In designing a VOIP network we will take different approaches to wholesale and retail models. The center of any VOIP network is the billing system. For both you will need a stable, robust bililng server that allows for backups and easy rate changes. For a wholesale network you will need to add in your profit margin and track the minutes going to your various gateways and providers. For a retail network you will need to do the same while also offering robust billing features to your customers, such as balance enquires, updating of personal information, adding additional DID's etc.

A retail system will rely on a gateway to both play IVR and assist in voicemail and other options. Under a wholesale buisness model it is possible to run a successful business relying solely on your billing server by reselling other's routes. The greatest profit however is in deploying your own gateways. By installing a network of gateways that have PRI, E1, T1, or POTS lines terminated to them you will be offering long distance service at the going market price, while yourself paying for local calls.

After we design a retail network with a robust billing system, configured IVR and other calilng options, and have succesfully tested calls we are only half way done. The next step is creating a custom website for the service. While it is easy in 2006 to get a website designed, the trick is to get one where customers may access their account. This is accomplished by interacting with the billing server database. This function must be 100% accurate and presented in a proffesional manner. Finally the system is ready to be marketed. We are able to assist here as well.

For a wholesale business it is a bit simplier. Many succesfull gateway owners operate with just a few customers. As you are at the top of the food chain you want just a few customers to have to deal with. They can then in turn resell the route. It is advisable to have a few large customers as well as access to the larger trading sites in the industry. When you start your route you want maximum minutes from the first day. We can assist in ensuring that your minutes have a buyer before you go live.

The two major competing standards for VoIP are the IETF standard SIP and the ITU standard H.323. Initially H.323 was the most popular protocol, though its popularity has decreased in the "local loop" due to its poor traversal of NAT and firewalls. For this reason as domestic VoIP services have been developed, SIP has been far more widely adopted. However in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones, and the vast majority of callers have little or no idea that their POTS calls are being terminated over VoIP. So really SIP is a useful tool for the "local loop" and H.323 is like the "fiber backbone". With the most recent changes introduced for H.323, however, it is now possible for H.323 devices to easily and consistently traverse NAT and firewall devices, opening up the possibility that H.323 may again be looked upon more favorably in cases where such devices encumbered its use previously.

Where VoIP travels through multiple providers' Soft Switches the concept of Full Media Proxy and signalling proxy are important. In H.323 the data is made up of 3 streams of data: 1) H.225.0 Call Signalling 2) H.245 3) Media. So if you are in London, your provider is in Australia, and you wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600 ms) and packet loss will be high. However in signalling proxy mode where only the signalling flows through the provider the delay will be reduced to a more user friendly 120-150 ms. These proxy concepts could lead the way to true global providers.

One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically to send a G.723.1 5.6 kbit/s compressed audio path will require 18 kbit/s of bandwidth based on standard sampling rates. The difference between the 5.6 kbit/s and 18 kbit/s is packet headers. There are a number of bandwidth optimisation techniques used such as silence suppression and header compression. This can typically save 35% on bandwidth used. But the really interesting technology comes from VoIP off shoots such as TDMoIP which take advantage of the concept of bundling conversations that are heading to the same destination and wrapping them up inside the same packets. These can offer near toll quality audio in a 6-7 kbit/s data stream.