What does VoIP packets using UDP experience during transmission?

Voice over Internet Protocol (VoIP) packets utilize the User Datagram Protocol (UDP) primarily because of its low-latency characteristics, which are essential for real-time communications. In this context, the correct option highlights that VoIP packets experience minimal delay, which is crucial for maintaining the quality of voice calls.

UDP is designed for speed and efficiency; it sends packets without establishing a connection, unlike its counterpart, TCP (Transmission Control Protocol), which guarantees reliable data delivery. This process allows UDP to have less overhead and lower latency, making it more suitable for applications like VoIP where timely delivery is more critical than complete reliability. While UDP does not ensure that all packets will arrive at their destination (i.e., they can be lost or arrive out of order), the trade-off allows for smoother, uninterrupted communication even if some packets are missing. This compromise is acceptable in many real-time scenarios where immediate voice delivery is prioritized over perfect accuracy.

In contrast, other options reference high reliability or guaranteed delivery, which are characteristics associated with TCP rather than UDP. VoIP does not offer significant overhead for packet management, as UDP is optimized for performance, not overhead, making the chosen option the most accurate in describing the behavior of VoIP packets during transmission.