300-075 Exam
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Q81. Scenario
There are two call control systems in this item. The Cisco UCM is controlling the Cisco Jabber for Windows Client, and the 7965 and 9971 Video IP Phone. The Cisco VCS is controlling the SX20, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
DP
Locations
CSS
Movi Failure
Movi Settings
CIPTV Topo
Subzone
Links
Pipe
A third collaboration call fails between the backbone site and the HQ site. After reviewing the exhibits, which of the following reasons could be causing this failure?
A. Not enough bandwidth has been allocated.
B. Device Pool.
C. Location.
D. The pipe is not functioning.
Answer: A
Explanation:
Based on the exhibit, each call is limited to no more than 128kbps per call, but the total available bandwidth is set to 256 kbps. This will allow the first to calls to go through, but there will be no more available bandwidth for the third call.
Q82. Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure?
A. SRST with MGCP fallback
B. SRST without MGCP fallback
C. Cisco Unified Communications Manager Express in SRST mode
D. SRST with VoIP dial peers to Cisco Unified Communications Manager Express
Answer: C
Q83. Which option indicates the best QoS parameters for interactive video?
A. 1% Max Loss, 150 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
B. 5% Max Loss, 5 s One-way Latency, 30 ms Jitter, 20% Overprovisioning
C. 0% Max Loss, 100 ms One-way Latency, 30 ms Jitter, 20% Overprovisioning
D. 1% Max Loss, 160 ms One-way Latency, 60 ms Jitter, 10% Overprovisioning
Answer: A
Q84. During device failover to the secondary Cisco Unified Communications Manager server, how does the phone recognize that the primary server is back?
A. The secondary server keeps sending keepalive message to the primary server and when it succeeds, it unregisters the phones to force them to register to the primary.
B. When the primary server goes online, it sends out an "ALIVE" message via broadcast so that the phones re-register.
C. The phones never re-register with the primary server until the active (secondary) one goes down.
D. The phone sends keepalive messages to the primary server frequently and when it succeeds, the phone re-registers with it.
Answer: D
Q85. A Cisco 3825 needs to be added in Cisco Unified Communications Manager as an H.323 gateway. What should the gateway type be?
A. H.323 gateway
B. Cisco 3825
C. Cisco 3800 series router. The specific model will be selected after the Cisco 3800 is selected.
D. The gateway can be added either as an H.323 gateway or Cisco 3800 series router.
E. The gateway can be added either as an H.323 gateway or Cisco 3825 series router.
Answer: A
Q86. Which two Cisco Extension Mobility attributes are available in the user device profile? (Choose two.)
A. regions
B. description
C. phone button template
D. NTP information
Answer: B,C
Q87. A voice-mail product that supports only the G.711 codec is installed in headquarters.
Which action allows branch Cisco IP phones to function with voice mail while using only the G.729 codec over the WAN link to headquarters?
A. Configure Cisco Unified Communications Manager regions.
B. Configure transcoding within Cisco Unified Communications Manager.
C. Configure transcoding resources in Cisco IOS and assign to the MRGL of Cisco IP phones.
D. Configure transcoder resources in the branch Cisco IP phones.
Answer: C
Q88. Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What value should be entered into the gatekeeper to support this bandwidth?
A. 768 kbps
B. 384 kbps
C. 512 kbps
D. 192 kbps
Answer: B
Explanation:
Incorrect Answer: A, C, D A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This sum does not include overhead. If the audio codec for a video call is G.729 (at 24 kb/s), the video rate increases to maintain a total bandwidth of 384 kb/s. Link:
http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08video.html#wp1059726
Q89. What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager?
A. EF/46
B. CS6/48
C. AF41/34
D. CS3/24
E. CS4/32
Answer: E
Q90. Refer to the exhibit.
Which CSS is used at the HQ Cisco Unified Communications Manager to reroute calls via the PSTN when the SAF network is unavailable?
A. the phone device CSS
B. the phone line CSS
C. the phone line/device combined CSS
D. the SAF CSS configured on the CCD requesting service
E. the phone AAR CSS configured at the phone device
F. No special CSS is required. If SAF patterns are accessible, the PSTN reroute is automatic.
Answer: E