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ONT – Optimizing Converged Cisco Networks : 642-845 Exam

Exam Number/Code: 642-845
Exam Name:ONT – Optimizing Converged Cisco Networks

“ONT – Optimizing Converged Cisco Networks”, also known as 642-845 exam, is a Cisco certification. With the complete collection of questions and answers Q&as with Expert Explanations, Pass4sure has assembled to take you through 312 Q&As to your 642-845 Exam preparation. In the 642-845 exam resources, you will cover every field and category in CCNP helping to ready you for your successful Cisco Certification.
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QUESTION 21:
Two PassGuide locations are using a multi-site centralized call processing model. The
voice gateway on the remote branch has lost IP connectivity to its Cisco
CallManager server. Which feature enables the remote gateway to take the role of
the call agent during the WAN failure?
A. Cisco CallManager Attendant Console
B. Automated alternate routing (AAR)
C. Real-Time Protocol (RTP)
D. Survivable Remote Site Telephony (SRST)
E. None of the above
Answer: D
Explanation:
In IP telephony environments, gateways support fallback scenarios for IP phones that
have lost IP connectivity to their call agent (that is, Cisco Unified CallManager). This
feature, called Cisco Survivable Remote Site Telephony (SRST), enables the gateway to
take the role of the call agent during WAN failure. Local calls can then proceed even if
IP connectivity to Cisco Unified CallManager is broken. In addition, Cisco SRST can
route calls out to the PSTN and, thus, use the PSTN as the backup route for calls toward
any site that is not reachable via IP.
642-845
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QUESTION 22:
A Cisco router is being used as a VOIP gateway to convert voice signals in the
PassGuide network. When a router converts analog signals to digital signals, what
three steps are always included in the process? (Select three)
A. Compression
B. Involution
C. Quantization
D. Encoding
E. Sampling
F. Companding
Answer: C, D, E
Explanation:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling
is a pulse amplitude modulation (PAM) signal.
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale
measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression: Optionally, voice samples can be compressed to reduce bandwidth
requirements. Analog-to-digital conversion is done by digital signal processors (DSPs),
which are located on the voice interface cards. The conversion is needed for calls
received on analog lines, which are then sent out to a packet network or to a digital voice
interface.
QUESTION 23:
The PassGuide network has offices around the country. With the use of Cisco Unified
CallManager, PassGuide has deployed a VoIP network in a multisite centralized
configuration. Which IOS gateway feature should be deployed to enable VoIP
devices to register with a local gateway and continue to function when the
connection to the Cisco Unified CallManager is broken?
A. Automatic Alternate Routing (AAR)
B. Gatekeeper multizone
C. Call Admission Control (CAC)
D. Survivable Remote Site Telephony (SRST)
E. None of the above
Answer: D
Explanation:
642-845
www.actualtest.org- The Power of Knowing
In IP telephony environments, gateways support fallback scenarios for IP phones that
have lost IP connectivity to their call agent (that is, Cisco Unified CallManager). This
feature, called Cisco Survivable Remote Site Telephony (SRST), enables the gateway to
take the role of the call agent during WAN failure. Local calls can then proceed even if
IP connectivity to Cisco Unified CallManager is broken. In addition, Cisco SRST can
route calls out to the PSTN and, thus, use the PSTN as the backup route for calls toward
any site that is not reachable via IP.
QUESTION 24:
A Cisco router is being used as a VOIP gateway to convert analog and digital voice
signals in the PassGuide network. Which two statements are true about analog to
digital conversion of voice signals for use in digital telephony networks? (Select two)
A. The output of the sampling process is a pulse code modulation (PCM) signal.
B. The three required steps in the analog to digital conversion are sampling, quantization,
and encoding.
C. The three required steps in the analog to digital conversion are sampling, encoding,
and compression.
D. The output of the sampling process is a pulse amplitude modulation (PAM) signal.
E. The three required steps in the analog to digital conversion are sampling, quantization,
and compression.
Answer: B, D
Explanation:
Analog to digital conversion steps include these are:
Step 1 Sampling: The analog signal is sampled periodically. The output of the sampling
is a pulse amplitude modulation (PAM) signal.
Step 2 Quantization: The PAM signal is matched to a segmented scale. This scale
measures the amplitude (height) of the PAM signal.
Step 3 Encoding: The matched scale value is represented in binary format.
Step 4 Compression: Optionally, voice samples can be compressed to reduce bandwidth
requirements.
Analog-to-digital conversion is done by digital signal processors (DSPs), which are
located on the voice interface cards. The conversion is needed for calls received on
analog lines, which are then sent out to a packet network or to a digital voice interface.
QUESTION 25:
Many of the Cisco VOIP gateways used in the PassGuide network contain FXS
interfaces. Which statement is true about Foreign Exchange Station (FXS) ports on
a router?
642-845
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A. The FXS interface connects directly to an IP phone and supplies ring, voltage, and dial
tone.
B. The FXS interface allows an analog connection to be directed at the public switched
telephone network (PSTN’s) central office.
C. The FXS interface connects directly to a standard telephone, fax machine, or similar
device and supplies ring, voltage, and dial tone.
D. The FXS interface connects directly to ISDN voice channels.
E. None of the above.
Answer: C