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QUESTION 61
When configuring AutoQoS VoIP on a Cisco Catalyst switch how is the configuration performed?
A. The auto qos voip command is applied to each interface.
B. The auto qos voip command is applied globally in the switch.
C. Each interface will need either the auto qos voip cisco-phone or auto qos voip trust on each interface depending on the upstream device.
D. Each interface will need either the auto qos voip trust cisco-phone or auto qos voip trust trust on each interface depending on the upstream device.
Correct Answer: C Section: (none) Explanation
Explanation/Reference: Explanation:
The QoS mechanisms on a Catalyst switch differ from those QoS mechanisms found on a router. For example, while a router uses LLQ as a priority queuing strategy, a Catalyst switch might use weighted round-robin (WRR) as a priority queuing strategy. Fortunately, the AutoQoS feature available on some Catalyst switch models applies voice-specific QoS features globally to a Catalyst switch and also at the port level. To configure AutoQoS on supported Catalyst switch platforms, issue the following command from interface configuration mode:
Switch(config-if)#auto qos voip [trust | cisco-phone] If the trust option is used in the previous command, the Catalyst switch makes queuing decisions based on Layer 2 Class of Service (CoS) markings. However, if the cisco-phone option is used, the Catalyst switch makes queuing decisions based on CoS markings originating from a Cisco IP phone. The switch detects the presence of a Cisco IP phone via the CDP.
http://www.cisco.com/en/US/docs/ios/12_2t/12_2t15/feature/guide/ftautoq1.html
QUESTION 62
Assuming no cRTP or header compression. How many VoIP G.729 calls can be made simultaneously over a 128-kb/s Frame Relay circuit (Layer 3) if 50 percent of the circuit is dedicated to voice and 50 percent is dedicated to data?
A. 1
B. 2
C. 3
D. 4
E. 5
Correct Answer: B Section: (none) Explanation
Explanation/Reference: Explanation:
Bandwidth Calculation Formulas These calculations are used: Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size) Codec bit rate = codec sample size / codec sample interval PPS = (codec bit rate) / (voice payload size) Bandwidth = total packet size * PPS
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml
QUESTION 63
How are firmware images implemented and which file type describes the contents of the firmware image?
A. Firmware images are implanted as firmware groups that are described by a file that has a .cnf suffix.
B. Firmware images are implemented as individual files that are described by a file that has a .loads suffix.
C. Firmware images are implemented as a file loader group and are described by a file that ends with a .sbn suffix.
D. Firmware images are implemented as file bundles that are described by a file that ends with a .loads suffix.
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Explanation:
QUESTION 64
Which three methods are used by a Cisco Unified Border Element to provide network hiding? (Choose three.)
A. Back-to-back user agent, replacing all SIP-embedded IP addressing
B. IP network security boundary
C. media flow-through
D. RSVP
E. IP network privacy
F. Intelligent IP address translation for RTP flows
Correct Answer: ABE Section: (none) Explanation
Explanation/Reference:
Explanation:
QUESTION 65
What is the function of class-based marking?
A. Marking packets is based only 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 frames or packets sets information in the Layer 2 and Layer 3 headers of a packet so that the frame or packet can be identified and distinguished from other frames or packets in the same traffic flow.
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.
E. Marking allows network devices to classify a packet or frame, based on a specific traffic descriptor.
Correct Answer: E Section: (none) Explanation
Explanation/Reference:
Explanation:
QUESTION 66
A small office needs to provide outbound dialing and in-bound DID without the cost of a T1 circuit. All signaling is loop start. Which analog port configuration will support these requirements?
A. voice-port 0/0/0 description fxs-did signal did loop-start !
voice-port 0/1/0
description fxo
signal loop-start
!
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
port 0/0/0
!
dial-peer voice 90 pots
“Pass Any Exam. Any Time.” – www.actualtests.com 43
Cisco 642-437 Exam
destination-pattern 9T
port 0/1/0
B. voice-port 0/0/0 signal loop-start ! voice-port 0/1/0 signal loop-start ! dial-peer voice 1 pots incoming called-number T direct-inward-dial ! dial-peer voice 90 pots destination-pattern 9T port 0/1/0
C. voice-port 0/1/0 signal did loop-start ! dial-peer voice 1 pots incoming called-number . ! dial-peer voice 90 pots destination-pattern 9T port 0/1/0
D. voice-port 0/0/0 signal did loop-start ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! dial-peer voice 90 pots destination-pattern 9T port 0/0/0
Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Explanation:
QUESTION 67
Which statement best describes dial peers in a voice gateway. (Choose two.)
A. Dial peers are call legs that are used to identify call source and destination endpoints and to define the characteristics that are applied to each call leg in the call connection.
B. Dial peers are configured with call legs that are essential to implementing dial plans and providing voice services over an IP packet network.
C. A dial peer is a physical addressable endpoint in a voice gateway.
D. Dial peers create physical connections called call legs to complete an end-to-end call.
Correct Answer: AC Section: (none) Explanation
Explanation/Reference:
Explanation:
QUESTION 68
Which QoS mechanism for VoIP works with weighted fair queuing (WFQ) and class-based weighted fair queuing (CBWFQ)?
A. Header compression
B. FRF.12
C. IP RTP Priority and Frame Relay IP RTP Priority
D. Multilink PPP

E. RSVP
Correct Answer: C Section: (none) Explanation
Explanation/Reference:
Explanation:
QUESTION 69
How does LLQ ensure that voice traffic is always expedited?
A. LLQ adds WRED to CBWFQ. This allows delay-sensitive data such as voice to be dequeued and sent first.
B. LLQ uses CBWFQ to prioritize voice traffic and by dequeuing the voice packets so they can be handled first.
C. The strict priority queue has a higher weight than the queues in CBWFQ. This weight allows the delay-sensitive data such as voice to be dequeued and sent first.
D. The LLQ strict priority queue allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic.
Correct Answer: D Section: (none) Explanation
Explanation/Reference: Explanation:
Without Low Latency Queueing, CBWFQ provides weighted fair queueing based on defined classes with no strict priority queue available for real-time traffic. This scheme poses problems for voice traffic that is largely intolerant of delay, especially variation in delay. For voice traffic, variations in delay introduce irregularities of transmission manifesting as jitter in the heard conversation. The Low Latency Queueing feature provides strict priority queueing for CBWFQ, reducing jitter in voice conversations. Configured by the priority command, Low Latency Queueing enables use of a single, strict priority queue within CBWFQ at the class level, allowing you to direct traffic belonging to a class to the CBWFQ strict priority queue.
http://www.cisco.com/en/US/docs/ios/12_0t/12_0t7/feature/guide/pqcbwfq.html
QUESTION 70
Refer to the exhibit.

Drag the appropirate IOS command from the left and drop them in the spaces on the right in order to
configure Cisco Unified Border Element.
The ITSP does not support early offer. Not all boxes are used.

Exhibit:

Select and Place: Correct Answer:
Section: (none) Explanation
Explanation/Reference:

Explanation:
Voice Service Voip
Allow-Connections sip to h323
Allow-Connections h323 to sip
H323
Call Start Interwork
SIP

Configuring an IP IP Gateway:
Call direction and translation section
voice service voip – Enters VoIP voice-service configuration mode allow-connections from-type to
to-type – Allows connections between specific types of endpoints in an Cisco Unified Border
Element. Arguments are as follows:
·from-type – Type of connection. Valid values: h323, sip. ·to-type – Type of connection. Valid values:
h323, sip.

Main protocol section
h323
call start interwork – Enables slow-start to fast-start interworking sip

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-config.html

QUESTION 71

Refer to the exhibit.

Drag the signaling methods from the left and drop them in the correct position in the graphic on the right. Some method are used more than once, and some method may not be used at all.
Select and Place:

Correct Answer:

Section: (none) Explanation
Explanation/Reference:

Explanation:
The one thing that distinguishes a PRI from other interfaces is the fact that the data that is received from the PSTN on the D-channel and needs to be carried in its raw form back to the Cisco CallManager to be processed. The gateway does not process or change this signalling data, it simply passes it onto the Cisco CallManager through TCP port 2428. The gateway is still responsible for the termination of the Layer 2 data. That means that all the Q.921 data-link layer connection protocols are terminated on the gateway, but everything above that (Q.931 network layer data and beyond) is passed onto the Cisco CallManager. This also means that the gateway does not bring up the D-channel unless it can communicate with Cisco CallManager to backhaul the Q.931 messages contained in the D-channel.
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00801da84e.shtml
QUESTION 72
The voice gateway selects an inbound VoIP dial peer by matching the information elements in the message with the dial-peer attributes. From the list on the left, drag the elements to the right and drop

Select and Place:

Correct Answer: Section: (none) Explanation

Explanation/Reference: QUESTION 73

Drag the delay type on the left and drop it on the correct description on the right.

Select and Place:

Correct Answer:

Section: (none) Explanation
Explanation/Reference:

Processing Delay: Coder delay is the time taken by the digital signal processor (DSP) to compress a block of PCM samples. This is also called processing delay (n). This delay varies with the voice coder used and processor speed. Serialization Delay: Serialization delay (n) is the fixed delay required to clock a voice or data frame onto the network interface. It is directly related to the clock rate on the trunk. Dejitter Buffer: Because speech is a constant bit-rate service, the jitter from all the variable delays must be removed before the signal leaves the network. In Cisco router/gateways this is accomplished with a de-jitter (n) buffer at the far-end (receiving) router/gateway. The de-jitter buffer transforms the variable delay into a fixed delay. It holds the first sample received for a period of time before it plays it out. This holding period is known as the initial play out delay. DSP Delay: The time the packet spends inside the DSP is known as DSP Delay. Sampling, Encoding, Decoding etc. takes place inside the DSP. Queuing Delay: After the compressed voice payload is built, a header is added and the frame is queued for transmission on the network connection. Voice needs to have absolute priority in the router/gateway. Therefore, a voice frame must only wait for either a data frame that already plays out, or for other voice frames ahead of it. Essentially the voice frame waits for the serialization delay of any preceding frames in the output queue. Queuing delay (.n) is a variable delay and is dependent on the trunk speed and the state of the queue. There are random elements associated with the queuing delay. Propagation Delay: Caused by the length a signal must travel via light in fiber or electrical impulse in copper-based networks http://www.cisco.com/en/US/tech/tk652/tk698/technologies_white_paper09186a00800a8993.shtml
QUESTION 74
Drag the components that make up Cisco Fax Relay and T.38 from the left and drop them under the

Select and Place:

Correct Answer: Section: (none) Explanation

Explanation/Reference:

Cisco fax relay is the oldest method of supporting fax on Cisco IOS gateways and has been supported since Cisco IOS Release 11.3. Cisco fax relay uses Real-Time Transport Protocol (RTP) as the method of transport. In Cisco fax relay mode, gateways terminate T.30 fax signaling by spoofing a virtual fax machine to the locally attached fax machine. The gateways use a Ciscoproprietary fax-relay RTP-based protocol to communicate between them.
T.38 Fax Relay provides an ITU-T standards-based method and protocols for fax relay. Data is packetized and encapsulated according to the T.38 standard. The encoding of the packet headers and the mechanism to switch from VoIP mode to fax relay mode are clearly defined in the specification. http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/voice_troubleshooting/old/vts_fax.html
QUESTION 75
Drag the function that are associated with H.245 from the list on the left ot the boxes on the right.

Select and Place:

Correct Answer:

Section: (none) Explanation
Explanation/Reference:

H.225 is responsible only for setting up the call and routing it to the proper destination. H.225 does not have any mechanism for exchanging capabilities or setting up and tearing down media streams. The called
H.323 device is responsible for sending the IP address and port number that are used to establish the TCP connections for H.245 signaling. This information can be sent by the called device in either the Alerting or Connect message. When the originating H.323 device receives the IP address and port number for H.245 negotiations, it initiates a second TCP connection to carry out the necessary capabilities exchange and logical channel negotiations. This TCP session is primarily used to do four things:Master/slave determination-This is used to resolve conflicts that might exist when two endpoints in a call request the same thing, but only one of the two can gain access to the resource at a time. Terminal capabilities exchange-This is one of the most important functions of the H.245 protocol. The two most important capabilities are the supported audio codecs and the basic audio calls. Logical channel signaling-This indicates a one-way audio stream. With H.323 version 2, it is possible to open and close logical channels in the middle of a call. Because H.245 messages are independent of the H.225 signaling, a call can still be connected in H.225 even if no logical channels are open. This is typical with such features as hold, transfer, and conference.DTMF relay-Because voice networks typically do not carry DTMF tones inband because of compression issues, these tones are carried on the signaling channel. Ensure that the type of DTMF relay configured on your gateway is compatible with your gatekeeper.
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/voice_troubleshooting/old/vts_h323.html#wp1068085
QUESTION 76

Select and Place:

Correct Answer:

Section: (none) Explanation
Explanation/Reference:

The Real-Time Transport Protocol (RTP) is an Internet protocol standard that specifies a way for programs to manage the real-time transmission of multimedia data over either unicast or multicast network services. RTP is commonly used in Internet telephony applications. RTP does not in itself guarantee real-time delivery of multimedia data; it does, however, provide the wherewithal to manage the data as it arrives to best effect. RTP combines its data transport with a control protocol (RTCP), which makes it possible to monitor data delivery for large multicast networks. When protocols are used in conjunction, RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number. Monitoring allows the receiver to detect if there is any packet loss and to compensate The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications. Since RTP is closely related to RTCP (Real Time Control Protocol) which can be used to control the RTP session, SRTP also has a sister protocol, called Secure RTCP (or SRTCP); SRTCP provides the same security-related features to RTCP, as the ones provided by SRTP to RTP. Utilization of SRTP or SRTCP is optional to the utilization of RTP or RTCP; but even if SRTP/SRTCP are used, all provided features (such as encryption and authentication) are optional and can be separately enabled or disabled. The only exception is the message authentication feature which is indispensably required when using SRTCP. On slow links, it may be advantageous to compress the IP/UDP/RTP headers using Compressed RTP (cRTP). If you use cRTP then the 40 bytes of overhead incurred by the IP/UDP/RTP headers can typically be compressed down to 2 to 4 bytes (2 bytes when no UDP checksums are sent, and 4 bytes when checksums are sent). Enabling compression on both ends of a low-bandwidth serial link can greatly reduce the network overhead if it carries a lot of RTP traffic. cRTP is supported on serial lines using Frame Relay, HDLC, or PPP encapsulation. It is also supported over ISDN interfaces. CRTP should not be used on links greater than 2 Mbps.
QUESTION 77
Refer to the exhibit.

Drag the appropriate IOS commands from the left and drop them in the space on the right in order to
configure thre dial peer for the Cisco Unified Border Element.
The IP WAN connection between the Cisco Unified Border Elements uses SIP. Not all commands and
spaces are used.

Exhibit:

Select and Place:

Correct Answer:

Section: (none) Explanation
Explanation/Reference:

Please beaware of signaling protocol.If signaling is not the same on incoming and outgoing, we must use media flow-around instat of media flow-through.

QUESTION 78
All call over the IP WAN use G.279. IP phones A and B use Cisco Unified Communications Manager Express. IP phone A is on a call with IP phone B.IP phone A conferences in analog phone C with IP phone
B. Software conference resources are not being used.Drag the appropriate DSP resource for each gateway from the list to the correct locations in the graphic so the call can be complete.

Select and Place: Correct Answer:
Section: (none) Explanation
Explanation/Reference: QUESTION 79

Drag the appropriate IOS commands from the left and drop them in the spaces om the right to create a dial
peer that will match all inbound call and prevent two-stage dialing on a T1 PRI cricuit.
Not all boxes are used and not all options are used.
Select and Place:

Correct Answer:

Section: (none) Explanation
Explanation/Reference:

In the case of Digital Interfaces, when the PBX or central office (CO) switch sends a setup message that contains all the digits necessary to fully route the call, those digits can be mapped to an outbound Voice over IP (VoIP) dial-peer (or hairpin to plain old telephone service (POTS) dialpeer directly). The router/gateway does not present a secondary dial tone to the caller and does not collect digits. It forwards the call directly to the configured destination. In the case of analog interfaces, the user only hears the dial tone once (either local or remote), and then dials the digits and gets through to the destination phone. This is called one stage dialing. When one receives an inbound call from a POTS interface, the Direct Inward Dial (DID) feature in dial-peers enables the router/gateway to use the called number (dialed number identification service (DNIS)) to directly match an outbound dial-peer. When DID is configured on the inbound POTS dial-peer, the called number is automatically used to match the destination pattern for the outbound call leg. The incoming called number command will match the dial-peer that has the DID configured. http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800e00d0.shtml
QUESTION 80

Select and Place: Correct Answer:
Section: (none) Explanation
Explanation/Reference:
DSP delay, Packetization delay, Serialization delay & Dejitter Buffer delay are Fixed delay types. Queuing and Buffering delay & Network delay are Variable Delay types. http://www.cisco.com/en/US/tech/tk652/tk698/ technologies_white_paper09186a00800a8993.shtml

QUESTION 81
Drag the signaling streams to support SIP Early Offer from thre left and drop them in the correct box in the graphic on the right.

Select and Place: Correct Answer:
Section: (none) Explanation
Explanation/Reference:
Call Flow of a Typical sip Session

QUESTION 82
Drag the attributes of a scaleable numbering plan from the left and place them in the boxes on the right.

Select and Place:

Correct Answer:

Section: (none) Explanation
Explanation/Reference:

When designing a large-scale dial plan, Cisco recommends you adhere to the following attributes:
.
Logic distribution: Good dial plan architecture relies on the effective distribution of the dial plan logic among the various components. Devices that are isolated to a specific portion of the dial plan reduce the complexity of the configuration. Each component focuses on a specific task accomplishment. Generally, the local switch or gateway handles details that are specific to the local point of presence (POP). Higher-level routing decisions are passed along to the gatekeepers and PBXs. A well-designed network places the majority of the dial plan logic at the gatekeeper devices.

.
Hierarchical design (scalability): You should attempt to keep the majority of the dial plan logic (routing decisions and failover) at the highest-component level. Maintaining a hierarchical design makes the addition and deletion of number groups more manageable. Scaling the overall network is much easier when configuration changes are made to a single component.

.
Simplicity in provisioning: Keep the dial plan simple and symmetrical when designing a network. Try to keep consistent dial plans on the network by using translation rules to manipulate the local digit dialing patterns. These number patterns are normalized into a standard format or pattern before the digits enter the VoIP core. Putting digits into a standard format simplifies provisioning and dial-peer management.

.
Reduction in postdial delay: Consider the effects of postdial delay in the network when you design a large-scale dial plan. Postdial delay is the time between the last digit dialed and the moment the phone rings at the receiving location. In the PSTN, people expect a short postdial delay and to hear ringback within seconds. The more translations and lookups that take place, the longer the postdial delay becomes. Overall network design, translation rules, and alternate pathing affect postdial delay. Therefore, you should efficiently use these tools to reduce postdial delay.

.
Availability and fault tolerance: Consider overall network availability and call success rates when you design a dial plan. Fault tolerance and redundancy within VoIP networks are most important at the gatekeeper level. By using an alternate path you help provide redundancy and fault tolerance in the network.

.
Conformance to public standards: Different geographical locations might impose restrictions to your dial plan. Therefore, familiarize yourself with any such limitations prior to designing your dial plan.
QUESTION 83
Assume a SIP voice network. Drag each characteristic to the type of SIP call setup the characteristics best describes.

Select and Place:

Correct Answer:

Section: (none) Explanation
Explanation/Reference:

Direct call setup:+ Nonscalable+ UA must keep data on large number of destinations+ Relies on cached information to resolve addresses Redirect Server Call Setup:+ Server reports back to a UA with destination coordinates Proxy Server Call Setup:+ Most dynamic address resolution capability+ All setup messages to through server+ UA incapable of establishing its own sessions http://www.cisco.com/en/US/tech/tk652/tk701/ technologies_configuration_guide_chapter09186a0080163444.html
QUESTION 84

Select and Place: Correct Answer:
Section: (none) Explanation
Explanation/Reference:
The H.323 setup is suspended before the destination phone, triggered by the H.225 alerting message, starts ringing. The RSVP reservation is made in both directions because a voice call requires a two-way speech path and therefore bandwidth in both directions. The terminating gateway ultimately makes the CAC decision based on whether or not both reservations succeed. At that point the H.323 state machine continues either with an H.225 Alerting/Connect (the call is allowed and proceeds), or with an H.225 Reject/Release (call is denied). The RSVP reservation is in place by the time the destination phone starts ringing and the caller hears ringback.

QUESTION 85
Click and drag the feature on the left to the category it belongs to on the right.

Select and Place:

Correct Answer:

Section: (none) Explanation

Gateway: Supports Analog Faxes and Modems on a Voip Network Performs Call Setup and teardown between Voip Networks & the PSTN CUBE: Interconnects segments of the same or different VoIP networks using different media types Interconnects segments of the same or different VoIP networks using different media types
Gateway Functionality : Gateways are responsible Media stream handling and speech path integrity, DTMF relay, Fax relay and pass-through, Digit translation and call processing, Dial peers and codec filtering, Carrier ID handling, Termination and re-origination of signaling and media The Cisco Unified Border Element is a session border controller designed to provide easy, secure, and cost-effective connectivity between independent unified communications networks or network domains for different enterprises. It provides interconnection between incompatible applications within the enterprise network, between different enterprises for business-to-business applications, and between enterprise networks and service provider Session Initiation Protocol (SIP) trunks. The Cisco Unified Border Element provides key session management capabilities, H.323 and SIP interworking functions, and network-to-network interface security and demarcation capabilities. It performs most of the same functions of a public switched telephone network (PSTN)-to-IP gateway but joins two VoIP call legs. Media packets can either flow through (thus hiding the networks from each other) or around the Cisco Unified Border Element platform http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gwoverview_ps10591_TSD_Products_Configuration_Guide_Chapter.html
QUESTION 86

Select and Place:

Correct Answer: Section: (none) Explanation

Explanation/Reference:

2) FXO: off-net 3) FXS: local 4) FXS or switch: on-net 5) E&M, FXO, FXS: PLAR

Explanation
PBX to PBX connections can use T1 or E1 with CAS or PRI: PBX can connect to a network through T1 or E1 lines with channel associated signaling (CAS) or Primary Rate Interface (PRI) signaling. For off-net calls, the typical connection between the router and the PSTN is through FXO port. A local call just needs FXS ports so it is the only choice for this type of call. We can make on-net calls through FXS port (phone directly connected to the router) or FXO port (phone connected to a PBX). The “switch” here means that we can connect an IP phone
through a
switch and place on-net calls through Cisco Unified Communications Manager.
A PLAR call can work with any type of signaling, including E&M, FXO, FXS interfaces.

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Cisco 642-437 Exam Materials, Discount Cisco 642-437 Dumps On Sale