the other option available is mainly integrating
the existing voice with IP systems at different
points along the chain. The PBX could be IP
enabled; only the trunks could be IP-based or
maybe just the phones. Technology exists to
do everything from single device IP integration
to complete infrastructure replacement.
According to a recent survey among VoIP ven-
dors a majority saw a rapid growth in the num-
ber of VoIP products that would interoperate
within a foreseeable future. The figures given
were that by year-end of 2002 72 % of the VoIP
products will interoperate, by 2004 88 %, by
2005 94 % and finally by year-end 2005 100 %.
The transition to this new order will likely occur
gradually, emerging from organizations back
offices and special application workgroups. The
current paradigm consists of a circuit switched
fabric for voice networks and a complex separate
LAN infrastructure for data. The hybrid model,
deployed by some enterprises already, is the CTI
(Computer Telephony Integration). While most
of the data transfer takes place on specific data
networks, some have selectively deployed CTI
systems for specific applications, generally those
designed to generate revenue, such as telemar-
keting, or minimize costs, such as customer sup-
port. In a typical CTI system the incoming
caller’s telephony number is transferred to the
systems database, i.e. computer network, which
transforms the customer’s telephony number to
specific customer information. This information
is then displayed on the screen in front of the
specialist, sales or support personnel. The con-
nections between the telephony and data net-
works are loose. The market is severely re-
strained because it relies on proprietary connec-
tions between insular systems. Unlike the world
of packet telephony CTI relies on short distance
relationships, over proprietary lines, between
complementary systems (vendors) that have
each been optimized bilaterally for their specific
purpose. The final (?) telephony paradigm con-
sists of telephony and data tightly coupled on
packet based multimedia networks. In this sce-
nario, data and voice share a common transport
network and equipment. Designed with this in
mind the fabrics are capable of growing to sup-
port new services like video conferencing and
video streaming and voice mail. To be able to
support all these kinds of services the fabrics
have to be equipped with features to cope with
streams with different QoS demands. In this sce-
nario the telephony calls become transparent, i.e.
the users will not be able to tell whether a call is
placed over the packet switched network, the cir-
cuit switched network or a combination of the
two. This new scenario also has the potential to
spark fundamental shifts in collective business
behaviour, as people exploit the simultaneous
and joined delivery of data applications and
voice over a single unified network. This will
most likely provide unprecedented opportunities
for new enterprises to provide innovative appli-
cations.Voice Coders
Key technical requirements for coders include:- Low bandwidth (8 kb/s or less);
- High quality for voice (3.5 or higher on the
MOS1)(mean opinion score) scale); - Low latency.
In real-time transmission, up to 30 % of the
packets in a transmission could be lost or
delayed to an extent where they have to be cal-
culated as lost. A successful IP telephony appli-
cation then needs to recover from lost packets by
effectively reconstructing the lost data. The com-
plexity of coding algorithms has an impact as
well. High complexity increases the cost of the
host platform. G.723.1 [5] is emerging as a pop-
ular coding choice. It is an algorithm for com-
pressed digital audio over telephone lines. The
enduring requirement for coders, however, is
that IP telephony systems will be capable of sup-
porting multiple coders and adding more as tech-
nology emerges and popularity increases.Echo Cancellation
VoIP, using ordinary telephones at the end-
points, will cause echo problems and the gate-
ways have to perform some kind of echo cancel-
lation. The ordinary telephone switches do not
generally perform any echo cancellation on local
lines. The echo is present due to the exchange of
information/signals between the two wire and
four wire systems. The echo is however not a
problem on the local lines, the latency is not
long enough to come back as a separate trans-
mission. When using long distance lines echo
cancellation is performed within the telephony
system. On these lines the time it takes for the
signal to propagate back to the sender is long
enough to receive a quite disruptive signal.
Comparing VoIP to ordinary telephony makes us
discover one of the big differences. When VoIP
is used, with ordinary telephones at the end sta-
tions and an IP network in between, local lines1)For many years the industry has employed a rather subjective scale to determine the quality of a
conversation, defined in [4]. This test is based on a number of volunteer testers who listen to a voice
sample and grade it according to the following scale; 5:excellent, 4:good, 3:fair, 2:poor and 1:bad.