What is a QoS requirement for VoIP calls?
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A. B. C. D.D.
https://www.ciscopress.com/articles/article.asp?p=357102#:~:text=One%2Dway%20latency%20(mouth%20to,and%20Layer%202%20media%The correct answer for this question is D. 150 ms of one-way latency from mouth to ear.
Quality of Service (QoS) is a set of techniques used to manage network resources and ensure that the traffic of critical applications, such as VoIP, is given priority over less important traffic. QoS is essential for VoIP because it is a real-time application that requires low latency, low jitter, and minimal packet loss to ensure a high-quality conversation.
Latency refers to the time it takes for a packet to travel from the source to the destination. In the case of VoIP, latency can cause delays in the conversation, making it difficult for participants to have a natural conversation. The recommended one-way latency for VoIP calls is 150 ms or less, from the mouth of the speaker to the ear of the listener. This ensures that the conversation is as close to real-time as possible and that there are no noticeable delays.
Jitter refers to the variation in packet delay, which can cause the audio to be choppy and unintelligible. The recommended jitter for VoIP calls is 150 ms or less.
Packet loss occurs when packets are dropped or lost in transit, which can cause the audio to cut out or be distorted. The recommended packet loss rate for VoIP calls is less than 1%.
Bandwidth is the amount of data that can be transmitted over a network in a given time period. VoIP calls require a minimum amount of bandwidth to ensure high-quality audio. The recommended amount of guaranteed bandwidth per phone for VoIP calls is 80 kbps.
In summary, the most critical QoS requirement for VoIP calls is the one-way latency of 150 ms or less from the mouth to the ear. This ensures a high-quality, natural conversation with minimal delays.