Many Faces of VoIP Applications

Based on scenario in Malaysia

VoIP Solutions come with Free Download from the Internet

The earliest products worked exclusively via software, and users are both ends of the connection needed a computer running the software, an Internet connection, a sound card, and a USB microphone. The connection was often only half duplex, making the conversational experience more like taking on a two-way radio than on a telephone.

There are still many software-based VoIP products in use and these tend to be less expensive and some maybe even free installation. These programmes are such as Skype by Kazaa, MSN Chat and Yahoo Chat, to name a few. The programmes can be easily downloaded from the internet.

As the saying goes ‘there is no free lunch’, these programmes will come with adwares, which normally will open a back door for the spy wares to lurk into our computer. These spy wares will check our daily usage of our computer, the websites that we normally surf, our e-mails and our data information. If we do not handle these spy wares properly, they will later create loses in memory to our PC thus slowing down our PC, creating lots of pop up’s during our surfing the internet, and sometimes do cost a breakdown to our PC.  


These softwares are made user-friendly especially with a PC, and they are meant for home and soho usage. Since it is very much depends on the PC, the PC need to be ‘on’ every time the call is made. The voice quality is good due to the bandwidth usage of 64 kbps download and uploads speed. It will go up to about 90 to 100 kbps combining with the IP headers. But voice quality cannot be assured if there is a burst in the bandwidth usage. Any large download of files could cause a delay and a drop in the voice quality. There is no control of bandwidth.

Calls cannot be transferred but can make three parties conference call. With a certain prepaid amount and some credit top-up, users could call to an outside or any ordinary phone all over the world. It is still cheaper that the normal telecom rate but this is very subjective because there is no constructive billing from the software party to the users.

In short, this is still the cheapest but a more conservative way to have VoIP call. It is more on a point-to-point (p2p) connections and it is not very flexible for corporate organizations. The target market for these VoIP softwares is more for the home users, to chat with friends from all around the world. Well, after all, it is free!

Third Party VoIP Discounted Calls Service Providers (ASP)

ASP means Applications Service Provider, which is also called the third party VoIP Discounteded Calls Service Provider, in Malaysia. These vendors provide discounted calls rates up to 88% (mostly on international IDD calls and outsation STD calls depending on the location) compared to the Telekom Malaysia rate, to the corporate organizations. Their competitive edges are better voice quality and a cheaper call rates.  

It is very beneficial for organizations with high volume of international IDD and outstation STD calls. It also comes with credit limit and credit facility of up to a month, where monthly itemized bills with call details are issued to these organizations. Dialers need to be installed if it is attached to the PBX. There is no other additional hardware or software required.

layout for ASP

Not a very wise choice if calls are to handphone like Maxis or Digi. The rate of these Discounted Calls Service Providers to handphone is at the range of RM 0.22 to RM 0.27 per minute. And these costs is not included with the last mile calls charges from Telekom, which is RM 0.08 for the first two minutes, follow by RM 0.04 for the subsequent minute. Compared to Maxis and Digi, the cost is at RM 0.15 per minute (condition applied) flat per minute respectively, where no last mile calls incurred.

Although the billing from these Discounted Calls Service Providers is postpaid, which is every month-end, the itemized billing is also very subjective due to

  • Billing does not state the Telecom last mile charges.
  • If carefully calculate, there might not be any saving for all the handphone calls.
  • Unsuccessful calls are not shown in the billing from these providers. For all unsuccessful calls which is due to insufficient ports at the third party VoIP Service Providers gateway server, there is definitely still a last mile charges from the Telekom, because the Telekom line is used (using the dialer) to call to the Service Providers gateway server. Example, for every ten (10) unsuccessful calls, the charges from Telekom will be RM 0.08 x 10 = RM 0.80.
  • To avoid unsuccessful calls happen to the customers, there are some of these Discounted Calls Service Providers offer a back up to the PSTN, if the calls could not connect to their gateway server. If these happen often, the customers might lose a lot. This explains why sometimes we do receive calls to our handphone with no number or stated ‘private number’ (calls got through the gateway server), and sometimes we receive calls from the same place, but the number is shown in our hand phone display (calls did not go through the gateway server but fall back to the normal dialing format which is through the PSTN/ISDN). 

VoIP through SIP (Session Initiation Protocol) Server (Voip using ATA device).

SIP is less complicated protocol compared to H.323 which was designed specifically for VoIP. SIP gateway providers compete with the above Discounted Calls Service Providers by abolishing the last mile Telecom changes because it uses a SIP dialer (ATA) to call to the SIP server, through the internet broadband. In addition to that, SIP Providers offer ‘free’ inter-branch calls through the internet broadband if the customers’ branches are also installed with the SIP dialer (ATA). Normally a SIP dialer or a SIP phone cost around RM 399 to RM 499 each and comes with 2 FXS ports to be to be attached to two single-line phones. SIP market is huge to consumers, home and small business network. It is not flexible and is very difficult to apply to a larger organization.

SIP providers do not earn any revenue from customers’ ‘inter-branch’ calls because it is ‘free’ through the broadband. Normally, customers are required to pay a certain amount like RM 25.00 monthly per site (which is installed with the SIP dialer). SIP providers are more interested to have customers to call ‘outside the branches’ like international IDD calls where they will earn some revenue. There will not be any monthly bill from these SIP providers. Prepaid is practice here where customers need to deposit some payment first before any outside calls is made. This might be also due to the SIP server is not own by the SIP providers where it might be own by their business partners and some SIP servers are located at overseas.


Sip providers will always say that their solutions take a simpler view of things and easy to install. Just plug the WAN port to the DSL modem and plug two single-line phones to the phone ports. Well, that is actually a short-coming, because communication systems are complex. H 323 is more complex, because it meets the requirements in order to build carrier class solutions. Even the global data backbone rides on H.323 (using mostly Cisco equipments) and is being used for millions of minutes per month in carrier networks.


Have you ever ask your SIP provider where does the voice packets routed to? They route to the SIP server. Who own the SIP server? It is owned by the third party, mostly overseas’ counterparts. Well, when every voice packets routed to the SIP server, definitely it is going to be a bottle-neck, especially where there is congestion in the flow of packets. Then what will happen to the voice quality? Indirectly, your organization is sharing the similar network to the SIP server with others.

Then again, who is controlling the SIP server? When all the packets routed to the SIP server, have your organization thinks of privacy and security? Can the SIP server cater-made for your organization’s requirements? Who is controlling your network?

Also, most of the insufficient bandwidth issues are actually created by the internal bandwidth usage in your own organization. Have the SIP providers take into consideration about your insufficient of bandwidth which will also affect your voice? Surfing the internet, video streaming, downloading movies and music files from Kazaa, Bit-torrent, edonkey are also very bandwidth consuming. And when the voice quality is bad, these SIP providers will blame on the broadband provider.   

How does the SIP gadget be implemented to your existing data network? Like leased line? Can it does call routing without using the DISA feature of the PBX? (because the voice volume will drop through the DISA function)

What about the dialing plan? Can SIP gadget do ‘hop-on’, using the similar dialing plan with the normal telco way? They can’t, because they want to earn your money, if you want to call outside your network through the SIP server. Your calls will be routed through a multiple service provider before reaching the destinated number.  

Who is earning the money for the routing of the calls? Well, you can say that the money is flowing out of the country, because we are paying using ‘prepaid’ to the management who handles the overseas’ SIP server.      

Anyway, SIP services, if managed well should be very useful to simple users and small organizations. It creates a very low investment for these groups to utilize VoIP applications. Total cost for two branches might be even less than RM 2,000. But, do you have room for error? This is because SIP gadgets (dialers) might also turn out to be a toy to your organization’s needs. 

VoIP System Integrators and IP Telephony Vendors.

VoIP System Integrators (SI) will assist organizations to an open infrastructure for telephony and communications through the interoperability with value in a multi vendor environment. The proper deployment will give your organization a feel of a virtual private network through real-time telephony, conferencing and secure enterprise-class softphone services, which can be accessed through a single user identity.

Have your organization ever thought of linking all your inter-branches using IP network to your existing PBX or key phone and make dialing as transparent as intercom, even calling outside the country? Have you thought of owning the whole network where the voice packets routed from your office to the destination without going into another third party server? Have you ever thought of making‘outbound’ calls (hop-on calls) and paying Telecom local rate charges without going through any Third Party Discounted Call Service Providers or SIP providers and pay at a more expensive rate?

Have your organization ever thought of an IP integrated Call Centre, even at a global standard or even integrating with Toll Free number and therefore save a lot, not only in terms of monetary but also in the human resources? Have you ever thought about the level of frustration in the customers, if they were asked to call another number at another location due to the insufficiency of transferring the calls? Have you ever thought of maintaining the customers loyalty and satisfaction where you can ensure that every call they made are answered by the person they need to talk too? Have you ever imagine that calls to your office can be transferred to your PDA hand-phone or any notebook attached to a soft phone? 

If your organization has considered the above solutions, then the product that you are looking for is an enterprise level IP media gateway to do the job for you. A high end IP media gateway with embedded server will be able to integrated most of your current communication method (platform) into IP platform. Having your own IP network is what most organizations are thinking of, as in the coming future your organization IP platform can also be integrated to the global IP platform. This is where in the near future, all communications and routing will be based on principles of IP.

A proper setup and implementation with the IP gateway will bring a lot of saving to your organization. The setup of IP gateway, will allow you to call directly, without going through any third party servers, thus abolishing all the charges from these third party vendors.

Choosing the right vendor whom is familiar with data and voice applications, together with the right proposal and at the right timing will eventually assist your organization to implement the whole transformation at the least cost. An experienced vendor will definitely take into consideration of your internal bandwidth usage, the IP packets classifications, the existing data network, firewall and security, and the IP gateway equipment provided.