Voice Portals and Speech Recognition system

Voice portals

In its most generic sense a voice portal can be defined as “speech enabled access to Web based information”. In other words, a voice portal provides telephone users with a natural language interface to access and retrieve Web content. An Internet browser can provide Web access from a computer but not from a telephone. A voice portal is a way to do that.

VUI (Voice User Interface)

For a voice portal to function, one of the most important technology we have to include is a good VUI (Voice User Interface).There has been a great deal of development in the field of interaction between human voice and the system. And there are many other fields they have started to get implemented. Like insurance has turned to interactive voice response (IVR) systems to provide telephonic customer self-service, reduce the load on call-center staff, and cut overall service costs. The promise is certainly there, but how well these systems perform-and, ultimately, whether customers leave the system satisfied or frustrated-depends in large part on the user interface.
Many IVR applications use Touch-Tone interfaces-known as DTMF (dual-tone multi-frequency)-in which customers are limited to making selections from a menu. As transactions become more complex, the effectiveness of DTMF systems decreases.
In fact, IVR and speech recognition consultancy Enterprise Integration Group (EIG) reports that customer utilization rates of available DTMF systems in financial services, where transactions are primarily numeric, are as high as 90 percent; in contrast, customers' use of insurers' DTMF systems is less than 40 percent.
Enter some more acronyms. Automated speech recognition (ASR) is the engine that drives today's voice user interface (VUI) systems. These let customers break the 'menu barrier' and perform more complex transactions over the phone. "In many cases the increase in self-service when moving from DTMF to speech can be dramatic," said EIG president Rex Stringham.
The best VUI systems are "speaker independent"-they understand naturally spoken dialog regardless of the speaker. And that means not only local accents, but regional dialects, local phrases such as "pop" versus "soda," people who talk fast (you know who you are), and all the various nuances of speech. Those nuances are good for human beings; they allow us to recognize each other by voice. For computers, however, they make the process much more difficult. That's why a hand held or pocket computer still needs a stylus, and why the 'voice dialing' offered by some cell-phone companies still seems high-tech.

Voice recognition is tough. And sophisticated packages not only can recognize a wide variety of speakers, they also allow experienced users to interrupt menu prompts ("barge-in") and can capture compound instructions such as "I'd like to transfer a thousand dollars from checking to savings" in one command rather than several.
These features are designed to not only overcome limitations of DTMF but to increase customer use and acceptance of IVR systems. The hope is that customers will eventually be comfortable telling a machine "I want to add a driver to my Camry's policy." Besides taking some of the load off customer service representatives, VUI vendors promise an attractive ROI to help get these systems into insurers' IT budgets.

ASR systems can be enabled with voice authentication, eliminating the need for PINs and passwords. Call centers themselves will likely transform into units designed to support customers regardless of whether contact comes from a telephone, the Web, e-mail, or a wireless device. At the same time, the 'voice Web' is evolving, where browsers or Wireless Application Protocol (WAP)-enabled devices display information based on what the user vocally asks for. "We're definitely headed toward multi-modal applications," Ehrlich predicts.
ASR vendors are working to make sure that VUI evolves to free staff from dealing with voice-related channels; it's better to have them supporting the various modes of service that are just now beginning to emerge.
.
Good VUI systems have multiple fallback strategies for speech recognition failure, such as asking callers to spell names or breaking a question into parts. The interface could revert to DTMF, especially if ASR was added to an existing DTMF-enabled system. Still, transferring an unintelligible customer to a human agent is frequently an early fallback option in order to minimize customer frustration.

Facilitating this evolution is XML's telephonic flavor, Voice XML. As its name implies, Voice XML is a markup language specification that defines the key elements of voice-enabled transactions, which also allows re purposing of this data across any number of platforms and systems

Next Generation Network Services

Today’s business environment is more competitive – and complex than ever before. Customer service is the key to success. And demand is growing for powerful new communications services as a strategic way to enhance customer service and build a competitive edge. At the center of these new services is the next-generation network (NGN).

What is Next generation network? It is the next step in the world communications traditionally enabled by three separate networks: the public switched telephone network (PSTN) voice network, the wireless network and the data network (the Internet). NGNs converge all three of these networks-voice, wireless and internet- into a common a common packet infrastructure. This intelligent, highly efficient applications and service opportunities.

Three types of services drive NGNs: real-time and non real-time communication services, content services, and transaction services. The services-driven NGN gives service provides greater control, security, and reliability while reducing their operating costs. Service providers can quickly and costs effectively build new revenue.

Built on open modular elements, standard protocols, and open interfaces, the NGN caters to the specific needs of all users - enterprises, remote offices, telecommuters, and small office / home office (SOHO) customers. It unites traditional wireline and wireless voice, video, and data using a packet-based transport. The new class of services it enables is more flexible, scalable, and costs efficient than services that have been offered in the past.

One key NGN service is voice portal, which provide callers with anywhere, anytime access to information like news, weather, stock quotes, and account balances using simple voice commands and any telephone. Voice portals are poised to become the next big thing in communications. The organizations using them will have a very real edge in the market place-differentiating themselves from their competitors, attracting loyal customers and growing their revenues.

Voice based service for the next-generation network; Intel Corporation has the hardware, software and services for the next generation network solutions, including a complete voice portal platform.

Next-generation Network Benefits

* More choices-Open systems promote multiple-vendor participation in the marketplace.
* Lower cost-Solutions built on open standards cut development time for solution providers.
* Innovative Services- Working to standards frees developers to concentrate on adding value.
* Lower Risk- Compatibility with products and technologies from many suppliers decreases the risk of ownership.

Deploying NGN Services

                                  Next-generation communications solutions are built with open, flexible, standards-based building blocks. The reasons are clear- with open solutions, there’s no need to start from scratch to add next-generation voice enabled E-business services. Using modular building blocks makes it easy and affordable to add new features, services, and value to existing systems. It all adds up to powerful, affordable solutions that protect your- and your customers – investments.

The NGN begins with media servers, which provide advanced media-processing capabilities. The value of a media server is its flexibility for supporting advanced media-processing services like basic voice announcements, interactive voice response (IVR), conferencing, messaging, text-to-speech (TTS), and speech recognition. Built with open, standard computing and voice-processing boards, media servers can be deployed many ways. For example, a voice portal platform is a media server that provides a speech driven user interface with simple speech recognition and TTS capability, providing voice-client access to Internet content, messages, or both.

Voice Recognition and Authorization

                                             Speech recognition and speaker authentication are often thought of as the same thing, but that this is a myth. Speech recognition's focus is on what is being said, and can not determine who is speaking. Speaker authentication's focus is on the speaker –“and does not know what is being said.”
Both, however, face similar hurdles. Markowitz said a handy rule is that "if it is a problem for speaker authentication, it is difficult for speech-recognition.”

Markowitz dispelled a second myth: that voice biometrics "can not resist tape recorders." But in fact, a challenge-response program for released criminals actually automates "which one of the questions it will ask you to repeat, so you never know what it is going to ask you say!" The challenge-response system phones home-arrestees, who are sentenced to house-arrest, and asks them to say something over the phone that they would not and could not know in advance - thus not being able to be duped by a pre-recorded taped response.

A third myth Markowitz dispels is that speaker authentication is not as accurate as other biometrics. "Voice is doing pretty darn well compared to other biometrics." She reminds us that "no form of security is 100 percent secure - but there are ways to combine belts and suspenders, and blend in partnership different kinds of techniques to create multi-level security." Here we would be talking only about the speaker verification that is speaker authentication and not speech recognition.

Speaker Verification

The speaker-specific characteristics of speech are due to differences in physiological and behavioral aspects of the speech production system in humans. The main physiological aspect of the human speech production system is the vocal tract shape. The vocal tract is generally considered as the speech production organ above the vocal folds, which consists of the following: (i) laryngeal pharynx (beneath the epiglottis), (ii) oral pharynx (behind the tongue, between the epiglottis and velum), (iii) oral cavity (forward of the velum and bounded by the lips, tongue, and palate), (iv) nasal pharynx (above the velum, rear end of nasal cavity), and (v) nasal cavity (above the palate and extending from the pharynx to the nostrils). The shaded area in figure 1 depicts the vocal tract.
The vocal tract modifies the spectral content of an acoustic wave as it passes through it, thereby producing speech. Hence, it is common in speaker verification systems to make use of features derived only from the vocal tract. In order to characterize the features of the vocal tract, the human speech production mechanism is represented as a discrete-time system of the form depicted in figure 1.

The acoustic wave is produced when the airflow from the lungs is carried by the trachea through the vocal folds. This source of excitation can be characterized as phonation, whispering, friction, compression, vibration, or a combination of these. Phonated excitation occurs when the airflow is modulated by the vocal folds. Whispered excitation is produced by airflow rushing through a small triangular opening between the arytenoid cartilage at the rear of the nearly closed vocal folds. Frication excitation is produced by constrictions in the vocal tract. Compression excitation results from releasing a completely closed and pressurized vocal tract. Vibration excitation is caused by air being forced through a closure other than the vocal folds, especially at the tongue. Speech produced by phonated excitation is called voiced, that produced by phonated excitation plus frication is called mixed voiced, and that produced by other types of excitation is called unvoiced.

It is possible to represent the vocal-tract in a parametric form as the transfer function H(z). In order to estimate the parameters of H(z) from the observed speech waveform, it is necessary to assume some form for H(z). Ideally, the transfer function should contain poles as well as zeros. However, if only the voiced regions of speech are used then an all-pole model for H(z) is sufficient. Furthermore, linear prediction analysis can be used to efficiently estimate the parameters of an all-pole model. Finally, it can also be noted that the all-pole model is the minimum-phase part of the true model and has an identical magnitude spectra, which contains the bulk of the speaker-dependent information.
The above discussion also underlines the text-dependent nature of the vocal-tract models. Since the model is derived from the observed speech, it is dependent on the speech. Figure 2 illustrates the differences in the models for two speakers saying the same vowel.

Choice of features:
The LPC features were very popular in the early speech-recognition and speaker-verification systems. However, comparison of two LPC feature vectors requires the use of computationally expensive similarity measures such as the Itakura-Saito distance and hence LPC features are unsuitable for use in real-time systems. Furui suggested the use of the cepstrum, defined as the inverse Fourier transform of the logarithm of the magnitude spectrum, in speech-recognition applications. The use of the cepstrum allows for the similarity between two cepstral feature vectors to be computed as a simple Euclidean distance. Furthermore, Atal has demonstrated that the cepstrum derived from the LPC features results in the best performance in terms of FAR and FRR for a speaker verification system. Consequently, we have decided to use the LPC derived cepstrum for our speaker verification system.
Speaker Modeling:
Using cepstral analysis as described in the previous section, an utterance may be represented as a sequence of feature vectors. Utterances spoken by the same person but at different times result in similar yet a different sequence of feature vectors. The purpose of voice modeling is to build a model that captures these variations in the extracted set of features. There are two types of models that have been used extensively in speaker verification and speech recognition systems: stochastic models and template models. The stochastic model treats the speech production process as a parametric random process and assumes that the parameters of the underlying stochastic process can be estimated in a precise, well defined manner. The template model attempts to model the speech production process in a non-parametric manner by retaining a number of sequences of feature vectors derived from multiple utterances of the same word by the same person. Template models dominated early work in speaker verification and speech recognition because the template model is intuitively more reasonable. However, recent work in stochastic models has demonstrated that these models are more flexible and hence allow for better modeling of the speech production process. A very popular stochastic model for modeling the speech production process is the Hidden Markov Model (HMM). HMMs are extensions to the conventional Markov models, wherein the observations are a probabilistic function of the state, i.e., the model is a doubly embedded stochastic process where the underlying stochastic process is not directly observable (it is hidden). The HMM can only be viewed through another set of stochastic processes that produce the sequence of observations. Thus, the HMM is a finite-state machine, where a probability density function p(x | s_i) is associated with each state s_i. The states are connected by a transition network, where the state transition probabilities are a_ {ij} = p (s_i | s_j). A fully connected three-state HMM is depicted in figure 3.
For speech signals, another type of HMM, called a left-right model or a Bakis model, is found to be more useful. A left-right model has the property that as time increases, the state index increases (or stays the same) -- that is the system states proceed from left to right. Since the properties of a speech signal change over time in a successive manner, this model is very well suited for modeling the speech production process.


Pattern Matching:
The pattern matching process involves the comparison of a given set of input feature vectors against the speaker model for the claimed identity and computing a matching score. For the Hidden Markov models discussed above, the matching score is the probability that a given set of feature vectors was generated by the model.

Speech Recognition Challenges


Bell labs designed the first speech-recognition system in the early 1950s.the technology was amazingly accurate at determining single digit spoken numbers. By early 1970s natural language speech understanding was demonstrated by Terry Winograd’s SHRDLU system, a robot that understood commands such as “move the red block on top of the small green one.” By the 1990s speech recognition was being applied to internet. In 1996 California State University at Northridge was able to demonstrate an experimental voice controlled web interface. By 2000 many voice driven portals were being developed.
Until recently, a number of issues made it almost impossible to develop a speech-recognition engine that would recognize fluent and natural speech. The basic challenges faced by voice application developers included the following:
1. Variability of speech patterns: Different people speak the same language differently and even speak the same word in many different ways. Interpreting speech variability has led to the development of complex pattern analysis. Understanding natural pauses, speaking rates and changes in volumes has been complex and difficult.
2. Processing power: In the mid-1980s, a new technique known as Hidden Markov’s Models improved the ability to recognize word relationships. This computation intensive technology eventually led to powerful speech recognition applications. Achieving real time voice recognition found in voice portals requires processing power that is not commercially viable until recently.
3. Extracting meaning: Very few speech Recognition applications are able to accurately determine the meaning of words. The quality of speech interpretation depends on the ability of the speech recognition engine to properly choose the best match for spoken sounds from its list of expected text phrases. A more advanced process was required to extract meaning from those words. Because of the many possible ways that people speak and the many words that are used to communicate the same concept, a full understanding of human factors was critical to properly interpret the meaning.
4. Background Noise: Before mobile phone users often access voice portals, background has been difficult for voice recognition developers to filter out The development of better microphones has helped, but issues such as wind, murmurs and music have made it a challenge to properly isolate voice from noise.
5. Continuous speech recognition: Designing systems that are powerful enough to understand and respond to continuous speech requires a large amount of processing power that was not available at reasonable cost in the past. When a person speaks at a natural rate, it has been difficult to distinguish which sounds were associated with specific words. For example the phrase “to recognize speech” could have been misunderstood to be “to wreck a nice beech.” Users do not naturally speak with pauses between words .As a result processing phrases in real time as they are naturally spoken has been a major challenge.
Many of the solutions to problem associated to speech recognition
are still being fine tuned. Powerful computers have provided the processing power to overcome many of the limitations of the past. However the most common speech-recognition systems of today are still very different from the way in which people naturally interact.

Speech-enabled Internet portals or voice portals are quickly becoming the hottest trend in E-commerce – broadening the access to internet content to everyone with the most universal communications device of all, a telephone. Voice portals purt all kinds of information at a customers fingertips anytime, anywhere. Customers just dial into the voice portal’s prescribed number and they use simple voice commands to access whatever information they need. It’s quick, easy and effective, even from a car or the airport.

The potential for voice portals is as wide as the reach of telephones, which today number 1.3 billion around the world. Compare that to the 250 million computers with Internet access and it is easy to understand while analysts believe voice-enabled web access will take off. Feasibility studies exhibit that by 2005, 45 million wireless users around the world will regularly use voice portals to handle their everyday cyber chores

Privacy Policy

Privacy policy for http://latesttechnologyarticles.blogspot.com/
Your privacy is important to us. We are committed to safeguarding your privacy while online at our site latesttechnologyarticles.blogspot.com. The following discloses the information gathering and dissemination practices for this Web site.
The Information We Collect
       We use statcounter, google analytics and other hosting service tools to collect the information. This includes internet protocol (IP) addresses, browser type, internet service provider (ISP), referring/exit pages, platform type, date/time stamp, and number of clicks to analyze trends, administer the site, track user's movement in the aggregate, and gather broad demographic information for aggregate use. IP addresses, etc. are not linked to personally identifiable information. The name and website URl are displayed on comments, but email address is secure with us. We do not share it with outside parties
Cookies
     A cookie is a piece of data stored on the user's computer tied to information about the user. Our site don't use cookies. However, Some of our business partners use cookies on our site (for example, advertisers). However, we have no access to or control over these cookies, once we have given permission for them to set cookies for advertising.
     Google’s use of the DART cookie enables it to serve ads to you based on your visit to this site and other sites on the Internet. You may opt out of the use of the DART cookie by visiting the Google ad and content network privacy policy.
Newsletter
If a user wishes to subscribe to our newsletter (powered by Feedburner), we ask for contact information such as name and email address. Out of respect for our users privacy we provide a way to opt-out of these communications.
Links
This Web site contains links to other sites. Please be aware that we are not responsible for the privacy practices of such other sites. We encourage our users to be aware when they leave our site and to read the privacy statements of each and every Web site that collects personally identifiable information. This privacy statement applies solely to information collected by this Web site.
Copy Limitations
Please don't copy the whole post. A short paragraph with content in quotes is acceptable with a link-back to the post at www.latesttechnologyarticles.blogspot.com/
Link Back
You need to link back to the article from which you derived the excerpt.
Advertisers
We use outside ad companies to display ads on our site. These ads may contain cookies and are collected by our ad companies, and we do not have access to this information. We work with the ad companies: Google Adsense, Adbrite, Axill, Komli etc. Please check their websites for respective privacy policies.
Contact Information
If users have any questions or suggestions regarding our privacy policy or our post, please Contact us.

Contact Us

VOICE OVER INTERNET PROTOCOL(VoIP) - IP TELEPHONY

VoIP or “Voice over Internet Protocol” refers to sending voice and fax phone calls over data networks, particularly the Internet. This technology offers cost savings by making more efficient use of the existing network.
Traditionally, voice and data were carried over separate networks optimized to suit the differing characteristics of voice and data traffic. With advances in technology, it is now possible to carry voice and data over the same networks whilst still catering for the different characteristics required by voice and data.

Voice over Internet Protocol

Voice-over-Internet-Protocol (VoIP) is an emerging technology that allows telephone calls or faxes to be transported over an IP data network. The IP network could be

• A local area network in an office
• A wide area network linking the sites of a large international organization
• A corporate intranet
• The internet
• Any combination of the above

There can be no doubt that IP is here to stay. The explosive growth of the Internet, making IP the predominate networking protocol globally, presents a huge opportunity to dispense with separate voice and data networks and use IP technology for voice traffic as well as data. As voice and data network technologies merge, massive infrastructure cost savings can be made as the need to provide separate networks for voice and data can be eliminated.

Most traditional phone networks use the Public Switched Telephone Network(PSTN), this system employs circuit-switched technology that requires a dedicated voice channel to be assigned to each particular conversation. Messages are sent in analog format over this network.

Today, phone networks are on a migration path to VoIP. A VoIP system employs a packet-switched network, where the voice signal is digitized, compressed and packetized. This compressed digital message no longer requires a voice channel. Instead, a message can be sent across the same data lines that are used for the Intranet or Internet and a dedicated channels is no longer needed. The message can now share bandwidth with other messages in the network.

Normal data traffic is carried between PC’s, servers, printers, and other networked devices through a company’s worldwide TCP/IP network. Each device on the network has an IP address, which is attached to every packet for routing. Voice-over-IP packets are no different.

Users may use appliances such as Symbol’s Net-vision phone to talk to other IP phones or desktop PC-based phones located at company sites worldwide, provided that a voice-enabled network is installed at the site. Installation simply involves assigning an IP address to each wireless handset.

VOIP lets you make toll-free long distance voice and fax calls over existing IP data networks instead of the public switched telephone network (PSTN). Today business that implement their own VOIP solution can dramatically cut long distance costs between two or more locations.
Methods:
PC to PC,PC to Phone,Phone to Phone

WORKING of VoIP

In VoIP, analog voice signals are digitized using PCM. These digital voice samples are then buffered on an IP gateway. This device converts the PCM data stream into a compressed IP packet stream using DSP’s (Digital Signal Processors). DSP’s are responsible for converting from analog to digital as well as compression. The set of PCM samples are analyzed as a discrete set of binary data. It checks the speech for all the moments of silence, which are a lot. Even when we speak, there are pauses in between that go unnoticed to the human ear, but are quiet discernible to the sampling device. The length and beginning of these pauses is noticed, while the remaining silence is removed from the data set. Similarly redundant data is also

Removed, making the data set more compact. Finally, an IP header is attached to this compressed data, which is then sent out on the network as discrete data packets. Once the voice
Packet is sent out; it finds its way to the destination just like any other data packet. It passes through various routers and switches to reach the destination gateway. Here, it gets
Decompressed, meaning all the periods of silence and redundant data are reinserted and is finally decoded to produce an approximation of the original sound. The compression algorithm
Used in the process can compress the voice signals and can even carry voice over a little as 5.3 kbps bandwidth

APPLICATIONS OF VOICE OVER INTERNET PROTOCOL

Voice communications will certainly remain a basic form of interaction for all of us. The PSTN simply cannot be replaced, or even dramatically changed, in the short term (this may not apply to private voice networks, however). The immediate goal for VoIP service providers is to reproduce existing telephone capabilities at a significantly lower "total cost of operation" and to offer a technically competitive alternative to the PSTN. It is the combination of VoIP with point-of-service applications that shows great promise for the longer term. Some of the example are:
• PSTN gateways: Interconnection of the Internet to the PSTN can be accomplished using a gateway, either integrated into a PBX (the iPBX) or provided as a separate device. A PC-based telephone, for example, would have access to the public network by calling a gateway at a point close to the destination (thereby minimizing long distance charges).
• Internet-aware telephones: Ordinary telephones (wired or wireless) can be enhanced to serve as an Internet access device as well as providing normal telephony. Directory services, for example, could be accessed over the Internet by submitting a name and receiving a voice (or text) reply.
• Inter-office trunking over the corporate intranet: Replacement of tie trunks between company-owned PBXs using an Intranet link would provide economies of scale and help to consolidate network facilities.
• Remote access from a branch (or home) office: A small office (or a home office) could gain access to corporate voice, data, and facsimile services using the company's Intranet
(emulating a remote extension for a PBX, for example). This may be useful for home-based agents working in a call center, for example.
• Voice calls from a mobile PC via the Internet: Calls to the 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. This could be ideal for submitting or retrieving voice messages.
• 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. Another VoIP application for call centers is the interconnection of multiple call centers.

H.323

The H.323 standard provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service (QoS). These networks dominate today’s corporate desktops and include packet-switched TCP/IP and IPX over Ethernet, Fast Ethernet and Token Ring network technologies. Therefore, the H.323 standards are important building blocks for a broad new range of collaborative, LAN-based applications for multimedia communications. It includes parts of H.225.0 - RAS, Q.931, H.245 RTP/RTCP and audio/video codecs, such as the audio codecs (G.711, G.723.1, G.728, etc.) and video codecs (H.261, H.263) that compress and decompress media streams.


Media streams are transported on RTP/RTCP. RTP carries the actual media and RTCP carries status and control information. The signalling is transported reliably over TCP. The following protocols deal with signalling:
• RAS manages registration, admission, status.
• Q.931 manages call setup and termination.
• H.245 negotiates channel usage and capabilities.
• H.235 security and authentication.

VOICE CODING ALGORITHMS

There are several approaches to digitizing the voice samples. These approaches vary by the information that is transmitted, the complexity of the algorithm, and the assumptions of the sound being transmitted (e.g. voice, fax, music). Broadly classified, the various coding algorithms fall into two broad categories: coding of waveform and modeling of the vocal track. Pulse Code Modulation and Sub Band Coding are examples of waveform coding algorithms while Linear Predictive Coding is an example of an algorithm that models the vocal tract.

The Pulse Code Modulation (PCM) algorithm makes no assumptions about the sound that is being digitized and therefore does the best job on various types of sounds. It also produces the highest bit-rate for the data and has the shortest delay. The basics of the various PCM algorithms is that the voice is sampled at fixed time intervals (i.e. 8,000 times/second) and then a number is generated from the data based on each sample.

Figure 1. PCM algorithms sample the voice at fixed time intervals

ADPCM (Adaptive Differential Pulse Code Modulation), a variant of PCM, samples the voice at fixed time intervals and then calculates the change from the previous sample and sends that information. To save bandwidth, these step sizes are specially coded so that the step size at low volume is different than the step size at high volume. ADPCM provides about a 2:1 reduction in the data compared to PCM.

CODING STANDARDS

There are a number of voice coding standards and the ITU is the most active of the groups in this area. For information on the details of any of these standards, go to the ITU web site (http://www.itu.int/ITU-T).

Table 1 provides a summary of several of the major voice coding algorithms. As can be seen, there is a range of data rates available. The column labeled MOS (Mean Opinion Score), is a subjective score that listeners give to each of these algorithms. For point of reference, G.711 is what is used in the US phone system.

Algorithm	Bit Rate (Kbits/sec)	Complexity (Mips)	Delay (milliseconds)	MOS
G.711 PCM	64	< 1	.25	4.4
G.723.1 MPMLQ	6.3	18	30	3.9
G.723.1 ACELP	5.3	18	30	3.6
G.726 ADPCM	32	1	.25	4.2
G.728 LD-CELP	16	30	3 - 5	4.2
G.729a CS-ACELP	8	20	10	4.2
GSM	13.2	4.5	40	3.7

Compelling VoIP Applications

The VoIP technology only becomes useful when compelling applications meet the needs of customers. Three such applications are corporations that replace PBX systems, cable operators offering telephony services using their plant, and video conferencing. These existing applications are the driving factors in allowing manufactures to make equipment, service providers to offer services, and customers to increase their productivity. Having an established market and a profitable business model allows VoIP to begin addressing the next generation of applications which then will allow the market to continue to grow.

Corporate

LAN-based telephony applications offer attractive business models to consumers today. The most important applications are the replacement of the traditional PBX, new client side applications using the PC, and the reduction in maintenance expenses for wiring changes.

LAN based PBX systems provide superior return on investment to traditional PBX systems. Although initial equipment costs are comparable, LAN PBXs typically cost much less than PBXs to install because they use the existing data infrastructure (Category 5 cabling) rather than separate voice wiring. Administration is also less burdensome because LAN and server administrators can manage the system without the need for dedicated telephony technicians.

The most compelling reason businesses consider IP telephony-type applications is for the integration of applications with voice. Over the years, a significant amount of work has gone into computer telephony integration (CTI) in traditional PBXs. These systems began to offer application-programming interfaces such as Telephony API (TAPI), Telephony Services API (TSAPI), and Java Telephony API (JTAPI). This work has resulted in advanced call center functions, including screen pops for agents and active call routing between call centers.

Client (end station) products offer LAN telephony services through the use of a software client on the user's PC, while others offer telephone instruments that plug into the LAN. When there is a need to relocate the equipment, it needs only to be unplugged from one data port and plugged in at the new destination. This process avoids the wiring changes typically done for convention phones and thus VoIP clients reduce the cost of ownership. Over time, these savings add up to the point that a LAN-based telephony system can offer considerable savings over traditional PBXs.

PacketCable

PacketCable™ is the telephony architecture for using VoIP over the Cable TV system. With buried Cable TV plant passing hundreds of millions of homes worldwide, it is logical to assume that cable operators desire to offer new services that make use of their installed system.

The PacketCable standards (1.0, 1.1 and 1.2) were developed by CableLabs®, the research group of the cable operators and makes use of the same equipment used for the cable modem services. The cable modem architecture, known as DOCSIS (Data over Cable Service Interface Specification), is able to meet the requirements of transporting VoIP packets. DOCSIS version 1.1 provides Quality of Service (QoS), security features, and the prioritization of packet traffic that is necessary for voice communication.

The PacketCable specification also incorporates the Network-based Call Signaling (NCS) protocol for signaling voice calls over cable networks. NCS leverages the existing Media Gateway Control Protocol (MGCP) and the protocol is sometimes referred to as MGCP NCS. NCS uses network-based call agents to negotiate cable-based IP telephony calls.

Traditional telephones draw all the power they need from the phone lines. Part of the reason that the public phone system has evolved to such a reliable state, is that it is essentially immune from the effects of power outages. Electrical utilities in most areas do not offer this degree of unfailing reliability and this reality is an important issue for the cable television plant.
Normally, head-end and customer-premises cable equipment rely solely on the local electric company for their power and this puts users at risk of losing phone service should a power outage occur. There is not universal agreement among the cable operators about offering "lifeline" telephony service.

Some Time Warner systems focus their telephony service towards the "second" phone line in the home. They suggest that the customer keep the existing primary line and then add the Time Warner telephony service for a business line, fax line, or children phone. AT&T has taken a different approach by enhancing many of their networks with alternate power sources thereby allowing lifeline telephony services to their customers.

Video Conferencing

Video conferencing includes the use of packetized voice and is an important application for the home and business. There are several video conferencing standards and one of the significant standards is known as H.32x where x can be 0, 1, 2, 3, or 4 representing video conferencing on different types of communications links. The H.32x model includes establishing connections between the source and destination devices and this method connection establishment has been adapted for use in VoIP gateways.

Table 1. The ITU-T H.32x Video Conferencing Standards

	H.320	H.321	H.322	H.323	H.324
Approval Date	1990	1995	1995	1996	1996
Network	Narrowband switched digital ISDN	Broadband ISDN ATM LAN	QoS packet switched networks	Non-QoS networks (Ethernet)	The analog phone system
Audio	G.711 G.722 G.728	G.711 G.722 G.728	G.711 G.722 G.728	G.711 G.722 G.728 G.723 G.729	G.723
Video	H.261	H.261/H.263	H.261/H.263	H.261/H.263	H.261/H.263
Control	H.230/H.242	H.242	H.230/H.242	H.245	H.245
Multiplexing	H.221	H.221	H.221	H.225	H.213
Comm. Interface	I.400	AAL  I.363  AJM I.361  PHY I.400	I.400&  TCP/IP	TCP/IP	V.34 Modem

The H.323 recommendation supports the largest number of audio coding standards. In fact, the audio codecs of the other recommendations are a subset of the audio codecs outlined in the H.323 recommendation. In addition, the H.323 is the recommendation that is made for a non-guaranteed bandwidth, packet switched networks such as the Internet.

To the QoS message, they must guarantee those parameters for the duration of the video conferencing session.

H.245 - Control Channel Protocol. Provides capability H.323 Video Conferencing
The H.323 recommendation covers the technical requirements for audio and video communications services in LANs that do not provide a guaranteed Quality of Service (QoS). The scope of H.323 does not include the LAN itself or the transport layer that may be used to connect various LANs. Only elements needed for interaction with the Switched Circuit Network (SCN) are within the scope of H.323.

H.323 defines four major components for a network-based communications system: Terminals, Gateways, Gatekeepers, and Multipoint Control Units (MCUs). The terminal must support voice transmission with data and video transfer being an option.

Figure 1. The H.323 Protocol Stack  
H.323 uses the following standards as part of the video conferencing protocol stack (Figure 1):
Q.931 QoS - Quality of Service. When the connection between devices is being established, the end stations and each data link in the route negotiate to determine what bandwidth is available, how much delay the application can tolerate, and how much jitter there will be in the packet arrival. Once the links agree negotiation between the two end-points such as voice compression algorithm to use, conferencing requests, etc. The H.245 channels transport must be reliable (e.g. TCP, SPX)
H.225 RAS - Registration, Admission, and Status (RAS) Protocol. Used to convey the registration, admissions, bandwidth change and status messages between IP Telephone devices and servers called Gatekeepers that provide address translation and access control to devices.
RTCP - Real-time Transport Control Protocol (RTCP). Provides statistics information for monitoring the quality of service of the voice call.
Control messages (Q.931 signaling, H.245 capability exchange and the RAS protocol) are carried over the reliable TCP layer. This ensures that important messages get retransmitted if necessary so they can make it to the other side. Media traffic is transported over the unreliable UDP layer and includes two protocols: RTP (Real-Time Protocol) that carries the actual media and RTCP (Real-Time Control Protocol) that includes periodic status and control messages. Audio and Video information is carried over UDP because it need not be retransmitted because if a sound packet is lost and then transmitted, it would most probably arrive too late to be used for the real-time conferencing.




GATEWAY PROTOCOLS

H.323
The most widely embraced standard is the ITU-T’s H.323 umbrella standard. Originally designed as an end to end communications standard for video conferencing over packet networks. H.323 was adapted for VoIP applications. The result was a standard that defines far more functionality than is necessary for most VoIP environments. Its complexity is hard to implement efficiently and causes problems in interoperably since there are various ways to interpret the standard.

MGCP (MEDIA GATEWAY CONTROL PROTOCOL)

MGCP is a protocol that addresses control of media gateways, but it does not, as H.323 does, specify a complete end-to-end communication. MGCP uses simple endpoints called Media gateways (MGs). An intelligent media gateway controller (MGC) or call agent (CA) provides services. The endpoints provides user interactions and interfaces, while the MGC provides centralized call intelligence.
A master/slave relationship is presented at all times between the MGC and the MGs. Typically MGCP message are sent over IP/UDP between the MG and MGC. The media connection itself is usually over IP/RTP. For security, MGCP uses IP sec to protect the signaling information.

SIP (SESSION INITIATION PROTOCOL)

SIP is an application layer signaling protocol that specifies call control for multiparty sessions, IP phone calls or multimedia distribution. Unlike H.323, which is based on binary encoding, SIP is a text based protocol that is much easier to implement. Much like H.323, SIP is a peer-to-peer architecture ( vs, master/slave for MGCP).

SIP depends on relatively intelligent endpoints, which require little or no interaction with servers. Each endpoints manages its own signaling both to the user and to other endpoints. SIP is more scalable than H.323 because it is inherently a distributed and stateless call model.
Perhaps the key advantage of SIP is that it is an Internet-model protocol from inception. It uses simple ASCII messaging based on HTTP/1.1. This means that SIP messaging is easy to decode and troubleshoot

SIP ARCHITECTURE

• SIP terminal: Supports real-time, 2 way communication with another SIP entity. Supports both signaling and media, similar to H.323 terminal.
• Proxy: Contacts one or more clients or next-hop servers and passes the call request further.
• Redirect Server: Accepts SIP requests, maps the address into zero or more new addresses and returns those addresses to the client. Does not initiate SIP requests or accept calls.
• Location Server: Provides information about a callers possible locations to redirect and proxy servers. May be co-located with a SIP server.

THE IMPORTANCE OF VOICE OVER IP

Of the key emerging technologies for data, voice, and video integration, voice over IP (Internet Protocol) is arguably very important. The most quality of service (QoS) sensitive of all traffic, voice is the true test of the engineering and quality of a network. Demand for Voice over IP is leading the movement for QoS in IP environments, and will ultimately lead to use of the Internet for fax, voice telephony, and video telephony services. Voice over IP will ultimately be a key component of the migration of telephony to the LAN infrastructure.

Significant advances in technology have been made over the past few years that enable the transmission of voice traffic over traditional public networks such as Frame Relay (Voice over Frame Relay) as well as Voice over the Internet through the efforts of the Voice over IP Forum and the Internet Engineering Task Force (IETF). Additionally, the support of Asynchronous Transfer Mode (ATM) for different traffic types and the ATM Forum's recent completion of the Voice and Telephony over ATM specification will quicken the availability of industry-standard solutions.

MGCP COMMANDS

•Endpoint Configuration (EPCF): Specify coding
•Notification Request (RQNT): Watch for event
•Notify (NTFY): Used by gateway to inform Call agent
•Create Connection (CRCX)
•Modify Connection (MDCX)
•Delete Connection (DLCX)
•Audit Endpoint (AUEP): Give me status
•Audit Connection (AUCX)
•Restart in Progress (RSIP): Used by gateway to indicate initialization/shutdown of endpoints/gateway

BENEFITS OF  VOIP

Widespread deployment of a new technology seldom occurs without a clear and sustainable justification, and this is also the case with VoIP. Demonstrable benefits to end users are also needed if VoIP products (and services) are to be a long-term success. Generally, the benefits of technology can be divided into the following four categories:

• Cost Reduction: Although reducing long distance telephone costs is always a popular topic and would provide a good reason for introducing VoIP, the actual savings over the long term are still a subject of debate in the industry. Flat rate pricing is available with the
Internet and can result in considerable savings for both voice and facsimile. It has been estimated that up to 70% of all calls to Asia are to send faxes, most of which could be replaced by FoIP.
• Simplification: An integrated infrastructure that supports all forms of communication allows more standardization and reduces the total
equipment complement. This combined infrastructure can support dynamic bandwidth optimization and a fault tolerant design.

• Consolidation: Since people are among the most significant cost elements in a network, any opportunity to combine operations, to eliminate points of failure, and to consolidate accounting systems would be beneficial. In the enterprise, SNMP-based management can be provided for both voice and data services using VoIP.

• Advanced Applications: Even though basic telephony and facsimile are the initial applications for VoIP, the longer term benefits are expected to be derived from multimedia and multiservice applications. For example, Internet commerce solutions can combine WWW access to information with a voice call button that allows immediate access to a call center agent from the PC.




ADVANTAGES OF IP TELEPHONY

Ubiquity

The IP standard is by far the world’s most popular network protocol. It is developing fast and is accepted by every major vendor. Users can benefit from end-to-end connectivity to every data-networking device available, a tremendous amount of research from firms focused on IP, and a unique global addressing scheme which allows an IP device to address the entire network, regardless of size or location.With more than 80 per cent of personal computers in business networks on a local area network (LAN), IP is an obvious transport of choice. It also seems to make sense to carry all forms of communication (data, voice, and video) over a common, ubiquitous medium.

Value Added Applications

Once an IP infrastructure is in place there is a seemingly unending number of add value applications that are available or being designed that are either far simpler than those enabled by regular telephony systems, or totally new applications like user registration that allows phone users to be identifiable and fully functional no matter which handset they’re using.

Cost

For most large businesses, overhauling a PBX-based telecoms infrastructure is a long-term investment decision. However, with telephony requirements becoming more demanding and new technology creating business advantage in shorter time-frames, IP telephony has become a more and more realistic alternative. And while cost projections will inevitably vary enormously from one company to another, most equipment vendors and IP telephony integration specialists would claim return on investment within one or two years.

Single Infrastructure

Putting the voice and data traffic on one set of wires instead of two also seems to make good commercial sense when compared to the cost of developing and supporting two separate infrastructures.

Overhead

A large part of that cost is a human one – and again there is a clear case for creating one pool of skills rather than two, so that voice calls essentially become just another application running on the network. In reality the network will only see information packets, some of this information originates and terminates within a data edge device whilst other packets of information on the network will originate and terminate in voice edge devices.

Call center

Several advantages arise from a Call Center environment based on an IP infrastructure. Web based forms; text chat sessions, email, and web collaboration are all examples of modes of communication, which are based on the IP standard. Voice is transformed to the IP standard in IP call centers. An IP infrastructure therefore facilitates integration of both web-based media and voice into the call center.
IP also puts companies in a better position to take advantage of future developments in terms of new channels because once an IP ‘skeleton’ is in place, functionality is just a question of what software is developed.