There are the following main factors impacting voice quality. Focusing on these improvements, it will help a lot on voice quality.
BANDWIDTH
It is the rate of data transfer, bit rate or throughput, measured in bits per second (bit/s).
It depends on the codec—how the data is compressed to be sent and received. When analog voice is digitized, if it is sampled 8,000 times per second. Each sample is encoded in 8-bits. So, we need to have a bandwidth of 64,000 bits per second (or 64 kbps) one way to send that voice data.
PACKET LOSS
Packet loss occurs when one or more packets of data traveling across a computer network fail to reach their destination. Packet loss is either caused by errors in data transmission, typically across wireless networks, or network congestion. Packet loss is measured as a percentage of packets lost with respect to packets sent.
You know when the voice drops in and out randomly on a phone call? That’s packet loss. Packets can be dropped randomly while files are being delivered from one phone to the other, resulting in those blank spots you might have experienced. The good news here: today’s IP network equipment is super reliable and even at a loss of around 3%, the quality that listeners hear is still better than what you’d hear on an average cellphone.
JITTER
A network with constant delay has no packet jitter. Packet jitter is expressed as an average of the deviation from the network mean delay.
Everyone’s been there—you’re on the phone and there’s a bit of a delay, so you keep talking over each other and finally have to take a pause or hang up. That’s jitter at work—another network enemy. Jitter occurs when packets are delivered across the network at delayed or inconsistent times. Some providers tackle this problem by introducing jitter buffers—essentially an agreed-upon delay. Jitter buffers can be a double-edged sword as a big buffer will introduce latency, and a very small one might not be effective.
LATENCY
Network delay is an important design and performance characteristic of a computer network or telecommunications network. The delay of a network specifies how long it takes for a bit of data to travel across the network from one node or endpoint to another. It is typically measured in multiples or fractions of seconds. Delay may differ slightly, depending on the location of the specific pair of communicating nodes.
Latency is introduced when voice data travels from point to point across the network—the more endpoints or other networks your call has to cross, the more latency you’re likely to experience. A network’s on-net infrastructure can help you minimize latency. Staying on-net means the call doesn’t have to bounce from carrier to carrier to be completed. The less hops a call has to take, the better the call will sound.
