Call For Free.

Well, instead of reading the article, I last mentioned, I got diverted. This is what I tried doing, I set up a website www.iCallWorldFree.com. This was supposed to let people call for free and run based on their donations and income from advertising before calls. Imagine, I was so excited that I had even purchased the website. And then, I talked to one of my friends, to get his reaction on it. :( Though his reaction was positive and encouraging, I got some hints, which I thought were not so good. So, people, back to the article.

Here's a page from the website that I had launched. (I was not done yet)

Here's the plan:

Call almost anywhere in the world for FREE, that too from your phone. One time registration fee $2. Initial system will only support calls from USA to a list of countries. (India included). People outside USA can still call by using one of the many free applications like Yahoo Messenger to call a USA number and then anywhere in the world through http://www.iCallWorldFree.com. (Please review terms and conditions of any such service). Another alternative for people not in USA could be to order a softphone either locally or through us and we will give you a US number for just $2 (one time fee). This will allow you to receive calls too. You will need an internet connection (not computer) to setup a softphone.

Does that mean I can make long distance calls for FREE? Yes sir, absolutely free. No setup costs, No per minute charges, free and UNLIMITED. Only $2 activation fee for each incoming or outgoing number. Just call the access number we provide to dial the destination number and you are done. No pins, and no punching too many keys. So, what is the catch? Read on.

You make your own plan, DONATE as much as you like and talk as much as you like. You calculate how much this system can reduce your call charges by, and donate us that money for a better cause. A large portion of profit will be donated for the needful. Isn't that just great?A short advertisement may be played before connecting the call, for some users.

Donations will be made on sole discretion of the company to any organisation that the company chooses.

To honor and encourage donations, we will include a page showing how much each person/company donated, for those interested to be listed. And how much of it was donated.

Expected Launch Date: October 1st, 2010.

P.S. I will post here when we have started accepting donations, until then please feel free to leave comments and reviews.

Will there be a solution? I am sure, I will find one.

The total cost for setting up a system like this would involve local calls within incoming and outgoing country, therefore, it would not be possible to setup a FREE system with only the server setup costs (one time investment) but it is a continuous investment of the local calls which would be the incoming cost in source country and outgoing cost at destination country.
What do I do now?
Well, right now I am reading this article:
http://www.faqs.org/docs/Linux-HOWTO/VoIP-HOWTO.html#ss8.1

SIP Failed, now what?

I went through a couple of articles and now I can not recall exactly what made me believe that it is still possible with SIP to do all this.
Probably the following paragraph:


So your decision will be taken considering PSTN line costs. In fact what VoIP does is the convert this:

Home Telephone1 --- (PSTN) --- Home Telephone2
             PSTN great distance calling cost

into this:

Home Telephone1 --- (PSTN) --- PC1   +
      PC2 ---- (PSTN) --- Home Telephone2  =
      --------------------------------------
          2 PSTN short distance calling costs

To save money you need that:

2 PSTN short distance calling costs < PSTN great distance calling cost 

or this one:

A typical application is like that:

Home telephone1 -- (PSTN) -- PC1 -- (Internet) -- PC2 -- (PSTN) -- Home telephone2

Home Telephone1 make a calls to PC1 phone number (using PSTN line, I mean classic telephone line).
PC1 answer to it.
Home telephone1 must tell PC1 what gateway use (PC2 in this case) by giving the IP address (from DTMF keyboard) and/or what number call (in this case Home telephone2).
After that PC1 will start to make an H323 call to PC2, then it will pass Home telephone2 to PC2 to make it call it throught PSTN line.
Home telephone2 answers to call and communication between Home telephone1 and Home telephone2 begins.

SIP CONTD.

I forgot to mention in the last post, the cost of outgoing number is ($36 /Year).

Quotes from Wikipedia :

SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format. SIP clients typically use TCP or UDP on port numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints.


Network Elements:

A SIP user agent (UA) is a logical network end-point used to create or receive SIP messages and thereby manage a SIP session. A SIP UA can perform the role of a User Agent Client (UAC), which sends SIP requests, and the User Agent Server (UAS), which receives the requests and returns a SIP response. These roles of UAC and UAS only last for the duration of a SIP transaction.

A SIP phone is a SIP user agent that provides the traditional call functions of a telephone, such as dial, answer, reject, hold/unhold, and call transfer. SIP phones may be implemented by dedicated hardware controlled by the phone application directly or through an embedded operating system (hardware SIP phone) or as a softphone, a software application that is installed on a personal computer or a mobile device, e.g., a personal digital assistant (PDA) or cell phone with IP connectivity. As vendors increasingly implement SIP as a standard telephony platform, often driven by 4G efforts, the distinction between hardware-based and software-based SIP phones is being blurred and SIP elements are implemented in the basic firmware functions of many IP-capable devices. Examples are devices from Nokia and Research in Motion.

Each resource of a SIP network, such as a User Agent or a voicemail box, is identified by a Uniform Resource Identifier (URI), based on the general standard syntax^[9] also used in Web services and e-mail. A typical SIP URI is of the form: sip:username:password@host:port. The URI scheme used for SIP is sip:. If secure transmission is required, the scheme sips: is used and SIP messages must be transported over Transport Layer Security (TLS).

Last paragraph confirms that I will need a SIP device at the user end. Maybe the ATA was the UAC here. Now, the problem is how do I route the numbers to existing PSTN.

Lets start all over again.

SIP and ATA

I found a post, http://hubpages.com/hub/Build-Your-Own-Cheap-VOIP-Phone-Connection and will summarize what I read here. This guy is trying to create an incoming phone line for himself for the cheapest possible way through VOIP. ATA (Analog Transmission Adapters) is a box that converts analog to digital signals and vice versa. "Unlocked" ATA (available at voxilla store, techtoast; cost under $100) hooked with the router and the phone and configured correctly with the SIPPHONE vendor allows the vendor to route calls to the phone connected.
SIPPHONE vendors: SIP (Signal Initiation Protocol) is an application layer protocol that  serves as a link between a phone and the internet and is widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). 
More on SIP coming up...
I feel that to establish your own VOIP system, the first and only thing required will be the SIP server. This is the one that routes calls to the ATA and must be the one that routes calls to the PSTN (Public Switched Telephone Network).

Day 1 - How it all started

Just like most of the people trying to make money, I decided to learn about Google Ad sense today. During the sign-in process I realized that this was not something new for me, I had once abandoned Google Ad sense a couple of years ago because of my laziness (and there was $0.23 in my earnings). I had already visited success stories of the so called internet "entrepreneurs" and said to myself, "Darn! Had I not abandoned this process and continued maybe I could have got more until now".

And then I decided to start on the same thing again. But this time, I realized I had to create a blog, and that was the easiest way to make money online. "Blog??!!, I am not a writer", I thought. You know what, let's just give it a try.

Deciding the topic was the hardest thing I have done yet with this whole making money thing. And I feel, that choosing VOIP was the right decision, and I hope I am not too late like I was with Google Ad sense.

You know what, let's just save the other minute details until this story becomes a success story. :P