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Question 1:
A user dials 9011841234567 to reach Vietnam. Which steps send the call to the PSTN provider as 011841234567?
A. Option A
B. Option B
C. Option C
D. Option D
Correct Answer: A
Question 2:
Which Cisco Unity Connection handler plays a greeting that announces the option to dial a user extension by default?
A. the Operator call handler
B. the Goodbye call handler
C. the Directory handler
D. the Interview handler
Correct Answer: B
Reference: https://www.cisco.com/en/US/docs/voice_ip_comm/unity/3x/administration/guide/313/SAG_0200.html
Question 3:
If a phone needs to register with cucm1.cisco.com, which network service assists with the phone registration process?
A. SMTP
B. ICMP
C. DNS
D. SNMP
Correct Answer: C
According to the Cisco Community website1, the phone uses DNS to resolve the hostname of the CUCM server (cucm1.cisco.com) to its IP address. DNS is a network service that translates domain names into IP addresses.
Question 4:
Which two features of Cisco Prime Collaboration Assurance require advanced licensing? (Choose two.)
A. real time alarm browse
B. multicluster support
C. call quality monitoring
D. email notifications
E. customizable events
Correct Answer: BC
https://www.cisco.com/c/en/us/td/docs/net_mgmt/prime/collaboration/11-6/assurance/advanced/guide/cpco_b_Cisco-Prime-Collaboration-Assurance-Guide-Advanced-11-6/cpco_b_Cisco-Prime-Collaboration-Assurance-Guide-Advanced-116_chapter_00.html#ID8
Question 5:
Which command in the MGCP gateway configuration defines the secondary Cisco Unified Communications Manager server?
A. mgcapp
B. ccm-manager fallback-mgcp
C. mgcp call-agent
D. ccm-manager redundant-host
Correct Answer: D
https://www.cisco.com/c/en/us/td/docs/ios/voice/cminterop/configuration/guide/12_4t/vc_12_4t_book/vc_ucm_mgcp_gw.html
Question 6:
Which application traffic does the DiffServ AF41 class according to the Cisco Collaboration System Solution Reference Network Design?
A. audio call
B. messaging
C. video call
D. signalling
Correct Answer: C
Question 7:
An engineer implements a new Cisco UCM based telephony system per these requirements:
1.
The local Ethernet bandwidth is sized based on the total bandwidth per call.
2.
A G.736 codec is used.
3.
The bit rate is 64 kbps.
4.
The codec sample interval is 10 ms.
5.
The voice payload size is 160 bytes per 20 ms.
What should the size of the Ethernet bandwidth be per call?
A. 31.2 kbps
B. 38.4 kbps
C. 55.2 kbps
D. 87.2 kbps
Correct Answer: D
To calculate the Ethernet bandwidth per call, we need to take into account the total number of bytes per second in each direction (transmit and receive) and add additional overhead for Ethernet, IP, and UDP headers.
The total number of bytes per second is calculated as follows:
160 bytes per 20 ms = (160 bytes/20 ms) x (50 packets/s) = 8000 bytes/s
The bit rate is 64 kbps, so we need to add an additional 8 kbps for overhead:
64 kbps + 8 kbps = 72 kbps
The total number of bytes per second including overhead is:
72,000 bps / 8 bits per byte = 9,000 bytes/s
Adding additional overhead for Ethernet, IP, and UDP headers, we can estimate that the total number of bytes per second will be approximately 12,000 bytes/s in each direction. Therefore, the Ethernet bandwidth per call should be:
12,000 bytes/s x 8 bits per byte = 96 kbps
Therefore, the correct answer is D. 87.2 kbps is not sufficient to support the required bandwidth per call.
Question 8:
Which DSCP marking is represented as 101110 in an IP header?
A. EF
B. CS3
C. AF41
D. AF31
Correct Answer: A
Question 9:
How many minutes does it take for automatic fallback to occur in a Presence Redundancy Group if the primary node lost a critical service?
A. 5 min
B. 10 min
C. 30 min
D. 60 min
Correct Answer: C
Enable Automatic Fallback “This parameter speci es whether to do automatic fallback. In the event of a failover, the IM and Presence Service moves users automatically from the backup node to the primary node thirty minutes after the
primary node returns to a healthy state.
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/im_presence/con gAdminGuide/11_5_1/cup0_b_con g-and-admin-guide- 1151su5/ cup0_b_imp-system-configuration-1151su5_chapter_0100.html
Question 10:
Which action is required if an engineer wants to have Cisco Unified Communications Manager control the configuration for an MGCP gateway?
A. Apply the ccm-manager configuration commands to the gateway.
B. Upload the custom configuration in the TFTP server in Cisco Unified CM.
C. From Cisco Unified CM > Device > Gateway > Add gateway, check the auto-configuration check box.
D. Configure the Cisco Unified CM\’s IP in voice service VoIP.
Correct Answer: A
Question 11:
In an eleven-class queueing model following Realtime. Best effort, and scavenger queuing rules. What recommended percentage of bandwidth must be allocated for interactive video?
A. 10%
B. 12%
C. 15%
D. 20%
Correct Answer: C
Reference: https://www.ciscopress.com/articles/article.asp?p=357102andseqNum=7#:~:text=If%20Scavenger%20and%20Bulk%20traffic,illustrated%20in%20Figure%2012%2D10
Question 12:
What is a QoS requirement for VoIP calls?
A. 150 ms of one-way latency from mouth to ear
B. 15% packet loss
C. 150 bps per phone of guaranteed bandwidth
D. 150 ms of jitter
Correct Answer: D
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/netstruc.html#20861
Question 13:
An engineer with troubleshoots poor voice quality on multiple calls. After looking at packet captures, the engineer notices high levels of jitter. Which two areas does the engineer check to prevent jitter? (Choose two.)
A. The network meets bandwidth requirements.
B. MTP is enabled on the SIP trunk to Cisco Unified Border Element.
C. Cisco UBE manages voice traffic, not data traffic.
D. All devices use wired connections instead of wireless connections.
E. Voice packets are classified and marked.
Correct Answer: AE
Reference: https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/20371-troubleshoot-qos-voice.html
Question 14:
Refer to the exhibit. Which codec should an engineer select for a call mode between “Dallas-REG” and “Austin-REG”?
A. MP4A-LATM
B. G.729
C. OPUS
D. G.711
Correct Answer: B
The codec preference list for the “Dallas-REG” region is “Factory Default low loss”. This list includes the following codecs in order of preference:
G.729
G.711 OPUS MP4A-LATM The codec preference list for the “Austin-REG” region is “Factory Default low loss”. This list includes the following codecs in order of preference:
G.729
G.711 OPUS MP4A-LATM Since both regions have the same codec preference list, the codec that will be used for a call made between “Dallas-REG” and “Austin-REG” is G.729. G.729 is a narrowband speech codec that was developed by the ITU-T in 1988. It is a low- bitrate codec that provides good quality speech at a bitrate of 8 kbps. G.729 is widely used in VoIP applications and is the default codec for many VoIP systems. G.711 is a wideband speech codec that was developed by the ITU-T in 1972. It is a high- bitrate codec that provides excellent quality speech at a bitrate of 64 kbps. G.711 is not as widely used as G.729 due to its high bitrate requirements. OPUS is a lossy audio codec that was developed by the IETF in 2012. It is a low-bitrate codec that provides good quality speech at a bitrate of 6 kbps. OPUS is widely used in VoIP applications and is the default codec for many VoIP systems. MP4A-LATM is a lossy audio codec that was developed by the IETF in 1999. It is a high- bitrate codec that provides excellent quality speech at a bitrate of 24 kbps. MP4A-LATM is not as widely used as G.729 or OPUS due to its high bitrate requirements.
Question 15:
Which external DNS SRV record must be present for Mobile and Remote Access?
A. _cisco-uds._tcp.example.com
B. _collab-edge._tls.example.com
C. _collab-edge._tcp.example.com
D. _cisco-uds._tls example.com
Correct Answer: B
Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/expressway/config_guide/X14-0-1/mra/exwy_b_mra-deployment-guide-x1401/exwy_m_requirements-for-mra.html