The answer is D. I think what's throwing most of us off is the wording "Mouth to ear".
In the current SRND, it explains this clearly -
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/netstruc.html#20861
Voice (Bearer Traffic)
The following list summarizes the key QoS requirements and recommendations for voice (bearer traffic):
Voice traffic should be marked to DSCP EF per the QoS Baseline and RFC 3246.
Loss should be no more than 1 percent.
One-way latency (mouth to ear) should be no more than 150 ms.
Average one-way jitter should be targeted at less than 30 ms.
A range of 21 to 320 kbps of guaranteed priority bandwidth is required per call (depending on the sampling rate, the VoIP codec, and Layer 2 media overhead).
Okay then why didn't you select D as your answer? I got this directly from the current SRND.
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/netstruc.html#20861
CLCOR 350-801 Official Cert Guide, p. 313.
The end goal of a properly designed voice and video network solution should be to maintain a delay of less than 150ms, jitter less than 30 ms, and a packet loss of less than 1 percent.
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