{"id":23423,"date":"2026-09-28T06:25:36","date_gmt":"2026-09-28T06:25:36","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23423"},"modified":"2026-09-28T06:25:36","modified_gmt":"2026-09-28T06:25:36","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-11-q201-220","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-11-q201-220\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 11 Q201-220"},"content":{"rendered":"

View Full <\/b>Cisco CCNP Collaboration 300-815 Exam Dumps<\/b><\/a> and Practice Test Dumps<\/b><\/h1>\n

 <\/p>\n

Question 201. A Cisco Unified Communications Manager administrator needs calls to a specific PSTN destination to use a particular gateway group without changing the routing of other destinations. Which configuration should be used?<\/b><\/p>\n

    \n
  1. <\/b> Device Mobility<\/span><\/li>\n
  2. <\/b> Region configuration<\/span><\/li>\n
  3. <\/b> Location configuration<\/span><\/li>\n
  4. <\/b> Route pattern associated with the appropriate route list<\/span><\/li>\n<\/ol>\n

    Correct Answer: 4. Route pattern associated with the appropriate route list<\/b><\/p>\n

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

    A route pattern defines how a dialed destination is routed and can be associated with a route list that identifies the route groups and gateways used for the call. By creating a destination-specific route pattern and associating it with the appropriate route list, an administrator can direct selected PSTN calls through a particular gateway path without changing unrelated destinations. Device Mobility changes device settings according to location, while regions and locations primarily address media and bandwidth behavior. Therefore, the combination of a suitable route pattern and route list provides the required destination-specific gateway routing.<\/span><\/p>\n

    Question 202. A SIP trunk requires a different SIP header format than the default generated by Cisco Unified Communications Manager. Which feature can modify SIP headers to meet the interoperability requirement?<\/b><\/p>\n

      \n
    1. <\/b> Location CAC<\/span><\/li>\n
    2. <\/b> SIP profile<\/span><\/li>\n
    3. <\/b> Route group<\/span><\/li>\n
    4. <\/b> Device pool<\/span><\/li>\n<\/ol>\n

      Correct Answer: 2. SIP profile<\/b><\/p>\n

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

      A SIP profile can modify SIP signaling to support interoperability requirements between Cisco Unified Communications Manager and external SIP systems. Depending on the supported configuration, SIP profiles can manipulate specific SIP headers or message elements so that signaling conforms to the expectations of the remote system. Location CAC controls bandwidth admission, route groups determine gateway or trunk selection, and device pools provide common device configuration. When the requirement specifically involves changing the structure or content of SIP signaling headers, a SIP profile is the appropriate feature to investigate.<\/span><\/p>\n

      Question 203. A branch phone moves from its normal office to another location where different calling privileges and codec requirements apply. Which Cisco Unified Communications Manager feature can automatically apply location-specific settings to the phone?<\/b><\/p>\n

        \n
      1. <\/b> Device Mobility<\/span><\/li>\n
      2. <\/b> Route group<\/span><\/li>\n
      3. <\/b> Translation pattern<\/span><\/li>\n
      4. <\/b> SIP session timer<\/span><\/li>\n<\/ol>\n

        Correct Answer: 1. Device Mobility<\/b><\/p>\n

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

        Device Mobility allows Cisco Unified Communications Manager to apply device-specific settings based on the physical or configured location where a phone is operating. This can affect settings such as calling search spaces, regions, locations, and other parameters required for the phone’s current environment. Route groups handle gateway or trunk selection, translation patterns manipulate digits, and session timers govern SIP dialog refresh behavior. When a phone moves between locations and needs different communication settings without manually reconfiguring the device each time, Device Mobility provides the appropriate mechanism.<\/span><\/p>\n

        Question 204. Two sites use different codec policies, and calls between them must use a compressed codec because of limited WAN bandwidth. Which Cisco Unified Communications Manager configuration determines the codec relationship between the sites?<\/b><\/p>\n

          \n
        1. <\/b> Calling Search Space<\/span><\/li>\n
        2. <\/b> Route list<\/span><\/li>\n
        3. <\/b> Translation pattern<\/span><\/li>\n
        4. <\/b> Region relationship<\/span><\/li>\n<\/ol>\n

          Correct Answer: 3. Region relationship<\/b><\/p>\n

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

          Cisco Unified Communications Manager regions define codec relationships between devices in different regions. Administrators can configure the preferred codec behavior for calls within a region and between different regions, allowing WAN calls to use a codec appropriate for the available bandwidth. Calling Search Spaces control which partitions a device can access, route lists determine gateway or trunk selection, and translation patterns manipulate digits. Therefore, when the requirement concerns codec selection between two sites with different bandwidth characteristics, the relevant configuration is the region relationship.<\/span><\/p>\n

          Question 205. A company wants to prevent users in one department from accessing international route patterns while still allowing them to make local and national calls. Which configuration provides this dial-plan restriction?<\/b><\/p>\n

            \n
          1. <\/b> Device pool and region<\/span><\/li>\n
          2. <\/b> Partition and Calling Search Space<\/span><\/li>\n
          3. <\/b> Route group and route list<\/span><\/li>\n
          4. <\/b> SIP profile and normalization<\/span><\/li>\n<\/ol>\n

            Correct Answer: 2. Partition and Calling Search Space<\/b><\/p>\n

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

            Partitions and Calling Search Spaces provide logical access control within the Cisco Unified Communications Manager dial plan. International route patterns can be placed in a dedicated partition, while users in the restricted department receive a CSS that does not include that partition. Local and national route patterns can remain in partitions accessible to the users. Route groups and route lists determine gateway selection after routing is permitted, while SIP profiles modify signaling. This partition-and-CSS design therefore provides granular control over which classes of destinations a group of users can reach.<\/span><\/p>\n

            Question 206. A SIP call is being routed through CUBE, but the outbound provider requires the enterprise’s internal extension to be converted into a complete E.164 number. Which configuration should be applied to the appropriate CUBE dial peer?<\/b><\/p>\n

              \n
            1. <\/b> Region configuration<\/span><\/li>\n
            2. <\/b> SIP session timer<\/span><\/li>\n
            3. <\/b> Device Mobility<\/span><\/li>\n
            4. <\/b> Translation profile using a translation rule<\/span><\/li>\n<\/ol>\n

              Correct Answer: 4. Translation profile using a translation rule<\/b><\/p>\n

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

              Cisco IOS translation rules can transform telephone digits, and translation profiles associate those rules with the appropriate call-processing direction or dial peer. For an outbound CUBE call, a translation profile can convert an internal extension or other enterprise-specific number format into the complete E.164 calling number required by the provider. Region configuration and Device Mobility do not perform digit manipulation, while SIP session timers control dialog refresh behavior. The translation profile must be applied at the correct point in the call flow so the desired number is presented to the external provider.<\/span><\/p>\n

              Question 207. An administrator observes that a SIP endpoint sends 183 Session Progress containing SDP before the remote party answers. What does the SDP in this provisional response commonly enable?<\/b><\/p>\n

                \n
              1. <\/b> Early media<\/span><\/li>\n
              2. <\/b> SIP registration<\/span><\/li>\n
              3. <\/b> Call transfer<\/span><\/li>\n
              4. <\/b> Endpoint authentication<\/span><\/li>\n<\/ol>\n

                Correct Answer: 3. Call transfer<\/b><\/p>\n

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

                183 Session Progress is a provisional SIP response that can contain SDP describing media parameters. When SDP is provided in an appropriate early-media signaling flow, media can be established before the call is answered. This can allow network-provided announcements, tones, or other early audio to reach the caller. SIP registration is performed with REGISTER, call transfer commonly involves REFER or other transfer mechanisms, and authentication uses SIP authentication procedures rather than SDP itself. Therefore, when 183 contains SDP before answer, the relevant capability is early media rather than registration or transfer.<\/span><\/p>\n

                Question 208. A CUBE administrator needs to ensure that SIP signaling uses the correct physical interface toward the service provider while media uses the appropriate enterprise-facing interface. Which configuration area should be investigated?<\/b><\/p>\n

                  \n
                1. <\/b> Calling Search Space<\/span><\/li>\n
                2. <\/b> Route pattern<\/span><\/li>\n
                3. <\/b> Signaling and media interface bindings<\/span><\/li>\n
                4. <\/b> Device Mobility<\/span><\/li>\n<\/ol>\n

                  Correct Answer: 1. Signaling and media interface bindings<\/b><\/p>\n

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

                  CUBE can use signaling and media interface or address binding configuration to control which interfaces and addresses are associated with SIP signaling and RTP media. This is particularly important in deployments with multiple interfaces, VRFs, NAT boundaries, or separate enterprise and provider networks. Calling Search Spaces and route patterns determine Cisco Unified Communications Manager dial-plan behavior, while Device Mobility applies location-specific device settings. If signaling is leaving through the wrong interface or media is being advertised from an inappropriate address, the administrator should inspect the CUBE signaling and media binding configuration.<\/span><\/p>\n

                  Question 209. A SIP endpoint is receiving repeated 401 Unauthorized responses during registration. Which issue should the administrator investigate first?<\/b><\/p>\n

                    \n
                  1. <\/b> Region codec selection<\/span><\/li>\n
                  2. <\/b> Route group distribution<\/span><\/li>\n
                  3. <\/b> SIP digest authentication credentials and configuration<\/span><\/li>\n
                  4. <\/b> Location bandwidth<\/span><\/li>\n<\/ol>\n

                    Correct Answer: 3. SIP digest authentication credentials and configuration<\/b><\/p>\n

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

                    A 401 Unauthorized response in SIP commonly indicates that the server is requesting authentication, typically through SIP digest authentication. During registration, the endpoint should respond to the challenge with valid authentication information. Repeated 401 responses can therefore indicate incorrect credentials, an authentication mismatch, or an improperly configured authentication relationship. Region codec selection, route group distribution, and location bandwidth do not normally determine whether SIP registration authentication succeeds. The administrator should inspect the username, authentication credentials, challenge-response behavior, and related SIP authentication configuration.<\/span><\/p>\n

                    Question 210. A SIP endpoint sends an INVITE, but the user hangs up before receiving a final response. Which SIP method is normally used to terminate the pending INVITE transaction?<\/b><\/p>\n

                      \n
                    1. <\/b> BYE<\/span><\/li>\n
                    2. <\/b> CANCEL<\/span><\/li>\n
                    3. <\/b> REFER<\/span><\/li>\n
                    4. <\/b> REGISTER<\/span><\/li>\n<\/ol>\n

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

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

                      CANCEL is used to terminate a pending SIP request such as an INVITE when the session has not yet reached an established state. If the caller abandons the attempt before receiving a final response, the originating endpoint can send CANCEL to request termination of the pending transaction. BYE is used to terminate an established dialog, REFER is associated with referral operations such as call transfer, and REGISTER is used for endpoint registration. The distinction between CANCEL and BYE is important: CANCEL applies to an outstanding request, while BYE terminates an established session.<\/span><\/p>\n

                      Question 211. A service provider requires CUBE to reject SIP requests from sources that are not part of the provider’s known signaling addresses. Which security mechanism is most directly relevant?<\/b><\/p>\n

                        \n
                      1. <\/b> Region configuration<\/span><\/li>\n
                      2. <\/b> Translation pattern<\/span><\/li>\n
                      3. <\/b> Source-based access control and trusted-peer policy<\/span><\/li>\n
                      4. <\/b> Device Mobility<\/span><\/li>\n<\/ol>\n

                        Correct Answer: 4. Source-based access control and trusted-peer policy<\/b><\/p>\n

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

                        CUBE security can restrict which SIP sources are permitted to communicate with the enterprise gateway. Source-based access controls and trusted-peer policies help ensure that signaling is accepted only from expected provider addresses or other authorized peers. This is especially important for reducing unauthorized SIP traffic and toll-fraud exposure. Regions control codec relationships, translation patterns manipulate digits, and Device Mobility changes device settings based on location. When the requirement specifically concerns accepting SIP signaling only from approved source addresses, source-based access controls and trusted-peer configuration should be investigated.<\/span><\/p>\n

                        Question 212. A company uses a SIP URI such as user@example.com for dialing. CUBE must route calls differently depending on the domain portion of the URI. Which feature supports this requirement?<\/b><\/p>\n

                          \n
                        1. <\/b> SIP URI-based dial-peer matching<\/span><\/li>\n
                        2. <\/b> Location-based CAC<\/span><\/li>\n
                        3. <\/b> Device pool<\/span><\/li>\n
                        4. <\/b> Route group distribution<\/span><\/li>\n<\/ol>\n

                          Correct Answer: 1. SIP URI-based dial-peer matching<\/b><\/p>\n

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

                          SIP URI-based dial-peer matching allows CUBE to select an appropriate dial peer based on information contained in a SIP URI, including the destination domain. This is useful in environments where multiple providers, tenants, or organizations use different SIP domains and calls must follow domain-specific routing policies. Location-based CAC controls bandwidth admission, device pools provide common configuration, and route group distribution determines how gateways or trunks within a route group are selected. Therefore, URI-based dial-peer matching is the appropriate mechanism when the destination domain is a key routing criterion.<\/span><\/p>\n

                          Question 213. A Cisco Unified Communications Manager administrator wants a route list to try one route group first and another route group if the first becomes unavailable. Which configuration supports this behavior?<\/b><\/p>\n

                            \n
                          1. <\/b> Calling Search Space<\/span><\/li>\n
                          2. <\/b> Device pool<\/span><\/li>\n
                          3. <\/b> Ordered route group selection<\/span><\/li>\n
                          4. <\/b> SIP profile<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 2. Device pool<\/b><\/p>\n

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

                            The intended mechanism for primary and backup gateway selection is an ordered route-group configuration within a route list. An ordered route group selection allows the first available route-group member to be attempted before subsequent alternatives. A device pool does not determine route-group failover behavior; it provides common settings for associated devices. Calling Search Spaces control partition access, and SIP profiles manipulate SIP signaling. Therefore, when implementing primary and backup routing, the route list and its ordered route-group configuration should be examined rather than device-pool settings.<\/span><\/p>\n

                            Question 214. A SIP trunk uses a security policy requiring encrypted signaling and encrypted RTP media. Which combination provides these protections?<\/b><\/p>\n

                              \n
                            1. <\/b> UDP and RTP<\/span><\/li>\n
                            2. <\/b> SIP compression and DTMF relay<\/span><\/li>\n
                            3. <\/b> TLS and SRTP<\/span><\/li>\n
                            4. <\/b> OPTIONS and REGISTER<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 3. TLS and SRTP<\/b><\/p>\n

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

                              TLS is used to protect SIP signaling by providing encryption and integrity for the signaling connection, while Secure Real-time Transport Protocol (SRTP) protects voice or other real-time media. Using both provides protection for the two major components of a SIP call: signaling and media. OPTIONS and REGISTER are SIP methods rather than encryption mechanisms. DTMF relay controls how keypad digits are transported and does not provide general media encryption. Therefore, a deployment requiring encrypted SIP signaling together with encrypted RTP media should use TLS and SRTP where supported and correctly configured.<\/span><\/p>\n

                              Question 215. A user can dial an internal extension but receives a routing failure when dialing an external number. The external route pattern exists, but the user’s CSS does not contain the partition where the route pattern resides. What should be changed?<\/b><\/p>\n

                                \n
                              1. <\/b> Add the route-pattern partition to the user’s effective CSS<\/span><\/li>\n
                              2. <\/b> Change the phone’s region<\/span><\/li>\n
                              3. <\/b> Modify the SIP session timer<\/span><\/li>\n
                              4. <\/b> Change the route group’s codec<\/span><\/li>\n<\/ol>\n

                                Correct Answer: 1. Add the route-pattern partition to the user’s effective CSS<\/b><\/p>\n

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

                                A route pattern is usable only when the calling device or line has access to the partition containing that route pattern through its effective Calling Search Space. If the route pattern exists but the CSS cannot search its partition, Cisco Unified Communications Manager will not select that route pattern for the call. The administrator should therefore add the appropriate partition to the user’s effective CSS while maintaining the organization’s dial-plan access policy. Region settings, session timers, and codec selection do not determine whether the user can access a specific route-pattern partition.<\/span><\/p>\n

                                Question 216. A CUBE call reaches the provider, but the provider reports that the SIP Contact or From information does not match its expected format. Which feature can be used to manipulate the SIP message fields?<\/b><\/p>\n

                                  \n
                                1. <\/b> Location CAC<\/span><\/li>\n
                                2. <\/b> Device Mobility<\/span><\/li>\n
                                3. <\/b> Route group<\/span><\/li>\n
                                4. <\/b> SIP profile or normalization<\/span><\/li>\n<\/ol>\n

                                  Correct Answer: 4. SIP profile or normalization<\/b><\/p>\n

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

                                  SIP profiles and normalization mechanisms can modify supported SIP message fields and headers, making them useful for interoperability with service providers that require specific signaling formats. Headers such as From, Contact, P-Asserted-Identity, and related SIP information may require adjustment depending on the provider’s requirements and the supported CUBE configuration. Location CAC controls bandwidth admission, Device Mobility applies location-specific device settings, and route groups determine gateway or trunk selection. Therefore, when the provider reports an incorrect SIP header format, the administrator should investigate SIP profile or normalization configuration.<\/span><\/p>\n

                                  Question 217. A remote site has limited WAN bandwidth. Calls to another site should be admitted only when sufficient bandwidth remains for the call. Which Cisco Unified Communications Manager feature addresses this requirement?<\/b><\/p>\n

                                    \n
                                  1. <\/b> Location-based Call Admission Control<\/span><\/li>\n
                                  2. <\/b> SIP profile<\/span><\/li>\n
                                  3. <\/b> Translation rule<\/span><\/li>\n
                                  4. <\/b> Route group<\/span><\/li>\n<\/ol>\n

                                    Correct Answer: 3. Location-based Call Admission Control<\/b><\/p>\n

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

                                    Location-based Call Admission Control (CAC) allows Cisco Unified Communications Manager to manage available bandwidth between configured locations. When a call would exceed the configured bandwidth allocation, CAC can prevent the call from being admitted over the constrained path. This helps protect voice quality by avoiding excessive oversubscription of WAN links. SIP profiles modify signaling, translation rules manipulate digits, and route groups select gateways or trunks. Therefore, when the requirement is to admit calls only when sufficient intersite bandwidth is available, location-based CAC is the relevant feature.<\/span><\/p>\n

                                    Question 218. A SIP endpoint needs to determine whether a remote SIP service is reachable and what SIP methods or capabilities it supports without establishing a call. Which SIP method is commonly used?<\/b><\/p>\n

                                      \n
                                    1. <\/b> BYE<\/span><\/li>\n
                                    2. <\/b> OPTIONS<\/span><\/li>\n
                                    3. <\/b> CANCEL<\/span><\/li>\n
                                    4. <\/b> REFER<\/span><\/li>\n<\/ol>\n

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

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

                                      The SIP OPTIONS method can be used to query the capabilities and availability of a SIP endpoint or service without establishing a normal voice call. It can provide information about supported methods, extensions, codecs, and other capabilities depending on the response. BYE terminates an established dialog, CANCEL terminates a pending request, and REFER is commonly associated with referral or transfer operations. OPTIONS is therefore a useful diagnostic and keepalive mechanism when an administrator needs to determine whether a remote SIP service is reachable and responsive.<\/span><\/p>\n

                                      Question 219. A branch phone loses connectivity to Cisco Unified Communications Manager and registers to the local router using SRST. The administrator needs to verify that PSTN calls can be placed during survivability mode. Which configuration should be examined?<\/b><\/p>\n

                                        \n
                                      1. <\/b> Unified Communications Manager route list<\/span><\/li>\n
                                      2. <\/b> Device Mobility settings<\/span><\/li>\n
                                      3. <\/b> Local SRST dial peers and PSTN gateway configuration<\/span><\/li>\n
                                      4. <\/b> Centralized SIP profile<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 4. Local SRST dial peers and PSTN gateway configuration<\/b><\/p>\n

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

                                        When a branch phone operates in SRST mode, the branch router provides local survivable call processing. External calls generally depend on appropriately configured local dial peers and connectivity to the local PSTN gateway or another available external voice path. Centralized Unified Communications Manager route lists may not be available during a WAN outage, and Device Mobility or SIP-profile configuration does not provide the local PSTN routing function. Therefore, troubleshooting external calls during SRST operation should begin with the local SRST configuration, dial peers, and PSTN gateway connectivity.<\/span><\/p>\n

                                        Question 220. An administrator captures SIP signaling for an established call and sees an UPDATE request containing changed session information. What is the primary purpose of UPDATE in this context?<\/b><\/p>\n

                                          \n
                                        1. <\/b> Register the endpoint with the SIP server<\/span><\/li>\n
                                        2. <\/b> Terminate the established call<\/span><\/li>\n
                                        3. <\/b> Acknowledge a reliable provisional response<\/span><\/li>\n
                                        4. <\/b> Modify session information within the existing SIP dialog<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 2. Modify session information within the existing SIP dialog<\/b><\/p>\n

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

                                          The SIP UPDATE method can be used to modify session information within an existing dialog without establishing a new dialog. It may be used in scenarios involving changes to session parameters, including SDP-related information, when the relevant SIP extensions and endpoint capabilities support it. REGISTER handles endpoint registration, BYE terminates an established dialog, and PRACK acknowledges reliable provisional responses. Therefore, when an UPDATE is observed during an established SIP session and it carries changed session information, its purpose is to modify the existing session parameters rather than register or terminate the call.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 201. A Cisco Unified Communications Manager administrator needs calls to a specific PSTN destination to use a particular gateway group without changing the routing of other destinations. Which configuration should be used? Device Mobility Region configuration Location configuration Route pattern associated […]<\/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\/23423"}],"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=23423"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23423\/revisions"}],"predecessor-version":[{"id":23424,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23423\/revisions\/23424"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23423"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23423"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23423"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}