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Optimizing Converged Cisco Networks

Exam Number: 642-845
Associated Certifications: CCNP
Duration: 90 minutes
Available Languages: English
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Exam Description Exam Topics Recommended Training Additional Resources
Exam Description
The Optimizing Converged Cisco Networks (642-845 ONT) is a qualifying exam for the Cisco Certified Network Professional CCNP®. The ONT 642-845 exam will certify that the successful candidate has important knowledge and skills in optimizing and providing effective QOS techniques for converged networks. The exam topics include implementing a VOIP network, implementing QoS on converged networks, specific IP QoS mechanisms for implementing the DiffServ QoS model, AutoQoS, wireless security and basic wireless management.

Exam Topics
The following information provides general guidelines for the content likely to be included on the exam. However, other related topics may also appear on any specific delivery of the exam. In order to better reflect the contents of the exam and for clarity purposes the guidelines below may change at any time without notice.

Describe Cisco VoIP implementations.
Describe the functions and operations of a VoIP network (e.g., packetization, bandwidth considerations, CAC, etc.).
Describe and identify basic voice components in an enterprise network (e.g. Gatekeepers, Gateways, etc.)

Describe QoS considerations.
Explain the necessity of QoS in converged networks (e.g., bandwidth, delay, loss, etc.).
Describe strategies for QoS implementations (e.g. QoS Policy, QoS Models, etc.).

Describe DiffServ QoS implementations.
Describe classification and marking (e.g., CoS, ToS, IP Precedence, DSCP, etc.).
Describe and configure NBAR for classification.
Explain congestion management and avoidance mechanisms (e.g., FIFO, PQ, WRR, WRED, etc.).
Describe traffic policing and traffic shaping (i.e., traffic conditioners).
Describe Control Plane Policing.
Describe WAN link efficiency mechanisms (e.g., Payload/Header Compression, MLP with interleaving, etc.).
Describe and configure QoS Pre-Classify.

Implement AutoQoS.
Explain the functions and operations of AutoQoS.
Describe the SDM QoS Wizard.
Configure, verify, and torubleshoot AutoQoS implementations (i.e., MQC).

Implement WLAN security and management.
Describe and Configure wireless security on Cisco Clients and APs (e.g., SSID, WEP, LEAP, etc.).
Describe basic wireless management (e.g., WLSE and WCS). Configure and verify basic WCS configuration (i.e., login, add/review controller/AP status, security, and import/review maps).
Describe and configure WLAN QoS.

QUESTION 1
You need to implement QoS for the Certkiller 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 talking 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 2
Certkiller 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 3
VOIP has been rolled out to every Certkiller 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.

QUESTION 4
Certkiller is rolling out an H.323 VOIP network using Cisco devices. Which IOS feature provides dial plan scalability and bandwidth management for H.323 VoIP implementations?

A. Digital Signal Processors
B. Call Routing
C. Gatekeeper
D. Call Admission Control
E. None of the above
Answer: C Explanation:
Enterprise voice implementations use components such as gateways, gatekeepers, Cisco
Unified CallManager, and IP phones. Cisco Unified CallManager offers PBX-like features to IP phones. Gateways interconnect traditional telephony systems, such as
analog or digital phones, PBXs, or the public switched telephone network (PSTN) to the
IP telephony solution. Gatekeepers can be used for scalability of dial plans and for bandwidth management when using the H.323 protocol.

QUESTION 5
A Cisco router is being used as a VOIP gateway to convert voice signals in the Certkiller network. What steps are taken when a router converts a voice signal from analog to digital form? (Select two)

A. Quantization B. Serialization C. Packetization D. Sampling
Answer: A, D 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

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Recommended Training about 642-845 exam PDF
The following courses are the recommended training for 642-845 exam PDF.
642-845 Q & A with Explanations
642-845 Audio Exam
642-845 Study Guide
642-845 Preparation Lab

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.
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