p>\n

              Question 107<\/b><\/h3>\n

              Which CUCM component determines the CUCM servers to which a device can register?<\/b><\/p>\n

                \n
              1. Route Group<\/span><\/li>\n
              2. Partition<\/span><\/li>\n
              3. CUCM Group<\/span><\/li>\n
              4. Translation Pattern<\/span><\/li>\n<\/ol>\n

                Correct Answer: 3<\/b><\/p>\n

                Explanation<\/b><\/p>\n

                A CUCM Group specifies the prioritized list of Cisco Unified Communications Manager servers that devices can use for registration and call processing. Device pools commonly reference a CUCM Group, allowing administrators to apply registration and redundancy behavior to groups of endpoints. Route groups organize gateways, partitions control dial-plan access, and translation patterns modify numbers. CUCM Groups are important for high availability because devices can use alternate servers if their preferred server becomes unavailable. When troubleshooting registration failures, administrators should verify the device pool, assigned CUCM Group, server availability, network connectivity, and TFTP configuration to determine where the registration process is failing.<\/span><\/p>\n

                Question 108<\/b><\/h3>\n

                Which SIP message is normally used to terminate an established SIP session?<\/b><\/p>\n

                  \n
                1. OPTIONS<\/span><\/li>\n
                2. REGISTER<\/span><\/li>\n
                3. INVITE<\/span><\/li>\n
                4. BYE<\/span><\/li>\n<\/ol>\n

                  Correct Answer: 4<\/b><\/p>\n

                  Explanation<\/b><\/p>\n

                  The SIP BYE message is used to terminate an established SIP session. After a call has been successfully established, either endpoint can send a BYE to request session termination. INVITE is used to establish or modify a session, REGISTER is used to register a SIP endpoint with a registrar, and OPTIONS can be used to query supported capabilities or availability. During troubleshooting, examining the SIP message sequence can reveal which endpoint initiated call termination and whether a network or application condition caused the disconnect. Understanding the purpose of common SIP methods is essential when analyzing call traces and identifying signaling problems.<\/span><\/p>\n

                  Question 109<\/b><\/h3>\n

                  Which CUCM feature associates media resources with devices that require them?<\/b><\/p>\n

                    \n
                  1. Media Resource Group List<\/span><\/li>\n
                  2. Route Pattern<\/span><\/li>\n
                  3. Calling Search Space<\/span><\/li>\n
                  4. Partition<\/span><\/li>\n<\/ol>\n

                    Correct Answer: 1<\/b><\/p>\n

                    Explanation<\/b><\/p>\n

                    A Media Resource Group List, or MRGL, provides an ordered collection of Media Resource Groups that can be assigned to devices and other supported configuration objects. The associated MRGs can contain resources such as conference bridges, transcoders, and music-on-hold resources. The MRGL helps CUCM determine which media resources are available to a device. Route patterns handle digit matching, calling search spaces control dial-plan access, and partitions organize destinations. If a phone cannot access a conference bridge or transcoder, administrators should verify the phone’s assigned MRGL, the MRG contents, and the operational status of the underlying media resources.<\/span><\/p>\n

                    Question 110<\/b><\/h3>\n

                    Which protocol is used to resolve hostnames and service records for Cisco collaboration components?<\/b><\/p>\n

                      \n
                    1. DHCP<\/span><\/li>\n
                    2. DNS<\/span><\/li>\n
                    3. RTP<\/span><\/li>\n
                    4. SRTP<\/span><\/li>\n<\/ol>\n

                      Correct Answer: 2<\/b><\/p>\n

                      Explanation<\/b><\/p>\n

                      Domain Name System, or DNS, resolves hostnames into IP addresses and can provide service records used by collaboration applications. Cisco collaboration deployments frequently depend on accurate DNS configuration for server discovery, certificates, Expressway services, and other components. DHCP can provide IP addressing information, but it does not perform hostname resolution itself. RTP and SRTP are media transport protocols. When collaboration services fail to discover servers or endpoints, administrators should verify forward and reverse DNS records, SRV records where applicable, DNS reachability, and hostname consistency with certificates. Correct DNS configuration is especially important in distributed and secure collaboration environments.<\/span><\/p>\n

                      Question 111<\/b><\/h3>\n

                      Which codec is commonly associated with wideband voice quality in Cisco collaboration environments?<\/b><\/p>\n

                        \n
                      1. G.729<\/span><\/li>\n
                      2. G.711<\/span><\/li>\n
                      3. G.722<\/span><\/li>\n
                      4. iLBC<\/span><\/li>\n<\/ol>\n

                        Correct Answer: 3<\/b><\/p>\n

                        Explanation<\/b><\/p>\n

                        G.722 is a wideband audio codec commonly used to provide higher-quality voice than traditional narrowband codecs. It can improve speech clarity when supported by both endpoints and the network path. G.711 provides high-quality narrowband audio but generally uses more bandwidth than compressed codecs, while G.729 provides compressed speech with lower bandwidth requirements. Codec selection can be influenced by endpoint capabilities, region configuration, trunks, and media resources. When troubleshooting unexpected codec selection, administrators should inspect the configured region relationships and negotiated media parameters. Wideband operation also requires that the entire call path support the selected codec and its associated capabilities.<\/span><\/p>\n

                        Question 112<\/b><\/h3>\n

                        Which Expressway component is normally positioned at the enterprise network edge for Mobile and Remote Access?<\/b><\/p>\n

                          \n
                        1. Expressway-C<\/span><\/li>\n
                        2. Expressway-E<\/span><\/li>\n
                        3. CUCM Publisher<\/span><\/li>\n
                        4. Unity Connection<\/span><\/li>\n<\/ol>\n

                          Correct Answer: 2<\/b><\/p>\n

                          Explanation<\/b><\/p>\n

                          Expressway-E is normally positioned at the external network edge in a Cisco Mobile and Remote Access deployment. It communicates with Expressway-C through a secure traversal relationship and helps provide remote access to internal collaboration services without directly exposing CUCM to the public network. Expressway-C is located closer to the internal collaboration environment, while CUCM provides call processing and Unity Connection provides voicemail. Administrators troubleshooting MRA should verify DNS discovery, certificates, traversal zones, search rules, authentication, firewall policies, and CUCM configuration. Correct separation between the edge and internal components is important for both security and reliable remote collaboration access.<\/span><\/p>\n

                          Question 113<\/b><\/h3>\n

                          Which CUCM object is used to organize gateways so they can be selected by a route list?<\/b><\/p>\n

                            \n
                          1. Route Group<\/span><\/li>\n
                          2. Region<\/span><\/li>\n
                          3. Device Pool<\/span><\/li>\n
                          4. Partition<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 1<\/b><\/p>\n

                            Explanation<\/b><\/p>\n

                            A Route Group organizes one or more gateways so they can be referenced by a route list. This provides a structured approach to gateway selection and supports redundancy by allowing multiple gateways to belong to the same logical group. Route lists can contain multiple route groups and establish an ordered routing strategy. Regions control codec and bandwidth relationships, device pools provide shared endpoint settings, and partitions control dial-plan visibility. When configuring outbound PSTN routing, administrators should verify that the correct gateways are members of the route group and that the route list and route pattern point to the intended routing structure.<\/span><\/p>\n

                            Question 114<\/b><\/h3>\n

                            Which SIP response code generally indicates that the requested operation was successful?<\/b><\/p>\n

                              \n
                            1. 404<\/span><\/li>\n
                            2. 486<\/span><\/li>\n
                            3. 503<\/span><\/li>\n
                            4. 200<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 4<\/b><\/p>\n

                              Explanation<\/b><\/p>\n

                              SIP response code 200 indicates successful completion of a requested operation. A common example is a 200 OK response to an INVITE after the called party accepts the session. A 404 response indicates that a requested resource was not found, 486 commonly indicates that the destination is busy, and 503 indicates service unavailability. Understanding SIP response codes helps administrators determine whether a call failed because of addressing, availability, service problems, or other signaling conditions. When troubleshooting SIP calls, administrators should examine the complete message sequence rather than focusing on one response alone, because earlier signaling messages can reveal the actual cause of failure.<\/span><\/p>\n

                              Question 115<\/b><\/h3>\n

                              Which CUCM feature is primarily used to provide different codec choices and bandwidth limits between locations?<\/b><\/p>\n

                                \n
                              1. Region<\/span><\/li>\n
                              2. Partition<\/span><\/li>\n
                              3. Route List<\/span><\/li>\n
                              4. CSS<\/span><\/li>\n<\/ol>\n

                                Correct Answer: 1<\/b><\/p>\n

                                Explanation<\/b><\/p>\n

                                Regions in CUCM define codec preferences and bandwidth relationships between groups of devices. They are useful when collaboration endpoints communicate across WAN connections where bandwidth must be managed carefully. Administrators can specify how devices in one region communicate with devices in another region and influence the codecs selected for those calls. Partitions and CSSs control dial-plan access, while route lists control gateway routing. If WAN calls are consuming excessive bandwidth or using an unexpected codec, administrators should inspect region assignments and region-to-region settings. Proper configuration helps maintain voice and video quality while controlling the amount of bandwidth consumed by collaboration traffic.<\/span><\/p>\n

                                Question 116<\/b><\/h3>\n

                                Which protocol is commonly used to transport audio and video packets after SIP signaling establishes a session?<\/b><\/p>\n

                                  \n
                                1. LDAP<\/span><\/li>\n
                                2. RTP<\/span><\/li>\n
                                3. DNS<\/span><\/li>\n
                                4. NTP<\/span><\/li>\n<\/ol>\n

                                  Correct Answer: 2<\/b><\/p>\n

                                  Explanation<\/b><\/p>\n

                                  Real-time Transport Protocol, or RTP, transports real-time audio and video packets after signaling establishes the session. RTP includes sequence numbers and timestamps that help receivers reconstruct media streams correctly. SIP is responsible for session signaling, while LDAP provides directory services, DNS resolves names, and NTP provides time synchronization. Because RTP traffic is sensitive to packet loss, delay, and jitter, collaboration networks should implement appropriate QoS policies. If signaling succeeds but users experience missing or one-way audio, administrators should investigate the RTP path, firewall rules, NAT behavior, routing, codec negotiation, and endpoint media configuration independently of the SIP signaling path.<\/span><\/p>\n

                                  Question 117<\/b><\/h3>\n

                                  Which Cisco collaboration component can normalize SIP signaling between systems that use different SIP implementations?<\/b><\/p>\n

                                    \n
                                  1. Cisco TFTP<\/span><\/li>\n
                                  2. Cisco Unity Connection<\/span><\/li>\n
                                  3. Cisco Unified Border Element<\/span><\/li>\n
                                  4. Cisco IM and Presence<\/span><\/li>\n<\/ol>\n

                                    Correct Answer: 3<\/b><\/p>\n

                                    Explanation<\/b><\/p>\n

                                    Cisco Unified Border Element, or CUBE, can perform SIP normalization and interoperability functions between different SIP systems. Service providers and enterprise systems may use different SIP headers, number formats, codecs, or signaling expectations. CUBE can help normalize these differences while also providing functions such as topology hiding, security controls, and controlled connectivity. TFTP handles configuration distribution, Unity Connection provides voicemail, and IM and Presence supports messaging and presence. When interoperability problems occur between CUCM and an external SIP provider, administrators can inspect CUBE dial peers, SIP profiles, codec configuration, digit manipulation, and SIP messages to identify incompatible parameters.<\/span><\/p>\n

                                    Question 118<\/b><\/h3>\n

                                    Which CUCM feature can provide backup call-processing functionality at a branch when WAN connectivity to the main CUCM environment fails?<\/b><\/p>\n

                                      \n
                                    1. SRST<\/span><\/li>\n
                                    2. LDAP<\/span><\/li>\n
                                    3. CDP<\/span><\/li>\n
                                    4. DNS<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 1<\/b><\/p>\n

                                      Explanation<\/b><\/p>\n

                                      Survivable Remote Site Telephony, or SRST, provides backup call-processing functionality for supported Cisco IP phones when connectivity to centralized CUCM servers is lost. SRST is commonly configured on a Cisco router located at a branch or remote site. During normal operation, phones register with CUCM, while SRST provides a fallback mechanism during WAN or centralized-service outages. LDAP provides directory integration, CDP provides neighbor discovery, and DNS provides name resolution. Administrators should configure the SRST reference appropriately in CUCM and verify the router’s SRST configuration. Testing failover is important to confirm that endpoints transition successfully and that essential calling functions remain available.<\/span><\/p>\n

                                      Question 119<\/b><\/h3>\n

                                      Which service is responsible for automatically providing IP addressing information to Cisco IP phones?<\/b><\/p>\n

                                        \n
                                      1. NTP<\/span><\/li>\n
                                      2. DHCP<\/span><\/li>\n
                                      3. SIP<\/span><\/li>\n
                                      4. RTP<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 2<\/b><\/p>\n

                                        Explanation<\/b><\/p>\n

                                        Dynamic Host Configuration Protocol, or DHCP, automatically provides IP addressing information to Cisco IP phones and other network devices. Depending on the configuration, DHCP can provide an IP address, subnet mask, default gateway, DNS servers, and additional vendor-specific information used during endpoint provisioning. NTP synchronizes time, SIP provides signaling, and RTP carries media. Cisco IP phones can use DHCP options to discover the appropriate TFTP server and obtain configuration information. When a phone fails to acquire an address or download its configuration, administrators should verify the DHCP scope, VLAN, relay configuration, DHCP options, network connectivity, and TFTP reachability.<\/span><\/p>\n

                                        Question 120<\/b><\/h3>\n

                                        Which security technology is primarily responsible for protecting SIP signaling rather than the media stream?<\/b><\/p>\n

                                          \n
                                        1. SRTP<\/span><\/li>\n
                                        2. RTP<\/span><\/li>\n
                                        3. TLS<\/span><\/li>\n
                                        4. DHCP<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 3<\/b><\/p>\n

                                          Explanation<\/b><\/p>\n

                                          Transport Layer Security, or TLS, is commonly used to secure SIP signaling between supported collaboration systems. It encrypts signaling traffic and helps provide authentication through certificates and trusted relationships. SRTP is used to protect real-time media, RTP transports unencrypted media, and DHCP provides network configuration. Secure SIP deployments require correctly configured certificates, security profiles, trust stores, and synchronized system time. When a secure SIP connection fails, administrators should inspect certificate validity, hostname matching, trust relationships, and TLS configuration. Separating signaling security from media security is important because a call can have secure signaling while media protection is configured independently through SRTP.<\/span><\/p>\n

                                           <\/p>\n","protected":false},"excerpt":{"rendered":"

                                          View Full Cisco CCNP Collaboration 350-801 Exam Dumps and Practice Test Dumps.   Question 101 Which CUCM configuration allows administrators to control access to specific dial-plan destinations? Region Calling Search Space Device Pool Route Group Correct Answer: 2 Explanation Calling Search Spaces are used in Cisco Unified Communications Manager to control which partitions a device, […]<\/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\/21266"}],"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=21266"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21266\/revisions"}],"predecessor-version":[{"id":21267,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21266\/revisions\/21267"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=21266"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=21266"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=21266"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}