[Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 441-450

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CCIE Collaboration (v1.1)

Question No: 441 – (Topic 4)

Which SIP request is used by Cisco Unified Communications Manager to signal DND status changes to a Cisco 9971 IP Phone?

  1. OPTIONS

  2. NOTIFY

  3. INFO

  4. REFER

  5. UPDATE

Answer: D Explanation:

Cisco Unified Communications Manager supports Do Not Disturb that a SIP device initiates or that a Cisco Unified Communications Manager device initiates. A DND status change gets signaled from a SIP device to Cisco Unified Communications Manager by using the SIP PUBLISH method (RFC3909). A DND status change gets signaled from a Cisco Unified Communications Manager to a SIP device by using a dndupdate Remote-cc REFER request. Cisco Unified Communications Manager can also publish the Do Not Disturb status for a device, along with the busy and idle status for the device.

Reference: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/6_1_1/ccmfeat/cmfsgd6 11/fsdnd.html

Question No: 442 – (Topic 4)

N NO: 61

Which statement about using the Answer File Generator to load a Cisco Unified Communications virtual machine is true?

  1. You must copy the output text to a file named platformConfig.txt.

  2. Each host should be copied to its own configuration file.

  3. The answer file can be used only when performing the new identity process to load the Cisco Unified Communications virtual machines.

  4. The configuration file should be placed inside an ISO file and mounted on the virtual machine.

Answer: B

e: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/install/9_0_1/CUCM_BK_I87B 437D_00_installing-cucm-90/CUCM_BK_I87B437D_00_installing-cucm- 90_chapter_0100.html#CUP0_TK_G0262E75_00

Question No: 443 – (Topic 4)

You are assisting a customer to troubleshoot a SIP early-offer problem with a SIP service provider. You have enabled Cisco CallManager trace and set the debug trace level to Detailed for SIP Call Processing trace on their standalone Cisco Unified Communications Manager 9.1 system. Using the RTMT tool, your customer has remote browsed to the Cisco UCM and asked you which trace file to download.

What is the trace file name syntax in which detailed SIP messages are logged?

  1. SDL

  2. SDI

  3. CCM

  4. Call logs

  5. Traces

Answer: A Explanation:

SDL files log SIP messages from CCM.

Question No: 444 – (Topic 4)

ON NO: 42

Which SIP request is used by a Cisco 9971 IP Phone to signal DND status changes to Cisco Unified Communications Manager?

  1. REGISTER

  2. NOTIFY

  3. INFO

  4. PUBLISH

  5. UPDATE

Answer: D Explanation:

Cisco Unified Communications Manager supports Do Not Disturb that a SIP device initiates or that a Cisco Unified Communications Manager device initiates. A DND status change gets signaled from a SIP device to Cisco Unified Communications Manager by using the SIP PUBLISH method (RFC3909). A DND status change gets signaled from a Cisco Unified Communications Manager to a SIP device by using a dndupdate Remote-cc REFER request. Cisco Unified Communications Manager can also publish the Do Not Disturb status for a device, along with the busy and idle status for the device.

Reference: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/6_1_1/ccmfeat/cmfsgd6 11/fsdnd.html

Question No: 445 – (Topic 4)

Which method does a Cisco Unified 9971 phone use to send keep-alive messages to Cisco Unified Communications Manager?

  1. SIP NOTIFY with Event set to keep-alive

  2. SIP OPTIONS

  3. SIP REGISTER with Expires set to zero

  4. SCCP StationRegister

  5. SCCP StationServerReq

Answer: C Explanation:

Phone registers with primary and establishes keepalive connection with secondary. Expires = 0 keepalive mechanism allows Cisco SIP Phones to more closely resemble the failover / fallback behavior of SCCP.

Question No: 446 – (Topic 4)

Which device is the initiator of a StationD message in a Cisco Unified Communications Manager SDI trace?

  1. SCCP IP phone

  2. SIP IP phone

  3. Cisco Unified Communications Manager

  4. MGCP analog gateway

  5. digital voice gateway

Answer: C Explanation:

All messages to and from a skinny device are preceded by either the words StationD or StationInit. StationD messages are sent from call manager to IP phone. Skinny message transmission such as this between the IP phone and CallManger occurs for every action undertaken by the IP phone, includinginitialization, registration, on-hook, off-hook, dialing of the digits, key press on the phone, and so much more.

Question No: 447 – (Topic 4)

Which Cisco Unified Communications Manager partition will be associated with a directory URI that is configured for an end user with a primary extension?

  1. null

  2. none

  3. directory URI

  4. default

  5. any partition that the Cisco Unified Communications Manager administrator desires

Answer: C Explanation:

Cisco Unified Communications Manager supports dialing using directory URIs for call addressing. Directory URIs look like email addresses and follow the username@host format where the host portion is an IPv4 address or a fully qualified domain name. A directory URI is a uniform resource identifier, a string of characters that can be used to identify a directory number. If that directory number is assigned to a phone, Cisco Unified Communications Manager can route calls to that phone using the directory URI. URI dialing is available for SIP and SCCP endpoints that support directory URIs.

Question No: 448 – (Topic 4)

Which two applications can connect directly with a Cisco Unified Communications Manager Session Management Edition cluster? (Choose two.)

  1. Cisco Unity

  2. Cisco Unified Meeting Place Express

  3. Cisco Unified Contact Center Enterprise

  4. Cisco Unified Contact Center Express

  5. Cisco Unified Communications Manager Attendant Console

  6. Cisco Emergency Responder

Answer: A,B Explanation:

The deployment of a Unified CM Session Management Edition enables commonly used applications such as conferencing or videoconferencing to connect directly to the Session Management cluster, thus reducing the overhead of managing multiple trunks to leaf

systems. Cisco Unity or other voicemail systems can be deployed at all sites and integrated into the Unified CM cluster.

Question No: 449 – (Topic 4)

Which statement describes the Maximum Serving Count service parameter of the Cisco TFTP service on Cisco Unified Communications Manager?

  1. It specifies the maximum number of files in the TFTP server disk storage.

  2. It specifies the maximum number of TFTP client requests to accept and to serve files at a given time.

  3. It specifies the maximum file support by the Cisco TFTP service.

  4. It specifies the maximum file counts, in cache as well as in disk, that are supported by the Cisco TFTP service.

  5. It specifies the maximum number of TFTP client requests to accept and to serve files in a 120-minute window.

Answer: B Explanation:

This parameter specifies the maximum number of client requests to accept and to serve files at a time. Specify a low value if you are serving files over a low bandwidth connection. You can set it to a higher number if you are serving small files over a large bandwidth connection and when CPU resources are available, such as when no other services run on the TFTP server. Use the default value if the TFTP service is run along with other Cisco CallManager services on the same server. Use the following suggested values for a dedicated TFTP server: 1500 for a single-processor system and 3000 for a dual-processor system. If the dual-processor system is running Windows 2000 Advanced Server, the serving count can be up to 5000.

This is a required field.

->Default: 200.

->Minimum: 1.

->Maximum: 5000.

Question No: 450 – (Topic 4)

What is the amount of audio bandwidth, in kilobits per second, that is used in the Cisco Unified Communications Manager location bandwidth calculation for a G.728 call?

  1. 8

  2. 16

  3. 24

  4. 29

  5. 80

Answer: B Explanation: Explanation:

G.728-Low-bit-rate codec that video endpoints support.It support kilobits per second

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[Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 431-440

Dumps4Cert 2017 Oct Cisco Official New Released 400-051
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CCIE Collaboration (v1.1)

Question No: 431 – (Topic 4)

Which two host portion format conditions are true for directory URI on Cisco Unified Communications Manager? (Choose two.)

  1. It is case sensitive.

  2. It cannot start with a hyphen.

  3. It must have at least one character.

  4. It supports IPv4 or IPv6 addresses, or fully qualified domain names.

  5. It cannot end with a hyphen.

  6. It supports the amp; character.

Answer: B,E Explanation:

Cisco Unified Communications Manager supports the following formats in the host portion of a directory URI (the portion after the @ symbol):

->Supports IPv4 addresses or fully qualified domain names.

->Accepted characters are a-z, A-Z ,0-9, hyphens, and dots.

->The host portion cannot start or end with a hyphen.

->The host portion cannot have two dots in a row.

->Minimum of two characters.

->The host portion is not case sensitive.

Reference: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/bat/9_1_1/CUCM_BK_C271A6 9D_00_cucm-bulk-administration-guide-91/CUCM_BK_C271A69D_00_cucm-bulk- administration-guide-91_chapter_01001110.html

Question No: 432 – (Topic 4)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming the calling SIP phone is associated with a SIP Dial Rule with a pattern value of 2001, which statement about the call setup process of this call is true?

  1. Each digit will arrive at Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML event, and Cisco Unified Communications Manager will extend the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of service configuration on both IP phones.

  2. Each digit will arrive at Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML event. When the collected digits match the extension of the SCCP IP phone, Cisco Unified Communications Manager will extend the call only if the class of service configuration on both phones permits this action.

  3. As soon as the user selects the Dial softkey, the SIP IP phone will forward all digits to Cisco Unified Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of service configuration on both IP phones.

  4. As soon as the user selects the Dial softkey, the SIP IP phone will forward all digits to Cisco Unified Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend the call only if class of service configuration on both phones permits this action.

  5. The SIP IP phone will wait for the interdigit timer to expire, and then send all digits to Cisco Unified Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of service configuration on both IP phones.

Answer: D Explanation:

Cisco Type B SIP Phones offer functionality based SIP INVITE Message. Every key the end user presses triggers an individual SIP message. The first event is communicated with a SIP INVITE, but subsequent messages use SIP NOTIFY messages. The SIP NOTIFY

messages send KPML events corresponding to any buttons or soft keys pressed by the user. Cisco Type B SIP IP Phones with SIP dial rules operate in the same manner as Cisco Type A phones with dial rules.

Question No: 433 – (Topic 4)

Which two applications must be connected to a leaf cluster in a Cisco Unified Communications Manager Session Management Edition deployment? (Choose two.)

  1. Cisco Unified Meeting Place

  2. Cisco Unified Contact Center Express

  3. H.323-based video conferencing systems

  4. Cisco Unity

  5. Cisco Unified Communications Manager

  6. fax servers

Answer: B,E Explanation:

The deployment of a Unified CM Session Management Edition enables commonly used applications, such as conferencing or videoconferencing to connect directly to the session management cluster, which reduces the overhead of managing multiple trunks to leaf systems.

Unified CM Session Management Edition supports the following applications:

Dumps4Cert 2017 PDF and VCE

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Unity, Unity Connection

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Meeting Place, Meeting Place Express

Dumps4Cert 2017 PDF and VCE

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SIP and H.323-based video conferencing systems

Dumps4Cert 2017 PDF and VCE

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Third Party voice mail systems

Dumps4Cert 2017 PDF and VCE

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Dumps4Cert 2017 PDF and VCE

Fax servers

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Cisco Unified Mobility

The following applications must connect to the leaf cluster:

Cisco Unified Presence Server

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Dumps4Cert 2017 PDF and VCE

Attendant Console

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Dumps4Cert 2017 PDF and VCE

Manager Assistant

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IP IVR

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Cisco Voice Portal

Reference: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/session_mgmt/deploy/8_5_1/o verview.html

Question No: 434 – (Topic 4)

When the Cisco Unified Communications Manager service parameter quot;Auto Call Pickup Enabledquot; is selected, which two softkeys on an IP phone connect you to an incoming call? (Choose two.)

  1. Pickup

  2. Gpickup

  3. CallBack

  4. Select

  5. Join

Answer: A,B Explanation:

Pickup softkey is used to receive a call that is ringing in another phone within the same pickup group and Gpickupsoftkey is used to receive calls that are ringing but that phone is another pickup group.

Question No: 435 – (Topic 4)

accept and use from a DHCP server?

  1. 2

  2. 3

  3. 4

  4. 5

Answer: A

Reference: http://www.techtronicssolution.com/blog/?p=1201

Question No: 436 – (Topic 4)

What is the maximum length of any numeric geographic area address in ITU recommendation E.164?

  1. 15

  2. 18

  3. 21

  4. 22

  5. 25

Answer: A Explanation:

E.164 defines a general format for international telephone numbers. Plan-conforming numbers are limited to a maximum of 15 digits. The presentation of numbers is usually prefixed with the character (plus sign), indicating that the number includes the international country calling code (country code), and must typically be prefixed when dialing with the appropriate international call prefix, which is a trunk code to reach an international circuit from within the country of call origination.

Question No: 437 – (Topic 4)

How many destinations can be configured for a SIP trunk on a Cisco Unified

Communications Manager 9.1 system when the destination address is an SRV?

  1. 1

  2. 2

  3. 3

  4. 8

  5. 16

Answer: A Explanation:

SIP trunks can be configured with up to 16 destination IP addresses, 16 fully qualified domain names, or a single DNS SRV entry.

Reference: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/trunks.html

Question No: 438 – (Topic 4)

When IP phone A was provisioned in a Cisco Unified Communications Manager, 2001 was configured as the directory number for its first line. Also, bob@cisco.com was defined as the only directory URI on the Directory Number configuration page for this line. A few days later, an end user was created in the same Cisco Unified Communications Manager and was associated with the same phone with the primary extension set to 2001. Also, bobby@cisco.com was defined as a directory URI for that end user.

Which option about the primary directory URI for IP phone A is true?

  1. bob@cisco.com

  2. bobby@cisco.com

  3. It depends on which radio button was selected next to the Directory URI entries on the Directory Configuration page.

  4. Both are primary directory URIs in a manner like a shared line for DNs.

  5. Neither are primary directory URIs for IP phone A.

Answer: B

Reference: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_ BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-

guide_chapter_0101111.html

Question No: 439 – (Topic 4)

Which statement about the effective path in the Enhanced Location Call Admission Control mechanism on Cisco Unified Communications Manager is true?

  1. It is a sequence of links and intermediate locations that connect a pair of locations.

  2. It is used to define the bandwidth that is available between locations.

  3. Only one effective path is used between two locations.

  4. There could be multiple effective paths between a pair of locations.

  5. It logically represents the WAN link.

Answer: C Explanation:

The effective path is the path used by Unified CM for the bandwidth calculations, and it has the least cumulative weight of all possible paths. Weights are used on links to provide a quot;costquot; for the quot;effective pathquot; and are pertinent only when there is more than one path between any two locations.

Question No: 440 – (Topic 4)

Which design restriction applies to Cisco Unified Communications Manager Session Management Edition clustering over the WAN deployment with extended round-trip times in Cisco Unified CM 9.1 and later releases?

  1. SIP and H.323 intercluster trunks are supported.

  2. Only SIP trunk is supported.

  3. SIP trunks and H.323 gateways are supported.

  4. A minimum of 1.544 Mb/s bandwidth is required for all traffic between any two nodes in the cluster.

  5. Only RSVP agents can be configured and registered to the SME cluster as media resources.

Answer: B Explanation:

Using only SIP trunks in the SME cluster allows you to deploy a quot;media transparentquot; cluster where media resources, when required, are inserted by the end or leaf Unified Communications system and never by SME. Using only SIP trunks also allows you to use extended round trip times (RTTs) between SME nodes when clustering over the WAN.

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[Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 421-430

Dumps4Cert 2017 Oct Cisco Official New Released 400-051
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CCIE Collaboration (v1.1)

Question No: 421 – (Topic 4)

When neither the active or standby Location Bandwidth Manager in the configured LBM group is available, what will the Cisco CallManager service on a subscriber Cisco Unified Communications Manager server do to make location CAC decisions?

  1. It will attempt to communicate with the first configured member in the Location Bandwidth Manager hub group.

  2. It will use the Call Treatment When No LBM Available service parameter with the default action to allow calls.

  3. It will use the Call Treatment When No LBM Available service parameter with the default action to reject calls.

  4. It will attempt to communicate with the local LBM service for location CAC decisions.

  5. It will allow all calls until communication is reestablished with any configured servers in the LBM group.

Answer: D Explanation:

By default the Cisco CallManager service communicates with the local LBM service; however, LBM groups can be used to manage this communication. LBM groups provide an active and standby LBM in order to create redundancy for Unified CM call control.

Question No: 422 – (Topic 4)

Router A and router B are Cisco IOS routers with hardware CFB resources that are registered to the same Cisco Unified Communications Manager server. Which Media Resource Group and Media Resource Group List configuration should be implemented if an administrator wants to make sure that all provisioned DSPs on router A are consumed

before router B#39;s DSP is used?

  1. Router A#39;s CFB and router B#39;s CFB should each be configured in its own MRG. Both MRGs should then be grouped into the same MRGL, but the MRG that contains router A#39;s CFB should be listed in higher order than the MRG that contains router B#39;s CFB. Finally, associate the MRGL to all conference resource consumers.

  2. Router A#39;s CFB and router B#39;s CFB should each be configured in its own MRG. Both MRGs should then be further separated into different MRGLs. Finally, associate the MRGL that contains router A#39;s CFB in higher order than router B#39;s CFB to all conference resource consumers.

  3. Router A#39;s CFB and router B#39;s CFB should both be configured in the same MRG with router A#39;s CFB listed higher than that of router B. Then associate the MRG with an MRGL and apply it to all conference resource consumers.

  4. Router A#39;s CFB and router B#39;s CFB should both be configured in the same MRG. Make sure router A#39;s CFB is listed in a higher alphabetical order than router B#39;s CFB. Then associate the MRG with an MRGL and apply it to all conference resource consumers.

  5. Router A#39;s CFB and router B#39;s CFB should both be configured in the same MRG. Use Cisco Unified Communications Manager service parameters to assign a higher priority to router A#39;s CFB. Then associate the MRG with an MRGL and apply it to all conference resource consumers.

Answer: A

Question No: 423 – (Topic 4)

Which option is a characteristic of the Enhanced Location Call Admission Control mechanism on Cisco Unified Communications Manager?

  1. It accounts for network protocol rerouting.

  2. It accounts for network downtime and failures.

  3. It supports dynamic bandwidth adjustments based on WAN topology changes.

  4. It supports asymmetric media flows such that different bit rates in each direction are deducted accordingly.

  5. Unidirectional media flows are deducted as if they were bidirectional.

Answer: E Explanation:

Network Modeling with Locations, Links, and Weights

Enhanced Location CAC is a model-based static CAC mechanism. Enhanced Location CAC involves using the administration interface in Unified CM to configure Locations and

Links to model the quot;Routed WAN Networkquot; in an attempt to represent how the WAN network topology routes media between groups of endpoints for end-to-end audio, video, and immersive calls. Although Unified CM provides configuration and serviceability interfaces in order to model the network, it is still a quot;staticquot; CAC mechanism that does not take into account network failures and network protocol rerouting. Therefore, the model needs to be updated when the WAN network topology changes. Enhanced Location CAC is also call oriented, and bandwidth deductions are per-call not per-stream, so asymmetric media flows where the bit-rate is higher in one direction than in the other will always deduct for the highest bit rate. In addition, unidirectional media flows will be deducted as if they were bidirectional media flows.

Reference: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/cac.htm l

Question No: 424 – (Topic 4)

Which two Cisco Unified Communications Manager SIP profile configuration parameters for a SIP intercluster trunk are mandatory to enable end-to-end RSVP SIP Preconditions between clusters? (Choose two.)

  1. Set the RSVP over SIP parameter to Local RSVP.

  2. Set the RSVP over SIP parameter to E2E.

  3. Set the SIP Rel1XX Options parameter to Disabled.

  4. Set the SIP Rel1XX Options parameter to Send PRACK If 1xx Contains SDP.

  5. Set the SIP Rel1XX Options parameter to Send PRACK for All 1xx Messages.

  6. Check the Fall Back to Local RSVP check box.

Answer: B,D Explanation:

Each Unified Communications Manager cluster and Unified CME should have the same configuration information. For example, Application ID should be the same on each Unified Communications Manager cluster and Unified CME. RSVP Service parameters should be the same on each Unified Communications Manager cluster.

Question No: 425 – (Topic 4)

What is the maximum number of Cisco Unified Communications Manager subscriber pairs in a megacluster deployment?

  1. 4

  2. 8

  3. 12

  4. 16

  5. 32

Answer: B Explanation:

There can be up to 8 pairs of subscribers, 16 subscribers total and must be in a 1:1 redundancy mode (8 active, 8 standby).

Question No: 426 – (Topic 4)

Which two call processing features have a lower priority than the Do Not Disturb settings on a Cisco IP phone? (Choose two.)

  1. park reversion for a locally parked call

  2. hold reversion

  3. intercom

  4. pickup notification

  5. terminating side of a call back

  6. originating side of a call back

Answer: D,E Explanation:

For the DND Ringer Off option, only visual notification gets presented to the device. For the DND Call Reject option, no notification gets presented to the device.

For the terminating side of the call, Do Not Disturb overrides call back:

->When the phone that terminates the call uses DND Ringer Off, the Callback Available screen will be displayed on the phone after the terminating side goes off hook and on hook.

When the phone that terminates the call has DND Call Reject enabled but the phone

becomes available (goes off hook and on hook), a new screen will be presented to the originating device as “lt;Extensiongt; has become available but is on DND-R”. Callback available notification will be sent only after the terminating side disables DND Call Reject.

Question No: 427 – (Topic 4)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming that the calling SIP phone is not associated with any SIP dial rules, which statement about how digits are forwarded to Cisco Unified Communications Manager for further call processing is true?

  1. Each digit is sent to Cisco Unified Communications Manager in a SIP NOTIFY message KPML event, at the time that the user enters the digit on the keypad.

  2. The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before sending each digit to Cisco Unified Communications Manager as a separate KPML event in a SIP NOTIFY message.

  3. The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before sending all digits to Cisco Unified Communications Manager in a SIP INVITE message.

  4. The SIP IP phone will wait for the interdigit timer to expire or for the Dial softkey to be selected before sending the first digit in a SIP INVITE and the subsequent digits in SIP INFORMATION messages.

  5. The SIP IP phone will send all digits to Cisco Unified Communications Manager in a SIP INVITE message as soon as the fourth digit is pressed.

Answer: A Explanation:

KPML procedures use a SIP SUBSCRIBE message to register for DTMF digits. The digits themselves are delivered in NOTIFY messages containing an XML encoded body. And it is Out of Band DTMF

Question No: 428 – (Topic 4)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

A user is going through a series of dialing steps on a SIP Type A IP phone to call a SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assume that the calling SIP phone is not associated with any SIP dial rules.

Which statement about how digits are forwarded to the Cisco Unified Communications Manager for further call processing is true?

  1. As each digit is pressed on the SIP IP phone, it is sent to the Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML event.

  2. The SIP IP phone waits for the inter-digit timer expiry and then sends each digit to the Cisco Unified Communications Manager as a separate KPML event in a SIP NOTIFY message.

  3. The SIP IP phone waits for the inter-digit timer expiry or for the Dial soft key to be selected before it sends all digits to the Cisco Unified Communications Manager in a SIP INVITE message.

  4. The SIP IP phone waits for the inter-digit timer expiry, or for the Dial soft key to be selected before it sends the first digit in a SIP INVITE and the subsequent digits in SIP NOTIFY messages.

  5. The SIP IP phone sends all digits to the Cisco Unified Communications Manager in a SIP INVITE message as soon as the fourth digit is pressed.

Answer: C

Explanation:

Because Type A SIP phone with no SIP dial rules sends digit in Enbloc style.All digits are sent to CUCM after the user completes the dialing and press the Dial softkey.

Reference: https://supportforums.cisco.com/document/87236/working-concept-sccp-sip- phones-and-dial-rules.

Question No: 429 – (Topic 4)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

All displayed devices are registered to the same Cisco Unified Communications Manager server and the phones are engaged in an active call. Assume that the provided configurations exist at the phone line level and multicast MOH is disabled cluster wide.

Which description of what will happen when the user of IP phone A presses the Hold soft key is true?

  1. IP phone B receives audio source 2 from MOH server A.

  2. IP phone B receives audio source 3 from MOH server A.

  3. IP phone B receives audio source 2 from MOH server B.

  4. IP phone B receives audio source 3 from MOH server B.

  5. IP phone B receives audio source 1 from MOH server A.

Answer: C Explanation:

Because audio source 2 is in top of the MRGL List and it will be selected locally first.

Question No: 430 – (Topic 4)

Which Cisco Unified Communications Manager deployment model for clustering over the IP WAN mandates a primary and a backup subscriber at the same site?

  1. multisite with centralized call processing

  2. multisite with distributed call processing

  3. local failover

  4. remote failover

  5. remote failover with Cisco Unified Communications Manager Express as SRST

Answer: C Explanation: Explanation:

Clustering Over the IP WAN

You may deploy a single Unified CM cluster across multiple sites that are connected by an IP WAN with QoS features enabled. This section provides a brief overview of clustering over the WAN. For further information, refer to the chapter on Call Processing.

Clustering over the WAN can support two types of deployments:

鈥ocal Failover Deployment Model

Local failover requires that you place the Unified CM subscriber and backup servers at the same site, with no WAN between them. This type of deployment is ideal for two to four sites with Unified CM.

鈥emote Failover Deployment Model

Remote failover allows you to deploy primary and backup call processing servers split across the WAN. Using this type of deployment, you may have up to eight sites with Unified CM subscribers being backed up by Unified CM subscribers at another site.

Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42models.html

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[Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 411-420

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CCIE Collaboration (v1.1)

Question No: 411 – (Topic 3)

When a Cisco IOS gatekeeper receives an LRQ, what is the first step it will take in an attempt to resolve the destination address?

  1. Check to see if the LRQ reject-unknown-prefix flag is set.

  2. Check to see if the destination address matches the technology prefix.

  3. Check to see if the destination address matches the hop-off technology prefix.

  4. Check to see if the destination address matches the remote zone prefix.

  5. Check to see if the LRQ forward-queries flag is set.

Answer: B Explanation:

LRQ – These messages are exchanged between gatekeepers and are used for inter-zone (remote zone) calls. For example, gatekeeper A receives an ARQ from a local zone gateway requesting call admission for a remote zone device. Gatekeeper A then sends an LRQ message to gatekeeper B. Gatekeeper B replies to the LRQ message with either a Location Confirm (LCF) or Location Reject (LRJ) message, which depends on whether it is configured to admit or reject the inter-zone call request and whether the requested resource is registered.

Question No: 412 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

During a QSIG tunneling over SIP call establishment, which two types of SIP messages can the OGW use to tunnel a waiting QSIG message? (Choose two.)

  1. SIP re-INVITE

  2. SIP NOTIFY

  3. SIP INFO

  4. SIP OPTIONS

  5. SIP UPDATE

  6. SIP REFER

Answer: A,E Explanation:

The TGW sends and the OGW receives a 200 OK response-the OGW sends an ACK message to the TGW and all successive messages during the session are encapsulated

into the body of SIP INFO request messages. There are two exceptions:

When a SIP connection requires an extended handshake process, renegotiation, or an update, the gateway may encapsulate a waiting QSIG message into a SIP re-INVITE or SIP UPDATE message during QSIG call establishment.

When the session is terminated, gateways send a SIP BYE message. If the session is terminated by notice of a QSIG RELEASE COMPLETE message, that message can be encapsulated into the SIP BYE message.

Reference: http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15- mt/sip-config-15-mt-book/voi-sip-tdm.html

Question No: 413 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

You received this debug output to troubleshoot a Cisco IOS MGCP gateway problem at a customer site. What is the purpose of this message?

  1. The MGCP gateway uses this message to respond to an RQNT message from Cisco Unified Communications Manager.

  2. The MGCP gateway uses this message to respond to an AUCX message from Cisco Unified Communications Manager.

  3. The MGCP gateway uses this message to respond to an AUEP message from Cisco Unified Communications Manager.

  4. The MGCP gateway uses this message to respond to a DLCX message from Cisco Unified Communications Manager.

  5. The MGCP gateway uses this message to respond to an NTFY message from Cisco Unified Communications Manager.

Answer: C Explanation:

This message requests the status of an endpoint. Information that can be audited with this includes Requested Events,DigitMap,SignalRequests,RequestIdentifier,QuarantineHandling, Notified Entity, Connection Identifiers, Detect Events, Observed Events, Event States, Bearer Information, Restart Method, Restart Delay, ReasonCode, PackageList, Max MGCP Datagram, and Capabilities. The response will include information about each of the items for which auditing info was requested.

Question No: 414 – (Topic 3)

Which statement about the iSAC on Cisco Unified Border Element is true?

  1. It is a narrow-band codec.

  2. It has a fixed frame of 30 milliseconds.

  3. It has an adaptive frame of up to 60 milliseconds.

  4. It is designed to deliver wideband sound quality in high-bit-rate applications only.

  5. It is not yet supported on the Cisco Unified Border Element (CUBE)

  6. It is not yet supported on Cisco Unified Border Element.

Answer: C Explanation:

iSAC-Internet Speech Audio Codec (iSAC) is an adaptive wideband audio codec, specially designed to deliver wideband sound quality with low delay in both low and medium-bit rate applications. Using an adaptive bit rate of between 10 and 32 kb/s, iSAC provides audio quality approaching that of G.722 while using less than half the bandwidth. In deployments with significant packet loss, delay, or jitter, such as over a WAN, iSAC audio quality is superior to that of G.722 due to its robustness. iSAC is supported for SIP and SCCP devices. The Cisco Unified Communications Manager IP Voice Media Streaming App (IPVMSApp), which includes Media Termination Point, Conference Bridge, Music on Hold Server, and Annunciator does not support iSAC. MGCP devices are not supported.

Question No: 415 – (Topic 3)

Which ITU-T recommendation defines the procedures for using more than one video channel in H.320-based systems?

A. H.324 B. H.230 C. H.239 D. H.264 E. H.329

Answer: C Explanation:

H.239 is the ITU standard for a second video channel; this is supported by all the major vendors, and is the only content channel standard supported by Cisco acquired Codian products or Cisco TelePresence Serial Gateway Series products on H.323 video calls. Cisco acquired Codian products need to be configured to enable H.239. Any H.323 endpoint that also supports the H.239 protocol can source this channel, as can a VNC connection, though some endpoints need to be configured to use H.239 instead of their proprietary solution.

Question No: 416 – (Topic 3)

What is the name of the logical channel proposal that is transmitted from the called entity to the calling entity in H.323 Fast Connect?

  1. Forward Logical Channel

  2. Backward Logical Channel

  3. Reverse Logical Channel

  4. Originator Logical Channel

  5. Destination Logical Channel

Answer: C

Explanation:

Unlike the OpenLogicalChannel request used by H.323 for video uni-directional logical channels, the request used by H.324 for opening video bi-directional logical channels specifies the temporalSpatialTradeOff Capability in both the forward and reverse directions-

-in the forwardLogicalChannelParameters.dataTypeandreverseLogicalChannelParameters.dataTy pe components, respectively. The semantics of temporalSpatialTradeOffCapability used in forward LogicalChannelParameters.dataType is described in the previous section. The semantics for its presence in the reverse direction is described in this section.

Question No: 417 – (Topic 3)

Which RAS message is used between two gatekeepers to resolve an alias address?

  1. GRQ

  2. ARQ

  3. IRQ

  4. LRQ

  5. RRQ

Answer: D Explanation:

LRQ – These messages are exchanged between gatekeepers and are used for inter-zone (remote zone) calls. For example, gatekeeper A receives an ARQ from a local zone gateway requesting call admission for a remote zone device. Gatekeeper A then sends an LRQ message to gatekeeper B. Gatekeeper B replies to the LRQ message with either a Location Confirm (LCF) or Location Reject (LRJ) message, which depends on whether it is configured to admit or reject the inter-zone call request and whether the requested resource is registered.

Question No: 418 – (Topic 3)

Which two VoIP protocols use SDP to describe streaming media sessions? (Choose two.)

A. SCCP

B. H.323

  1. SIP

  2. MGCP

  3. RAS

  4. cRTP

Answer: C,D Explanation:

The Session Description Protocol (SDP), defined in RFC 2327, describes the content of sessions, including telephony, Internet radio, and multimedia applications. SDP includes information about [8]:

Dumps4Cert 2017 PDF and VCE

Media streams: A session can include multiple streams of differing content. SDP currently defines audio, video, data, control, and application as stream types, similar to the MIME types used for Internet mail.

Dumps4Cert 2017 PDF and VCE

Addresses: SDP indicates the destination addresses, which may be a multicast address, for a media stream.

Dumps4Cert 2017 PDF and VCE

Ports: For each stream, the UDP port numbers for sending and receiving are specified.

Dumps4Cert 2017 PDF and VCE

Payload types: For each media stream type in use (for example, telephony), the payload type indicates the media formats that can be used during the session.

Dumps4Cert 2017 PDF and VCE

Start and stop times: These apply to broadcast sessions, for example, a television or radio program. The start, stop, and repeat times of the session are indicated.

Dumps4Cert 2017 PDF and VCE

Originator: For broadcast sessions, the originator is specified, with contact information. This may be useful if a receiver encounters technical difficulties.

Topic 4, Cisco Unified Communications Manager (CUCM)

Question No: 419 – (Topic 4)

What is the maximum one-way delay, in milliseconds, between any two Cisco Unified Communications Manager servers in a non-Session Management Edition cluster over an IP WAN?

  1. 20

  2. 40

  3. 80 D. 160 E. 250

Answer: B Explanation:

The maximum one-way delay between any two Unified CM servers should not exceed 40 msec, or 80 msec round-trip time. Propagation delay between two sites introduces 6 microseconds per kilometer without any other network delays being considered. This equates to a theoretical maximum distance of approximately 3000 km for 20 ms delay or approximately 1860 miles. These distances are provided only as relative guidelines and in reality will be shorter due to other delay incurred within the network.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/7x/uc7_0/models.html

Question No: 420 – (Topic 4)

What does a comma accomplish when it is used in a SIP Dial Rule pattern that is associated with a Cisco 9971 IP Phone that is registered to Cisco Unified Communications Manager?

  1. It inserts a 500-millisecond pause between digits.

  2. It causes the phone to generate a secondary dial tone.

  3. It is a delimiter and has no significant dialing impact.

  4. It indicates a timeout value of 5000 milliseconds.

  5. It is an obsolete parameter and will be ignored.

Answer: B

Explanation:

Comma is accepted in speed dial as delimiter and pause. -Comma used to delineate dial string, FAC, CMC, and post connect digits For post connect digits, commas insert a 2 second delay Commas may be duplicated to create longer delays.

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CCIE Collaboration (v1.1)

Question No: 401 – (Topic 3)

Which SIP reason phrase maps to SIP response reason code 181?

  1. Ringing

  2. Call is Being Forwarded

  3. Session in Progress

  4. Unknown Number

  5. Call Does not Exist

Answer: B Explanation:

1xx-Provisional Responses

100 Trying

Extended search being performed may take a significant time so a forking proxy must send a 100 Trying response.

  1. Ringing

    Destination user agent received INVITE, and is alerting user of call.

  2. Call is Being Forwarded

    Servers can optionally send this response to indicate a call is being forwarded.[1]:搂21.1.3

  3. Queued

    Indicates that the destination was temporarily unavailable, so the server has queued the

    call until the destination is available. A server may send multiple 182 responses to update progress of the queue.

  4. Session in Progress

This response may be used to send extra information for a call which is still being set up.

199 Early Dialog Terminated

Can be used by User Agent Server to indicate to upstream SIP entities (including the User Agent Client (UAC)) that an early dialog has been terminated.

Reference: http://en.wikipedia.org/wiki/List_of_SIP_response_codes

Question No: 402 – (Topic 3)

Which two data frame lengths are supported by iLBC? (Choose two.)

  1. 10 milliseconds

  2. 20 milliseconds

  3. 30 milliseconds

  4. 40 milliseconds

  5. 50 milliseconds

  6. 60 milliseconds

Answer: B,C Explanation:

iLBC-Internet Low Bit Rate Codec (iLBC) provides audio quality between that of G.711 and G.729 at bit rates of 15.2 and 13.3 kb/s, while allowing for graceful speech quality degradation in a lossy network due to the speech frames being encoded independently. By comparison, G.729 does not handle packet loss, delay, and jitter well, due to the dependence between speech frames. iLBC is supported for SIP, SCCP, H323, and MGCP devices.

Question No: 403 – (Topic 3)

In Key Press Markup Language, which SIP request is used to deliver the actual DTMF

digits?

  1. SUBSCRIBE

  2. INFO

  3. NOTIFY

  4. INVITE

  5. ACK

Answer: C Explanation:

KPML procedures use a SIP SUBSCRIBE message to register for DTMF digits. The digits themselves are delivered in NOTIFY messages containing an XML encoded body.

Question No: 404 – (Topic 3)

Which SIP request method enables reliability of SIP 1xx response types?

  1. ACK

  2. PRACK

  3. OPTIONS

  4. CANCEL

  5. REGISTER

Answer: B Explanation:

In order to achieve reliability for provisional responses, we do nearly the same thing. Reliable provisional responses are retransmitted by the TU with an exponentialbackoff. Those retransmissions cease when a PRACK message is received. The PRACK request plays the same role as ACK, but for provisional responses. There is an important difference, however. PRACK is a normal SIP message, like BYE. As such, its own

Question No: 405 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

If this SIP call is initiated using delayed offer, which SIP message will UA#2 use to communicate its media capability to UA#1?

  1. INVITE

  2. 180 Ringing

  3. 200 OK

  4. ACK

  5. RTP Media

Answer: C Explanation:

200 OK Indicates the request was successful.

Question No: 406 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

If this SIP call is initiated using early offer, which SIP message will UA#2 use to communicate its media capability to UA#1?

  1. INVITE

  2. 180 Ringing

  3. 200 OK

  4. ACK

  5. RTP Media

Answer: C Explanation:

In Early offer, SIP Send SDP in the invite, the other node will send the SDP in the 200 message.

Question No: 407 – (Topic 3)

Which two SDP content headers can be found in a SIP INVITE message? (Choose two.)

  1. Expires

  2. Contact

  3. Connection Info

  4. Media Attributes

  5. Allow

  6. CSeq

Answer: C,D Explanation:

Connection info is optional field in SDP whether Media attributes decide the codec and media type for that call.

Question No: 408 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

You received this debug output to troubleshoot a Cisco IOS MGCP gateway call quality issue at a customer site. Which statement about this message is true?

  1. The MGCP gateway is responding to an RQNT message from Cisco Unified Communications Manager to poll the call statistics of an active call.

  2. The MGCP gateway is responding to an AUEP message from Cisco Unified Communications Manager to poll the call statistics of a terminating call.

  3. The MGCP gateway is responding to an MDCX message from Cisco Unified Communications Manager during a call setup.

  4. The MGCP gateway is responding to an AUCX message from Cisco Unified Communications Manager about an active call.

  5. The MGCP gateway is responding to a DLCX message from Cisco Unified Communications Manager about a terminating call.

Answer: E Explanation:

DeleteConnection – used by a call agent to instruct a gateway to delete an existing connection. DeleteConnection can also be used by a gateway to release a connection that can no longer be sustained.

Question No: 409 – (Topic 3)

How are DTMF digits transported in RFC 2833?

  1. In the RTP stream with the named telephone events payload format.

  2. In the RTP stream with the regular audio payload format.

  3. In SIP NOTIFY messages.

  4. In SIP INFO messages.

  5. In SIP SUBSCRIBE messages.

Answer: A Explanation:

DTMF digits and named telephone events are carried as part of the audio stream, and MUST use the same sequence number and time-stamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway. The default clock frequency is 8,000 Hz, but the clock frequency can be redefined when assigning the dynamic payload type.

Question No: 410 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

You received this debug output to troubleshoot a Cisco IOS MGCP gateway media-related problem at a customer site. What is the purpose of this message?

  1. The MGCP gateway is responding to an RQNT message from Cisco Unified Communications Manager to poll the media capabilities on its endpoints.

  2. The MGCP gateway is responding to an AUEP message from Cisco Unified Communications Manager to poll the media capabilities on its endpoints.

  3. The MGCP gateway is responding to an AUCX message from Cisco Unified Communications Manager to poll the active calls on its endpoints.

  4. The MGCP gateway is responding to an MDCX message from Cisco Unified Communications Manager during a call setup.

  5. The MGCP gateway is responding to a CRCX message from Cisco Unified Communications Manager during a call setup.

Answer: E

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[Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 391-400

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CCIE Collaboration (v1.1)

Question No: 391 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

If this SIP call is initiated using delayed offer, which SIP message will UA#1 use to communicate its media capability to UA#2?

  1. INVITE

  2. 180 Ringing

  3. 200 OK

  4. ACK

  5. RTP Media

Answer: D Explanation:

In the Delayed Offer process, the calling does not send its offer in the SIP INVITE Message. The callee sends the offer within the SDP fields of its answer (SIP 200 OK). The calling answers within the ACK message.

Question No: 392 – (Topic 3)

Which two compression formats for high-definition video have technical content that is

identical to H.264? (Choose two.)

  1. MPEG-4 Part 10

  2. MPEG-4 Part 14

  3. MPEG-2 Part 7

  4. AVC

  5. VC3

  6. VP8

Answer: A,D Explanation:

MPEG-4 Part 10, also known as MPEG-4 AVC (Advanced Video Coding), is actually defined in an identical pair of standards maintained by different organizations, together known as the Joint Video Team (JVT). While MPEG-4 Part 10 is a ISO/IEC standard, it was developed in cooperation with the ITU, an organization heavily involved in broadcast television standards. Since the ITU designation for the standard is H.264, you may see MPEG-4 Part 10 video referred to as either AVC or H.264. Both are valid, and refer to the same standard.

Question No: 393 – (Topic 3)

Which SIP response is considered a final response?

  1. 183 Session in Progress

  2. 199 Early Dialog Terminated

  3. 200 OK

  4. 180 Ringing

  5. 100 Trying

Answer: C Explanation:

200 OK Indicates the request was successful. Whether other options state the request is still in progress or request is initiated.

Question No: 394 – (Topic 3)

Which element was added to H.225 messages to enable Fast Connect in H.323 version 2?

  1. fastStart

  2. fastConnect

  3. H.245 PDU

  4. User-User Information

  5. Connection Information

Answer: A Explanation:

Fast start allows for H323 media connections to be started at the beginning of a call. This is helpful for ringback scenarios, and also reduces the amount of time calls take to establish media. H245 is still negotiated later, but the actual media can be done earlier through H225 messages.

Question No: 395 – (Topic 3)

Which procedure uses H.225 messages to exchange H.245 Master-Salve Determination information?

  1. H.323 Fast Connect

  2. H.245 tunneling

  3. H.225 tunneling

  4. H.323 early media

  5. H.245 terminal capability set

Answer: B Explanation:

The H.245 protocol is a media control protocol that is a part of H.323 protocol suite. The

H.245 protocol is used primarily to negotiate master-slave relationship between communicating endpoints. These endpoints exchange terminal capabilities and logical channel manipulations (open, close, modify).The H.245 messages can be encapsulated and carried between H.225 controlled endpoints within H.225 messages. This way of quot;piggy-backingquot; an H.245 message to H.225 message is referred to as H245 Tunneling.

The H.245 Tunneling method is optional and negotiable between communicating H.323 endpoints. If both endpoints support this option, usually the H.245 Media Controlled messages are exchanged via the Tunneling method.

Question No: 396 – (Topic 3)

To which SIP response class do the SIP response codes 300 to 399 belong?

  1. Provisional

  2. Client Failure

  3. Server Failure

  4. Successful

  5. Redirection

Answer: E Explanation:

Redirection – further action needs to be taken in order to complete the request. That is what this class implies.

Question No: 397 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

You received this debug output to troubleshoot a Cisco IOS MGCP gateway problem at a customer site. Which statement about this endpoint on the Cisco MGCP gateway is true?

  1. This endpoint is on a T1 Controller 0/1/0.

  2. This endpoint is on an E1 Controller 0/1/0.

  3. This endpoint is on a T1 Controller 0/1/1.

  4. This endpoint is on an E1 Controller 0/1/2.

  5. This endpoint is on an T1 Controller 0/1/2.

Answer: A Explanation:

The s0/Su1/DS1-0 refers to the slot and port information (0/1/0). It is also a DS1 as shown by this output, which means it is a T1 not an E1.

Question No: 398 – (Topic 3)

In a SIP REFER-based call transfer, which SIP message is being used by the recipient to notify the originator that the final recipient was successfully contacted?

  1. 200 OK

  2. NOTIFY with a message body of 200 OK

  3. 202 Accepted

  4. 100 Trying

  5. 200 BYE

Answer: B Explanation:

The Refer method always begins within the context of an existing call and starts with the originator. The originator sends a Refer request to the recipient (user agent receiving the Refer request) to initiate a triggered Invite request. The triggered Invite request uses the SIP URL contained in the Refer-To header as the destination of the Invite request. The recipient then contacts the resource in the Refer-To header (final-recipient), and returns a SIP 202 (Accepted) response to the originator. The recipient also must notify the originator of the outcome of the Refer transaction-whether the final-recipient was successfully or unsuccessfully contacted. The notification is accomplished using the Notify Method, SIP’s event notification mechanism. A Notify message with a message body of SIP 200 OK indicates a successful transfer, while a body of SIP 503 Service Unavailable indicates an unsuccessful transfer. If the call was successful, a call between the recipient and the final-recipient results.

Question No: 399 – (Topic 3)

Which two types of devices on Cisco Unified Communications Manager support iSAC? (Choose two.)

  1. MGCP

  2. SIP

  3. SCCP

  4. Music on Hold server

E. H.323

Answer: B,C Explanation:

iSAC-Internet Speech Audio Codec (iSAC) is an adaptive wideband audio codec, specially designed to deliver wideband sound quality with low delay in both low and medium-bit rate applications. Using an adaptive bit rate of between 10 and 32 kb/s, iSAC provides audio quality approaching that of G.722 while using less than half the bandwidth. In deployments with significant packet loss, delay, or jitter, such as over a WAN, iSAC audio quality is superior to that of G.722 due to its robustness. iSAC is supported for SIP and SCCP devices. The Cisco Unified Communications Manager IP Voice Media Streaming App (IPVMSApp), which includes Media Termination Point, Conference Bridge, Music on Hold Server, and Annunciator does not support iSAC. MGCP devices are not supported.

Question No: 400 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

Which DTMF relay method is advertised when the originating SIP gateway sends an INVITE message with a Call-Info header shown?

  1. RFC 2833

  2. SIP INFO

  3. SIP NOTIFY

  4. SIP KPML

  5. In-band audio

Answer: C Explanation:

You can develop user-specific applications that reside on your network entity and have the ability to subscribe for event services supported by the IMG. If the network entity wants the ability to detect an entered DTMF digit (only telephone event of “###” are currently supported) from the TDM-side of a call to the IP side of a call, the entity can subscribe to the IMG for these events and receive SIP NOTIFY events containing the digit event.

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[Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 381-390

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CCIE Collaboration (v1.1)

Question No: 381 – (Topic 3)

Which H.245 information is exchanged within H.225 messages in H.323 Fast Connect?

  1. Terminal Capability Set

  2. Open Logical Channel

  3. Master-Slave Determination

  4. Call Setup

  5. Call Progress

Answer: B Explanation:

With the standard H.245 negotiation, the two endpoints need three round-trips before they agree on the parameters of the audio/video channels (1. master/slave voting, 2. terminal capability set exchange, and finally, 3. opening the logical channels). In certain situations and especially with high-latency network links, this can last too long and users will notice the delay.

Question No: 382 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

Which SIP message header is used to tunnel QSIG messages across the SIP network when the OGW receives a call bound for the TGW?

  1. Content-TypE. application/sdp

  2. Content-TypE. application/qsig

  3. Content-TypE. message/ISUP

  4. Content-TypE. message/external-body

  5. Content-TypE. application/x-q931

Answer: B Explanation:

Tunneling over SIP

The Cisco gateway receives QSIG messages from the PBX side and then identifies the destination of the message (or call). The QSIG messages received from the PBX are encapsulated within SIP messages as Multipurpose Internet Mail Extensions (MIME) bodies and are sent (tunneled) across the IP network to the recipient gateway.

When encapsulating a QSIG message (for switch type primary-qsig) inside a SIP message, Cisco gateways include the QSIG message in a MIME body of the SIP request or response using media type

->application/QSIG:

->Content-Type: application/QSIG

Reference:

http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/empowered-branch- solution/white_paper_c11_459092.html

Question No: 383 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

Which SIP trunk deployment model is shown in this enterprise VoIP topology?

  1. mixed TDM and VoIP

  2. centralized

  3. hybrid

  4. traditional TDM

  5. distributed

Answer: C Explanation:

Hybrid SIP Trunk Model In a hybrid SIP trunk deployment, some of the businesses’ sites conform to a distributed SIP trunk deployment model. In this model each site has direct SIP session connectivity to the IP PSTN, and other sites conform to a centralized SIP trunk deployment, accessing the IP PSTN through a central hub, which has SIP session connectivity to the IP PSTN (Figure 3). The hybrid SIP trunk deployment model may have multiple “central” hubs in different geographic regions, or for specific business functions, such as call centers.

Figure 3 Hybrid SIP Trunk Deployment Mode

Dumps4Cert 2017 PDF and VCE

Reference: http://www.cisco.com/c/dam/en/us/products/collateral/unified-

communications/unified-border-element/cis_45835_cube_assets_wp1e.pdf

Question No: 384 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

How many SIP signaling dialog(s) took place in this SIP message exchange between two SIP user agents?

  1. 1

  2. 2

  3. 3

  4. 4

  5. 5

  6. 6

Answer: A Explanation:

During the establishment, maintenance and termination of a SIP session, signaling messages are exchanged between the two SIP endpoints. There are two different kinds of signaling “conversations” that those messages take part in: transactions and dialogs.

A transaction is a SIP message exchange between two user-agents that starts with a request and ends with its final response (it can also contain zero or more provisional responses in between). For example, during the termination of a SIP session, one user releases the call by sending a BYE request and the other party replies back with a 200 OK response. This message exchange is called a transaction.

But what happens in the case of the INVITE request? The establishment of a SIP session starts basically with an INVITE request and is considered as completed upon the receipt of the ACK. In this case, the transaction starts with the INVITE request and ends with the 200 OK, so the ACK is not part of the transaction. The ACK can be considered as a transaction on its own. However, when the final response to an INVITE is not a 2xx response, then the ACK is considered as part of the transaction. A dialog is a complete exchange of SIP messages between two user-agents. That means that transactions are actually parts of a dialog. For example, in the case of a SIP session establishment, a dialog starts with the INVITE-200 OK transaction, continues with the ACK and ends with the BYE-200 OK transaction.

The picture below depicts the dialog and transactions that take place during the establishment of a SIP session:

Dumps4Cert 2017 PDF and VCE

Note: There can also be subsequent requests that belong to the same dialog, such as a BYE or a re-INVITE message. As out-of-dialog requests are considered messages such as an initial INVITE request for a new session or an OPTIONS message for checking capabilities.

There are different SIP headers/parameters that identify the dialogs and transactions, and they will be analyzed in later posts.

Reference: https://telconotes.wordpress.com/2013/03/13/sip-transactions-vs-dialogs/

Question No: 385 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

A user is going through a series of dialing steps on an SCCP IP phone (extension 1001) to call another SCCP IP phone (extension 2003). Both phones are registered to the same Cisco Unified Communications Manager cluster. Which user inputs are sent from the calling IP phone to the Cisco Unified Communications Manager, in forms of SCCP messages, after the user pressed the Dial softkey? Note that the commas in answer choices below are logical separators, not part of the actual user input or SCCP messages.

  1. A separate SCCP message is sent to Cisco Unified Communications Manager for each of the following user inputs: 2, 0, 0, 3.

  2. A separate SCCP message is sent to Cisco Unified Communications Manager for each of the following user inputs: 2, 0, 1, lt;lt;, 0, 3.

  3. A single SCCP message is sent to Cisco Unified Communications Manager to report that digits 2003 have been dialed.

  4. A single SCCP message is sent to Cisco Unified Communications Manager to report that digits 201lt;lt;03 have been dialed.

  5. A separate SCCP message is sent to Cisco Unified Communications Manager for each of the following user inputs: 2, 0, 1, lt;lt;, 2, 0, 0, 3.

Answer: C Explanation:

After the user delete phone stop the digit by digit dialing and send it as a whole setup.

Question No: 386 – (Topic 3)

Which two SCCP call signaling messages are sent by an IP phone to Cisco Unified Communications Manager? (Choose two.)

  1. SoftKeyEvent

  2. OpenReceiveChannelAck

  3. StartMediaTransmission

  4. SelectSoftKeys

  5. CloseReceiveChannel

  6. StopTone

Answer: A,B Explanation:

This message indicates which soft key was pressed. Upon receipt of this message, CallManager invokes the action associated with the pressed soft key. For example, if Hold was the pressed soft key, CallManager places the active call on user hold. In some trace files you might see a soft key number without the corresponding description. The following list defines each soft key number.

Question No: 387 – (Topic 3)

What is the minimum number of TCP sessions needed to complete a H.323 call between two H.323 gateways using slow start?

  1. 0

  2. 1

  3. 2

  4. 3

  5. 4

Answer: C Explanation:

H.323 has two modes of operation: slow start and fast start. The initiation of a call may proceed in a slow start or fast start in H.323. In a slow start, H.323 signaling consists of Setup, Call Proceeding, Alerting, and Connect steps. After these steps, the H.245 media negotiation is performed. When a call is initiated in H.323 fast start, the H.245 media negotiation is performed within the initial Setup message. With slow start, multiple TCP connections are needed for an H.323 call, such as one H.225 signaling channel and one

H.245 signaling channel if required (minimum of these two).

Question No: 388 – (Topic 3)

Which statement about G.722.1 codec support on Cisco Unified Communications Manager is true?

  1. It is always preferred by Cisco Unified Communications Manager over G.711.

  2. It is a high-complexity wideband codec.

  3. It operates at bit rates of 15.2 and 13.3 kb/s.

  4. It is supported for SIP and SCCP devices.

  5. It is supported for SIP and H.323 devices.

Answer: E Explanation:

G.722.1 is a low-complexity wideband codec operating at 24 and 32 kb/s. The audio quality approaches that of G.722 while using at most half the bit rate. As it is optimized for both speech and music, G.722.1 has slightly lower speech quality than the speech-optimized iSAC codec. G.722.1 is supported for SIP and H.323 devices.

Question No: 389 – (Topic 3)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

You received this debug output to troubleshoot a Cisco IOS MGCP gateway media-related problem at a customer site. What is the purpose of this message?

  1. The MGCP gateway is responding to an RQNT message from Cisco Unified Communications Manager to poll the media capabilities on its endpoints.

  2. The MGCP gateway is responding to an AUEP message from Cisco Unified Communications Manager to poll the media capabilities on its endpoints.

  3. The MGCP gateway is responding to an AUCX message from Cisco Unified Communications Manager to poll the active calls on its endpoints.

  4. The MGCP gateway is responding to an MDCX message from Cisco Unified Communications Manager during a call setup.

  5. The MGCP gateway is responding to a CRCX message from Cisco Unified

Communications Manager during a call setup.

Answer: D Explanation:

See MGCP packet debugging examples and their meanings at the Reference link below.

Reference: Sample of Debug MGCP Packets

http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified- communications-manager-callmanager/42104-debug-mgcp.html

Question No: 390 – (Topic 3)

When a Cisco IOS gatekeeper receives an ARQ from a registered endpoint, what is the first step it will take in an attempt to resolve the destination address?

  1. Check to see if the destination address is locally registered.

  2. Check to see if the destination address matches the technology prefix.

  3. Check to see if the destination address matches the local zone prefix.

  4. Check to see if the destination address matches the remote zone prefix.

  5. Check to see if the destination address matches the default technology prefix.

Answer: B Explanation:

Admission Request (ARQ) and Location Request (LRQ) are the two H.225 Registration, Admission, Status (RAS) messages that trigger a gatekeeper to initiate the call routing decision process.

->ARQ-Local zone messages that are sent by H.323 endpoints (usually gateways)

to the Cisco gatekeeper. Gatekeepers receive ARQs from an endpoint if.

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[Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 371-380

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CCIE Collaboration (v1.1)

Question No: 371 – (Topic 2)

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

Assuming that the administrator has never performed any manual custom uploads, which two file types can be found when you choose Software Upgrades, followed by TFTP File Management on the Cisco Unified Operating System Administration web page for Cisco Unified Communications Manager? (Choose two.)

  1. IP phone configuration files

  2. announcement audio files

  3. ringer files

  4. IP phone license files

  5. sample music-on-hold audio files

  6. softkey template files

Answer: B,C Explanation:

The two file types that we get are Announcement Audio Files and Ringer Files.

Topic 3, Telephony Standards and Protocols

Question No: 372 – (Topic 3)

: 43

Refer to the exhibit.

Dumps4Cert 2017 PDF and VCE

Which SIP response message should the TGW send if it cannot process the tunneled QSIG messages from the OGW?

  1. 405 Method Not Allowed

  2. 406 Not Acceptable

  3. 412 Conditional Request Failed

  4. 415 Unsupported Media Type

  5. 485 Ambiguous

Answer: D Explanation:

Fallback from QSIG Tunneling

In some situations, QSIG tunneling will fail or need to fall back:

  • Remote party does not support multipart MIME body: In this case, the remote side sends a quot;415 Media Not Supportedquot; response. Upon receiving this response, OGW will fall back to normal mode and send an INVITE request without any tunneled content. This procedure helps ensure that at least the basic call will work normally.

  • Remote party does not understand tunneled content: If the remote side does not support the tunneled content, it should drop the tunneled content and continue as a normal call; because all essential parameters are present in the original INVITE, the call can go through without the need for fallback.

    ence: http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/empowered- branch-solution/white_paper_c11_459092.html

    Question No: 373 – (Topic 3)

    Which two SCCP call signaling messages are initiated by Cisco Unified Communications Manager to an IP phone? (Choose two.)

    1. SoftKeyEvent

    2. CloseReceiveChannelAck

    3. CallState

    4. KeypadButton

    5. OpenReceiveChannel

    6. Offhook

    Answer: C,E Explanation:

    Upon receiving an OpenReceiveChannelmessage, the IP phone selects the UDP port number it wants to use to receive RTP packets and reports this information to call manager.

    With the SCCP protocol architecture, the majority of the H.323 processing power resides in an H.323 proxy – the Cisco CallManager. The end stations (IP phones) run the Skinny client, which consumes less processing overhead. The client communicates with CallManager using connection-oriented (TCP/IP-based) communication to establish a call with another H.323-compliant end station. Once Cisco CallManager has established the call, the two H.323 end stations use connectionless (UDP/IP-based) communication for audio transmissions.

    Question No: 374 – (Topic 3)

    Which two statements describe characteristics of Binary Floor Control Protocol? (Choose two.)

    1. Its binary encoding is designed to work in high-bandwidth environments.

    2. It is designed for audio or video conference sessions of three or more participants.

    3. It enables management of shared content resources independent of video streams.

    4. It supports TLS-based authentication.

    5. It supports SIP as well as H.323.

    Answer: C,D

    Explanation:

    BFCP is a deliverable developed as part of the Internet Engineering Task Force (IETF) XCON Centralized Conferencing working group. The IETF XCON working group was formed to focus on delivering a standards-based approach to managing IP conferencing while promoting broad interoperability between software and equipment vendors.

    Question No: 375 – (Topic 3)

    Refer to the exhibit.

    Dumps4Cert 2017 PDF and VCE

    Which user agent has the recipient role in this SIP REFER call transfer?

    1. user agent A

    2. user agent B

    3. user agent C

    4. user agent B and C

    5. user agent A and B

    Answer: B

    Explanation:

    The Refer method has three main roles:

  • Originator-User agent that initiates the transfer or Refer request.

  • Recipient-User agent that receives the Refer request and is transferred to the final- recipient.

  • Final-Recipient-User agent introduced into a call with the recipient.

  • Reference: http://www.cisco.com/c/en/us/td/docs/ios_xr_sw/iosxr_r3- 4/sbc/configuration/guide/sbc_c34/sbc34stx.pdf

    Question No: 376 – (Topic 3)

    What is the minimum number of H.225 messages required to establish an H.323 call with bidirectional media?

    1. 1

    2. 2

    3. 3

    4. 4

    5. 5

    Answer: B Explanation:

    A typical H.245 exchange looks similar to below figure

    After this exchange of messages, the two endpoints (EP) in this figure would be transmitting audio in each direction. The number of message exchanges is numerous, each has an important purpose, but nonetheless takes time.

    For this reason, H.323 version 2 (published in 1998) introduced a concept called Fast Connect, which enables a device to establish bi-directional media flows as part of the

    H.225.0 call establishment procedures. With Fast Connect, it is possible to establish a call with bi-directional media flowing with no more than two messages, like in figure 3.

    Fast Connect is widely supported in the industry. Even so, most devices still implement the complete H.245 exchange as shown above and perform that message exchange in parallel to other activities, so there is no noticeable delay to the calling or called party.

    Dumps4Cert 2017 PDF and VCE

    Question No: 377 – (Topic 3)

    To which SIP response category does 301 Moved Permanently belong?

    1. Provisional

    2. Successful

    3. Redirection

    4. Client Failure

    5. Server Failure

    Answer: C Explanation:

    The 301 response from the Web server should always include an alternative URL to which redirection should occur. If it does, a Web browser will immediately retry the alternative URL. So you never actually see a 301 error in a Web browser, unless perhaps you have a corrupt redirection chain e.g. URL A redirects to URL B which in turn redirects back to URL

    A. If your client is not a Web browser, it should behave in the same way as a Web browser

    i.e. immediately retry the alternative URL.

    Question No: 378 – (Topic 3)

    Which device is the initiator of a StationInit message in a Cisco Unified Communications Manager SDI trace?

    1. Cisco Unified Communications Manager

    2. MGCP gateway

    3. Cisco Music on Hold server

    4. SCCP IP phone

    5. SIP Proxy Server

    Answer: D Explanation:

    Station Init means that an inbound Transmission Control Protocol (TCP) message from a Skinny station reached CallManager. A Skinny station is any endpoint that uses the Skinny protocol to communicate with CallManager.

    Question No: 379 – (Topic 3)

    Which two responses are examples of client error responses in SIP protocol? (Choose two.)

    1. 302 Moved Temporarily

    2. 404 Not Found

    3. 503 Service Unavailable

    4. 502 Bad Gateway

    5. 604 Does Not Exist Anywhere

    6. 408 Request Timeout

    Answer: B,F Explanation:

    Client Error (400 to 499)-Request contains bad syntax or cannot be fulfilled at this server. This class of 400 to 499 contains only error messages.

    Question No: 380 – (Topic 3)

    Refer to the exhibit.

    Dumps4Cert 2017 PDF and VCE

    How many SIP signaling transaction(s) took place in this SIP message exchange between two SIP user agents?

    1. 1

    2. 2

    3. 3

    4. 4

    5. 5

    6. 6

    Answer: C Explanation:

    During the establishment, maintenance and termination of a SIP session, signaling

    messages are exchanged between the two SIP endpoints. There are two different kinds of signaling “conversations” that those messages take part in: transactions and dialogs.

    A transaction is a SIP message exchange between two user-agents that starts with a request and ends with its final response (it can also contain zero or more provisional responses in between). For example, during the termination of a SIP session, one user releases the call by sending a BYE request and the other party replies back with a 200 OK response. This message exchange is called a transaction.

    But what happens in the case of the INVITE request? The establishment of a SIP session starts basically with an INVITE request and is considered as completed upon the receipt of the ACK. In this case, the transaction starts with the INVITE request and ends with the 200 OK, so the ACK is not part of the transaction. The ACK can be considered as a transaction on its own. However, when the final response to an INVITE is not a 2xx response, then the ACK is considered as part of the transaction. A dialog is a complete exchange of SIP messages between two user-agents. That means that transactions are actually parts of a dialog. For example, in the case of a SIP session establishment, a dialog starts with the INVITE-200 OK transaction, continues with the ACK and ends with the BYE-200 OK transaction.

    The picture below depicts the dialog and transactions that take place during the establishment of a SIP session:

    Dumps4Cert 2017 PDF and VCE

    Note: There can also be subsequent requests that belong to the same dialog, such as a BYE or a re-INVITE message. As out-of-dialog requests are considered messages such as an initial INVITE request for a new session or an OPTIONS message for checking capabilities.

    There are different SIP headers/parameters that identify the dialogs and transactions, and they will be analyzed in later posts.

    Reference: https://telconotes.wordpress.com/2013/03/13/sip-transactions-vs-dialogs/

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    [Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 361-370

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    CCIE Collaboration (v1.1)

    Question No: 361 – (Topic 2)

    Which statement about Cisco EnergyWise domain member neighbor formation is true?

    1. Cisco EnergyWise supports static neighbors, but the neighbor relationship is only possible if a noncontiguous domain member and a contiguous domain member have a static neighbor entry pointing to each other.

    2. Cisco EnergyWise static neighbors can be formed even if domain members are not physically contiguous.

    3. Static neighbors can be manually defined on Cisco EnergyWise domain members, but TCP protocols must be used.

    4. Static neighbors can be manually defined on Cisco EnergyWise domain members, but they have a lower priority compared to the autodiscovered members.

    5. Static neighbors can be manually defined on Cisco EnergyWise domain members and the TCP or UDP protocol can be used.

    Answer: B

    Reference: http://www.cisco.com/en/US/docs/solutions/Enterprise/Borderless_Networks/Energy_Mana gement/energywisedg.html?referring_site=smartnavRD#wp554384

    Question No: 362 – (Topic 2)

    Which two statements about the Cisco UC on UCS specs-based virtualization support model are true? (Choose two.)

    1. It has a configuration-based approach.

    2. It has a rule-based approach.

    3. It has less hardware flexibility compared to the third-party server specs-based support model.

    4. It has less hardware flexibility compared to the UC on UCS TRC support model.

    5. VMware vCenter is optional with this support model.

    Answer: B,C

    Reference: http://docwiki.cisco.com/wiki/UC_Virtualization_Supported_Hardware#UC_on_UCS_Teste d_Reference_Configurations

    Question No: 363 – (Topic 2)

    Refer to the exhibit.

    Dumps4Cert 2017 PDF and VCE

    Assuming that the administrator has never performed any manual custom uploads, which two file types can be found when you choose Software Upgrades, followed by TFTP File Management on the Cisco Unified Operating System Administration web page? (Choose two.)

    1. IP phone configuration files

    2. sample music-on-hold audio files

    3. Identity Trust List files

    4. IP phone license files

    5. Mobile Voice Access audio files

    6. softkey template files

    Answer: C,E Explanation:

    We get option for Identity Trust list Files and Mobile Voice Access audio files.

    Question No: 364 – (Topic 2)

    Which four attributes are needed to determine the time to complete a TFTP file transfer process? (Choose four.)

    1. file size

    2. file type

    3. network interface type

    4. round-trip time

    5. packet loss percentage

    6. response timeout

    7. network throughput

    Answer: A,D,E,F Explanation:

    Four attributes that are needed to determine the time to complete TFTP file transfer process is:

    ->File Size

    ->Round-trip time

    ->Packet loss percentage

    ->Response timeout

    Reference: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6882/ps6884/white_paper_c11- 583891_ps10451_Products_White_Paper.html

    Question No: 365 – (Topic 2)

    Which two statements about the Peer Firmware Sharing option for IP phone firmware distribution are true? (Choose two.)

    1. This option uses a parent-child hierarchy in which a firmware image is downloaded by a parent phone to up to three directly associated child phones.

    2. This option must be enabled on Cisco Unified Communications Manager service parameters for Cisco TFTP.

    3. This option mandates that the parent phone and child phones be identical, selected phone models.

    4. This option allows firmware transfers between phones in different subnets, as long as the round-trip delay is less than 5 milliseconds.

    5. This option uses a parent-child hierarchy that must be manually defined by the Cisco Unified Communications Manager administrator.

    6. This option allows falling back to the TFTP server in the Cisco Unified Communications Manager cluster.

    Answer: C,F Explanation:

    Peer Firmware Sharing works by setting up a parent-child hierarchy of the phones in which a firmware image is downloaded by the parent phone to a child phone. The advantage of using Peer Firmware Sharing is that instead of all phones individually retrieving a software image, they pass the image along from one phone to another phone on the same subnet.

    Advantage of PFS:

    ->Hierarchy is automatic

    ->One download per phone model on a subnet

    ->Uses TCP

    ->Fails back to TFTP

    ->Speeds up LAN upgrades

    ->Reduces TFTP CPU load during upgrade

    Question No: 366 – (Topic 2)

    Which statement about the Cisco UC on UCS TRC and the third-party server specs-based virtualization support model is true?

    1. Both the UC on UCS TRC and the third-party servers spec-based support models have rule-based approaches.

    2. The UC on UCS TRC support model has a rule-based approach and the third-party servers spec-based support model has a configuration-based approach.

    3. The UC on UCS TRC support model requires a high level of virtualization experience while the third-party server spec-based support model requires a low to medium level virtualization experience.

    4. VMware vCenter is mandatory for the UC on UCS TRC support model but it is optional for the third-party server spec-based support model.

    5. VMware vCenter is optional for the UC on UCS TRC support model but it is mandatory for the third-party server spec-based support model.

    Answer: E Explanation:

    VMware vCenter is

    ->optional when deploying on UC on UCS Tested Reference Configuration hardware

    ->mandatory when deploying on UC on UCS Specs-based and Third-party Server Specs-based hardware.

    ->vCenter Statistics Level 4 logging is mandatory so that Cisco TAC is able to provide effective support.

    ->Click here for how to configure VMware vCenter to capture these logs. If not configured by default, Cisco TAC may request enabling these settings in order to provide effective support.

    ->Also note that enablement of specific VMware vSphere management features may require vCenter and/or a higher feature Edition of vSphere ESXi.

    ->Cisco Collaboration does not require its own dedicated vCenter.

    ->Note that when VMware vCenter is not required and is not used, then VMware vSphere ESXi#39;s default management interface is its free/included VMware vSphere Client (formerly branded VI Client).

    Reference: http://docwiki.cisco.com/wiki/Unified_Communications_VMware_Requirements

    Question No: 367 – (Topic 2)

    Company ABC is planning to migrate from MCS-hosted Cisco Unified Communications Manager applications to Cisco UC on UCS B-Series servers. Which statement about installation media support is true for this migration?

    1. The install log can be written to a USB flash drive that is attached to the UCS server.

    2. The answer file that is generated by the Answer File Generator (platformConfig.xml) can be read from a USB flash drive to perform an unattended installation on the UCS server.

    3. The Cisco Music on Hold USB audio sound card can be mapped to a virtual USB port on a VMware virtual machine on the UCS server.

    4. The answer file that is generated by the Answer File Generator (platformConfig.xml) can be read from an FLP image that is mounted in a virtual floppy drive.

    5. The Cisco Music on Hold USB audio sound card can be mapped to a virtual serial port on a VMware virtual machine on the UCS server.

    Answer: D Explanation:

    Using the AFG will allow you to get this license mac before even touching the server. It is provided after filling in the main form of the AFG but it can also be found by looking at the last few lines of your platformconfig.xml file.

    Once you have the xml files, you will need to map those to the floppy drive of the VM (no

    usb support on the VM OVA). There are many ways to do this. I simply use a freeware virtual floppy app that I drop the platformconfig.xml file on and then copy the*.flp image out to the datastore. I’ll end up with a directory on my datastore called AFG that has the host named *.flp images that I will use during install. It also serves as archival of these files in the event the server needs to be re-imaged. This is important because the license mac will change if every parameter is not entered exactly as it was prior. If the license mac changes, you will have to go through the process of requesting new license files to be generated.

    Reference: http://angryciscoguy.com/jello/cisco-answer-file-generator-to-the-rescue/

    Question No: 368 – (Topic 2)

    Which protocol does the Cisco Prime LAN Management Solution application use to communicate with Cisco EnergyWise domain members?

    1. UDP broadcast

    2. Cisco Discovery Protocol

    3. UDP unicast

    4. TCP

    5. multicast

    Answer: D Explanation:

    Cisco Prime LMS 4.1 uses TCP port 43440.

    Question No: 369 – (Topic 2)

    Which two mechanisms does Cisco EnergyWise use for neighbor discovery? (Choose two.)

    1. multicast

    2. LLDP-MED

    3. UDP broadcast

    4. Cisco Discovery Protocol

    5. TCP

    Answer: C,D Explanation:

    Cisco EnergyWise Neighbor Discovery Process

    The Cisco EnergyWise neighbor discovery process is the mechanism by which domain members discover each other and populate their Cisco EnergyWise neighbor tables. Cisco EnergyWise queries can subsequently be distributed to all domain members using the neighbor relationships to monitor and control the power usage of devices within a domain. Cisco EnergyWise domain members automatically discover their neighbors through one of two mechanisms:

    ->Cisco EnergyWise UDP broadcast packet

    ->Cisco EnergyWise CDP packets

    UDP broadcast packets are automatically sent out switch ports which support Cisco EnergyWise, regardless of whether the interfaces are configured with the no energywise interface-level command. CDP packets are sent when CDP is configured for the switch ports.

    Reference: http://www.cisco.com/en/US/docs/solutions/Enterprise/Borderless_Networks/Energy_Mana gement/energywisedg.html?referring_site=smartnavRD#wp555927

    Question No: 370 – (Topic 2)

    Which two statements about using the Load Server option for IP phone firmware distribution are true? (Choose two.)

    1. This option must be enabled on at least two servers in a Cisco Unified Communications Manager cluster.

    2. This option must be enabled on Cisco Unified Communications Manager service parameters for Cisco TFTP.

    3. Phone firmware must be manually copied to any applicable load servers.

    4. The load server will not function if its IP address is not in the same subnet as the IP phones.

    5. This option is only available for newer IP phone models.

    6. This option does not accommodate falling back to Cisco TFTP on error.

    Answer: C,F Explanation:

    Choosing the Right Distribution Method

    Which of the three different image-distribution methods discussed so far is the best for a customer deployment? Each method has advantages and disadvantages, and they are summarized in Table 1.

    Table 1. Summary of Distribution Models

    Peer Firmware Sharing Load Server Traditional TFTP Advantages

    • Hierarchy is automatic

    • One download per phone model on a subnet

    • Uses TCP

    • Fails back to TFTP

    • Speeds up LAN upgrades

    • Reduces TFTP CPU load during upgrade

    • Has same download time as LAN image distribution

    • Distributes TFTP load over multiple TFTP servers

    • Proven distribution

    • Default behavior

      Disadvantages

    • Must be enabled on each phone

    • Hierarchy is formed for each phone model

    • Hierarchy is limited to subnet

    • IP must be set on each phone

    • Administrator must manually file copy to load server

    • No failback to TFTP on error

    [Free] 2017(Oct) Dumps4cert Testinsides Cisco 400-051 Dumps with VCE and PDF Download 351-360

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    CCIE Collaboration (v1.1)

    Question No: 351 – (Topic 1)

    Which two SCCP call states support the CallBack softkey? (Choose two)

    1. On Hook

    2. Remote In Use

    3. Connected Transfer

    4. Ring In

    5. Off Hook

    6. Connected Conference

    Answer: A,C

    Question No: 352 – (Topic 1)

    An engineer notices that two Cisco unity Connection servers in a cluster are in split-brain mode. The engineer corrects a network issue that allows the two servers to communicate again. Which two statements describe negative effects of this event? (Choose two)

    1. A user calling in to check their voicemail during the recovery may be informed that their messages are not available.

    2. Message waiting indicators can become out of sync after the split-brain recovery, forcing the administrator to run an MWI Synchronization.

    3. The replication between the nodes becomes defunct, requiring the administrator to run utils cuc cluster activate to re-establish intracluster communication.

    4. A message left on the subscriber server during the outage may be lost during the cluster recovery.

    5. The replication between the nodes becomes defunct, requiring the administrator to run utils cuc cluster renegotiate to re-establish intracluster communication.

    6. The Unity Connection Database can become corrupted, causing the need to reinstall the subscriber server.

    Answer: A,C

    Question No: 353 – (Topic 1)

    Refer to the exhibit.

    Dumps4Cert 2017 PDF and VCE

    Cisco unity connection site A has two locations and Cisco Unity connection Site B has one Location. Which protocol connect the location and servers together for messaging and replication?

    1. 1 SMTP

  • – HTTP/HTTPS, SMTP

  • None

    1. 1 HTTP/HTTPS, SMTP 2 SMTP

      3 None

    2. 1 – HTTP/HTTPS, SMTP 2 – HTTP/HTTPS, SMTP

      3 – HTTP/HTTPS, SMTP

    3. 1 SMTP 1 SMTP

    1 SMTP

    Answer: A

    Topic 2, Cisco Collaboration Infrastructure

    Question No: 354 – (Topic 2)

    What is the maximum number of call-processing subscribers in a standard deployment of a Cisco Unified Communications Manager Session Management Edition cluster?

    1. 3

    2. 4

    3. 5

    4. 8

    5. 16

    Answer: D Explanation:

    There is no deployment difference between CUCM amp; CUCM session management Edition cluster. The only difference is that CUCM SME is designed to support a large number of trunk to trunk connections. Thus, 8 subscribers.

    Question No: 355 – (Topic 2)

    Which definition is included in a Cisco UC on UCS TRC?

    1. storage arrays such as those from EMC or NetApp, if applicable

    2. configuration of virtual-to-physical network interface mapping

    3. step-by-step procedures for hardware BIOS, firmware, drivers, and RAID setup

    4. server model and local components (CPU, RAM, adapters, local storage) at the part number level

    5. configuration settings and patch recommendations for VMware software

    Answer: D Explanation:

    What does a TRC definition include?

    ->Definition of server model and local components (CPU, RAM, adapters, local storage) at the orderable part number level.

    ->Required RAID configuration (e.g. RAID5, RAID10, etc.) – including battery backup cache or SuperCap – when the TRC uses DAS storage

    ->Guidance on hardware installation and basic setup (e.g. click here).

    ->Design, installation and configuration of external hardware is not included in TRC definition, such as:

    ->Configuration settings, patch recommendations or step by step procedures for VMware software are not included in TRC definition.

    ->Infrastructure solutions such as Vblock from Virtual Computing Environment may also be leveraged for configuration details not included in the TRC definition.

    Reference: http://docwiki.cisco.com/wiki/UC_Virtualization_Supported_Hardware#UC_on_UCS_Teste d_Reference_Configurations

    Question No: 356 – (Topic 2)

    Which capability is support by LLDP-MED but not by Cisco Discovery Protocol?

    1. LAN speed discovery

    2. network policy discovery

    3. location identification discovery

    4. power discovery

    5. trust extension

    Answer: A Explanation:

    LLDP-MED supports both LAN speed and duplex discovery. Cisco Discovery Protocol supports duplex discovery only, but this limited support is not seen as a problem because if there is a speed mismatch, LLDP-MED and Cisco Discovery Protocol cannot be exchanged and thus cannot be used to detect the mismatch.

    Question No: 357 – (Topic 2)

    Which definition is included in a Cisco UC on UCS TRC?

    1. required RAID configuration, when the TRC uses direct-attached storage

    2. configuration of virtual-to-physical network interface mapping

    3. step-by-step procedures for hardware BIOS, firmware, drivers, and RAID setup

    4. configuration settings and patch recommendations for VMware software

    5. server model and local components (CPU, RAM, adapters, local storage) by name only; part numbers are not included because they change over time

    Answer: A

    Explanation:

    Definition of server model and local components (CPU, RAM, adapters, local storage) at the orderable part number level.

    ->Required RAID configuration (e.g. RAID5, RAID10, etc.) – including battery backup

    cache or SuperCap – when the TRC uses DAS storage

    ->Guidance on hardware installation and basic setup.

    ->Design, installation and configuration of external hardware is not included in TRC definition, such as:

    ->Configuration settings, patch recommendations or step by step procedures for VMware software are not included in TRC definition.

    ->Infrastructure solutions such as Vblock from Virtual Computing Environment may also be leveraged for configuration details not included in the TRC definition.

    Question No: 358 – (Topic 2)

    In a Cisco EnergyWise domain, which two terms describe a Cisco IP phone? (Choose two.)

    1. endpoint

    2. domain member

    3. child domain member

    4. EnergyWise agent

    5. Cisco power distribution unit

    Answer: A,C

    Reference: http://www.cisco.com/en/US/docs/switches/lan/energywise/phase2_5/ios/configuration/guid e/one_ent.html

    Question No: 359 – (Topic 2)

    Which statement describes a disadvantage of using the Cisco TFTP service to serve IP phone load files?

    1. The Cisco TFTP services can run on only one Cisco Unified Communications Manager server in a cluster.

    2. Because TFTP operates on top of UDP, there is a high risk of corrupted load file delivery at the completion of the TFTP process due to undetected data loss in the network.

    3. If a response is not received in the timeout period, the TFTP server will not resend the data packet.

    4. Packet loss can significantly increase the TFTP session completion time.

    5. Because TFTP operates with an adaptive timeout period, the time to complete the file transfer is unpredictable.

    Answer: D Explanation: Explanation:

    Voice traffic cannot recapture lost packets. Rather than retransmitting a lost network connection, the phone resets and attempts to reconnect its network connection.

    Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/6921_6941_6961/7_1_2/english/ admin/guide/6921trb.html#wp1031181

    Question No: 360 – (Topic 2)

    Which capability is supported by Cisco Discovery Protocol but not by LLDP-MED?

    1. LAN speed and duplex discovery

    2. Network policy discovery

    3. Location identification discovery

    4. Power discovery

    5. Trust extension

    Answer: E Explanation:

    Cisco Discovery Protocol provides an additional capability not found in LLDP-MED that allows the switch to extend trust to the phone. In this case, the phone is now trusted to mark the packets received on the PC port accordingly. This feature can be used to off-load the switch because now it does not need to police the information being received from the phone.

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