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Cisco 350-801 Exam Questions Learning Material in Three Different Formats

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Cisco 350-801 Exam and Its Target

The Cisco 350-801 exam is designed mainly for the network engineers, systems engineers, and network administrators who want to improve their professional expertise to move their career forwards.

The IT specialists who are going to pass the Cisco 350-801 exam should first make sure that they possess the requisite knowledge and skills. They must be able to implement and handle basic collaboration technologies that include collaboration applications, QoS, Call Control, media resources, Cisco IOS gateway, and codecs and endpoints. Their skills in implementing and deploying key collaboration technologies will be an added advantage.

Additionally, the students should demonstrate a specializing knowledge of the basic computer networking terms, including routing, switching, LANs, and WANs. Moreover, they must have a thorough understanding of the fundamentals of digital interfaces, Voice over IP (VoIP), and Public Switched Telephone Networks (PSTNs). The applicants should also have the general familiarity with Cisco Unified Communications Manager deployment, data networks, and voice.

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Cisco Implementing and Operating Cisco Collaboration Core Technologies Sample Questions (Q224-Q229):

NEW QUESTION # 224
A Cisco Unity Connection Administrator must set a voice mailbox so that it can be accessed from a secondary device. Which configuration on the voice mailbox makes this change?

A. Attempt Forward routing rule
B. Mobile User
C. Alternate Extensions
D. Alternate Names

Answer: A

Explanation:
Section: Collaboration Applications

NEW QUESTION # 225
Which DHCP option must be set up for new phones to obtain the TFTP server IP address?

A. option 6
B. option 120
C. option 66
D. option 15

Answer: C

Explanation:
Section: Infrastructure and Design
Explanation/Reference: https://blog.router-switch.com/2013/03/dhcp-option-150-dhcp-option-66/

NEW QUESTION # 226
An engineer implements a new Cisco UCM based telephony system per these requirements:
- The local Ethernet bandwidth is sized based on the total bandwidth
per call
- A G.736 codec is used.
- The bit rate is 64 kbps
- The codec sample interval is 10 ms.
- The voice payload size is 160 bytes per 20 ms.
What should the size of the Ethernet bandwidth be per call?

A. 87.2 kbps
B. 31.2 kbps
C. 55.2 kbps
D. 38.4 kbps

Answer: A

Explanation:
All details match with G711 protocol, for which the calculation is :
(160+(40+18))/160 x 64 =87.2 kbps
https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html Codec Information Bandwidth Calculations Codec & Bit Rate (Kbps) Codec Sample Size (Bytes) Codec Sample Interval (ms) Mean Opinion Score (MOS) Voice Payload Size (Bytes) Voice Payload Size (ms) Packets Per Second (PPS) Bandwidth MP or FRF.12 (Kbps) Bandwidth w/cRTP MP or FRF.12 (Kbps) Bandwidth Ethernet (Kbps)
G.711 (64 Kbps) 80 Bytes 10 ms 4.1 160 Bytes 20 ms 50 82.8 Kbps 67.6 Kbps 87.2 Kbp

NEW QUESTION # 227
Which option must be used when configuring the Local Gateway for a Cisco Webex Calling trunk?

A. Auth-based
B. mutual TLS
C. local authentication
D. certificate-based

Answer: D

Explanation:
A certificate-based trunk is a type of trunk that uses certificates to authenticate the connection between Webex Calling and the Local Gateway1. A Local Gateway is a supported session border controller that terminates the trunk on the premises2. A certificate-based trunk requires a certificate authority (CA) to issue and manage the certificates for both Webex Calling and the Local Gateway1.

NEW QUESTION # 228
Refer to the exhibit.

Which codec should an engineer select for a call made between "Dallas-REG' & "Austin-REG'?

A. G.729
B. MP4A-LATM
C. G.711
D. OPUS

Answer: A

Explanation:
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.

NEW QUESTION # 229
......

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