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NEW QUESTION: 1
Which two different traffic types have the most similar sensitivity to latency, jitter, and packet loss? (Choose two.)
A. streaming video
B. peer-to-peer file sharing
C. SQL transactions
D. video conferencing
E. voice
F. voice signaling
Answer: D,E
Explanation:
Voice quality is directly affected by all three QoS quality factors such as loss, delay, and delay variation.
Loss causes voice clipping and skips. Industry standard codec algorithms can correct for up to 30 ms of lost voice. Cisco Voice over IP (VoIP) technology uses 20 ms samples of voice payload per VoIP packet. Only a single Real Time Transport (RTP) packet could be lost at any given time. If two successive voice packets are lost, the 30 ms correctable window is exceeded and voice quality begins to degrade. Delay can cause voice quality degradation if it is above 200 ms. If the end-to-end voice delay becomes too long, the conversation sounds as if two parties are talking over a satellite link or a CB radio. The ITU standard for VoIP, G.114, states that a 150 ms one-way delay budget is acceptable for high voice quality. With respect to delay variation, there are adaptive jitter buffers within IP Telephony devices. These buffers can usually compensate for 20 to 50 ms of jitter.
NEW QUESTION: 2
Answer:
Explanation:
Explanation
Box 1: Group 1 only
First rule applies
Box 2: Group1 and Group2 only
Both membership rules apply.
References: https://docs.microsoft.com/en-us/sccm/core/clients/manage/collections/create-collections
NEW QUESTION: 3
Which of the following items are definitive of Network Performance? (Choose three.)
A. Utilization
B. Response time
C. Packet size
D. Throughput
E. Efficiency
Answer: A,B,D
Explanation:
You can perform assessments on network performance prior to implementing latency-sensitive applications and use them afterward for monitoring. Examples of performance variables are network throughput, user response times, and line utilization.
Reference: Cisco p.483.
NEW QUESTION: 4
Refer to the exhibit.
The exhibit shows the output of debug isdn q931. An inbound PSTN call was received by a
SIP gateway that is reachable via a SIP trunk that is configured in Cisco Unified Communications Manager. The call failed to ring extension 3001. If the phone at extension 3001 is registered and reachable through the gateway inbound CSS, which three actions can resolve this issue? (Choose three.)
A. Configure a voice translation profile in the SIP Cisco IOS gateway with a voice translation rule that truncates the number from ten digits to four digits.
B. Configure a called-party transformation CSS on the gateway in Cisco Unified Communications Manager that includes a pattern that transforms the number from ten digits to four digits.
C. Change the significant digits for inbound calls to 4 on the SIP trunk configuration in Cisco Unified Communications Manager.
D. Configure the digit strip 4 on the SIP trunk under Incoming Called Party Settings in Cisco Unified Communications Manager.
E. Configure a translation pattern in Cisco Unified Communications Manager that can be accessed by the trunk CSS to truncate the called number to four digits.
F. Configure the Cisco IOS command num-exp 2288223001 3001 on the gateway ISDN interface.
Answer: A,C,E
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