For VoIP applications, one-way mouth-to-ear delay should ideally be less than 150 ms for natural conversations [1].
Other factors, such as jitter, transmission delay, queuing delay, etc., also impact quality. However, if the delay occurs mid-transmission (e.g., due to network congestion or routing inefficiencies), there’s little that can be done beyond optimizing at the endpoints.
Other factors, such as jitter, transmission delay, queuing delay, etc., also impact quality. However, if the delay occurs mid-transmission (e.g., due to network congestion or routing inefficiencies), there’s little that can be done beyond optimizing at the endpoints.
[1] https://www.wikiwand.com/en/articles/Latency_(audio)#Telepho...