By Kevin Davis
Adapted from “Sources of Latency” Whitepaper

When network users call the Help Desk to report poor application performance, you don’t typically hear things like “The router’s CPU is too busy!,” “The network utilization is above 70%!,” or “The carrier path has failed-over to a sub-optimal path.” Instead, what you’re likely to hear is “The network is slow” or “The calls on my IP phone sound terrible.”

Complaints that end-users lodge are nearly always based their quality of experience using the application. And their quality of experience is almost always reliant on time.

Anytime a significant delay occurs in the delivery of network data, application performance suffers. Depending on the type of application and how it works, variances in network delay can have a severe impact on application performance thereby degrading end-user’s experiences.

Two important measurements of time intervals in network transmission systems are referred to as “latency” and “jitter”. Understanding latency and jitter sources and how their values vary in network architectures is critical to engineering application performance and optimizing information resources. For many regular readers, this will be old-hat, but we’ll go over it again.

Network latency is the amount of time it takes for a packet to be transmitted end-to-end across a network and is composed of five variables:

Network Latency = (Distance Delay) + (Serialization Delay) + (Queue Delay) + (Forwarding Delay) + (Protocol Delay)

Serialization Delay refers to the amount of time it takes for a network interface (such as a router’s interface or computer’s NIC) to perform bitwise transmission of a frame unto the outbound media, Forwarding Delay is the amount of time it takes a network device to process a frame/packet by performing a destination address lookup and forwarding the frame/packet to the outbound interface, and Protocol Delay is the amount of time that access or transmission algorithms may contribute to the delay of a network frame, and is typically introduced at the endpoints of the data transmission system.

Serialization delay, on a per-packet basis, becomes insignificant at data rates above 1.544 Mbits/s – or a T1. Forwarding delay is typically insignificant in modern routers and switches (when appropriately configured – significant delay can occur in misconfigured routers.) And Protocol delay typically occurs at the access layer or the end points. So the two major variables that have the most effect on network latency are Distance Delay and Queue Delay.

Distance Delay is simply the minimum amount of time that it takes the electrical signals that represent bits to travel down the physical wire. Optical cable sends bits at about ~5.5 µs/km, copper cable sends it at ~5.606 µs/km, and satellite sends bits at ~3.3 µs/km. (There are a few additional microseconds of delay from amplifying repeaters in optical cable, but compared to distance, the delay is negligible.)

Distance delay can have a significant impact on application performance for applications that require a large number of network round trips in order to complete a transaction – for example, custom transactional based applications, database queries, and VoIP, which begins do degrade when one-way end-to-end latency exceeds 200-220 milliseconds.

One of the biggest sources of end-user ire are database queries designed to run over a LAN ported to the WAN. For example if a user executes a SQL database query that requests 100 rows of a database table, one row at a time, over a link with a latency due to distance of 60 ms, it would take approximately 6 seconds (60 ms * 100 turns) to complete the transaction. The same query executed by a user on a LAN connected to the same database server would take less than 2-3 ms to be completed, as the latency due to distance across the LAN is insignificant.

Queue Delay is the amount of time a packet must spend in a network buffer waiting its turn to be transmitted. Network interfaces transmit one frame at a time, typically one bit at a time. As such, when two or more packets are forwarded to a network interface at the same time, or close to the same time – one packet is transmitted while the others are put in a queue on the interface buffer to await their turn at the interface. Packets that are put into the queue must wait until they can be transmitted, adding milliseconds of delay.

Increases in Queue Delay can be measured and detected by monitoring traffic along a given network path. Typically, most intermittent increases in latency above the baseline distance latency can be attributed to network congestion. (In order to reduce the possibility of excessive queue delay, application servers that are members of the same application architecture should be placed on the same Ethernet switch and on the same VLAN to ensure they do not have to compete for uplink bandwidth when problems like the one pictured above occur.)

Worse still, if the problem gets worse and packets wait in increasingly longer lines within the queue, the buffer may become full and the packets may be dropped. Packet drop, in turn, causes TCP connections to throttle back on the rate of transmission.

Those are some of the main causes of latency – but what about jitter?

Jitter is a term that refers to the variance in the arrival rate of packets from the same data flow, and abnormal jitter values can negatively impact real-time applications like VoIP and video. Jitter is typically created by three different mechanisms in a network: variance in Serialization Delays due to variance in packet sizes, variance in per-packet Queue Delay due to packet spacing from multiple sources at a common outbound interface, or packets taking different routes from source to destination – perhaps due to per-packet load sharing or routing issues.

The most effective way to deal with jitter is by using low-latency queuing for VoIP and video traffic on network interfaces with large serialization and/or queue delays. In addition, endpoints (such as IP phones) can use jitter buffers or playout delay buffers in order to deliver received packets at a constant rate to the end consumer. These buffers are typically 30-50 ms in depth, and thus they attempt to manage jitter values within these values on any single one-way path. While these buffers technically add 30-50ms in latency, they significantly reduce jitter. Since human beings don’t start to notice latency in VoIP or VideoIP applications till it hits about 200ms, if latency can be kept to under 150 milliseconds, then jitter can be significantly reduced using this method.