WebRTC Embedded

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

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

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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:

Interference and noise in wireless connections

Generally, wireless networks experience more issues with packet loss than wired networks. Challenges such as radio frequency interference, weak signals, distance from routers and physical barriers like budlings can all cause wireless networks to drop packets.

Software bugs and security threats to your device

Your device’s OS is the first stop data packets make in their journey across the internet. If your device is not up to date, operating optimally or is infected with malware, then the internet connection may be affected resulting in internet packet loss.

Network congestion and bandwidth limitations

Network congestion occurs when there’s too much traffic in one network overwhelming the bandwidth. Just like highways get congested with cars (data packets) on all lanes (available bandwidth) during peak hours, networks can experience delays when they reach their maximum capacity. As a result, the packet may be ignored or discarded during transmission.

Latency and delay issues

As we mentioned above, in a typical network packet loss scenario, the cause is probably due to network congestion. Modern networks are intelligent and if too much congestion occurs, the network intentionally drops some packets to relieve the congestion.

However, not all packets are dropped and the successful packets will still take some time to be processed. If it takes too long for the successful packets to reach their destination, the successful packets could also be lost altogether.

Inadequate network infrastructure or faulty hardware

The rate at which hardware becomes outdated these days is another challenge for your network. Hardware such as routers, switches, cables, and firewalls consume a lot of computing power and can considerately weaken network signals.

Additionally, some hardware like firewalls can result in false positives, where the firewall blocks out data packets from reliable sources. System infrastructure is another challenge, if it’s not adequately equipped to handle and prevent packet loss, then most of your data will be lost.

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.

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.
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.
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.
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:

Packet loss measurement and monitoring tools

Use network monitoring tools to measure and monitor packet loss rates in real-time. These tools provide insights into the severity and patterns of packet loss incidents.

Analyzing network performance metrics and identifying bottlenecks

Monitor metrics such as latency, jitter, and throughput to identify potential bottlenecks or network issues contributing to packet loss.

Troubleshooting techniques for diagnosing and resolving packet loss issues

Employ network diagnostic tools to identify and resolve packet loss issues, such as analyzing network configurations, inspecting firewall settings, or investigating potential hardware faults.

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.

Choose a server with a low-latency network connection

The first step in reducing latency on RTC media servers is to ensure that the server is connected to the internet through a low-latency network connection. This will ensure that messages are transmitted quickly between the server and the participants.

Optimize the server configuration

You can also optimize the server configuration by using a high-performance server with a powerful processor and sufficient memory. Additionally, you can configure the server to use efficient codecs and other settings that minimize latency. One way to do this is by setting up your server to use UDP instead of TCO, this will help reduce packet loss caused by latency issues.

Use a Content Delivery Network (CDN)

By using a CDN, you can reduce the distance between the participants and the server, which can significantly reduce latency.

Use Adaptive Bitrate (ABR) streaming

As mentioned in our article, ABR streaming allows the server to adjust the quality of the video or audio stream based on the available network bandwidth. By using ABR streaming, you can ensure that the communication experience remains smooth and uninterrupted even if network conditions change.

Load Balancing and Redundancy for Resilience

You can also distribute your payload among different servers to ensure that the server are resources are utilized effectively which also reduces the risk of overloading which might lead to packet loss. Redundancy on the other hand ensures that you always have a backup in case data is dropped or lost.

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.

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.
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.
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.

Bandwidth

Packet loss can lead to increased bandwidth usage, as the application will need to resend the lost packets. This can lead to increased costs, especially if you are using a metered bandwidth plan.

WebRTC Media server

If you are using a third-party media server, packet loss can lead to increased costs, as the server will need to store and process the re-sent packets.

License

Some WebRTC implementations require a license fee. If your application is experiencing high packet loss, you may need to purchase a more expensive license that includes features to mitigate packet loss.

The mechanisms in place to handle packet loss

As discussed earlier, you need to ensure the WebRTC application is optimized to handle packet loss through mechanisms such as error-handling techniques which will incur additional costs to implement.

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.

FAQs

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

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

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

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

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

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

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

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