Implementing Cisco IP Telephony and Video Part 2

405 Questions and Answers

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The Implementing Cisco IP Telephony and Video Part 2 (CIPTV2) Practice Exam is a targeted resource for IT and voice professionals preparing for the Cisco 300-075 certification. This exam focuses on advanced IP telephony and video solutions in a multi-site deployment, including inter-site dial plans, call control discovery, and conferencing.

Designed to match real exam standards, this practice test features comprehensive, scenario-based questions that challenge your ability to implement and troubleshoot enterprise-level collaboration environments. Every answer includes a detailed explanation to reinforce your understanding of critical VoIP and video technologies.

Topics Covered:

 

  • Globalized call routing and inter-site dialing

  • Call Control Discovery (CCD) and SAF configuration

  • Cisco Unified Border Element (CUBE) and SIP trunking

  • Call admission control (CAC) and URI-based dialing

  • ILS and GDPR in multi-cluster environments

  • Media resource management and conferencing

  • QoS considerations for voice and video

  • Troubleshooting multi-site IP telephony deployments

This exam is ideal for collaboration engineers, voice administrators, and network professionals tasked with designing, implementing, and supporting Cisco Unified Communications in complex enterprise networks.

Sample Questions and Answers

What does the SIP message “180 Ringing” indicate?

A) The called party is being alerted
B) The call is being processed silently
C) The user is unavailable
D) Media is not negotiated

Answer: A) The called party is being alerted
Explanation: “180 Ringing” signals that the called device is being alerted.

What is required to support SIP URI routing on Cisco CUBE?

A) Dial peers using destination URI match
B) MGCP fallback enabled
C) SRTP profiles set
D) RSVP Agent license

Answer: A) Dial peers using destination URI match
Explanation: URI routing on CUBE needs proper dial-peer configuration based on URI patterns.

Which parameter in CUCM helps define a unique user identity for URI dialing?

A) Directory URI
B) Calling Search Space
C) Device Pool
D) Partition

Answer: A) Directory URI
Explanation: The Directory URI defines the user’s email-like identity used in URI dialing.

How does RSVP support CAC in voice networks?

A) Reserves bandwidth before establishing a call
B) Disables redundant trunks
C) Compresses video streams
D) Encrypts SIP messages

Answer: A) Reserves bandwidth before establishing a call
Explanation: RSVP ensures that required resources are reserved before the call proceeds.

What is the function of SIP OPTIONS messages?

A) Used for SIP trunk keepalive
B) Transmit RTP streams
C) Used for failover signaling
D) Block codec negotiation

Answer: A) Used for SIP trunk keepalive
Explanation: SIP OPTIONS is a method to verify remote system availability.

What happens if a SIP trunk security profile requires TLS but the peer supports only TCP?

A) Trunk fails to register or make calls
B) Fallback to TCP automatically
C) Delayed offer is enforced
D) Route pattern is overridden

Answer: A) Trunk fails to register or make calls
Explanation: Mismatched transport types prevent SIP trunk communication.

What is a key feature of URI-based routing in CUCM?

A) Allows email-style address dialing
B) Enables inter-site redundancy
C) Prevents TFTP downloads
D) Supports only analog gateways

Answer: A) Allows email-style address dialing
Explanation: URI dialing enables calling users by standardized names like “[email protected]”.

In which scenario is Media Termination Point (MTP) required?

A) When SIP Early Offer is enabled but device cannot process it
B) When phones are in the same VLAN
C) When using MGCP for call routing
D) When DTMF tones are blocked

Answer: A) When SIP Early Offer is enabled but device cannot process it
Explanation: MTP is needed to handle codec or signaling incompatibilities in SIP call flows.

What does “Delayed Offer” mean in SIP?

A) SDP is sent after receiving a SIP INVITE
B) RTP is sent immediately
C) The call is denied
D) DTMF is sent out-of-band

Answer: A) SDP is sent after receiving a SIP INVITE
Explanation: In Delayed Offer, media details (SDP) are included in the 200 OK message instead of the INVITE.

Which Cisco tool can validate SIP signaling and media streams?

A) RTMT
B) UCCX
C) Unity Connection
D) TFTP

Answer: A) RTMT
Explanation: RTMT (Real-Time Monitoring Tool) provides diagnostics for signaling, registration, and media paths.

What is the maximum number of Directory URIs allowed per user in CUCM?

A) 5
B) 1
C) 10
D) Unlimited

Answer: C) 10
Explanation: CUCM supports up to 10 Directory URIs per user for advanced routing options.

What is required to support URI dialing in Cisco Jabber?

A) Directory URI configured on user profile
B) Static IP addressing
C) SIP domain blacklisting
D) MGCP trunking

Answer: A) Directory URI configured on user profile
Explanation: URI dialing in Jabber depends on proper user-level URI setup.

What does the SAF Call Control Discovery (CCD) feature do?

A) Advertises CUCM patterns dynamically via routing protocols
B) Provides voicemail to Jabber
C) Creates MGCP routes
D) Secures DTMF transmission

Answer: A) Advertises CUCM patterns dynamically via routing protocols
Explanation: CCD allows CUCM to advertise and discover call routing information across a network.

What is a main benefit of SIP Normalization?

A) Customizes SIP headers for interoperability
B) Removes all SIP headers
C) Converts RTP to TCP
D) Eliminates need for dial-peers

Answer: A) Customizes SIP headers for interoperability
Explanation: Normalization scripts modify SIP messages to ensure compatibility between vendors.

 

What is the main purpose of Intercluster Lookup Service (ILS)?

A) Synchronize user voicemails across clusters
B) Route media between Expressway-E pairs
C) Share user and URI dialing information across clusters
D) Encrypt all SIP signaling

Answer: C) Share user and URI dialing information across clusters
Explanation: ILS facilitates the discovery and exchange of directory URIs and user information between CUCM clusters.

Which of the following must be configured to support ILS in a multi-cluster environment?

A) Hub and spoke topology with ILS trust
B) Expressway-C configuration
C) SRST fallback rules
D) TFTP multicast zones

Answer: A) Hub and spoke topology with ILS trust
Explanation: ILS requires a hub-and-spoke or mesh topology with proper trust relationships.

What is the role of the SAF Forwarder in CCD?

A) Registers route patterns with the routing protocol
B) Encodes video streams for telepresence
C) Assigns IP addresses dynamically
D) Provides PSTN backup

Answer: A) Registers route patterns with the routing protocol
Explanation: SAF Forwarders advertise learned CUCM route patterns to other routers using EIGRP.

What is a benefit of SIP Trunk Early Offer?

A) Reduces call setup time
B) Prevents codec negotiation
C) Delays media establishment
D) Eliminates need for MTP

Answer: A) Reduces call setup time
Explanation: Early Offer sends media capabilities in the INVITE, which can speed up call setup.

In a centralized call processing model, what ensures local survivability?

A) SRST
B) MGCP gateway
C) TFTP proxy
D) ILS

Answer: A) SRST
Explanation: Survivable Remote Site Telephony (SRST) provides limited call control during WAN failure.

What does a SIP “183 Session Progress” message typically indicate?

A) Call setup is in progress and media can begin
B) Final acceptance of the call
C) A redirect to a new URI
D) Codec negotiation failure

Answer: A) Call setup is in progress and media can begin
Explanation: SIP 183 allows early media without full call establishment.

What does the CUCM feature “Enhanced Location CAC” offer?

A) Video bandwidth enforcement
B) Alternate routing to backup trunks
C) Static routing tables
D) Enhanced voice prompt customization

Answer: A) Video bandwidth enforcement
Explanation: Enhanced CAC supports bandwidth control for audio, video, and immersive video streams.

What is a requirement for SIP Normalization scripting?

A) TCL interpreter support in CUCM
B) Java runtime on voice gateways
C) External syslog support
D) Use of Unity Connection

Answer: A) TCL interpreter support in CUCM
Explanation: Normalization scripts are based on Cisco’s TCL interpreter and run natively in CUCM.

Why is a Media Termination Point (MTP) needed in some SIP trunk scenarios?

A) To handle DTMF or codec negotiation mismatches
B) To register softphones
C) To terminate MGCP call legs
D) To provide email integration

Answer: A) To handle DTMF or codec negotiation mismatches
Explanation: MTPs bridge differences in signaling and codec capabilities between endpoints.

What is a “route pattern” used for in CUCM?

A) Match dialed numbers to route calls
B) Translate video codecs
C) Monitor bandwidth usage
D) Configure Expressway firewall rules

Answer: A) Match dialed numbers to route calls
Explanation: Route patterns define how CUCM matches dialed digits and determines call routing.

What is the function of a “Transformation Pattern” in CUCM?

A) Change calling or called number before routing
B) Encrypt RTP streams
C) Modify codec lists for devices
D) Block SRST fallback

Answer: A) Change calling or called number before routing
Explanation: Transformation patterns allow number manipulation for routing or display purposes.

Which CUCM component controls digit analysis for calls?

A) Call Routing Table
B) Dial Plan Analyzer
C) Call Control Engine
D) Route List

Answer: C) Call Control Engine
Explanation: The Call Control Engine in CUCM performs digit analysis to determine call paths.

How is redundancy achieved in SAF CCD?

A) Configuring multiple SAF Forwarders
B) Adding multiple TFTP servers
C) Creating SIP clusters
D) Setting multiple user profiles

Answer: A) Configuring multiple SAF Forwarders
Explanation: Redundancy is provided by deploying more than one SAF Forwarder.

What is a “Learned Pattern” in CCD?

A) A pattern received from another CUCM cluster via SAF
B) A route pattern based on MGCP
C) A codec preference setting
D) A static pattern used for TFTP

Answer: A) A pattern received from another CUCM cluster via SAF
Explanation: CCD allows CUCMs to advertise and learn patterns from other clusters.

What happens if a SIP trunk is misconfigured with the wrong destination IP?

A) Calls will fail with unreachable errors
B) Calls are rerouted automatically
C) CUCM disables the trunk permanently
D) SRST becomes active

Answer: A) Calls will fail with unreachable errors
Explanation: SIP trunks require correct destination IPs for call setup.

What is the use of CSS in SIP trunk configuration?

A) Controls which route patterns are accessible
B) Assigns a codec list
C) Defines VLAN mapping
D) Specifies encryption profiles

Answer: A) Controls which route patterns are accessible
Explanation: Calling Search Spaces (CSS) determine which partitions and routes a SIP trunk can use.

What must be true to use URI dialing in Jabber?

A) The user must have a Directory URI and the service profile must enable URI dialing
B) A separate Jabber server must be deployed
C) RTP tunneling must be disabled
D) Expressway-C must be removed

Answer: A) The user must have a Directory URI and the service profile must enable URI dialing
Explanation: Directory URI is essential for SIP URI-based dialing.

How are learned CCD routes displayed in CUCM?

A) As dynamic patterns in the call routing menu
B) As static device pools
C) In the gateway configuration page
D) Under phone button templates

Answer: A) As dynamic patterns in the call routing menu
Explanation: CCD learned patterns appear dynamically for administrative review.

Which two components are essential for ILS communication?

A) HTTPS and TCP port 8443
B) TFTP and DNS SRV
C) MGCP and HTTP
D) RTMT and UDP port 5060

Answer: A) HTTPS and TCP port 8443
Explanation: ILS uses secure HTTPS on TCP port 8443 for replication and synchronization.

What is the default behavior of CUCM when a learned SAF route becomes unreachable?

A) The route is removed from the routing table
B) The system falls back to MGCP
C) The route remains until a manual deletion
D) It fails over to the next route list

Answer: A) The route is removed from the routing table
Explanation: SAF dynamically updates routes, removing unreachable ones.

Which call flow scenario typically requires a SIP Normalization Script?

A) When interworking between CUCM and a third-party PBX
B) When SRST is unavailable
C) When using multicast RTP
D) When configuring dial-peers

Answer: A) When interworking between CUCM and a third-party PBX
Explanation: SIP normalization ensures compatibility between different SIP vendors.

What does the CUCM SIP Profile setting “Use Fully Qualified Domain Name” affect?

A) SIP URI format in headers
B) Codec negotiation
C) RTP payload size
D) Authentication mode

Answer: A) SIP URI format in headers
Explanation: This setting affects how CUCM formats SIP requests and responses using FQDN.

What does CUCM’s Call Admission Control (CAC) do?

A) Prevents oversubscription of voice/video bandwidth
B) Encrypts all SIP and RTP traffic
C) Blocks unauthorized TFTP downloads
D) Monitors voicemail traffic

Answer: A) Prevents oversubscription of voice/video bandwidth
Explanation: CAC ensures that calls do not exceed available network resources.

What is one advantage of using URI dialing over traditional E.164?

A) More user-friendly addressing
B) Eliminates SIP headers
C) Bypasses firewall rules
D) Requires MGCP

Answer: A) More user-friendly addressing
Explanation: URI dialing uses familiar formats like email addresses, improving usability.

When is Media Bypass used in CUCM?

A) To allow endpoints to establish direct RTP paths
B) To encrypt SIP signaling
C) To route all calls through CUBE
D) To enable multicast conferencing

Answer: A) To allow endpoints to establish direct RTP paths
Explanation: Media Bypass reduces latency and bandwidth usage by keeping RTP off CUCM.

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