{"id":23431,"date":"2026-09-28T06:34:45","date_gmt":"2026-09-28T06:34:45","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23431"},"modified":"2026-09-28T06:34:45","modified_gmt":"2026-09-28T06:34:45","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-15-q281-300","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-15-q281-300\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 15 Q281-300"},"content":{"rendered":"

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

 <\/p>\n

Question 281. A Cisco Unified Communications Manager SIP trunk sends an INVITE toward a remote SIP endpoint. The endpoint returns 183 Session Progress with SDP before the call is answered. What is the primary purpose of this response?<\/b><\/p>\n

    \n
  1. <\/b> To terminate the SIP dialog before answer<\/span><\/li>\n
  2. <\/b> To request a new SIP registration<\/span><\/li>\n
  3. <\/b> To establish early media before the final response<\/span><\/li>\n
  4. <\/b> To indicate that the remote endpoint has answered the call<\/span><\/li>\n<\/ol>\n

    Correct Answer: 3. To establish early media before the final response<\/b><\/p>\n

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

    A SIP 183 Session Progress response can carry SDP and is commonly used to establish early media while the call is still progressing. Early media allows audio, such as ringback, announcements, or network-generated tones, to be transmitted before the called party sends a final 200 OK response. This is different from a 180 Ringing response, which generally indicates alerting but does not necessarily establish media. In troubleshooting, receiving a 183 with SDP can therefore explain why media is heard before call answer. The response does not mean that the remote party has answered, because the final call acceptance normally occurs with a 200 OK response.<\/span><\/p>\n

    Question 282. During SIP call setup, a UCM administrator wants a reliable provisional response mechanism for a SIP endpoint that supports 100rel. Which SIP method is associated with acknowledging reliable provisional responses?<\/b><\/p>\n

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

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

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

      PRACK, or Provisional Response ACKnowledgment, is used to acknowledge reliable provisional SIP responses. When 100rel is supported and required, provisional responses such as 183 Session Progress can be transmitted reliably instead of relying solely on the normal SIP retransmission behavior. The PRACK request provides confirmation that the provisional response was received. This mechanism is particularly relevant when early media, resource reservation, or other signaling information must be delivered reliably before the final call response. UPDATE serves a different purpose, such as modifying session information during an established SIP dialog, while BYE terminates a session and REFER is commonly associated with call transfer or referral operations.<\/span><\/p>\n

      Question 283. A SIP call is placed on hold and then resumed. Which type of SIP signaling is most directly associated with communicating this mid-call session change?<\/b><\/p>\n

        \n
      1. <\/b> REGISTER<\/span><\/li>\n
      2. <\/b> OPTIONS<\/span><\/li>\n
      3. <\/b> CANCEL<\/span><\/li>\n
      4. <\/b> Re-INVITE or UPDATE<\/span><\/li>\n<\/ol>\n

        Correct Answer: 4. Re-INVITE or UPDATE<\/b><\/p>\n

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

        SIP uses mid-call signaling to communicate changes to an existing session. A re-INVITE can be used to renegotiate session parameters, including changes associated with putting a call on hold or resuming it. Depending on the implementation and scenario, UPDATE can also modify session information within an established dialog. Hold is commonly represented through SDP changes, such as placing the media stream in a send-only, inactive, or equivalent state. REGISTER is used for endpoint registration, OPTIONS is generally used for capability or availability checks, and CANCEL terminates an outstanding request before it receives a final response. Therefore, re-INVITE or UPDATE is the relevant signaling category for these mid-call changes.<\/span><\/p>\n

        Question 284. A SIP endpoint sends an INVITE but does not receive a response within the expected interval. Which SIP mechanism helps control how long the transaction waits and how retransmissions are handled?<\/b><\/p>\n

          \n
        1. <\/b> SIP session timer<\/span><\/li>\n
        2. <\/b> RTP payload type<\/span><\/li>\n
        3. <\/b> SIP transaction timers and retransmission behavior<\/span><\/li>\n
        4. <\/b> SDP codec preference<\/span><\/li>\n<\/ol>\n

          Correct Answer: 3. SIP transaction timers and retransmission behavior<\/b><\/p>\n

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

          SIP transaction timers determine how long a SIP transaction waits for responses and how requests or responses are retransmitted under applicable transport conditions. They are different from SIP session timers, which are used to refresh an established dialog and detect stale sessions. When troubleshooting an INVITE that appears to disappear or repeatedly retransmit, transaction timing and retransmission behavior should be examined. SDP codec preferences determine media capabilities rather than signaling transaction timing. RTP payload types identify media formats. Understanding SIP timers is important because apparent call failures can result from delayed responses, unreachable destinations, or timer expiration rather than an incorrect dial plan alone.<\/span><\/p>\n

          Question 285. A SIP call connects successfully, but the caller cannot hear DTMF tones when navigating an IVR. Which area should be investigated first?<\/b><\/p>\n

            \n
          1. <\/b> SIP registration expiration<\/span><\/li>\n
          2. <\/b> DTMF signaling and negotiation<\/span><\/li>\n
          3. <\/b> SIP URI synchronization<\/span><\/li>\n
          4. <\/b> Device pool date and time settings<\/span><\/li>\n<\/ol>\n

            Correct Answer: 2. DTMF signaling and negotiation<\/b><\/p>\n

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

            When voice calls work but an IVR does not recognize keypad presses, DTMF signaling should be investigated. SIP collaboration environments can transport DTMF using methods such as RTP-NTE, SIP INFO, or other negotiated mechanisms depending on the endpoint and trunk configuration. The selected method must be compatible across the signaling path. Codec negotiation alone does not guarantee correct DTMF operation. Administrators should examine the SIP messages, SDP information, dial-peer configuration where applicable, and DTMF settings on connected systems. Registration timers, URI synchronization, and device pool time settings do not normally explain a condition where the voice path works but DTMF digits fail to reach the IVR application.<\/span><\/p>\n

            Question 286. A Cisco Unified CME router must provide local SIP phone registration for several endpoints. Which configuration concept is required to enable CME to act as the SIP registrar for those phones?<\/b><\/p>\n

              \n
            1. <\/b> A SIP registrar configuration under voice service<\/span><\/li>\n
            2. <\/b> A CUBE tenant for each phone<\/span><\/li>\n
            3. <\/b> A media termination point for each phone<\/span><\/li>\n
            4. <\/b> A route list containing each extension<\/span><\/li>\n<\/ol>\n

              Correct Answer: 1. A SIP registrar configuration under voice service<\/b><\/p>\n

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

              Cisco Unified CME can provide SIP registration services for supported SIP phones. The CME configuration must establish the router as the registrar and define the appropriate authentication and registration parameters. The SIP phones then register their directory identities with CME, allowing CME to maintain the registration information used for call routing. CUBE tenants are associated with trunk and multi-tenant border-element designs rather than basic local phone registration. Media termination points are used for specific media-processing requirements, and route lists are Cisco Unified CM dial-plan constructs rather than the mechanism that makes CME a SIP registrar. Proper registrar configuration is therefore fundamental when deploying SIP endpoints on CME.<\/span><\/p>\n

              Question 287. A branch router running Cisco Unified CME has several phones whose users must be prevented from placing certain external calls. Which CME capability should be used to control which destinations users are permitted to dial?<\/b><\/p>\n

                \n
              1. <\/b> SRST fallback<\/span><\/li>\n
              2. <\/b> SIP OPTIONS<\/span><\/li>\n
              3. <\/b> Calling privileges and dial-plan restrictions<\/span><\/li>\n
              4. <\/b> RTP codec negotiation<\/span><\/li>\n<\/ol>\n

                Correct Answer: 3. Calling privileges and dial-plan restrictions<\/b><\/p>\n

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

                Cisco Unified CME can implement calling privileges to control which users or phones can access particular destinations. COR-like calling restrictions and related dial-plan mechanisms can separate permitted and restricted call paths. This is useful when some users should be allowed to place local calls while others can access long-distance or international destinations. SRST is designed to provide survivability during loss of connectivity to Cisco Unified Communications Manager and is not primarily an authorization mechanism. SIP OPTIONS is a signaling capability, while RTP codec negotiation concerns media compatibility. Calling privileges should therefore be examined when the requirement is to control which destinations particular CME users or phones can dial.<\/span><\/p>\n

                Question 288. During WAN connectivity loss, a branch must continue providing basic IP telephony services. The gateway should preserve local call processing and provide survivability for registered phones. Which technology addresses this requirement?<\/b><\/p>\n

                  \n
                1. <\/b> SIP REFER<\/span><\/li>\n
                2. <\/b> Cisco SRST<\/span><\/li>\n
                3. <\/b> Cisco Unified Mobility<\/span><\/li>\n
                4. <\/b> Cisco Extension Mobility<\/span><\/li>\n<\/ol>\n

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

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

                  Cisco Unified Survivable Remote Site Telephony, or SRST, provides call-processing survivability at a remote site when the normal connection to Cisco Unified Communications Manager becomes unavailable. A Cisco IOS gateway can take over selected call-control functions so supported phones can continue making and receiving calls during the outage. SRST is particularly important in branch deployments where WAN failure should not completely eliminate voice service. Cisco Unified Mobility instead focuses on extending reachability to mobile or remote destinations, while Extension Mobility allows users to log into supported phones. SIP REFER is a call-control mechanism used for referral or transfer scenarios and does not provide branch call-processing survivability.<\/span><\/p>\n

                  Question 289. A CUBE deployment receives calls from multiple service providers. The administrator needs separate SIP trunk behavior and configuration contexts for different providers on the same CUBE. Which feature is designed for this requirement?<\/b><\/p>\n

                    \n
                  1. <\/b> SIP UPDATE<\/span><\/li>\n
                  2. <\/b> Local Route Group<\/span><\/li>\n
                  3. <\/b> SRST<\/span><\/li>\n
                  4. <\/b> CUBE tenants<\/span><\/li>\n<\/ol>\n

                    Correct Answer: 4. CUBE tenants<\/b><\/p>\n

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

                    CUBE tenants provide a mechanism for separating configuration and behavior associated with multiple trunks or service-provider environments on a single Cisco Unified Border Element. This can be useful when different providers require distinct dial plans, signaling parameters, security controls, or other trunk-specific behavior. A tenant-based design can help prevent one provider’s configuration from unintentionally affecting another. Local Route Group is a Cisco Unified Communications Manager routing construct, while SRST provides survivability and SIP UPDATE is a signaling method. When multiple independent trunk environments must coexist on one CUBE, tenant configuration is therefore a relevant design and troubleshooting area.<\/span><\/p>\n

                    Question 290. A CUBE administrator needs to change the calling-party number format before a call is sent to a carrier. Which CUBE feature is specifically designed to manipulate telephone numbers?<\/b><\/p>\n

                      \n
                    1. <\/b> Codec preference list<\/span><\/li>\n
                    2. <\/b> Voice translation rules and profiles<\/span><\/li>\n
                    3. <\/b> SIP session timer<\/span><\/li>\n
                    4. <\/b> Media flow-through<\/span><\/li>\n<\/ol>\n

                      Correct Answer: 2. Voice translation rules and profiles<\/b><\/p>\n

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

                      Voice translation rules and profiles are used to manipulate calling and called numbers in Cisco IOS voice configurations, including CUBE deployments. A translation rule defines the matching and replacement behavior, while the associated translation profile determines where that transformation is applied, such as to incoming or outgoing calling or called numbers. This makes translation profiles useful when a provider expects a different numbering format from the format used internally. Codec preference lists address media codec selection, SIP session timers concern dialog refresh and expiration, and media flow-through affects media handling. Therefore, when the requirement specifically involves changing telephone number presentation, voice translation rules and profiles are the appropriate mechanism.<\/span><\/p>\n

                      Question 291. A CUBE receives an SDP offer using a codec that is not preferred by the enterprise. The administrator wants CUBE to select from an ordered list of acceptable codecs. Which feature should be configured?<\/b><\/p>\n

                        \n
                      1. <\/b> Codec preference list<\/span><\/li>\n
                      2. <\/b> SIP registrar<\/span><\/li>\n
                      3. <\/b> Voice translation profile<\/span><\/li>\n
                      4. <\/b> Dial peer huntstop<\/span><\/li>\n<\/ol>\n

                        Correct Answer: 1. Codec preference list<\/b><\/p>\n

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

                        A codec preference list allows a Cisco voice gateway or CUBE configuration to define the preferred order of codecs for call negotiation. When multiple compatible codecs are available, the configured preference influences which codec is selected for the call. This is particularly useful when an enterprise wants to prioritize a particular codec for bandwidth, quality, interoperability, or operational reasons. Voice translation profiles manipulate telephone numbers rather than codec selection. A SIP registrar handles endpoint registration, and dial-peer huntstop controls dial-peer hunting behavior. Codec negotiation also depends on what codecs the remote side offers, so the preference list does not force a codec that the other endpoint does not support.<\/span><\/p>\n

                        Question 292. A company uses CUBE between UCM and a service provider. The provider requires a specific SIP header to be modified before outbound calls are accepted. Which CUBE capability can perform this signaling-header modification?<\/b><\/p>\n

                          \n
                        1. <\/b> SRST<\/span><\/li>\n
                        2. <\/b> DTMF relay<\/span><\/li>\n
                        3. <\/b> SIP profiles<\/span><\/li>\n
                        4. <\/b> RTP codec transcoding<\/span><\/li>\n<\/ol>\n

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

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

                          Cisco CUBE SIP profiles provide a mechanism for manipulating SIP messages, including selected SIP headers and message elements, when interoperability requirements differ between connected systems. This is useful when a service provider expects a particular header value, format, or signaling behavior that does not match what UCM normally sends. SIP profiles can therefore address protocol interoperability without requiring the application or endpoint configuration to be completely redesigned. DTMF relay handles digit signaling, SRST provides survivability, and codec transcoding changes media formats. When troubleshooting a provider rejection caused by an unexpected SIP header, examining the SIP messages and applying an appropriate SIP profile is a logical approach.<\/span><\/p>\n

                          Question 293. A CUBE administrator wants to prevent unauthorized external callers from using the organization’s gateway to place outbound toll calls. Which CUBE security consideration directly addresses this threat?<\/b><\/p>\n

                            \n
                          1. <\/b> Toll fraud prevention<\/span><\/li>\n
                          2. <\/b> Codec preference<\/span><\/li>\n
                          3. <\/b> SDP c-line modification<\/span><\/li>\n
                          4. <\/b> SIP keepalive tuning<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 1. Toll fraud prevention<\/b><\/p>\n

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

                            Toll fraud occurs when unauthorized users exploit voice infrastructure to place calls that generate charges for the organization. CUBE deployments should use appropriate dial-plan restrictions, trusted signaling sources, authentication or access controls where applicable, and explicit routing policies to reduce this risk. Toll fraud prevention is therefore a core operational concern when a CUBE is exposed to external signaling or interconnected with untrusted environments. Codec preferences and SDP manipulation address media negotiation rather than authorization. SIP keepalives can help monitor signaling relationships but do not themselves prevent unauthorized call origination. A secure CUBE design should ensure that only intended call sources can reach permitted outbound dial peers and destinations.<\/span><\/p>\n

                            Question 294. An enterprise has UCM clusters in multiple geographic locations and wants users to dial remote users using normalized directory numbers while maintaining consistent routing across clusters. Which Cisco UCM capability is specifically intended to distribute global dial-plan information between clusters?<\/b><\/p>\n

                              \n
                            1. <\/b> Extension Mobility<\/span><\/li>\n
                            2. <\/b> Global Dial Plan Replication<\/span><\/li>\n
                            3. <\/b> Call Park<\/span><\/li>\n
                            4. <\/b> Local Route Group<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 2. Global Dial Plan Replication<\/b><\/p>\n

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

                              Global Dial Plan Replication is designed to distribute dial-plan information between Cisco Unified Communications Manager clusters so that users and administrators can maintain consistent knowledge of remote destinations. This supports multisite deployments where directory numbers and routing information exist across separate clusters. The feature works with globalized numbering concepts to help establish predictable intercluster call routing. Extension Mobility concerns user login and phone profiles, Call Park provides temporary call parking, and Local Route Group provides routing flexibility within UCM dial-plan designs. When troubleshooting why one cluster does not recognize or properly route to destinations known by another cluster, global dial-plan information and its replication status should be examined.<\/span><\/p>\n

                              Question 295. A company wants users to place calls using SIP URIs such as user@example.com instead of traditional numeric directory numbers. Which Cisco UCM dial-plan element is most directly associated with matching SIP URI destinations?<\/b><\/p>\n

                                \n
                              1. <\/b> Route Group<\/span><\/li>\n
                              2. <\/b> Translation Pattern<\/span><\/li>\n
                              3. <\/b> SIP Route Pattern<\/span><\/li>\n
                              4. <\/b> Hunt Pilot<\/span><\/li>\n<\/ol>\n

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

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

                                SIP Route Patterns are used in Cisco Unified Communications Manager to route calls based on SIP URI information. URI-based calling allows users to address destinations using formats such as user@example.com rather than relying exclusively on numeric telephone numbers. The SIP route pattern provides the matching logic required to determine how a URI should be routed. Route Groups contain gateways or trunks used for routing, translation patterns transform or manipulate dialed digits, and hunt pilots provide call-distribution behavior. In a multisite collaboration environment, URI routing can simplify interorganization and intercluster addressing when the associated SIP trunks, domains, and routing configuration are correctly established.<\/span><\/p>\n

                                Question 296. A UCM administrator needs to ensure that emergency or priority calls can be routed through an alternate path when the normal gateway is unavailable. Which dial-plan concept can provide an alternate route by defining multiple route groups under a route list?<\/b><\/p>\n

                                  \n
                                1. <\/b> Route List<\/span><\/li>\n
                                2. <\/b> Translation Pattern<\/span><\/li>\n
                                3. <\/b> Directory Number<\/span><\/li>\n
                                4. <\/b> SIP Profile<\/span><\/li>\n<\/ol>\n

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

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

                                  A Cisco Unified Communications Manager Route List provides an ordered collection of Route Groups that can be used for call routing. When the primary Route Group cannot complete the call, UCM can attempt another configured group according to the route-list design and applicable routing rules. This makes route lists useful when redundancy or alternate gateway selection is required. A translation pattern changes or transforms the dialed digits and does not itself provide gateway redundancy. A directory number represents an endpoint address, while a SIP profile controls selected SIP behavior. Administrators should therefore examine route lists and their associated route groups when implementing or troubleshooting alternate outbound routing paths.<\/span><\/p>\n

                                  Question 297. A user receives a call on a desk phone and wants the same call to be extended to a configured mobile destination while maintaining the ability to use UCM mobility features. Which UCM feature provides this capability?<\/b><\/p>\n

                                    \n
                                  1. <\/b> Call Park<\/span><\/li>\n
                                  2. <\/b> Unified Mobility<\/span><\/li>\n
                                  3. <\/b> Extension Mobility<\/span><\/li>\n
                                  4. <\/b> Hunt Pilot<\/span><\/li>\n<\/ol>\n

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

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

                                    Cisco Unified Mobility extends a user’s enterprise calling experience beyond the desk phone by allowing configured mobile destinations to participate in call handling. Depending on the deployment, a call can be extended to a user’s mobile phone while UCM continues to manage the enterprise call context. Unified Mobility is therefore associated with reachability and mobility between enterprise and mobile endpoints. Extension Mobility serves a different purpose: it allows users to log into supported Cisco phones and obtain their configured device profile. Call Park temporarily places a call in a parking slot, while a Hunt Pilot distributes calls among configured members. These distinctions are important when troubleshooting mobile call behavior.<\/span><\/p>\n

                                    Question 298. An administrator configures Extension Mobility but users report that their personal phone profiles do not appear when they log in. Which area should be checked first?<\/b><\/p>\n

                                      \n
                                    1. <\/b> CUBE codec preference<\/span><\/li>\n
                                    2. <\/b> SIP DTMF relay<\/span><\/li>\n
                                    3. <\/b> Extension Mobility configuration and user-device association<\/span><\/li>\n
                                    4. <\/b> SRST dial peers<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 3. Extension Mobility configuration and user-device association<\/b><\/p>\n

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

                                      Extension Mobility depends on correct configuration of the Extension Mobility service, user credentials, device profiles, and the association between users and their permitted devices or profiles. If a user can reach the login service but does not see the expected personal configuration, administrators should verify that the correct device profile exists and is associated with the user, and that the phone is configured to support Extension Mobility. CUBE codec settings and DTMF relay affect voice signaling or media behavior, not the profile-login process. SRST dial peers are relevant to survivable gateway call routing. Troubleshooting should therefore begin with the UCM Extension Mobility and user-device configuration.<\/span><\/p>\n

                                      Question 299. A branch connected to UCM experiences WAN congestion, and calls to a remote site are being rejected because available bandwidth cannot support additional calls. Which UCM mechanism is intended to control call admission based on configured bandwidth resources?<\/b><\/p>\n

                                        \n
                                      1. <\/b> Call Admission Control<\/span><\/li>\n
                                      2. <\/b> Extension Mobility<\/span><\/li>\n
                                      3. <\/b> SIP REFER<\/span><\/li>\n
                                      4. <\/b> Global Dial Plan Replication<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 1. Call Admission Control<\/b><\/p>\n

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

                                        Cisco Unified Communications Manager Call Admission Control, or CAC, helps control whether calls are allowed across constrained network links based on configured bandwidth and location relationships. Its purpose is to prevent excessive voice traffic from consuming available WAN resources and degrading active calls. When the configured bandwidth allocation cannot support another call, CAC can prevent the call from being admitted across the constrained path. Extension Mobility manages user profiles, SIP REFER supports call referral or transfer operations, and Global Dial Plan Replication distributes dial-plan information between clusters. CAC should therefore be examined when calls are being rejected specifically because configured network bandwidth resources are insufficient.<\/span><\/p>\n

                                        Question 300. A UCM deployment uses a SIP trunk between two sites. Calls are established, but media is one-way. Signaling appears correct, and SDP contains private IP addresses that are not reachable from the remote site. What should be investigated first?<\/b><\/p>\n

                                          \n
                                        1. <\/b> Extension Mobility login parameters<\/span><\/li>\n
                                        2. <\/b> Route-list digit manipulation<\/span><\/li>\n
                                        3. <\/b> SIP URI synchronization<\/span><\/li>\n
                                        4. <\/b> Media path and SDP address reachability<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 4. Media path and SDP address reachability<\/b><\/p>\n

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

                                          One-way media with apparently successful SIP signaling often indicates a media-path problem rather than a basic call-routing failure. SDP advertises the IP addresses and ports that endpoints or media resources should use for RTP. If an unreachable private address is advertised across a routed or NAT boundary, the remote side may send RTP to an address it cannot reach. Troubleshooting should therefore examine the SDP offer and answer, routing, NAT, firewall policies, media flow, and any CUBE or media-resource behavior affecting the RTP path. SIP signaling can complete successfully even when RTP cannot traverse the network, so signaling success alone does not confirm two-way audio.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 281. A Cisco Unified Communications Manager SIP trunk sends an INVITE toward a remote SIP endpoint. The endpoint returns 183 Session Progress with SDP before the call is answered. What is the primary purpose of this response? To terminate the SIP […]<\/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\/23431"}],"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=23431"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23431\/revisions"}],"predecessor-version":[{"id":23432,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23431\/revisions\/23432"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23431"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23431"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23431"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}