{"id":23411,"date":"2026-09-28T06:07:56","date_gmt":"2026-09-28T06:07:56","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23411"},"modified":"2026-09-28T06:07:56","modified_gmt":"2026-09-28T06:07:56","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-5-q81-100","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-5-q81-100\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 5 Q81-100"},"content":{"rendered":"

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

 <\/p>\n

Question 81. A Cisco Unified Communications Manager administrator wants to route calls to a specific SIP trunk when the dialed destination matches a defined pattern. Which configuration object provides the destination-number match?<\/b><\/p>\n

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

    Correct Answer: 4. Route pattern<\/b><\/p>\n

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

    A route pattern defines a destination-number pattern that Cisco Unified Communications Manager uses during digit analysis. When the dialed number matches the pattern, call processing can send the call toward a configured route list, gateway, or SIP trunk depending on the routing design. A route group organizes gateways or trunks for path selection, while a device pool provides common device configuration. A media resource group organizes resources such as media termination or transcoding resources. Route patterns are therefore a fundamental component when implementing destination-based call routing through PSTN gateways or SIP trunks.<\/span><\/p>\n

    Question 82. A user is permitted to call internal extensions but receives a reorder tone when dialing a PSTN destination. The PSTN route pattern exists, but its partition is not included in the user’s CSS. What is the most likely cause?<\/b><\/p>\n

      \n
    1. <\/b> Incorrect codec selection<\/span><\/li>\n
    2. <\/b> Calling Search Space does not include the route-pattern partition<\/span><\/li>\n
    3. <\/b> RTP ports are blocked<\/span><\/li>\n
    4. <\/b> SRST is enabled<\/span><\/li>\n<\/ol>\n

      Correct Answer: 2. Calling Search Space does not include the route-pattern partition<\/b><\/p>\n

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

      Cisco Unified Communications Manager uses Calling Search Spaces to determine which partitions a calling device or line can search. If a PSTN route pattern is placed in a partition that is absent from the caller’s CSS, the route pattern cannot be selected even though it is correctly configured. Codec and RTP issues generally affect media after routing and call establishment, while SRST is relevant to survivability during loss of centralized call control. Comparing the CSS of the affected user with that of a working user is a useful troubleshooting technique. The administrator should verify both line-level and device-level CSS configuration.<\/span><\/p>\n

      Question 83. A company wants users to dial four-digit internal extensions while the actual directory numbers are eight digits. Which Unified Communications Manager feature can transform the abbreviated number into the required full number?<\/b><\/p>\n

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

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

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

        Translation patterns allow Cisco Unified Communications Manager to manipulate dialed digits before continuing call processing. An abbreviated four-digit number can be transformed into an eight-digit directory-number format by adding or modifying the required digits. The translated result can then be routed according to the applicable dial plan. Route groups are used for gateway selection, while regions influence codec and bandwidth relationships and device pools provide common configuration settings. When implementing abbreviated dialing, administrators should also consider partitions and CSS because the translation pattern itself must be reachable by the calling user.<\/span><\/p>\n

        Question 84. An enterprise has a primary PSTN gateway and a secondary gateway at another site. The administrator wants Unified Communications Manager to attempt the primary gateway first and use the secondary gateway if necessary. Which configuration should be used?<\/b><\/p>\n

          \n
        1. <\/b> Translation pattern only<\/span><\/li>\n
        2. <\/b> Device Mobility group<\/span><\/li>\n
        3. <\/b> Route list with ordered route groups<\/span><\/li>\n
        4. <\/b> Region configuration<\/span><\/li>\n<\/ol>\n

          Correct Answer: 3. Route list with ordered route groups<\/b><\/p>\n

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

          Route lists provide an ordered routing structure that references route groups. Each route group can contain one or more gateways or trunks, allowing Unified Communications Manager to use a preferred path and then consider an alternate path according to the configured routing logic. Translation patterns manipulate digits, Device Mobility groups support location-specific device behavior, and region configuration controls codec and bandwidth relationships. For gateway redundancy, the engineer should verify the route pattern points to the intended route list and that the route groups contain the correct gateways in the desired order.<\/span><\/p>\n

          Question 85. A collaboration administrator needs to limit the audio bandwidth available for calls between two Unified Communications Manager sites. Which configuration should be examined?<\/b><\/p>\n

            \n
          1. <\/b> SIP profile<\/span><\/li>\n
          2. <\/b> Region configuration<\/span><\/li>\n
          3. <\/b> Route group<\/span><\/li>\n
          4. <\/b> Partition<\/span><\/li>\n<\/ol>\n

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

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

            Region configuration controls codec and bandwidth relationships between Unified Communications Manager regions. Administrators can specify the maximum audio and video bandwidth characteristics between regions so that collaboration traffic aligns with WAN capacity. This is particularly important when sites are connected through constrained links. SIP profiles modify SIP signaling, route groups determine gateway selection, and partitions organize dial-plan objects. For a complete WAN design, region settings should be considered together with locations and call admission control. The actual negotiated codec should also be verified because endpoint capabilities and other policies can affect the final media characteristics.<\/span><\/p>\n

            Question 86. A user moves a Cisco IP phone from the headquarters office to a branch office. The phone should retain its extension but receive the branch’s calling search space and region settings. Which feature supports this requirement?<\/b><\/p>\n

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

              Correct Answer: 4. Device Mobility<\/b><\/p>\n

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

              Device Mobility allows Unified Communications Manager to apply location-specific configuration when a supported device registers from a different location. The phone can retain its identity and extension while receiving settings appropriate to the new site, such as device pool, region, location, and calling search space information. SIP OPTIONS provides capability or reachability checks, route groups control gateway selection, and translation patterns manipulate digits. Device Mobility is especially useful for organizations where users or phones move between offices. The administrator should verify device mobility groups, physical locations, and associated device pools when troubleshooting unexpected settings after a move.<\/span><\/p>\n

              Question 87. An enterprise wants only authorized users to reach a group of international route patterns. Which design provides logical access control within Unified Communications Manager?<\/b><\/p>\n

                \n
              1. <\/b> Partitions combined with Calling Search Spaces<\/span><\/li>\n
              2. <\/b> Region bandwidth settings<\/span><\/li>\n
              3. <\/b> RTP port restrictions<\/span><\/li>\n
              4. <\/b> Codec preference lists<\/span><\/li>\n<\/ol>\n

                Correct Answer: 1. Partitions combined with Calling Search Spaces<\/b><\/p>\n

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

                Partitions and Calling Search Spaces provide the core logical access-control mechanism for Unified Communications Manager dial plans. International route patterns can be placed in a dedicated partition, and only users whose CSS includes that partition can access those destinations. Other users can receive a CSS that excludes the restricted partition. Region settings control media characteristics, RTP port restrictions affect media transport, and codec preference lists influence codec negotiation. This approach allows the administrator to maintain a complete enterprise dial plan while controlling which users can access sensitive or costly destination classes.<\/span><\/p>\n

                Question 88. A CUBE router receives a SIP INVITE with a called number format that does not match the internal dial plan. The enterprise needs to add a prefix before forwarding the call to Unified Communications Manager. Which capability should be considered?<\/b><\/p>\n

                  \n
                1. <\/b> NTP authentication<\/span><\/li>\n
                2. <\/b> OSPF redistribution<\/span><\/li>\n
                3. <\/b> Translation rules and profiles<\/span><\/li>\n
                4. <\/b> SRST fallback<\/span><\/li>\n<\/ol>\n

                  Correct Answer: 3. Translation rules and profiles<\/b><\/p>\n

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

                  Translation rules and profiles can transform telephone digits as calls pass through Cisco voice gateways and CUBE. In this case, a required prefix can be added to the called number so that the resulting number matches the internal Unified Communications Manager dial plan. NTP authentication protects time synchronization, OSPF redistribution exchanges routing information, and SRST provides survivable call processing. The engineer should verify the incoming called number, the required transformation, the point at which the translation is applied, and the resulting dial-peer or SIP-trunk match. Testing should confirm that only the intended calls are modified.<\/span><\/p>\n

                  Question 89. A SIP provider requires CUBE to change a specific SIP header before forwarding an INVITE. Which feature is designed for SIP message manipulation?<\/b><\/p>\n

                    \n
                  1. <\/b> Region configuration<\/span><\/li>\n
                  2. <\/b> Calling Search Space<\/span><\/li>\n
                  3. <\/b> Route group<\/span><\/li>\n
                  4. <\/b> SIP profile or SIP normalization<\/span><\/li>\n<\/ol>\n

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

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

                    SIP profiles and SIP normalization mechanisms allow selected SIP messages and headers to be modified as they pass through supported Cisco collaboration components. This capability is useful when an enterprise and service provider have different expectations for SIP header values or formatting. Region configuration controls media bandwidth and codec relationships, Calling Search Spaces control dial-plan access, and route groups organize gateway selection. Before applying a SIP manipulation rule, the engineer should capture the signaling and identify the exact header that needs to change. Narrowly scoped rules help avoid unintended effects on unrelated SIP transactions.<\/span><\/p>\n

                    Question 90. A SIP trunk is configured with TLS, but calls fail during signaling because the two systems cannot establish a trusted TLS session. Which area should be checked first?<\/b><\/p>\n

                      \n
                    1. <\/b> Certificates and TLS trust configuration<\/span><\/li>\n
                    2. <\/b> RTP codec preference<\/span><\/li>\n
                    3. <\/b> Calling Search Space<\/span><\/li>\n
                    4. <\/b> Route-group order<\/span><\/li>\n<\/ol>\n

                      Correct Answer: 1. Certificates and TLS trust configuration<\/b><\/p>\n

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

                      TLS-protected SIP signaling depends on successful certificate validation and appropriate trust configuration between the communicating systems. If the TLS session cannot be established, the engineer should verify certificates, certificate validity periods, subject names or SANs where applicable, trusted certificate authorities, supported TLS versions, and the configured trust relationships. RTP codec preferences affect media after signaling succeeds, while Calling Search Spaces and route groups affect call routing. A SIP trace or TLS diagnostic can help identify whether the failure involves certificate validation, protocol compatibility, or another security parameter.<\/span><\/p>\n

                      Question 91. A SIP call establishes successfully, but neither endpoint receives RTP audio. SDP negotiation appears valid. Which troubleshooting area should receive immediate attention?<\/b><\/p>\n

                        \n
                      1. <\/b> Calling Search Space<\/span><\/li>\n
                      2. <\/b> RTP path and firewall\/NAT behavior<\/span><\/li>\n
                      3. <\/b> Translation pattern<\/span><\/li>\n
                      4. <\/b> Device pool naming<\/span><\/li>\n<\/ol>\n

                        Correct Answer: 3. RTP path and firewall\/NAT behavior<\/b><\/p>\n

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

                        When SIP signaling and SDP negotiation succeed but no audio flows in either direction, the RTP media path should be investigated. The engineer should verify the media addresses and ports advertised in SDP, routing between the endpoints, firewall policies, NAT translations, CUBE media bindings, and whether RTP packets are actually being transmitted and received. Calling Search Spaces and translation patterns primarily affect call routing, while device pool naming does not control RTP forwarding. Packet captures and voice-gateway statistics can help determine whether the media is generated but blocked, sent to an incorrect address, or never transmitted.<\/span><\/p>\n

                        Question 92. A CUBE router has an internal interface and an external provider-facing interface. The provider receives an SDP address from the enterprise that contains an unreachable private IP address. Which configuration should be investigated?<\/b><\/p>\n

                          \n
                        1. <\/b> Calling Search Space<\/span><\/li>\n
                        2. <\/b> Media interface or signaling\/media binding configuration<\/span><\/li>\n
                        3. <\/b> Route list<\/span><\/li>\n
                        4. <\/b> Region bandwidth<\/span><\/li>\n<\/ol>\n

                          Correct Answer: 2. Media interface or signaling\/media binding configuration<\/b><\/p>\n

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

                          CUBE can have multiple interfaces and routing contexts, so selecting the correct address for SIP signaling and RTP media is important. If SDP contains an unreachable private address, the engineer should inspect media bindings, interface selection, NAT behavior, and related CUBE configuration. The provider must receive an address that it can actually use to send RTP toward the enterprise. Calling Search Spaces and route lists influence call routing, while region bandwidth affects codec and bandwidth policy. Comparing the SDP addresses with the actual network topology is an effective way to identify media-interface selection problems.<\/span><\/p>\n

                          Question 93. A branch router is operating in SRST mode after losing connectivity to centralized call control. Internal extension-to-extension calls work, but PSTN calls fail. Which component should be checked?<\/b><\/p>\n

                            \n
                          1. <\/b> Local SRST dial peers and PSTN gateway configuration<\/span><\/li>\n
                          2. <\/b> Central CUCM device pool<\/span><\/li>\n
                          3. <\/b> Remote region relationship<\/span><\/li>\n
                          4. <\/b> SIP provider’s video codec<\/span><\/li>\n<\/ol>\n

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

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

                            When phones successfully operate under SRST but cannot place PSTN calls, local outbound call-routing configuration should be investigated. SRST depends on local dial peers, destination patterns, voice gateway configuration, and available PSTN connectivity. Centralized device-pool settings and remote region relationships are not the immediate mechanisms controlling calls while the branch is operating independently. A video codec is also unrelated to basic PSTN routing. The engineer should verify the phone’s SRST registration, dial-peer matching, gateway status, and physical or SIP connectivity toward the PSTN service.<\/span><\/p>\n

                            Question 94. A Unified Communications Manager administrator wants to restrict a user to internal calls and selected external destinations while keeping the overall dial plan unchanged. Which configuration provides this control?<\/b><\/p>\n

                              \n
                            1. <\/b> Region configuration<\/span><\/li>\n
                            2. <\/b> Media resource group<\/span><\/li>\n
                            3. <\/b> Calling Search Space and partition design<\/span><\/li>\n
                            4. <\/b> RTP port range<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 4. Calling Search Space and partition design<\/b><\/p>\n

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

                              Calling Search Spaces and partitions allow administrators to control which dial-plan objects a user can access. Internal directory numbers and selected external route patterns can be placed in appropriate partitions, while the user’s CSS can include only the partitions required for authorized calling. This approach changes the user’s access without requiring the overall dial plan to be removed or duplicated. Regions control codec and bandwidth behavior, media resource groups organize media resources, and RTP port ranges affect media transport. Administrators should verify both device-level and line-level CSS settings when implementing or troubleshooting calling restrictions.<\/span><\/p>\n

                              Question 95. Two Unified Communications Manager sites use a constrained WAN. The administrator wants to prevent additional calls from being admitted after the configured WAN voice bandwidth is exhausted. Which feature should be used?<\/b><\/p>\n

                                \n
                              1. <\/b> SIP normalization<\/span><\/li>\n
                              2. <\/b> Location-based call admission control<\/span><\/li>\n
                              3. <\/b> Route pattern<\/span><\/li>\n
                              4. <\/b> Device Mobility<\/span><\/li>\n<\/ol>\n

                                Correct Answer: 2. Location-based call admission control<\/b><\/p>\n

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

                                Location-based call admission control allows Unified Communications Manager to control how much audio and video traffic is admitted between locations. Once the configured bandwidth allocation is consumed, additional calls can be denied or otherwise prevented from exceeding the planned capacity. This protects existing calls and helps maintain predictable voice quality over constrained WAN links. SIP normalization modifies signaling, route patterns perform destination matching, and Device Mobility provides location-specific device configuration. CAC should be designed using realistic bandwidth values and validated against the codecs actually used between the locations.<\/span><\/p>\n

                                Question 96. A CUBE router receives calls from a SIP provider and must route them to different internal destinations based on the called number. Which configuration element is primarily responsible for matching the destination and selecting an outbound call path?<\/b><\/p>\n

                                  \n
                                1. <\/b> Dial peer<\/span><\/li>\n
                                2. <\/b> NTP server<\/span><\/li>\n
                                3. <\/b> OSPF neighbor<\/span><\/li>\n
                                4. <\/b> DHCP relay<\/span><\/li>\n<\/ol>\n

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

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

                                  CUBE uses dial peers to define call-routing behavior and determine how matching calls should be forwarded. Depending on the deployment, dial peers can use destination patterns, URI matching, session targets, and other criteria to select an outbound destination. NTP provides time synchronization, OSPF establishes routing relationships, and DHCP relay forwards address-assignment requests. When troubleshooting CUBE call routing, engineers should inspect both inbound and outbound dial-peer matching and verify that the selected session target is reachable. Correct dial-peer configuration is essential for reliable SIP and PSTN interoperability.<\/span><\/p>\n

                                  Question 97. A SIP endpoint is behind NAT and can establish signaling, but direct RTP connectivity is not possible. Which technology can provide a framework for discovering and selecting viable media paths?<\/b><\/p>\n

                                    \n
                                  1. <\/b> ICE<\/span><\/li>\n
                                  2. <\/b> SRST<\/span><\/li>\n
                                  3. <\/b> CUBE dial-peer matching<\/span><\/li>\n
                                  4. <\/b> DHCP option 150<\/span><\/li>\n<\/ol>\n

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

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

                                    Interactive Connectivity Establishment, or ICE, is designed to discover and evaluate possible media paths between endpoints, including scenarios involving NAT. ICE uses candidate addresses and can work with mechanisms such as STUN and TURN to determine whether direct or relayed media connectivity is possible. SRST provides survivable call processing, CUBE dial-peer matching determines call-routing behavior, and DHCP option 150 assists Cisco IP phone provisioning. When troubleshooting NAT-related media problems, engineers should verify whether the endpoints and intermediary systems support the required ICE behavior and whether the resulting candidates are reachable.<\/span><\/p>\n

                                    Question 98. A provider requires secure SIP signaling and secure RTP media for an enterprise SIP trunk. Which combination should be configured where supported?<\/b><\/p>\n

                                      \n
                                    1. <\/b> UDP SIP and ordinary RTP<\/span><\/li>\n
                                    2. <\/b> TLS for signaling and SRTP for media<\/span><\/li>\n
                                    3. <\/b> OSPF authentication and GRE<\/span><\/li>\n
                                    4. <\/b> DHCP snooping and IP SLA<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 4. TLS for signaling and SRTP for media<\/b><\/p>\n

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

                                      TLS provides protection for SIP signaling, while SRTP provides security for the real-time media stream. Using both mechanisms allows an enterprise to protect call-control messages and voice media independently. Ordinary UDP-based SIP and RTP do not provide equivalent cryptographic protection. OSPF authentication and GRE address network-layer functions, while DHCP snooping and IP SLA serve security and monitoring purposes unrelated to SIP-media encryption. Secure SIP deployments also require compatible certificate and trust configuration for TLS and compatible SRTP negotiation between the endpoints or intermediary devices.<\/span><\/p>\n

                                      Question 99. An engineer sees periodic SIP OPTIONS messages exchanged between CUBE and a provider even when no calls are active. What is the most likely purpose of these messages?<\/b><\/p>\n

                                        \n
                                      1. <\/b> To transfer an active call<\/span><\/li>\n
                                      2. <\/b> To negotiate an RTP codec for an established call<\/span><\/li>\n
                                      3. <\/b> To check SIP peer reachability or capabilities<\/span><\/li>\n
                                      4. <\/b> To terminate a SIP dialog<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 3. To check SIP peer reachability or capabilities<\/b><\/p>\n

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

                                        SIP OPTIONS can be used to query the availability or capabilities of a SIP peer without establishing a normal call. Periodic OPTIONS exchanges are commonly useful for keepalive or reachability monitoring between voice systems. REFER is associated with referred operations such as call transfer, codec negotiation occurs through SDP during session establishment or modification, and BYE terminates an established dialog. When troubleshooting SIP trunk availability, engineers can examine OPTIONS requests and responses for connectivity, response codes, timing, and possible firewall or routing problems. A successful OPTIONS exchange does not by itself guarantee successful end-to-end calling.<\/span><\/p>\n

                                        Question 100. A SIP call connects and initially has two-way audio, but audio becomes one-way after a call-transfer operation. Which area should be investigated first?<\/b><\/p>\n

                                          \n
                                        1. <\/b> Mid-call SIP signaling and SDP\/media renegotiation<\/span><\/li>\n
                                        2. <\/b> Phone display language<\/span><\/li>\n
                                        3. <\/b> NTP stratum<\/span><\/li>\n
                                        4. <\/b> Directory number description<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 2. Phone display language<\/b><\/p>\n

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

                                          The most relevant troubleshooting area for audio changing after a call transfer is mid-call SIP signaling and SDP\/media renegotiation. A transfer can introduce a new media path, endpoint, or session description, so the engineer should compare the SDP and SIP signaling before and after the transfer. The advertised RTP addresses and ports, CUBE media handling, NAT, firewall policies, and interface selection should also be checked. Phone display language, NTP stratum, and directory-number descriptions do not control RTP media. Packet captures can help determine whether the new media path is reachable in both directions.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 81. A Cisco Unified Communications Manager administrator wants to route calls to a specific SIP trunk when the dialed destination matches a defined pattern. Which configuration object provides the destination-number match? Route group Device pool Media resource group Route pattern Correct […]<\/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\/23411"}],"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=23411"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23411\/revisions"}],"predecessor-version":[{"id":23412,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23411\/revisions\/23412"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23411"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23411"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23411"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}