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The Importance of Latency for the User Experience

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The Importance of Latency on the User Experience

Network latency refers to the time it takes for a data packet to travel from one point in a network to another. It's also important to consider round-trip time (RTT), which is the time it takes for data to travel from one point in a network to another and back again. RTT is a significant measurement because it directly impacts the user experience regardless of bandwidth.

Latency and TCP data

TCP data is used for business critical applications, cloud services, web pages, and other purposes. The performance of TCP data depends on the RTT for both upstream and downstream data transmission. If delay occurs in either direction then it can affect the user experience. If the delay gets too long the user connections will disconnect, resulting in the worst possible user experience.

TCP data is loss intolerant, which means if a packet is lost in transmission it needs to be recovered. All the data that follows a missing packet cannot be processed, and the recovery process dramatically increases the RTT, including late packets. Visualware NCS devices can detect data quality issues such as retransmissions, duplicates, and losses (bytes and packets).

A statement we hear frequently is: "I have a 1 Gbps connection, that's all that matters!"

In reality, latency matters more. The maximum route throughput - the amount of data that can be transmitted over a network connection in a given amount of time - is determined by the latency. For example, the maximum route throughput for a 15 ms round-trip is 34.95Mbps. Why is this?

The TCP receive window size is the amount of receive data (524,280 bits) that can be buffered during a connection. The sending computer can send only that amount of data before it must wait for an acknowledgment and window update from the receiving computer. The Windows OS is designed to self-tune and uses larger default window sizes when possible.

Divide the round-trip time into the window size (524,280 / 15) to get the maximum throughput achievable for one window of data, which in this case is 34.95Mbps.

As the Windows OS self-tunes, the window size may be increased, but don't know by how much.

It's important to note, sending more data doesn't necessarily help. It leaves less bandwidth for other services or applications trying to use the connection at the same time.

Having one window of data performing at 100% efficiency and quality of service is preferable over many windows performing inadequately.

Latency and UDP data (VoIP, Video streaming)

In the case of real-time voice data transmitted over UDP, the RTT can have a significant impact on the quality of the voice data transmission.

UDP is a low-latency protocol that does not provide error checking or retransmission of lost packets, unlike TCP. This means that if a packet is lost during transmission, there is no mechanism for the receiver to request that the sender resend the packet. As a result, lost packets can have a significant impact on the quality of the voice transmission.

The RTT can affect UDP voice data transmission in several ways:

To minimize the impact of RTT on UDP voice data transmission, it is important to have a low-latency network connection with a stable and consistent RTT. Other techniques, such as using jitter buffers, can help mitigate the effects of jitter and packet loss on voice data transmission. Additionally, using compression techniques, such as silence suppression, can reduce the amount of data that needs to be transmitted, which can help minimize the impact of RTT on UDP voice data transmission.

RTT Consistency

Another important factor regarding latency is the consistency of the trip-time (RTT). It's important for several reasons.

Improving RTT

Here are some ways to improve RTT.

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