Having over the last two years acquired the appropriate licenses and agreements, Cameroon being the latest addition, Nvirocom has dedicated a significant amount of time, research and development for the North African Market.
Nvirocom’s goal is to install 25 new termination facilities in strategic sites to optimize services across the country by May 2012. As Nvirocom expands its operating services and broadening business relationship in Cameroon, we have become a fully integrated company established in Cameroon securing local employment for local people benefiting all concerned.
Nvirocom have and is currently in negotiations to secure 3g and 4g licenses with the Minister of Posts and Telecoms for the Cameroon Government also securing an ISP (Internet Service Provider) license. This will allow Nvirocom to provide a more efficient service to Cameroon and the rest of the African market becoming a market leader within the telecommunications sector at this time.
Nvirocom has studied and researched many destinations globally; our findings suggest that South America will probably be the next continent for expansion. The company will look to explore this avenue and develop infrastructure to support and install another 25 new sites within a 9 month period Subject to the requisite licensing being in place we intend to commence business in June 2012.
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The Supply of VoIP Termination through Nvirocom’s own secure servers. A service where demand has and is continuing to rise as one of the fastest growing industry in the world.
Immigration can be defined as the relocation in a country of a person or group of people native of another country. (More often motivated by a job search, and the perspective of a better life.) These movements of deportation are observed mostly in poor countries or countries at war as those of Africa, Asia, South America and the Eastern Europe.
According to a report of the UN, in 2005 191 million people, approximately 3% of the world population live outside their country of origin and according to the same report, this figure has increase by 15 million more immigrants for end of 2011 and expected to increase by another 50 million by 2015.
This resettlement of people always nourished the need to communicate between communities in the world and is set to continue and increase at a high rate.
Only just 7 to 8 years ago, it was already possible with the fixed line telephone to communicate overseas but remained all the same limited because of technical problems bound to that period in third world countries. Today, the choice is vast in communication, 3G 4G and Internet trends and capabilities has paved the way to make easier access to inaccessible areas, This means demand is booming and it makes telecoms the fastest growing sector to date.
Boosted by this perpetual need to exchange information through communication manifested by people, Nvirocom bridges the gap for user demand and ultimately offering considerable support on old networks of communication with innovative technologies. Nvirocom’s system is ultimately offering high quality communication routes to fit any existing telecoms network but capable of lowering prices to meet user demands.
In a market where telecommunications operators are everywhere, Nvirocom makes the difference.
The seminar on costs and tariffs organized by the BDT in May 2011 in Botswana with the participation of Cameroon focused on the contribution rate of mobile termination and roaming mobile in Cameroon. It has stimulated the opening of the telecommunications sector Cameroon, which should now enjoy a period of rapid expansion and development of the infrastructure.
Nvirocom Cameroon is leading the way in Cameroon as one of the leading suppliers of telecommunications services. Nvirocom’s expertise and focus includes the termination of traffic in which it enjoys a significant market share with Government approval. The need for termination services has been promoted in Cameroon since the seminar due to the lack of transportation means.
Nvirocom Cameroon has created the necessary infrastructure and used its unique technology to implement the termination of traffic across Cameroon. Its strong relationships with established giants in the telecommunications sector have also allowed it to take a significant market share. Nvirocom’s unique technology and the size of its market share allows Cameroonian users to be connected with foreign countries at a low cost.
The creation of Nvirocom Cameroon is the culmination of the vision of Mr.Simon Williams and all his team. The rapid expansion and development of Nvirocom is expected to continue in 2012.
According to statistics the penetration rate for broadband internet is 4%. It is an interesting niche market additional for Nvirocom in Cameroon and one that it has taken control of. The increased use of and demand for the internet in Cameroon is reflective of young Cameroonians strong desire to develop computer skills as the communications sector expands and becomes more accessible. Nvirocom also has the option of offering computers and PCs in Cameroon at very low cost and also potentially exempt from customs duty. Other opportunities are open to Nvirocom in Cameroon, particularly in the telecommunications sector and it is anticipated that the use of fibre optics and its unique technology will be a catalyst for new products and services during a period of further expansion as the market develops. The future is certainly bright for both Nvirocom and for the telecommunications market in Cameroon.
Follow http://wp.me/1GQwP or www.nvirocom.com for more information on Nvirocom.
Nvirocom (UK) Ltd
The Nvirocom group of companies deliver global telecommunication services and gateways with a particular focus and presence in Africa. Since Nvirocom (UK) Ltd commenced business in March 2011 the company has rapidly expanded its network and gateways. It is an exciting period for Nvirocom as it is poised to make a significant impact within the telecoms industry. It will achieve the group aim of becoming a major player within its chosen sector of the telecoms market ahead of schedule.
The achievements and growth of Nvirocom to date are impressive. Its notable milestones to date are:
The securing of licenses to commence telecom operations in Cameroon;
Three fully operational telecom gateways in Africa; A further seven telecom gateways under construction and on schedule to become operational by the end of the first quarter of 2012; and, it has secured private equity finance through an E.I.S Inland Revenue Approved Investment Scheme with lucrative returns for investors.
As a result of the licences already obtained and the securing of funding Nvirocom has a unique opportunity to expand rapidly in the coming months. New applications for licences in other jurisdictions are currently being submitted as planned growth continues. Watch this space and stay in touch with Nvirocom (UK) Ltd. If you are interested in becoming an E.I.S Approved Investor please contact Nvirocom’s Public Relations Officer at email@example.com for further information and details of how to apply.
The soft switch contains a database of users and phone numbers. If it doesn’t have the information it needs, it hands off the request downstream to other soft switches until it finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests. It sends back all the relevant information to the softphone or IP phone, allowing the exchange of data between the two endpoints.
Soft switches work in tandem with network devices to make VoIP possible. For all these devices to work together, they must communicate in the same way. This communication is one of the most important aspects that will have to be refined for VoIP to take off.
As we’ve seen, on each end of a VoIP call we can have any combination of an analogue, soft or IP phone as acting as a user interface, ATAs or client software working with a codec to handle the digital-to-analogue conversion, and soft switches mapping the calls. How do you get all of these completely different pieces of hardware and software to communicate efficiently to pull all of this off? The answer is protocols.
There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP. They also include specifications for audio codecs. The most widely used protocol is H.323, a standard created by the International Telecommunication Union (ITU). H.323 is a comprehensive and very complex protocol that was originally designed for video conferencing. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP. Actually a suite of protocols, H.323 incorporates many individual protocols that have been developed for specific applications.
H.323 Protocol Suite
As you can see, H.323 is a large collection of protocols and specifications. That’s what allows it to be used for so many applications. The problem with H.323 is that it’s not specifically tailored to VoIP.
An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP). SIP is a more streamlined protocol, developed specifically for VoIP applications. Smaller and more efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the process. Media Gateway Control Protocol(MGCP) is a third commonly used VoIP protocol that focuses on endpoint control. MGCP is geared toward features like call waiting. You can learn more about the architecture of these protocols at Protocols.com: Voice Over IP.
One of the challenges facing the worldwide use of VoIP is that these three protocols are not always compatible. VoIP calls going between several networks may run into a snag if they hit conflicting protocols. Since VoIP is a relatively new technology, this compatibility issue will continue to be a problem until a governing body creates a standard universal protocol for VoIP.
VoIP is a vast improvement over the current phone system in efficiency, cost and flexibility. Like any emerging technology, VoIP has some challenges to overcome, but it’s clear that developers will keep refining this technology until it eventually replaces the current phone system.
A codec, which stands for coder-decoder, converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay. It’s the essence of VoIP.
Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance, a G.711 codec samples the audio at 64,000 times a second. It converts each tiny sample into digitized data and compresses it for transmission. When the 64,000 samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear, it sounds like one continuous second of audio signal. There are different sampling rates in VoIP depending on the codec being used:
• 64,000 times per second
• 32,000 times per second
• 8,000 times per second
A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used codec in VoIP.
Codecs use advanced algorithms to help sample, sort, compress and packetize audio data. The CS-ACELP algorithm (CS-ACELP = conjugate-structure algebraic-code-excited linear prediction) is one of the most prevalent algorithms in VoIP. CS-ACELP organizes and streamlines the available bandwidth. Annex B is an aspect of CS-ACELP that creates the transmission rule, which basically states “if no one is talking, don’t send any data.” The efficiency created by this rule is one of the greatest ways in which packet switching is superior to circuit switching. It’s Annex B in the CS-ACELP algorithm that’s responsible for that aspect of the VoIP call.
The codec works with the algorithm to convert and sort everything out, but it’s not any good without knowing where to send the data. In VoIP, that task is handled by soft switches.
For example 208 555 2323
The switches use “208” to route the phone call to the area code’s region. The “555” prefix sends the call to a local exchange, and the network routes the call using the last four digits, which are associated with a specific location. Based on that system, no matter where you’re in the world, the number combination “(208) 555” always puts you in the same local exchange, which has a switch that knows which phone is associated with “2323.”
The challenge with VoIP is that IP-based networks don’t read phone numbers based on NANP. They look for IP addresses, which look like this:
IP addresses correspond to a particular device on the network like a computer, a router, a switch, a gateway or a telephone. However, IP addresses are not always static. They’re assigned by a DHCP server on the network and change with each new connection. VoIP’s challenge is translating NANP phone numbers to IP addresses and then finding out the current IP address of the requested number. This mapping process is handled by a central call processor running a soft switch.
The central call processor is hardware that runs a specialised database/mapping program called a soft switch. Think of the user and the phone or computer as one package — man and machine. That package is called the endpoint. The soft switch connects endpoints.
Soft switches know:
• Where the network’s endpoint is
• What phone number is associated with that endpoint
• The endpoint’s current IP address
VoIP technology uses the Internet’s packet-switching capabilities to provide a phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 20-minute phone call we talked about earlier consumed 20 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional system. And this example doesn’t even factor in the use of data compression, which further reduces the size of each call.
If you and your friend both have conection through a VoIP provider. You both have your analogue phones hooked up to the service-provided ATAs. Let’s take another look at that typical telephone call, but this time using VoIP over a packet-switched network:
1. You pick up the receiver, which sends a signal to the ATA.
2. The ATA receives the signal and sends a dial tone. This lets you know that you have a connection to the Internet.
3. You dial the phone number of the party you wish to talk to. The tones are converted by the ATA into digital data and temporarily stored.
4. The phone number data is sent in the form of a request to your VoIP Company’s call processor. The call processor checks it to ensure that it’s in a valid format.
5. The call processor determines to whom to map the phone number. In mapping, the phone number is translated to an IP address (more on this later). The soft switch connects the two devices on either end of the call. On the other end, a signal is sent to your friend’s ATA, telling it to ask the connected phone to ring.
6. Once your friend picks up the phone, a session is established between your computer and your friend’s computer. This means that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.
7. You talk for a period of time. During the conversation, your system and your friend’s system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analogue audio signal that you hear. Your ATA also keeps the circuit open between itself and your analogue phone while it forwards packets to and from the IP host at the other end.
8. You finish talking and hang up the receiver.
9. When you hang up, the circuit is closed between your phone and the ATA.
10. The ATA sends a signal to the soft switch connecting the call, terminating the session.
One of the most compelling advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers do.
It will still be at least a decade before communications companies can make the full switch over to VoIP. As with all emerging technologies, there are certain hurdles that have to be overcome.
A packet-switched phone network is a better alternative to circuit switching. While you’re talking, the other party is listening, which means that only half of the connection is in use at any given time. Based on that, we can surmise that we could cut the file in half, down to about 4.7 MB, for efficiency. Plus, a significant amount of the time in most conversations is dead air — for seconds at a time, neither party is talking. If we could remove these silent intervals, the file would be even smaller. Then, instead of sending a continuous stream of bytes (both silent and noisy), what if we sent just the packets of noisy bytes when you created them?
Data networks do not use circuit switching. Your Internet connection would be a lot slower if it maintained a constant connection to the Web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching.
While circuit switching keeps the connection open and constant, packet switching opens a brief connection — just long enough to send a small chunk of data, called a packet, from one system to another. It works like this:
• The sending computer chops data into small packets, with an address on each one telling the network devices where to send them.
• Inside of each packet is a payload. The payload is a piece of the e-mail, a music file or whatever type of file is being transmitted inside the packet.
• The sending computer sends the packet to a nearby router and forgets about it. The nearby router sends the packet to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router, and so on.
• When the receiving computer finally gets the packets (which may have all taken completely different paths to get there), it uses instructions contained within the packets to reassemble the data into its original state.
Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well.