QoS

What is QoS?

QoS is Quality of Service. With limited bandwidth resources, QoS allocates bandwidth for various services and provides end-to-end quality of service assurance for the services. For example, voice, video, and critical data applications can be prioritized for service in network devices by configuring QoS.

QoS Metrics

Factors that affect the quality of a network include the bandwidth of the transmission link, message delivery delay and jitter, and packet loss rate, which become metrics of QoS.

Bandwidth

Bandwidth, also known as throughput, is the maximum number of bits of data that can be transmitted from one end of the network to the other in a fixed period of time (1 second), or it can be interpreted as the average rate of a particular data stream between two nodes of the network. There are two common bandwidth-related concepts in networking: uplink rate and downlink rate. The uplink rate is the rate at which data is transmitted when a user sends information to the network, and the downlink rate is the rate at which the network sends information to the user. For example, if a user uploads a file to the network via FTP, it is the upstream rate that affects the speed of the uploaded file; and if a user downloads a file from the network, it is the downstream rate that affects the speed of the downloaded file.

Latency

Latency is the delay time required for a message or packet to travel from the sending end to the receiving end of a network, and is generally composed of transmission delay and processing delay. In the case of voice transmission, for example, the delay is the time from when the speaker begins to speak to when the other party hears what is being said. Generally people do not notice a delay of less than 100 milliseconds. When the delay is between 100 milliseconds and 300 milliseconds, the speaker can detect a slight pause in the other party's reply, a pause that may make both parties to the call feel uncomfortable. Beyond 300 milliseconds, the delay becomes noticeable and users begin to wait for each other to reply. When one side of the call does not receive the desired reply in time, the speaker may repeat what was said, which will collide with the delayed reply at the far end, resulting in a repetition.

Jitter

Jitter is important for real-time transmission, especially for real-time services like voice and video.  Jitter also affects the processing of some network protocols. Some protocols send interactive messages at fixed intervals, and too much jitter can cause the protocol to oscillate. All transmission systems have jitter, and as long as the jitter is within specified tolerances it will not affect the quality of service. Excessive jitter can be overcome by utilizing caching, but this will increase latency.

Packet Loss Rate

Packet loss rate is the number of packets lost during network transmission as a percentage of the total number of packets transmitted. A small amount of packet loss does not have a significant impact on the service. For example, in voice transmission, the loss of a bit or a packet of information is often unnoticed by both parties to the call. In video transmission, the loss of a bit or a packet may cause momentary waveform interference on the screen but can quickly return to normal. Transmitting data using TCP can handle a small number of packet losses because TCP allows lost information to be retransmitted. However, a large number of dropped packets can affect transmission efficiency. In QoS, we are concerned with the statistics of packet loss, also known as the packet loss rate. So for normal transmission, the network packet loss rate should just be kept within a certain range.

Importance of QoS

Services in an IP network can be categorized into real-time services and non-real-time services. Real-time services often occupy a fixed bandwidth and have a clear perception of changes in network quality, requiring high stability of network quality, such as voice services. Non-real-time services occupy bandwidth that is difficult to predict, and bursty traffic often occurs. Burst traffic can lead to network quality degradation, which can cause network congestion, increase forwarding delay, and in severe cases, packet loss, resulting in service quality degradation or even unavailability.

The best way to solve network congestion is to increase the network bandwidth, but from the operation and maintenance costs, this is unrealistic, the most effective solution is to apply a "guaranteed" policy to manage network traffic.

QoS is generally used to protect the quality of important services when there is unexpected traffic in the network. If the service for a long time can not meet the quality of service requirements (such as service traffic for a long time more than the bandwidth limit), you need to consider the network expansion or the use of specialized equipment based on the upper layers of the application to control the service.

In recent years, there has been an explosion in the use of video, with almost everyone now owning a smartphone capable of shooting high-resolution video anywhere, anytime. Meanwhile, with the emergence of social networking sites, sharing and posting videos has become everyone's daily behavior, and people can post their own videos to share with others no matter where they are. For enterprises, HD video conferencing, HD video surveillance and other applications also generate a lot of HD video traffic in the network. Compared with voice traffic, video traffic takes up more bandwidth and is more unstable, especially some interactive video, which has very high real-time requirements.