{"id":23415,"date":"2026-09-28T06:24:12","date_gmt":"2026-09-28T06:24:12","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23415"},"modified":"2026-09-28T06:24:12","modified_gmt":"2026-09-28T06:24:12","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-7-q121-140","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-7-q121-140\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 7 Q121-140"},"content":{"rendered":"

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

<\/h1>\n

Question 121. A Unified Communications Manager administrator wants a single dial-plan object to determine which route list should be used when users dial international numbers. Which configuration object should be created?<\/b><\/p>\n

    \n
  1. <\/b> Device pool<\/span><\/li>\n
  2. <\/b> Route pattern<\/span><\/li>\n
  3. <\/b> Region<\/span><\/li>\n
  4. <\/b> Media resource group<\/span><\/li>\n<\/ol>\n

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

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

    A route pattern provides destination-number matching within the Unified Communications Manager dial plan. The administrator can define a pattern for international destinations and associate it with an appropriate route list or other routing destination. The route pattern therefore determines how a dialed number enters the routing logic. A device pool provides common device settings, a region controls codec and bandwidth relationships, and a media resource group organizes media resources. When configuring international routing, the engineer should also verify the route pattern’s partition and the Calling Search Space of users who need access to it.<\/span><\/p>\n

    Question 122. A SIP trunk between Unified Communications Manager and a service provider must use encrypted signaling. The provider specifies TCP-based TLS for SIP communication. Which configuration is required?<\/b><\/p>\n

      \n
    1. <\/b> RTP-NTE only<\/span><\/li>\n
    2. <\/b> SRST fallback<\/span><\/li>\n
    3. <\/b> SIP OPTIONS polling<\/span><\/li>\n
    4. <\/b> TLS configuration with appropriate certificates and trust**<\/span><\/li>\n<\/ol>\n

      Correct Answer: 4. TLS configuration with appropriate certificates and trust<\/b><\/p>\n

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

      TLS protects SIP signaling by providing encryption and authentication mechanisms between participating systems. For a secure SIP trunk, Unified Communications Manager and the service provider must have compatible TLS settings, valid certificates, and appropriate trust relationships. RTP-NTE handles DTMF, SRST provides survivable call processing, and SIP OPTIONS can monitor reachability but does not encrypt signaling. The engineer should verify certificate validity, trusted certificate authorities, endpoint identity, supported TLS versions, and the SIP trunk security profile. A packet or SIP trace can help determine whether the TLS session fails before SIP signaling begins.<\/span><\/p>\n

      Question 123. A user can reach a route pattern from one phone but not another. Both phones use the same directory-number range, but their line configurations use different CSS values. What should the administrator compare first?<\/b><\/p>\n

        \n
      1. <\/b> Effective Calling Search Space<\/span><\/li>\n
      2. <\/b> RTP port ranges<\/span><\/li>\n
      3. <\/b> SRST dial peers<\/span><\/li>\n
      4. <\/b> CUBE media bindings<\/span><\/li>\n<\/ol>\n

        Correct Answer: 1. Effective Calling Search Space<\/b><\/p>\n

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

        Calling Search Space determines which partitions a device or line can access during digit analysis. If two phones behave differently even though they use similar numbering, their effective CSS configuration should be compared first. The administrator should inspect both line-level and device-level settings and determine which CSS is actually applied to each call. RTP port ranges and CUBE media bindings affect media transport rather than route-pattern accessibility, while SRST dial peers are relevant to local survivable routing. A CSS mismatch is a common cause when one user can access a destination and another cannot.<\/span><\/p>\n

        Question 124. A CUBE router must remove a leading digit from an incoming telephone number before sending the call toward Unified Communications Manager. Which feature is designed for this requirement?<\/b><\/p>\n

          \n
        1. <\/b> SIP session timer<\/span><\/li>\n
        2. <\/b> Route group<\/span><\/li>\n
        3. <\/b> Voice translation rule and profile<\/span><\/li>\n
        4. <\/b> Region relationship<\/span><\/li>\n<\/ol>\n

          Correct Answer: 3. Voice translation rule and profile<\/b><\/p>\n

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

          Voice translation rules and profiles are used to manipulate telephone numbers as calls pass through Cisco voice gateways and CUBE. A translation rule can remove a leading digit, add a prefix, or otherwise transform a number according to a defined pattern. The translation profile determines where the transformation is applied, such as to a called or calling number. SIP session timers control dialog refresh behavior, route groups select gateways or trunks, and regions influence codec and bandwidth policies. The engineer should verify the original number, translation-rule matching, profile application, and resulting number before it enters the Unified Communications Manager dial plan.<\/span><\/p>\n

          Question 125. An enterprise has two WAN-connected regions. Calls between them must use G.729 whenever possible because of limited bandwidth. Where should the administrator define this inter-region codec preference?<\/b><\/p>\n

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

            Correct Answer: 4. Region configuration<\/b><\/p>\n

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

            Region configuration controls codec preferences between Unified Communications Manager regions. By configuring the appropriate codec relationship between the two regions, the administrator can influence which codec is selected for calls across the constrained WAN. Calling Search Spaces and partitions affect dial-plan access, while route lists provide routing destinations. Codec selection should also consider endpoint capabilities and any additional media policies in the deployment. After configuration, the engineer should verify the actual negotiated codec in call statistics or signaling rather than assuming that the configured preference was necessarily used for every call.<\/span><\/p>\n

            Question 126. A remote office phone registers to SRST after WAN connectivity is lost. The user can call local extensions but cannot dial the PSTN. Which configuration is most relevant to the failed external call?<\/b><\/p>\n

              \n
            1. <\/b> Local SRST dial peers and PSTN gateway configuration<\/span><\/li>\n
            2. <\/b> Centralized CUCM region configuration<\/span><\/li>\n
            3. <\/b> Device Mobility group<\/span><\/li>\n
            4. <\/b> SIP session refresh interval<\/span><\/li>\n<\/ol>\n

              Correct Answer: 1. Local SRST dial peers and PSTN gateway configuration<\/b><\/p>\n

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

              When a branch is operating in SRST mode, local call processing and gateway configuration become critical for external calls. The engineer should verify that SRST has appropriate dial peers for PSTN destinations, that the destination patterns match the user’s digits, and that the local gateway or trunk is operational. Centralized region settings are not the primary routing mechanism while centralized call control is unavailable. Device Mobility and SIP session timers are also unrelated to basic PSTN route selection. Troubleshooting should begin with phone SRST registration, dial-peer matching, gateway status, and connectivity to the PSTN provider.<\/span><\/p>\n

              Question 127. A SIP provider sends a 183 Session Progress response containing SDP, and the caller begins hearing announcements before the call is answered. What does this behavior indicate?<\/b><\/p>\n

                \n
              1. <\/b> The call has already been terminated<\/span><\/li>\n
              2. <\/b> Early media is being established<\/span><\/li>\n
              3. <\/b> The endpoint has permanently registered<\/span><\/li>\n
              4. <\/b> The SIP dialog has been canceled<\/span><\/li>\n<\/ol>\n

                Correct Answer: 3. Early media is being established<\/b><\/p>\n

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

                A 183 Session Progress response can carry SDP and establish an early-media path before the called party answers. This allows the caller to hear provider announcements, ringback generated by the network, or other media while the call is still in the provisional state. A 180 Ringing response normally indicates that the called endpoint is being alerted, while BYE terminates an established dialog and CANCEL terminates a pending request. When troubleshooting early media, engineers should inspect the 183 response, SDP addresses and ports, provisional-response reliability if applicable, and the resulting RTP flow.<\/span><\/p>\n

                Question 128. A SIP endpoint sends an INVITE, receives a provisional response, and then the caller hangs up before the remote party answers. Which SIP method normally terminates the pending call attempt?<\/b><\/p>\n

                  \n
                1. <\/b> BYE<\/span><\/li>\n
                2. <\/b> REFER<\/span><\/li>\n
                3. <\/b> UPDATE<\/span><\/li>\n
                4. <\/b> CANCEL<\/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 unanswered INVITE transaction. It is different from BYE, which terminates an established SIP dialog after the call has been answered. REFER is commonly used for call-transfer operations, while UPDATE can modify session information within an existing dialog when supported. During troubleshooting, engineers should correlate the CANCEL with the original INVITE transaction and examine the resulting responses. Understanding these SIP methods is important because incorrectly interpreting CANCEL as a normal established-call teardown can lead to inaccurate conclusions about where a call failure occurred.<\/span><\/p>\n

                  Question 129. An administrator wants to configure a group of gateways so that Unified Communications Manager can select among them as part of a route list. Which object should contain those gateways?<\/b><\/p>\n

                    \n
                  1. <\/b> Route group<\/span><\/li>\n
                  2. <\/b> Translation pattern<\/span><\/li>\n
                  3. <\/b> Region<\/span><\/li>\n
                  4. <\/b> Device pool<\/span><\/li>\n<\/ol>\n

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

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

                    A route group contains gateways or trunks that Unified Communications Manager can use for call routing. Route groups can then be referenced by route lists, allowing administrators to build ordered or redundant routing structures. Translation patterns are primarily used for digit manipulation, regions define codec and bandwidth relationships, and device pools provide common device configuration. When designing gateway redundancy, the engineer should verify that the intended gateways are members of the correct route group and that the route group is referenced by the appropriate route list and route pattern.<\/span><\/p>\n

                    Question 130. A SIP trunk intermittently drops calls after several minutes. The signaling trace shows that session refreshes are not being received. Which configuration should be investigated?<\/b><\/p>\n

                      \n
                    1. <\/b> DTMF relay<\/span><\/li>\n
                    2. <\/b> SIP session timers<\/span><\/li>\n
                    3. <\/b> DHCP option 150<\/span><\/li>\n
                    4. <\/b> Route-group priority<\/span><\/li>\n<\/ol>\n

                      Correct Answer: 4. SIP session timers<\/b><\/p>\n

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

                      SIP session timers maintain an established session by requiring periodic refresh requests. If the expected refresh does not arrive before the negotiated interval expires, one endpoint may terminate the session. When calls consistently disconnect after a predictable duration, the engineer should inspect Session-Expires, Min-SE, refresher selection, and the actual UPDATE or re-INVITE messages exchanged. DTMF relay concerns keypad signaling, DHCP option 150 assists Cisco IP phone provisioning, and route-group priority controls routing. Firewall policies or SIP normalization can also interfere with refresh messages, so the signaling path should be examined end to end.<\/span><\/p>\n

                      Question 131. A CUBE router receives SIP requests from several sources. The administrator wants to ensure that only the provider’s known signaling addresses can initiate calls. Which control is most appropriate?<\/b><\/p>\n

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

                        Correct Answer: 3. Source-based access control and trusted-peer restrictions<\/b><\/p>\n

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

                        Restricting SIP signaling to known and trusted source addresses is an important part of securing CUBE. Access-control policies and trusted-peer mechanisms can reduce exposure to unauthorized SIP traffic and help limit toll-fraud risks. Region configuration controls codec and bandwidth relationships, translation patterns manipulate digits, and Device Mobility manages location-specific endpoint settings. Security policies should be designed so that legitimate provider traffic is permitted while unexpected sources are blocked. Engineers should also monitor SIP logs and access-control counters to verify that the intended traffic is being accepted and unauthorized traffic is being rejected.<\/span><\/p>\n

                        Question 132. An administrator needs a common set of Cisco IP phone settings for devices at the same site, including region, location, and related configuration. Which object is commonly used to group these settings?<\/b><\/p>\n

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

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

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

                          Device pools provide a way to assign common configuration characteristics to groups of Unified Communications Manager devices. Depending on the deployment, device pools can associate devices with regions, locations, calling search spaces, date\/time settings, and other shared configuration elements. Route groups are used for gateway or trunk selection, translation patterns manipulate digits, and SIP profiles modify SIP signaling. Device pools are therefore useful for maintaining consistent site-specific behavior without configuring every phone independently. When troubleshooting a phone’s inherited configuration, the administrator should inspect the device pool as well as any device- or line-level settings that override inherited values.<\/span><\/p>\n

                          Question 133. A CUBE engineer observes that the SIP signaling is correct, but the SDP advertises the wrong IP address for RTP. Which component should be examined first?<\/b><\/p>\n

                            \n
                          1. <\/b> Calling Search Space<\/span><\/li>\n
                          2. <\/b> Route list<\/span><\/li>\n
                          3. <\/b> CUBE media-interface and NAT configuration<\/span><\/li>\n
                          4. <\/b> Directory-number partition<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 4. CUBE media-interface and NAT configuration<\/b><\/p>\n

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

                            The SDP connection address is critical because it tells the remote endpoint where to send media. If CUBE advertises the wrong RTP address, the engineer should inspect media-interface selection, signaling and media bindings where applicable, NAT configuration, and the actual network topology. Calling Search Spaces and partitions affect call-routing access, while route lists determine routing destinations. A packet capture or SIP trace can confirm the advertised address and help compare it with the interface that should carry RTP. The engineer should also verify whether NAT is rewriting the address correctly between the enterprise and provider networks.<\/span><\/p>\n

                            Question 134. A company wants users at one site to access a PSTN route pattern, while users at another site must not see that route pattern during digit analysis. Which design provides this separation?<\/b><\/p>\n

                              \n
                            1. <\/b> Separate partitions with appropriate CSS assignments<\/span><\/li>\n
                            2. <\/b> Different RTP port ranges<\/span><\/li>\n
                            3. <\/b> Different SIP session timers<\/span><\/li>\n
                            4. <\/b> Different DHCP scopes only<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 1. Separate partitions with appropriate CSS assignments<\/b><\/p>\n

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

                              Partitions and Calling Search Spaces allow Unified Communications Manager administrators to separate access to destinations. The PSTN route pattern can be placed in a partition accessible through the CSS assigned to authorized users, while other users receive a CSS that excludes that partition. RTP ports and SIP session timers do not control dial-plan visibility, and DHCP scopes primarily provide network configuration. This design is scalable because multiple route patterns can coexist while different user groups receive different access. The engineer should verify the effective CSS at both the line and device levels when testing the policy.<\/span><\/p>\n

                              Question 135. A SIP trunk requires the enterprise to authenticate the remote system using certificates. Which security mechanism provides encrypted SIP signaling and certificate-based trust?<\/b><\/p>\n

                                \n
                              1. <\/b> TLS<\/span><\/li>\n
                              2. <\/b> RTP-NTE<\/span><\/li>\n
                              3. <\/b> SRTP only<\/span><\/li>\n
                              4. <\/b> SIP REFER<\/span><\/li>\n<\/ol>\n

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

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

                                Transport Layer Security, or TLS, is used to secure SIP signaling and can provide certificate-based authentication and trust between participating systems. TLS protects SIP messages in transit and requires compatible certificate and trust configuration. SRTP protects media rather than SIP signaling, RTP-NTE carries DTMF information, and REFER is a SIP method used for referred operations. For a certificate-based SIP trunk, the engineer should verify certificate validity, trusted certificate authorities, endpoint identity, TLS versions, and the SIP trunk security profile. Successful IP connectivity alone does not confirm that a TLS-protected SIP session can be established.<\/span><\/p>\n

                                Question 136. A user reports that a phone moved to another office but still receives the original site’s region and calling behavior. Which feature should the administrator review?<\/b><\/p>\n

                                  \n
                                1. <\/b> SIP OPTIONS<\/span><\/li>\n
                                2. <\/b> Route group<\/span><\/li>\n
                                3. <\/b> Device Mobility<\/span><\/li>\n
                                4. <\/b> DTMF relay<\/span><\/li>\n<\/ol>\n

                                  Correct Answer: 3. Device Mobility<\/b><\/p>\n

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

                                  Device Mobility is designed to apply location-specific configuration when a supported phone moves between locations. If a phone retains the original site’s settings after moving, the administrator should verify Device Mobility configuration, device mobility groups, physical locations, and the device pools associated with those locations. SIP OPTIONS handles reachability or capability checks, route groups organize gateway selection, and DTMF relay handles keypad signaling. Device Mobility can affect settings such as region, location, and calling search space, making it important in environments where phones regularly move between offices.<\/span><\/p>\n

                                  Question 137. A service provider requires a SIP header to contain a particular value that is not generated by Unified Communications Manager by default. Which feature can be used to alter SIP messages before they leave the enterprise?<\/b><\/p>\n

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

                                    Correct Answer: 1. SIP profile or normalization<\/b><\/p>\n

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

                                    SIP profiles and normalization mechanisms can manipulate SIP messages and headers to accommodate interoperability requirements between different systems. They can be used when a provider expects a particular header value or formatting that is not produced by default. Route groups determine which gateways or trunks are selected, location CAC controls media admission, and device pools provide shared endpoint settings. The engineer should first capture the outbound SIP message, identify the exact header requirement, and then implement a narrowly scoped normalization rule. Afterward, a new SIP trace should confirm that the intended header is being generated correctly.<\/span><\/p>\n

                                    Question 138. A WAN has limited bandwidth between two collaboration sites. The administrator wants Unified Communications Manager to reject additional calls once the configured voice bandwidth allocation is exhausted. Which feature should be configured?<\/b><\/p>\n

                                      \n
                                    1. <\/b> SIP URI matching<\/span><\/li>\n
                                    2. <\/b> Location-based Call Admission Control<\/span><\/li>\n
                                    3. <\/b> Translation rules<\/span><\/li>\n
                                    4. <\/b> Device Mobility<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 4. Location-based Call Admission Control<\/b><\/p>\n

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

                                      Location-based Call Admission Control helps Unified Communications Manager manage the amount of media traffic admitted between locations. By defining available bandwidth for voice and video, the system can prevent additional calls from consuming more bandwidth than the network design allows. SIP URI matching determines SIP routing behavior, translation rules modify digits, and Device Mobility applies location-specific endpoint configuration. Effective CAC planning requires accurate knowledge of WAN capacity and the bandwidth requirements of the codecs being used. Administrators should monitor call behavior after implementation to ensure that the configured limits correspond to the actual network environment.<\/span><\/p>\n

                                      Question 139. A CUBE router must route calls differently depending on whether the SIP destination URI belongs to <\/b>example.com<\/b> or <\/b>partner.net<\/b>. Which mechanism can provide this type of routing decision?<\/b><\/p>\n

                                        \n
                                      1. <\/b> DHCP option 150<\/span><\/li>\n
                                      2. <\/b> SIP URI-based dial-peer matching<\/span><\/li>\n
                                      3. <\/b> Region bandwidth<\/span><\/li>\n
                                      4. <\/b> SRST fallback<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 2. SIP URI-based dial-peer matching<\/b><\/p>\n

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

                                        SIP URI-based dial-peer matching allows CUBE to make routing decisions based on SIP URI information such as the destination domain. This is useful in SIP environments where calls are identified by addresses rather than only by numeric destination patterns. The engineer can define matching criteria for the relevant domains and associate the appropriate outbound session targets. DHCP option 150 supports Cisco phone provisioning, region bandwidth controls media policy, and SRST provides survivable call processing. When troubleshooting URI routing, the exact incoming Request-URI and dial-peer matching criteria should be inspected.<\/span><\/p>\n

                                        Question 140. During a SIP troubleshooting session, an engineer sees a successful INVITE transaction followed by a BYE shortly after the call connects. What does the BYE indicate?<\/b><\/p>\n

                                          \n
                                        1. <\/b> The endpoint is requesting registration<\/span><\/li>\n
                                        2. <\/b> A provisional response is being acknowledged<\/span><\/li>\n
                                        3. <\/b> The established SIP dialog is being terminated<\/span><\/li>\n
                                        4. <\/b> A new route pattern is being selected<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 3. The established SIP dialog is being terminated<\/b><\/p>\n

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

                                          BYE is the SIP method normally used to terminate an established SIP dialog. If an INVITE successfully establishes a call and a BYE follows shortly afterward, the engineer should investigate why one endpoint or intermediary decided to end the established session. Useful evidence includes the BYE source, accompanying SIP response codes, Reason headers when present, session timers, policy decisions, and any preceding signaling or media failures. REGISTER handles endpoint registration, PRACK acknowledges reliable provisional responses, and route patterns are Unified Communications Manager routing objects rather than SIP methods. The timing of the BYE can provide an important clue.<\/span><\/p>\n

                                          <\/h1>\n","protected":false},"excerpt":{"rendered":"

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps Question 121. A Unified Communications Manager administrator wants a single dial-plan object to determine which route list should be used when users dial international numbers. Which configuration object should be created? Device pool Route pattern Region Media resource group Correct Answer: 2. Route […]<\/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\/23415"}],"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=23415"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23415\/revisions"}],"predecessor-version":[{"id":23416,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23415\/revisions\/23416"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23415"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23415"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23415"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}