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 ' Vâ„¢oice Ëœoâ„¢ver ËœIâ„¢nternet ËœPâ„¢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 didnâ„¢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). VOIPâ„¢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.
Traditional Analog Phone Connection

VoIP Gateway System


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.

Traditional Office Communication System

VoIP Gateway System

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:

Figure 1 PC-to-PC Scenario
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:


Figure 2 PC to Phone or Phone to PC Scenario
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.

Paper Presentation
On
(VOICE OVER INTERNET PROTOCOL)

---

Prepared By:
Nirali M.Patel Hiral R. Patel
B.E.5th (I.T) B.E.5th (I.T)
Sankalchand Patel College of Engg.
Visnagar.
Gujarat

---

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 ' Vâ„¢oice Ëœoâ„¢ver ËœIâ„¢nternet ËœPâ„¢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 didnâ„¢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). VOIPâ„¢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.
Traditional Analog Phone Connection
VoIP Gateway System
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.
Traditional Office Communication System
VoIP Gateway System
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:
Figure 1 PC-to-PC Scenario
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:
Figure 2 PC to Phone or Phone to PC Scenario
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:
Figure 3 Phone-to-Phone Scenarios
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 userâ„¢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 userâ„¢s microphone and receive, decode and play inbound voice information through the userâ„¢s speaker or headsets. The other type of client, known as a Ëœvirtualâ„¢ 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 couldnâ„¢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 didnâ„¢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.
Enterprise Environment Application:
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 Enterprise IP Solution: A large organization with multiple offices located in different areas can stay connected through an IP system with any arrangement of endpoints. One option is the desktop IP phone solution. The desktop IP phone can take the place of any traditional office phone, without having to maintain the traditional connection. The existing phone lines can be physically removed from an office because the IP phone only requires a computer connection. Calls can be made between offices or between continents with no difference in cost, while retaining access to the same voicemail, email, and other office systems.
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.
Enterprise IP Solutions:
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 talkerâ„¢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 Ëœoverheadâ„¢ 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 mustnâ„¢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 thatâ„¢s one of the reasons why a fully implemented solution wonâ„¢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.

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What is VOIP ?

Voice over Internet Protocol transfers voice through IP packets through the Internet. VoIP can use special

hardware or a PC environment to achieve this purpose. IP Telephony is defined as the use of networks to

transmit both voice and data packets



How does VOIP works:

Voice (source) - - ADC - - - - Internet - - - DAC - - Voice (dest)


DEFINITIONS OF IMP TERMS

ZoneThe collection of a gatekeeper and the endpoints registered with it is called a zoneNetwork AddressFor

each H.323 entity, a network address is assigned and this address uniquely identifies the H.323 entity on the

network TSAP Identifier These TSAP identifiers allow multiplexing of several channels sharing the same network

address 


PACKET FREQENCY

Packet frequency is the number of packets containing voice samples which are sent per second


Components of H.323( it is a terminal)
Different Terminals
Gateways
Gatekeepers


H.323 TERMINAL SUPPORTING OTHER PROTOCOLS


H.245 : allowing the usage of the channels.

Q.931: call signaling and setting up the call

RTP: carries voice packets

RAS: interacting with the gatekeeper





Interface between the PSTN and the Internet

Translator : H.225 to H.221 or audio to video
GATEKEEPERS
Controls the end points under its zone

Function

Address Translation
Admissions Control
Call signaling
Call Authorization
Bandwidth Management

H.323 PROTOCOL STACK (layers)
CALL SETUP IN H.323(working)
Endpoint enters the call setup phase
Discovering a gatekeeper which would take the management of that endpoint
The capability exchange takes place between the endpoint and the gatekeeper
Registration of the endpoint with its gatekeeper 
The call is established 
MULTIPOINT CONTROL UNITS (MCU)
SESSION INITIATION PROTOCOL (SIP)



It is an application layer control protocol for
creating, modifying and terminating sessions with one or more participants.
SIP can also invite participants to already existing sessions, such as multicast conferences.
Media can be added to (and removed from) an existing session.
SIP transparently supports name mapping and redirection services, which supports personal mobility “
SIP MESSAGES(commands)


INVITE: for inviting a user to a call
BYE: for terminating a connection between the two end points
ASK: invitation for reliable exchange of messages
OPTIONS: for getting information about the capabilities of a call
REGISTER: gives information about the location of a user to the SIP registration server.
CANCEL: for terminating the search for a user

SAMPLE SIP OPERATION


SUPPORTING PROTOCOLS(along SIP)
VOIP AND PHONE
VoIP Networking “ The Problem
A VoIP network can grow out of control and needs a solid control layer and network management system to maintain

revenue assurance and allow further growth
Configurations in each gateway are complex and prone to error
Multiple sources of CDRs “ complex CDR analysis and billing
No control over traffic demand “ high demand causes gateway overload and lost revenue
Multiple gateways and no common routing point


CONCLUSION
Like everything else, as the technology changes so At first, only a few companies like Cisco and Lucent offered

VoIP services, but the large telecommunications carriers “ such as AT&T and Sprint -- are catching on

VoIP is predominately used for personal instead of enterprise-wide use, but will the people who use it.

CONCLUSION(contd¦)
VoIP offers opportunity to increase robustness and decrease costs
So future of communications is IP TELEPHONY.


THE END



THANK YOU

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INTRODUCTION

VoIP (Voice Over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service
VoIP is therefore telephony using a packet based network instead of the PSTN (circuit switched).
The first VoIP application was introduced in 1995 - an "Internet Phone". An Israeli company by the name of "VocalTec" was the one developing this application. The application was designed to run on a basic PC. The idea was to compress the voice signal and translate it into IP packets for transmission over the Internet. This "first generation" VoIP application suffered from delays (due to congestion), disconnection, low quality (both due to lost and out of order packets) and incompatibility.
VocalTec's Internet phone was a significant breakthrough, although the application's many problems prevented it from becoming a popular product. Since this step IP telephony has developed rapidly. The most significant development is gateways that act as an interface between IP and PSTN networks.

What is Voice Over IP?
Voice over IP (VoIP) is a blanket description for any service that delivers standard voice telephone services over Internet Protocol (IP). Computers to transfer data and files between computers normally use Internet protocol.
"Voice over IP is the technology of digitizing sound, compressing it, breaking it up into data packets, and sending it over an IP (internet protocol) network where it is reassembled, decompressed, and converted back into an analog wave form.." The transmission of sound over a packet switched network in this manner is an order of magnitude more efficient than the transmission of sound over a circuit switched network.
As mentioned before, VoIP saves bandwidth also by sending only the conversation data and not sending the silence periods. This is a considerable saving because generally only one person talks at a time while the other is listening. By removing the VoIP packets containing silence from the overall VoIP traffic we can reach up to 50% saving. In a circuit switched network, one call consumes the entire circuit. That circuit can only carry one call at a time.
In a packet switched network, digital data is chopped up into packets, sent across the network, and reassembled at the destination. This type of circuit can accommodate many transmissions at the same time because each packet only takes up what bandwidth that is necessary.. Internet Telephony simply takes advantage of the efficiencies of packet switched networks.
Gateways are the key component required to facilitate IP Telephony. A gateway is used to bridge the traditional circuit switched PSTN with the packet switched Internet.
Requirements of a VoIP
The requirements for implementing an IP Telephony solution to support Voice Over IP varies from organization to organization, and depends on the vendor and product chosen. The following section aims to identify the fundamental requirements in the general case and is split into 3 sections:
Hardware Requirements
Protocol Requirements
Software Requirements
Software Requirements
The software package chosen will reflect the organizational needs, but should contain the following modules as defined in the Technology Guide Series - Voice Over IP Publication, and other sources.
Voice Processing Module This aspect of the software is required to prepare voice samples for transmission. The functionality provided by the voice processing module should support:
A PCM Interface is required to receive samples from the telephony interface (e.g. a voice card) and forward them to the Voice Over IP software for further processing.
Echo Cancellation is required to reduce or eliminate the echo introduced as a result of the round trip exceeding 50 milliseconds.
Idle Noise Detection is required to suppress packet transmission on the network when there are no voice signals to be sent. This helps to reduce network traffic as up to 60% of voice calls are silence and there is no point in sending silence.
A Tone Detector is required to discriminate between voice and fax signals by detecting DTMF (Dial Tone Multi frequency) signals.
The Packet Voice Protocol is required to encapsulate compressed voice and fax data for transmission over the network.

A Voice Playback Module is required at the destination to buffer the incoming packets before they are sent to the Codec for decompression.
Call Signaling Module This is required to serve as a signaling gateway which allows calls to be established over a packet switched network as opposed to a circuit switched network (PSTN for example).
Packet Processing Module This module is required to process the voice and signaling packets ready for transmission on the IP based network.
Network Management Protocol Allows for fault, accounting and configuration management to be performed.
Hardware Requirements
The exact hardware, which would be required, again, depends on organizational needs and budget. The list below highlights the most general hardware required.
The most obvious requirement is the existence (or installation) of an IP based network within the branch office gateway is required to bridge the differences between the protocols used on an IP based network and the protocols used on the PSTN.
The gateway takes a standard telephone signal and digitizes it before compressing it using a Codec. The compressed data is put into IP packets and these packets are routed over the network to the intended destination.
The PC's attached to the IP based network require the voice/fax software outlined above. They also require Full Duplex Voice Cards which allow both communicating parties to speak at the same time - as often happens in reality.
As an alternative to installing Voice Cards, IP Telephones can be attached to the network to facilitate Voice Over IP. A secondary gateway should be considered as a backup in the event of the failure of the primary gateway.
Protocol Requirements
There are many protocols in existence but the main ones are considered to be the following:
H.323 is an ITU (International Telecommunications Union) approved standard which defines how audio/visual conferencing data is transmitted across a network. H.323 relies on the RTP (Real-Time Transport Protocol) and RTCP (Real Time Control Protocol) on top of UDP (User Datagram Protocol) to deliver audio streams across packet based networks.
G.723.1 defines how an audio signal with a bandwidth of 3.4KHz should be encoded for transmission at data rates of 5.3Kbps and 6.4Kbps. G.723.1 requires a very low transmission rate and delivers near carrier class quality. The VoIP Forum as the baseline Codec for low bit rate IP Telephony has chosen this encoding technique.
G.711. The ITU standardized PCM (Pulse Code Modulation) as G.711. This allows carrier class quality audio signals to be encoded for transmission at data rates of 56Kbps or 64Kbps. G.711 uses A-Law or Mu-Law for amplitude compression and is the baseline requirement for most ITU multimedia communications standards.
Real-Time Transport Protocol (RTP) is the standard protocol for streaming applications developed within the IETF (Internet Engineering Task Force).
Resource Reservation Protocol (RSVP) is the protocol which supports the reservation of resources across an IP network. RSVP can be used to indicate the nature of the packet streams that a node is prepared to receive.

MAIN TYPES OF VoIP
VoIP has broadly three main branches, which can and do overlap.
VoIP over the Internet This is probably the best known and most publicized, talking PC to PC. Basically free telephone calls. The call is only free if both parties to the call have access to the public Internet at zero cost..
Advantage. free calls regardless of distance or length of call.
Disadvantage. often the voice quality is bad due to the lack of bandwidth available for the call.
Other factors Have to use a PC or other computer running VoIP software.
Office to Office A large multinational company will have offices across the whole country. They have a fixed data network connecting all the offices together. This allows every computer access to every other computer in the company. By installing a VoIP Gateway in each office and connecting it to the office legacy PBX and to the data network, employees use the data network for voice calls between offices.
Advantages. Interoffice calls are free, since the company already has the bandwidth between offices. The technology is transparent to the user, and requires minimum training. The only new equipment required is a gateway at each office. Voice quality is good, because the company has control over the bandwidth.
Disadvantage. Extra bandwidth may be required between offices, which offset the savings.
Other factors... The carrier providing the interoffice bandwidth will almost certainly offer an alternative solution including management of the internal telephone traffic.

IP PBX A traditional Private Branch Exchange (PBX) connects all the phones within an organization to the public telephone network. Essentially IP PBX replaces all the internal phones with VoIP telephones. The IP PBX has standard telephone trunk connections to the public telephone network. The IP PBX is a PBX with VoIP, but it also has the ability to support VoIP over the Internet and Office to Office VoIP.
Advantages. Single cable infrastructure. The technology is transparent to the user, and requires minimum training. Future proof technology.
Disadvantages. Primarily useful for Greenfield sites, but can be adapted to work with existing technology.

WORKING OF VOIP

How VoIP works : Part 1
Let us look at very simple VoIP call. Consider two VoIP telephones connected via an IP network .In this example both VoIP telephones are connected to a local LAN. Sallyâ„¢s phone has an IP address of 192.168.1.1, Billâ„¢s phone is 192.168.1.2, and the IP addresses uniquely identify the telephones. Both our phones are configured to use a widely used VoIP standard called H.323.
Bill wants to talk to Sally and his phone knows the IP address of Sallyâ„¢s phone. Bill lifts the handset and 'dials' Sally, the phone sends a call setup request packet to Sally's phone, Sallyâ„¢s phone starts to ring, and responds to Bill's phone with a call proceeding message. When Sally lifts the handset the phone sends a connect message to Bill's phone. The two phones will now exchange the data packets containing the speech. At the end of the call Bill replaces his handset and phone stops sending voice data sends a disconnect message and Sally's phone responds with a release message. The call is now complete. all the messages contain the Q931 ISDN protocol.
Having introduced VoIP I will now talk about three main 'types' of VoIP installed in the market place today.
How VoIP works part 2 : The Protocols.
I have made an assumption that both ends of a VoIP telephone conversation are compatible. This compatibility only happens if both ends agree to use the same protocol. All manufacturers who claim to be producing industry standard voice over IP either support SIP or H.323 protocol.
H.323
Over the next few years, the industry will address the bandwidth limitations by upgrading the Internet backbone to asynchronous transfer mode (ATM), the switching fabric designed to handle voice, data, and video traffic. Such network optimization will go a long way toward eliminating network congestion and the associated packet loss. The Internet industry also is tackling the problems of network reliability and sound quality on the Internet through the gradual adoption of standards. Standards-setting efforts are focusing on the three central elements of Internet telephony: the audio codec format; transport protocols; and directory services.
H.323 Call Sequence :

As such, H.323 addresses the core Internet-telephony applications by defining how delay-sensitive traffic, (i.e., voice and video), gets priority transport to ensure real-time communications service over the Internet. (The H.324 specification defines the transport of voice, data, and video over regular telephony networks, while H.320 defines the protocols for transporting voice, data, and video over integrated services digital network (ISDN).

How VoIP works part 3: Encoding
The call control part of H.323 sets up the parameters for the full duplex voice path between source telephone and destination telephone. I will continue with my analogies to explain how your voice gets transported across the Internet.
In terms of H.323 there is a trade off between call quality and bandwidth, in general the higher the quality the greater the bandwidth required
During the call setup portion of H.323 the phones have to decide which speech encoder/decoder to use when they send the speech to the other phone, Bill and Sally both have phones that support G.723.1, G.711 and G729.
The main difference between each of these encoders is the amount of bandwidth they use, G.711 uses 64kbit/s and G.723.1 can use as little as 5.3kbit/s. Although it would seem obvious to use the encoder with the lowest bandwidth, there is a loss of quality with a lower bandwidth.. At the same time a stream of G723.1 encoded voice data starts being sent from each phone to the other phone.
How VoIP works part 4 :Hear the Quality.
The performance of the speech encoders at each end, the number of packets lost on route, Latency and Jitter.
In the early days of voice calls via satellite there would be an annoying echo. As the technology improved the echo disappeared. Echo suppression is very key to good quality VoIP calls. I do not dwell on the subject since the mathematics is beyond my comprehension. Good echo suppression makes for quality calls.
Be warned that because a manufacturer has a G.723.1 encoder it may not sound the same as another manufacturer who claims to have G.723.1, quality does vary. As a general rule the occasional lost packet will not affect too drastically the quality of a call, but lose 5 in a row and an entire word is lost and this will be a problem. So if you are going to have lost packets make sure they are only lost in a regular distributed manner. 5% lost packets distributed evenly will not result in the loss of words lose 5% of the words by clustering the packets and the effect is bad.
PROS AND CONS :
Advantages of VoIP
There are many advantages to be gained from implementing an IP Telephony solution within the organization. The following list aims to highlight some of the advantages of such a strategy:
Single network infrastructure. When installing VoIP in the office only a single cable is required to the desk, for both telephone and data. Eliminating separate telephone wiring.
VoIP uses "soft" switching which eliminates most of the legacy PBX equipment. Reducing the cost of installing a communications infra-structure and the maintenance cost once installed.
Simple upgrade path. The VoIP PBX technology is software based. It is easier to expand, upgrade and maintain than its traditional telephony counterparts.
Bandwidth efficiency. VoIP can compress more voice calls into available bandwidth than legacy telephony.. IP Telephony helps to eliminate wasted bandwidth by not transporting the 60% of normal speech which is silence
Only one physical network is required to deal with both voice/fax and data traffic instead of two physical networks. Having only one physical network has the following advantages:
lower physical equipment cost ,lower maintenance costs.

Weaknesses:
While there are many aspects of VoIP which provide considerable benefits, the technology is still very young and problems remain. The following section looks at some of the weaknesses of this technology and their consequences.
The Internet is not the best medium for real time communications. Individual packets can take different routes and varying delays can be encountered and packets lost in transit. Waiting for delayed packets or retransmission of lost packets can result in considerable degradation of quality. Long delays in transit can affect quality so much that the technology can become unusable, though many vendors do have solutions which aim to negate the degradation suffered due to transit delays.
While some standards have been set by the ITU, the technology is not fully standardized and there is no guarantee that products from different vendors will be interoperable. Some vendors are trying to resolve this problem by forming groups and making guarantees about the products in the group but this is only a partial solution - vendors outwits the group cannot guarantee interoperability.
Heavy congestion on the network can result in considerable degradation of service as IP is not good at providing QoS (Quality of Service) guarantees. Feedbacks to Lucent Technologies customers reflect this worry. Major companies are planning to install IP Telephony capabilities at some point and have carried out initial investigations.
Since only one physical network for both data and voice/fax transmissions is required, failure of the network could be catastrophic, as all communications capabilities are lost.

Opportunities
Many vendors offer the ability to incorporate Virtual Private Networking (VPN) with relative ease into the IP Telephony solutions they provide. This allows any transmission to be encrypted using a number of cryptographic techniques and providing security by transmitting the communications through a 'tunnel' which is set up using PPTP (Point-to-Point Tunneling Protocol) before commencing communications.
IP Telephony allows companies to exploit Computer Telephony Integration to its full extent.
The convergence of communications technologies allows greater control over communications; most vendors provide logging and accounting facilities whereby all usage can be monitored.


Conclusion :
Without a doubt, the data revolution will only gain momentum in the coming years, with more and more voice traffic moving onto data networks. Vendors of voice equipment will continue to develop integrated voice and data devices based on packetized technology. Users with ubiquitous voice and data service integrated over one universal infrastructure will benefit from true, seamless, transparent interworking between voice and all types of data.

BIBLIOGRAPHY

1. Computer Networks by Andrew S.Tanenbaum
2. Internetworking with TCP/IP by Douglas E.comer
3. WWW.WIKIPEDIA.COM
3. iec.org.com
4. telogy.com
5. rad.com
6. http://seminarsprojects.in

please send this report to me

This article is presented by:
Andrew Quinn

Implementing SIP for VoIP Algorithms
JM-3

ABSTRACT
Internet Protocol (IP) Telephony has many issues that have to be overcome before it can be considered a rival to the existing telephony infrastructure. One such issue is the quality of service. The use of play-out buffering at the receiver helps to improve the quality of Voice over IP (VoIP). A buffering algorithm has been proposed by Narbutt, which uses a dynamic approach to buffering. This algorithm is adjusted automatically according to an estimate of the network delay. This is more suitable to the changing network conditions usually experienced. The buffer has been implemented using the H.323 signalling protocol.
The aim of this project is to incorporate Narbutt’s adaptive buffering algorithm into the Session Initiation Protocol (SIP). SIP has been shown to be much easier to implement and update than H.323. The integration of the algorithm was doing using VOCAL, a VoIP software library based around SIP. This report describes IP telephony and the protocols surrounding it, and the software used is also described. The manipulation of VoIP software to implement the play-out buffer and the issues involved in doing this are discussed.
INTRODUCTION
The ability to communicate properly over long distances has become an integral part of society today. Businesses are expanding to different regions in the world, but need to keep the same deadlines. This means it is necessary for employees in two different regions to communicate with each other over long distances, cheaply and trouble free. The public switched telephone network (PSTN) has developed itself to accommodate these requirements.

The internet has become a very popular means of communication in a very short period of time. It was set up as a network where people could share files and access other peoples work. It has since established itself as a massive communications infrastructure that provides many services such as electronic mail. In the recent years it has further developed itself into providing Internet Telephony or Voice over Internet Protocol (VoIP). This allows users to make voice or video calls over the internet. All the user needs is a computer with a network connection, a soundcard, and a microphone. VoIP enables a lot of big companies to combine their communications and their networking infrastructures. This is the biggest advantage that VoIP has over the regular telephone system. It means that voice and multimedia services are joined together. This means that a number of calls can be made on the one line, as well as having a multimedia broadcast. The fact that you are putting elements that would use one line each, down a single line, means that costs are significantly cut in the management and leasing of lines. There is even no need to change the communication infrastructure that already exists in the company. The companies PABX (private automatic branch exchange) only has to connect to a VoIP gateway, so IP calls can be made. Although VoIP seems to be taking off more with the corporate market the emergence and interest of the general public with Broadband should mean that IP telephony service could soon be implemented to its full extent in the home environment.


For more information about this article,please follow the link:
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What is VoIP
VoIP (voice over Internet protocol) describes the technology that enables consumers to make phone calls over the Internet instead of through a traditional landline or cell phone telecommunications network. Some of the benefits of VoIP technology include saving money on long distance phone calls, voice mail that can be received as e-mail messages, and the option of adding a toll-free number that will reach the user's main number.
VoIP technology has grown in recent years because small business customers and consumers are clamoring for this technology because of its easy-to-use and sophisticated features that surpass those of traditional phones, its software upgrade potential, and its bandwidth efficiency. Examples of some cost-efficient residential VoIP services include Vonage, Packet8 and Skype, while small business customers have favored Skyy Consulting and GalaxyVoice. Additionally, several large cable companies and telcos, such as Time Warner and AT&T, are offering VoIP services, but they are calling them digital phone services.
According to a new study by Info-Tech Research analyst George Goodall, the rapid adoption of VoIP is siphoning off traditional telephony, with 50 percent of small-to mid-sized enterprises (SMEs) expecting to rely on VoIP by 2008. He expects the majority of these SMEs to have converted at least part of their telephony networks to VoIP within the next five years.
History
• 1974 — The Institute of Electrical and Electronic Engineers (IEEE) published a paper titled "A Protocol for Packet Network Interconnection."
• 1981 — IPv4 is described in RFC 791.
• 1985 — The National Science Foundation commissions the creation of NSFNET.
• 1995 — VocalTec releases the first commercial Internet phone software.
• 1996 —
o ITU-T begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the H.323 standard. US telecommunication companies petition the US Congress to ban Internet phone technology.
• 1997 — Level 3 began development of its first softswitch, a term they coined in 1998.
• 1999 —
o The Session Initiation Protocol (SIP) specification RFC 2543 is released.
o Mark Spencer of Digium develops the first open source Private branch exchange (PBX) software (Asterisk).
• 2004 — Commercial VOIP service providers proliferate.
• 2005 — OpenSER (later Kamailio and OpenSIPS) SIP proxy server is forked from the SIP Express Router.
• 2006 — FreeSWITCH open source software is released.
How VoIP works
If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls. VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.
How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you're bypassing the phone company (and its charges) entirely.
•
VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems. VoIP providers like Vonage have already been around for a while and are growing steadily. Major carriers like AT&T are already setting up VoIP calling plans in several markets around the United States, and the FCC is looking seriously at the potential ramifications of VoIP service.
Above all else, VoIP is basically a clever "reinvention of the wheel." In this article, we'll explore the principles behind VoIP, its applications and the potential of this emerging technology, which will more than likely one day replace the traditional phone system entirely.
The interesting thing about VoIP is that there is not just one way to place a call. There are three different "flavors" of VoIP service in common use today:
• ATA -- The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it's a very straightforward setup.
• IP Phones -- These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Wi-Fi phones allow subscribing callers to make VoIP calls from any Wi-Fi hot spot.
• Computer-to-computer -- This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.
If you're interested in trying VoIP, then you should check out some of the free VoIP software available on the Internet. You should be able to download and set it up in about three to five minutes. Get a friend to download the software, too, and you can start tinkering with VoIP to get a feel for how it works.

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Abstract:-
The Voice-Over Internet Protocol (VoIP) technology allows the voice information to pass over IP data networks. This technology results in huge savings on the amount of physical resources required to communicate by voice over long distance. It does so by exchanging the information in packets over a data network.
The basic functions performed by a VoIP include – signalling, data basing, call connect and disconnect, and coding/decoding. The steps involved in originating and internet telephone call are the conversion of the analogue voice signal to digital format and compression/translation of the signal into internet protocol (IP) packets for transmission over the internet; the process is reversed at the receiving end. VoIP software’s like Vocal TEC or Net 2 Phone are available for the user. With the exception of phone to phone, the user must posses an array of equipment which should at minimum include VoIP software, an internet connection, and a multimedia computer with a sound card, speakers, a microphone and a modem.
The VoIP network acts as a gateway to the existing PSTN network. This gateway forms the interface for transportation of the voice content over the IP networks. Gateways are responsible for all call origination, call detection, analogue to digital conversion of voice, and creation of voice packets.
Introduction:-
The development of very fast, inexpensive microprocessors and special-purpose switching chips, coupled with highly reliable fibre-optic transmission systems, has made it possible to build economical, ubiquitous, high-speed packet-based data networks. Similarly, the development of very fast, inexpensive digital signal processors (DSPs) has made it practical to digitise and compress voice and fax signals into data packets. The natural evolution of these two developments is to combine digitised voice and fax packets with packet data, creating integrated data-voice networks. The voice-over-Internet protocol (VoIP) technology allows voice information to pass over IP data networks. Primarily, the cost savings that accrue from operating a single, shared network have motivated this convergence of telecommunications and data communications working in the diagram given below :
This short note offers a quick overview of voice communications using internet technology to carry the voice signal. The purpose is to assist a public policy decision
maker who wishes to understand some of the current debate regarding VoIP. The main
point is: the phrase voice over internet protocol (VoIP) is deceptively simple. VoIP
refers to dozens of activities that differ greatly in their quality, cost, and relationship to
traditional regulatory boundaries.
VoIP Technology in a Nutshell
Data communications over the Internet relies on dozens of protocols—rules defining the
required steps for a communications task. For example, part of the protocol for telephone
calls includes (1) waiting for dial tone before dialing and (2) the different sounds that
indicate that the called telephone is either ringing or busy. The internet protocol (IP) is
the basic protocol at the heart of the Internet. An IP message is the data communications
equivalent of a postcard—it carries the recipient’s and sender’s addresses, a block of data,and little else. Other protocols, such as TCP and HTTP, build on IP to create additional capabilities.
IP is a flexible building block—it has become the data communications equivalent of
2by4s and plywood—one can build anything using IP. IP is often used for communications that never touch the Internet—such as communications between a
computer and a nearby printer or communications within an organization’s private
network.
Naturally enough, one kind of data that can be carried by IP is digitally encoded voice.
Thus, voice over IP or VoIP. But, VoIP is not just one thing—it is many different things.
The differences arise from many sources, including the nature of the network used to
carry VoIP, the terminal equipment used to generate the VoIP signals, and the software
used to provide the VoIP connection.
Network Differences
Using VoIP over an unreliable network or a network with excessive delay results in poor
voice quality. In contrast, VoIP over an uncongested, minimal-delay network can
provide voice quality as good as (or even better than) that provided by traditional
telephone networks.
VoIP connections over the public Internet encounter a wide range of network conditions.
Sometimes such connections work well; other times they perform poorly.
VoIP over a network designed and operated to provide high-quality connections, such as
a corporate data network or a lightly used local area network, provides connections that
sound just as good as traditional telephone service. Similarly, if a telecommunications
carrier chooses to use VoIP in providing telephone service, the carrier can manage its
network so that voice quality matches that of traditional telephone service.
Terminal Equipment Differences
The terminal equipment used with VoIP connections affects the speech quality and
calling experience. At one extreme is the home computer with a microphone and
speakers. Generally speaking, these systems provide poor speech quality because of
problems with echoes and volume adjustment. Replacing the speaker and microphone
with a headset alleviates many of these problems.
The other extreme is a telephone designed to plug into an Ethernet local area network.
Cisco sells the Cisco IP Phone 7970G, shown in the figure below.
This unit is hearing aid compatible, and the dial pad meets ADA requirements.3 It can be
used as a PBX extension with no VoIP connection to the outside world, or it can connect
over a data network to other VoIP phones.
Between these two extremes is the integrated access device—an adapter that allows a
traditional phone to appear to a data network as a VoIP device.
Software Differences
VoIP software can be specialized for telephone-like conversations, or it may have a voice
capability that is only incidental to its primary purpose. AOL’s Instant Messenger
software has a talk button. Selecting the talk button with the mouse and clicking on it
begins the process of establishing a voice connection in parallel with an existing text chat
session. As far as I can tell, hardly anybody uses this capability of AOL Instant
Messenger but it may be a valuable option for individuals that have difficulty reading.
Microsoft sells a combined hardware/software product called Microsoft Sidewinder
Game Voice that provides both the ability to use voice commands to control computer
games and a voice communications capability among the players of a multiplayer game.
That voice capability permits groups of players to coordinate attacks on other players and
allows players to taunt their opponents.
The Cisco phone described above contains software that permits it to function like a
traditional telephone with voice mail, selectable ring tones, etc.
Gateways and Telephone Numbers
For many VoIP applications such as voice communications among the players of a
computer game, there is no need for connections to the traditional telephone network.
But VoIP applications that provide telephone service need the ability to connect to the
existing public switched telephone network (PSTN). The connecting points, which allow
traffic to flow between the VoIP network and the PSTN, are called gateways. Gateways
allow VoIP telephones to receive calls dialed to telephone numbers and permit VoIP
telephones to place calls to traditional telephones. That is, a gateway permits a telephone
number to be associated with a specific VoIP user. Gateway service is not a built-in
element of Internet service. Rather, gateway service is usually purchased separately from
suppliers such as Vonage or BellSouth.
Examples of Uses of VoIP
The discussion above provides some examples of VoIP use—such as that by computer
gamers to enhance their game playing or as part of AOL Instant Messenger.
Microsoft’s Windows Messenger software provides voice, video, whiteboard, program
sharing, file sharing, and text chat capabilities to users with compatible software and
connected by an appropriate network such as the public Internet or a private corporate
network. Windows Messenger is provided with the Windows XP operating system and is
also available for the Apple Macintosh. Thus, the vast majority of desktop computers
have installed on them software that provides voice communications over the Internet.
Hardly anybody uses those capabilities today.
Vonage provides a VoIP service that resembles traditional telephone service. However,
uses must “bring their own access.” That is, Vonage provides gateway service, but
Vonage’s customers must arrange for their own access to the public Internet.5 Calls are
carried over the public Internet from the user’s premises to the Vonage Gateway.
Vonage customers have phone numbers and can make calls to and receive calls from
traditional telephones.
In December 2003, Cox Cable began providing voice telephone service in Roanoke,
Virginia, using VoIP technology. In the Roanoke system Cox Cable provides the access
connection, using cable modems, as well as providing the gateway.6 The service
connects to the existing inside wiring in the home and is intended to appear to the
consumer as a direct substitute for the telephone service provided by the local telephone
company. The Cox Cable service uses the customer’s own telephones and provides
common telephone services such as call waiting, caller ID, and 911 service.

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Introduction
Companies and organizations around the world want to reduce rising communication costs. The consolidation of separate voice and data networks offers an opportunity for significant reduction in communication costs. Accordingly, the challenge of integrating voice and data networks is becoming a rising priority for network managers. Since data traffic is growing much faster than telephone traffic, a need has been identified to transport voice over data networks, as opposed to the transmission of data over voice networks. This brought about the rise for Voice over IP (VoIP). VoIP has become especially attractive given the low-cost, flat rate pricing of the public Internet. Many components will have to be designed to accommodate voice over data networks, such as the access gateways that page link the data and the telephony networks among others. Applications that offer Voice over IP services will have to include a comprehensive technology set that reduces the impairments caused by sending voice over data networks that were not designed to handle it. An important factor to be considered by network designers is the problem of Quality of Service (QoS). This is because IP is a best effort service and therefore provides no guarantees on delivery and data integrity. Voice processing will need to handle greater and variable delays, jitter, and cancel echoes that will be introduced from the telephony side. It will also have to include an appropriate algorithm to mask the gaps caused by dropped packets due to congestion on the network. A protocol needs to be implemented which guarantees bandwidth for the duration of a session and also better compression technologies need to be put in place. An understanding of how to handle call set up translation for different types of networks, connections, and internetworking is essential for competent handling of every call. A Voice over IP application meets the challenges of combining legacy voice networks by allowing both voice and signaling information to be transported over IP.
Advantages and Benefits of Voice over IP
One of the main reasons and probably the most significant interest in the race to send voice over IP is the cost advantage that this process offers organizations due to the flat rate, low cost of Internet traffic. Generally the benefits of technology can be divided into three categories:
1)Cost reduction:- Although reducing long distance telephone costs is always a popular topic and provides a good reason to introduce VoIP, the actual savings over a long term are still under scrutiny and debate. These savings from lower prices are however, based on avoiding telephony access charges and settlement fees, rather than actually reducing resource costs. The sharing of equipment and operations costs across both data and voice users can also improve network efficiency, since excess bandwidth on one network can be used by the other, thereby creating economies of scale.
2)Simplification:- An integrated infrastructure that supports all forms of communication allows more standardization and reduces the total equipment complement. The economies of putting all forms of traffic over an IP based network will pull companies in this direction, simply because IP will act as the unifying agent regardless of the underlying architecture. This combined infrastructure can support dynamic bandwidth optimization and a fault tolerant design.
3)Consolidation:- People are the most significant cost elements in a network, so any opportunity to combine operations and eliminate points of failure and to consolidate accounting systems would be beneficial. In the enterprise, SNMP based management with the appropriate MIB structures can be provided for both voice and date services using VoIP. Universal use of the IP protocol for all applications will reduce complexity and provide more flexibility.
Transmitting Voice over Internet Protocol is also beneficial for the following reasons:-
1)Since the Internet is a packet switched or "connectionless" network, the individual packets of each voice signal travel over separate network paths for reassembly in the proper sequence at their ultimate destinations. This makes for a more efficient use of network resources and more reliability than the circuit switched PSTN.
2)Private voice networks require n (n-1) access links. Whereas private data networks require only ‘n’ access links.
3)Voice has per-minute distance sensitive charge, whereas data on the other hand has flat time-sensitive charges.
4)Data transmission has no 64 kbps bandwidth limitation, which means that we can provide high fidelity voice transmissions very easily.
Applications implementing VoIP
1) PSTN gateways: Interconnection of the Internet to the PSTN can be accomplished using a gateway. A PC-based telephone for example would have access to the public network by calling a gateway point close to the destination.
2) nternet-aware telephones: Ordinary telephones can be enhanced to serve as an Internet access device as well as providing normal telephony. Inter-office trunking over the corporate intranet: replacement of tie trunks between company owned PBX’s using an Intranet page link would provide for economies of scale and help to consolidate network facilities.
3) Remote access from a branch office: A small office could gain access to corporate voice, data, and facsimile services using the companies Intranet services.
4) Voice calls from a mobile PC via the Internet: Calls to an office can be achieved using a multimedia PC that is connected via the Internet. One example would be using the Internet to call from a hotel instead of using expensive hotel telephones.
5) Internet call center access: Access to call center facilities via the Internet is emerging as a valuable adjunct to electronic commerce applications. Internet call center access would enable a customer who has questions about a product being offered over the Internet to access customer service agents online.
Voice Over IP Design and Development Challenges
The goal of VoIP developers is to add telephone calling capabilities (both voice transfer and signaling) to IP based networks and interconnect these to the public telephone network and to private voice networks, in such a way as to maintain current voice standards and preserve the features everyone expects from the telephone. VoIP development needs to take place in five specific areas:
1)Voice quality should be comparable to what is available using the PSTN, even over networks having variable levels of QoS.
2)The underlying IP network must meet strict performance requirements and criteria including minimizing call refusals, network latency, packet loss, and disconnects. This is required even when there is heavy congestion in the network or when resources have to be shared among multiple users.
3)Call control (the actual signaling) should be done transparent to the user in such a way that they should be unaware of what technology is actually implementing the service.
PSTN/ VoIP service internetworking (and equipment interoperability) involves gateways between the voice and data network environments.
4)System management, security, addressing, and accounting must be provided, preferably consolidated with the PSTN operation support systems.
Quality of Service Issues in IP Networks
The advantages of reduced cost and bandwidth savings of carrying voice over data networks are associated with some Quality of Service (QoS) issues unique to packet networks. Delivering quality voice signals from one point to another cannot be considered successful unless the quality of the delivered signal satisfies the recipient. Providing a level of quality that at least equals the PSTN (this is usually referred to as "toll quality voice") is viewed as a basic requirement. Although QoS usually refers to the fidelity of the transmitted voice and facsimile document it can also be applied to network availability, telephone feature availability, and scalability. Many factors have been identified that play a big role in determining the quality of service. They are as follows:
1) Delay
Two problems that result from high end-to-end delay in a voice network is echo and talk over lap. Echo is caused by signal reflections of the speaker’s voice from the far end telephone equipment back into the speaker’s ear. Echo becomes a problem when the round trip delay exceeds 50 milliseconds. Since echo is perceived as a significant quality problem, Voice over IP systems have to address the need for echo control and implement means for echo cancellation. Talkers overlap is the problem of one caller stepping on the other talker’s speech. This becomes significant if the one-way delay becomes greater than 250 milliseconds. Delay can be subdivided into two sub-components. They can be fixed delay components as well as variable delay components. Fixed delay components include propagation, serialization, and processing. The variable delay components include the queuing delay, jitter buffers as well as variable packet sizes.
2)Jitter (Delay Variability)
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 in turn causes additional delay. The conflicting goals of minimizing delay and removing jitter has led to the development of various schemes to adapt the jitter buffer size to match the time varying requirements of network jitter removal.
3) Packet Loss
IP networks cannot provide a guarantee that packets will be delivered at all, much less in order. Packets will be dropped under peak loads and during periods of network congestion. But due to the time sensitivity of voice transmissions, however the normal TCP- based retransmission schemes are not suitable. Approaches that compensate for packet loss have to be developed to overcome this problem.
4) Bandwidth Availability
Bandwidth is the portion of the network that is available to an application to transfer information on the network. The level of reliability and sound quality that is acceptable among users has not yet been reached and this is primarily because of bandwidth limitations and this also leads to packet loss. In voice communications, packet loss shows up in the form of gaps or periods of silence in the conversation, thus leading to a "clipped speech" effect that is unsatisfactory for most users and unacceptable in business communications.
Proposed solutions for problems associated in sending Voice over IP
Maintenance of acceptable voice quality levels despite inevitable variations in network performance is achieved using a variety of techniques. These techniques and solutions to problems that have been detailed above with regard to transmission of voice over IP are as follows.
1) One of the main problems of a very big end-to-end delay is the problem of echoes. The ITU standard G.165 defines performance requirements for echo cancellers. The way the echo cancellers work is that when the echoes are generated from the telephone network toward the packet network, the echo canceller compares the voice data received from the IP network to the voice data that is being transmitted to the IP network. The echo from the telephone network is removed by a digital filter on the transmit path to the IP network.
2) The approach to remove jitter involves counting the number of packets that arrive late and create a ratio of these packets to the number of packets that are successfully processed. This ration is then in turn used to adjust the jitter buffer to target a predetermined allowable late packet ratio. This approach works best with networks with highly variable packet and inter-arrival intervals such as IP.
3) Lost packets are a big problem in networks. Some schemes called lost packet compensation schemes used by voice over IP to overcome the problem of lost packets are as under.
a) Interpolate for lost speech packets by replaying the last packet received during the interval when the last packet was supposed to be played out. This works well when the incidence of lost frames is infrequent. It does not work very well for bursty loss of packets.
b) Another way is to send redundant information at the expense of bandwidth utilization. The basic approach replicates and sends the n’th packet of voice information along with the (n+1)th packet. This method has the advantage of being able to exactly correct for the lost packet. However, this approach uses more bandwidth and creates greater delay.
c) An alternative approach is to develop an algorithm in the digital signal processor that detects missing packets, and then replays the last successfully received packet at a decreased volume in order to fill the gaps.
d) Another problem is that of Out of Order Packets. When an out of order condition is detected in the network, the missing packet is replaced by its predecessor, as if it was lost. 6. Software Support to enable Voice over IP

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