Understanding Call Quality: Key Terminology for Telephony

Call quality is a critical aspect of telephony that directly affects the overall communication experience. Clear and reliable voice communication is essential, whether it’s a business call, customer service interaction, or personal conversation. To better understand call quality in the context of telephony, let’s explore some key terminology used in the field.

What is Call Quality

Call quality refers to the overall performance and clarity of a phone call, typically assessed by the participants’ perceived audio and communication experience. It encompasses various factors that can affect the quality of the call, including voice clarity, call stability, absence of disruptions or distortions, and overall user satisfaction.

Good call quality means that the participants’ voices are clear and easy to understand, with no noticeable delays, distortions, or disruptions during the call. It also implies that the call is stable and reliable, with consistent audio quality throughout the conversation.

What is Latency

Latency is the amount of time it takes for a data packet to travel from its source to its destination in a network. Latency is also referred to as delay and is measured in milliseconds (ms). Anything under 100ms will be usable. Under 50ms is preferable

Latency is an important metric in networking as it affects the performance of real-time applications such as voice and video conferencing, online gaming, and financial transactions. A high latency results in slow response times, making these applications difficult to use or even causing them to fail.

Latency can be influenced by several factors such as network congestion, limited bandwidth, distance, and the number of network devices (hops) between the source and destination. Network administrators can monitor and manage latency to ensure optimal performance by adding bandwidth, reducing network congestion, and implementing quality of service (QoS) measures to prioritize critical traffic.

What is Jitter

Jitter refers to the variation in latency (delay) in a network communication. Jitter occurs when data packets are transmitted over a network and the time it takes for each packet to reach the destination varies. The term jitter refers to the deviation from the average latency in a series of packets.

Jitter can be caused by several factors, including network congestion, limited bandwidth, distance, and interference from other devices. Jitter can negatively impact the quality of real-time applications such as voice and video conferencing, where even small variations in latency can cause noticeable disruptions in audio and video quality. Jitter should not exceed 30ms.

In a networked environment, jitter can be monitored and managed using network performance monitoring tools. Solutions to jitter include adding bandwidth, reducing network congestion, and implementing quality of service (QoS) measures to prioritize critical traffic. In some cases, jitter can also be addressed through the use of error correction protocols or by reducing the number of hops (network devices) between the source and destination of the data.

What is a Packet

A logically grouped unit of data. Packets contain a payload (the information to be transmitted), originator, destination and synchronizing information. The idea with packets is to transmit them over a network so each individual packet can be sent along the most optimal route to its. Packets are assembled on one end of the communication and re-assembled on the receiving end based on the header addressing information at the front of each packet. Routers in the network will store and forward packets based on network delays, errors and re-transmittal requests from the receiving end.

What is Packet loss

Packet loss is the percentage of data packets that fail to reach their intended destination. Packet loss can occur for a variety of reasons, including network congestion, limited bandwidth, technical errors or interference from other devices. In a data transmission, packets of information are sent from one location to another. Packet loss occurs when one or more of these packets fail to reach the intended destination. This can result in a degradation of the quality of the transmission, as well as delays and errors.

In a networked environment, packet loss can be a serious problem, as it can lead to slow response times, disconnections and other issues that affect the overall performance of the network. To address packet loss, network administrators may need to implement a variety of solutions, such as optimizing network configurations, adding bandwidth, or upgrading hardware and software. In some cases, packet loss can also be prevented by using quality of service (QoS) measures, such as prioritizing certain types of traffic or using error correction protocols.

What is Packet Switching

Packet switching is a method of transmitting data over a network in which data is divided into small packets and transmitted to the destination in a more efficient manner.

In packet switching, data is divided into small packets that each contain a portion of the original data and the destination address. These packets are transmitted individually and can take different paths to reach the destination. Once the packets arrive at the destination, they are reassembled into the original data.

The main advantage of packet switching is that it allows for more efficient use of network resources. Packets can be transmitted over different paths in the network, reducing the impact of network congestion and increasing the overall speed of data transmission.

Packet switching is used in most modern data networks, including the Internet, local area networks (LANs), and wide area networks (WANs). It is also used in telecommunications networks for voice and video communication.

What is MOS (Mean Opinion Score)

MOS, or Mean Opinion Score, is a widely used subjective measurement that assesses the perceived quality of a voice call or audio transmission. It is typically used to evaluate the quality of telecommunications systems, such as telephony or voice over IP (VoIP), and is based on human perception of the call quality.

MOS is typically rated on a scale from 1 to 5, with 1 being the lowest and 5 being the highest score. The MOS score is determined by human listeners who subjectively rate the quality of a call based on their perception of various factors, such as voice clarity, background noise, delay, echo, and overall call experience.

What is Echo

Echo is the reflection of voice during a phone call, which can cause an echo effect that disrupts conversation and affects call quality. Echo can be caused by various factors such as poor acoustic design, faulty equipment, or network issues.

What is Background Noise

Background noise refers to any unwanted noise in the background during a phone call, such as static, hum, or interference. Background noise can degrade call quality and make it difficult to understand the conversation, especially in noisy environments or during conference calls.

What is Compression

Compression is the process of reducing the size of voice data packets for efficient transmission over a network. Compression can affect call quality, as excessive compression can result in loss of voice quality. Balancing compression settings is crucial to ensure optimal call quality.

What is Bandwidth

Bandwidth refers to the amount of data that can be transmitted over a network within a given time period. Sufficient bandwidth is required for high-quality voice transmission in telephony. Inadequate bandwidth can result in call quality issues such as latency, jitter, or packet loss.

What is HD Voice

HD Voice (High Definition Voice) is a technology that provides superior call quality with wider frequency range, higher voice clarity, and reduced background noise compared to standard voice calls. HD Voice is typically used in modern telephony systems to enhance call quality and provide a more immersive and natural communication experience.

In conclusion, understanding the key terminology related to call quality in telephony is essential for ensuring clear and reliable voice communication. Factors such as latency, jitter, packet loss, MOS, echo, background noise, compression, bandwidth, and HD Voice all play a crucial role in determining call quality. Telephony providers and users alike should be familiar with these terms to effectively diagnose, troubleshoot, and improve call quality, leading to better communication experiences for all parties involved.