Voice Over IP (VoIP)
Using the corporate IP-based network and even the Internet to carry non-data traffic is nothing new. Like many new IT initiatives, VoIP to many people will be a handset that plugs into a USB port allowing you to chat to your friend in Hong Kong on the cheap- even for free. Well things have moved on, the big names from data have moved into telephony, the telephone folk have moved into data - there is a whole new market and its big business.

John Blake of British Telecom

Voice over IP is not "if" or "when" - it's now. When many businesses initially look at VoIP, they often focus primarily on inexpensive international calls as a reason to upgrade their legacy analogue systems. Though saving money on long distance calls may be the reason many businesses begin exploring VoIP, it is certainly not where they will realise the highest Return on Investment. Businesses today are finding that the right VoIP system will deliver savings in ongoing system management, future expansion costs, and on wire to the desktop. With a cutting edge VoIP system, the administrator simply enters the relevant information into the management interface and the user is automatically provisioned with an extension, voicemail, presence and instant messaging. Expanding a legacy phone system can be expensive and quite disruptive, often requiring the addition of entirely new switching and control units. With IP based systems, businesses can add more users without the need to add additional hardware. Our systems are secure, scaleable and provide cost efficiencies unimaginable with traditional analogue systems.

The main challenge of providing a packetized voice service is quality. When considering the IP environment this is mainly provided by Bandwidth (network integrity) and the ability of the network management to provide acceptable voice Quality of Service. This is not currently available over the public Internet (internet telephony), mainly due to the lack of regulations in place to handle bandwidth. However VoIP is possible over managed IP networks where QOS (Quality of Service) solutions and SLA 's (Service level agreements) exist between providers.

The potential of this market is vast with many vendors and solution providers offering a confusing array of products and services that tend to be linked directly to their own (often proprietary) equipment. React's VoIP solutions are based upon standards based technology, enabling open systems communications over both public and private IP networks. Our sole objective is to increase customer satisfaction through improved productivity and reduced operational costs in the managed IP network - independent of manufacturer or VoIP protocol.

 Investment
(network infrastructure, software and cables)

 Maintenance
(cost for running service teams including training)

 Operation
(connection fees)

In migrating to converged VoIP Technology, Investment costs can be kept to a minimal as much of the required infrastructure already exists. Maintenance costs can be reduced, e.g. technical issues resolved by one IT- department, and Operation costs minimised, e.g. calls within the data network are at no additional charge.

Voice / Data integration and advanced services
VoIP implementations can be managed remotely e.g. managed moves, adds and changes from virtually anywhere, with Phone connections initiated with a mouse clicks, or speech be recorded to files or attached to emails. These features are additional to those provided previously via the PBX.

Scalability
With IP technologies, new segments and equipment are easily introduced, enabling the network to be scaled quickly and therefore cost efficiently, allowing for minimal network (and user) disruption.

SIP / H.323
SIP is used to signal interactive communication sessions. It is increasingly being used to provide Voice over IP, Presence and Instant Messaging in Next Generation Networks, and being mandated for many new applications, including 3G telephony.

The SIP protocol is used to control interactive communication sessions. The uses for such sessions include voice, video, chat and instant messaging, as well as interactive games and virtual reality. SIP can be used to control Internet multimedia conferences, Internet telephone calls and multimedia distribution, in both the core and the periphery of the communications network.

SIP invitations are used to create sessions and carry session descriptions that allow participants to agree on a set of compatible media types. In this way, SIP is not restricted to any particular media type, and can therefore handle the expanding range of media technologies. SIP enables user mobility through a mechanism that allows requests to be proxied or redirected to the user's current location. Users can register their current location with their home server.

Members in a SIP session can communicate using multicast or unicast relations, or a combination of these. In addition, SIP is independent of the lower-layer transport protocol, which allows it to take advantage of new transport protocols. Software implementing the basic SIP protocol can be extended with additional capabilities and is actively being exploited for many media applications.

H.323 is an International Telecommunications Union (ITU) standard that provides specification for computers, equipment, and services for multimedia communication over networks that do not provide a guaranteed quality of service. H.323 computers and equipment can carry real-time video, audio, and data, or any combination of these elements.

Users can connect with other people over the Internet and use varying products that support H.323, just as people using different makes and models of telephones can communicate over Public Switched Telephone Network (PSTN) lines. H.323 defines how audio and video information is formatted and packaged for transmission over the network. Standard audio and video codecs encode and decode input/output from audio and video sources for communication between nodes.