Cisco Voice Communications and QoS v8.0 (CVOICE v8.0): 642-437 Exam

The Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) 642-437 is the exam associated with the CCNP Voice certification. This exam tests a candidate’s knowledge of how to implement and operate gateways, gatekeepers, Cisco Unified Border Element, Cisco Unified Communications Manager Express and QoS in a voice network architecture. Candidates can prepare for this exam by taking the CVOICE v8.0 Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) 642-437 course.

PDF Collections:

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VCE:

http://www.examcollection.com/cisco/Cisco.ActualTest.642-437.v2012-04-11.by.Panjarakuttan.86q.vce

http://www.examcollection.com/cisco/Cisco.CertifyMe.642-437.v2012-08-29.by.Cody.97q.vce

QUESTION NO: 1
What is the function of class-based marking?
A. Marking packets based on CoS value, IP precedence value, or DSCP value allows Layer 3 frames to be
identified and distinguished from other packets.
B. Marking frames based only on CoS value or IP precedence value allows Layer 2 frames to be identified
and distinguished from other frames.
C. Marking packets or frames sets information in the Layer 2 and Layer 3 headers of a packet so that the
packet or frame can be identified and distinguished from other packets or frames.
D. Marking frames only sets information in the Layer 2 headers of a frame so that the frame can be identified
and distinguished from other packets or frames.
Answer: C
QUESTION NO: 2
Voice packets are arriving at a destination with a variance of between 20 and 50 ms. If the jitter buffer has a
capacity of 30 ms, what is the impact on the audio at the receiving IP phone?
A. The jitter buffer will replay the previous voice packets to replace those packets that exceed 30 ms to avoid
speech gaps.
B. The audio stream at the receiving IP phone will be delayed and garbled.
C. 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.
D. 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.
Answer: B
QUESTION NO:
Your PSTN carrier sends digits to your TI 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 serial interface that is associated with the T1 controller needs to include the isdn incoming-voice
command.
C. The T1 controller needs to include the isdn overlap-receiving command.
D. The serial interface that is associated with the T1/E1 controller needs to include the isdn overlap-receiving
command.
Answer: D
QUESTION NO:
You have a Cisco Unified Border Element configured to provide H.323 to SIP interworking. Which command
will verify that you have a single H.323 and a single SIP call leg when the call is placed?
A. show call active voice
B. debug voip ipipgw
C. show dialpeer voice
D. debug voice dialpeer
Answer: