Cisco 400-051 Online Practice Questions and Exam Preparation

Cisco 400-051 Online Questions & Answers

• Question 31:

  Which route pattern is matched in Cisco Unified Communications Manager version 11.0 when a user dials 2001?

  A. 200X configured with urgent priority
  B. 20[02-4]1 configured with urgent priority
  C. 200[1-4] configured with urgent priority
  D. 200[^2-4] configured with urgent priority
  E. 200! configured with urgent priority
  C. 200[1-4] configured with urgent priority

• Question 32:

  A Cisco collaboration architect is evaluating a list of codecs to use in a voice infrastructure. Which three facts are associated with iSAC and should be considered in the decision? (Choose three)

  A. The codec has better quality with less bandwidth for sideband applications.
  B. The codec will not be supported in TDM voice gateways.
  C. The codec will adjust its bandwidth consumption to the network conditions.
  D. The codec will not be available for H.323 and MGCP devices.
  E. The codec will not support low complexity.
  F. The codec will not be supported by SCCP configured on DSPFARMS.
  A. The codec has better quality with less bandwidth for sideband applications.
  C. The codec will adjust its bandwidth consumption to the network conditions.
  E. The codec will not support low complexity.

• Question 33:

  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?

  A. 200 OK
  B. NOTIFY with a message body of 200 OK
  C. 202 Accepted
  D. 100 Trying
  E. 200 BYE
  B. NOTIFY with a message body of 200 OK

  ---

  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 34:

  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?

  A. 20
  B. 40
  C. 80
  D. 160
  E. 250
  B. 40

  ---

  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 35:

  Which DNS SRV must the Expressway-E resolve when routing an outbound call to Microsoft clients on Lync or Skype for Business?

  A. _sips._tls.domain.com
  B. _sipfederationtls._tls.domain.com
  C. _sip._tcp.domain.com
  D. _sipfederationtls._tcp.domain.com
  E. _sips._tcp.domain.com
  C. _sip._tcp.domain.com

• Question 36:

  How are DTMF digits transported in RFC 2833?

  A. In the RTP stream with the named telephone events payload format.
  B. In the RTP stream with the regular audio payload format.
  C. In SIP NOTIFY messages.
  D. In SIP INFO messages.
  E. In SIP SUBSCRIBE messages.
  A. In the RTP stream with the named telephone events payload format.

  ---

  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 37:

  In a single split Cisco Meeting Server deployment, which software components are required to allow API access? (Choose three.)

  A. Web Admin
  B. Database
  C. XMPP Server
  D. Web Bridge
  E. TURN Server
  F. Load Balancer
  G. Call Bridge
  A. Web Admin
  B. Database
  G. Call Bridge

• Question 38:

  A collaboration engineer is configuring toll-fraud prevention in the dial plan. Which two sets of patterns allow Cisco Unity Connection to transfer calls to local and long distance numbers while blocking all other patterns? (Choose two.)

  EF

• Question 39:

  To which QoS tool category does compressed RTP belong?

  A. classification
  B. marking
  C. link efficiency
  D. queuing
  E. prioritization
  C. link efficiency

  ---

  LLQ is a feature that provides a strict PQ to CBWFQ. LLQ enables a single strict PQ within CBWFQ at the class level. With LLQ, delay-sensitive data (in the PQ) is dequeued and sent first. In a VoIP with LLQ implementation, voice traffic is placed in the strict PQ.

• Question 40:

  Refer to the exhibit. How many SIP transactions are involves in this call flow?

  

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