{"id":21286,"date":"2026-09-24T11:51:37","date_gmt":"2026-09-24T11:51:37","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=21286"},"modified":"2026-09-24T11:51:37","modified_gmt":"2026-09-24T11:51:37","slug":"cisco-ccnp-collaboration-350-801-practice-test-questions-and-exam-dumps-part16-q301-320","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-350-801-practice-test-questions-and-exam-dumps-part16-q301-320\/","title":{"rendered":"Cisco CCNP Collaboration 350-801 Practice Test Questions and Exam Dumps Part16 Q301-320"},"content":{"rendered":"

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

 <\/p>\n

Question 301<\/b><\/h3>\n

Which SIP response code indicates that a request was successfully processed?<\/b><\/p>\n

    \n
  1. 404 Not Found<\/span><\/li>\n
  2. 503 Service Unavailable<\/span><\/li>\n
  3. 486 Busy Here<\/span><\/li>\n
  4. 200 OK<\/span><\/li>\n<\/ol>\n

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

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

    The SIP 200 OK response indicates that a request was successfully received, understood, and accepted. In a normal call setup, a 200 OK response to an INVITE indicates that the called endpoint has accepted the session and provides session information, usually through SDP. The caller then sends an ACK to confirm receipt of the final response. Other SIP response codes indicate different conditions, such as 404 for an unavailable destination, 486 when the destination is busy, and 503 when a service is temporarily unavailable. Understanding SIP response codes is important when troubleshooting call setup and signaling problems in Cisco collaboration environments.<\/span><\/p>\n

    Question 302<\/b><\/h3>\n

    Which CUCM configuration determines which devices can access a particular route pattern?<\/b><\/p>\n

      \n
    1. Calling Search Space<\/span><\/li>\n
    2. Region<\/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 Calling Search Space determines which partitions a calling device or directory number can access. CUCM uses the combination of partitions and Calling Search Spaces to control which destinations are reachable. When a user places a call, CUCM searches the partitions available through the caller’s CSS and evaluates matching patterns. This provides an important mechanism for controlling local, national, international, emergency, and restricted dialing. Regions are primarily used for codec and bandwidth control, while device pools provide common device configuration and location-related settings. Media Resource Groups control access to media resources such as transcoders and conference bridges rather than determining route-pattern accessibility.<\/span><\/p>\n

      Question 303<\/b><\/h3>\n

      What is the primary purpose of a CUCM translation pattern?<\/b><\/p>\n

        \n
      1. To provide NTP synchronization<\/span><\/li>\n
      2. To transform dialed digits before call routing<\/span><\/li>\n
      3. To assign an IP address to a phone<\/span><\/li>\n
      4. To encrypt RTP packets<\/span><\/li>\n<\/ol>\n

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

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

        A translation pattern is used in CUCM to manipulate dialed digits before the call is routed toward its destination. It can add, remove, or modify digits and can also influence the calling or called party information depending on configuration. Translation patterns are useful when integrating different numbering plans, normalizing numbers, or adapting internal dialing to external service-provider requirements. Unlike route patterns, which primarily select a route for a call, translation patterns modify the number and then allow CUCM to continue call processing. They are therefore an important component of CUCM dial-plan design and digit manipulation.<\/span><\/p>\n

        Question 304<\/b><\/h3>\n

        Which protocol is commonly used to synchronize time across Cisco collaboration servers and endpoints?<\/b><\/p>\n

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

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

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

          Network Time Protocol, or NTP, is used to synchronize clocks across network devices and servers. Accurate time is particularly important in Cisco collaboration environments because certificates, security mechanisms, logs, call records, and troubleshooting activities can depend on synchronized timestamps. CUCM, Unity Connection, IM and Presence, and other collaboration components can use NTP to maintain consistent system time. Without proper synchronization, authentication and certificate-related operations may experience problems, and troubleshooting logs from different servers can become difficult to correlate. TFTP is primarily used for configuration and firmware delivery, while LDAP provides directory services and RTP carries media.<\/span><\/p>\n

          Question 305<\/b><\/h3>\n

          What is the main function of a CUCM route group?<\/b><\/p>\n

            \n
          1. To synchronize database information<\/span><\/li>\n
          2. To provide directory authentication<\/span><\/li>\n
          3. To encrypt signaling<\/span><\/li>\n
          4. To contain gateways or trunks that can be selected for routing<\/span><\/li>\n<\/ol>\n

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

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

            A CUCM route group contains a prioritized list of gateways or trunks that can be used to route calls. When CUCM selects a route list, the route list references one or more route groups. The route group then determines the specific gateway or trunk that should be attempted based on its configured order and availability. This architecture provides flexibility and redundancy in call routing. If one gateway is unavailable, CUCM can attempt another member of the route group. Route groups therefore form an important part of CUCM dial-plan architecture and help distribute calls across available external connectivity resources.<\/span><\/p>\n

            Question 306<\/b><\/h3>\n

            Which protocol carries real-time audio and video media after a SIP session is established?<\/b><\/p>\n

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

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

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

              Real-time Transport Protocol, or RTP, carries audio and video media for many IP-based collaboration sessions. SIP is generally responsible for signaling and establishing, modifying, or terminating the session, while RTP transports the actual media streams. RTP packets contain sequence numbers and timestamps that help receivers handle packet ordering and timing. RTCP can also provide information about media quality and transmission statistics. In Cisco collaboration deployments, understanding the separation between signaling and media is essential for troubleshooting. A call can have successful SIP signaling while still experiencing RTP-related problems such as packet loss, jitter, latency, or one-way audio.<\/span><\/p>\n

              Question 307<\/b><\/h3>\n

              Which component provides voicemail services in a traditional Cisco collaboration deployment?<\/b><\/p>\n

                \n
              1. Cisco CUBE<\/span><\/li>\n
              2. Cisco Unity Connection<\/span><\/li>\n
              3. Cisco Expressway-E<\/span><\/li>\n
              4. Cisco IM and Presence<\/span><\/li>\n<\/ol>\n

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

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

                Cisco Unity Connection provides voicemail and unified messaging capabilities in Cisco collaboration environments. It can integrate with CUCM so that unanswered or busy calls are redirected to appropriate voicemail boxes. Unity Connection also provides features such as voicemail access, greetings, message management, and automated attendant functionality. CUCM handles core call processing and signaling, but voicemail functionality is provided by Unity Connection when that architecture is deployed. CUBE is primarily used for voice interoperability and external SIP connectivity, Expressway-E supports edge and remote-access functions, and IM and Presence provides instant messaging and presence services.<\/span><\/p>\n

                Question 308<\/b><\/h3>\n

                Which technology allows remote Cisco collaboration clients to securely connect without requiring a traditional VPN?<\/b><\/p>\n

                  \n
                1. SRST<\/span><\/li>\n
                2. CUBE<\/span><\/li>\n
                3. Expressway Mobile and Remote Access<\/span><\/li>\n
                4. DHCP<\/span><\/li>\n<\/ol>\n

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

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

                  Cisco Expressway Mobile and Remote Access, commonly called MRA, allows supported collaboration clients to connect securely from outside the corporate network without requiring a traditional VPN connection. The deployment uses Cisco Expressway-C inside the organization and Expressway-E at the network edge. Appropriate DNS, certificates, authentication, and traversal configuration are required for successful operation. MRA can provide services such as voice, video, messaging, and presence depending on the client and deployment. SRST is intended to provide limited call-processing functionality during WAN outages, while CUBE focuses on voice and SIP interoperability rather than remote client access.<\/span><\/p>\n

                  Question 309<\/b><\/h3>\n

                  Which CUCM component groups devices that share common configuration characteristics such as date\/time settings and call-processing servers?<\/b><\/p>\n

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

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

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

                    A CUCM device pool provides a collection of common configuration settings that can be applied to multiple devices. Device pools can define items such as CUCM groups, regions, date\/time groups, SRST references, and other operational parameters. Assigning devices to an appropriate device pool simplifies administration and ensures consistent configuration across groups of phones or other endpoints. A route group is concerned with gateways and trunks used for call routing, while a partition controls directory-number and pattern visibility. Translation patterns manipulate dialed digits. Device pools therefore provide an efficient way to standardize configuration across endpoints located in similar environments.<\/span><\/p>\n

                    Question 310<\/b><\/h3>\n

                    Which SIP message is normally used to begin establishing a new call session?<\/b><\/p>\n

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

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

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

                      The SIP INVITE request is used to initiate a session, such as a voice or video call. It contains information about the proposed session and commonly includes an SDP body describing media capabilities, codecs, IP addresses, and ports. The receiving endpoint processes the INVITE and responds with provisional or final SIP responses. If the call is accepted, a 200 OK response is returned and the originating endpoint sends an ACK. BYE terminates an established session, REGISTER associates an endpoint with a SIP service, and OPTIONS can be used to query capabilities. Understanding SIP message roles is essential for collaboration troubleshooting.<\/span><\/p>\n

                      Question 311<\/b><\/h3>\n

                      What is the primary purpose of SRTP in a Cisco collaboration environment?<\/b><\/p>\n

                        \n
                      1. To assign phone IP addresses<\/span><\/li>\n
                      2. To secure real-time media streams<\/span><\/li>\n
                      3. To provide DNS resolution<\/span><\/li>\n
                      4. To synchronize system clocks<\/span><\/li>\n<\/ol>\n

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

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

                        Secure Real-time Transport Protocol, or SRTP, provides protection for RTP media streams. It can provide confidentiality through encryption and integrity protection so that media cannot be easily intercepted or modified during transmission. SRTP is commonly used when secure voice or video communications are required. It is different from TLS, which is generally used to protect signaling or other application-layer communications. DHCP is used for IP addressing and related network parameters, DNS resolves names, and NTP provides time synchronization. When troubleshooting secure collaboration calls, engineers should verify that endpoints and network components support compatible security mechanisms and that required certificates or keying methods are properly configured.<\/span><\/p>\n

                        Question 312<\/b><\/h3>\n

                        Which CUCM object can contain multiple route groups and determine the order in which those groups are attempted?<\/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 is a CUCM dial-plan object that contains one or more route groups and determines the order in which those route groups are selected for call routing. This structure provides flexibility and redundancy because multiple gateways or trunks can be organized behind route groups and route lists. When a route pattern points to a route list, CUCM evaluates the route list and selects an available route group according to its configuration. Partitions control pattern visibility, device pools provide common endpoint settings, and media resource groups organize resources such as conference bridges and transcoders. Route lists are therefore central to scalable CUCM routing designs.<\/span><\/p>\n

                          Question 313<\/b><\/h3>\n

                          Which protocol is commonly used by CUCM to communicate with a corporate directory service for user information?<\/b><\/p>\n

                            \n
                          1. RTP<\/span><\/li>\n
                          2. LDAP<\/span><\/li>\n
                          3. SRTP<\/span><\/li>\n
                          4. SIP<\/span><\/li>\n<\/ol>\n

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

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

                            LDAP, or Lightweight Directory Access Protocol, is commonly used to integrate CUCM with external directory services. Directory integration can allow CUCM to synchronize user information from systems such as Microsoft Active Directory. This can simplify user administration and provide centralized identity information for collaboration applications. Depending on the deployment, directory synchronization can include usernames, names, telephone numbers, and other attributes. LDAP is not a media or call-signaling protocol. RTP and SRTP transport media, while SIP provides session signaling. Proper LDAP configuration requires connectivity, authentication details, search bases, and appropriate directory attributes to be defined.<\/span><\/p>\n

                            Question 314<\/b><\/h3>\n

                            Which device is commonly deployed at the enterprise edge to provide SIP interoperability and voice-network demarcation?<\/b><\/p>\n

                              \n
                            1. CUCM Group<\/span><\/li>\n
                            2. CUBE<\/span><\/li>\n
                            3. Unity Connection<\/span><\/li>\n
                            4. IM and Presence<\/span><\/li>\n<\/ol>\n

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

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

                              Cisco Unified Border Element, or CUBE, is commonly deployed at the boundary between an enterprise voice network and an external SIP service provider or another voice network. CUBE can provide SIP normalization, interoperability, security controls, topology hiding, and media handling. It commonly uses dial peers to determine how calls are processed and routed. CUCM remains responsible for centralized call processing within the collaboration environment, while Unity Connection provides voicemail and IM and Presence provides messaging and presence services. CUBE is especially useful when enterprise SIP behavior must be adapted to meet service-provider requirements or when network boundaries need controlled voice connectivity.<\/span><\/p>\n

                              Question 315<\/b><\/h3>\n

                              What is the primary purpose of a CUCM partition?<\/b><\/p>\n

                                \n
                              1. To organize dialable numbers and patterns into groups for access control<\/span><\/li>\n
                              2. To provide media encryption<\/span><\/li>\n
                              3. To synchronize clocks<\/span><\/li>\n
                              4. To transport RTP packets<\/span><\/li>\n<\/ol>\n

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

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

                                A CUCM partition groups directory numbers, route patterns, translation patterns, and other dial-plan objects into logical collections. Calling Search Spaces then determine which partitions a device or user can access. This combination provides an important mechanism for controlling dialing permissions. For example, an organization can place international dialing patterns in a restricted partition and allow access only to authorized users. Partitions do not encrypt media, synchronize clocks, or carry RTP. They are logical dial-plan constructs used to control pattern visibility and routing. Proper partition and CSS design is essential when implementing scalable and secure CUCM dialing policies.<\/span><\/p>\n

                                Question 316<\/b><\/h3>\n

                                Which SIP response indicates that the called endpoint is currently busy and cannot accept the call?<\/b><\/p>\n

                                  \n
                                1. 200 OK<\/span><\/li>\n
                                2. 404 Not Found<\/span><\/li>\n
                                3. 486 Busy Here<\/span><\/li>\n
                                4. 503 Service Unavailable<\/span><\/li>\n<\/ol>\n

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

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

                                  SIP response code 486 Busy Here indicates that the destination endpoint is currently busy and cannot accept the incoming call in its present state. The response is a client error response generated by the endpoint or server handling the request. CUCM or another call-control system may then apply configured call-routing behavior, such as forwarding the call to voicemail or another destination. A 404 response indicates that the requested destination cannot be found, while 503 indicates that a service is temporarily unavailable. A 200 OK indicates successful processing. Understanding these response codes helps identify where call establishment is failing.<\/span><\/p>\n

                                  Question 317<\/b><\/h3>\n

                                  Which Cisco collaboration component can provide media conversion when endpoints require incompatible codecs?<\/b><\/p>\n

                                    \n
                                  1. DHCP server<\/span><\/li>\n
                                  2. Transcoder<\/span><\/li>\n
                                  3. DNS server<\/span><\/li>\n
                                  4. TFTP server<\/span><\/li>\n<\/ol>\n

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

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

                                    A transcoder provides media conversion between codecs when two endpoints or network components cannot directly communicate using the same codec. For example, one side of a call may use G.711 while another side requires G.729. CUCM can invoke an available transcoder when codec compatibility or other media requirements make direct communication impossible. Transcoding consumes processing resources, so capacity should be considered when designing the collaboration environment. DHCP supplies network configuration, DNS resolves names, and TFTP can provide configuration files and firmware. Media resources such as transcoders are typically organized and accessed through CUCM media resource configuration.<\/span><\/p>\n

                                    Question 318<\/b><\/h3>\n

                                    Which CUCM configuration controls the maximum bandwidth and preferred codecs between groups of devices?<\/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

                                      CUCM regions are used to control codec selection and bandwidth usage between groups of devices. Administrators can define relationships between regions and specify preferred codecs for calls between them. This is particularly useful when endpoints are separated by WAN links with limited bandwidth. For example, a deployment may use a high-bandwidth codec within a local region while using a more bandwidth-efficient codec across a WAN. Partitions and CSSs control dial-plan access, while route lists determine routing through route groups. Region configuration therefore plays an important role in managing voice and video bandwidth within distributed Cisco collaboration deployments.<\/span><\/p>\n

                                      Question 319<\/b><\/h3>\n

                                      Which component can provide limited call-processing functionality for Cisco IP phones when communication with CUCM is lost?<\/b><\/p>\n

                                        \n
                                      1. Expressway-E<\/span><\/li>\n
                                      2. CUBE<\/span><\/li>\n
                                      3. SRST<\/span><\/li>\n
                                      4. Unity Connection<\/span><\/li>\n<\/ol>\n

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

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

                                        Survivable Remote Site Telephony, or SRST, provides limited call-processing functionality at a remote site when IP phones lose connectivity with their primary CUCM servers. An SRST-capable router can allow supported phones to register locally and maintain basic calling services during a WAN outage. The goal is to preserve essential voice communication until normal connectivity is restored. SRST is not intended to replace the full feature set of CUCM. Expressway-E is primarily an edge collaboration component, CUBE handles voice interoperability and border functions, and Unity Connection provides voicemail. Proper SRST configuration requires appropriate references and network connectivity at the remote site.<\/span><\/p>\n

                                        Question 320<\/b><\/h3>\n

                                        Which protocol is commonly used to securely protect SIP signaling between collaboration components?<\/b><\/p>\n

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

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

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

                                          Transport Layer Security, or TLS, can be used to protect SIP signaling by providing encryption and integrity for signaling messages between supported collaboration components. Secure SIP deployments often use TLS certificates to authenticate endpoints or servers and establish an encrypted signaling channel. This protects call-control information from unauthorized observation or modification while it travels across the network. TLS protects signaling rather than carrying the actual audio or video media. RTP carries media, DHCP provides network configuration, and LDAP provides directory access. When implementing secure collaboration, administrators must ensure certificates, trust relationships, supported TLS versions, and endpoint configuration are compatible.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 350-801 Exam Dumps and Practice Test Dumps.   Question 301 Which SIP response code indicates that a request was successfully processed? 404 Not Found 503 Service Unavailable 486 Busy Here 200 OK Correct Answer: 4 Explanation The SIP 200 OK response indicates that a request was successfully received, understood, and […]<\/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\/21286"}],"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=21286"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21286\/revisions"}],"predecessor-version":[{"id":21287,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21286\/revisions\/21287"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=21286"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=21286"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=21286"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}