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Free 300-815 Exam Braindumps (page: 4)

Page 4 of 45
PDF with 212 Questions

What is first preference condition matched in a SIP-enabled incoming dial peer?

  1. incoming uri
  2. target carrier-id
  3. answer-address
  4. incoming called-number

Answer(s): A


Reference:

https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8



Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally. What are two possible solutions? (Choose two.)

  1. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
  2. Ask the firewall administrator to change the ports to TCP.
  3. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
  4. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
  5. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

Answer(s): C,D


Reference:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/9_1_1/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide- 91_chapter_01.html



Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

  1. Analysis Manager > Inventory > Trace File Repositories
  2. System > Tools > Trace and Log Central
  3. Voice/Video > Session Trace Log View > Real Time Data
  4. Voice/Video > Session Trace Log View > Open From Local Disk

Answer(s): C


Reference:

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified- communications-manager-callmanager/213583-procedure-to-analyse-call-flow-of-sip-ca.html



Which description of RTP timestamps or sequence numbers is true?

  1. The sequence number is used to detect losses.
  2. Timestamps increase by the time “carrying” by a packet.
  3. Sequence numbers increase by four for each RTP packet transmitted.
  4. The timestamp is used to place the incoming audio and video packets in the correct timing order
    (playoutdelay compensation).

Answer(s): D


Reference:

https://www.cs.columbia.edu/~hgs/rtp/faq.html






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