View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps

 

Question 1. A SIP endpoint sends an INVITE and receives a provisional response indicating that the called party has been alerted. The caller hears ringback generated by the network rather than by the originating endpoint. Which SIP response most commonly indicates this early call-progress condition?

1. 180 Ringing
2. 183 Session Progress
3. 200 OK
4. 486 Busy Here

Correct Answer: 2. 183 Session Progress

Explanation :-

A 183 Session Progress response is a provisional SIP response commonly used to provide early media or additional session-progress information before the final response to an INVITE. It can allow media such as announcements or network-generated ringback to be established before the call is answered. A 180 Ringing response indicates that the destination is being alerted, but it does not by itself establish early media. A 200 OK represents successful final call establishment, while 486 Busy Here indicates that the called endpoint is unable to accept the call. Troubleshooting early ringback therefore requires examining the provisional responses and associated SDP information.

Question 2. During troubleshooting of a SIP call, an engineer observes that the called endpoint does not receive reliable information about provisional responses that require acknowledgment. Which SIP mechanism is specifically designed to provide reliable provisional responses?

1. UPDATE
2. REFER
3. OPTIONS
4. PRACK

Correct Answer: 4. PRACK

Explanation :-

PRACK, or Provisional Response Acknowledgment, provides a mechanism for reliably acknowledging certain provisional SIP responses. It is particularly relevant when reliable provisional response handling is required during call setup. Unlike PRACK, UPDATE is used to modify session information before or during a dialog, REFER is commonly associated with call transfer, and OPTIONS is used to query capabilities or availability. When troubleshooting signaling involving provisional responses, engineers should inspect the SIP messages and determine whether the endpoints and intermediary devices correctly support the required reliability mechanism. Incorrect handling of PRACK can result in interoperability problems during call establishment.

Question 3. A SIP session is already established. One endpoint needs to modify session parameters without terminating the existing dialog. Which SIP method is specifically designed to update session information within an existing dialog?

1. UPDATE
2. REGISTER
3. MESSAGE
4. CANCEL

Correct Answer: 1. UPDATE

Explanation :-

The SIP UPDATE method allows session parameters to be modified within an existing dialog without waiting for a re-INVITE. It can be used during call setup or an established dialog depending on the signaling scenario. REGISTER is used to associate a SIP address-of-record with a contact location, while MESSAGE is used for instant messaging or similar content exchange. CANCEL terminates a pending request before a final response is received. During advanced SIP troubleshooting, an engineer should examine UPDATE messages when session information changes unexpectedly or when interoperability depends on modifying session parameters without creating a new dialog.

Question 4. A remote SIP endpoint is behind a NAT device and cannot receive RTP directly from another endpoint. The organization wants to use a framework that discovers possible media paths and selects a viable path through NAT. Which technology should be considered?

1. SRST
2. SIP OPTIONS
3. ICE
4. MGCP

Correct Answer: 3. ICE

Explanation :-

Interactive Connectivity Establishment, or ICE, is a framework used to discover and evaluate possible media paths between endpoints, including paths that traverse NAT devices. ICE works with mechanisms such as STUN and TURN to determine candidate addresses and connectivity options. STUN can help an endpoint discover its public-facing address, while TURN can provide media relay when direct connectivity is not possible. SRST provides survivable call processing, SIP OPTIONS exchanges capability information, and MGCP is a gateway-control protocol. For a NAT traversal problem involving media-path discovery and selection, ICE is therefore the most relevant technology.

Question 5. A Cisco Unified Communications Manager Express deployment must support SIP phones that register directly to the router. Which configuration area is most important for enabling this registration model?

1. OSPF redistribution
2. HSRP tracking
3. DHCP snooping
4. SIP phone registration and authentication configuration

Correct Answer: 4. SIP phone registration and authentication configuration

Explanation :-

Cisco Unified Communications Manager Express can provide call-processing services for SIP phones, but the phones must be properly configured and permitted to register with the CME system. Registration-related configuration includes the appropriate SIP phone definitions, directory numbers, authentication or credentials where required, and the relevant voice-service settings. OSPF redistribution and HSRP tracking address routing and gateway availability rather than SIP registration. DHCP snooping is a Layer 2 security function and does not establish SIP phone registration. When a SIP phone cannot register with CME, the engineer should first examine registration configuration and the signaling exchanged between the phone and CME.

Question 6. A branch router provides Cisco Unified SRST for remote phones. During a WAN outage, the phones must continue placing calls through the local PSTN gateway. Which SRST capability is directly responsible for maintaining local call-processing availability?

1. Local survivable call processing on the branch router
2. Centralized DNS resolution
3. BGP route reflection
4. Cisco Discovery Protocol

Correct Answer: 1. Local survivable call processing on the branch router

Explanation :-

Cisco Unified SRST provides survivable call-processing functionality when phones lose connectivity to their centralized Cisco Unified Communications Manager. During a WAN failure, supported phones can register with the local branch router and continue using locally configured call-processing and gateway resources. This allows essential calling to continue even though the centralized call-control system is temporarily unreachable. DNS, BGP route reflection, and Cisco Discovery Protocol may support other network functions but do not provide the local call-control function required by SRST. Proper SRST configuration, dial peers, voice-port or SIP gateway settings, and phone registration behavior should be verified when testing branch survivability.

Question 7. A CUBE deployment receives calls from a SIP provider and must select different outbound destinations based on the called number. Which configuration element is primarily used to determine the outbound call-routing decision?

1. SIP registrar
2. Codec preference list
3. Dial peer
4. TLS certificate

Correct Answer: 3. Dial peer

Explanation :-

Cisco Unified Border Element uses dial peers as a fundamental mechanism for determining how calls are routed between voice networks. Destination patterns, URI matching, session targets, and other dial-peer characteristics can determine which outbound path is selected for a call. A codec preference list influences media negotiation rather than the basic destination-routing decision. A SIP registrar handles registration functions, while a TLS certificate supports secure signaling authentication and encryption. When troubleshooting CUBE call routing, engineers should examine dial-peer matching and selection, including destination patterns or URI-based matching, before focusing on media or security parameters.

Question 8. A CUBE router must remove a specific SIP header from an incoming message and insert a different value before forwarding the message to a service provider. Which feature is designed for this type of SIP message manipulation?

1. SRST
2. SIP profile
3. E.164 pattern map
4. Codec preference list

Correct Answer: 2. SIP profile

Explanation :-

SIP profiles allow Cisco voice gateways and CUBE to manipulate selected SIP headers and message elements as calls pass through the device. They can be used when interoperability with a service provider or another SIP system requires a header to be modified, removed, or inserted. E.164 pattern maps are primarily associated with digit-pattern matching, while codec preference lists control codec selection. SRST provides survivable call processing rather than SIP-header manipulation. When troubleshooting an interoperability problem caused by incompatible SIP signaling, the engineer should examine the SIP messages and determine whether a SIP profile is required to transform the signaling into a format expected by the remote system.

Question 9. A SIP trunk successfully establishes calls, but users cannot reliably send DTMF digits to an automated IVR system. Voice quality is otherwise normal. Which area should be investigated first?

1. DTMF relay interoperability
2. OSPF adjacency timers
3. DHCP lease duration
4. SRST registration timeout

Correct Answer: 1. DTMF relay interoperability

Explanation :-

DTMF problems on SIP trunks commonly result from incompatible DTMF signaling methods between endpoints, CUBE, gateways, and service providers. DTMF can be transported using different mechanisms, including RTP-based methods such as RFC 2833/4733, SIP signaling methods, or in-band audio depending on the deployment. If voice calls are established and audio is normal but IVR digit recognition fails, the engineer should inspect the negotiated and configured DTMF relay mechanisms. OSPF timers, DHCP lease duration, and SRST registration settings do not directly control DTMF transport. Interoperability testing should verify that both sides support the selected signaling method.

Question 10. A company wants to ensure that a CUBE router selects a preferred codec when negotiating media with a SIP provider. Which configuration feature is most directly relevant?

1. Voice VRF
2. SIP registrar
3. E.164 pattern map
4. Codec preference list

Correct Answer: 4. Codec preference list

Explanation :-

A codec preference list controls the preferred order of codecs during voice-media negotiation. When multiple codecs are supported by both endpoints, the preference configuration can influence which codec is selected. This is particularly important when an enterprise and service provider support different codec sets or when bandwidth and transcoding requirements must be considered. Voice VRF separates routing contexts, an E.164 pattern map helps match telephone-number patterns, and a SIP registrar handles registration. If a call establishes but uses an undesired codec or fails because of codec incompatibility, the engineer should inspect the codec capabilities and the configured codec preference lists.

Question 11. A CUBE router has separate interfaces for an internal voice network and an external SIP provider. The engineer wants signaling to use one interface and media to use a specific interface. Which CUBE configuration concept is designed to control these paths?

1. SRST fallback
2. Signaling and media bindings
3. OSPF passive interfaces
4. E.164 pattern maps

Correct Answer: 2. Signaling and media bindings

Explanation :-

CUBE signaling and media bindings allow the engineer to associate SIP signaling and RTP media with appropriate interfaces or addresses. This becomes important when a router has multiple interfaces, VRFs, or network paths and the signaling and media must follow specific connectivity requirements. Incorrect bindings can result in one-way audio, unreachable media addresses, or signaling that leaves through an unintended interface. SRST addresses survivable call processing, E.164 pattern maps support digit matching, and OSPF passive interfaces affect routing-protocol participation. For a CUBE design involving distinct internal and external voice paths, signaling and media bindings should be carefully reviewed.

Question 12. A service provider requires encrypted SIP signaling and encrypted media for calls traversing CUBE. Which combination best addresses these requirements?

1. UDP signaling and RTP without encryption
2. OSPF authentication and GRE
3. TLS for SIP signaling and SRTP for media
4. DHCP snooping and IP SLA

Correct Answer: 3. TLS for SIP signaling and SRTP for media

Explanation :-

TLS can provide encryption and authentication for SIP signaling, while Secure Real-time Transport Protocol, or SRTP, protects voice media. Using both mechanisms allows a deployment to protect the signaling exchanged between SIP entities and the actual RTP media carrying the conversation. Plain UDP signaling and ordinary RTP do not provide the same cryptographic protection. OSPF authentication protects routing-protocol exchanges rather than SIP and voice media, while DHCP snooping and IP SLA address unrelated network functions. In a CUBE deployment requiring secure voice communication, engineers should verify TLS profiles, certificates, SRTP negotiation, and compatibility between the connected SIP systems.

Question 13. A CUBE router must support two SIP trunks that use different routing contexts and security requirements. The engineer wants to isolate the traffic associated with each trunk. Which technology can help provide separate voice routing contexts?

1. Voice VRF and multi-VRF configuration
2. CDP
3. DHCP relay
4. NTP authentication

Correct Answer: 1. Voice VRF and multi-VRF configuration

Explanation :-

Voice VRF and multi-VRF capabilities can be used to separate routing contexts for different voice networks or SIP trunks. This is useful when a CUBE deployment connects multiple providers or tenants that require independent routing, addressing, or security policies. By separating routing tables, the administrator can control how traffic associated with each voice environment is forwarded. CDP is a neighbor-discovery protocol, DHCP relay forwards DHCP requests between networks, and NTP authentication protects time synchronization. None of those mechanisms provides the routing isolation required for multiple independent voice connectivity contexts.

Question 14. A SIP call is placed through CUBE using a destination expressed as a SIP URI rather than a traditional digit string. Which dial-peer matching capability is particularly useful for this scenario?

1. URI matching
2. DSP resource allocation
3. SRST fallback
4. Codec transcoding

Correct Answer: 1. URI matching

Explanation :-

URI matching allows CUBE dial peers to make routing decisions based on SIP Uniform Resource Identifiers rather than relying solely on traditional digit-pattern matching. This is useful for business-to-business SIP connectivity and deployments where calls are addressed using domains, usernames, or other URI components. DSP resources and codec transcoding relate to media processing, while SRST provides survivable call processing. When troubleshooting URI-based call routing, engineers should examine the SIP Request-URI, From or To information as appropriate, and the corresponding dial-peer configuration to determine whether the expected URI pattern is being matched.

Question 15. A CUBE deployment must route calls using an E.164 numbering pattern that is reused across several dial peers. The engineer wants to define the pattern once and reference it from multiple dial peers. Which feature is appropriate?

1. SIP profile
2. E.164 pattern map
3. TLS profile
4. Voice VRF

Correct Answer: 2. E.164 pattern map

Explanation :-

An E.164 pattern map provides a reusable method for defining telephone-number patterns that can be referenced by dial peers. This can simplify dial-plan configuration when multiple dial peers need to match the same numbering patterns. SIP profiles manipulate SIP messages, TLS profiles define secure signaling parameters, and Voice VRF provides routing separation. When a CUBE dial plan becomes complex, reusable pattern definitions can reduce duplication and make troubleshooting easier. Engineers should still verify the overall dial-peer selection logic because pattern matching is only one part of the call-routing process.

Question 16. A SIP endpoint places a call successfully, but after the remote party answers, the call is disconnected unexpectedly after a predictable interval. SIP traces show periodic session refresh messages. Which feature should be investigated?

1. DTMF relay
2. E.164 pattern maps
3. Session timers
4. DHCP snooping

Correct Answer: 3. Session timers

Explanation :-

SIP session timers provide a mechanism for periodically refreshing an established session so that endpoints can determine whether the dialog is still active. If refresh requests or responses are not handled correctly, a session may eventually be terminated even though the call initially establishes successfully. During troubleshooting, engineers should inspect Session-Expires and related signaling, determine which endpoint acts as the refresher, and verify that intermediary devices correctly process the refresh messages. DTMF relay affects digit transport, E.164 pattern maps affect dial-plan matching, and DHCP snooping is unrelated. Predictable call termination after establishment makes session-timer behavior particularly important to investigate.

Question 17. A user places a call and puts the remote party on hold. The call resumes, but the media stream uses different RTP information than before the hold operation. Which SIP behavior should the engineer investigate?

1. REGISTER expiration
2. Mid-call signaling and SDP renegotiation
3. DHCP address allocation
4. BGP route reflection

Correct Answer: 2. Mid-call signaling and SDP renegotiation

Explanation :-

Hold and resume operations can require mid-call SIP signaling and changes to the SDP describing the media session. Depending on the implementation, a re-INVITE or UPDATE can communicate new media information, including changes to connection addresses, ports, directions, or media state. Engineers troubleshooting hold and resume should therefore inspect the SIP messages and compare the SDP before and after the operation. REGISTER controls endpoint registration, DHCP provides IP addressing, and BGP route reflection is unrelated to SIP media renegotiation. Problems such as one-way audio after resume can often be traced to incorrect SDP or media-path handling.

Question 18. A company wants an external SIP business partner to communicate with its internal Unified Communications environment without exposing internal call-control infrastructure directly to the Internet. Which component is commonly deployed at the network boundary for this purpose?

1. Cisco Unified Communications Manager Express only
2. Cisco Unified Border Element
3. DHCP server
4. OSPF route reflector

Correct Answer: 2. Cisco Unified Border Element

Explanation :-

Cisco Unified Border Element, or CUBE, commonly operates at the boundary between an enterprise collaboration environment and an external SIP network. It can provide SIP interworking, signaling and media control, security features, dial-plan manipulation, and topology-related functions between the internal collaboration environment and external service providers or business partners. Deploying an appropriate border element helps avoid exposing internal call-control systems directly to external networks. CME provides local call control, while DHCP and OSPF provide network services unrelated to SIP border control. CUBE should be designed with appropriate security and toll-fraud protections when exposed to external SIP connectivity.

Question 19. During a SIP call-transfer investigation, an engineer observes a SIP REFER message sent by one endpoint. What operation is the REFER method commonly associated with?

1. Requesting another SIP endpoint to initiate a referenced operation, such as call transfer
2. Registering a contact address
3. Establishing an IPsec security association
4. Negotiating an RTP codec list

Correct Answer: 1. Requesting another SIP endpoint to initiate a referenced operation, such as call transfer

Explanation :-

The SIP REFER method is commonly used to request that another SIP entity perform a referenced operation, particularly in call-transfer scenarios. A device can use REFER to instruct another endpoint to contact a specified destination, depending on the supported transfer model and signaling behavior. REGISTER is used for SIP registration, IPsec security associations are negotiated by VPN protocols rather than SIP REFER, and codec negotiation is represented through SDP offer/answer exchanges. When troubleshooting attended or blind transfer behavior, the engineer should examine REFER messages and the subsequent NOTIFY or related SIP signaling to determine whether the requested operation completed correctly.

Question 20. A CUBE administrator wants to reduce the risk of unauthorized toll calls originating through the enterprise voice gateway. Which security practice is most directly relevant?

1. Increasing the OSPF dead interval
2. Disabling all SIP responses
3. Implementing toll-fraud prevention controls and restricting call routing
4. Increasing the RTP packet size

Correct Answer: 3. Implementing toll-fraud prevention controls and restricting call routing

Explanation :-

Toll fraud can occur when unauthorized users exploit voice gateways or SIP services to place expensive calls. CUBE and related voice infrastructure should therefore use controls that restrict which sources can place calls, which destinations are permitted, and which signaling relationships are trusted. Appropriate dial-plan restrictions, authentication, access controls, trusted-source policies, and other toll-fraud prevention mechanisms can reduce exposure. OSPF timers and RTP packet size do not address unauthorized call origination, while disabling SIP responses would disrupt legitimate communication. Security should be incorporated into the dial-plan and SIP-border design rather than treated as an afterthought.