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.
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.
Existing phone systems are driven by a very reliable but somewhat inefficient method for connecting calls called circuit switching.
Circuit switching is a very basic concept that has been used by telephone networks for wee over 100 years. When a call is made between two parties, the connection is maintained for the duration of the call. Because you’re connecting two points in both directions, the connection is called a circuit. This is the foundation of the Public Switched Telephone Network (PSTN).
This how a typical telephone call works:
1. You pick up the receiver and listen for a dial tone. This lets you know that you have a connection to the local office of your telephone carrier.
2. You dial the number of the party you wish to talk to.
3. The call is routed through the switch at your local exchange to the party you are calling.
4. A connection is made between your telephone and the other party’s line using several interconnected switches along the way.
5. The phone you are calling rings, and someone answers.
6. The circuit opens.
7. You talk for a period of time and then hang up.
8. Hanging up, causes the circuit to close, freeing your line and all the lines in between.
Let’s say you talk for 20 minutes. During this time, the circuit is continuously open between the two phones. In the early phone system, up until 1960 or so, every call had to have a dedicated wire stretching from one end of the call to the other for the duration of the call. So if you were in London and you wanted to call Manchester, the switches between London and Manchester would connect pieces of copper wire all the way across the United Kingdom. You would use all those pieces of wire just for your call for the full 20 minutes. You paid a lot for the call, because you actually owned a 220-mile-long copper wire for 20 minutes.
Telephone conversations over today’s traditional phone network are somewhat more efficient and they cost a lot less. Your voice is digitized and your voice along with thousands of others can be combined onto a single fiber optic cable for much of the journey (there’s still a dedicated piece of copper wire going into your home though). These calls are transmitted at a fixed rate of 64 kilobits per second (Kbps) in each direction, for a total transmission rate of 128 Kbps. Since there are 8 kilobits (Kb) in a kilobyte (KB), this translates to a transmission of 16 KB each second the circuit is open, and 960 KB every minute it’s open. In a 20-minute conversation, the total transmission is 19,200 KB, which is roughly equal to 20 megabytes If you look at a typical phone conversation, much of this transmitted data is wasted.
Bandwidth requirements are going through the roof with increasing user demands. Trends today are fast growing towards the new technologies and the api market which means more bandwidth is required to share video, download music and a array of office tools for those on the move. The stats are building up and so are the demands. This is why Nvirocom is in a great position with this explosive market to help meet just some of these demands. The question is how far will we push it with our technologies before the demand cannot be met. Lets hope never but we need more companies like Nvirocom to push and place networks to be sure of this.
Today 11th july 2011 our moroccan switch has gone live. We have had great success with its output and can offer 80% ASR 4 min ACD. This in the telecoms world are amazing stats and we intend to keep up the level of service throughout the whole network we are building.
Nvircom Ltd is a telecommunications company born to create Eco friendly networks and to lower user costs. We are keen to find the latest technologies which are designed to lower costs, higher productivity and put them into practice so we invite anyone to talk to us about new ideas and what advantages they may bring. We also like to keep a keen eye on what’s happening within the environment and as with our network and servers we are always looking to find alternative energy to do our bit to lowering our Co2 footprint.
VoIP phone users can make calls from anywhere there’s a broadband connection.
Chances are good you’re already making VoIP calls any time you place a long-distance call. Phone companies use VoIP to streamline their networks. By routing thousands of phone calls via a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they’re using for the long haul. Once the call is received by a gateway on the other side of the call, it’s decompressed, reassembled and routed to a local circuit switch.
Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced with packet-switching technology. IP telephony just makes sense, in terms of both economics and infrastructure requirements. More and more businesses are installing VoIP systems with Nvirocom, and the technology will continue to grow in popularity as it makes its way into our homes. Perhaps the biggest draws to VoIP for the home users that are making the switch are price and flexibility.
With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATAs broadcast their info over the Internet, they can be administered by the provider anywhere there’s a connection. So business travelers can take their phones or ATAs with them on trips and always have access to their home phone. Another alternative is the softphone. A softphone is client software that loads the VoIP service onto your desktop or laptop. May soft-phones have an interface on your screen that looks like a traditional telephone. As long as you have a headset/microphone, you can place calls from your laptop anywhere in the broadband-connected world.
We (nvirocom) are offering minute-rate plans structured like cell phone bills for as little as £20 per month. On the higher end, some offer unlimited plans for as little as £50 a month which is a great option for heavy users. With the elimination of unregulated charges and the suite of free features that are included with these plans, it can be quite a savings.
Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service plan.
• Caller ID
• Call waiting
• Call transfer
• Repeat dial
• Return call
• Three-way calling
There are also advanced call-filtering options available from some carriers. These features use caller ID information to allow you make a choice about how calls from a particular number are handled.
• Forward the call to a particular number
• Send the call directly to voice mail
• Give the caller a busy signal
• Play a “not-in-service” message
• Send the caller to a funny rejection hotline
With many VoIP services, you can also check voice mail via the Web or attach messages to an e-mail that is sent to your computer or handheld. Not all VoIP services offer all of the features above. Prices and services vary, so if you’re interested, it’s best to do a little shopping.