Preparing Your TCP/IP Data Network For Voice Traffic

Posted on January 13, 2007 
Filed Under VoIP

As much as vendors would like you to believe, employing voice
applications over your existing TCP/IP data network is certainly
not as simple as plugging in VoIP-enabled phones and installing
software to make them work. Combining voice and data networks
into one seamless operation can be tricky.

Before you attempt to run voice communication over your TCP/IP
network, familiarize yourself with the following key issues in
order to avoid any unpleasant surprises.

Voice vs Data

VoIP enables the human voice to be sent over networks as data
“packets”. These packets are then reorganized into the human
voice upon reaching their final destination. One would think
that non-voice traffic travels over the network in the same
manner as data traffic. After all, data is data, right?

Wrong. The reasons are that TCP/IP networks do not generally
deliver “packets” of data in the same order, along the same
route, or even within the same time frame. This is not a problem
for normal data downloads or data transfer, but for voice
conversations it is critical that “packet” information is
transferred without packet loss or latency.

Bandwidth

It goes without saying that in order to run voice over a TCP/IP
network, sufficient bandwidth is required. Most network services
customers are familiar with the raw bandwidth of each of their
connections. The key issue here is not to confuse “available”
bandwidth with “total” bandwidth. For example, a T-1 devoted to
data networking may have 1.5 Mb of raw bandwidth. That does not
mean, however, that the entire 1.5 MB of bandwidth will be
available for voice applications.

Packet Loss

Inherent in any network is the inevitability of “packet loss”.
Packet loss refers to the percentage of data packets that travel
the network then fail to reach their final destination. Packet
loss can be tested and measured using network analysis tools. If
you test and determine a packet loss of 3% or more, your
existing network will not successfully handle voice traffic.

Keep in mind that packet loss increases dramatically when a
network is overloaded with traffic. In fact, a network may even
become unusable for voice applications when approaching their
maximum bandwidth capabilities.

Jitter

Packets of voice information traveling across a network take
varying amounts of time to go from one end to the other. This
variation is referred to as “jitter”. The receiving end of a
VoIP voice call “buffers” packet information so it can be played
as a smooth and unbroken stream of voice audio. The depth of
jitter (measured in milliseconds) can and should be measured.
Always be sure that jitter settings match the behavior of the
network. Dropouts may occur if the setting is too low, and
delays in the audio will occur if the setting is too high.

Latency

The total amount of time it takes for a packet of voice
information to get from one end of the network to the other is
called latency. Latency is also measured in milliseconds. A
latency of 200 or more milliseconds can result in echo,
especially if the connections at the receiving end are not all
digital. A latency of more than 400 milliseconds results in both
parties of the call constantly “interrupting” each other, then
waiting for the other person to finish. This situation is simply
not acceptable for even the most patient of callers.

Codecs

A codec is responsible for converting the analog voice signal of
a phone call to digital packets of information - then converting
them back to analog voice audio. There are many types of codecs
available depending on available bandwidth and the quality of
the audio that is desired. First determine the amount of voice
data traffic you anticipate having, then choose the appropriate
codec. The G.711 codec is widely used throughout North America
and although it consumes up to 83 kB per second of bandwidth it
provides toll-quality voice connections.

Configuration for Quality of Service (QOS)

The most complicated and difficult issue you will encounter will
be how to successfully configure the network to handle both data
and voice packets simultaneously. File downloads and other data
transfers that occur at the same time as voice calls can easily
interfere and even interrupt these voice conversations if the
network is not configured properly.

It is the job of the routers to treat voice packet information
in a special way. Without routers giving voice packets special
treatment, they will almost always lose the battle when in
direct competition with data packets. The configuration of
routers to do this properly is called “Quality of Service”, or
QOS. There are four types of configurations of QOS. Each provide
different levels of efficiency for handling voice and data
traffic simultaneously.

1) Best-Effort QOS
This configuration is the most
inefficient and one that most network routers are configured by
default. Voice traffic may sound fine with this configuration,
although any large data downloads will easily interrupt voice
conversations.

2) Differentiated Service
One way to solve the problem of
competition between voice and data packets is to configure
routers to simply determine the difference between the two types
of information, then handle them accordingly. Differentiated
service allows for routers to use different schemes for handling
the two types of traffic.

3) Dedicated Service
Routers can be configured to ensure
that sufficient bandwidth is always available for voice traffic.
This configuration tells the router to never use the dedicated
bandwidth for data transmission. Although it can be complicated
to configure routers with dedicated service, it does a good job
of eliminating the problem of data traffic interfering with
voice communications. One major disadvantage, however, is that
the “dedicated” portion of the network will go unused when there
is no voice traffic.

4) Guaranteed Service
The most complex and expensive
option to packet competition is guaranteed service. This
configuration allows routers to set up dedicated but temporary
bandwidth for each individual call. When a call has ended, the
bandwidth then becomes available for other voice calls or data
traffic.

The ability to use data networks for voice applications is an
attractive one although not always simple and straightforward.
Proper planning and testing will help you avoid the inevitable
pitfalls of configuring voice applications over data networks.

Author: Robert Potter
www.teconassociates.com

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