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Cisco 300-815 Practice Test - Questions Answers, Page 2

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Question 11

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The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real- Time Transport Protocol traffic that had the one-way audio call.

You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).

H.245 Terminal Capability Set

H.245 Terminal Capability Set

H.245 Open Logical Channel

H.245 Open Logical Channel

H.225 Connect

H.225 Connect

H.245 Open Logical Channel Ack

H.245 Open Logical Channel Ack

Suggested answer: B
asked 10/10/2024
Yannik Huith blu Systems GmbH
36 questions

Question 12

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Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios

(H.323 to SIP interworking)? (Choose two.)

DTMF

DTMF

BFCP

BFCP

VIDEO

VIDEO

FAX

FAX

AUDIO

AUDIO

Suggested answer: A, B
asked 10/10/2024
Damir M
42 questions

Question 13

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When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?

ALERTING

ALERTING

PROCEEDING

PROCEEDING

CONNECT

CONNECT

RINGING

RINGING

Suggested answer: A
asked 10/10/2024
Lukas Reker
36 questions

Question 14

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End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

Contact: header of the 200 OK response

Contact: header of the 200 OK response

Allow: header if the 200 OK response

Allow: header if the 200 OK response

o= line of SDP content

o= line of SDP content

c= line of SDP content

c= line of SDP content

Suggested answer: D
asked 10/10/2024
Nestor Quintero
43 questions

Question 15

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Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?

The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.

The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.

Cisco Unified Communications Manager invoked media termination point resources.

Cisco Unified Communications Manager invoked media termination point resources.

The RTP traffic is arriving beyond the jitter buffer on the receiving end.

The RTP traffic is arriving beyond the jitter buffer on the receiving end.

A firewall in the media path is blocking TCP ports 16384-32768.

A firewall in the media path is blocking TCP ports 16384-32768.

Suggested answer: C
asked 10/10/2024
Mareah Allawi
39 questions

Question 16

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An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?

debug H.323 messages

debug H.323 messages

debug H.225 asn1

debug H.225 asn1

debug H.246 asn 1

debug H.246 asn 1

debug H.225 media

debug H.225 media

debug H.323 asn 1

debug H.323 asn 1

Suggested answer: B
asked 10/10/2024
femke vroome
53 questions

Question 17

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What is first preference condition matched in a SIP-enabled incoming dial peer?

incoming uri

incoming uri

target carrier-id

target carrier-id

answer-address

answer-address

incoming called-number

incoming called-number

Suggested answer: A
asked 10/10/2024
Marinus Johannes Klomp
45 questions

Question 18

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Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally. What are two possible solutions? (Choose two.)

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767

Ask the firewall administrator to change the ports to TCP.

Ask the firewall administrator to change the ports to TCP.

Ask the firewall administrator to change the range of UDP ports to 16384-32767.

Ask the firewall administrator to change the range of UDP ports to 16384-32767.

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.

Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

Suggested answer: A, C
asked 10/10/2024
Duc Hai
46 questions

Question 19

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Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

Analysis Manager > Inventory > Trace File Repositories

Analysis Manager > Inventory > Trace File Repositories

System > Tools > Trace and Log Central

System > Tools > Trace and Log Central

Voice/Video > Session Trace Log View > Real Time Data

Voice/Video > Session Trace Log View > Real Time Data

Voice/Video > Session Trace Log View > Open From Local Disk

Voice/Video > Session Trace Log View > Open From Local Disk

Suggested answer: C
asked 10/10/2024
Miles Greenyer
43 questions

Question 20

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Which description of RTP timestamps or sequence numbers is true?

The sequence number is used to detect losses.

The sequence number is used to detect losses.

Timestamps increase by the time ''carrying'' by a packet.

Timestamps increase by the time ''carrying'' by a packet.

Sequence numbers increase by four for each RTP packet transmitted.

Sequence numbers increase by four for each RTP packet transmitted.

The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).

The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).

Suggested answer: D
asked 10/10/2024
Amin Dashti
55 questions
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