350-801 Exam Dumps - Implementing Cisco Collaboration Core Technologies (350-801 CLCOR)

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Question # 49

Which two features will access layer switches provide support providing high-quality voice and taking advantage of the full voice feature set? (Choose two.)

Deploy RSVP to improve VoIP QoS only where it will have a positive impact on quality and functionality where there is limited bandwidth and frequent network congestion.

Use multiple egress queues to provide priority queuing of RTP voice packet streams and the ability to classify or reclassify traffic and establish a network trust boundary.

Map audio and video streams of video calls (AF41 andAF42) to a class-based queue with weighted random early detection.

Implement IP RTP header compression on the serial interface to reduce the bandwidth required per voice call on point-to-point links.

Use 802 1Q trucking and 802.1p for proper treatment of Layer 2 CoS packet marking on ports with phones connected.

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Question # 50

What are two reasons that AF41 is marked for the audio and video channels of a video call? (Choose two.)

to prioritize video over other high -priority traffic classes

to give video calls a higher priority than AP41 in the QoS policy

to allow high-definition quality calls over low-speed links

to preserve lip synchronization between the audio and video channels

to provide separate classes for audio calls and video calls

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Question # 51

Which characterstic of distributed class- based weighted fair queueing addresses jitter

prevention?

It provides additional granularity by allowing a user to create classes

It minimizes jitter by implementing a priority queue for voice traffic

It uses a priority queue for voice traffic to avoid jitter.

It provides additional granularity by allowing a user to define custom class

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Question # 52

An engineer must create a route pattern for a SIP trunk with the pattern named Unity_Con_SIP_Trunk and a voicemail pilot number 8000/< None > in Cisco Unity Connection. Which value must the engineer enter for the Gateway/Route List field?

Option A

Option B

Option C

Option D

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Question # 53

Refer to tho exhibit. A customer reports that SIP voice calls are using G.711 codec instead of G.729. The administrator checks the SIP signaling in the logs and sees the SDP shown. What will the administrator see in the logs if the right configuration change was made to resolve the issue?

m=audio 16444 RTP/AVP 4 101

m=audio 16444 RTP/AVP 15 101

m=audio 16444 RTP/AVP 9 101

m=audio 16444 RTP/AVP 18 101

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Question # 54

What is an indicator of network congestion in VoIP communications?

jitter increase due to variable delay

discards in the interface of routers and switches

video loss due to video frame corruption

gaps in the voice due to packet loss

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Question # 55

Which two configuration elements are part of the Cisco UCM toll-fraud prevention?(Choose two.)

feature control policy

partition

SIP trunk security profile

SUBSCRIBE Calling Search Space

Calling Search Space

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Question # 56

An end user at a remote site is trying to initiate an Ad Hoc conference call to an end user at the main site. The conference bridge is configured to support G.711. The remote user's phone only supports G.729. The remote end user receives an error message on the phone: "Cannot Complete Conference Call." What is the cause of the issue?

The remote phone does not have the conference feature assigned.

A software conference bridge is not assigned.

A Media Termination Point is missing.

The transcoder resource is missing.

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