Implementing Cisco IP Telephony and Video, Part 1 (CIPTV1) Practice Exam
Prepare to excel in the Implementing Cisco IP Telephony and Video, Part 1 (CIPTV1) exam with Exam Sage’s comprehensive and expertly crafted practice test. This exam is a critical step for IT professionals aiming to validate their skills in deploying and managing Cisco IP telephony and video solutions, a key component of modern unified communications infrastructure.
What Is the CIPTV1 Exam?
The CIPTV1 exam assesses your knowledge and hands-on skills related to Cisco Unified Communications Manager (CUCM), Cisco IP phones, and video endpoints. It tests your understanding of call control protocols, device configuration, signaling, call routing, quality of service, and troubleshooting within Cisco’s IP telephony environment. Passing this exam demonstrates your ability to design and implement efficient and secure Cisco collaboration networks.
What You Will Learn
Cisco Unified Communications Manager (CUCM) Architecture: Gain insights into CUCM services, clustering, device pools, and deployment models.
Device Configuration and Management: Learn to register and configure Cisco IP phones and video endpoints, including SCCP and SIP protocols.
Call Routing and Signaling: Understand route patterns, partitions, calling search spaces, and Cisco’s call admission control.
Quality of Service (QoS): Explore strategies to prioritize voice and video traffic to maintain call clarity.
Cisco IP Telephony Features: Master call forwarding, transfer, hunt groups, call park, and more.
Security and High Availability: Learn best practices to secure CUCM environments and ensure continuous service with redundancy.
Cisco Video Integration: Understand video endpoint management and integration with TelePresence solutions.
Covered Topics
CUCM architecture and deployment
Device registration and provisioning
Call routing and dial plans
Call features and services
Codec and media resource management
QoS and bandwidth management
Troubleshooting IP telephony and video
Cisco Jabber and collaboration endpoints
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ExamSage.com offers a trusted, up-to-date, and detailed practice exam designed specifically for the CIPTV1 certification. Our practice tests provide realistic multiple-choice questions with in-depth explanations to help you understand core concepts and apply them confidently. With Exam Sage, you’ll benefit from:
High-quality, expert-crafted questions aligned with the latest Cisco exam objectives
Detailed explanations to reinforce learning and clarify difficult topics
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Regular updates reflecting Cisco’s evolving technologies and exam changes
Focused preparation to boost your confidence and maximize your exam success
Start your journey toward Cisco collaboration certification with Exam Sage’s CIPTV1 practice exam. Achieve mastery in implementing Cisco IP telephony and video solutions and unlock new career opportunities in the fast-growing field of unified communications.
Sample Questions and Answers
1. Which protocol is used by Cisco Unified Communications Manager to signal between endpoints?
A. H.323
B. SIP
C. MGCP
D. SCCP
Answer: D. SCCP
Explanation: SCCP (Skinny Client Control Protocol) is a proprietary Cisco protocol used between CUCM and IP Phones.
2. What is the primary function of a Region in CUCM?
A. Assign IP addresses
B. Control codec selection between sites
C. Manage DHCP settings
D. Set time zones for phones
Answer: B. Control codec selection between sites
Explanation: Regions are used to manage bandwidth and codec selection between locations.
3. What is the purpose of a route pattern in CUCM?
A. To route IP packets to the PSTN
B. To define how digits are interpreted and routed
C. To match call types to classes of service
D. To define user roles
Answer: B. To define how digits are interpreted and routed
Explanation: Route patterns determine how dialed digit strings are matched and routed in CUCM.
4. Which of the following provides digit manipulation in CUCM?
A. Translation Pattern
B. Route Group
C. Calling Search Space
D. Device Pool
Answer: A. Translation Pattern
Explanation: Translation patterns allow modification of dialed digits before routing decisions.
5. What is a Calling Search Space (CSS) in CUCM?
A. It determines the dial plan route
B. It defines accessible partitions
C. It assigns phone IPs
D. It connects to the gateway
Answer: B. It defines accessible partitions
Explanation: CSS controls which partitions a device or pattern can access.
6. What is the function of a Route List in CUCM?
A. Routes RTP streams
B. Routes control messages
C. Groups route groups for call routing
D. Acts as a backup CUCM server
Answer: C. Groups route groups for call routing
Explanation: Route Lists enable multiple route groups to be used for call routing.
7. SIP uses which port by default for signaling?
A. 5060
B. 2000
C. 16384
D. 1720
Answer: A. 5060
Explanation: SIP uses port 5060 for unencrypted signaling.
8. Which CUCM component selects which codec to use between endpoints?
A. Partition
B. Region
C. Route Pattern
D. Device Pool
Answer: B. Region
Explanation: Regions determine allowable codecs between locations based on device configuration.
9. What is the role of a Device Pool in CUCM?
A. Assign phone numbers
B. Assign IP addresses
C. Provide configuration elements like region, date/time group
D. Filter dial patterns
Answer: C. Provide configuration elements like region, date/time group
Explanation: Device Pools provide regional settings and configuration groups.
10. Which protocol is used for media stream transport in Cisco voice networks?
A. SIP
B. SCCP
C. H.323
D. RTP
Answer: D. RTP
Explanation: Real-Time Transport Protocol (RTP) is used for transporting media like audio.
11. What does MGCP stand for?
A. Media Gateway Control Protocol
B. Multi-Gateway Configuration Protocol
C. Managed Gateway Cisco Protocol
D. Modular Gateway Channel Protocol
Answer: A. Media Gateway Control Protocol
Explanation: MGCP is a control protocol used between CUCM and gateways.
12. Which two protocols can be used for video endpoint signaling in CUCM? (Choose two)
A. SCCP
B. RTP
C. SIP
D. H.323
Answer: C. SIP and D. H.323
Explanation: Both SIP and H.323 can be used for signaling video endpoints.
13. Which type of pattern provides digit manipulation after call routing is determined?
A. Translation Pattern
B. Route Pattern
C. Transformation Pattern
D. Hunt Pilot
Answer: C. Transformation Pattern
Explanation: Transformation patterns modify digits post-call routing, unlike translation patterns.
14. Which CUCM component defines access to dial plan elements like route patterns and translation patterns?
A. Partition
B. Route Group
C. Location
D. Region
Answer: A. Partition
Explanation: Partitions contain dial plan elements and are accessed through CSS.
15. What is the purpose of SAF in CUCM?
A. Enabling SIP trunking
B. Facilitating call control redundancy
C. Enabling dynamic advertisement of dial plan information
D. Routing video calls
Answer: C. Enabling dynamic advertisement of dial plan information
Explanation: SAF (Service Advertisement Framework) allows CUCM clusters to share dial plan information.
16. What is ILS used for in Cisco UC?
A. Secure voice encryption
B. Call routing to PSTN
C. Sharing user directory information between clusters
D. Codec negotiation
Answer: C. Sharing user directory information between clusters
Explanation: Intercluster Lookup Service (ILS) facilitates URI and DN sharing.
17. What is the function of RSVP in a Cisco UC environment?
A. Encrypts call signaling
B. Provides NAT traversal
C. Reserves bandwidth for voice/video
D. Connects IP phones to CUCM
Answer: C. Reserves bandwidth for voice/video
Explanation: RSVP (Resource Reservation Protocol) ensures bandwidth for QoS.
18. Which of the following ensures redundancy and load balancing in CUCM call routing?
A. Device Pool
B. SAF
C. Route List
D. SIP Trunk
Answer: C. Route List
Explanation: Route Lists can include multiple Route Groups for load balancing and redundancy.
19. What type of codec is G.729?
A. Wideband
B. Narrowband
C. Uncompressed
D. Video-only
Answer: B. Narrowband
Explanation: G.729 is a compressed narrowband codec typically used for WAN links.
20. Which component is not a part of the CUCM dial plan?
A. Route Pattern
B. Location
C. Translation Pattern
D. Hunt Pilot
Answer: B. Location
Explanation: Location is used for CAC (Call Admission Control), not part of dial plan matching.
21. What defines the maximum number of active calls between sites?
A. Region
B. Device Pool
C. Location
D. CSS
Answer: C. Location
Explanation: Locations control bandwidth and active call limits between sites.
22. What does URI dialing allow in CUCM?
A. Calling with phone numbers only
B. Calling using usernames or email addresses
C. Location-based routing
D. Dynamic IP routing
Answer: B. Calling using usernames or email addresses
Explanation: URI dialing allows use of email-like addresses instead of numbers.
23. Which dial plan element can reroute calls to multiple extensions in a sequence?
A. Translation Pattern
B. Hunt Pilot
C. Device Pool
D. SIP Trunk
Answer: B. Hunt Pilot
Explanation: Hunt Pilots route calls to a group of extensions based on algorithms like top-down.
24. Which of the following supports video on a Cisco phone?
A. G.711
B. H.264
C. G.729
D. G.722
Answer: B. H.264
Explanation: H.264 is a video codec used by Cisco endpoints.
25. Which is a benefit of using SIP over SCCP?
A. Lower latency
B. More Cisco support
C. Greater interoperability
D. Simplified codec support
Answer: C. Greater interoperability
Explanation: SIP is a standardized protocol and offers broader compatibility.
26. Which trunk type is most commonly used between CUCM clusters?
A. SIP
B. MGCP
C. H.323
D. SCCP
Answer: A. SIP
Explanation: SIP trunks are the standard for intercluster communication.
27. How does CUCM handle video calls when using low-bandwidth links?
A. Blocks all calls
B. Switches to G.711
C. Prioritizes voice
D. Uses regions and locations to manage codec and bandwidth
Answer: D. Uses regions and locations to manage codec and bandwidth
Explanation: CUCM uses regions and locations for bandwidth management.
28. What feature allows Cisco phones to call external PSTN numbers through CUCM?
A. SAF
B. Route Pattern
C. MGCP
D. SIP Trunk
Answer: B. Route Pattern
Explanation: Route Patterns direct calls to external destinations via gateways or trunks.
29. What is required for URI dialing to work in a multi-cluster environment?
A. MGCP gateway
B. Intercluster Lookup Service (ILS)
C. Device Pool
D. Hunt List
Answer: B. Intercluster Lookup Service (ILS)
Explanation: ILS shares URI and DN info across clusters to enable URI dialing.
30. What is a Hunt List in CUCM?
A. A collection of dial peers
B. A set of translation rules
C. A list of line groups for routing calls
D. A group of partitions
Answer: C. A list of line groups for routing calls
Explanation: Hunt Lists route calls through line groups in a specific order.
31. What is the purpose of a Location in CUCM?
A. Define codec selection
B. Assign phone numbers
C. Enforce call admission control
D. Provide time zone settings
Answer: C. Enforce call admission control
Explanation: Locations manage bandwidth limits between sites to control the number of active calls.
32. Which Cisco Unified Communications feature allows digit manipulation at the gateway?
A. Route List
B. Voice Translation Rule
C. Transformation Pattern
D. Dial-peer
Answer: B. Voice Translation Rule
Explanation: Voice translation rules are configured on gateways to manipulate digits before or after routing.
33. What is a SIP profile used for in CUCM?
A. Codec preference configuration
B. SIP trunk security
C. Define SIP-specific parameters like timers and behaviors
D. Assign user privileges
Answer: C. Define SIP-specific parameters like timers and behaviors
Explanation: SIP profiles manage SIP behavior, timers, and capabilities on trunks and devices.
34. Which call routing element determines the matching dialed number range?
A. Route Group
B. Partition
C. Route Pattern
D. Device Pool
Answer: C. Route Pattern
Explanation: Route Patterns match dialed numbers and determine routing paths.
35. A CUCM system has multiple sites with low-bandwidth links. What feature can be enabled to prevent poor call quality?
A. RSVP Agent
B. Locations-based CAC
C. G.722 codec
D. SAF
Answer: B. Locations-based CAC
Explanation: Locations enforce bandwidth limits, ensuring QoS by denying calls if resources are insufficient.
36. What must be configured on both CUCM clusters to enable inter-cluster URI dialing?
A. MGCP Gateways
B. SIP Proxy Server
C. ILS and SIP Trunks
D. CSS and Partitions only
Answer: C. ILS and SIP Trunks
Explanation: Intercluster Lookup Service (ILS) and SIP trunks enable URI-based inter-cluster dialing.
37. What is the role of the Media Termination Point (MTP)?
A. Encrypts SIP signaling
B. Enables DTMF relay between incompatible devices
C. Prioritizes video packets
D. Sends SRTP key exchanges
Answer: B. Enables DTMF relay between incompatible devices
Explanation: MTPs handle signaling incompatibilities and allow features like DTMF relay between mismatched devices.
38. Which of the following is a video-enabled endpoint in a Cisco Unified Communications environment?
A. Cisco 3905
B. Cisco Jabber
C. Cisco VG310
D. Cisco ATA
Answer: B. Cisco Jabber
Explanation: Cisco Jabber supports video communication and presence.
39. How can video bandwidth be limited between CUCM locations?
A. Route Groups
B. Regions and Locations
C. CSS and Partitions
D. Translation Patterns
Answer: B. Regions and Locations
Explanation: Regions control codecs; Locations enforce bandwidth restrictions between sites.
40. Which device would use MGCP to register with CUCM?
A. SIP trunk
B. IP Phone
C. H.323 Gateway
D. MGCP Gateway
Answer: D. MGCP Gateway
Explanation: MGCP gateways register with CUCM and are fully controlled by it.
41. Which codec is recommended by Cisco for high-quality wideband audio?
A. G.711
B. G.729
C. G.722
D. iLBC
Answer: C. G.722
Explanation: G.722 provides wideband (HD) audio with high quality at a low bandwidth cost.
42. Which of the following is an advantage of SIP over H.323 in CUCM?
A. Easier NAT traversal
B. Higher codec quality
C. Simplified dial plan
D. Reduced packet loss
Answer: A. Easier NAT traversal
Explanation: SIP is more NAT-friendly and offers modern protocol enhancements compared to H.323.
43. What allows a Cisco gateway to route calls when CUCM is down?
A. MGCP
B. SIP Proxy
C. SRST (Survivable Remote Site Telephony)
D. Route Group
Answer: C. SRST (Survivable Remote Site Telephony)
Explanation: SRST enables IP phones to register with the router during a CUCM outage.
44. What is the typical use case of a Route Group?
A. To list available translations
B. To combine gateways for outbound call routing
C. To define partitions
D. To assign user licenses
Answer: B. To combine gateways for outbound call routing
Explanation: Route Groups group gateways or trunks for use in Route Lists.
45. What feature ensures phones receive their proper regional settings like time zone and network locale?
A. Partition
B. Device Pool
C. SIP Profile
D. Transformation Pattern
Answer: B. Device Pool
Explanation: Device Pools assign regional and configuration parameters to endpoints.
46. In SIP, what does CUCM use to route a call to a SIP trunk?
A. Call Leg
B. Route List
C. SIP Profile
D. SIP Route Pattern
Answer: D. SIP Route Pattern
Explanation: SIP Route Patterns are used to direct calls to SIP trunks based on digit matches.
47. What component is required on a Cisco ISR to provide SRST functionality?
A. SIP Trunk
B. MGCP Configuration
C. SRST license and IOS feature
D. CME GUI
Answer: C. SRST license and IOS feature
Explanation: The SRST feature on IOS routers allows phone registration in fallback mode.
48. What distinguishes an ICT (Intercluster Trunk) in CUCM?
A. It uses MGCP
B. It is a gateway-based trunk
C. It connects two CUCM clusters directly
D. It requires SRST
Answer: C. It connects two CUCM clusters directly
Explanation: Intercluster Trunks enable communication between different CUCM clusters.
49. When implementing video calling in a CUCM network, what element is critical to successful media negotiation?
A. CSS
B. Region
C. Partition
D. Hunt List
Answer: B. Region
Explanation: Regions determine which codecs (including video) can be used between devices.
50. What is the impact of placing a SIP trunk in a different CSS?
A. It won’t impact routing
B. It will block media streams
C. It will restrict access to route patterns and partitions
D. It will reduce codec availability
Answer: C. It will restrict access to route patterns and partitions
Explanation: CSS controls access to dial plan elements; incorrect CSS can cause call failures.
51. What does DTMF mean in Cisco Unified Communications?
A. Digital Transfer Media Format
B. Dual-Tone Multi-Frequency
C. Digit Transfer Management Function
D. Dual Terminal Matching Format
Answer: B. Dual-Tone Multi-Frequency
Explanation: DTMF represents tones generated when pressing keys on a phone keypad.
52. What is a SIP trunk security profile used for?
A. DTMF negotiation
B. RTP encryption
C. Defining transport, port, and security options
D. Codec fallback
Answer: C. Defining transport, port, and security options
Explanation: SIP Trunk Security Profiles define security features such as TLS and SRTP.
53. What is needed to allow a SIP trunk to support early offer?
A. MGCP configuration
B. Media Termination Point
C. Early offer support for voice and video enabled in trunk profile
D. SIP Proxy
Answer: C. Early offer support for voice and video enabled in trunk profile
Explanation: Early Offer is configured in the SIP Profile and enables SDP in the INVITE.
54. What is the maximum number of characters for a URI in CUCM?
A. 32
B. 64
C. 128
D. 254
Answer: D. 254
Explanation: CUCM supports URI dialing up to 254 characters.
55. Which option allows video calls from an IP Phone to a video conferencing system?
A. Codec negotiation only
B. SIP URI dialing
C. Location bandwidth bypass
D. MGCP configuration
Answer: B. SIP URI dialing
Explanation: URI dialing enables seamless video calling using standardized identifiers.
