In the field of networking, the delay that occurs while processing network data is known as ‘network latency.’ It’s the interval between requesting and getting the data. This network latency, sometimes referred to as ‘lag,’ indicates how long a data transfer takes to go from one communication endpoint to another through a data communication network.
Network latency may take on a wide variety of shapes and scenarios within an IT environment and dedicated server setup. For example, network latency can occur when the IT infrastructure in one data center is connected to an IT environment in another data center via a data center interconnect (DCI). It could be a hospital, which has stored sensitive patient data in its on-premises server room, while data from the server room can be accessed from IT infrastructure deployed on dedicated servers in a third-party data center. In the context of the cloud, network latency may for example occur when a developer at a customer organization is sending data to or retrieving data from the platform of a cloud service provider (CSP) through the network available. When talking about multi-cloud, network latency might occur when a business application in one cloud instance is communicating to another business application in a different cloud instance. No matter what shape it takes, network latency may have a big impact on a business and an effective utilization of its IT applications.
Cloud, Finance, Gaming, Video Conferencing
Low network latency is certainly important for latency-sensitive sectors such as finance and trading, for example, as well as gaming, and cloud gaming more specifically, but also for cloud environments in general. Actually, network latency is an important factor for a multitude of types of businesses, especially since almost every organization nowadays works with cloud-based solutions. In the case of real-time communication requirements and real-time tracking, network latency requirements tend to be higher, but actually it plays a role in the performance of any cloud-based or otherwise hosted (web) application.
Video conferencing is a good example of an application featuring real-time communication principles that’s used by many businesses where the network latency affects how quickly audio and video data can be transferred and received between participants. Even a slight delay in video conferencing might result in issues like grainy video, stuttering audio, and delayed responses to messages or commands. Participants may find it challenging to communicate clearly as a result which might frustrate them and lower productivity. Real-time communication sure is better off by establishing the lowest network latency, guaranteeing that audio and video data are transferred rapidly and without delay. Lower latencies will result in fewer interruptions, making a dialogue flow more naturally, facilitating better collaboration and communication.
What Factors Affect Network Latency?
The speed at which data is transmitted from the server source to the user is heavily influenced by network latency. While distance is a primary factor, it’s not the only factor that contributes to network latency that may come with a server setup in a data center environment. The efficiency of the network backbone to which the server is connected also plays a crucial role in determining latency. At Worldstream, we have built a worldwide network backbone over the years that is strategically constructed with ultra-low latency, often putting distance as a secondary concern.
The massive bandwidth of more than 10Tbit/s that the Worldstream network delivers at 45% utilization is another aspect. This enormous quantity of bandwidth also contributes to desired low network latency values. Bandwidth and latency are closely linked when it comes to network performance and creating optimal connections for a server environment. If there is a lot of available bandwidth, this will benefit limiting network latency values. If less bandwidth is available, this will have a detrimental effect on the latency values effectively achieved.
Intelligent routing is an important element within a network architecture that can impact latency. Data is usually transmitted across multiple autonomous systems. The network backbone of Worldstream is also constructed in this manner. Data travels over several networks, and routers are responsible for processing and transferring it to its final destination. The data transmission process will take longer the more networks and Internet Exchange (IX) points it traverses. To properly route this network traffic to its intended location, the efficiency of network routers plays a crucial role. How rapidly data may be transported depends heavily on how effective these routers are and how efficiently this equipment does its job. The amount of time it takes for data to reach its destination may be considerably impacted by how quickly routers analyze and handle the data.
So, distance is not the only factor that can contribute to network latency within a server setup in a data center. Network efficiency, routing, and router efficiency all play critical roles in determining the speed at which data is transmitted. By considering these factors, server operators can address the network latency and provide their (web) application users with an optimal experience.
Network latency will also be impacted by a server’s capacity, speed, and configuration. Although we won’t go into detail about the server configuration’s impact on network latency in this blog article, it is important to keep in mind that it too contributes to achieving low network latency values. Only the network’s function in generating ideal latency values will be covered in this article.
Network Bandwidth vs. Latency
As said, bandwidth and latency are two critical factors that are related to each other while they can greatly impact the performance and user experience of servers and the applications running on them. However, their significance varies based on the type of server as well as the operations it performs.
In the case of gaming servers, for example, network bandwidth is a critical factor when considering multi-player use cases. It determines how many gamers can connect and play the game simultaneously. High network bandwidth allows more players to connect and participate in the game without any lag or delay. On the other hand, if the server has a low bandwidth connection and multiple players are playing the game simultaneously, it can lead to lag, and players may experience observable delays between their input and the gaming character’s behavior, leading to a subpar gaming experience. The effect of bandwidth can be less important though if there are not that many players gaming at once on the same connection.
For (live) streaming applications running on servers, both bandwidth and latency are crucial factors. Low bandwidth can result in lengthy buffering times or poor video quality, causing a frustrating experience for users. In contrast, high latency can lead to a noticeable delay in live streaming, making it difficult for users to enjoy real-time content.
Web hosting applications running on servers also heavily rely on bandwidth and latency. Low bandwidth can cause websites to load slowly, which can lead to a higher bounce rate and lower conversion rate. On the other hand, high latency can create a frustrating wait at the beginning, causing users to lose interest and abandon the website.
So, bandwidth and latency are both crucial factors that can significantly impact the performance and user experience of servers and the applications running on it. Servers that require high data transfer rates, such as with gaming and (live) streaming servers, usually require high bandwidth connections to ensure optimal performance. Web hosting servers, on the other hand, tend to require a balance of both bandwidth and latency to provide a fast and reliable user experience.
Network Latency – How to Measure It
Measuring network latency can be essential for server deployments because it directly affects the performance of network-dependent applications and services deployed and operated on these servers. Latency can be measured using two key metrics: Time to First Byte (TTFB) and Round Trip Time (RTT). TTFB calculates the time it takes for the first byte of data to reach the origin server after a client sends a request, while RTT measures the time it takes for a data packet to travel from the user’s browser to a network server and back. While ultra-low latency networks require analysis in nanoseconds (ns), administrators typically monitor TTFB and RTT in milliseconds (ms).
To measure network latency, administrators commonly use three methods: Ping, traceroute, and MTR (My Traceroute). Ping checks a host’s reachability on an IP network and provides information on around half the latency value of the network. Traceroute tests reachability and records the route packets take to reach the host, while MTR combines the ping and traceroute methods more thoroughly. By using these methods, administrators can gain insights into the network latency and identify any issues that may be impacting performance.
In terms of an overall understanding of network performance and the effects it may have on a server environment and the applications running on it, we are still not there yet with only elaborating on the concepts of network latency and bandwidth. Throughput, jitter, and packet loss are other variables that may affect network performance and therefore the performance of server-deployed IT applications in general.
Throughput refers to the volume of network traffic moving at any given moment from a source or collection of sources to a particular destination or group of destinations. Essentially, it measures the speed and efficiency of data transfer. Throughput can be expressed as a number of packets, bytes, or bits per second, with the most common unit of measurement being Mbit/s, or megabits per second. Understanding throughput as part of the overall network operations is crucial for ensuring efficient data transfer for any server-deployed IT application.Throughput determines the number of packets/messages that can be delivered successfully to their intended destinations, making it another essential metric for evaluating network performance. High throughput is achieved when a majority of messages are delivered successfully, whereas a low success rate will lead to reduced throughput. A decrease in throughput may directly affect network performance, probably resulting in poor service quality. A proper packet delivery can be crucial to connect and communicate effectively. For instance, during a VoIP Voice over IP connection, low throughput can result in audio skips and poor communication quality for its users. So, it is essential to maintain good throughput levels for an optimal network performance.
Several factors can contribute to poor network throughput, but ineffective hardware performance is one of its key causes. The terms network bandwidth and throughput are often used interchangeably, although these sure are two network terms that give meaning to different network characteristics. Consider bandwidth to be the boundaries of a network connection. Throughput, on the other hand, is the actual pace at which data is sent through the network.
Just as with bandwidth, bitrate units are used to measure throughput. The bitrate is the quantity of bits processed in a given amount of time. Bits per second (bit/s) or kilobits per second (kbit/s) are the usual units of measurement. It calculates the volume of data that is moved from one network endpoint to another in a certain period of time.
Jitter – Streaming Video and Audio
In addition to network latency, bandwidth, and throughput, jitter is also something that can play a role in the performance of network communication and server-deployed IT applications using a network. It’s the variation in delay between packet flows from one network point to another that is referred to as jitter. Just as with network latency, jitter is measured in milliseconds.
Low levels of jitter won’t likely have a noticeable effect on a network experience, so these won’t necessarily cause a significant issue. There may even be brief, unpredictable anomalous jitter variations in certain circumstances. Jitter is less of an issue under these circumstances.
Streaming audio and video services are most affected by jitter. To reiterate the example of VoIP applications, it might be possible that jitter is to blame for VoIP conversations which are temporarily reduced in quality or even completely interrupted with significant portions of the conversation being lost or unclear.
Jitter in fact represents the degree of unpredictability in latency throughout a network, while latency is the amount of time it takes for data to travel from one network point to another and complete a round trip. High latency might be unpleasant, but jitter, or (unexpected) latency, can be just as annoying and bad for a service provider’s business operations. To ensure continuous network quality, especially when it comes to employing services like VoIP and live streaming on a server infrastructure, jitter sure must be addressed when it comes to establishing the highest network performance.
Jitter – How to Measure It
As with latency, jitter values can also be measured. To quantify network jitter and determine its impact on server-deployed IT applications, the average packet-to-packet delay time may be calculated. Different absolute packet delays in sequential network communications might be measured as an alternative. The kind of network traffic will have its influence on how to measure jitter precisely. Depending on whether there’s control over one or both network endpoints, the procedure for measuring jitter in VoIP communication will vary.
Calculating the mean round-trip time and the minimum round-trip time for a set of packets will allow for a ping jitter test if there’s only control over one network endpoint from a user perspective. The variation between the sending and receiving intervals for a single packet is referred to as a real-time jitter measurement, and it may be used to quantify jitter if there’s control over both network ends. When multiple packets are being transmitted, jitter can be calculated as the average difference between real-time jitter measurements and the average real-time jitter across all network packets.
Network Packet Loss
When packets are transported via a network but one or more of them are lost in the transmission, this is known as packet loss. Applications that need real-time data transfer are the largest sufferers of packet loss. Online video games, Voice over IP, and video-based collaboration tools are a few examples of these. Network congestion, malfunctioning or old network gear, as well as software issues may contribute to packet loss.
Network congestion is one of the most frequent causes of network packet loss. When a connection is operating at near to its maximum throughput, packets might start to be lost. Other common reasons include malfunctioning hardware, generic radio-related problems, and sometimes, equipment may intentionally lose packets to accomplish goals like reducing traffic throughput or for routing.
Packet loss will often slow down a network connection’s throughput or speed. When it comes to latency-sensitive protocols or applications like streaming video, video games, or VoIP, this might sometimes cause a drop in service quality.
By discussing the concepts of latency, bandwidth, throughput, jitter, and packet loss, we have touched on the key factors that determine the performance of a network. How the various network values can be determined can help tailor your specific server-deployed IT applications and a network design closely matching them.
Doing any of the computations mentioned in this article might be intimidating to some. If this is the case, for example with determining jitter levels, another practical way to examine jitter is via bandwidth testing. If you need support, with any kind of network measurement as well as a desired alignment with user applications, you can always consult Worldstream’s experts. Our engineers are very knowledgeable in this area and our support team will be happy to assist you.
To conclude, network latency is a delay that occurs while processing network data. It can take on a variety of shapes and scenarios within an IT environment and dedicated server setup. Low network latency is an important factor for a variety of businesses, especially those using cloud-based solutions. It also affects how quickly audio and video data can be transferred and received between participants, resulting in issues like grainy video, stuttering audio, and delayed responses. Lower latencies will result in fewer interruptions and better collaboration and communication.
It can especially be an important metric for latency-sensitive sectors such as finance and trading, gaming, videoconferencing, and VoIP, as well as for cloud environments in general. Network latency requirements tend to be higher in real-time communication requirements and real-time tracking, but in fact it plays a role in the performance of any cloud-based or otherwise hosted (web) application.
Network latency is in fact a major factor in data transmission. That’s why Worldstream has built a worldwide network backbone with ultra-low latency. Latency is also closely linked with bandwidth, while intelligent routing is an important element within a network architecture that can impact latency. Data travels over networks and routers are responsible for processing and transferring it to its destination. Router efficiency, routing, and router efficiency all play a critical role in determining the speed at which data is transmitted. Worldstream’s 10Tbit/s global network backbone is constructed in this manner, allowing for ultra-low latency levels.
Server capacity, speed, and configuration also contribute to achieving low network latency values, although this is beyond the scope of this blog article. What we can say about it in this article is the following. Worldstream’s server offerings delivered to our customers globally are unmanaged solutions that can always be tailored to customers’ latency needs and application requirements. It uniquely enables our clients to create end-to-end low-latency setups including the deployment of the network and the accompanying server configurations.
Network bandwidth and throughput are often used interchangeably, although these two terms give meaning to different network characteristics. Throughput is the volume of network traffic moving at any given moment from a source or collection of sources to a particular destination or group of destinations. Understanding throughput as part of the overall network operations is crucial for ensuring efficient data transfer for any server-deployed IT application. Several factors can contribute to poor network throughput, but ineffective hardware performance is one of its key causes. Worldstream’s network backbone is built with hardware of the highest quality and the most up-to-date technology, both of which contribute to the industry-leading throughput values that we’re able to achieve in our global network backbone.
Jitter must also be addressed when it comes to establishing the highest network performance. Jitter is a variation in delay between packet flows from one network point to another that can affect the performance of network communication and server-deployed IT applications. Jitter represents the degree of unpredictability in latency, while latency is the amount of time it takes for data to travel from one network point to another and complete a round trip. Streaming audio and video services are particularly affected by jitter. The fact that Worldstream is able to successfully service a large number of clients in this market segment is, of course, also indicative of the low jitter values of the network backbone that is provided to these customers.
About Worldstream & Its Global Network Backbone
Worldstream was founded in 2006 by childhood friends with a shared passion for gaming. Dissatisfied with the high costs and unreliability of game servers, they came up with the idea of offering better solutions. Since then, the Westland-based IT company has grown into an international player of IT infrastructure (IaaS).
Worldstream aims to uncomplicate the lives of IT leaders at tech companies. As a provider of data center, hardware, and network services, Worldstream serves various business markets, including Managed Service Providers (MSPs), System Integrators (SIs), Independent Software Vendors (ISVs), and web hosting companies. The key business objective of Worldstream is to give IT leaders peace of mind by providing high-quality infrastructure, industry-leading service, and strong partnerships that will get them excited about their IT infrastructure again.
Worldstream’s proprietary global network has 10 Tbit/s of bandwidth capacity available. This network’s maximum bandwidth usage is only 45%, guaranteeing server users exceptional scalability and ultimate DDoS protection. Our ultra-low latency global network backbone is reason for several customers to use it in their data center environment. Our experienced and knowledgeable engineering support department is available 24/7 to assist customers with their network and server deployments.
More about Worldstream’s global network backbone offering can be found here.
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