1). Introduction:
This document explains about VoIP systems. Recent happenings like Internet diffusion at low cost, new integration of dedicated voice compression processors have changed common user requirements allowing VoIP standards to diffuse. This how to tries to define some basic lines of VoIP architecture.
What is VoIP?
VoIP stands for ' V’oice ‘o’ver ‘I’nternet ‘P’rotocol. As the term says VoIP tries to let go voice (mainly human) through IP packets and, in definitive through Internet. VoIP can use accelerating hardware to achieve this purpose and can also be used in a PC environment.
How does it work?
Many years ago we discovered that sending a signal to a remote destination could have be done also in a digital fashion: before sending it we have to digitalize it with an ADC (analog to digital converter), transmit it, and at the end transform it again in analog format with DAC (digital to analog converter) to use it.
VoIP works like that, digitalizing voice in data packets, sending them and reconverting them in voice at destination.
Digital format can be better controlled. We can compress it, route it, convert it to a new better format, and so on; also we saw that digital signal is more noise tolerant than the analog one (see GSM vs. TACS).
TCP/IP networks are made of IP packets containing a header (to control communication) and a payload to transport data: VoIP use it to go across the network and come to destination.
Voice (source) - -ADC - -Internet - - DAC- - Voice (dest.) 2). Background:
The past:
For the past 100 years people have relied on the PSTN for voice communication. The two parties using the line. No other information can travel over the line, although there is often during a call between two locations, the line is dedicated to plenty of bandwidth available.
Later, as data communications emerged, companies paid for separate data lines so their computers could share information, while voice and fax communications were still handled by the PSTN.
More than 30 years ago Internet didn’t exist. Interactive communications were only made by telephone at PSTN line cost.
Data exchange was expansive (for a long distance) and no one had been thinking to video interactions (there was only television that is not interactive, as known).
The present:
Today we can see a real revolution in communication world: everybody begins to use PCs and Internet for job and free time to communicate each other, to exchange data (like images, sounds, documents) and, sometimes, to talk each other using applications like Net meeting or Internet Phone. Particularly starts to diffusing a common idea that could be the future and that can allow real-time vocal communication: VoIP.
Today, with the rapid adoption of IP, we now have a far-reaching, low-cost transport mechanism that can support both voice and data. A VOIP solution integrates seamlessly into the data network and operates alongside existing PBXs, or other phone equipment, to simply extend voice capabilities to remote locations. The voice traffic essentially "rides for free" on top of the data network using the IP infrastructure and hardware already in place.
The future:
We cannot know what is the future, but we can try to image it with many computers, Internet almost everywhere at high speed and people talking (audio and video) in a real time fashion. We only need to know what will be the means to do this: UMTS, VoIP (with video extension) or other? Anyway we can notice that Internet has grown very much in the last years, it is free (at least as international means) and could be the right communication media for future.
3). Requirement:
Hardware Requirement:
To create a little VoIP system you need the following hardware:
1. PC 386 or more
2. Sound card, full duplex capable
3. A network card or connection to internet or other kind of interface to allow communication between 2 PCs
Software requirement:
We can choose what O.S. To use:
1. Win9x
2. Linux
4). VOIP’s implementation:
The IP Gateway: The gateway was the first "stand alone" form of IP technology; a separate piece of hardware that is placed onto an Intranet above a phone endpoint. Once in place, it converts an existing network of traditional analog phones into a network of Voice over IP phones, while continuing to allow the phones to place calls through the PSTN.
When a call is placed, standard voice transmission from the phone is compressed and transferred in a gateway-to-gateway format.
The IP Phone:
The IP phone implements the same technology, packet zing voice data and transmitting it over data signaling lines; but it combines this technology with the features of an office phone network in one platform.
The primary advantage to the phone is having IP capability without having to add any hardware to the communication chain. It appears in an office environment as a standard desktop phone, but delivers the functionality and savings of IP technology.
5). Classification of connection: The VoIP connection can be classified by the type of devices performing an Internet call. Please note that the term PC can be applied to any device capable of transmitting voice over data network. It does not necessarily have all the features of a standard computer. It could just look like a traditional telephone with the basic elements of a computer to execute an Internet call. We have the following generic classifications.
PC to PC:
For users who already have an Internet access and an audio-capable PC. This scenario can take advantage of integration with other Internet services such as World Wide Web, instant messaging, e-mail, etc.
PC to telephone or telephone to PC:
In this scenario, PC-callers may reach also the PSTN users. A gateway converting the Internet call into a PSTN call has to be used. Traditional telephone users also can make a call to a PC going through the gateway that connects the IP network with PSTN.
Telephone to telephone:
The IP network can be a dedicated backbone to connect PSTN. Gateways should connect PSTN to the IP network.
6). BASIC SYSTEM COMPONENTS OF VoIP:
There are three major system components to VoIP technology: clients, servers, and gateways.
Clients:
The client comes in two basic forms. It is either a suite of software running on a user’s PC that allows the user, through a GUI, to set-up and clear voice calls, encode, packetize and transmit outbound voice information from the user’s microphone and receive, decode and play inbound voice information through the user’s speaker or headsets. The other type of client, known as a ‘virtual’ client, does not have a direct user interface, but resides in gateways and provides an interface for users of POTS.
Servers:
In order for IP Telephony to work and to be viable as a commercial enterprise, a wide range of complex database operations, both real-time and non-real-time, must occur transparently to the user. Such applications include user validation, rating, accounting, billing, revenue collection, revenue distribution, routing (least cost, least latency or other algorithms), management of the overall service, downloading of clients, fulfillment of service, registration of users, directory services, and more.
Gateways:
Void technology allows voice calls originated and terminated at standard telephones supported by the PSTN to be conveyed over IP networks. VoIP "gateways" provide the bridge between the local PSTN and the IP network for both the originating and terminating sides of a call. To originate a call, the calling party will access the nearest gateway either by a direct connection or by placing a call over the local PSTN and entering the desired destination phone number.
The VoIP technology translates the destination telephone number into the data network address (IP address) associated with a corresponding terminating gateway nearest to the destination number. Using the appropriate protocol and packet transmission over the IP network, the terminating gateway will then initiate a call to the destination phone number over the local PSTN to completely establish end-to-end two-way communications. Despite the additional connections required, the overall call set-up time is not significantly longer than with a call fully supported by the PSTN.
The gateways must employ a common protocol - for example, the H.323 or SIP or a proprietary protocol - to support standard telephony signaling. The gateways emulate the functions of the PSTN in responding to the telephone's on-hook or off-hook state, receiving or generating DTMF digits and receiving or generating call progress tones. Recognized signals are interpreted and mapped to the appropriate message for relay to the communicating gateway in order to support call set-up, maintenance, billing and call tear down.
7). Protocols of VOIP:
There are two protocol classifications that used in the VoIP system that is the protocol for the signaling and for sending the conversation data in the IP medium. For the signaling protocol class there are several methods, while protocol to send the conversation data, which is de facto use a method, named RTP/RTCP (Real Time Protocol/Real Time Control Protocol).
1. Signaling Protocol/Call Control:
There are two standard institutions that made the signaling protocol, which are dominated the VoIP application protocol IETF and ITU-T. The IETF published the SIP and S/MGCP protocol; in the meanwhile ITU-T published the H.323. On the next development, IETF and ITU-T make cooperation to enhance the MGCP protocol to be MEGACO, which is hopefully, become the VoIP signaling protocol standard in the near future.
H323:
At the beginning, H.323 was designed to use in the videoconference application, but then quickly enhance on the VoIP application. On this protocol standard is used the other protocol standards in the ITU-T. The standard that used as the reference on H.323 is H.225-0 for call control protocol and H.245 for logical channel protocol. The function of the call control function is starting the call setup, in the meanwhile the logical channel protocol, is capability and bandwidth control and the control of the channel number that will use to transport the conversation data.
S/MGCP:
The S/MGCP is the text base protocol as the other IETF protocol. The S/MGCP has the different model from the H.323. On the S/MGCP is only recognized the two elements that are the Media Gateway and the Media Gateway Controller as discussed. The Media Gateway is the end point of the conversation on the S/MGCP equal to the terminal on the H.323 protocol. The Media Gateway Control availability on S/MGCP is absolute since the Media Gateway couldn’t make the direct call setup with the other Media Gateway. The Media Gateway should always make the coordination with the Media Gateway Control. As the conversation formed, the Media Gateway makes the direct conversation connection with the destination Media Gateway using the RTP/RTCP protocol.
SIP:
The other signaling protocol from IETF is the SIP, which is text base. The used model on SIP is the same as the model that used on H.323, but it is simpler. On the SIP is recognized the terminal, the proxy server that occurred as the gatekeeper on H.323 and the gateway. The call procedure is also take the same model, between the terminal can make the direct call setup or through the proxy server.
MEGACO:
MEGACO is the future VoIP protocol that the result of cooperation between MEGACO working group on IETF and H.GCP group in ITU-T. By the MEGACO protocol availability, it is to be hope that the interoperability level between the VoIP equipment is growing better, so that can be equal to the interoperability level of the circuit switch network.
2. Transport Media Protocol:
The protocol on this step is defined the VoIP quality since it was used during the communication. By this time the RTP/RTCP is the defector protocol.
RTP/RTCP:
The RTP protocol responsible to control the voice packets consecutively and then run through the IP network. Then on the receiver side responsible to re-arrange those packets to the form of the voice signal. On this receiver side the RTP protocol become the most important part in the VoIP system.
The first function of the RTP is made the incoming packets buffering. The buffering is important to exceed the incoming packet that didn’t guarantee when the time of the incoming is. Through those packets should consecutively control. There are several probabilities that will happen during the way of the packet to the destination:
- Has the different taken time that caused the jitter?
- Missing along the way
- Come on the wrong sequence
The buffer solves the first problem, so that will decrease the jitter effect. As bigger the buffer as decreased the jitter effects but will increase the delay time of the signal to the listener.
If there is the missing packet on the way, so that packet will be replaced by the previous packet but with the deducted volume. On this circumstance the packet that comes lately, that packet was ignored and assumed as the missing packet.
8). VoIP and FoIP – The story so far:
Voice over Internet Protocol (VoIP) and Fax over Internet Protocol (FoIP) are cost-efficient methods of transmitting voice and fax transmissions over the public and private Data Network utilizing packet switching. Voice and fax services have traditionally relied upon the circuit switched network, while other services such as switched data and video conferencing have used packet transmission protocols on the data network. By adopting VoIP and FoIP, the full family of telecommunication services including voice and fax may be handled on the data network utilizing packet switching, hence avoiding the majority of associated circuit switched network costs.
VoIP and FoIP are now graduating to Voice over Frame Relay (VoFR) and Voice over ATM (VoATM) consistent with the data network evolution.
VoIP / FoIP applications should have particular appeal for Internet Service Providers (ISPs), Enhanced Service Providers (ESPs) Internet Telephony Service Providers (ITSPs), or companies operating a private network.
There are four methods by which we can pass fax over Internet.
PC to PC or PC to Fax:
Allows Internet users to exclusively utilize PC programs (Word, Excel, E-mail, etc…) to compose facsimiles, and then through the use of a software or hardware application, send the newly composed document over the Internet as a fax document to a PC or fax machine. Generally, these methods use an Internet fax software application.
Fax machine to Fax machine or Fax machine to PC:
Allows Internet users to send and receive fax document as usual, however, this method utilizes a traditional fax machine with Internet fax capabilities, a software application or other application that enables you to send a document to the fax machine, software program or fax enabling device to send the document to a fax machine, software, program, printer etc.
9). Benefits of VoIP:
Voice communications will certainly remain as basic form of interaction among people. A simple replacement of PSTN is hard to implement in short term. The immediate goal for many VoIP service providers is to reproduce existing telephone capabilities at a significantly lower cost and offer a quality of service competitive to PSTN. In general, the benefits of VoIP technology can be the following:
Low cost:
By avoiding traditional telephony access charges and settlement, a caller can significantly reduce the cost of long distance calls. Although the cost reduction is somewhat related to future regulations, VoIP certainly adds an alternate option to existing PSTN services.
Network efficiency:
Packetized voice offers much higher bandwidth efficiency than circuit-switched voice because it does not take up any bandwidth in listening mode or during pauses in a conversation. It is a big saving when we consider a significant part of a conversation is silence. The network efficiency can also be improved by removing the redundancy in certain speech patterns. If we were to use the same 64 Kbps Pulse Code Modulation (PCM) digital-voice encoding method in both technologies, we would see that bandwidth consumption of packetized voice is only a fraction of the consumption of circuit-switched voice. The packetized voice can take advantage of the latest voice-compression algorithms to improve efficiency.
Simplification and consolidation:
An integrated infrastructure that supports all forms of communication allows more standardization and reduces the total equipment and management cost. The combined infrastructure could support bandwidth optimization and a fault tolerant design. Universal use of the IP protocols for all applications reduces both complexity and more flexibility. Directory services and security services could be more easily shared.
Even though basic telephony and facsimile are the initial applications for VoIP, the longer-term benefits are expected to be derived from multimedia and multi-service applications. Combining voice and data features into new applications will provide a significant return over the longer term. In out project we will discuss what are the development challenges and basic system components to implement VoIP technology. We will also present the two most relevant protocols - H.323 and SIP - and compare them.
10). Applications of VoIP:
Voice over IP is marketed to multi-location businesses looking to reduce toll charges associated with intra-office calling. It is designed to help you maximize investments you've already made in your data and voice network infrastructure. Some examples of the many applications for a voice network include the following:
Office-to-Office Communication:
A VOIP network can be as small as two offices or as large as hundreds of offices. Each office installs and configures a VOIP solution on their nework to begin placing calls or sending faxes to the other offices on the VOIP network.
Off-Net Calling:
Telecommuters or customers off the IP network can make toll-free long distance calls by dialing into a local VOIP solution and placing calls to any other location on the VOIP network. You can even have a VOIP solution at a remote site dial a local phone number for a free person-to-person long distance call.
Create Off-Premise Extensions:
Extend the reach of your PBX into home office locations. Simply connect a VOIP solution to the PBX at the corporate office, and another VOIP solution at the remote office. Now, anyone can place calls to the remote office by simply dialing an extension number.
Replace Expensive Tie Lines:
A corporation that utilizes Tie lines to connect branch office PBXs to the corporate PBX can now use the company's IP-based Wide Area Network to complete the call.
Communication is the key to big businesses performance. Without a good communication system, an organization can fall quickly behind from repetition, poor service, and missed opportunities.
A VoIP system greatly reduces the chances of such pitfalls by consolidating the communication chain and unifying the office environment. Any organization could quickly replace their traditional analog system with an IP system, largely because the structure is already in place. As was discussed before, to employ a gateway system, phone system, or a combination of the two, only involves a change of endpoints. Once implemented, an IP system provides many features that aren't available with PBX technology. One example is the "voice-button", a button that is placed on an organizations web page that, when pressed, automatically connects a call to a selected department. This allows the company to maximize a customer's interest, at the time of interest.
Another example is the combination of voice mail and email systems. In a VoIP network it is possible to have voice mail transmitted to a users email box and then played on the recipients PC or to have email retrieved by a voice mail system and read back over the phone.
The
In the same situation, gateways can be installed at individual workstations to IP enable specific departments, using the phones and extension systems that are already in place.
There is also the "in the closet option". Which refers to IP enabling all the phones on a current system by installing a multi port gateway further up in the communication chain? Locating the gateway in the electrical center of a building, or "the closet" allows the office environment to remain unchanged.
The Residential Application:
The residential application represents the final implementation of IP technology. In the future, IP systems will be present in every household, providing reliable service around the world, minimizing cost, and tying together all available media. But when considering the residential application, it is possible to get a clear picture of its future and current benefits.
Traditional Residential Communication
The Residential Solution:
The benefits of a residential IP system are quickly becoming apparent as it is applied in a small environment that is meted by tight budgets and monthly bills.
Currently a household can only gain access to Internet calling through an Internet Service Providers (ISP's); but both parties are rewarded from this symbiotic relationship. Subscribers can purchase blocks of long distance calling each month for a static base rate, dramatically reducing the cost of typical long distance communication; while the ISPs only have to pay for the local charges incurred before gateway-to-gateway transmission.
In this application when the subscriber places a call, it is first sent to the ISP's nearest gateway, where it is then packetized and transmitted over the IP network to the gateway that is closest to a calls destination. Once there, the packets are translated and switched to the PSTN to make the final connection. By employing the IP system, providers avoid the long distance tariffs that govern the PSTN.
While this previous scenario allows IP communication in a residential application today, the benefits of a true residential solution have yet to be realized. As the larger communications companies become more involved with the IP market, a residential unification of incoming media is beginning to evolve. Advances in gateway technology are now being aimed the consolidation of residential entertainments multiple media signals. These "household gateways" are being designed to act as a communications hub for a home, receiving all transmissions from a single line and providing channels for Internet connection, television programming, and telephone communication throughout the house.
11). Development challenges:
The goal of VoIP developers is to add telephone calling capabilities to IP-based networks and interconnect these to traditional public telephone network and to private voice networks maintaining current voice quality standards and preserve the features everyone expects from the telephone. We can summarize the technical challenges as the following.
Quality of Service (QoS):
The voice quality should be comparable to what is available using the PSTN, even over networks of varying levels of QoS. The following factors decide the VoIP quality:
Packet loss:
In order to operate a multi-service packet based network at a commercially viable load level, random packet loss is inevitable. This is particularly true with communications over the Internet where traffic profiles are highly unpredictable and the competitive nature of the business drives corporations to load their networks to the maximum.
Packetizing voice codecs are becoming better at reducing sensitivity to packet loss. The main approaches are smaller packet sizes, interpolation (algorithmic regeneration of lost sound), and a technique where a low-bit-rate sample of each voice packet is appended to the subsequent packet. Through these techniques, and at some cost of bandwidth efficiency, good sound quality can be maintained even in relatively high packet loss scenarios.
As techniques for reducing sensitivity to packet loss improve, so a new opportunity for the achievement of even greater efficiencies is presented. This refers to the suppression of the transmission of voice packets whose loss is determined by the encoder to be below a threshold of tolerability at the decoder. This is particularly attractive in the packet based networking world where statistical multiplexing favors the reuse of freed-up bandwidth.
Delay:
Two problems that result from high end-to-end delay in a voice network is echo and talker overlap. Echo becomes a problem when the round-trip delay is more than 50 milliseconds. Since echo is perceived as a significant quality problem, VoIP systems must address the need for echo control and implement some means of echo cancellation. Talker overlap (the problem of one caller stepping on the other talker’s speech) becomes significant if the one-way delay becomes greater than 250 milliseconds. The end-to-end delay budget is therefore the major constraint and driving requirement for reducing delay through a packet network.
Propagation delay (the time taken for the information wave-front to travel a given distance through a given media), jitter buffering, packetization, analog to digital encoding and digital to analog decoding delays are responsible for most of the overall delay. Service and wait time through the switching and transmission elements of the network may be considered trivial given the small packet sizes and relatively wide bandwidths prevalent on the Internet. It is generally true that when considering the achievable quality of a given service, the overall geographic distance traveled by a call is far more important than the complexity of its routing, (i.e. the number of intermediary nodes or "hop-count").
Jitter:
Jitter is the variation in inter-packet arrival time as introduced by the variable transmission delay over the network. Removing jitter requires collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence, which causes additional delay. The jitter buffers add delay, which is used to remove the packet delay variation that each packet is subjected to as it transits the packet network.
Overhead:
Each packet carries a header of various sizes that contains identification and routing information. This information, necessary for the handling of each packet, constitutes ‘overhead’ not present with circuit switching techniques. Small packet size is important with real-time transmissions since packet size contributes directly to delay and the smaller the packet size, the less sensitive a given transmission would be to packet loss. Various new techniques such as header compression are evolving to reduce the packet overhead in IP networks. It is likely that packet based networks, of one form or another, will eventually approach the efficiency, with respect to overhead, of circuit-based networks.
User friendly design:
The user need not know what technology is being used for the call. He should be able to use the telephone as he does right now.
Easy configuration:
An easy to use management interface is needed to configure the equipment. A variety of parameters and options such as telephony protocols, compressing algorithm selections, dialing plans, access controls, PSTN fall back features, port arrangement etc. are to be taken care of.
Addressing/Directories:
Telephone numbers and IP addresses need to be managed in a way that it is transparent to the user. PCs that are used for voice calls may need telephone numbers. IP enabled telephones IP addresses or an access to one via DHCP protocols and Internet directory services will need to be extended to include mappings between the two types of addresses.
Security issues:
VOIP networks introduce some new risks to carriers and their customers, risks that are not yet fully appreciated. Responding to these threats requires some specific techniques, comprehensive, multi-layer security policies, and firewalls that can handle the special latency and performance requirements of VoIP.
It is important to remember that a VoIP network is an IP network. Any VoIP device is an IP device, and it's therefore vulnerable to the same types of attacks as any other IP device. In addition, a VoIP network will almost always have non-VoIP devices attached to it and be connected to other mission-critical networks.
Every IP network, regardless of how private it is, eventually winds up connected to the global Internet. Even if it is not possible to directly route a packet from the "private" network onto the Internet, it is extremely likely that some host on the "private" network will also be connected to a less private network. Compromising this host provides an attacker with a gateway into the presumed secure private network. It's important, therefore, to secure all IP networks, but VoIP networks have special security requirements. Specific techniques, comprehensive policies, and VoIP-capable firewalls are needed to do the job right.
Billing issues:
VOIP gateways must keep track of successful and unsuccessful calls. Call detail records should be produced. But the major issue is the suitable billing model selection. The following billing models can be applied.
| | |
| |
| | | |||
| | |
| |
1.Time based: Metered by flow duration, time-of-day, time-of week.
2. Destination, Carrier based: Rated by called and calling station IDs associated with the sequence of stages used to support the call.
3. QoS based: rated by established service parameters such as priority, selected QoS, and latency
12). VoIP solutions:
From the wide variety of VOIP solutions available today, the one you select depends on the size of your business, the level of networking expertise available, the amount of integration with legacy equipment, and the level of voice quality.
Routers:
Router solutions usually replace an existing network router and keep voice and data all in a single box. However, this solution requires networking expertise, and can be costly to install, while placing network services at risk during deployment and maintenance.
VoIP server cards:
VOIP server cards can be an economical VOIP solution. However, they must be compatible with the server and operating system and installations can be complex.
IP-Based PBX:
The IP-based PBX is usually software running on a computer-based server. However, it often requires a forklift upgrade of the existing PBX or, at a minimum, an extensive software and/or hardware upgrade. An IP-based PBX is typically marketed to new installations where no legacy system is in place.
PC-Based telephony:
PC-based telephony software is by far the cheapest VOIP solution, but it is also the clumsiest. It requires users to make phone calls using their PC instead of a phone. This usually requires user training and an investment in speakers and microphones for each PC. Plus, many users complain that voice quality for this solution is not adequate for business communications.
IP Gateway:
An IP gateway, like Multi-Tech's MultiVOIP, is often the most suitable VOIP solution for small to midsize businesses and remote sites. It does not disturb your existing data infrastructure because it simply drops into the Ethernet network. Furthermore, it operates alongside existing PBXs or other phone equipment to extend voice capabilities to remote locations or users. An IP gateway requires only a minimal investment in product, installation, and user training.
13). IS VoIP the future of telecommunications?
VoIP means that the technology used to send data over the Internet is now being used to transmit voice as well. The technology is known as packet switching. Instead of establishing a dedicated connection between two devices (computers, telephones, etc.) and sending the message "in one piece", this technology divides the message into smaller fragments, called 'packets'. These packets are transmitted separately over a decentralized network and when they reach the final destination, they're reassembled into the original message.
VoIP allows a much higher volume of telecommunications traffic to flow at much higher speeds than traditional circuits do, and at a significantly lower cost. VoIP networks are significantly less capital intensive to construct and much less expensive to maintain and upgrade than legacy networks (traditional circuit-switched networks). Since VoIP networks are based on Internet protocol, they can seamlessly and cost-effectively interface with the high technology, productivity-enhancing services shaping today's business landscape. These networks can seamlessly interface with web-based services such as virtual portals, interactive voice response (IVR), and unified messaging packages, integrating data, fax, voice, and video into one communications platform that can interconnect with the existing telecommunications infrastructure.
Industry experts see VoIP as a tool that will become the standard platform for the international calling market. At present, VoIP constitutes 2% of the international calling market, estimated by Frost and Sullivan, a prominent marketing research group, at 325 billion for the year 2000. That is approximately 50% of the total telecom market value of $700 billion. We strongly believe that the profit realization VoIP will trigger in the global telecommunications industry will dwarf the impact of the now ancient "digital revolution" of 20 years ago.
As with any promising new technology, a myriad of companies are trying to climb aboard the VoIP bandwagon. Currently, however, the industry is characterized by a high degree of confusion. Most companies, including large resource-rich national and international telecommunications carriers, are experiencing enormous difficulty in building effective international VoIP networks. They are unable to harness the power of VoIP or effectively communicate the benefits of VoIP to their customers.
We mustn’t ignore the problem of scalability. The system has to be designed so that it can grow. And each segment within the system must be able to grow. Creating an architecture that can handle billions of minutes of use per month requires a solution with high call processing capabilities. If we are looking at a global solution, we have to start from the beginning with a global approach. And that’s one of the reasons why a fully implemented solution won’t be available tomorrow.
But when will it be available?
Figure 1 shows Probe Research’s take on the future. As you can see, there won’t be an awful lot of activity for at least two years. But four years from now—wow! That, of course, means that companies that are serious about the telecommunications business must get ready. For most, it won’t be a sprint. On the contrary, we will likely watch providers train for a long and grueling marathon—with VoIP at the finish line.
14). Conclusion:
The applications presented in this paper focused on the possibilities that become available to the user of an IP system. As discussed, the advances made in Voice over IP field stand to revolutionize the communications industry as new products are developed. Interest has grown in recent years as Telecom industry reports announced data transfer surpassing voice transfer usage in 1997, increasing demand and fuelling the search for successful standards-compliant IP platforms.
Since its inception, e-tel Corporation has been working to develop H.323 standards based platforms that present viable IP solutions for most applications. Shortly after the successful release of its Free Ride Gateway, e-tel announced the completion of one of the first standards-compliant IP phone.
TO DOWN LOAD REPORT AND PPT
