Topics Covered: Voice Quality, Bandwidth, Packet Loss, Jitter, and Latency
Factors That Impact Call Quality
When making a Voice over IP (VoIP) phone call, the sound of your voice is broken into thousands of packets. These packets travel various paths on the internet to Simplicity VoIP, and on to their final destination, where they are reassembled. Many factors can affect packets on this journey, and thus impact the quality of the call. The four most common: bandwidth, packet loss, jitter, and latency.
Bandwidth
Bandwidth is defined as the bit rate measure of available or consumed data communication resources expressed in bits/second or multiples of it (kilobits/s, megabits/s etc.).
The bandwidth needed for VoIP transmission will depend on a few factors: the compression technology, packet overhead or network protocol. For example, if we are using G.711 codec for voice encoding we need 64 Kbps for this process alone. Then, we need to add packet overheads. In total, it is estimated that we need 80 Kbps to transmit the one call.
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.
The highest rate of packet loss for the voice to be heard with enough quality must be under 1%. But it depends on the codec used. When the codec compression is higher, this effect will be more dangerous. A 1% packet loss degrades the voice more if the communication is using G.729 codec instead of G.711 codec.
Jitter
Jitter in IP networks is the variation in the latency on a packet flow between two systems, occurring when some packets take longer to travel from one system to the other. Jitter results from network congestion, timing drift and route changes
Real time communications (for example VoIP) usually have quality problems due to this effect. In general, it is a problem in slow-speed links or with congestion.
Jitter between the starting and final point of the communication should be less than 80 ms.
Latency
The latency is also known as delay. It's the amount of delay (or time) it takes to send information from one point to the next. Latency is usually measured in milliseconds or ms. It's also referred to (during speed tests) as a ping rate.
Real time communications (for instance VoIP) and full-duplex communications are aware of this problem. Like jitter, it is a common problem in slow-speed or congested connections.
Latency between the starting and final point of the communication should be less than 100 ms.