{"id":23413,"date":"2026-09-28T06:08:11","date_gmt":"2026-09-28T06:08:11","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23413"},"modified":"2026-09-28T06:08:11","modified_gmt":"2026-09-28T06:08:11","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-6-q101-120","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-6-q101-120\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 6 Q101-120"},"content":{"rendered":"

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

 <\/p>\n

Question 101. A Cisco Unified Communications Manager deployment has several SIP trunks, but an administrator wants to provide a single logical routing destination that can select among multiple trunks based on route-group priority. Which object should be configured?<\/b><\/p>\n

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

    Correct Answer: 3. Route list<\/b><\/p>\n

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

    A route list provides a logical routing destination that can reference one or more route groups. Unified Communications Manager can use the ordered route groups within the route list to select an available gateway or trunk according to the configured routing design. A translation pattern is primarily used for digit manipulation, a device pool supplies common device configuration, and a Calling Search Space controls access to partitions. Route lists are particularly useful when multiple PSTN gateways or SIP trunks provide primary and alternate paths. The administrator should also verify the associated route pattern and the ordering of the route groups.<\/span><\/p>\n

    Question 102. A user can dial internal extensions but cannot reach a SIP trunk route pattern. The route pattern is configured correctly and belongs to partition PT-PSTN. Which setting should be verified on the user’s phone or line?<\/b><\/p>\n

      \n
    1. <\/b> Calling Search Space<\/span><\/li>\n
    2. <\/b> Region codec preference<\/span><\/li>\n
    3. <\/b> Device Mobility group<\/span><\/li>\n
    4. <\/b> SIP session timer<\/span><\/li>\n<\/ol>\n

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

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

      A Calling Search Space determines which partitions a device or line can access during digit analysis. If the PSTN route pattern is placed in PT-PSTN but the user’s applicable CSS does not contain that partition, Unified Communications Manager will not select the route pattern for that user. Region configuration affects codec and bandwidth relationships, Device Mobility controls location-specific device settings, and session timers concern SIP dialog refresh behavior. When troubleshooting this situation, the engineer should check both the line-level and device-level CSS because the effective calling search space can depend on how those settings are configured.<\/span><\/p>\n

      Question 103. A CUBE deployment must send SIP signaling through one interface and RTP media through another interface because of the network topology. Which configuration area is most directly relevant?<\/b><\/p>\n

        \n
      1. <\/b> Route list<\/span><\/li>\n
      2. <\/b> Partition<\/span><\/li>\n
      3. <\/b> Device Mobility<\/span><\/li>\n
      4. <\/b> Signaling and media interface bindings<\/span><\/li>\n<\/ol>\n

        Correct Answer: 4. Signaling and media interface bindings<\/b><\/p>\n

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

        CUBE can use interface-related configuration to control where SIP signaling and media traffic are sourced or received. This becomes important when an enterprise has separate internal and external interfaces, multiple routing domains, or specific NAT and firewall requirements. Route lists and partitions are Unified Communications Manager call-routing constructs, while Device Mobility controls location-specific device settings. When troubleshooting an incorrect SIP source address or an SDP that advertises the wrong media address, the engineer should inspect CUBE interface bindings, routing, NAT behavior, and the resulting SIP and SDP messages.<\/span><\/p>\n

        Question 104. A company uses four-digit abbreviated dialing, but users in one site must have their calls transformed into the site’s full E.164 destination format before routing. Which feature is appropriate?<\/b><\/p>\n

          \n
        1. <\/b> Location CAC<\/span><\/li>\n
        2. <\/b> Translation pattern<\/span><\/li>\n
        3. <\/b> Region configuration<\/span><\/li>\n
        4. <\/b> SIP OPTIONS<\/span><\/li>\n<\/ol>\n

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

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

          Translation patterns are designed to transform dialed digits during Unified Communications Manager call processing. They can add, remove, or modify digits so that an abbreviated number becomes the appropriate full destination format. Once transformed, the resulting number can continue through normal route selection. Location CAC controls bandwidth admission, region configuration controls codec and bandwidth relationships, and SIP OPTIONS is used for peer reachability or capability checks. When implementing abbreviated dialing, the administrator should verify the translation pattern’s partition, calling search space access, transformation settings, and whether the translated number matches the intended route pattern.<\/span><\/p>\n

          Question 105. A branch office loses its WAN connection to Unified Communications Manager. Phones successfully register with the branch router in SRST mode. Which capability allows those phones to continue placing basic local calls?<\/b><\/p>\n

            \n
          1. <\/b> Local SRST call processing<\/span><\/li>\n
          2. <\/b> Centralized route lists<\/span><\/li>\n
          3. <\/b> Unified Communications Manager Extension Mobility<\/span><\/li>\n
          4. <\/b> SIP OPTIONS monitoring<\/span><\/li>\n<\/ol>\n

            Correct Answer: 1. Local SRST call processing<\/b><\/p>\n

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

            Cisco Unified SRST provides survivable call processing on a branch router when phones can no longer reach their centralized Unified Communications Manager. After entering SRST operation, supported phones can register locally and continue essential calling functions. Local dial peers and gateway configuration determine how external calls are handled. Centralized route lists and Extension Mobility depend on centralized Unified Communications Manager functionality, while SIP OPTIONS provides monitoring or capability information rather than local call processing. Engineers should verify SRST configuration, phone registration status, local dial peers, and PSTN connectivity when troubleshooting calls during WAN outages.<\/span><\/p>\n

            Question 106. A SIP provider sends a reliable provisional response containing important session information. The enterprise endpoint must explicitly acknowledge that provisional response. Which SIP method is used?<\/b><\/p>\n

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

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

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

              PRACK, or Provisional Response Acknowledgement, is used to acknowledge reliable provisional SIP responses. It is associated with the SIP reliability mechanism for provisional responses and can be important when the remote system requires reliable delivery of responses such as 183 Session Progress. BYE terminates an established dialog, REFER is commonly associated with call-transfer operations, and OPTIONS can be used for reachability or capability checks. When troubleshooting PRACK-related interoperability, engineers should inspect the provisional response, the presence of the required reliability mechanism, the PRACK transaction, and any resulting SIP error responses.<\/span><\/p>\n

              Question 107. A SIP call remains established, but one endpoint stops sending session-refresh messages. The other side eventually terminates the call after the configured interval expires. Which SIP mechanism is involved?<\/b><\/p>\n

                \n
              1. <\/b> SIP URI matching<\/span><\/li>\n
              2. <\/b> SIP session timers<\/span><\/li>\n
              3. <\/b> DTMF relay<\/span><\/li>\n
              4. <\/b> Route-group hunting<\/span><\/li>\n<\/ol>\n

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

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

                SIP session timers provide a mechanism for periodically refreshing an established SIP session. Depending on the negotiated timer behavior, one endpoint acts as a session refresher and sends the required refresh request. If refreshes are not received within the expected interval, the session can eventually be terminated. SIP URI matching controls how SIP destinations are identified, DTMF relay transports keypad digits, and route-group hunting controls call-path selection. When calls disconnect at predictable intervals, engineers should inspect Session-Expires and Min-SE values, refresher behavior, UPDATE or re-INVITE exchanges, and whether intermediate devices are permitting the refresh messages.<\/span><\/p>\n

                Question 108. A CUBE router must prevent unauthorized SIP systems from sending calls through the enterprise voice gateway. Which security approach should be considered?<\/b><\/p>\n

                  \n
                1. <\/b> Increase RTP packet size<\/span><\/li>\n
                2. <\/b> Change the directory-number format<\/span><\/li>\n
                3. <\/b> Disable all SIP responses<\/span><\/li>\n
                4. <\/b> Restrict trusted SIP sources and apply appropriate access controls<\/span><\/li>\n<\/ol>\n

                  Correct Answer: 4. Restrict trusted SIP sources and apply appropriate access controls<\/b><\/p>\n

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

                  CUBE security should restrict which external systems are permitted to establish SIP relationships with the enterprise. Trusted-source restrictions, access-control policies, authentication where appropriate, and carefully defined dial peers can reduce the risk of unauthorized call attempts or toll fraud. Increasing RTP packet size does not provide signaling security, changing directory-number format is unrelated to peer authorization, and disabling SIP responses would disrupt normal interoperability. Administrators should also apply least-privilege routing, monitor unexpected SIP traffic, and verify that inbound calls are accepted only from intended provider or peer addresses.<\/span><\/p>\n

                  Question 109. A SIP endpoint sends an INVITE containing an SDP offer. The receiving system responds with SDP describing the media parameters it will use. Which protocol format carries these media descriptions?<\/b><\/p>\n

                    \n
                  1. <\/b> SNMP<\/span><\/li>\n
                  2. <\/b> LDAP<\/span><\/li>\n
                  3. <\/b> SDP<\/span><\/li>\n
                  4. <\/b> RTP<\/span><\/li>\n<\/ol>\n

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

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

                    Session Description Protocol, or SDP, describes session and media parameters within signaling messages. It can identify media types, codecs, IP addresses, ports, and other information needed to establish the media session. RTP is responsible for carrying real-time media packets after the media path is established, while SNMP is used for network management and LDAP commonly provides directory or authentication-related services. Engineers troubleshooting SIP audio should inspect SDP in the INVITE and subsequent responses to determine whether the advertised media addresses, ports, and codec capabilities are appropriate and reachable.<\/span><\/p>\n

                    Question 110. A user receives a new IP phone that must automatically obtain its Cisco Unified Communications Manager configuration after connecting to the network. Which DHCP option is commonly used by Cisco IP phones to identify the TFTP server?<\/b><\/p>\n

                      \n
                    1. <\/b> DHCP option 3<\/span><\/li>\n
                    2. <\/b> DHCP option 150<\/span><\/li>\n
                    3. <\/b> DHCP option 66 exclusively<\/span><\/li>\n
                    4. <\/b> DHCP option 82<\/span><\/li>\n<\/ol>\n

                      Correct Answer: 2. DHCP option 150<\/b><\/p>\n

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

                      Cisco IP phones commonly use DHCP option 150 to learn the address of the TFTP server from which they can obtain configuration information. The TFTP infrastructure can then provide the files needed for phone provisioning and Unified Communications Manager registration. DHCP option 3 identifies the default gateway, while option 82 is associated with DHCP relay information. Option 66 can also identify a TFTP server in some environments and devices, but Cisco IP phone deployments commonly use option 150. When troubleshooting phone provisioning, engineers should verify DHCP responses, network reachability, TFTP availability, and the configured Unified Communications Manager information.<\/span><\/p>\n

                      Question 111. An administrator needs to change the SIP From header on calls leaving CUBE while leaving the dialed digits unchanged. Which mechanism is most appropriate?<\/b><\/p>\n

                        \n
                      1. <\/b> Location bandwidth configuration<\/span><\/li>\n
                      2. <\/b> Translation rule<\/span><\/li>\n
                      3. <\/b> SIP profile or normalization<\/span><\/li>\n
                      4. <\/b> Route group priority<\/span><\/li>\n<\/ol>\n

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

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

                        SIP profiles and normalization mechanisms are intended for controlled manipulation of SIP messages and headers. If the requirement is to modify the From header while keeping the called digits unchanged, SIP message manipulation is more appropriate than digit translation. Translation rules and profiles are primarily used to transform telephone numbers, while route groups determine gateway or trunk selection and location bandwidth settings govern call admission. The engineer should identify the exact header syntax required by the provider, apply the smallest necessary manipulation, and validate the resulting outbound INVITE with a SIP trace or packet capture.<\/span><\/p>\n

                        Question 112. A CUBE engineer wants to identify which outbound dial peer is being selected for a particular SIP call. Which troubleshooting approach is most useful?<\/b><\/p>\n

                          \n
                        1. <\/b> Inspect dial-peer matching and voice-routing diagnostics<\/span><\/li>\n
                        2. <\/b> Change the phone wallpaper<\/span><\/li>\n
                        3. <\/b> Modify region bandwidth<\/span><\/li>\n
                        4. <\/b> Disable SIP timers<\/span><\/li>\n<\/ol>\n

                          Correct Answer: 1. Inspect dial-peer matching and voice-routing diagnostics<\/b><\/p>\n

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

                          CUBE call routing depends heavily on inbound and outbound dial-peer matching. Voice-routing diagnostic commands and call-processing logs can show which dial peer is selected and why a particular call path is being used. This is especially useful when multiple dial peers have overlapping destination patterns or URI-based matching criteria. Phone wallpaper and region bandwidth do not determine CUBE dial-peer selection, and disabling SIP timers is unrelated to initial routing. The engineer should inspect the received calling and called information, dial-peer configuration, session target, and any transformations applied before forwarding the call.<\/span><\/p>\n

                          Question 113. A Unified Communications Manager administrator wants to ensure that a route pattern can be accessed only by a specific class of users. Which design should be implemented?<\/b><\/p>\n

                            \n
                          1. <\/b> Put the route pattern in a partition and provide that partition through an appropriate CSS<\/span><\/li>\n
                          2. <\/b> Change the region codec preference<\/span><\/li>\n
                          3. <\/b> Enable SRST on every phone<\/span><\/li>\n
                          4. <\/b> Configure an RTP media resource group<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 2. Put the route pattern in a partition and provide that partition through an appropriate CSS<\/b><\/p>\n

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

                            Partitions and Calling Search Spaces are the standard Unified Communications Manager mechanisms for controlling access to dial-plan destinations. The administrator can place a route pattern in a dedicated partition and assign a CSS containing that partition only to users who are authorized to access it. Region settings affect codec and bandwidth relationships, SRST provides survivability, and media resource groups organize media resources. This design allows multiple classes of users to share the same overall dial plan while receiving different access privileges. The administrator should verify that the effective CSS includes the required partition and does not unintentionally expose restricted destinations.<\/span><\/p>\n

                            Question 114. A SIP trunk repeatedly fails after the certificate on the remote system was replaced. Other network connectivity tests succeed. What should the engineer investigate?<\/b><\/p>\n

                              \n
                            1. <\/b> RTP packetization interval only<\/span><\/li>\n
                            2. <\/b> Route-pattern wildcard syntax<\/span><\/li>\n
                            3. <\/b> TLS certificate trust and certificate identity<\/span><\/li>\n
                            4. <\/b> Device Mobility settings<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 3. TLS certificate trust and certificate identity<\/b><\/p>\n

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

                              Replacing a certificate can affect TLS-based SIP connectivity even when basic IP reachability remains intact. The engineer should verify that the new certificate is valid, trusted by the local system, and identifies the remote endpoint in a manner consistent with the configured TLS expectations. Certificate chains, expiration, trusted authorities, and relevant certificate names should all be examined. RTP packetization affects media, route-pattern syntax affects call routing, and Device Mobility concerns endpoint location behavior. TLS diagnostics and SIP traces can help distinguish certificate validation failures from other SIP trunk configuration problems.<\/span><\/p>\n

                              Question 115. A user reports that digits entered during an IVR call are not recognized, even though the voice path itself is clear. Which area should be investigated?<\/b><\/p>\n

                                \n
                              1. <\/b> DTMF relay configuration<\/span><\/li>\n
                              2. <\/b> Region name length<\/span><\/li>\n
                              3. <\/b> Route-group description<\/span><\/li>\n
                              4. <\/b> Device pool time zone only<\/span><\/li>\n<\/ol>\n

                                Correct Answer: 1. DTMF relay configuration<\/b><\/p>\n

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

                                DTMF digits can be transported separately from normal voice audio, depending on the selected signaling method. If an IVR does not recognize digits while the voice path remains clear, the engineer should investigate DTMF relay configuration and interoperability between the endpoints, CUBE, SIP trunk, and provider. Common mechanisms include RTP-NTE and SIP-based methods, depending on the environment. Region names, route-group descriptions, and device-pool time-zone settings do not normally determine DTMF transport. A SIP and RTP capture can help confirm whether the digits are generated, transmitted using the expected method, and received correctly by the IVR.<\/span><\/p>\n

                                Question 116. A SIP endpoint sends a CANCEL because the caller has abandoned an unanswered call attempt. What does the CANCEL request normally terminate?<\/b><\/p>\n

                                  \n
                                1. <\/b> An established media session<\/span><\/li>\n
                                2. <\/b> A pending SIP transaction that has not yet completed<\/span><\/li>\n
                                3. <\/b> A completed call-transfer dialog<\/span><\/li>\n
                                4. <\/b> The endpoint’s registration permanently<\/span><\/li>\n<\/ol>\n

                                  Correct Answer: 4. A pending SIP transaction that has not yet completed<\/b><\/p>\n

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

                                  SIP CANCEL is used to terminate a pending request, such as an unanswered INVITE transaction, before the request has completed. It is not the normal method for terminating an established call; BYE is used for that purpose. CANCEL also does not permanently remove an endpoint’s registration or inherently terminate a completed transfer operation. Understanding the distinction between CANCEL and BYE is important when analyzing SIP traces. Engineers should examine the original transaction, response sequence, and timing to determine whether the CANCEL is being generated correctly and whether the remote system responds as expected.<\/span><\/p>\n

                                  Question 117. A Unified Communications Manager deployment has multiple sites with different WAN capacities. The administrator needs to control which codec is preferred between each pair of sites. Which configuration should be reviewed?<\/b><\/p>\n

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

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

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

                                    Region configuration determines codec preferences and bandwidth relationships between Unified Communications Manager regions. Administrators can define which codecs are preferred or allowed for communication between different regions and can associate bandwidth limits with those relationships. Calling Search Spaces control dial-plan access, translation patterns manipulate digits, and route groups organize gateway or trunk selection. When troubleshooting unexpected codec selection, the engineer should inspect the regions assigned to the devices, the configured region relationship, endpoint capabilities, and any additional policies that could influence the final negotiated codec.<\/span><\/p>\n

                                    Question 118. A provider uses SIP URIs instead of traditional telephone-number matching. CUBE must select a specific outbound dial peer based on the SIP destination domain. Which matching capability is relevant?<\/b><\/p>\n

                                      \n
                                    1. <\/b> Location CAC<\/span><\/li>\n
                                    2. <\/b> Device Mobility<\/span><\/li>\n
                                    3. <\/b> SIP URI matching<\/span><\/li>\n
                                    4. <\/b> DHCP option 150<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 4. SIP URI matching<\/b><\/p>\n

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

                                      SIP URI matching allows CUBE to use SIP URI information, including domains or user portions, when selecting appropriate dial peers. This is useful for SIP environments where destinations are represented as addresses such as user@domain rather than traditional numeric telephone numbers. Location CAC controls bandwidth admission, Device Mobility provides location-specific endpoint behavior, and DHCP option 150 supports Cisco phone provisioning. When implementing URI-based routing, the engineer should verify the incoming SIP URI format, matching rules, dial-peer configuration, and session target so that calls are forwarded to the intended SIP domain.<\/span><\/p>\n

                                      Question 119. A SIP call is established through CUBE, but the provider reports that the enterprise sends a private IP address in the SDP. Which evidence would most directly confirm the problem?<\/b><\/p>\n

                                        \n
                                      1. <\/b> A SIP trace showing the SDP connection address<\/span><\/li>\n
                                      2. <\/b> A route list containing two route groups<\/span><\/li>\n
                                      3. <\/b> A phone’s Calling Search Space<\/span><\/li>\n
                                      4. <\/b> A Unified Communications Manager device pool name<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 3. A SIP trace showing the SDP connection address<\/b><\/p>\n

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

                                        The SDP connection address identifies where the remote endpoint should send media. A SIP trace containing the SDP body can directly reveal whether CUBE is advertising an inappropriate private address to the provider. This evidence is more directly relevant than a route list, Calling Search Space, or device pool name because those settings do not show the media address presented in the SIP session. After identifying the incorrect SDP address, the engineer should investigate CUBE media bindings, NAT, interface selection, and related configuration to determine why the private address is being advertised.<\/span><\/p>\n

                                        Question 120. A SIP endpoint sends an UPDATE during an established dialog to modify session information without creating a separate new dialog. Which statement best describes the purpose of UPDATE?<\/b><\/p>\n

                                          \n
                                        1. <\/b> It permanently unregisters the endpoint<\/span><\/li>\n
                                        2. <\/b> It selects a Unified Communications Manager route group<\/span><\/li>\n
                                        3. <\/b> It terminates the established call<\/span><\/li>\n
                                        4. <\/b> It can modify session information within an existing SIP dialog<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 1. It can modify session information within an existing SIP dialog<\/b><\/p>\n

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

                                          The SIP UPDATE method can modify session information within an existing dialog without establishing a separate new dialog. Depending on the negotiated capabilities and deployment, UPDATE can carry SDP and support changes to session parameters during call setup or an established dialog. REGISTER is used for endpoint registration, BYE terminates an established dialog, and route groups are Unified Communications Manager routing constructs rather than SIP methods. When troubleshooting UPDATE behavior, engineers should inspect the dialog state, Allow and Supported headers where relevant, SDP contents, and the response generated by the remote SIP system.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 101. A Cisco Unified Communications Manager deployment has several SIP trunks, but an administrator wants to provide a single logical routing destination that can select among multiple trunks based on route-group priority. Which object should be configured? Translation pattern Device pool […]<\/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\/23413"}],"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=23413"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23413\/revisions"}],"predecessor-version":[{"id":23414,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23413\/revisions\/23414"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23413"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23413"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23413"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}