1. Roaming the VoIP Inner Side
a general depiction of realizing voice communication through the Internet.
The basic VoIP components are made up by specific protocols informing both
parties as well as any intermediary (gateway, VoIP provider and so on) on
the kind of content to be transmitted (either audio or video), how the call
to be both commenced and terminated, along with the digitalization as well
as compression-decompression codec to be used.
Up until a while ago, when incoming-call providers facilitating VoIP applications
started making their appearance, communication between VoIP users and any fixed
telephone communication system had not been feasible.
Communication between two VoIP applications or appliances,
initially involves support by common protocols and codecs at the same time.
In case such a communication is rendered unlikely to be accomplished, it is
possible that an in-between provider will offer a codec conversion of some
sort, so that communication is carried out [Transcoding].
Real-Time Transport Protocol [RTP] & Real-Time Control Protocol
is practically used in any audio-video transmission by all networks. It has
taken on the-as far as possible-correct audio-video packets transmission
from point A to point B, avoiding thus delay mishaps, jitter and other inter-network
awkward situations. It does not carry out any other procedures, like, for
instance, service quality control or network resource management.
SIP & H.323
No more nor less than the two
biggest VoIP players. They both set up a career around 1995, when the need
for audio-video network transmission
showed up. They have undertaken the task of notifying a receiver that there
has been a requirement for communication by the caller. The next step is to
sort out the type of communication, who is going to take part in it and so
on and so forth.
The basic differentiation between the two protocols is focused on the fact
that H.323 is dependent on a relatively “crabbed code”, while the
Session Initiation Protocol [SIP] is based on text instructions, like the HTTP
protocol. The former (H.323) comes from traditional telephone networks and
likewise cooperation with such networks becomes easier, yet the location as
well as handling of any problems-should they arise-is more difficult when compared
to the SIP.
The H.323 is in fact an “umbrella” of protocols specializing in
a range of communication sections [H.225, H.245, T.38 used in fax over IP and
so on), while both are using several different Internet protocols to be able
to cover any gaps coming up when achievement of voice communication is in progress.
Here, for the sake of information, it would be appropriate if we referred to
the Session Description Protocol [SDP], the Resource Reservation Setup Protocol
[RSVP] to ensure high voice quality and of course the UDP/TCP/IP ones, used
to break up packets and route them to the Internet.
Even though there is a sort of a conflict at a development and management providers
level, it is more than evident that the SIP tends to become the model of today’s
Internet voice communication, not to mention the inclusion of video while calling,
when circumstances become favorable.
Other VoIP protocols
Apart from the H.232 and the SIP there are other voice communication protocols
achieved by net infrastructures, like the H.248 and the Media Gateway Control
Protocol [MGCP], which are specialized in gateway communication. Gateways
are the links between a network-ie, the IP-with either a fixed [PSTN] or
telephony network. There are other VoIP protocols too, which are either closed
ones or they cooperate with specific applications-telephone exchanges. For
example, the particularly popular Skype is a restrictive VoIP protocol by
itself, whereas the Inter-Asterisk Exchange [IAX] is being utilized by the
open software telephone exchange. Finally, the Skinny Call Protocol has been
developed by Cisco.
2. The Codecs
VoiP codecs have been created to serve the primary function
of optimizing voice over IP networks. Yet, there are other codecs specializing
in music-video transmission.
The quality as well as the features attributed to any VoIP applications or
devices affect to a great extent the general quality of Internet calls. Like
already mentioned at the beginning of this article, codecs take over our
voice compression-decompression-which, in the meanwhile, has been digitalized-,
that it occupies the least possible volume, while it is being transmitted
in the Internet. At the same time, they maintain voice quality at satisfactory
To make a VoIP call feasible, both parties have to not only adopt the same
communication protocol, but the same codec as well.
The main codec features are as follows:
- Sampling rate: 8 to 16 Khz are more than enough to achieve quality while
- Codec bit rate: In this case rating fluctuates between 5 and 64 kbps.
When referring to data flow, it is widely accepted that what is meant focuses
on the rate of
compression performed by the codec. A high rate means little or zero compression,
therefore higher sound quality. On the other side, low compression is responsible
for a big volume of packets and as a result, an increase in delays and
- Nominal bandwidth: It refers to the bandwidth needed for our connection
with the Internet or in a LAN, so that the codec is able to achieve maximum
performance. Apart from each codec’s data flow, the “load’ of
various protocols being in control of the flow of packets like the UDP, RTP,
and others, also
has to be counted in, in the total bandwidth required.
- Time required
to the creation of voice packets: The more time required to create voice
packets, the less network load is achieved. On the contrary,
mean that the codec will have to compress a rather large volume of sound
data, resulting thus in delaying data transmission as well as their reception.
acceptable compression times range between 10 and 40 ms.
MOST WIDESPREAD CODECS
G.711. It is the codec being used by digitalized fixed telephony networks.
It relies on the Pulse Code Modulation [PCM] and provides relatively
low compression. This means high quality when it comes to voice transmission;
yet, wide bandwidth
is required while connection is being attempted.
G.723.1. The high compression achieved by this particular codec relates
to a minimum bandwidth to achieve VoIP communication. However, the quality
accomplished reaches mediocrity, while delays in telephone conversation
increases. The fact that it has been designed to accommodate tele-conferences
communications carried out by plain telephone lines is more than obvious.
G.729. It is normally found in VoIP SIP applications. It provides double
compression than the one offered by the G.711, meaning that a rather
high voice quality
can be achieved, though the times required, to prepare coded voice packets
GSM. None other than the quite well known and qualitatively tested codec
applied to the GSM mobile telephony networks.
ILBC. Its full name is the portrait of its capacities, [Internet Low
Bitrate Codec]. It is provided at no cost, though some limitations are
by the Global IP Sound. It is designed to facilitate use in low speed
networks and incorporates special voice improvement techniques for lost
Speex. It is an open software codec with very good technical features
when compared to several “restrictive” as well as expensive
Skype. Last but not least, comes the codec referring to the most popular
VoIP application. It has been developed by the Global IP Sound and
into exclusive utilization by the Skype.
3. Common Problems Found With VoIP
Any selection of a VoIP application has to be based on the existing
Internet connection a user has been provided with. If, for example, the G.711
codec is to be used with a 56K connection, there will be a sort of communication
that is not to be performed. For such a connection the G.723.1 and G.729 codecs
are considered to be the ideal ones.
Jitter. Assuming that a codec is capable of providing voice packets every 30ms,
the person’s to be called telephone appliance or application will have
to receive these packets every 30ms. Unfortunately, the delays due to the Internet,
with the intervention of dozens of networks and routers, seem to be the cause
to the problem. Yet, it looks that buffers in both VoIP applications and appliances
are a partial solution to the problem.
Delay. Any delays while coded packets are being created by codecs are added
to the one caused by the Internet, adding thus to the appearance of problems.
instance, Internet delay while a VoIP communication is being performed must
range between 150 and 450ms and the codec delay to follow must not exceed 40ms
voice packets are being created.
Lost packets. While a data transfer in the Internet is carried out, loss of
some packets is expected up to a point and can be dealt with, with the help
techniques. In voice transmission though, the margins to the loss of packets
are not many, since the problems arising are to be immediately “heard” by
both parties involved. The selection of an ISP with a high quality network
and an efficient bandwidth may be the most effective measure of precaution
NAT/Firewall. Fear for crackers along with hackers has resulted in the addition
of firewalls to each PC as well as network, be them either big or small ones.
However, both firewalls and NATs are cause for trouble in VoIP connections.
All applications provide different ways to override these obstacles.
Echo. The really disturbing echo case [listening to our voice while the voice
of the person being called is being heard at the same time], is a common
occurrence with VoIP communication and sometimes with mobile telephony
calls. The echo
is a direct outcome of the delay in communication between the two parties.
techniques to handle the problem are being provided by all VoIP applications
and appliances as well, not to mention communication codecs themselves.
MORE INFORMATION ON VoIP
Voice over Internet Protocol is not just a single, unique protocol. It involves
a group of technologies-protocols, devices and applications, which allow
voice call compression, coding, transport and routing to IP networks, like,
for example, the LAN, theWLAN, theWAN, and naturally the Internet, by means
of overriding fixed public telephone networks. Of course, an internet voice
call may be commenced or end up to a fixed public service, while VoIP technologies
are likely to be set to operational status in some sections of a conventional
voice communication. In short, the cases setting VoIP to action are as follows:
1st VoIP case
VoIP technologies are being utilized by providers in the
telecommunications sector in this case, either in setting up their internal
network or in the so-called “last mile.” To facilitate their internal
substructure, numerous telecommunications services in the private sector have
selected to equip
themselves with cheaper TCP/IP-VoIP networks rather than developing a rather
extravagant transmission line network.
2nd VoIP case
New and profitable services offered by emergent telecommunications
VoIP providers are what this case comprises from. They are related to the relatively
cheap communication developed between VoIP subscribers throughout the world and
fixed as well as mobile networks via the Internet, overriding this way any fixed
public telephony services.
3rd VoIP case
VoIP user-VoIP user
It has to do with the birth of the Internet telephony, the two VoIP users
achieving peer- to- peer communication, through the Internet of course,
charged. Such a communication can be achieved through a computer, the
use of headphones and a microphone or any USB/DECT telephone appliance
4. The Key To Success
The information to follow includes all the details needed to a successful
Until the VoIP situation somehow gets stabilized,
VoIP novice users are in for a rough time as soon as they set themselves to
new world.” The dozens of VoIP applications, the five different communication
models (either open or closed ones), the at least five differing codecs as
well as the thousands of VoIP telecommunication providers founded all over
the world, providing various-occasionally differentiated-services at varying
costs and within ranging geographical limitations, are amongst the causes to
Things are getting rougher when the two parties engaging in VoIP, wish direct
communication with no providers involved, like for example the Windows Messenger,
the Skype or any other, since the firewall, the NAT, VPN-even differing operating
systems in each party-may generate a number of varying situations. Yet, all
turn out evidently well in case an intermediary VoIP access provider or a communications
application of some kind is in hand, as the user is guided in all stages from
opening their account and downloading the application to handling their account.
This way, of course, allows a sort of “binding” with the specific
VoIP service provider, whereas, if you wish to change it, a new application
is required along with a fresh account being opened, new adjustments, and so
FIRST STEP. We shall start with the computer and
the connection to the Internet, which, as shown by all indications, will have
to be an ADSL one at 256/128
kbps at least. Even though VoIP communication via a 56kbps dialup is attainable,
the limitation are many, both at an applications as well as facilities provided
level and at voice quality. The basic gear will have to include a microphone-headphones
set, the installation and proper function of which is required. In case there
is an ADSL connection [the faster the better], a specialized ADSL VoIP router
or Analog Terminal Adapter [ATA] can be installed. Thus, the connection of
the telephone sets already existing at home to achieve VoIP calls is facilitated
even if our computer is off.
SECOND STEP. You will have to make a list of your telecommunication needs.
VoIP telephony still has a lot of stability problems to come up against with,
due to the big number of the intermediary parties involved and to the varying
quality of the services offered. That means that for no reason whatsoever should
you abandon fixed telephony, at least for the next couple of years. To continue,
you must select a provider to your liking, on the grounds of its geographical
coverage and of course the cost of the services it has to offer. One thing
you should keep in mind is that the cost of international calls compared with
the ones made through fixed telephony networks is still high.
THIRD STEP. A prerequisite to step into the VoIP world is to own a credit card.
The aforesaid condition is a deterrent in itself, to a lot of people. However,
a credit card is necessary to pre-buy communication time that will be later
consumed. The only solution to avoiding credit cards is the employment of local
VoIP providers as well as alternative ways of payment like a bank deposit or
a deposit with the Western Union.
A cautious selection of the telecommunication VoIP equipment has to be made
should you decide that the PC-microphone-headphones combination
is not functional. As a matter of fact we have been conditioned to telephone
conversation rather than making use of microphones connected to our PC. The
devices will definitely have to support SIP and H.323 protocols. In case
you opt to use a “restrictive” VoIP provider, like the Skype for example,
it is of great importance that you check out whether the device you have bought
supports it. Analog Terminal Adapters [ATA] call for the purchase of the models
suggested by the provider you have subscribed with. You will have to check
though, whether they are “locked” for exclusive use with the specific
provider. Depending on the model, utilization of a great number of services
offered by different VoIP networks is allowed.
5. The Good, Old, Software
VoIP TOOLS AND DRAWBACKS
We have the computer, we are provided with a connection to the Net; what is
left is the VoIP to be made use of. Right at this time, the VoIP market is
being dominated by the Skype. Over 30% of the calls made all over the world,
are being carried out by the Skype! Other than the Skype there is a “coalition” of
providers based on the SIP protocol. The Skype is in itself a restricted protocol
and does not allow communication with other networks’ SIP users. On the
other hand, generally speaking, the SIP service subscribers are able-either
the easy or the hard way-to communicate with each other.
Unfortunately, a lack of common numbering and interconnection among VoIP
providers, make up what is called the Achille’s heel of the system in
general. Although communication from VoIP networks to fixed networks around
the world is now being carried out at characteristically easy and cheap levels,
reverse communication has just started making its first steps. An organized
attempt to integrate these two worlds is on the way [ENUM]. A specialized DNS
servers network will allow easy location of subscribers in both sides of voice
THE GOOD, OLD SOFTWARE
The bulk of the VoIP communications provided today are
carried out by an application of some kind [Softphone]. On the one side we
have Internet telephony applications, which have developed to telecommunication
[ie the Skype Out service provided by the Skype], whereas on the other side
there are VoIP telecommunication providers offering applications that suit
ie the X-Lite. Among the various VoIP groups, there are the specialized messengers
[MSN, ICQ], offering voice communication in addition to the written one.
Normally, each application supports its own communication protocol along with
the rest of the widely spread VoIP protocols. For instance, the X-Lite is based
on SIP, while the JaJah application is based on a restrictive protocol/codec,
supporting though, at the same time SIP, H.323, Skype, IAX2 calls-in all the
widely spread codecs, to be more specific. Likewise, similar services are being
provided by the Firefly application supported by the FreshTel, based on the
IAX protocol while it supports SIP as well.
As you may have well understood, voice communication through a kind of a Softphone
it is required that an activated computer as well as a microphone and headphones
To make our communications more functional, the producers of each application/service
propose specific telephone devices, capable of being connected to a PC either
through a USB or an Ethernet port. These appliances can be purchased either
in every VoIP provider website or in local markets where a variety of models
Before setting your self to the purchase of such a device, you will have
to check out with the provider you are about to connect that the aforesaid
with the services provided. Several major VoIP services/applications will
be presented bellow.
MSN Messenger (Messenger.msn.com) The improved Windows Messenger Edition
is considered as one of the best applications in the Internet voice communication
era. Its use is really simple and as a
result its application is accessible even to a beginner, while its performance
effective. The procedure the user is required to follow is simple: A communication
window with the called user opens and a click on the icon displaying a microphone
is requested. If there is no technical trouble of any sort, voice communication
is carried out both ways [full duplex]. While conversation is on, the user
is able to silence the microphone in the software itself, without having to
with the Windows volume mixer/control. The philosophy of a simple software
run expands to the ways it can be utilized with problematic connections. In
of lack in speed, the quality of voice communication drops automatically, so
that the bandwidth requirements of the connection are reduced. The user is
capable of no obvious selection in such a shift. Should a problem related to
connection arises, a user is simply informed that their connection is just
an unsuccessful one, without being informed on the measures they have to take
solve the problem.
ICQ (www.icq.com) It has been some years since they have been offering voice
communication, but they insist on making life hard to the user. To start with,
the desired ability
is not available with the application right after initial installment. Both
the users wishing to join in a voice communication have to download an under-sized
element to be added to the application [from the www.icq.com website for the
2003b version]. As soon as the aforesaid addition is installed an icon displaying
an acoustic telephone appears next to the name and the list of connection introduced
by each user. Voice communication develops in a separate window. In general
voice quality is not as high as the one provided by the MSN Messenger, yet
ICQ seems capable of effectively carrying out connections amongst users, which
cause for other direct communication applications to give up.
Despite of having recently made its appearance in the VoIP era, it has developed
into the dominating force in it, having obtained a share amounting over 30%!
From the beginning it developed as a high quality and minimum requirements
with regard to the connection speed voice application. As an application it
a simple program. It provides voice and-for the last few months-written communication,
thus poking its way in the Messenger’s field. It supports a list of contacts,
which has started being stored in the company servers, so that the service is
available to any users being facilitated by third parties’ computers.
Voice communication is achieved via double-clicking on the user name required
with while it is completed with pressing the button relative to the red icon.
Performance is what amazes anyone with this application. The sound is really
clear and time delay is practically non-existent. The application has a minimum
of requirements as far as bandwidth is concerned, which, if not met, may cause
some problems. For example, with a 56K connection, if the line gets loaded,
the sound will come up with cut offs and discontinuity. Finally, it has to
that the program seems to have particularly high requirements as far as processing
power is concerned, when compared to other applications in the field. The Skype
provides connection to fixed as well as mobile networks all over the world
at low costs, while a service through which a specific number for incoming
from any fixed or mobile network is set, has made its appearance. The service
is called Skypeln and as for now it is available in the United States, the
United Kingdom, Sweden, France, Finland, Denmark and Poland.
Free World Dialup started-and still works-as the undertaking of VoIP users
to facilitate free of charge voice calls and has met with great appeal. As
already mentioned, to enable a voice call under a SIP protocol the intervention
of a server-central service- is required, which undertakes the task of connecting
the users who are involved. While the communication in question is to be directly
carried out by the users, the server will take on to intervene to the facilitation
of the commencement of such communication. Most servers of the kind are provided
as part of a payable provider. Free World Dialup is the most widely known charge-free
VoIP provider. Up until recently it used to offer a pre-adjusted X-Lite version,
but in the last few months it has been proposing the pulver.communicator [yes,
there is a dot in-between], which provides cooperation with the Skype network-on
condition that it is installed in the same computer.
Another challenging service provided by the FWD is the ability catered to the
user to be registered in a telephone network of the United Kingdom, free of
charge. What follows is that the user may have at his disposal a telephone
under the British international Code. In other words, one might call the aforesaid
number assuming that the user the number belongs to is connected to the FWD
server, while the call will appear as a SIP in the application running in their
Of course, it should be stated right at this point, that dialing the above
phone number will cost more than calling any “ordinary” number
in the same country and the caller will have to be charged more, yet on the
other hand the
FWD user does not have to pay a penny.
Now, we will move on to a comparison, indicative of the VoIP service charges.
Starting off, we would like to make clear that any communication between
two parties carried out in the Internet is free of charge. For instance,
between a Skype subscriber A with a Skype subscriber B is not charged, while
communication between two network subscribers, ie VoIP SIP, under
Free World Dialup, is usually availed at no cost, as long as such connection
When the connection ends up in a fixed network though, ie any public or mobile
network around the world, it is charged. Value added tax services, like,
for example, the ability to facilitate incoming calls from fixed networks
provided at a cost. VoIP providers offer charge packets at a flat rate charge and some “free” moments
of conversation and of course the typical charge depending on use.