Thinking of creating your own VoIP application?

Voice-Over-Internet Protocol, a.k.a VoIP, was labeled as a disruptive technology during its infancy. Disruptive in a sense that it will have a massive effect on the current Telecom industry. There were even rumors that it will totally replace the traditional Public Switched Telephone Network (PSTN), or what Cisco-fellows commonly call as POTS for Plain-Old Telephone Service. My belief is that VoIP was and should be developed to work with PSTN, not make it obsolete. Using PSTN and VoIP technology together is even better.

Contrary to public notion, VoIP is not that new to Telecom providers or carriers. In fact, a lot of them has been using VoIP for years now. Carriers mostly use VoIP in transferring international calls and trunking with other major carriers around the globe. They still use Signaling System 7 or SS7 (SS7 is the standard signaling method/protocol of PSTN, which is digital as well) as the primary method of signaling for the majority of their calls, but a consumer doesn't know that sometimes, to cut costs and connect with other carriers faster, a specific international call made by a subscriber is routed using VoIP to connect to another country. Then once the call hits the terminating carrier, the call is then transformed and/or encoded back to analog, then routed using SS7 to the destination telephone number. This method has been saving carriers around the world thousands of dollars.

The beauty of VoIP communication on a technology level is you do not need a dedicated physical or virtual circuit, always reserved for a communication to take place. With IP technology, the voice traffic can use the current available bandwidth of a circuit, then make it open and available for other applications or traffic once the communication has ended. This has been made possible because of Time-Division Multiplexing (TDM) technology.

The current state of VoIP is amazing. Despite not yet being fully-mature in my opinion, hundreds of start-up companies are now offering top-notch carrier-grade, easy to setup VoIP technology. I for one uses Skype as my major tool of communication with my family in the Philippines. I am hoping that someday Skype will decide to make public their proprietary P2P Signaling Protocol that makes their Skype Video Chat light-years ahead with the current competition.

If you want to test this, go and make a video chat session using the latest version of Yahoo Messenger. Observe and compare the delay and quality of the audio and video. Now launch Skype, and you will immediately notice the difference. The WiFi latency inside our house in Manila averages between 190 to 250ms when pinging a US gateway, but this doesn't seem to have a big effect with the quality of my Skype Video Chat session. I live here in the Bay Area, using Comcast Cable Internet with basic 1024K up and 325K download. My laptops are hooked on wireless so I can blog even while I watch my roommates outside skating on our homemade half-pipe at our backyard. Skype is utilizing an excellent proprietary protocol for their signaling resulting in an excellent performance of their VoIP product.

If you are code geek, someone who can easily develop their own application using various programming languages, you will find it relatively easy to develop your own VoIP application.

So what makes up a VoIP application?

A conceptual model has been developed by various leading companies and developers in the VoIP industry. The Internet Engineering Task Force (IETF) is one of the major contributors for the success of VoIP because of the excellent Standardization and Drafts members contributed.

Remember the Open System Interconnect Model a.k.a. OSI Model? Traditional Data Network guys use this conceptual model as a guide in developing and troubleshooting applications and processes that are made for transferring data from one network to another, regardless of its geographical location. A computer network's primary function is to transfer data in form of packets or radio signals from point A to point B. Everything else is optional and for maintenance purposes.

A VoIP application in a nutshell is composed of 3 Layers:

Layer 3: Application
Layer 2: Call Control
Layer 1: Packet Infrastructure

Layer 1, the Packet Infrastructure in a nutshell would map to the Transport Layer, (Layer 4) of the OSI Model. On this layer, you define if your application will use TCP, UDP, RTP over UDP or a combination of the identified standard communication protocols for your VoIP application to establish communication channels to carry the signal and actual voice payload from source to receiver.

Layer 2, Call Control, or Signaling, is the layer where you define how your VoIP application will be able to establish a connection from source to destination. This is where you define if your VoIP application will be using the signaling standards such as Session Initiation Protocol (SIP), H.323, MeGaCo and others to name a few.

Layer 3, Application, defines the actual capabilities of your VoIP application. Features such as Call Waiting, 3-way conferencing, Hold Music, Voicemail and Click-to-Call functionalities are defined here. This is where you make your application unique and stand-out among other VoIP products.

Stay tuned and I will explain and breakdown the 3 Core Layers of VoIP in details on my next post. I will include sample applications and opensource source codes that developers out there can use as a guide in discovering the inner-works of a VoIP application. Just be sure to credit me if you were able to create a wonderful Skype-like proprietary VoIP application after reading this series of blog :-)

Reach for the sky!
Ron