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Cisco Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) : 642-437
- Exam Code: 642-437
- Exam Name: Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0)
- Updated: Sep 14, 2026 Q & A: 257 Questions and Answers
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Cisco 642-437 Exam Syllabus Topics:
| Section | Objectives |
|---|---|
| Topic 1: Quality of Service (QoS) | - Traffic classification and marking - QoS models and mechanisms - Congestion management and avoidance - Queuing techniques and bandwidth management |
| Topic 2: Troubleshooting and Operations | - Voice and QoS troubleshooting methodologies - Common VoIP deployment issues |
| Topic 3: Call Admission Control and Performance | - Voice network performance considerations - Call admission control mechanisms |
| Topic 4: Signaling and Call Control | - Gateways and gatekeepers - SIP, H.323, and MGCP protocols - Dial plans and call routing |
| Topic 5: VoIP Fundamentals | - Voice codecs and compression techniques - Voice traffic signaling basics - Voice over IP architecture and components |
Cisco Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Sample Questions:
Question #1
Which type of voice port supports Dialed Number Identification Service signaling for outbound calls, and inbound calls carry only ANI?
- A. FXS
- B. E&M
- C. FXO
- D. FXS-DID
Correct Answer: C 🗳️
Question #2
Which type of delay describes the amount of time it takes to place a frame onto a physical medium?
- A. propagation delay
- B. serialization delay
- C. queuing delay
- D. processing delay
Correct Answer: B 🗳️
Question #3
Voice packets are arriving at a destination with a variance of between 20 and 50 milliseconds. If the jitter buffer has a capacity of 30 milliseconds, what is the impact on the audio at the receivers IP phone? Voice packets are arriving at a destination with a variance of between 20 and 50 milliseconds. If the jitter buffer has a capacity of 30 milliseconds, what is the impact on the audio at the receivers IP phone?
- A. The DSP will automatically increase the jitter buffer size after sampling the range of incoming voice packets to accommodate the wider range in variation of voice packet arrival times to avoid voice gaps.
- B. The jitter buffer will replay the previous voice packets to replace those packets that exceed 30 milliseconds to avoid speech gaps.
- C. The IP phone will negotiate in mid-call a lower bandwidth codec to reduce the delay in the arrival of voice packets to avoid voice gaps.
- D. There will be no impact the audio stream because the audio packets are arriving in the jitter buffer window.
Correct Answer: D 🗳️
Question #4
Which feature can be used to avoid traffic congestion by dropping low-priority packets rather than dropping high-priority packets and is not recommended for voice networks?
- A. weighted random early detection
- B. tail-drop
- C. low latency queuing
- D. class-based weighted fair queuing
Correct Answer: A 🗳️
Question #5
In the destination patterns, which wildcard symbol indicates a single-digit placeholder?
- A. . (period)
- B. +
- C. %
- D. ()
Correct Answer: A 🗳️
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