Understanding and Preventing Packet Loss in WebRTC

10 min read
July 31, 2023

In the world of networking, data travels at lightning speed across vast distances, bringing people and devices closer than ever before. However, this does not happen without challenges, one of the major ones being packet loss.

Packet loss may seem harmless, but in reality, It can significantly disrupt your video calls, slow down your downloads, and even crash your online gaming session. No matter the size or scope of your business, network connectivity and performance are essential for any business operation.

Table of Contents 

    1. What is packet loss in WebRTC?
    2. Reasons behind packet loss in WebRTC
    3. What is the role of network protocols in packet loss?
    4. Preventing packet loss in WebRTC
    5. Assessing acceptable packet loss percentage
    6. How to improve WebRTC streaming servers and media servers
    7. How does packet loss impact networking?
    8. How does packet loss impact WebRTC fees?
    9. Enjoy flowless real-time interactions for your business needs with Digital Samba
  2. In this article, we are going to explore what is packet loss, its causes, the concept of an acceptable packet loss rate and what you can do to prevent packet loss.

    What is packet loss in WebRTC?

    Data is constantly transmitted between networks in units known as data packets, which contain information such as your routing details, and video and audio data.  Data packets are shared in almost every online interaction between devices, most of which you are aware of, such as watching a movie on a streaming platform, playing online video games and video conferences.  

    However, the packets can be interrupted when moving from one device to another leading to what we know as packet loss.

  3. Essentially, packet loss can be described as the loss of data as it’s being transmitted between networks.

  4. In the world of WebRTC, data is transmitted over IP networks and in many ways, network packet loss is simply a part of the design of the network itself.

Reasons behind packet loss in WebRTC

Packet loss can be frustrating when using WebRTC, as it can disrupt the smooth flow of real-time communication. Here are some common packet loss reasons:

What is the role of network protocols in packet loss?

In the world of WebRTC, transport protocols such as UDP and TCP play a crucial role in facilitating seamless communication. Understanding the functions and characteristics of these protocols is key to comprehending their impact on packet loss.

What is the impact of UDP and TCP on packet loss?

UDP, or User Datagram Protocol in full, is well-known for its speed and low latency. This makes it quite ideal for real-time communication applications like video conferencing. However, UDP falls short by lacking an error recovery and congestion mechanism which can result in packet loss.

TCP (Transmission Control Protocol) on the other hand, ensures reliable and ordered data delivery through its error recovery mechanisms and congestion control algorithms. Although TCP effectively minimises packet loss, it may end up introducing latency or delays, which are not optimal for real-time communication.

How to use congestion control algorithms to mitigate packet loss?

To address packet loss concerns, congestion control algorithms like TCP-friendly rate control come into play. These algorithms dynamically adjust the sending rate based on network conditions, optimising data flow and reducing congestion-related packet loss.

What are the benefits of SCTP in WebRTC?

Stream Control Transmission Protocol (SCTP) is often used by the VoIP community for server-to-server communication. However, when it’s used in WebRTC it’s for peer-to-peer communication with the aim of providing built-in error recovery, congestion control, and multi-streaming capabilities.

By implementing SCTP, you can enhance the robustness and fault tolerance in your WebRTC applications and contribute to a reduction in packet loss incidents.

Preventing packet loss in WebRTC

Packet loss can significantly impact the quality of your WebRTC communication. Fortunately, by learning how to get rid of packet loss, you can ensure a more seamless communication experience.

Let's explore some key strategies and techniques how to get rid of packet loss.

Optimising network conditions

To minimise packet loss, optimising your network conditions is paramount. To begin, let’s look at a few techniques on how to lower packet loss.

  1. Make sure you allocate sufficient bandwidth for your WebRTC application and always prioritise real-time traffic to reduce congestion on the network which consequently reduces packet loss.
  2. Employ a traffic shaping mechanism to ensure that your network prioritises WebRTC traffic over other non-important data. This will ensure that no critical data is lost, reducing packet loss.
  3. Also make sure you enhance your network by using efficient routing protocols, establishing direct connections or by utilising CDNs. As a result, your packets will use the most efficient route to reduce latency as well as avoid jitter.
  4. Outdated hardware will always cause packets at one point in time. So, make sure you’re working with new equipment or rather up to date whenever possible. Additionally, ensure your network infrastructure is well-equipped and configured.

Implementing error correction mechanisms

One of the most effective methods to handle packet loss is by implementing error correction mechanisms. Let’s consider the following approaches on how to get rid of packet loss:

  • Forward Error Correction (FEC)

FEC adds information into packets multiple times. This ensures that the receiver can reconstruct data even when some of the data is lost during transmission. Consequently, this improves reliability and reduces the need for retransmission.

  • Real-time retransmission techniques

You can also implement protocols such as the SCTP or other proprietary protocols that support real-time transmission. Although this technique effectively reduces latency, it does so at the expense of reliability. Therefore, it’s crucial to evaluate the trade-offs between latency and reliability when handling packet loss.

Network Monitoring and Diagnostic Tools

Monitoring and diagnosing packet loss issues are also essential for effective prevention. Utilise the following packet loss monitoring tools and techniques:

Assessing acceptable packet loss percentage

When it comes to data transmission across the internet, packet loss is an inherent challenge. This is not any different when it comes to WebRTC applications. Therefore, understanding what constitutes an acceptable packet loss percentage is important especially when you are dealing with real-time communication or data transfer.


So, what is acceptable packet loss?

The concept of acceptable packet loss simply refers to an acceptable threshold at which packet loss now becomes noticeable and detrimental to the quality of WebRTC applications.

This threshold varies depending on the application, the data being transmitted and its specific use case. For instance, applications that transmit financial data or medical data require a minimal to zero packet loss rate. They require the data to be received in a smooth and uninterrupted way.  

On the other hand, data transfer on video conferences like online classes may be able to accommodate slightly higher packet loss rates.

So how do you determine the acceptable threshold?

To determine an acceptable packet loss rate, you should first consider the specific requirements of your WebRTC application, the data you intend to transfer and your users’ expectations.

However, independent of these factors, it’s good practice to maintain a packet loss percentage below 1%. For instance, a packet loss rate of 0.5% or lower is generally good for gaming while less than 1% is typically acceptable for VoIP.

How to improve WebRTC streaming servers and media servers

To enhance the performance and reliability of your WebRTC streaming and media servers, you need to implement optimisation techniques and leverage functionalities to reduce/improve packet loss. Let’s have a look at some strategies.

How does packet loss impact networking?

Packet loss in networking can have significant consequences on user experience, particularly in real-time communication applications. Let's explore the various aspects and implications of high packet loss.

  1. Packet loss can cause degraded audio and video quality which results in choppy and distorted media playback. If the packet loss rate is too high, users can experience noticeable delays, interruptions, or cause frozen screens which negatively impacts user experience.
  2. Packet loss can also disrupt video and audio synchronisation which can cause a mismatch between the audio and video streams. This results in a disjointed viewing experience which is very confusing and frustrating to end users.
  3. Real-time collaboration applications such as online gaming, online trading or important video conferences rely on very low latency and minimal to no packet loss. Significant packet loss in such applications can lead to huge financial losses, misinterpretation of information or ineffective collaboration which impacts user experience negatively.

How does packet loss impact WebRTC fees?

One of the reasons why WebRTC is the most widely used developer technology for real-time communication worldwide is that it’s open source. It also comes with the added advantage that it’s embedded and available in all modern browsers. This means that anyone can leverage WebRTC without paying any upfront licensing fee or royalties.

Although WebRTC is free, the web applications built with WebRTC are not. It might be free, but the applications built on top of it still need to cover the cost of getting their conferencing application online and dealing with traffic. 

When it comes to packet loss in WebRTC, here are some key factors to consider when choosing a WebRTC application.

Enjoy flawless real-time interactions for your business needs with Digital Samba

Having a comprehensive understanding of packet loss in WebRTC is crucial for smooth and reliable communication. Equipped with the knowledge from this guide, you can now monitor, analyse and proactively prevent packet loss and improve WebRTC performance in your network.

And here's the great news - you don't have to do it all alone! Consider leveraging third-party video conferencing platforms like Digital Samba, which offer the mechanisms to reduce latency, prevent bandwidth issues, and ensure a smooth, high-quality experience.

With Digital Samba you’ll enjoy flawless real-time interactions for your business needs.


What is video packet loss?

Video packet loss occurs when data packets transmitted over a network fail to reach their destination.

Does packet loss cause a delay?

Packet loss can cause delays, affecting real-time data transmission like video calls.

What is packet loss in ISP?

Packet loss in ISP refers to data packets lost within an internet service provider's network.

How do I troubleshoot network latency and packet loss?

Troubleshoot latency and packet loss by checking connections, restarting routers, and using network diagnostic tools.

What causes packet loss?

Causes of packet loss include network congestion, faulty hardware, software bugs, and signal interference.

What does 100% packet loss mean?

100% packet loss means all data packets are lost, resulting in complete transmission failure.

What is good packet loss?

Good packet loss is minimal to none; ideally, it should be 0% for optimal network performance.


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