(1) Early telemedicine networks used dedicated telecommunications circuits, but more recently the Internet has become widely used for telemedicine as it allows data transmission between any internet-connected locations. (2) There are several key performance criteria for evaluating digital networks for telemedicine - bandwidth, packet loss, end-to-end delay, jitter, and security and privacy. (3) If a telemedicine application is not performing satisfactorily over a network, a network engineer should be consulted to diagnose potential issues and make improvements.

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EDUCATION AND PRACTICE Technology
" Network basics for telemedicine
Jill Gemmill
University of Alabama at Birmingham, USA
Summary
Early telemedicine networks employed dedicated telecommunications circuits (e.g. leased digital lines) in which
the sender and receiver were connected by a private circuit. More recently, the Internet has become widely
available for general use, including telemedicine. The Internet was engineered to permit network paths to be
shared by all users, so data transmission is fundamentally different from traditional, circuit-switched networks.
Early telemedicine applications demonstrated the feasibility of Internet Protocol transmission. The basic
performance criteria to use in evaluating newer digital communications technologies that carry both voice and
data are: (1) bandwidth; (2) packet loss; (3) end-to-end delay; (4) jitter; (5) privacy and security. Network
engineering involves performance trade-offs between the hardware, architecture, security and the budget
available. A telemedicine application may be running over a network whose design is entirely under the user’s
control, or the application may employ some part of the Internet whose design is unknown to the user. If an
application is not running to satisfaction, then a network engineer should be consulted.
Introduction
In telecommunications generally, convergence is
taking place between video, voice and data networks.
For example, telephone companies now offer Internet
services in addition to voice, and Internet service
providers (ISPs) are offering digital voice services.
Convergence of the various technologies used for
health care at a distance can also be expected to occur.
Early telemedicine networks employed dedicated
telecommunications circuits. The underlying
communications medium was some form of digital
line, typically Integrated Services Digital Network
(ISDN) or T-1 lines. These special lines were required
because the transmission capacity of an ordinary
telephone line is not adequate for most telemedicine
applications. Permanent circuits, such as a T-1 line, are
paid for at a fixed annual cost and the charge is the
same whether the circuit is used a lot or a little. Dial-up
ISDN lines are usually paid for like telephone lines
where there is a fixed monthly rental and then a fee
per call.
From circuits to packets
Digital lines, such as T-1 or ISDN lines, are examples of
circuit-switched telecommunication. The sender and
receiver are connected by a private circuit. More
recently, the Internet has become widely available for
general use, including telemedicine. Data transmission
via the Internet is an example of packet-switching, i.e.
it takes place in a fundamentally different way from
transmission in circuit-switched networks.
The Internet was engineered to permit network paths
to be shared by all users. The Internet Protocol (IP) is a
standard that segments all data, voice and video into a
series of variable-sized chunks of information called
packets. Each packet has the destination address added
as a header. In the Internet itself, devices called routers
read the address on a data packet and forward it in the
general direction of the final destination. Ultimately,
after passing through several routers, an individual
packet will reach its destination. Packets from multiple
locations become interspersed as they flow along the
Internet, something like a conveyor belt that is being fed
packages from many delivery trucks at the same time.
Some packets are sent with delivery guaranteed, and
some are not (the reasons for not wanting a guarantee
in some cases are explained below). Information sent
along a circuit remains in order and travels along the
path defined by that circuit; in contrast, consecutive IP
Journal of Telemedicine and Telecare 2005; 11: 71–76
Correspondence: Jill Gemmill, IT Academic Computing, 701 20th Street
South AB 719-B, Birmingham, AL 35294-0107, USA
(Fax: þ 1 205 975 2855; Email: jgemmill@uab.edu)

2. packets may or may not take the same route to reach a
destination, and they may also arrive in any order. All
these delivery details occur invisibly to the user. This IP
engineering allows the Internet to be used by millions
of people simultaneously for multiple purposes such as
email, Web browsing, watching videos, telephone and
videoconferencing activities to and from any other
Internet-connected location. The routers can re-
configure themselves to avoid network paths that may
have become blocked due to damage or equipment
failure; this improves the resilience of the Internet.
Early telemedicine applications demonstrated the
feasibility of IP transmission. The basic performance
criteria to use in evaluating newer digital communi-
cations technologies that carry both voice and data are:
(1) bandwidth;
(2) packet loss;
(3) end-to-end delay;
(4) jitter;
(5) privacy and security.
Bandwidth
Bandwidth refers to the space required in the network
path so that all packets can get through – think of that
conveyor belt full of packages: the wider the belt, the
more packages that can be carried along without delay
and without falling off. If the conveyor belt is too
narrow for the number of trucks attempting to place
packages on it, there will be overcrowding and
packages will fall off or be delayed while waiting for
space to open up. Bandwidth must be available end
to end; if there is a bandwidth-limited segment
anywhere along the path, then it will act as the limiting
bottleneck. Think of six lanes of cars negotiating
their way into a two-lane tunnel; eventually, they
will all get through the tunnel but not at the speed at
which they were originally travelling on the six-lane
highway.
Network engineers agree that it is not a good idea to
operate network segments at maximum capacity (an
average bandwidth utilization rate of 100%). However,
there are differences of opinion as to what average
utilization is the best – 65–70% is frequently cited.1
In
other words, IP networks work best with at least one-
third excess average bandwidth available in order
to keep packets flowing and prevent congestion.
Bandwidth may not be provisioned symmetrically at
the edges of the Internet. For example, Figures 1 and 2
show real network utilization data from the University
of Alabama at Birmingham. The bandwidth available at
Figure 1 A network utilization graph. This is a snapshot of a gigabit router port connecting one
building to the campus network backbone at the University of Alabama. Note that this link is full duplex
and the average utilization as seen in 5-min sampling intervals was less than 1%. This graph was
produced by open source software called Multi Router Traffic Grapher (MRTG), available at http://
people.ee.ethz.ch/~oetiker/webtools/mrtg/mrtg.html
Figure 2 This graph shows utilization at the point that the entire campus backbone connects to the
public Internet. This connection currently operates at 100 Mbit/s and at this location in the same
network, utilization averages 70% of what is available
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72 Journal of Telemedicine and Telecare Volume 11 Number 2 2005

3. the edge of the campus, at the connection point to the
public Internet, is only a tenth of what is available at
the core of the campus network.
Symmetric bandwidth means that each end can send
and receive the same quantity of data at about the same
rate. In contrast, home ADSL lines, for example, usually
provide more bandwidth into the home than out of it.
Table 1 lists the common types of non-wireless Internet
connections, their associated bandwidth and example
applications that work well with that bandwidth
available. Note that a ‘bit’ is the smallest unit of digital
information; eight bits are required to represent one
character such as ‘a’ or ‘z’.
Packet loss
Packet loss occurs when packets fail to arrive at the
destination. Packet loss errors occur most commonly
on network links that are noisy, such as microwave,
satellite or other wireless links, but loss can occur in
any type of network. Over reliable transmission media
such as copper wires or optical fibres, packet loss is
usually the result of inadequate bandwidth somewhere
along the path. The Internet routers mentioned earlier
have a finite amount of storage space in which to
temporarily hold incoming packets while the
destination address is scanned and the appropriate
outgoing route is determined. Congestion causing
packet loss occurs when a large number of incoming
packets arrive faster than the router can handle them;
soon there is no more temporary storage in the router
and any additional incoming packets are discarded.
Packet loss is not necessarily cause for alarm because
the engineering of the Internet provides methods for
recovering from the problem.
As mentioned above, the Internet supports both
reliable and unreliable packet delivery. Reliable delivery
makes use of the IP Transmission Control Protocol
(TCP). If TCP finds that a packet on the receiving side is
missing, incomplete or corrupted, it will request that
this packet be resent. In addition, packets that have
been successfully received are acknowledged back to
the sending side. Because of TCP, data files such as
email messages or images can be transferred across the
Internet with both sender and receiver assured that the
material has arrived exactly as sent. Where TCP is used,
any packet loss that occurs will cause the TCP to
request retransmission. All the data will eventually be
delivered, but the total transfer time may be longer
than expected.
There are certain situations, however, where
unreliable Internet delivery is desirable. This is usually
for applications where near realtime delivery is needed,
for example when live audio and/or video is being
transmitted. If you are watching a live conference over
the Internet, and some packets get lost on their way to
you, the unreliable transport would skip over what is
missing and keep going – this makes sense, since you
want to keep up with the live content and are not
interested in a piece of sound or image that arrives 1 s
out of sequence. Audio and video that are streamed
across the Internet for live listening and viewing are
typically sent using unreliable transport, the User
Datagram Protocol (UDP) rather than TCP (see
Figure 3).
In summary, some minor packet loss can occur for
both reliable and unreliable transport without ill effect
on the application. Obviously, if the rate of packet loss
is too high, it will result in excruciatingly slow file
transfer or unintelligible audio/video, either of which
would be unacceptable for telemedicine.
End-to-end delay
Delay, sometimes referred to as latency, is the interval
between sending a packet and its reception at the far
end. Digital telecommunication, particularly over long
distances, depends on optical fibre transmission, where
information is transmitted at the speed of light. Even at
light speed, on a very long distance network path there
may be a latency of several tens of milliseconds.
Jitter
Jitter refers to the random variation in packet delay.
This delay is principally due to the length of time that a
packet spends waiting inside routers. Traffic patterns on
the Internet have been studied extensively and the
pattern of activity is bursty in nature, meaning that
longer periods of brief activities are interspersed with
big traffic spikes. A high rate of jitter corresponds to a
‘jerky’ arrival rate that could be unsuitable for operat-
ing a sensitive remote instrument or for satisfactory
videoconferencing. Jitter is one of the reasons that
packets may arrive in a different order than they
were sent.
Security and privacy
Part of the challenge in moving from circuit to packet
transmission is that rather than having a private
communication channel, the packets traverse a very
public network, the Internet. Because of privacy and
confidentiality concerns, any telehealth application
will require assurance that the information in transit is
secure.
A common solution is to use a virtual private
network (VPN). VPNs provide an encrypted
communication channel from the Internet to the edge
of an institution’s network. This approach would be
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Journal of Telemedicine and Telecare Volume 11 Number 2 2005 73

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suitable for a doctor to use when accessing records
held at his/her clinic from home. An encryption
scheme such as the Federal Information Processing
Standard (FIPS) for the Advanced Encryption Standard
(AES) encodes the content of each packet so that it is
unreadable while transiting the Internet. Note that
encryption adds about 30% overhead to the size of a
transmission, and some additional delay is added
due to the time it takes to encrypt and then decrypt
each packet.
Using Public Key Infrastructure (PKI), it is possible to
sign email messages, encrypt messages and guarantee
that only one specific person can decode and read the
message. PKI relies on digital certificates provided by a
certificate authority which (1) vouches that it has
verified your identity and (2) issues you with a pair of
digital keys, the private and public keys. You can then
sign messages with your private key, and recipients
can use your public key to verify the validity of your
signature. By using the public key A of the person to
whom your message is addressed and your own private
key B, it is possible to encrypt a message so that only
the holder of the private key A matching public key A
can decipher the message.
Securing general communication from any location
to any other location is, however, still a difficult
problem. VPNs are created so that you can access one
particular location securely, but not any location you
like. PKI is not yet widely deployed and is not yet well
integrated with video and voice applications. An
excellent description of how difficult it is to secure
Voice Over IP communications entitled Security
Considerations for Voice Over IP Systems was recently
published by the National Institute of Standards.2
In taking the steps necessary to secure enterprise
networks or important application servers from break-
ins or ‘denial of service’ attacks, you may feel that the
cure is worse than the disease. Security devices such as
firewalls and network address translator (NAT) devices
protect your important network elements by hiding
them and making them inaccessible from the public
Internet. Unfortunately, some applications such as
video and voice are likely to stop working. The ViDe
Videoconferencing Cookbook3
section entitled ‘Network
Matters’ contains a good description of what causes
firewalls and NATs to behave as insurmountable or at
least very unfriendly obstacles for videoconferencing
and voice over IP. One possible solution is to establish
a ‘demilitarized zone’ (DMZ) outside the firewall to
host services that are expected to be more publicly
accessible. A slightly different approach uses a
gateway or proxy to more carefully manage inbound
traffic that might be eligible to pass through the
firewall.
Evaluating networks for telemedicine
Network engineering involves performance trade-offs
between the hardware, architecture, security and the
budget available. In general, you will want to obtain as
much bandwidth as possible, monitoring average
utilization to make sure that you have adequate but not
excessive capacity. Where it is not possible to obtain
the necessary bandwidth, it may be possible to use
special router techniques, such as tagging video and
voice packets so that they are treated as priority traffic.
In a recent project in Alabama, the use of packet-
tagging was demonstrated in a teledermatology project
using only a T-1 Internet connection that was simul-
taneously being used for other purposes.
If a specific application does not seem to be perform-
ing well over your network, a network engineer who
understands the characteristics of the applications may
be able to solve the problem. The ‘duplex mismatch
problem’ is a fairly common and even frequently
Figure 3 Functional layers of an Ethernet network. Each layer has a well-defined interface to the layer
above and the layer below; the application and other upper layers have no need to know about what
physical medium (e.g. copper wire or optical fibre) may lie below
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Journal of Telemedicine and Telecare Volume 11 Number 2 2005 75

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undiagnosed problem. Most workstations, servers and
other network equipment are fitted with internal
network cards that are factory configured to ‘auto-
negotiate’ a correct setting when connected to your
network. Auto-negotiation is a protocol to determine
the highest speed available at both ends of the link (the
standard speeds being 10,100 or 1000 Mbit/s). Auto-
negotiation also involves selecting either half duplex
(the link is in either send or receive mode, but never
both simultaneously) or full duplex (the link can both
send and receive at the same time, which effectively
doubles the available bandwidth).
Unfortunately, auto-negotiation does not always
work correctly and a bad choice can cause terrible and
perplexing performance issues. The speed is usually
negotiated correctly, but the duplex may not be
correctly matched. ‘One of the most common causes of
performance issues on 10/100 Mbit/s ethernet links is
when one port on the link is operating at half duplex
while the other port is operating at full duplex’.4
Microsoft workstations will display the configuration
you requested, but not the configuration the devices
may have actually selected. Re-negotiation of settings
at a router may introduce a significant amount of jitter
into a previously well-behaved application. It is
possible to detect duplex mismatch errors by
examining network and router switch statistics. If one
end of a link is set to full duplex while the other end is
half duplex, statistics on the full duplex side will show
a high level of CRC or alignment errors while the half
duplex side reports a high number of late collisions.
Vigilant monitoring of all network devices, and even
automated monitoring with alarm conditions, is a
good management practice.
Conclusion
A telemedicine application may be running over a
network whose design is entirely under your control, or
your application may use some part of the Internet
whose design is unknown to you. If your application is
not running to your satisfaction, then a network
engineer should be consulted. There are a number of
tests5
that can be run by the user which may help
diagnose whether there is a bandwidth bottleneck,
duplex mismatch or other identifiable problem in the
path of interest. Everyone involved should have a
common understanding of what is meant by ‘secure’
and ‘satisfactory performance’. Guidelines for specific
telemedicine applications, such as dermatology or
psychiatry, may be available.6
References
1 Lantronix: Learning Center. http://www.lantronix.com/learning/
net-tutor-switching.html (last checked 22 December 2004)
2 Kuhn RD, Walsh TJ, Fries S. Security Considerations for Voice Over IP
Systems Recommendations of the National Institute of Standards and
Technology. http://csrc.nist.gov/publications/drafts/NIST_SP800-58-
040502.pdf (last checked 15 November 2004)
3 ViDe Videoconferencing Initiative. ViDe VideoConferencing
Cookbook. http://www.vide.net/cookbook/ (last checked 22
December 2004)
4 Cisco Systems. Configuring and Troubleshooting Ethernet 10/100/
1000Mb Half/Full Duplex Auto-Negotiation. Document ID: 10561.
http://www.cisco.com/warp/public/473/3.html (last checked
18 January 2005)
5 Burgiss S, Sprang R, Tracy J, eds. Telehealth: Technology Guidelines.
http://telehealth.hrsa.gov/pubs/tech/techhome.htm (last checked
18 January 2005)
6 SURFnet Detective. http://detective.surfnet.nl/en/ (last checked 22
December 2004)
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76 Journal of Telemedicine and Telecare Volume 11 Number 2 2005