Cisco 400-051 Online Practice Questions and Exam Preparation

Cisco 400-051 Online Questions & Answers

• Question 351:

  What is the maximum number of subclusters supported on a Cisco IM and Presence server?

  A. 3
  B. 5
  C. 2
  D. 1
  E. 4
  A. 3

• Question 352:

  A collaboration engineer has been asked to implement secure real-time protocol between a Cisco Unified CM and its SIP gateway. Which option is a consideration for this implementation?

  A. only T.38 and Cisco fax protocol are supported
  B. SIP require the all time be sent in GMT
  C. Call hold RE-INVITE is not supported
  D. SRTP is supported only in cisco IOS 15.x and higher
  B. SIP require the all time be sent in GMT

• Question 353:

  Which three Cisco Unity Connection call handler greetings can be overridden by the internal greeting? (Choose three.)

  A. holiday
  B. alternate
  C. error
  D. busy
  E. closed
  F. standard
  A. holiday
  E. closed
  F. standard

  ---

  This greeting overrides the Standard, Closed, and Holiday greetings but only for internal callers or users defined in Cisco Unity Connection because the mentioned three greetings are defined for externals users.

• Question 354:

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

  A. MGCP
  B. SIP
  C. SCCP
  D. Music on Hold server
  E. H.323
  B. SIP
  C. SCCP

  ---

  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 355:

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

  A. SCCP
  B. H.323
  C. SIP
  D. MGCP
  E. RAS
  F. cRTP
  C. SIP
  D. MGCP

  ---

  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]:

  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.

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

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

  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.

  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.

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

• Question 356:

  Refer to the exhibit.

  

  How many inbound calls can be handled simultaneously between ephone 1 and ephone 2 before a user busy tone is returned?

  A. 6
  B. 7
  C. 8
  D. 9
  E. 11
  A. 6

  ---

  Because hunt stop channel is set to 6 as it enables call hunting to up to six channels of this ephone-dn and remaining 2 channels are available for outgoing call features. References: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/command/reference/cme_cr/cme_e1ht.html

• Question 357:

  Assume that the customer domain is customer.com. Which external DNS SRV record is needed to facilitate service discovery so that the Cisco Jabber client can automatically find the appropriate servers to connect to while off-net?

  A. _cisco-uds._tcp.customer.com
  B. _collab-edge._tls.customer.com
  C. _cisco-uds._tls.customer.com
  D. _collab-edge._tcp.customer.com
  E. _cuplogin._ tcp.customer.com
  B. _collab-edge._tls.customer.com

• Question 358:

  Which Cisco Unified CM service is responsible for periodically checking disk usage and deleting old Call Management Records files?

  A. Cisco CallManager
  B. Cisco CDR Agent
  C. Cisco CDR Repository Manager
  D. Cisco SOAP ?CallRecord Service
  E. Cisco Extended Functions
  C. Cisco CDR Repository Manager

• Question 359:

  Refer to the exhibit.

  

  Which out-of-dialog SIP OPTIONS ping response put dial-peer tag 1111 into its current operational state?

  A. 501 Not Implemented
  B. 504 Server Time-out
  C. 408 Request Timeout
  D. 486 Busy Here
  E. 503 Service Unavailable
  E. 503 Service Unavailable

  ---

  SIP 503 Service Unavailable is commonly seen in a VoIP network when a SIP device (such as a SIP server) is knowingly unable to process a call. Typically when this happens the endpoint that originated the Invite will try the next available host it receives in the SIP Contact header.

• Question 360:

  Which call hunt mechanism is only supported by the voice hunt group in a Cisco Unified Communications Manager Express router?

  A. sequential
  B. peer
  C. longest idle
  D. parallel
  E. overlay
  D. parallel

  ---

  Parallel Hunt-Group, allows a user to dial a pilot number that rings 2-10 different extensions simultaneously. The first extension to answer gets connected to the caller while all other extensions will stop ringing. A timeout value can be set whereas if none of the extensions answer before the timer expires, all the extensions will stop ringing and one final destination number will ring indefinitely instead. The final number could be another voice hunt-group pilot number or mailbox. The following features are supported for Voice Hunt-Group: Calls can be forwarded to Voice Hunt-Group Calls can be transferred to Voice Hunt-Group Member of Voice Hunt-Group can be SCCP, ds0-group, pri-group, FXS or SIP phone/trunk Max member of Voice Hunt-Group will be 32