When a call is placed, the real-time audio

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Have you ever experienced a phone call that sounds as if the person on the other end has a tinny, almost robotic, voice? Have you experienced calls that cut out or sound choppy? Or maybe you have experienced two callers continuously talking over one another because the timing of the sound was off. All of this is often the result of a poorly prioritized or overloaded network.

is transported in network israel telegram packets. If you have a high level of bandwidth, you can usually expect to have high definition phone calls without all of those aforementioned problems. However, even when bandwidth is high, other issues may still persist in terms of VoIP call quality if the network is improperly prioritized. These issues are even more likely to occur when bandwidth is low.

Network Latency
Network latency is the time it takes to transfer audio packets from point A to point B. To maintain decent sound quality, network latency shouldn’t exceed 150ms (milliseconds). However, for best results, it’s recommended to strive for packet transfer within the range of 75 to 100ms in a VoIP call.

Latency above this general threshold can cause the timing of the sound in a call to be perceptibly delayed. This introduces the issue of two callers talking over one another.

Network Congestion & Prioritization
In most businesses, it is common for bandwidth to be shared among all devices connected to the wireless network, including VoIP calls; this is far more cost-effective and convenient. However, this can introduce problems if the network isn’t properly prioritized. For example, an email may be allocated more bandwidth than an audio packet on a VoIP call. Proper bandwidth allocation should prioritize VoIP calls since they take place in realtime, and increases in latency can cause all of those familiar issues, also known as “jitter.”

Congestion and a lack of prioritization can cause network latency, and can also introduce a number of other issues as well. Many packets traveling at once can use up all available bandwidth, causing the network to create a queue. In other words, an audio packet may be waiting in line behind a software download. This is not good for call quality and leads to jitter.