low latency

Bandwidth vs Latency in Real-Time Communication

9 minutes, 40 seconds Read

Real-time communication (RTC) has become fundamental to daily life in today’s digital environment. From online meetings to video conversations with loved ones, Bandwidth vs Latency the need for seamless, high-quality communication has never been stronger. 

The global real-time communication market is anticipated to develop at a compound annual growth rate of 41.7% and reach $45.91 billion by 2027. As the demand for real-time communication keeps growing, it’s critical to understand the key components that make it possible.

Bandwidth and latency are two of the most essential elements of real-time communication. Despite their similarity in sound, these names refer to two quite different aspects of communication. 

Anyone trying to maximize their real-time has to be aware of the differences between bandwidth and latency. 

In this article, we’ll explore the difference between these two terms and how they impact the quality of real-time communication.

What is Bandwidth?

The term “bandwidth” defines how much data sent via a network in a specific length of time. It calculates the amount of data that transferred or received through a connection. Bits per second (bps) or its multiples as kilobits per second (Kbps), megabits per second (Mbps), and gigabits per second (Gbps), are commonly used to measure bandwidth.

The quality of audio and video transmission in the context of RTC is greatly influenced bandwidth. The quality of the sent media improves with increasing bandwidth. High bandwidth is not assured, especially when several users are utilizing the same network. This may lead to a decrease in media quality or possibly a total loss of communication.

What is Latency?

Latency, on the other hand, refers to the delay or lag between the transmission and receipt of data over a network. Measured in milliseconds (ms) and represents the time it takes for data to travel from its source to its destination. Latency can be influenced by a variety of factors, including the distance between the sender and receiver, the number of devices on the network, and the quality of the network infrastructure.

In RTC, high latency can lead to delays in audio and video transmission, resulting in poor-quality communication. Latency issues are particularly problematic in real-time applications where communication is time-sensitive, such as online gaming or live video call.

Bandwidth vs. Latency in Causing RTC Issues

Although latency and bandwidth are two different concepts, they both have the potential to affect RTC. Inadequate bandwidth communication to be interrupted or of poor quality. Bandwidth-related problems appear when this happens. As a result of network saturation and congestion brought on by numerous users using a single network, this can occur.

Contrarily, latency-related problems appear data transmission is delayed, causing a pause in communication. The transmitter and receiver may be far apart, or many devices connected to the network, which can cause this to occur.

Bandwidth vs Latency Issues

Users can take several actions to enhance their network connectivity to address latency and bandwidth problems with RTC.

Here are a few solutions:

  1. Check network connection: Users should routinely test their network connections to make sure they are receiving the maximum amount of bandwidth. They ought to consider upgrading or changing their internet service provider (ISP) if necessary.
  2. Optimize settings: Users can adjust their audio and video settings to lessen the amount of data delivered. This may entail lowering the video’s resolution or frame rate or deactivating it entirely.
  3. Connect using a wired connection: Poor network performance might result from wireless connections that are vulnerable to signal loss and interference. Reducing latency and enhancing network reliability, wired connections can be used.
  4. Reduced distance between devices: Reduced distance between devices is one method for dealing with latency problems. Data can move more quickly, decreasing latency devices are placed physically close to one another.

Achieve Better Real-Time Communication Quality with the Third-party SDKs

Although the aforementioned fixes can assist with latency and bandwidth problems, they might not be sufficient to offer a smooth real-time communication experience. And only an API provider can help with that. 

Any in-app chat API provider offers a comprehensive SDK for real-time communication that includes features for audio and video settings, in-call quality statistics, and other tools to manage audio and video quality. 

  1. Overview

Maximizing bandwidth and lowering latency is the finest real-time communication experience possible. Accomplished via a number of features that are designed specifically to meet the requirements of real-time communication.

  1. Audio and Video Settings

The audio and video settings offered by a third-party SDK enable users to customize their listening and viewing experiences. This contains options for video resolution, frame rate, and audio quality.

  1. In-Call Quality Statistics

In-call quality statistics is another feature of an in-app chat SDK that offers in-call feedback on the transmission’s audio and video quality in real-time. Users can utilize this to find and fix any problems that might be creating latency or bandwidth problems.

  1. Additional Ways to Manage Bandwidth and Latency Issues

Below are a few others ways to optimize bandwidth consumption like,

  1. Assessing VLans
  2. Mapping network topology
  3. Using load balancing and caching
  4. Opting for networking monitoring tools or use APIs that offer
  5. Timely network updates

Together, these features and ways can help to provide a seamless real-time communication experience, even in situations where bandwidth and latency may be a challenge.


Bandwidth and latency are critical factors in real-time communication that can lead to issues such as poor call quality and dropped calls. To combat these problems, users can turn to a customizable video chat solution like the MirrorFly SDK, which offers a range of features to improve audio and video quality, including robust security, ultra low latencies , maintains an uptime SLA of 99.999%SLA, and more.

So, if you’re looking for an effective solution to your real-time communication needs, drop us a line in the comments section below, we will reach out to you. 

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