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Cisco Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Sample Questions:

1. Your PSTN carrier sends digits to your T1 PRI circuit in a digit-by-digit format. How must the T1 PRI circuit be configured to support this capability?

A) The T1 PRI controller supports either en-bloc or digit-by-digit formats natively.
B) The T1 controller needs to include the isdn overlap-receiving command.
C) The serial interface that is associated with the T1 controller needs to include the isdn incoming-voice command.
D) The serial interface that is associated with the T1/E1 controller needs to include the isdn overlap-receiving command.

2. Which command should you use to associate a Session Initiation Protocol phone using a tag of 1 with a directory number with a tag of 20?

A) button 1:20
B) number 20 dn 1
C) button 20:1
D) number 1 dn 20

3. What is the command sequence to configure an FXS port as a signal loop-start?

A) router(config)# voice-port slot/port router(config-voiceport)# signal loopstart router(config-voiceport)# shutdown router(config-voiceport)# no shutdown
B) router(config)# voice-port slot/port router(config-voiceport)# signal groundstart router(config-voiceport)# shutdown router(config-voiceport)# no shutdown
C) router(config)# voice-port slot/port router(config-voiceport)# signal loopstart
D) router(config)# voice-port slot/port router(config-voiceport)# signal groundstart

4. Which statement about the DiffServ model is true?

A) DiffServ is a way to provide the assurance of sufficiently low delay and packet loss for certain types of applications or traffic.
B) DiffServ is based on building a virtual circuit in the Internet using the bandwidth reservation technique.
C) DiffServ provides traffic management using QoS classifications.
D) DiffServ specifies a mechanism for classifying and managing network traffic and providing QoS on modern IP networks.

5. 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) There will be no impact the audio stream because the audio packets are arriving in the jitter buffer window.
B) 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.
C) The jitter buffer will replay the previous voice packets to replace those packets that exceed 30 milliseconds to avoid speech gaps.
D) 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.

Solutions: