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Cisco 350-801 Certification Exam is designed to test the knowledge and skills of IT professionals in implementing and operating Cisco Collaboration Core Technologies. 350-801 exam is part of the requirements for obtaining the CCNP Collaboration certification, which is aimed at professionals who are involved in designing, deploying, and maintaining collaboration solutions in a variety of business environments.

Cisco 350-801 certification exam is suitable for network administrators, system administrators, and IT professionals who want to develop their skills in Cisco collaboration technologies. Implementing and Operating Cisco Collaboration Core Technologies certification helps candidates to validate their skills and knowledge in collaboration technologies and enhance their career prospects. Implementing and Operating Cisco Collaboration Core Technologies certification also provides an opportunity for professionals to gain recognition within the industry and demonstrate their expertise in Cisco collaboration technologies.

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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.

Cisco Implementing and Operating Cisco Collaboration Core Technologies Sample Questions (Q165-Q170):

NEW QUESTION # 165
What is the most important consideration when designing a QoS voice framework?

Answer: D


NEW QUESTION # 166
Which attribute contains an XMPP stanza?

Answer: B


NEW QUESTION # 167
Refer to the exhibit.

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

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 # 168
What happens when a Cisco IP phone loses connectivity to the duster during an active call?

Answer: D

Explanation:

https://community.cisco.com/t5/unified-communications/call-in-progress-when-one-cucm-go- down/td-p/2453662


NEW QUESTION # 169
Refer to the exhibit.


A call is failing to establish between two SIP Devices. The Called device answer with this SDP. Which SDP parameter causes this issue?

Answer: D


NEW QUESTION # 170
......

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