p4s ont Updated: November 4th , 2008
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Product Description

Exam Number/Code: 642-845

Exam Name: 642-845 Value Pack

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p4s 642-845 Questions and Answers : 341 Q&As
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* Price: $129.99$89.99
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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, Passguide has assembled to take you through 341 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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* Preparation Lab: 14 Labs
* Updated: November 4th , 2008
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642-845 Preparation Labs

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Feedbacks

* “So after 1 yr and a lot of study i finally did it..CCNP
I cleared ONT today with 989/1000. Do not worry – Prepare and use P4S and you are guaranteed a pass.
If you actually KNOW QoS you can almost pass this test on that knowledge alone. Probably 80/10/10 split between QoS/Wireless security/VoIP. Very easy test in my opinion, but then again, I have lots of experience with QoS and VoIP. Cisco ONT class material is all you need for this one.
ONT is the easist exam i have done for CCNP.
So, i am going on to CCDA and CCDP”
* — Mohamad (Oct, 13 2008)
* “Passed!!!! With a 1000/1000.

Make sure you review:

Analog to Digital = (1) Sampling, (2) Quanitization, (3) Coding, & (4) (the optional Compression)

Digital to Analog = 1 Decompression, 2 Decoding, 3 Reconstruction of Analog Signal

& review Queuing Techniques / differences between them

Next … maybe CCIE or CCSP…still need to think about it..

Thanks Passguide!!!”
* — Dietmar (Oct, 12 2008)
* “Passed this morning with a 944. All questions but maybe 2 are straight out of P4S.
Got all from the P4S: the drag-and-drops, the SDM Simlet, and the other new citrix simlet.

Now that I’ve renewed my CCNP, I’m off to re-new my CCDP and CCIE written. Let this be a lesson to all you kiddies out there, do NOT let your certs expire, talk about a PITA to renew…”
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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.
Preparing for the 642-845 exam? Searching 642-845 Test Questions, 642-845 Practice Exam, 642-845 Dumps?

With the complete collection of questions and answers, Passguide 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.

QUESTION 32:
Router CK1 is being used as a VOIP gateway at a remote Certkiller office. Which

statement is true about the functions provided by a Cisco voice gateway router?
A. ISDN Q signaling is not supported on a voice gateway router.
B. On a voice gateway router, analog signals are converted to digital signals before the
encapsulation occurs.
C. Analog devices cannot be directly connected to a Cisco voice gateway device.
D. Voice gateways always require a call agent to process calls.
E. None of the above.

QUESTION 33:
You need to optimize the Certkiller VOIP network. To have the best possible voice
quality and to effectively utilize the available bandwidth, which queuing and
compression mechanisms need to be used? (Select two)
A. Class-based weighted fair queuing (CBWFQ)
B. TCP header compression
C. UDP header compression
D. Real-Time Transport Protocol (RTP) header compression
E. Priority Queuing (PQ) or Custom Queuing (CQ)
F. Low Latency Queuing (LLQ)

Questions and Answers : 312 Q&As
Updated: 2008-09-05

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QUESTION NO: 1
You need to implement QoS for the Certifyme.com VOIP network. Which three
statements are true about the data traffic characteristics of voice traffic? (Select
three)

A. Voice packets require TCP for rapid retransmission of dropped packets.
B. Latency is not a concern as long as jitter is kept below 30 ms.
C. Voice packets require a fairly constant bandwidth reserved for voice control traffic as
well as for the voice payload.
D. Voice packets do not require a specific type of queuing.
E. Latency must be kept below 150 ms.
F. Voice packets are rather small
Answer: C, E, F
Explanation:
QoS refers to the ability of a network to provide improved service to selected network
traffic over various underlying technologies including Frame Relay, ATM, Ethernet and
802.3 networks, SONET, and IP-routed networks.
QoS features provide improved and more predictable network service by offering the
following services:
1. Dedicated bandwidth
2. Improved loss characteristics
3. Congestion management and avoidance
4. Traffic shaping
5. Prioritization of traffic

Voice quality is directly affected by all three QoS quality factors such as loss, delay, and
delay variation.

Loss causes voice clipping and skips. Industry standard codec algorithms can correct for
up to 30 ms of lost voice. Cisco Voice over IP (VoIP) technology uses 20 ms samples of
voice payload per VoIP packet. Only a single Real Time Transport (RTP) packet could
be lost at any given time. If two successive voice packets are lost, the 30 ms correctable
window is exceeded and voice quality begins to degrade.

Delay can cause voice quality degradation if it is above 200 ms. If the end-to-end voice
delay becomes too long, the conversation sounds as if two parties are talguide over a
satellite link or a CB radio. The ITU standard for VoIP, G.114, states that a 150 ms
one-way delay budget is acceptable for high voice quality. With respect to delay
variation, there are adaptive jitter buffers within IP Telephony devices. These buffers can
usually compensate for 20 to 50 ms of jitter.
QUESTION NO: 2
Certifyme.com uses G.711 for the VOIP calls. When analog signals are digitized using the
G.711 codec, voice samples are encapsulated into protocol data units (PDUs)
involving which three headers? (Select three)

A. UDP
B. RTP
C. IP
D. TCP
E. Compressed RTP
F. H.323
Answer: A, B, C
Explanation:
When a VoIP device, such as a gateway, sends voice over an IP network, the digitized
voice has to be encapsulated into an IP packet. Voice transmission requires features not
provided by the IP protocol header; therefore, additional transport protocols have to be
used. Transport protocols that include features required for voice transmission are TCP,
UDP, and RTP. VoIP utilizes a combination of UDP and RTP.
QUESTION NO: 3
VOIP has been rolled out to every Certifyme.com location. What are three features and
functions of voice (VOIP) traffic on a network? (Select three)
A. Voice traffic is bursty
B. Voice traffic is retransmittable
C. Voice traffic is time-sensitive
D. Voice traffic is bandwidth intensive
E. Voice traffic is constant
F. Voice traffic uses small packet sizes
Answer: C, E, F
Explanation:
The benefits of packet telephony networks include
i. More efficient use of bandwidth and equipment: Traditional telephony networks use
a 64-kbps channel for every voice call. Packet telephony shares bandwidth among
multiple logical connections.
ii. Lower transmission costs: A substantial amount of equipment is needed to combine
64-kbps channels into high-speed links for transport across the network. Packet
telephony statistically multiplexes voice traffic alongside data traffic. This consolidation
provides substantial savings on capital equipment and operations costs.
iii. Consolidated network expenses: Instead of operating separate networks for voice
and data, voice networks are converted to use the packet-switched architecture to create a
single integrated communications network with a common switching and transmission
system. The benefit is significant cost savings on network equipment and operations.

iv. Improved employee productivity through features provided by IP telephony: IP
phones are not only phones, they are complete business communication devices. They
offer directory lookups and access to databases through Extensible Markup Language
(XML) applications. These applications allow simple integration of telephony into any
business application. For instance, employees can use the phone to look up information
about a customer who called in, search for inventory information, and enter orders. The
employee can be notified of a issue (for example, a change of the shipment date), and
with a single click can call the customer about the change. In addition, software-based
phones or wireless phones offer mobility to the phone user.

Passguide ccnp ont 642-845 Questions and Answers : 312 Q&As
Updated: 2008-09-05
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QUESTION 34:
Voice and Video are being used on the Certkiller IP network. Which statement is
true about the comparison of voice traffic with video traffic?
A. Video traffic requires the retransmission capabilities of TCP whereas voice uses UDP.
B. Video conferencing traffic tends to be much more bursty in nature and is more prone
to have an impact on other traffic flows.
C. Video traffic is less sensitive to dropped packets than for voice traffic.
D. Voice traffic tends to be much more bursty in nature and is more prone to have an
impact on other traffic flows.
E. Latency requirements are less stringent for video traffic than for voice traffic.
F. Voice traffic requires the retransmission capabilities of TCP whereas video uses UDP.
G. None of the above.
Answer: B
Explanation:
Packets that carry voice traffic are typically very small, they cannot tolerate delay and

delay variation as they traverse the network. Voices will break up, and words will
become incomprehensible. On the other hand, packets that carry file-transfer data are
typically large and can survive delays and drops. It is possible to retransmit part of a
dropped data file, but it is not feasible to retransmit a part of a voice conversation. The
constant, small-packet voice flow competes with bursty data flows. Unless some
mechanism mediates the overall flow, voice quality will be severely compromised at
times of network congestion. The critical voice traffic must get priority. Voice and video
traffic is very time-sensitive. It cannot be delayed or dropped, or the quality of voice and
video will suffer.
QUESTION 35:
You need to improve the quality of calls in the Certkiller VOIP network. What is a
method for preventing general voice-quality degradation caused by too many
concurrent active calls?
A. Compressed Real-Time Protocol (cRTP)
B. Committed Access Rate (CAR)
C. Call Admission Control (CAC)
D. Low Latency Queuing (LLQ)
E. None of the above
Questions and Answers : 312 Q&As Updated: October 30th , 2008 Price: $129.99 $89.99
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