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To prepare for the Cisco 350-801 exam, candidates can take advantage of a variety of study resources, including Cisco’s official training courses and study materials, practice exams, and online communities. Additionally, candidates should have hands-on experience with Cisco collaboration technologies in a professional setting.

Cisco 350-801 certification exam is an excellent opportunity for IT professionals who want to develop their skills in Cisco collaboration technologies. Implementing and Operating Cisco Collaboration Core Technologies certification is designed to test the candidate's knowledge and skills in various areas such as Cisco Unified Communications Manager, Cisco Unity Connection, Cisco Unified Contact Center Express, and Cisco Collaboration Edge Architecture. Implementing and Operating Cisco Collaboration Core Technologies certification provides an opportunity for professionals to validate their skills, gain recognition within the industry, and enhance their career prospects.

Cisco 350-801 Exam, also known as Implementing and Operating Cisco Collaboration Core Technologies, is a certification exam that tests the knowledge and skills of candidates in the field of Cisco Collaboration technology. 350-801 exam is designed for IT professionals who want to demonstrate their expertise in implementing and operating Cisco Collaboration solutions in enterprise environments. 350-801 exam covers various topics related to Cisco Unified Communications Manager, Cisco Unity Connection, Cisco Instant Messaging and Presence, and Cisco TelePresence.

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Cisco Implementing and Operating Cisco Collaboration Core Technologies Sample Questions (Q171-Q176):

NEW QUESTION # 171
When a remote office location is set up with limited bandwidth resources, which codec would allow the most voice calls with the limited bandwidth?

A. G.729
B. G.723
C. G.711
D. G.722

Answer: A

NEW QUESTION # 172

Refer to the exhibit. An administrator configures fax dial-peers on a Cisco IOS gateway and finds that faxes are not working correctly Which change should be made to resolve this issue?

A. codec g726r32
B. codec g723ar63
C. codec g729br81
D. codec g711ulaw

Answer: D

NEW QUESTION # 173

Refer to the exhibit. A customer wants the SIP 200 OK shown to advertise codecs in the following order:

After correcting the codec preferences. What should the audio payload show in the SIP Traces?

A. Option D
B. Option B
C. Option C
D. Option A

Answer: A

NEW QUESTION # 174
Refer to the exhibit.
https://i.postimg.cc/C57TkczG/image.png

A call is falling to establish between two SIP Devices The called device answers with these SOP Which SOP parameter causes issue?

A. The media stream is set to send only
B. The payload for G.711ulaw must be 18.
C. The calling device did not offer a ptime value
D. The RTP port is set to 0.

Answer: D

Explanation:
a) The calling device did not offer a ptime value: The ptime value is used to specify the amount of time between each media packet. If the calling device does not offer a ptime value, the called device will use the default value of 20 milliseconds.
b) The media stream is set to sendonly: The media stream is set to sendonly when the called device is only able to send media packets, and not receive them. This is not a problem, and the call will still succeed.
c) The payload for G.711ulaw must be 18: The payload for G.711ulaw is the type of media packet that is used. The payload must be set to 18 for G.711ulaw, but this is not a problem, and the call will still succeed.
Explanation:
The RTP port is used to send and receive media packets during a call. If the RTP port is set to 0, the called device will not be able to send or receive media packets, and the call will fail.
The other options are not correct because:

NEW QUESTION # 175
When a phone is registered over Mobile and Remote Access, where does it register?

A. Expressway-C
B. Expressway-E
C. Cisco Unified Communications Manager
D. Cisco Unified presence server

Answer: C

NEW QUESTION # 176
......

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