Network congestion and queueing

When driving a car, travel times on road networks often vary most due to traffic congestion. Similarly, the variability of packet delays on computer networks is most often due to queueing [25]:

  • • If several different nodes simultaneously try to send packets to the same destination, the network switch must queue them up and feed them into the destination network link one by one (as illustrated in Figure 8-2). On a busy network link, a packet may have to wait a while until it can get a slot (this is called network congestion). If there is so much incoming data that the switch queue fills up, the packet is dropped, so it needs to be resent—even though the network is functioning fine.
  • • When a packet reaches the destination machine, if all CPU cores are currently busy, the incoming request from the network is queued by the operating system until the application is ready to handle it. Depending on the load on the machine, this may take an arbitrary length of time.
  • • In virtualized environments, a running operating system is often paused for tens of milliseconds while another virtual machine uses a CPU core. During this time, the VM cannot consume any data from the network, so the incoming data is queued (buffered) by the virtual machine monitor [26], further increasing the variability of network delays.
  • • TCP performs flow control (also known as congestion avoidance or backpressure), in which a node limits its own rate of sending in order to avoid overloading a network link or the receiving node [27]. This means additional queueing at the sender before the data even enters the network.
If several machines send network traffic to the same destination, its switch queue can fill up. Here, ports 1, 2, and 4 are all trying to send packets to port 3

Figure 8-2. If several machines send network traffic to the same destination, its switch queue can fill up. Here, ports 1, 2, and 4 are all trying to send packets to port 3.

Moreover, TCP considers a packet to be lost if it is not acknowledged within some timeout (which is calculated from observed round-trip times), and lost packets are automatically retransmitted. Although the application does not see the packet loss and retransmission, it does see the resulting delay (waiting for the timeout to expire, and then waiting for the retransmitted packet to be acknowledged).

TCP Versus UDP

Some latency-sensitive applications, such as videoconferencing and Voice over IP (VoIP), use UDP rather than TCP. It’s a trade-off between reliability and variability of delays: as UDP does not perform flow control and does not retransmit lost packets, it avoids some of the reasons for variable network delays (although it is still susceptible to switch queues and scheduling delays).

UDP is a good choice in situations where delayed data is worthless. For example, in a VoIP phone call, there probably isn’t enough time to retransmit a lost packet before its data is due to be played over the loudspeakers. In this case, there’s no point in retransmitting the packet—the application must instead fill the missing packet’s time slot with silence (causing a brief interruption in the sound) and move on in the stream. The retry happens at the human layer instead. (“Could you repeat that please? The sound just cut out for a moment.”)

All of these factors contribute to the variability of network delays. Queueing delays have an especially wide range when a system is close to its maximum capacity: a system with plenty of spare capacity can easily drain queues, whereas in a highly utilized system, long queues can build up very quickly.

In public clouds and multi-tenant datacenters, resources are shared among many customers: the network links and switches, and even each machine’s network interface and CPUs (when running on virtual machines), are shared. Batch workloads such as MapReduce (see Chapter 10) can easily saturate network links. As you have no control over or insight into other customers’ usage of the shared resources, network delays can be highly variable if someone near you (a noisy neighbor) is using a lot of resources [28, 29].

In such environments, you can only choose timeouts experimentally: measure the distribution of network round-trip times over an extended period, and over many machines, to determine the expected variability of delays. Then, taking into account your application’s characteristics, you can determine an appropriate trade-off between failure detection delay and risk of premature timeouts.

Even better, rather than using configured constant timeouts, systems can continually measure response times and their variability (jitter), and automatically adjust timeouts according to the observed response time distribution. This can be done with a Phi Accrual failure detector [30], which is used for example in Akka and Cassandra [31]. TCP retransmission timeouts also work similarly [27].

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