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Appeared on: Wednesday, October 19, 2005
VoIP Essentials

1. Roaming the VoIP Inner Side
VoIP represents 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 is 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 [RTCP]

It 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 handling first 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 a mobile 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 Asterisk 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 we have already mentioned at the beginning of this article, codecs take over our voice compression-decompression-which, in the meanwhile, has been digitalized-, so 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 levels.

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:


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 of voice accomplished reaches mediocrity, while delays in telephone conversation increases. The fact that it has been designed to accommodate tele-conferences and telephone 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 are longer.

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 being imposed by the Global IP Sound. It is designed to facilitate use in low speed networks and incorporates special voice improvement techniques for lost packets.

Speex. It is an open software codec with very good technical features when compared to several “restrictive” as well as expensive codecs.

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 naturally falls into exclusive utilization by the Skype.

3. Common Problems Found With VoIP
Bandwidth. 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. For 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 while 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 of certain 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 to be taken.

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. Various techniques to handle the problem are being provided by all VoIP applications and appliances as well, not to mention communication codecs themselves.


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
VoIP user-PSTN
PSTN-VoIP user

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, without being charged. Such a communication can be achieved through a computer, the use of headphones and a microphone or any USB/DECT telephone appliance been required.

4. The Key To Success
The information to follow includes all the details needed to a successful VoIP access.

Until the VoIP situation somehow gets stabilized, VoIP novice users are in for a rough time as soon as they set themselves to accessing this “brave 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 confusion.

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 on.

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.

FOURTH STEP. 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

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 communication networks.


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 providers [ie the Skype Out service provided by the Skype], whereas on the other side there are VoIP telecommunication providers offering applications that suit them best, 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 be used.

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 can be found.

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 device is compatible 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 is really 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 intervene 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 case 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 a successful 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 to 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 terms, 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 are cause for other direct communication applications to give up.

Skype (www.skype.com) 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 is just 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 to communicate 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 be noted 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 calls 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.

FWD (www.freeworlddialup.com) 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 we have 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 call number 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 computer. 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.

VoIP charges

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, communication 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 is provided.

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 are normally 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.

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