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CCNP Voice FAQ: Cisco VoIP Implementations

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CCNP Voice FAQ: Cisco VoIP Implementations

Q1. Which one of the following is not a benefit of VoIP compared to traditional circuit-switched telephony?
A. Consolidated network expenses
B. Improved employee productivity
C. Access to new communication devices
D. Higher voice quality

Answer: D

Q2. Which one of the following is not considered a packet telephony device?
A. IP phone
B. Call agent
C. PBX
D. Gateway

Answer: C

Q3. Which one of the following is not an analog interface?
A. FXO
B. BRI
C. FXS
D. E&M

Answer: B

Q4. Which one of the following digital interface descriptions is incorrect?
A. T1 CAS with 30 voice channels
B. T1 CCS with 23 voice channels
C. BRI with 2 voice channels
D. E1 with 30 voice channels

Answer: A

Q5. Which one of the following is not one of the three stages of a phone call?
A. Call setup
B. Call maintenance
C. Call teardown
D. Call processing

Answer: D

Q6. Which one of the following is not a step in analog-to-digital signal conversion?
A. Sampling
B. Quantization
C. Encoding
D. Decompression

Answer: D

Q7. Based on the Nyquist theorem, what is the appropriate sampling rate for an analog voice signal with a maximum frequency of 4000 Hz?
A. 8800
B. 8000
C. 4000
D. 4400

Answer: B

Q8. Which of the following accurately describes the 8-bit encoding?
A. 1 polarity bit, 3 segment bits, 4 step bits
B. 1 polarity bit, 4 segment bits, 3 step bits
C. 4 polarity bits, 3 segment bits, 1 step bit
D. 3 polarity bits, 4 segment bits, 1 step bit

Answer: A

Q9. Which of the following codec descriptions is incorrect?
A. G.711 PCM 64 Kbps
B. G.726 ADPCM 8 Kbps
C. G.728 LD-CELP 16 Kbps
D. G.729 CS-ACELP 8 Kbps

Answer: B

Q10. Which of the following is not a telephony application that requires usage of a DSP?
A. Voice termination
B. Conferencing
C. Packetization
D. Transcoding

Answer: C

Q11. Which of the following is a false statement?
A. Voice needs the reliability that TCP provides.
B. Voice needs the reordering that RTP provides.
C. Voice needs the time-stamping that RTP provides.
D. Voice needs the multiplexing that UDP provides.

Answer: A

Q12. Which of the following correctly specifies the header sizes for RTP, UDP, and IP?
A. 8 bytes of RTP, 12 bytes of UDP, and 20 bytes of IP
B. 20 bytes of RTP, 12 bytes of UDP, and 8 bytes of IP
C. 8 bytes of RTP, 20 bytes of UDP, and 12 bytes of IP
D. 12 bytes of RTP, 8 bytes of UDP, and 20 bytes of IP

Answer: D

Q13. Which of the following is not a factor influencing VoIP media bandwidth?
A. Packet rate
B. Packetization size
C. TCP overhead
D. Tunneling or security overhead

Answer: C

Q14. If 30 ms of voice is packetized, what will the packet rate be?
A. 50 packets per second
B. 60 packets per second
C. 30 packets per second
D. 33.33 packets per second

Answer: D

Q15. With G.711 and a 20-ms packetization period, what will be the bandwidth requirement over Ethernet (basic Ethernet with no 802.1Q or any tunneling)?
A. 87.2 kbps
B. 80 kbps
C. 64 Kbps
D. 128 Kbps

Answer: A

Q16. With G.729 and 20 ms packetization period, what will be the bandwidth requirement over PPP if cRTP is used with no checksum?
A. 8 Kbps
B. 26.4 Kbps
C. 11.2 Kbps
D. 12 Kbps

Answer: C

Q17. Which of the following is not a factor in determining the amount of bandwidth that can be saved with VAD?
A. Type of audio (one-way or two-way)
B. Codec used
C. Level of background noise
D. Language and character of the speaker

Answer: B

Q18. Which of the following is not a voice gateway function on a Cisco router (ISR)?
A. Connect traditional telephony devices
B. Survivable Remote Site Telephony (SRST)
C. CallManager Express
D. Complete phone feature administration

Answer: D

Q19. Which of the following is not a Cisco Unified CallManager function?
A. Converting analog signal to digital format
B. Dial plan administration
C. Signaling and device control
D. Phone feature administration

Answer: A

Q20. Which of the following is not an enterprise IP Telephony deployment model?
A. Single site
B. Single site with clustering over WAN
C. Multisite with either centralized or distributed call processing
D. Clustering over WAN

Answer: B

Q21. List at least three benefits of packet telephony networks.

Answer: The benefits of packet telephony networks include these:

  • More efficient use of bandwidth and equipment
  • Lower transmission costs
  • Consolidated network expenses
  • Improved employee productivity
  • Access to new communication devices

Q22. List at least three important components of a packet telephony (VoIP) network.

Answer: Following are the components of a packet telephony (VoIP) network:

  • Phones
  • Gateways
  • Multipoint control units
  • Application servers
  • Gatekeepers
  • Call agents
  • Video end points

Q23. List three types of analog interfaces through which legacy analog devices can connect to a VoIP network.

Answer: The analog interfaces through which legacy analog devices can connect to a VoIP network include these:

  • FXS
  • FXO
  • E&M

Q24. List at least two digital interface options to connect VoIP equipment to PBXs or the PSTN.

Answer: The digital interface options to connect VoIP equipment to PBXs or the PSTN include the
following:

  • BRI
  • T1/E1 CAS
  • T1/E1 CCS

Q25. List the three stages of a phone call.

Answer: The three stages of a phone call are:

  1. Call setup
  2. Call maintenance
  3. Call teardown

Q26. What are the two main models of call control?

Answer: The two main models of call control are distributed call control and centralized call control. Examples of distributed call control include H.323 and SIP. An example of centralized call control is MGCP.

Q27. List the steps for converting analog signals to digital signals.

Answer: The steps for converting analog signals to digital signals include the following:

  1. Sampling
  2. Quantization
  3. Encoding
  4. Compression (optional)

Q28. List the steps for converting digital signals to analog signals.

Answer: Following are the steps for converting digital signals to analog signals:

  1. Decompression of the samples, if compressed
  2. Decoding
  3. Reconstruction of the analog signal from PAM signals

Q29. Based on the Nyquist theorem, what should be the minimum sampling rate of analog signals?

Answer: The sampling rate must be at least twice the maximum signal frequency. Because the maximum voice frequency over a telephone channel was considered 4000 Hz, based on the Nyquist theorem, a rate of 8000 samples per second is required.

Q30. What are the two main quantization techniques?

Answer: The two main quantization techniques are linear quantization and logarithmic quantization.

Q31. Name and explain the quantization methods used in North America and in other countries.

Answer: µ-Law is used in the United States, Canada, and Japan. A-Law is used in other countries. Both
methods are quasi-logarithmic; they use logarithmic segment sizes and linear step sizes within each segment. For communication between a µ-Law and an A-Law country, the µ-Law country must change its signaling to accommodate the A-Law country.

Q32. Name at least three main codec/compression standards, and specify their bandwidth requirements.

Answer: The main codec/compression standards and their bandwidth requirements are as follows:

  • G.711 (PCM)—64 Kbps
  • G.726 (ADPCM)—16, 24, or 32 Kbps
  • G.728 (LDCELP)—16 Kbps
  • G.729 (CS-ACELP)—8 Kbps

Q33. What is MOS?

Answer: MOS stands for mean opinion score. It is a measurement of voice quality derived from the judgment of several subscribers. The range of MOS scores is 1 to 5, where 5 is the perfect score for direct conversation.

Q34. What is a DSP?

Answer: DSP stands for digital signal processor. It is a specialized processor used for the following
telephony applications: voice termination, conferencing, and transcoding.

Q35. Which TCP/IP protocols are responsible for transporting voice? What are the sizes of those protocol headers?

Answer: The TCP/IP protocols that are responsible for transporting voice are RTP (12 bytes), UDP (8 bytes), and IP (20 bytes)

Q36. What features does RTP provide to complement UDP?

Answer: RTP provides sequence numbering (reordering) and time-stamping to complement UDP.

Q37. What is cRTP?

Answer: cRTP stands for Compressed RTP. cRTP reduces the IP, UDP, and RTP headers from 40 to 2
bytes (without a checksum), and to 4 bytes (with a checksum). cRTP provides significant bandwidth savings, but it is only recommended for use on slow links (less than 2 Mbps).

Q38. List at least three factors that influence bandwidth requirements of VoIP.

Answer: Packet rate, packetization size, IP overhead, data link overhead, and tunneling overhead influence the bandwidth requirements of VoIP.

Q39. What is the relationship between the packet rate and the packetization period?

Answer: The packet rate and packetization period are reciprocal. For example, if the packetization
period is 20 milliseconds (0.020 seconds), the packet rate is equal to 1 over 0.020, or 50 packets per second.

Q40. What are the sizes of Ethernet, 802.1Q, Frame Relay, and Multilink PPP (MLP) overheads?

Answer: The sizes of Ethernet, 902.1Q, Frame Relay, and Multilink PPP (MLP) overheads are as
follows:

  • Ethernet—18 bytes
  • 802.1Q+Ethernet—4 + 18 = 22 bytes
  • Frame Relay—6 bytes
  • MLP—6 bytes

Q41. Name at least three tunneling and security protocols and their associated overheads.

Answer: Following are the tunneling and security protocols and their associated overheads:

  • IPsec transport mode—30 to 53 bytes
  • IPsec Tunnel mode—50 to 73 bytes
  • L2TP—24 bytes
  • GRE—24 bytes
  • MPLS—4 bytes
  • PPPoE—8 bytes

Q42. Briefly list the steps necessary to compute the total bandwidth for a VoIP call.

Answer: Following are the steps necessary to compute the total bandwidth for a VoIP call:

  1. Determine the codec type and packetization period.
  2. Gather the link information. Determine whether cRTP, any type of tunneling, or IPsec is used.
  3. Calculate the packetization size or period.
  4. Add all the headers to the packetization size.
  5. Calculate the packet rate.
  6. Calculate the total bandwidth

Q43. What is VAD?

Answer: VAD stands for voice activity detection. It suppresses the transmission of silence; therefore,
it might result in up to 35 percent bandwidth savings. The success of VAD depends on the types of audio, the level of background noise, and other factors.

Q44. List at least three important components of enterprise voice implementations.

Answer: The components of the enterprise voice implementations include the following:

  • Gateways
  • Gatekeepers
  • IP phones
  • Cisco Unified CallManager

Q45. List at least three voice gateway functions on a Cisco router.

Answer: On a Cisco router, voice gateway functions include the following:

  • Connect traditional telephony devices
  • Convert analog signals to digital and vice versa
  • Encapsulate digital voice into IP packets
  • Perform voice compression
  • Provide DSP resources for conferencing and transcoding
  • Provide Cisco survivable remote site telephony (SRST)
  • Act as the call agent (Cisco Unified CallManager Express)

Q46. List the main functions of Cisco Unified CallManager.

Answer: The main functions of Cisco Unified CallManager include the following:

  • Call processing
  • Dial plan administration
  • Signaling and device control
  • Phone feature administration
  • Directory and XML services
  • API for external applications

Q47. List the four main enterprise IP Telephony deployment models.

Answer: The main enterprise IP Telephony deployment models are as follows:

  • The single site
  • Multisite with centralized call processing
  • Multisite with distributed call processing
  • Clustering over WAN

Q48. What is CAC?

Answer: CAC stands for call admission control. CAC artificially limits the number of concurrent voice
calls to prevent oversubscription.

Q49. With QoS features in place, there can be up to ten concurrent VoIP calls over a company WAN link. Is there a need for CAC? With no CAC, what will happen when there are more than ten concurrent calls?

Answer: CAC complements QoS features, and it is necessary. If more than ten calls become concurrently active in this case, all calls experience packet loss and extra delays. Therefore, the quality of all calls will drop.

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