How Much Bandwidth is Required for VoIP Phones?

A long-standing question for potential VoIP (Voice over Internet Protocol) consumers is “How much bandwidth does a VoIP phone require to make quality telephone calls?”

First of all, Bandwidth is defined as the ability to transfer data (such as a VoIP telephone call) from one point to another in a fixed amount of time. The higher the bandwidth speed you have, the more data you can send over your Broadband Internet connection.

There are two types of bandwidth at your location: upload bandwidth and download bandwidth. The Upload Bandwidth is the amount of data you can send to the Internet and download bandwidth is the amount of data you can receive from the Internet. The more Internet bandwidth you have from your ISP (Internet Service Provider) the better.

In most cases, the normal VoIP telephone call will use up 90 Kbps (kilobits per second). If you have a Broadband Internet service provider that doesn’t offer much bandwidth then most VoIP providers give you the option to lower the VoIP voice quality by lowering the bandwidth used for VoIP calls to 60 Kbps or, to really conserve your bandwidth, 30 Kbps. Most people can't tell the difference between the three settings. We suggest you use the high sound quality setting (90 Kpbs in most cases), if bandwidth is not an issue. High VoIP voice quality is generally the default setting but if you are running into a situation where your bandwidth is limited then you can adjust your VoIP bandwidth to one of the lower settings. Some consumers with 128 Kbps upload connections can receive less VoIP service quality due to a poor quality ISP (Internet Service Provider). By selecting a lower quality VoIP bandwidth setting, this problem can be avoided.

If you plan on using a VoIP service provider, should you get a DSL or a Cable Internet access provider? In general, DSL upload bandwidth starts at 128k where as Cable Internet upload bandwidth starts at around 600k. Cable Internet is a little bit more expensive, but it is also about 4-5 times faster than residential DSL and a bit friendlier to a VoIP telephone call. Having said that, both DSL and cable modem high-speed services provide sufficient broadband Internet access bandwidth to support any of the top VoIP service providers. If you are experiencing low Broadband Internet Service provider bandwidth, we suggest you try Packet8 VoIP. Packet8 VoIP boasts an advanced compression technology in which each active voice line uses approximately only 23Kbps of total data throughput, upstream and downstream.

The amount of bandwidth that a VoIP provider requires to make a quality telephone call is only one thing to consider when choosing a VoIP service provider. In fact, there are many things to consider when choosing a VoIP provider. An educated consumer generally results in a satisfied consumer.

Things to Consider When Selecting a VoIP Provider

The following are very important factors to consider when you are selecting a Voice over IP provider. Educate yourself and be informed before you choose.



Monthly costs: A VoIP provider can save you up to 75% on your telephone/long distance expenses. There are many Voice over IP providers out there so it will benefit you from shopping around. Unlimited calling packages can range from $19.95/month to as high as $54.95/month. Usually the lower priced providers have more customers and are able to offer the service at a lower price due to a lower overhead per subscriber



VoIP Product Features: Not all VoIP providers are created equal. Voice over IP offers a great value to the consumers because of the drastically reduced long distance costs as well as inexpensive local phone service with lots of enhanced features. Some providers offer more features than others. Features like Call Waiting, 3 Way Calling, etc. are usually included in the VoIP monthly cost where as the traditional phone companies will charge up to and above $5/month per feature. When shopping for a VoIP provider, be sure to compare VoIP providers by features as well as by monthly price.



Keeping Your Number: Some providers allow you to transfer (port) your current phone number to the VoIP service and some providers do not. It is not recommended to switch your home number to the VoIP service immediately. It is recommended that you try out the service and see if you are satisfied before you request that your current number be switched. Keep in mind that if you have DSL service, you must retain a phone number with the service provider of the DSL because the DSL service is provided over that telephone line. If you want to get rid of your current phone company all together, then we suggest you use a Cable Internet Service Provider.



911 Service: Most of the Voice over IP carriers offer E911 service, but not all. Be sure to check if the Voice over IP provider offers E911 because it is not a given. If the VoIP provider does not offer E911, then we suggest that you either have a cell phone or traditional landline to use in case of an emergency. (Note: It is also important to point out that if you take your VoIP phone when traveling, E911 has no way of knowing where you are when you call 911 if you are away from the registered address.)



International Calling: If you make a lot of international calls, you will want to do a lot of research on International Rates as they vary by provider. There are a few carriers that offer unlimited calling to certain countries.



Money Back Guarantee: Since VoIP is a relatively new product; most all VoIP providers will offer a free money back guarantee. Be sure to check with each provider as we have seen the money back guarantees range from a 14-day to a 30-day money back guarantee. (Note: Be sure to keep the original packaging that your equipment came in just in case you need to send it back)



This is only a short list. In fact, there are many things to consider when choosing a VoIP provider. An educated consumer generally results in a satisfied consumer.

VoIP CONCLUDING REMARKS

Implementation of VoIP is growing rapidly. There will also be further convergence of media, data, voice and video. In order to achieve these further levels of convergence, interoperability and continued work on standards will be critical. SIP is emerging as the major protocol to underpin IP-based advanced services.

Another key driver will be the development of wireless VoIP, which will provide students and staff with different ways to access learning environments while integrated services will open up the possibilities of ubiquitous computing. In future, learning may take place in a shared virtual world of social computing, similar to an online, multiplayer game, and VoIP will be one of the components of such an environment.

VoIP gateways have become the world's de facto standard in building voice and video over IP products and services. These gateways provide real-time voice and video IP communications. Many large SMEs are already looking to VoIP service providers for their needs. Many developing countries are also looking to VoIP to provide services to remote areas that are currently not reachable through traditional means.

Today, hundreds of thousands of people around the world use commercially deployed IP-centric products and services for next generation networks. However, the focus now is not just getting the VoIP service running but in delivering secure and high quality services.

VoIP in Thailand and South East Asia

There are two Telecom Providers in Thailand. TOT Corporation (formerly, the Telephone Organization of Thailand) to serve a Domestic voice communications and CAT Corporation (formerly, the Communication Authority of Thailand) serves an International voice communications and Data communications. TOT and CAT are both operators and regulators. They give concessions based on the BTO scheme to winning bidders on selected services. The concessions could be of the same kind due to overlapping roles and authority.

QoS of VoIP in Thailand. No QoS enforced both in terms of latency and accessibility. TOT VoIP domestic service claims to have delay comparable to that of mobile phones. CAT VoIP international service still faces problems about accessibility and latency.

Future Directions (Thailand)

The establishment of the National Telecommunications Committee (NTC) as an independent regulatory body. The Telecommunications Act is in the approval process from the Parliament. Long-term policy goal is set for liberalization.

VoIP in Other SEA Countries

Malaysia, the Philippines, and Singapore has VoIP licenses. Cambodia and Laos are available but still illegal. Cambodian military police have detained 20 foreigners on an alleged IDD bypass scam using an unlicensed IDD gateway. For Vietnam, the Posts and Telematics Ministry has suspended the sales of FPT Telecom’s Internet phone card service. Allow only outbound Internet-based calls from computers to computers and from computers to telephones while prohibiting inbound Internet phone calls.

QoS of VoIP in SEA. Only Singapore has clear QoS results on System Accessibility, Service Activation and Number of complaints. Philippines ask for QoS but does not enforce. Thailand has informal user surveys.

Future of VoIP in SEA

Move toward enterprise VoIP and Private leased-line infrastructure. IP telephony gateway over conventional PABX. More consolidation.

The Benefits of VoIP

In basic terms, it can lower the phone bills, deliver a wider range of unique services and improve employee productivity.

But while cost savings are keys, enhanced functionality and mobility are also huge factors in VoIP’s growing appeal. VoIP simply does things that are extremely expensive, or even impossible, with traditional phone networks. For example, many users enjoy the convenience of having their voicemail system notify them by text-message or e-mail whenever a new message arrives.

Best of all, most VoIP providers offer flat-priced bundles for both high-speed data and voice. One fixed monthly cost provides a fast, reliable Internet connection, as well as unlimited local and long-distance calling. While the initial investment in specialized VoIP equipment could involve significant up-front cost, that equipment generally pays for itself by reducing voice and data service costs and eliminating long-distance calling costs, within the first 12-18 months or even sooner.

Cost Savings

· Calls can be made between multiple offices for free using inexpensive equipment and an existing Internet connection

· Unlimited voicemail boxes

· No need for separate cabling since both data and voice are sent over the same network

Mobility

· Users can take their phone to another office or anywhere around the world and still receive calls at the normal number with any standard Internet connection

Functionality

· Easily integrates with contact list in contact databases, including Microsoft Outlook. Simply click on a contact and the phone will start dialing.

· Incoming calls can automatically activate pertinent information for that caller, which can enhance customer service

* Users can check and configure voicemail through an Internet web portal

“The Internet has forever changed the way that we view images and receive information, now it’s transforming the way we talk to one another. It’s an amazing technology that has wide ranging implications for today’s cost- and customer-conscious businesses,” states Jason.

Peer-to-peer networking (e.g. Napster, Kazaa, Skype)

The type of VoIP implementation most well known to consumers and students uses client-based, peer-to-peer (P2P) technologies such as Napster and Skype (Jones, 2005). P2P systems are far more decentralized than other networking or computer systems, with little or no reliance on the idea of clients being served by a central server. Clients (individual computers or telephony end points) act as nodes and are as likely to be ‘providing’ resources to other nodes (its peers) as to be consuming them from other nodes.

It is fairly well known within higher and further education that such client-based P2P applications can cause problems on IP networks because they make it much harder to manage the use of bandwidth and to exercise some level of control over the use of the network. There are particular problems with applications such as Kazaa and Napster, which are used to download music, films and games and the challenge is growing with the further development of P2P applications such as BitTorrent and Groove Networks’ Virtual Office, and the increasing popularity of instant messaging (IM) which is also a P2P application (see section 3.2 for more on IM). The use of Skype within college networks has raised similar concerns.

Many institutions see Skype as a good thing as it enables overseas students to keep in touch with their families at low cost and, as a consumer technology, has encouraged renewed interest in the potential for new developments in distance learning. However, because of the way it works, Skype is also considered by many to pose potential security threats, as well as possibly creating a major overhead on the network over which it is running.

Skype works by forming an ad hoc decentralized network of ordinary nodes and super nodes. The only centralized service is the Skype login server that stores user login and password details: the handling of voice calls is undertaken entirely by the decentralized network of nodes and super nodes. In order to make a telephone call, ordinary nodes must connect to an available super node (Baset and Schulzrinne, 2004). In this arrangement ‘any node with a public IP address having sufficient CPU, memory and network bandwidth is a candidate to become a super node’ and a Skype client “cannot prevent itself from becoming a super node” (Baset and Schulzrinne, 2004, pp.1 and 2).

In addition to the concerns over bandwidth usage it is worth remarking that Skype differs from some of the other VoIP services in that it is based on the company’s own, proprietary protocols, rather than the standard SIP protocol and one implication of this is that Skype users can only connect with other Skype users (Jones, 2005). Skype is not the only IP telephony system and that 'Alternatives that are standards-based may prove easier to manage and provide a more predictable service'. For example, VoIP systems such as the Gizmo Project use the SIP protocol and can connect effortlessly to multiple VoIP networks and SIP-based PBX systems.

A further consideration with regard to Skype is that it is considered by many to pose a potential security threat because of the way it works. Skype traffic is encrypted and uses a random combination of IP addresses and ports. This means it is hard to detect Skype calls because they run through their own, encrypted ‘tunnel’ over a network and there is concern that this is a possible way in which viruses or other problems could be introduced into a network, without being easily detected (Blackwell, 2005).

It is possible to block unauthorized Skype traffic. Blocking specific types of traffic over an IP network is usually done by blocking ports or denying access to specific IP addresses. But Skype traffic, because of the way it travels over the network and in particular its use of random combinations of IP addresses and ports, causes problems for traditional port blocking filters.

Skype traffic can be identified, and therefore blocked, by investigating the headers of every IP packet crossing the network. The challenge is to do this quickly, so other network services are not affected. VoIP blocking is often a function added to existing network or security management software, such as Narus’s IP Platform, Verso Technologies’ NetSpective 2.0, and SonicWALL’s enterprise appliances. Other systems able to help manage and control IP networks include Ellacoya’s IP Service Control System, Sandvine Broadband Network Management, and software from P-Cube, now owned by Cisco.

A proxy appliance, widely used to apply controls to Web traffic, can also be used to block specified unwanted traffic, including voice calls, if necessary. Deep Packet Inspection is another approach, developed from firewall technology that can also assist in intrusion detection and prevention.

List of some common VoIP softwares

· Skype

· GoogleTalk

· Windows Live Messenger

· Yahoo! Messenger

Technology, standards and challenges

As already mentioned, the traditional PSTN telephone infrastructure has been built up over the last one hundred years or so and has developed into a highly reliable voice communications system. In contrast, VoIP is a relatively new technology with a fledgling architecture that is built on inherently less reliable data networks. This means that there are therefore justifiable concerns around the extent to which it is deployed. The purpose of this section is to examine the underlying protocols and technologies used in VoIP and to discuss the potential challenges inherent in their deployment.

How VoIP is used, its background and history

Most people are aware of VoIP through the Skype consumer telephone service which has gained large-scale public recognition recently, particularly since its purchase by Ebay. However, VoIP has not suddenly appeared in the last few years as an opportunity afforded by the World Wide Web. Skype is only one particular implementation of VoIP and its related technologies and it is important to understand that VoIP has an important technological history, intertwined with the telecommunications industry in general, in order to appreciate the complexities of VoIP technologies and applications.

The idea of voice over IP has been discussed since the 1970s but it was the mid-1990s before commercial products became available with the introduction, in 1995, by Israeli company Vocaltec, of the first commercial system (Varshney et al., 2002). These early VoIP systems were designed to connect one PC to another and required each PC to have a sound card, speakers, microphone, modem and VoIP software. The software encoded and compressed the voice signal, converting it into IP packets that could be transmitted over the Internet. With this approach, both users used headsets, plugged into their PCs. The calls could only be made between PCs and could not connect to the PSTN network.

In parallel, from the 1970s onwards, traditional telecommunications carrier companies were developing new systems that introduced IP-enabling software for traditional telephony equipment. Human speech is an analogue wave signal and historically, voice telephone calls had been made over networks using analogue circuits which provided a temporary end-to-end connection, through the network, for each call (Sherburne and Fitzgerald, 2004). This is known as circuit switched, and builds on the original phone network of local telephone exchanges, in which wires between households were literally connected together for the duration of the call by a telephone operator. The companies that provided these services were often public agencies, usually part of a country’s post office service, and such networks became known as Post Office Telephone Systems (POTS), sometimes also referred to (post privatization) as Plain Old Telephone Systems. The Public Switched Telephone Network was the name given to the overall network created by telephone companies.

Between the 1950s and the 1990s these analogue systems were replaced by digital networks and telephone exchanges which made use of high-speed leased lines (known as T1 lines) and modern digital computer technology in the telephone exchanges, and digital signaling protocols such as ISDN between exchanges. However, these newer systems still relied on the circuit switch concept for end-to-end connection so for the consumer, things remained analogue since, by and large, the connection between the local exchange and the household remained a simple copper wire.

In the 1990s, with the Internet and Web boom, telephone equipment manufacturers and telecoms companies also began to make increasing use of the idea of transmitting digital information between exchanges through IP-based packets. This was in part driven by the lower costs associated with transmitting voice calls in this manner, as bandwidth use is more efficient. From the mid-1990s onwards, telephone equipment manufacturers added IP capabilities to their existing PBX telephony switches and, more recently, software has been developed to enable users to plug a VoIP adaptor into their traditional telephone. In this way, VoIP calls can start and end on the PSTN, but are then routed, via a software gateway, over the Internet.

This history means that VoIP is operating in a heterogeneous environment that extends way beyond the Internet. Voice calls need to have the potential to be carried over a variety of different networks including local networks, PBXs, PSTN and the Internet. Advances in VoIP technology mean PC telephony software is available from many software developers. Gateway servers with voice-processing cards are also available, to act as an interface between the Internet and the PSTN, enabling users to make calls either from their PCs, or from an IP phone, into the traditional telephone networks. Calls can also be made using IP handsets, which look similar to traditional phones, but which are plugged into an IP-based network rather than into the traditional telephony network, and have more features and capabilities than traditional telephones. The result is that there are now a number of ways in which VoIP can be implemented:

· PC to PC. Both the caller and recipient use headsets plugged into their PC.

· PC to PSTN. Only the caller uses a headset. The recipient receives the call in the

traditional way.

· PSTN to PSTN. The caller uses an IP adaptor on their traditional telephone and

The call is received on a traditional phone. But the call travels over an IP network.

· IP phone to PSTN. The caller uses an IP phone, and the call transfers from the

IP network to the telephone network via a gateway.

· IP phone to IP phone. The call travels over an end-to-end IP network.

It should be noted that there is confusion amongst communications professionals and industry commentators as to the use of terms like “VoIP”, “Internet Telephony” and “IP Telephony'. In this report we shall use the term VoIP to refer to the set of technologies that allow voice to be transported over an IP infrastructure (in effect, an IP-enabled PSTN) and the term IP telephony (IPT) to refer to VoIP technologies that also incorporate and build on the more advanced functionality provided by the old PBX systems.

What you need to make a VoIP cal

To make a VoIP call, the consumer user requires VoIP software and a broadband connection to the Internet. The software will handle the call routing to make sure the call reaches the intended destination as well as providing the codec. The software can be installed on a variety of hardware devices including traditional telephone handsets (using an adaptor that plugs into the telephone) or a PC or wireless device such as a Personal Digital Assistant (PDA). This use of software-enhanced end-user devices is one of the key distinguishing features of VoIP. Whereas the traditional telephone system contains its 'intelligence' within the network, VoIP makes use of the Internet model of intelligence at the edge of the network. This is often known as the end-to-end principle.

In order to make a call, an account with a VoIP service provider is also required. Different types of VoIP service are available, including services from traditional telephone carriers such as BT, and from specialized VoIP providers such as US firm Vonage and Luxembourg-based Skype. Some VoIP providers provide support only for PC-to-PC calls, while others provide the ability to make and receive calls from IP-enabled devices to users on the PSTN and on mobile networks.

To use VoIP, all you need is:

· High speed internet service (like cable or DSL)

· A credit card account w/ a US or Puerto Rico shipping address (no PO boxes)

· A touch-tone phone

Your connection needs a minimum upstream and downstream capacity of 90kb/s. If you're not sure if you have this, you can contact your service provider to find out.

Of course, these are just the minimum requirements. Many customers benefit from having an expandable cordless phone system. This will allow you to place a phone is any room you desire without running wires, etc.

How VoIP works

The Basics

The basic process involved in a VoIP call is as follows:

1. Conversion of the caller’s analogue voice signal into a digital format

2. Compression and translation of the digital signal into discrete Internet Protocol packets

3. Transmission of the packets over the Internet or other IP-based network

4. Reverse translation of packets into an analogue voice signal for the call recipient.

The digitization and transmission of the analogue voice as a stream of packets is carried out over a digital data network that can carry data packets using IP and other, related Internet related protocols (see section 2.1). This network may be an organization’s internal LAN, a leased network, the PSTN or the open Internet (Gradwell, 2006). The compression process is carried out by a codec, a voice-encoding algorithm, which allows the call to be transmitted over the IP network within the network's available bandwidth.

What is VoIP?

1An introduction to VoIP

VoIP is a set of technologies that enable voice calls to be carried over the Internet (or other networks designed for data), rather than the traditional telephone landline system — the Public Switched Telephone Network, or PSTN.

The term VoIP was coined by the VoIP Forum which was set up in May 1996 as an industry group concerned with promoting and developing product interoperability and a high quality of service for Internet telephony products. Initially, one of the main drivers in developing VoIP was the potential to cut the cost of telephone calls. Traditional voice calls, running over the PSTN, are made using circuit switching, where a dedicated circuit or channel is set up between two points before the users talk to one another — just like old-fashioned operators, plugging in the wires to connect two callers. The advantage of this is that once the circuit is set up, the call quality is very good, because it is running over a dedicated line. But this type of switching is expensive because the network needs a great deal of (mostly under-used) capacity.

The development of VoIP represents a major change in telecommunications. VoIP uses IP protocols, originally designed for the Internet, to break voice calls up into digital ‘packets’. In order for a call to take place the separate packets travel over an IP network and are reassembled at the far end. The breakthrough was in being able to transmit voice calls, which are much more sensitive to any time delays or problems on the network, in the same way as data.

Whereas calls over the PSTN are metered, so the user pays for the amount of time taken by their call, Internet usage is not metered. The user pays a set fee for their Internet service and their VoIP service and can then use the Internet to get free phone calls to other users on the same VoIP service, or pay a small fee to call users on other VoIP services or on the PSTN.

Packetized voice also enables much more efficient use of the network because bandwidth is only used when something is actually being transmitted. Also, the network can handle connections from many applications and many users at the same time, unlike the dedicated circuit-switch approach. This greater efficiency is one of the main reasons that all major carriers, such as BT with its 21CN (21^st Century Network) project, are changing their own networks so that they are IP-enabled.

VoIP's BACKGROUND

A. VoIP, Internet Telephony, Voice-over-the-Internet: What are they?

The terms Voice-over-Internet Protocol (“VoIP”), IP telephony, Internet telephony, and Voice-over-the-Internet (“VoN”) are given different meanings by different commentators and in fact have no universally agreed-upon meaning. There are, however, distinctions to be kept in mind, for IP can be used in various ways for the transmission of voice.

· VoIP is a generic term that refers to all types of voice communication using Internet protocol (IP) technology instead of traditional circuit switched technology. This includes use of packet technologies by telecommunications companies to carry voice at the core of their networks in ways that are not controlled by and not apparent to end users.

· VoN, also called Internet telephony, on the other hand is a service that end users decide to use -- it is a specialized form of VoIP in which a regular voice telephone call is transmitted via the public Internet, thus bypassing all or part of the public switched telephone network (PSTN). Internet telephony can occur between computers (computer-to-computer), between a computer and a phone (computer-to-phone), and between phones (phone-to-phone).

B. Transmission Of Voice Using IP Networks: How Does It Work?

Here is how a VoIP transmission is completed:

Step 1: Because all transmissions must be digital, the caller’s voice is digitized. This can be done by the telephone company (which is how carriers use IP in their networks), by an Internet service provider (ISP), or by a PC on your desk.

Step 2: Next using complex algorithms the digital voice is compressed and then separated into packets; and using the Internet protocol, the packets are addressed and sent across the network to be reassembled in the proper order at the destination. Again, this reassembly can be done by a carrier, and ISP, or by one’s PC.

Step 3: During transmission on the Internet, packets may be lost or delayed, or errors may damage the packets. Conventional error correction techniques would request retransmission of unusable or lost packets, but if the transmission is a real-time voice communication that technique obviously would not work, so sophisticated error detection and correction systems are used to create sound to fill in the gaps. (This process stores a portion of the incoming speaker’s voice, and uses a complex algorithm to “guess” the contents of the missing packets and create new sound information to enhance the communication.)

Step 4: After the packets are transmitted and arrive at the destination, the transmission is assembled and decompressed to restore the data to an approximation of the original form.

As this explanation suggests, technology that works fine for sending data may be less than perfect for voice transmissions. The technology is improving, but still the quality of a voice transmission using packet technology is inferior to a circuit-switched connection, and that difference in quality would normally be obvious to any listener. As IP technology improves, the quality advantage for voice communication enjoyed by the circuit-switched will decrease, but most experts see parity in quality as still a distant prospect.

C. Advantages of IP for Voice

Telecommunications carriers around the world have already introduced IP into their networks because it provides economic benefits over traditional telecommunications networks.

Greater Efficiency: The conventional circuit-switched technology of the PSTN requires a circuit between the telephone company’s switch and the customer’s premise to be open and occupied for the entire duration of a call, regardless of the amount of information transmitted. In contrast, on IP networks, all content -- whether voice, text, video, computer programs, or numerous other forms of information -- travels through the network in packets that are directed to their destination by diverse routes, sharing the same facilities most efficiently.

Lower Cost: IP systems will offer a more economical means for providing communication connections. Also -- and this is one of the sources of concern on the part of incumbent voice long distance carriers -- Internet technology makes available to anyone with a personal computer and modem the ability to bypass the long distance PSTN.

Higher Reliability: In some respects, IP networks also offer the potential for higher reliability than the circuit-switched network because IP networks automatically re-route packets around problems such as malfunctioning routers or damaged lines. Also, IP networks do not rely on a separate signaling network, which is vulnerable to outages.

Supporting Innovation: IP is a nonproprietary standard agreed on by hardware and software developers, and is free to be used by anyone. This open architecture allows entrepreneurial firms to develop new hardware and software that can seamlessly fit into the network. In contrast, the circuit switched network operates as a closed system, thus making it more difficult for innovative developers to build and implement new applications.

Telecom carriers using IP for their internal networks can reap these benefits. However, individual users seeking to use VoIP over their PCs encounter other limitations. Specifically, IP technologies currently lack a guaranteed quality of service. The ordinary telephone network (if properly installed and maintained) is designed to offer end users a very high quality of service for real-time communications. The Internet Protocol was not designed for voice; instead, it is based on a "best efforts" principle, which means that some packets are "lost" and have to be resent, introducing time delays that are inconvenient at best for voice communications.

Despite quality of service concerns and the policy issues that need to be resolved (discussed more fully in the next section), there are several arguments why carriers stand to benefit from VoIP even in end-to-end applications:

· Faced with an uncertain landscape and increased competition, incumbents must retain customers. By offering VoIP, in and of itself, carrier's can retain customers and increase traffic.

· Moreover, introduction of IP allows carriers to offer integrated services (voice, text, audio, and video) over a single connection, thereby further enhancing value to their customers and contributing to profits.

· Especially given the possibility of long distance savings, VoIP can boost consumer demand for local telephone service.

· Carriers stand to realize substantial cost savings as the IP switching equipment becomes less expensive.

· IP helps spur innovation and development. Infrastructure development on IP can take far less time and cost much less compared to the enormous costs of building out and maintaining a state-of-the-art PSTN network.

VoIP INTRODUCTION

Technologies that use the Internet and Internet protocol (“IP”) networks to deliver voice communications have the potential to reduce costs, support innovation, and improve access to communications services within developing countries and around the world.

However, these new technologies pose challenges for regulators, because they do not fit neatly within the regulatory model of the recent past, which in many countries has traditionally treated voice and data services differently. Especially in end-to-end applications, the introduction of voice over IP poses issues regarding: collection of universal service funds and obligations to provide universal service; payment of access or other settlement charges among local and long distance carriers and international carriers; quality of service; and the impact on the revenue of still-existing monopolies for long distance and international voice service.

Nevertheless, there seems to be a growing recognition in national and international bodies that these issues need to be resolved in ways that permit IP technologies to be used for voice communications. On balance, removing regulatory barriers to the use of IP technologies for voice can (a) advance the goal of affordable service and (b) support the deployment of the broadband networks and services that represent the future of communications in the era of digital convergence.

"VoIP" Voice Over Internet Protocal

VoIP is a set of technologies that enable voice calls to be carried over the Internet (or other networks designed for data), rather than the traditional telephone landline system — the Public Switched Telephone Network, or PSTN. The potential for very low-cost or free voice calls is driving the use of the technology but in the long-term, VoIP is more significant than just free phone calls - it represents a major change in telecommunications. The fact that VoIP transmits voice as digitized packets over the Internet means that it has the potential to converge with other digital technologies, which in turn will result in new services and applications becoming available.

However, the adoption of VoIP is not without complications. The traditional PSTN telephone infrastructure has been built up over the last one hundred years or so and has developed into a robust voice communications system that provides reliability figures of nearly 100%. In contrast, VoIP is a relatively new technology with a fledgling architecture that is built on inherently less reliable data networks. This means that there are therefore justifiable concerns around the extent to which it is deployed.

VoIP has developed considerably in recent years and is gaining widespread public recognition and adoption through consumer solutions such as Skype and BT’s strategy of moving to an IP-based network. In the long-term, VoIP is likely to impact on some of the bigger picture developments within further and higher education such as virtual universities, identity management, and integration with enterprise-level services and applications.