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Internship report Rutger Rienks (9706291)

Internship report from Rutger Rienks (9706291)

‘Decision making and research during the construction of VOIP

network components’

January 30th 2002 - April 28th 2002

Surya InfotechSuite 7B, Kirti Towers, Tilak Road, Baroda, India

Surya Infotech, Baroda, India


Company supervisor: Mr. Parag AminInternship supervisor: Mr. Job Zwiers

Surya Infotech, Baroda, India2


Summary

This report describes the AIESEC Internship from Rutger Rienks at Surya Infotech, an IT company based in Baroda, India. The goal of the Internship was to keep the level of knowledge about the emerging VOIP world on the side of Surya Infotech on such a level, to be able to improve and discuss the work performed and the decisions made by the students participating in ‘Project VOIP’ from Nirma’s Center of excellence based in Ahmedabad, India.

After three month the project started implementing and designing an IP Phone and for an IP Gateway all specifications and options were defined. The decision to start developing the gateway was pending on funds. But furthermore all students had achieved great results. The understanding of the technology had grown from nothing to sufficient to start implementing. This was achieved by choosing components which were considered the best amongst all the various components needed.

From the internship point of view, expanding knowledge by understanding the various components as well as helping the company in making choices and debating on technical and economical topics regarding their project was an experience hard to forget. The complete project positively contributed to personal development and the acquisition of new skills.

The time in India was one of great contrasts. Poverty on the streets and malicious systems everywhere around influenced the way of handling and feelings about things. The absurd richness of a very small minority and the total different culture on the other hand were an absolute eye opener. Good was it to see that lots of good initiatives are taken to heal the illnesses of the country. This all inclusive made it, together with the Indian working environment and the nice people, that it was a time never to be forgotten.



Table of contents

1. Introduction........................................................................................................................7

2. The Internship....................................................................................................................8

Personal expectations..........................................................................................................8AIESEC.............................................................................................................................8

The country...........................................................................................................................9India..................................................................................................................................9Baroda..............................................................................................................................9

3. The riots............................................................................................................................11

Background........................................................................................................................11

What happened in Baroda..................................................................................................12

4. The Company...................................................................................................................13

Corporate Profile................................................................................................................13

The working environment...................................................................................................14

5. The organization of ‘Project VOIP’.................................................................................16

Introduction.........................................................................................................................16

Nirma Institute....................................................................................................................16

The goal of the project........................................................................................................17

The way of working.............................................................................................................17

6. An Introduction to VOIP..................................................................................................19

Introduction to IP telephony................................................................................................19Why VOIP.......................................................................................................................19The origin........................................................................................................................20The (future) applications.................................................................................................20The technology...............................................................................................................20

Example network................................................................................................................21

7. A closer look at the major VOIP network components.................................................22



H.323..................................................................................................................................22

RSVP..................................................................................................................................23

G.168..................................................................................................................................23

G.723.1...............................................................................................................................23

T.37/T38.............................................................................................................................24

8. The IP Phone reference design.......................................................................................25

The IP Phone hardware architecture reference design......................................................25

The IP Phone software architecture reference design........................................................26User interface software...................................................................................................27Voice processing software..............................................................................................28Telephony signaling gateway software...........................................................................29Network management software......................................................................................30Network interface software.............................................................................................31System services software...............................................................................................31

9. The IP Gateway reference design...................................................................................33

The IP Gateway hardware reference design......................................................................33Microprocessor...............................................................................................................33DSP resources................................................................................................................33PSTN Interface Modules.................................................................................................34Packet or network Interface Modules..............................................................................34Things not to forget.........................................................................................................34

The IP Gateway software reference design........................................................................35

10. The progress: from one phase to another...................................................................37

Phase 1: The initial sessions..............................................................................................37

Phase 2: Comparison.........................................................................................................38

Phase 3: Which products to build?.....................................................................................39IP Phone, IP Gateway and IPPBX..................................................................................39

Phase 4: Which building blocks to use for our products?...................................................40

Phase 5: The later decisions and the future.......................................................................43

11. Evaluation.......................................................................................................................44

The company......................................................................................................................44



The traineeship...................................................................................................................45

India in general...................................................................................................................46

The time with AIESEC and the trainees.............................................................................47

12. Conclusion......................................................................................................................49

13. Recommendations.........................................................................................................50

The project..........................................................................................................................50

The company......................................................................................................................50

AIESEC.............................................................................................................................. 51

14. Literature en references*...............................................................................................52

APPENDIX A “A comparison between SIP and H323”....................................................54

APPENDIX B “A comparison amongst MEGACO gateways”.........................................57

APPENDIX C “A comparison amongst IP Phones”........................................................60

APPENDIX D “ATMEL and TI IP Phone chipsets compared”.........................................61

APPENDIX E “The Gateway Specifications”...................................................................63

APPENDIX F “The IP Phone Specifications”..................................................................64



1. IntroductionThis report is consists of four major sections, the fist part consisting out of four chapters explains about the background of the internship, the riots during the internship, the company and the internship itself. Chapter two focuses on the expectations of the traineeship, AIESEC as an organization providing the traineeship and it gives background information about India and Baroda, respectively the country and city of where the company is based. During the traineeship there were riots in Baroda that affected the community. Information about the riots and the political background can be found in chapter three. In chapter four the company in general is discussed, it tells about its working conditions and its people. To conclude the first part, chapter five explains the organization of the internship and the targets that had to be met.

The second part describes theory about the upcoming VOIP-networks is in general. Chapter six gives an introduction to VOIP. The reasons why VOIP is expected to become a booming business in the coming years as well as the (future) applications are addressed. In chapter seven a closer look at some of the protocols used in a VOIP environment is given. Important issues like H323, RSVP and codec protocols are discussed. In the chapters eight and nine a reference design is given for both software and hardware implementation for the phone and the gateway.

The third part which consists of chapter ten describes the development process; it explains about the decisions made and gives an overview of the work done from our side as well as from the side of the institute divided into different phases. It is made clear how we decided on the products that had to be developed, how we were going to develop them and with what building blocks.

The fourth and final part evaluates the traineeship in chapter eleven. Conclusions of the project are given in chapter twelve and chapter thirteen concludes the report by giving recommendations to all the organizations involved during the traineeship.



2. The InternshipIn this introduction a description is given about the expectations of the internship that existed before any action was taken, how the internship was obtained from AIESEC and how the country as well as the city during the stay in India was like.

Personal expectationsSince the university curriculum demands from their students that they have to gain some practical experience in the field, everyone has to leave the college campus for three month. In these three months you are to be confronted with a working situation in a real company, this as an addition to the regular, more theoretical, courses.

My expectations were a quite unsure. I had always been able to complete my exams, but programming in a serious way on my own, apart from websites, I had actually never done. Nor had I participated in a project being the only one responsible for the deliverables. So therefore I was quite excited.

During my first four years at the university I had done a lot of extracurricular activities. Most of them dealt with the European student association AEGEE-Enschede. Here, after being a member of several committees and even joining the board in 1999, it became clear to me that I really liked to be confronted with different people coming from different countries. Their different approach to solving problems and being confronted with another world view really intrigued me during these years.

Hence I came up with the plan to apply for the AIESEC exchange program. They offered a unique chance to combine both an internship as well as an international experience. I applied and enrolled.

My expectations of working in the company where I would end up, was working hard on an IT project all day, being flexible and show them my best side. Eager to learn something new, gaining experience and fulfill both mine as well as their expectations where the main goals.

What I expected regarding the working conditions were, although knowing the destination being India, like normal standards, working 40 hours a week, having a normal table with a decent computer etc. On the salary part I didn’t really mind what the company would pay. The Internship had to be done, not short for money, I didn’t care. It was all about the experience!



AIESECAIESEC is the world’s largest student association and is spread through 83 countries, covering 800 universities worldwide. [@11] The facilitation of global student exchange through traineeships is one of their key assets, offering both voluntary as well as commercial traineeships; they undoubtedly have a lot to offer.

Their mission is to contribute to the development of the countries and their people with an overriding commitment to international understanding and co-operation.

AIESEC assures that ‘The experience of a traineeship will undoubtedly affect the trainee forever’ in such a way that the experience where you have gone through is not one easily to be forgotten. ‘The new concepts the trainee acquires combined with the confrontation of mindsets from trainees coming from all over the world will change the way of reasoning and acting underlining the mission.’ All this really suited my ears and off I was.

The countryThe city where the company of my traineeship was based is called Baroda, a ‘normal’ sized Indian city. This paragraph gives a brief overview of the India as well as Baroda.

IndiaThe civilization of the valley of the river Indus between the Himalayas, Myanmar and Afghanistan is one of the oldest in the world and goes back at least 5,000 years. Aryan tribes from the northwest invaded the land in about 1500 B.C. The Inhabitants from that era created the classical Indian culture. By the 19th century, The British colonized the Indian lands by defeating all the different empires. One of these Empires was the Mughal Empire. One of these Mughal Kings was responsible for India’s main monument to be built: ‘The Taj Mahal’. Mahatma Gandhi and his nonviolent resistance to British colonialism led to independence in 1947. The subcontinent was divided into the state of India, containing most of the Hindus, and into the smaller Muslim state of Pakistan. After a third war between the two countries in 1971 this also resulted in East Pakistan becoming the nation of Bangladesh.

Fundamental concerns in India include the ongoing dispute with Pakistan over Kashmir and the massive overpopulation of over one billion people living on 3.3 million square kilometers, especially in the big cities such as Bombay having 18 million inhabitants. Environmental degradation, extensive poverty (A GDP per capita from $2,200 vs. $24.400 in the Netherlands), and ethnic strife are other main concerns. All of this despite impressive gains in economic investment and output.



BarodaBaroda is situated in one of the western states of India, in Gujarat. Baroda has a rich historical background. The historian can trace Baroda’s history over 2000 years back. However, the recent situation actually originated where the Mughal rule over the city came to an end in 1732. This happened when Pilaji brought the Maratha activities in Southern Gujarat to a head and finally captured the city. Except for a short break, Baroda continued to be in the hands of the Gaekwads from 1734 to 1949.

The greatest period in the Maratha rule of Baroda started with the accession of Maharaja Sayajirao III in 1875. It was an era of great progress and constructive achievements in all fields. It was the Maharadja who visualized a general scheme of development in all branches of knowledge, with the Maharaja Sayajirao University of Baroda as absolute crown. Modern Baroda owes its beauty, its educational institutions and its masterpieces of architecture, like the arts faculty (having the second biggest dome in India) to the vision of this great ruler. The city has built further on the academic infrastructure established by him.

The present educational foundation rests on over 20 public schools and over 100 private schools. The university has around 30.000 students. As you could say, Baroda is synonymous with education.

Figure 1: ‘The Baroda arts faculty’

The people of Baroda like to tell visitors that their city is a ‘Sanskari Nagari’, that is, a ‘cultured city’. The city has rich cultural traditions. And today’s Barodians have quite a full and hectic cultural life throughout the year. Nowadays Baroda is having almost 2 million inhabitants.



3. The riotsDuring the stay in Baroda, all the trainees were confronted with circumstances that they were not used to. Besides from the cultural differences, the situation even became totally unsafe. Riots started from February 28thth onwards in the complete state of Gujarat between Hindu’s and Muslims killing more than 1000 people. All the trainees, including myself, were forced to stay indoors for several days. This chapter gives background information on the origin of the riots and tells how we, as a group of trainees dealt with the situation.

BackgroundAyodhya is the capital of the descendants of Suryavanshi (Sun dynasty) and said to be the birthplace of the Hindu god Rama is a city full of temples that inhabits a majority of Hindu people. Mir Baqi, a Mughal governor, decided to build a mosque in Ayodhya. This happening took place from 1528 until 1529. The reason for this was to praise Babar, the founder of the Mughal Empire.

From this moment in time, the land where the mosque was build has turned out to be a bone of contention between Hindus and Muslims. The district Magistrate of Ayodhya declared the mosque a disputed property after a statue of Rama, a Hindu god, miraculously appeared in the mosque on December 22nd 1949 which Hindus called a miracle. Following a petition, the magistrate on January 5th 1950 ordered locking the doors of the mosque and appointed Ayodhya Municipal Corporation as a receiver in the case. It was on February 1st, 1986 the district judge on recommendation by the district collector ordered opening of the locks of the disputed site to the Hindus for prayers. Congress led Rajiv Gandhi government supported this move.

The Vishwa Hindu Parishad (VHP), a political party, announced in 1986 about its plan to carry out shilanyas (laying foundation stones for a Hindu temple) at the site. The Government agreed to the shilanyas and the VHP carried out the program on November 11, 1986.

The Kalyan Singh government on October 7th 1991 acquired the land in the premises of the site, which Muslims claim to be holy property. The VHP announced its plan to go ahead with temple construction at Ayodhya and announced Kar Sewa (initiating the building process by acquiring volunteers or Kar Sewaks) on December 6, 1992. Kar Sewaks entered the disputed structure and demolished it. The demolition of the disputed structure led to violence in several parts of India and the Government later acquired the land adjoining the site.

In 2001, the VHP decided to go ahead with construction of Ram Temple at Ayodhya from March 12th, 2002. VHP announced schedule for Purna Ahuti yagya (another ritual) and Kar Sewaks assembled in Ayodhya on February 2002. Tension gripped the country after the



Sabarmati Express, (a train) carrying Kar Sewaks returning from Ayodhya was burnt in Godhra, Gujarat on February 28th. This incident commenced the series of riots that lasted until the day that I left.



What happened in Baroda From the moment that the actual violence broke out in the state, all the trainees expected that it would pass away quickly and everyone was hopeful that the situation would return normal very soon.

After a couple of days however more incidents of violence were reported and people in the city started to burn shops. The spiral went on for a couple of days as over a 100 shops were burnt and 20 people were killed in Baroda. This despite the fact that the police imposed curfew in several area’s in the city. Luckily the majority of the violence took place in the old city and houses and areas from the trainees weren’t directly involved. After two more days the army was send to Ahmedabad and Baroda in order to restore peace. ‘Shoot at sight’ were the instructions for the army man. Everyone had to stay at home for four days on end, dependant on the media and the food in store.

After two days of sitting at home and reading books, we decided to go out on the streets to see whether we could buy some food somewhere in the neighborhood. Living on nothing but cookies and some lemonade for two days was long enough. Ellora Park, the neighborhood where I stayed was quite far from the old city, where most of the violence happened. Some of the shops in the streets luckily had their iron roll curtains half open but long queues were standing in front. We managed to buy some food and stuff here though, which made the following days slightly more comfortable. The days at home brought the trainees closer together and even one evening we spend with a small party. The fifth day the tension became less, the curfew was relaxed and we were all able to return to work again.

In the days that followed, now and then the curfew was imposed again, depending on the violence of the previous day. The streets in most of the city parts were deserted due to the curfews. The atmosphere remained tense and some of the trainees wanted to return home. With the newspaper as the only resource for information we did not really know what was going on. This made the situation very strange. On one hand everyone was aware that huge clashes were going on outside in the streets, but on the other hand we couldn’t do anything. Seeing the city burning from the rooftop was a terrible sight. My company demanded from their employees to show up for four Sundays on end to compensate for the days lost during the major gulf of violence. And so we did.

At this time of writing the violence still is not over and communal clashes are still going on. Now and then curfew is imposed again and in the old city a permanent curfew from 23:00 in the evening is still issued. Rumors are going that the police in Ahmedabad are instructed by the local political parties to misinform the army and prevent them from stopping the riots. Hopefully peace will come back quickly and things can turn back to normal.



4. The CompanyIn this chapter Surya Infotech, the company where I had my internship, is introduced by looking on some previous and on going projects. Furthermore the working conditions in the company are discussed.

Corporate Profile Surya InfoTech Private Limited is an information technology solutions company and a subsidiary of a US based technology and research corporation RADIQAL, LLC“ The company is providing cutting edge technology solutions to large international companies in the areas of satellite communications, gaming and wagering, mobile commerce, ebusiness, Video on Demand (VoD) and financial markets“. [@16] Surya InfoTech is a specialized software and technology solutions provider and has developed applications that are patent protected and copyrighted

. Figure 2: ‘The logo of Surya Infotech’

Currently the company is working on a product that provides real time video on demand services to their clients and comprises of industry leading technologies like personal VCR (PVR), record on demand and 20 channel simultaneous video broadcast servers. Applications have been developed for global GSM service providers to enable them in providing services like mobile banking; mobile commerce and SMSC based transaction clearance systems to clear credit and debit card transactions.

Surya InfoTech is involved with the regional academic institutes and partner with them in developing leading edge technologies like VoIP Phone, VoIP Enterprise Gateway and VoIP IPPBX. These systems are state of the art telecommunications solutions and are developed by students and faculty of nationally recognized universities along with technical know how and mentoring from companies like them.

By helping the state government in drafting the IT policy and procedures the company is actively involved in major industry forums across the state. The company recognizes that innovation cannot stop at the technological level but must extend to all facets of the organization.



To elevate the global thinking within the company, Surya Infotech regularly takes up international interns from associations like AIESEC. These student interns help in understanding the needs and demands of global companies; along with understanding the work ethics and culture of foreign nations and in-turn learn about the company and the Indian culture. This rich interaction sets Surya Infotech apart from their competition.

The working environmentSurya Infotech is one of the major IT companies in Baroda. The Director, Mr. Parag Amin has spent several years in the US during his studies and therefore the working conditions differ from a general Indian company.

The office is located on the seventh floor of the Kirti Towers, a twelve storey high building, with two corporate offices on every floor. The atmosphere inside the company was quite western. Marble floors and walls from glass give the company a nice look. The working hours from 9:00 until 18:00 are quite reasonable. However, working on Saturdays is considered normal and due to the fact that we couldn’t work for four days due to the riots, the company expected us to win the lost days back by working on Sundays, and so we did.

During the day there is one break: ‘lunchtime’. All the employees sit on the floor of one of the rooms from the office and all share their lunch prepared by their wife or mother. Since I had no one cooking for me, all the employees were very willing to share their hot, but tasty Indian food. Being amongst them, I used to bring my own bread, jam and honey and let them taste some of the lunches that I was used to.

The employees are all nice and talented people. All more or less afraid of the big boss Mr. Parag Amin (always referred to as Sir.) and his ‘independent woman’ office manager. This however brings the employees close to each other and goes along with a lot of hilarious moments.

The workspace in the office that I had during the internship was the smallest I ever had in my whole life. Although there was one computer for every employee, there was not more than that. On the other hand the chair was top notch. This also goes for the air-conditioning keeping us far from the 40˚ degrees outside. The figure 2 shows the room where I worked during my internship. Desk one was the place were I used to work and Ekta, my Indian trainee buddy used to work on desk two.



Figure 3: ‘The working environment’

As far as the hierarchy is concerned all of the projects were led by Parag himself. Some of them had an extra layer of leadership. Looking at our project for example, Deepa (nr 6 on the picture) was the supervisor of Ekta and me. Most of the interaction is going over MSN, which is not favorable, but due to the fact that the space is small and there are many other people in the same room it was the only option.

The employees are working without any contract about working conditions and days off. The salary is based on the time the employees are working with the company. Ranging from RS.3000 to RS.10.000 which is comparatively low compared to other IT company’s in India. This all is possible due to the fact that Surya Infotech is a private company without stakeholders. In the next chapter an introduction is given of the project where the traineeship was about and the work that had to be done in and for the company.


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5. The organization of ‘Project VOIP’This chapter gives the background information of the project of the traineeship in India. First light is shed on the institute which actually initiated the project, thereafter the project goals are defined, and the role from our side is explained as well as the way of working during the project.

IntroductionThe traineeship was involved with a project that again was part of a bigger project from the prestigious Nirma Institute. Nirma’s intention of the project was to let their highly talented students go through a complete product (IP Phone and IP Gateway) development cycle. In order to provide their students with more practical and commercial knowledge they asked Surya Infotech to assist the students and the project. It is clear that this way of working is cheap for the product development and valuable for the students.

Mr Parag Amin, the project leader from Surya Infotech who has more than 17 years of experience in the IT world and Dr. Madhu Metha from Anjaleem, responsible for nearly the complete development of the current telephone network in India, were the two industrialists who assisted Nirma Institute in order to give the students this unique chance.

Nirma InstituteNirma Institute of Technology (NIT) is a self financed technical institute offering degree level engineering programs and postgraduates in master of computer applications [@10]. It was established in 1995 by the in India philanthropist and industrialist, Shri Karsanbhai K. Patel under the aegis of Nirma Education and Research Foundation (NERF). The institute is recognized by the All India Council for Technical Education (AICTE) and State Government of Gujarat. Located about 15 kilometers from Ahmedabad and accessible from the State capital Gandhinagar. NIT shares a campus of 100 acres with two other premier institutions, Nirma Institute of Management and Nirma Institute of Diploma Studies.

Figure 4: ‘The Nirma campus’



The environment gives full scope for group activities, as also to individual pursuits for development on preferred tracks. Their vision is to set up an institute of international repute. The management believes that to withstand global competition and to fulfill the huge demand of quality professionals, an organization must constantly grow, constantly innovate, build strength, improve efficiency and strive to become more and more self-reliant. The growth of the institute under the dynamic leadership of management has been stupendous. The dedication and commitment with which student, faculty and staff are working for institution building are remarkable. The institute is the most preferred one among the students of Gujarat and they have been securing top ranks in examinations.

The Nirma centre of excellenceThis specific project is carried out by the center of excellence wherein Nirma exposes and trains their students on cutting-edge-technologies which hold the potential to ‘radically alter the future and impact daily life’. The center of excellence consists of 40 students. Since this is the first project for the center, this time the Nirma students were able to join the project just on interest. For the next projects however, the students will be screened on their skills. This all however implies that all the 40 students working on the project belong to the best students in Gujarat.

The goal of the projectBy the institute the Voice Over the Internet Protocol or VOIP has been selected as a first project for the Nirma center for Excellence.

The goal of the project was:” to let the students experience how it is to through the development cycle of producing a VOIP Gateway as well as a VOIP Phone.”

Together with Ekta Shah I was responsible to keep the knowledge from the company side on at least the same level of the students of Nirma Institute. In this way Surya Infotech was able to help the students and speedup the development and also to make better decisions and choices regarding the development process.

The way of workingThe project was organized mainly by the Nirma Institute. Dr. Madhu Metha (From now on referred to as Madhubhai, how he actually used to be called) and Parag Amin were guiding the students’ trough their way in developing their VOIP product. With weekly sessions at Nirma, the students had to show their progress by means of presentations every Saturday. Unclear points or aspects were debated and deadlines were set for the upcoming week. The complete project was scheduled to last 8 month and the students were expected to work on the project in their free hours and evenings. The budget from the beginning was unclear and



dependant on our demands. But since Nirma is a quite well funded institute there were no major hazards expected.

Our side (the team for Surya Infotech) of the project team consisted of Mr. Parag Amin, Deepa, Ekta and Me. During the sessions both Madhubhai and Parag stood in front of the students. Madhubhai’s focus was more on the technical aspect focusing on the progress, the level of understanding of the different concepts, the protocols and the opportunities. Parag on the other side spoke mainly from the business point of view (The product should be cheap, successful, fast to produce, the teams should work efficient etc.).

In order to keep up with the students we had to do a lot of research at our end to see that the way the students were going was the right one, that the decisions they made were founded on solid argumentation and that they did not overlook things or made decisions too quickly.

In the beginning a lot of reading through all the different aspects of IP telephony had to be done. Thereafter comparative studies amongst protocols, products and applications were made. Our as well as their results were discussed at Nirma and the students used it as guidance to speed up their research and development. All the time our objective was to stay ahead.

Ekta and I had to do most of the research. Comparative studies had to be made, concepts had to be cleared and we were expected to tell Deepa how things were supposed to work and what the essential building blocks were to be kept in mind. We discussed the things that arose in meetings with Madhubhai, who visited our office quite regular. During the research a lot of reading had been done. It extended our knowledge substantially. In the meetings with the students we, as Parag’s technical backup, had to explain how techniques were to be applied and what the pro’s and con’s of the various systems or components were.

Involving expertsBesides our company, Nirma Institute invited experts to meet the students and to discuss the topics they were deciding and working on. The students even had the opportunity to give presentations about their project to India’s richest man and IT guru, Mr. Azim Premji. Although he had just 7 minutes for us and did hardly give us any feedback, it was an amazing experience. The preparation of the presentations and so, however were still useful in way that the students had to reconsider their position and their goal in and of the project. In the next chapter a general introduction to VOIP is given.



6. An Introduction to VOIPThis chapter explains about VOIP and its background. It describes why VOIP is expected to be a huge growth market, what the origin of VOIP is, applications are discussed and an example network is given to provide a basic understanding level for the issues involved in the VOIP world.

Introduction to IP telephonyAn IP Phone is a telephone device able to transport voice over a network using data packets instead of using a circuit switched connection. IP Telephony in that sense refers to the transfer of voice over the Internet Protocol (IP). Other Voice Over Packet (VOP) standards exist for Frame Relay and ATM networks but many people however use the terms Voice over IP (VOIP) or “IP Telephony” for voice over any packet network.

Implementation can be seen in various ways e.g. phone to phone, phone to PC or PC to PC. In the future even more than two parties might talk with each other simultaneously. Refer for more information about next generation applications to [MC00]. The number of options is endless as long as there is an IP network somewhere in between the connection lines.

A major advantage of VOIP and Internet telephony is that it avoids the tolls charged by ordinary telephone providers, resulting in a likely substantial influence on the telephony market. Before going in any further technical detail it is good to know why the market actually is moving into the direction of VOIP.

Why VOIPThere are three main reasons according to the Internet Engineering Consortium [@06] for the fact that VOIP is interesting for both the consumer as well as the producer. All are listed below:

Cost reductionIP is everywhere. It is on our desktops and it is what the Internet is based on. Many people view the Internet as a "free transport" for data and voice services. With the introduction of Net2Phone and other similar "free" services, many people are making phone calls over the Internet. In addition, businesses and individuals have turned to higher-quality commercial products and services to make voice calls based on IP. The improvement of the IP network, like switches and routers, continues to reduce the cost of providing VOIP.

Simplification and consolidationAn integrated infrastructure that supports all forms of communication could allow more standardization and could reduce the total equipment. This provides clarity for both



customers as well as for the producers. The differences between the traffic patterns of voice and data offer further opportunities for significant efficiency improvements. Universal use of IP for all applications, voice and data, holds out the promise of both reduced complexity and more flexibility.

Advanced applicationsEven though basic telephony and faxing 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 World Wide Web access to information with a voice call button that allows immediate access to a call center agent from a PC. In addition, voice is an integral part of conferencing systems that could also include shared screens, white boards, etc. Combining voice and data features into new applications will provide the greatest returns over the longer term. This all will create new possibilities, markets and options for everyone.

The originThe origin of IP Phones originally comes from client software running on PCs for low-cost PC-to-PC communications over the Internet. The problems associated with the Internet and the PC resulted in poor voice quality (Low Quality of Service, or QoS) due to long and variable delay and network congestion resulting in lost packets. Nowadays, the QoS provided by the Internet continues to improve as the infrastructure uses faster backbone links and switches resulting in congestion reduction, cut down latency due to faster connections like ADSL, and new protocols like RSVP providing a technique to prioritize amongst packet types.

The (future) applicationsMost of the business focus regarding VOIP is currently concentrating on two applications. The first one is a private business network application. Most businesses with remote branch offices are already connected together via a corporate intranet for data services and can take the huge advantage in cost reducing by adding voice and fax services using VOIP technologies. The second application is VOIP over public networks. ISPs are interested in VOIP as a new way of offering value-added services like voice and fax on a per minute usage basis at rates significantly less than the old voice and fax rates for service through the PSTN. In the long term, as already mentioned, IP networks are more efficient for a wide range of new applications, particularly multimedia applications enabling convergence of voice, video, data, and fax. For example, web-enabled call centers will enhance the ability of companies to deliver new ways of customer service.



The technologyFrom the technology point of view, the development VOIP solutions include the advent of low-cost, low-power and high performance digital signal processors (DSPs) and RISC cores to perform the CPU-intensive function of packetization and mainly (de-)compressing voice and fax data. Another issue is that the arrival of industry standards for voice over packet systems will allow the interoperability of devices from different manufacturers. Recent standards include ITU H.323 (voice) and for T.38 (fax).

Example networkThe next figure shows an example VOIP network. Ordinary analog phones, instead of IP Phones, call each other via an Internet Service Provider (ISP) who transfers the call via the IP network. The gateway from the ISP has to digitize, compress and packetize the call data before transferring it. This is made possible by the network server together with the name server both assuring that the originating ISP gateway is able to connect to the called phone and that the call is to be transferred to the de destination ISP gateway who is able to decode and regenerate the call-data for transferring onto the PSTN again. Now the connection with the called party can be established.



Figure 5: ‘The example VOIP network’

The big profit in these kind of systems is that the customer saves a lot of costs due to the fact that the internet can bridge a huge distance for hardly any costs and on the other side, the customer is still using its ‘legacy’ phone system, which prevents the user from investing in a new telephone. Of course other network topologies exist, this is just meant as an example. In the next chapters the concept of IP Phone and IP Gateway will be examined more closely, but first some major VOIP concepts are explained in more detail.



7. A closer look at the major VOIP network componentsThis chapter focuses on some of the major components in a VOIP network. It will explain the network components based on the H323 concept. The RSVP protocol for prioritizing voice packets over data packets will be discussed, as well as the G.168 protocol for echo cancellation and the G.723.1 protocol for voice compression.

H.323Since VOIP services need to be able to connect to traditional circuit-switched voice networks. The ITU-T has made this possible by defining H.323, a set of standards for packet-based multimedia networks. The basic elements of the H.323 network are shown in the network diagram below [Bo00] where H.323 terminals such as PC-based phones connect to existing ISDN, PSTN and wireless devices.

Figure 6 : ’ Typical H.323 network’

H.323 terminals that are endpoints such as a IP Phone for voice transmission on a LAN, they have at least one CODEC (like G.723.1) to compress the voice. Terminals also need to support signaling functions that are used for call setup, tear down, etc. Gateways are the interface between the LAN and switched circuit network, a gatekeeper on the other hand performs admission control, address translation and routing. The idea behind defining the gatekeeper entity is to allow H.323 designers to separate the raw processing power of the gateway from intelligent network-control functions that can be performed in the gatekeeper The MCU (Multipoint Control Unit) is the last part of the network and enables conferencing between three or more endpoints.



The signaling standard is H.225.0 which is a subset of Q.931 (ISDN) signaling; H.245, used to exchange capabilities such as compression standards between H.323 entities; and RAS (Registration, Admission, Status) that connects a terminal to a gatekeeper.

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 as defined in IETF RFC 1889 [@15]: RTP (Real-Time Protocol) that carries the actual media and RTCP (Real-Time Control Protocol) that includes periodic status and control messages. Media is carried over UDP because it would not make sense for it to be retransmitted: should a lost sound fragment be retransmitted, it would most probably arrive too late to be of any use in voice reconstruction.

RSVPThis protocol [@08] is used to assure Quality of Service (QoS) from a network, on behalf of an application data stream. RSVP takes care after the data stream throughout the network, at each node, RSVP attempts to make a resource reservation for the stream. In VOIP networks it is important for the voice data to be delivered as soon as possible. The RSVP protocol comes into picture when there is both ordinary data traffic as well as voice traffic over the same network. In this case it is the task of the protocol to prioritize and schedule voice traffic above the ordinary traffic.

To make a resource reservation at a node, the RSVP daemon communicates with two local decision modules: admission control and policy control. Admission control determines whether the node has sufficient available resources to supply the requested QoS and a policy control determines whether the user has administrative permission to make the reservation. If either check fails, the RSVP program returns an error notification to the application process that originated the request. If both checks succeed, the RSVP daemon sets parameters in a packet classifier and packet scheduler to obtain the desired QoS. The packet classifier in each of the network nodes determine the QoS class for each packet and the scheduler orders packet transmission to achieve the promised QoS for each stream.

G.168G.168 is a line echo canceller, which is a voice operated device placed in the 4-wire circuit and is used to reducing the echo by subtracting an estimated echo from the circuit echo. Echo cancellers are able to cancel the delay up to a range of 120-125 ms



G.723.1G.723.1 is a dual rate speech coder standard from International Telecommunication Union–Telecommunication standardization sector (ITU-T). It is used for compressing quality speech, normally sampled at 8000 samples per second at a rate of 64 Kbps.

This codec has two compressing bit rates, 5.3 and 6.3 Kbps. Both bit rates share the same short-term analysis techniques for processing the speech. For long-term analysis of speech, the algorithms used are different. For 5.3 Kbps coder, Algebraic Code Excited Linear Prediction (ACELP) principles are used where as in 6.3 Kbps coder, Multi Pulse-Maximum Likelihood Quantization (MP-MLQ) techniques are used.

The coder works on a frame of 240 speech samples (30 ms). Besides, there is a look ahead of 60 samples (7.5 ms). So the total algorithmic delay for the coder is 37.5 ms, other codecs are shown in figure 7 [BO00].

Figure 7 : ’Different kinds of codecs’

T.37/T38T.38 is the fax transmission protocol selected for H.323. Traditionally there have been two approaches for sending Fax over Packet networks: the real-time method and store and forward method. The primary difference in service between these two approaches is in the delivery and method of receipt confirmation. The ITU and IETF are working together to continue to evolve both the real-time Fax over IP network standard (T.38) as well as the store and forward Fax over IP network standard (T.37). Both T.37 and T.38 were approved by the ITU in June, 1998.



8. The IP Phone reference designIn this chapter a closer look is taken towards the features of a user terminal, in this case a VOIP Phone. This reference design is made as a reference for the final design as discussed in later chapters. The main difference with a legacy telephone and an IP Phone is that the IP Phone is directly connected to the internet instead of to the PSTN.

The IP Phone hardware architecture reference designThinking about the hardware architecture it is easy to come across the user interface you will need. Other more in depth or underlying components are not so easy to come up with without studying some literature in advance. Figure 8 below shows a block diagram of a complete reference design of a possible IP Phone. This IP Phone consists of the following components:

User Interface Voice Interface Network Interface Processor Core and associated logic.

All the major components needed are discussed into more depth below.



Figure 8: ’Architecture of an IP Phone’

The User Interface provides the traditional user interface functions of a telephone. At a minimum, this consists of a keypad for dialing numbers (0-9, *, #) and an audible indicator for announcing incoming calls to the user. On more sophisticated telephone sets, additional keys are provided for features such as mute, redial, hold, transfer, conferencing, etc. A display is also typically provided for displaying user prompts, number dialed, CallerID information for incoming calls, etc. In certain models, the telephone can be equipped with a serial interface to allow communications to a device such as a PDA to allow synchronization of phone information, facilitate automatic dialing, etc.

The Voice Interface provides the conversion of analog voice into digital samples. Speech signals from the microphone are sampled at a rate of 8 KHz to create a digitized 64kbps data stream to the processor via a pulse code modulation (PCM) codec. Similarly, the processor passes a 64kbps data stream in the return path to the speaker through the PCM codec to convert digital samples back into speech.

The Network Interface allows transmission and reception of voice packets from and to the telephone. For corporate LANs this is most often either 10BaseT or 100BaseT Ethernet running TCP and IP protocols. The IP Phone may offer a second RJ-45 Ethernet connector to allow a PC to plug in and share one connection to the wall jack.

The Processor Core performs the voice processing, call processing, protocol processing, and network management software functions of the telephone. As shown in figure 8, this consists of a Digital Signal Processor (DSP) for the voice-related functions and a Micro Controller Unit (MCU) for the remaining functions.

The IP Phone software architecture reference designThe software architecture of an IP Phone based on the ITU H.323 standard for VOIP is shown in figure 9 [Mo00]. The software consists of the following major subsystems:

User Interface Voice Processing Telephony Signaling Gateway Network Interface Protocols Network Management Agent System Services



These subsystems are all described below. The most important software parts are listed. However, as software as well as hardware is subjected to changes. It might happen that some of the protocols listed are outdated.

Figure 9: ’ Software overview of an IP Phone’

User interface softwareThe User Interface subsystem provides the software components that handle the interface to the user of the IP Phone and consists of the following software modules:

Display Driver Controls the hardware that generates characters to the display.

Keypad DriverPerforms keypad scanning and detects key presses entered by the user.



Audible DriverPerforms control of the hardware that generates the ringing of the telephone.

User ProceduresControls the information displayed by the Display Driver and processes user key inputs and converts them into primitives for Call Processing.

Voice processing softwareThis software module is used for the processing of the voice coming from the IP network or from the user as well as creating sound over the speakers indicating actions performed by the user.

Pulse Code Module Interface UnitReceives PCM samples from the analog interface and forwards then to the appropriate DSP software module for processing. It also forwards processed PCM samples to the analog interface.

Tone Generator Generates call progress tones to the user and generates in-band Dual Tone Multi Frequency or DTMF digits to the network based on key presses relayed from the user interface. For certain voice codecs, the compression algorithm does not permit faithful transmission of DTMF tones. For those algorithms, e.g., G.723, the software generates an in-band message to the network that is used by the remote IP Phone (or gateway) to regenerate the DTMF tone.

Echo Canceller UnitThe Unit Performs ITU G.168 compliant echo cancellation on sampled full-duplex voice port signals. The echo from the telephone network is removed by a digital filter on the transmit path into the packet network.

Voice Activity Detector (VAD)Detects voice activity and activates or deactivates the transmission of packets in order to optimize bandwidth. When activity is not detected, the encoder output will not be transported across the network. This software also measures Idle Noise characteristics of the interface and reports this information to the Packet Voice Protocol for periodic forwarding to the remote IP Phone or gateway. Idle noise is reproduced by the remote end when there is no voice activity so that the remote user does think the other party hung up. Voice Codec Unit



Performs packetization of the 64 kbps data stream received from the user. Options are G.711 making just packets without any compression. Another one is G.729 compressing to 6.3 or 5.3 Kbps or G.728 compressing to 16 Kbps. For more details see the previous chapter.

Packet Playout UnitPerforms compensation for network delay, network jitter and dropped packets. Many proprietary techniques are used to address these problems since there are currently no standards in place for packet playout.



Packet Protocol Encapsulation UnitPerforms encapsulation of the packet voice data destined for the network interface. For VOIP this encapsulation goes via the Real-time Transport Protocol (RTP) layer which runs directly on top of UDP.

Voice EncryptionProvides optional encryption of the voice packet data before transmission over the network to ensure privacy.

Control UnitCoordinates the exchange of monitor and control information between the Voice Processing Module and Telephony Signaling and Network Management modules. The information exchanged includes software download, configuration data, signaling information and status reporting.

Telephony signaling gateway softwareThe Telephony Signaling Gateway (TSG) subsystem performs the functions for establishing, maintaining and terminating a call. These functions can also be done by an H323 gatekeeper, but in this case they are embedded in the IP Phone itself. The TSG consists of the following software modules:

Call ProcessingPerforms the state machine processing for call establishment, call maintenance and call tear down.

Address Translation and ParsingPerforms digit collection and parsing to determine when a complete number has been dialed and makes this dialed number available for address translation.

Network SignalingPerforms signaling functions for establishment, maintenance and termination of calls over the IP network. Using : H.323, SIP as well as SGCP/MGCP.

H.323 ProtocolsH.323 is an ITU standard that describes how multimedia communications occur between user terminals, network equipment and assorted services on Local and Wide Area IP networks. The following H.323 standards are used for VOIP in an IP Phone: H.225—Call Signaling Protocols. Performs signaling for establishment and termination of call connections based on Q.931.



H.245—Control Protocol. Provides capability negotiation between the two end-points such as voice compression algorithm to use, conferencing requests, etc.

RAS—Registration, Admission, and Status (RAS) Protocol. Used to look after the registration, admissions, bandwidth change and status messages between IP Phone devices and servers called Gatekeepers which 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.

SGCP/MGCP ProtocolsSimple Gateway Control Protocol (SGCP) is a standard that describes a master/slave protocol for establishing VOIP calls. The slave side or client resides in the IP Phone and the master could reside in an entity referred to as a Call Agent. SGCP has been adopted by the Cable Modem industry as part of the DOCSIS standard. SGCP is evolving to the Multimedia Gateway Control Protocol (MGCP).

SIPThe Session Initiation Protocol (SIP), is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging

Network management softwareThe Network Management subsystem supports remote administration of the IP Phone by a Network Management System. The Network Management Agent should consists of at least the following software modules:

Network Management AgentPerforms the network management functions of the IP Phone, including status monitoring and alarm reporting, gathering of statistics in response to SNMP queries, etc. from a network management system.

Embedded Web ServerThe EWS provides administration support via a standard web browser. It presents the user with web pages for configuring the IP Phone and gathering statistics information. Besides that, it may provide Java applets for loading to the user's PC, e.g., for status polling.

SNMPPerforms the Simple Network Management Protocol (SNMP) functions for processing Management Information Base (MIB) Gets and Sets and generation of Alarm Traps.



TFTPTrivial File Transport Protocol (TFTP) is used to download software updates into Flash memory.

Network interface softwareThe Network Interface Protocols support communications over the Local Area Network (LAN) and consists of the following software modules:

TCPThe Transport Control Protocol (TCP) provides reliable transport of data including retransmission and flow control. It is used for web queries and call signaling functions.

UDPThe User Datagram Protocol (UDP) provides efficient but unreliable transport of data. It is used for the transport of real-time voice data since retransmission of real-time data would add too much delay to the voice conversation and be unacceptable. UDP is also used for SNMP and TFTP network management traffic.

IPThe Internet Protocol (IP) provides a standard encapsulation of data for transmission over the network. It contains a source and destination address used for routing.

MAC/ARPPerforms Media Access Control (MAC) management functions and handles Address Resolution Protocol (ARP) for the device.

Ethernet DriverConfigures and controls the Ethernet controller hardware, including setting up Direct Memory Access operations.

System services softwareThese services are software modules needed in order to make the phone ready for use when switched on and to be able to run the other protocol stacks etc.

Startup/Initialization To initialize the phone on startup

POST



Power On Self Test

RTOS Real time Operating System

BSP Board Support package to let the RTOS support the hardware via hardware interface drivers of the BSP

Watchdog Timer Driver This Driver controls the WDT in order not to let the phone lock up due to a soft or a hardware failure

Flash memory managerThis manager is used for reading and writing to the memory

DSP interface manager This manager is needed for information exchange between the MCU and DSP



9. The IP Gateway reference designSince an IP Phone might want to call to a normal legacy phone there is a need for a device able to bridge the gap between two different networks. Such a gap bridging device is called a gateway. In our case the IP Gateway should be able to bridge the gap between the PSTN and the packet network. In this chapter the hardware part as well as the software part of an IP Gateway is examined.

The IP Gateway hardware reference designThe hardware of the gateway is actually quite simple compared to that of a phone. Figure 10 [in01] shows an abstract implementation of an IP Gateway. All the major system parts of a gateway are shown and are further discussed below.

Figure 10: ’ IP Gateway components’

MicroprocessorThe processor needs to handle signaling, and management and control functions as required by the gateway application. When using H323, the microprocessor is for example required to handle the control portion of H.323. Program and data memory is also required in the system for the microprocessor and the DSP. The choice of microprocessor depends on the processing power required for the application, in addition to scalability requirements and cost.



DSP resourcesThe VOIP Gateway should have more than one DSP, since there are many simultaneous telephone conversations going through the gateway. The DSP performs the actual conversion of the voice streams between the PSTN and packet worlds. It transforms the voice signal form analogue to digital and compresses it by using a codec. The selection of a DSP is based on several criteria. Port density is here quite an important factor. The number of voice and fax channels that can be served by a DSP depends on the available MIPS and the memory to run the algorithms

PSTN Interface ModulesThe PSTN Interface Modules provide the interface to the Public Switched Telephone Network. The connection to the PSTN can be through digital interfaces like T1 or E1. The T1 or E1 interface multiplexes 24 (T1) or 30 (E1) channels of voice, along with signaling, into a single data stream. The connection to PSTN however can also be made through an analog interface if there is a codec able to convert the analog signals into an appropriately coded digital stream

Packet or network Interface ModulesThe Packet Interface Modules provide the interface to the packet switch network. The two most prevalent networks are ATM and IP. 10 or 100Base T Ethernet connection is the standard nowadays.

Things not to forgetThe figure shown on the previous page shows only an abstract gateway. Some other components that should not be forgotten are listed below.

Backplane InterfaceThere are many different backplane interfaces that can be used in systems such as these, for communication with other devices for example. Most typical are PCI and cell bus variants like a utopia interface for ATM networks.

Alarm Monitor & Control (M&C) ModuleThe Alarm Monitor & Control (M&C) Module should perform the overall network management for the equipment.

Call Processing ModulesThe Call Processing Modules perform call establishment and call tear down for the system and performs bridging functions between the PSTN and packet network.



Aggregation logicIn order to concentrate a large number of channels, some aggregation logic is required to route incoming packets from the Backplane or Packet Network Interface to the appropriate DSP



The IP Gateway software reference designThere are many features that must be implemented for IP Gateways. The software should be designed to minimize delay & maximize scalability. This all includes:

Efficient and adaptive algorithms Low-latency implementation Features to off-load the “host” processor to enhance overall channel density

Looking at the figure below it becomes clear that there are three main parts apart from the interfaces. The fax relay should take care of the fax data packetization and relaying. The voice should be (de)compressed, (de)packetized and forwarded and the signaling part should do the translation amongst the signaling used in the PCM and the packet network.



Figure 11: ’ possible gateway software’

The architecture is quite similar to the phone. There are three main parts. The first one is Voice, looking after the echo canceling, conferencing, compressing, VAD, VPU and the packetization. The second one is Fax, which is optional. The third part is the signaling part, looking after the tone detection, caller ID detection and signaling support.

The main difference with the software architecture of an IP Phone is the number of channels that have to be considered as well as the signaling which now also have to deal with the analogue phones. Another big issue not to forget is the management to be carried out by the host processor for the various voice lines on the DSP’s.

The next chapter will describe the project development phases and the decisions made during the internship.



10. The progress: from one phase to anotherThis chapter will focus on the processes of the project and the different phase threads of research carried out in the first three months of the project. There are five phases that can be identified in these months of basically research. An attempt has been made to split up the work according to the work done at our end and the work performed by the students at Nirma. Some of the decisions made however were made in the interactive sessions on Saturday’s, others in phone calls or in meetings where Madhubhai visited our company.

Phase 1: The initial sessionsThis part describes the initial stage in the development process. Madhubhai and Parag both started the project on the 30th of January 2002.

The work at NirmaIn the beginning the situation at Nirma was far from structured. The main issues raised in these days were the things that had to be developed. There was an introduction on why VOIP was chosen as a research environment for the center of excellence. The professor explained was expected from the students taking part in the center of excellence and groups were made in order to perform some first research. The groups formed all had a specific task assigned. The main groups were:

The gateway group The IP Phone group The OS group Protocol study group

The work from our sideThe work in these days mainly consisted out of reading, reading and reading. The VOIP world is not an easy one to understand. It took both Ekta and me around one week before we were able to distinguish amongst the various protocols and solutions available. A lot of the whitepapers Deepa had gathered, before the actual project started, were studied. Hereafter we used around one week reading articles on the web and reading through RFC and ITU standards like MEGACO, H323 and SIP. Besides all this we had to read through the current telephony systems such as the PSTN functionality and SS7 signaling.

Deepa initially had identified three steps in the development of the product.



1) The first product should be an IP Phone that is able to talk to another IP Phone on the same LAN.

2) The second step was to let two legacy phones talk via two gateways and one gateway controller (MGCP) on a same LAN.

3) And the third step would be to let the two legacy phones talk via their own gateway over IP on a different location.

The philosophy was based on the idea that by using an IP network a company could cut huge costs by communicating amongst different branch offices over IP. Also, when using a legacy analog phone it should be possible to call to the office, there the IP Gateway should be able to transfer it to IP, connect it to the branch closest to the called party and from there redirect the call over PSTN.

However these plans were far future music and abandoned later due to political limitations. A closer look had to be taken at the various components involved in the system.

Phase 2: ComparisonAfter a lot of research the first steps were made into the right direction. The various aspects of the system were identified and thoroughly studied.

The work at NirmaThe students at this point of time made a lot of comparative studies. They put a lot of effort in comparing the different operating systems that could be implemented on the phone as well as the gateway. The cost factor was here of major importance as well as the speed of it. After some research and company responses it was quite clear that as far as the OS was concerned, we had to go with Linux. Other competitors like VxWorks or Nucleus were left far behind regarding costs and real time performance.

Other fields of research were about the first specifications for the IP Phone. How should it look and what functionality should it have. The discussions at Nirma were quite good and amusing. The debate on whether to use an LCD was quite nice. A decision was made to have at least support for LCD on the hardware level and if it wasn’t too costly two lines of sixteen characters. Based on Pixels in order to be able to use local languages and with backlight support so that the vendors could use it in the dark. The discussion ended by deciding on a four line display having sixteen to twenty characters. See appendix F for more details.



Many more aspects were debated. One of them was the target market. At this time, according to Madhubhai, the main focus had to be the phone stalls you can find everywhere in India along the roads. Since most of the Indians have only the possibility to make local telephone calls at their home this was the market we had to address. The markets for VOIP are mostly focusing on cutting costs of inter local and international traffic.

The work at our sideIn these days several components were picked out of the network and in depth studies were made of them. Especially the different protocols had to be studied more specific. It was clear that H323 is the standard, but how and why? In this moment in time the opinion at our side was that the MEGACO protocol might be the best to go with since it gained a substantial part of the market that is still expanding. It uses both SIP and H323. One of the documents made at that time is the H323 vs. SIP comparison that can be found in appendix A. The protocol to start with (decided on April 20th) turned out to be H323. This due to the limitations of the law prohibiting us from the MEGACO use and another important aspect was the benefit of error handling of H323 above SIP.

Apart from that a database with over 50 different gateways was built. Going through all of them provided more details about the actual components used. In these days it became clear that most of the gateways were using an MIPS [@13] or ARM [@12] microprocessor and a DSP farm from either DSP groups [@14] or Texas Instruments [@05]. Finding these components in the beginning was like looking for coca cola’s receipt. Most of the vendors did not talk about their internals. Luckily some were saying more than others.

With this database a comparative study of the gateways supporting MEGACO was made. The results can be found in appendix B. After this, the next goal was to come up with three phones that were the best ones to buy to do some performance testing with them. Three phones were selected. After comparing around 20, the 7 best ones were selected and after being compared more thoroughly. The TNETV1001 from Texas instruments, de Aplio IP Phone from Aplio and the IP Phone from Netergy came out as the best ones to use. Reasons for this were that the phones were not using the appropriate stacks, that they just weren’t providing enough details (E.g. the ‘freeride’ series from e-tel) or the phones were using complete PC’s for implementation (E.g. Nortel’s i2050) . The comparison can be found in appendix C

Phase 3: Which products to build?Since the market demands the products now, the plan of buying and playing was dropped and a process was started of building the components from scratch. So the idea of buying an IP Phone was abandoned and the knowledge that was gathered had to be used in a different way. On the other hand, regulations amongst the legalization appeared. From April 1st 2002



VOIP was to be legalized in India. One of the conditions was that no PSTN lines were connected to IP and vice versa. This smashed the idea of transferring calls via company branches. New ideas had to be made. Another bad thing was that the implementation had to use either SIP or H323, so the MEGACO efforts also became, apart from the knowledge level, quite useless.

IP Phone, IP Gateway and IPPBXIn the next meetings with Madhubhai the decision was made to build an IP Phone, an IP Gateway and an IPPBX. The ideas behind this were the ablillity to target on the major market components. A small enterprise gateway having 8 ports targets basically on cyber cafes enabling them to plug in normal phones on their Ethernet to provide telephony service for their customers. The IPPBX should be there as an add-on for the same gateway. The IPPBX was supposed to extend the gateway from a functionality perspective. For more information refer to [SH00]. The IPPBX had to support billing, Interactive Voice Response (IVR), Call lookup and voicemail and is intended for call centers, hotels and hospitals. The demand for the IP Phone is obvious; it should be plugged into the Ethernet and work. The main market would be the phone boot on the Indian streets.

The work at NirmaMore research was done in the field of DSP’s and microprocessors. After some comparisons at their end only two chipsets remained for the phone. The Texas Instruments C5471 and the Atmel AT75c220, both were having an ARM7 core and a strong DSP. Furthermore there were documents appearing about the requirements for the phone and gateway and things were cleared about compression algorithms and their memory usage. The one page spec sheet for the gateway can be found in appendix E

The work at our sideIn this phase the chipsets most wanted for the implementation were all looked after. On the other side some attention went to some of the VOIP toolkits. One of the things that appeared was that most of the vendors were not providing any specific implementation details. Again looking for coca cola’s receipt? However for the microcontroller there were three different companies we could trace: MIPS, ARM and PowerPC. On the side of the DSP, there were again the Oak DSP and Teak DSP cores from DSP groups and the c54x series from Texas Instruments that appeared as good solutions.

Suddenly Ekta and I had to change our field of research again. This time we had to come up with a complete reference implementation of both the IP Phone and the gateway. The results can be found in the chapters 7 and 8.



Phase 4: Which building blocks to use for our products?This phase took over one month. Apart from the riots that kept us indoors for five days, the only decisions taken were those to go for Linux as the OS and, that an IP Phone, an 8 port IP Gateway and an IPPBX would be built.

The work at NirmaThe students in these days were doing research on the QOS protocols like RSVP and IntServ. They came up with the idea of using a combination of both which provides the best performance. They also were working on 30 pager documents to nail down all the specifications the gateway and the phone should have.

The IPPBX also referred to as the PC based Enterprise Gateway (PEG) after a while was shifted to be developed later. This was due to the growing time to market pressure; Madhubhai was receiving two phone calls per day from people asking him whether he had the products ready for production.

On the other hand still a lot of research was going on concerning the different protocols. Groups were restructured. Now there were two main groups of students, one for the phone and one for the gateway. Gateway tasks performed by subgroups for example on March 20 th

were:

Getting the listing of Open Source of H323 & others and study them Specifying the PEG software components & PEG board definition Billing & Call Data Recording module SIP team H323 team Open source DSP software and algorithm gathering

The different team findings were all discussed on Saturday meetings and most of them put a lot of effort in it to really understand the more complex codecs and protocols in depth in a way to be able to really do something use full with it on an implementation level afterwards.

The work at our endEkta and I started by comparing complete solutions on chip, or systems on chip for the gateway. Around 10 different options were compared. It was found out that the Monet architecture from Real would be a very nice product to go with. We also came across the TI’s TNETV2020 what appeared to be a complete gateway solution. After various attempts from our side it was not clear whether TI was actually producing the solution as a chip or not. Several emails were written awaiting an answer from their side.



Attention however was again shifted. This time a list of minimal features had to be made for both the gateway and the IPPBX. The latter one mostly contains extra software features like billing and statistics. This gateway specification was just a list of minimal features likely to have on our side. The results were all debated with the students on Saturdays.

Another task performed was listing the free protocols needed on our end according to the specified features. It became clear that most of the protocols were available. OpenH323.org is a huge initiative on the web where they provide almost all of the suite protocols we need for an H323 implementation. However the decision whether to go with the free protocols or not was not debated until more info from the chipset vendor was obtained and the decision was made which vendor to use.

On the 7th of April finally an important decision was made. Together with the students the decision was made to go with the ATMEL At75c220 processor for the IP Phone. The processor is based on the ARM7 core and uses the OAK DSP from DSP groups. Although TI’s C5471 could process more instructions per second the Atmel solution was cheaper and sufficient to its needs. For the complete comparison refer to appendix D.

Along with this the decision was taken to order the Development Kit (DK) that goes along with it. This DK contains μ-CLINUX a Linux flavor used on embedded DSP systems as well as a physic development board including the physical peripheral connectors like the RJ45 interface. Coming back to the free software or not. Most of the compression codecs could be obtained for 1 USD extra per chip. This option was taken since it would safe a lot of production time and energy.

At this time the project was able to start developing on the IP Phone thread. For the gateway however more information was needed to come to the same level. Work was continued on previous efforts to find chips that were useable for a gateway, this since ATMEL was not providing any.

Our research on the gateway was concluded on April 16 th. It became clear that the TNETV2020 from Texas Instruments was actually available. Since most of the Solution on chip (or SOC) options were providing too many concurrent channels and other market ready solutions like the Audacity t2 or the Azurite 3000 from Virata were no competition to TI looking at the solution they provided and according to Madhubhai also TI’s reliability was generally known as much higher than the other two players. Besides that the DSP’s used in both the other options were from TI anyway.

So this was the last major decision to take; apart from which IP telephony protocol to go with. The decision came quite soon. The protocol stacks with the H323 implementation from



Radvision for example were costing USD 20.000. This was too expensive. The decision was made to go with the free H323 stacks from the openH323.org project.

In the last week of the stay after sever contact with TI it became clear that apart from the TNETV2020 chipset which would cost around 75 to 80 USD. Along with buying those chips the buyer is forced to go with VxWorks as an operating system. This costs an extra 20.000 USD for universities and the worst of all was that the Telogy software that had to come with it would cost another 150.000+ USD. This was quite a drawback expecting that everything was included in the 86 dollars as TI told the first time.

Now there were three paths to follow for the gateway.

Asking Nirma to pay the software Making it an academic exercise to develop the software on the TI chip Going for a existing gateway and building value added service applications like billing

on top of that framework

During the last meeting at Nirma (April 20th), the students we opting for option one or two. This was dependant on the funding of the institute. Option one would go along with a lot of expectations and demands from the investors, on the other hand it would be a tremendous challenging project to work on and this would bring a lot of motivation. The marketing issue was also in the race for the third option. To that extend however too much of the energy that had been put in would be lost. The second option finally didn’t make it because the procedure would take to long and this would not motivate the students. Awaiting for money the decision was made to proceed with the phone development and to wait for the results of the funding negotiations to finalize the decision.

Phase 5: The later decisions and the future.At this point in time for the phone we made a great leap forward. A laboratory at Nirma was set up in order to provide the students with a development environment. The products were ordered and a project plan was to be made in MS project. The phone it self, or at least a prototype was to be completed before the graduation of the eight semester students in July so that they would have something concrete at hand before leaving the project.

In the next chapter the complete traineeship is evaluated



11. Evaluation Looking back towards the traineeship in general, there are many things that I learned and will bring back to Holland. This chapter is an evaluation of the company, the project and of India in General.

The companyLooking at the company there are three major aspects to consider, the office conditions, the products and the people working there.

The office has a nice look and feel and is modern even compared to western standards. It was a nice office to work in, this although the room where most of the employees work has small working space. The facilities were good enough. My computer having 64 MB RAM and 250 MHz processor was sufficient for the job. Besides that there was an A/C keeping the employees far from the 40 degrees outside and cool water was brought every half an hour or so.

Starting somewhere between 9:00 and 10:00 AM, the working conditions were flexible. But keeping in mind that the official working hours were from 9:00 till 18:00 with one hour lunch break from 13:00 till 14:00 it was weird to see that the lunch most of the time took 10 minutes and that most of the employees stayed until 19:30 or sometimes even until 21:00. Most of the times however, I left at 18:00. On suggesting having a 10 minute coke break downstairs the office manager didn’t allow us to go down for some reason. A proposed lunch together outside was immediately transformed into ‘maybe going out together for a movie once’. So to that extend the employees themselves did not know whether they were allowed. For my farewell however Parag wanted us all to go for lunch. So things were possible but the employees just didn’t ask!

Apart from that there was little interaction amongst the employees. Most of conversations were via MSN, on a personal level there appeared to be not much activity. On the other hand people were always willing to help each other out if one was stuck with something, but before they actually asked someone to help them too much time was spilled.

One other thing was working on Saturdays. Although considered a half day you were allowed to leave at 16:00. There were no regulations or facilities for employees for having days off and payments were compared to other IT companies in the city relatively low. This was all possible by the fact that Surya Infotech is a private company without stakeholders and responsibility to anyone except their customers.



Looking to the projects that were running in the office, some of the employees were working on more than one project. The employees however all had their own working place. By grouping the employees working places according to their project they would produce better and faster results. Since most of the conversations took place over MSN, it is recommendable to stimulate interaction by having a daily meeting to crystallize tasks and assure progress.

Looking to the products the company produces, it appeared that a lot of effort was put into their products in order to deliver good results. The ability to ‘think out of the box’ is highly stimulated and has led to various successful company products. This was a good exposure. Why walk the way most people walk if we can do it better? That mentality has brought them towards success. One example is the development of a modem over the power line; a system that has been patented in the United States.

What will remain in my mind forever are the people, the employees that made the company the company. Ekta my trainee buddy was good company and as far as I can qualify them, they are all nice and well educated people. For some reason they were all quite afraid of Parag, although he is very nice person they all referred to him as ‘Sir’ and whenever he said something they just did it, whatever demand there was.

The work the employees actually performed could be more. Many people spend at least one hour per day on chatting and checking email. This culture I adapted quite easily and this was in my eyes made up by the work on Saturdays and only one break per day. More deadlines and (control) pressure will certainly increase the output. Two weeks before I left however, a system was introduced that applied to this problem. Employees had to report daily their progress in a program that was meant to monitor the work performance.

The traineeshipLooking back to the traineeship it was a wonderful time, this although no single line of code was written. The idea of contributing to a big project with a huge potential market made the wish disappear. In the first week I really thought that the company was working on a mission impossible and it would require employees that have been in the field for years to be able to make the products they wanted to make. Due to the guidance of Parag and Madhubhai however, by the time that I left, the status of the project had been brought into a stage that the all the ingredients needed for the phone to be built were there, the board designs were made and the first prototype was expected by the end of July. A complete research had to be conducted on all the products to be designed. The specification of the 8 port gateway as well as the specification of the IPPBX had been delivered. A huge benefit was there for the company who had spent next to nothing on the research so far.



The students at Nirma faced some motivational problems in the beginning. There was one core group of students driving the car while others had to be motivated by the group leaders. This resulted in drop outs or sometimes relatively low progress from the groups. Progress was also frustrated when either Parag or Madhubai was not attending parts of, or even whole meetings.

It was good to be part of the whole process and contributing to it made me feel even better. In the beginning I felt really small compared to the industrialists and the best students of the state, but after a while I realized that without my performance the project might have gone into another, different way. All the components we decided on were chosen with the best arguments. Sometimes decisions were pending too long, but finally things turned out to be ok. Knowing exactly what we wanted to make, why we want to make it and how we were going to make it was compared to all the theory I was used to finally something real.

In the beginning the amount of reading that had to be done was really a lot. It was hard stuff to go through, confusing in the beginning and a lot of concepts had to be cleared. This again was not always easy. Much time was spent in searching on the net for explanations of terms and for finding additional and refined materials. The research conducted amongst the numerous amounts of products and vendors took a lot of time. However it finally gave an inside view of the architecture what later served as a guidance to make the decisions and designs from our side.

As far as the structure is concerned there were some big differences with the Netherlands. The whole project in fact started quite chaotic, there was no project plan or schedule or anything in the beginning. The only thing that was clear was that we were going to assist the students at Nirma from a commercial perspective, control their efforts and debate their options. This idea of the project was not known to me by the time I left Holland. The way of working after a while made me sometimes feel as if I was swimming in a lake without seeing any land. But after some guidance from Deepa we turned out to make sense out of the situation and provided the students with information and hands to push the development.

Nice was it to attend the meetings with Parag and Madhubhai as well as the meetings at Nirma, these meetings were sometimes quite unstructured and without any agenda resulting in the two industrialists to speak on different levels. Their approach however was a good one. I learnt a lot from their experience in the field and from their way of decision making. “Do not shoot from the hip, but aim and make sure you hit the target at once.” These industrialists turned into idols quite quickly.

The students of Nirma have gone through a process which results they will carry with them the rest of their lives. This also goes for me; it was a unique opportunity and journey to go



through; making such a complex product in a way that all the aspects were addressed. We did it all together and learnt a lot from it. With the valuable input from the two top industrialists it made it an experience never to be forgotten.

India in generalAlready from the moment arriving on the airport you realize that India is completely different from the Dutch culture. The impression was that the airport was on fire! After a while you realize that it is just pollution, nothing special.

A part that really attracted me was the relaxed mentality of the people: “No worry, chicken curry.” or “Maybe tomorrow” are really sentences hitting the hart of the Indian society. It took me a while to get adapted and start to enjoy the beauty of doing nothing at home and being in a restaurant for two hours. The best example is of course the traffic, at first sight it is complete chaos, but after a while you realize that most things go nice and smooth. On my bicycle I really enjoyed riding to my work everyday. The roads covered with camels, monkeys, pigs, dogs and lots and lots of cows made me realize that this other way, was a different way, but also a good way.

The huge amount of temples on the streets and all the different beliefs made me feel a real stranger. Why do they spend so much time on praying a day and what is the use of it at all? Anyway, on the other hand it led to several communal clashes. The situation during the riots was far from nice and quite scaring in fact. The governmental democracy is fragile and some say that it is based on a divide and conquer strategy. The poverty and corruption I’ve been exposed to made me realize how fortunate we are in Holland. But let’s not forget that India also has a lot of good things to offer. The beautiful nature and architecture and the different colors on the always lively streets make the country a lust for the eye and absolute worth exploring.

The riots confronted everyone with the fact that politic and governmental organizations were not functioning similar to the ones in Holland. Bribery and caste or friend politics are daily matters. During a village trip it even shocked the local AIESEC’ers that bonded labor, caste exploitation and discrimination exists on large scales nowadays. Mostly Ignorance and disbelief prevail in the higher casts and the government. “Why should they change it if it is in their own benefit?” is a matter preventing governmental institutes from serious action. Luckily lots of NGO’s are fighting for the rights of these people and due to their efforts the combat to change the inhuman mindset and to raise awareness amongst the public has begun.

The time with AIESEC and the traineesFrom the moment of arrival in Baroda, until the day that I left, the AIESEC’ers have been great company and were helping when- and where-ever they could. On January 29 th 2002 at



6:00 am in the morning, while making my first steps on the Barodian soil, the AIESEC’ers provided me with a warm welcome and introduced me to the city where I had to spend the next three month.

During these first days I was lodged in a huge house of the family of one of the AIESEC’ers. The family looked after me very well and it was really great to with them in for me a complete new and different culture. The day after my arrival work started at the company and after a couple of days they shifted me to one of the two trainee houses, where I stayed during the rest of my traineeship with five other trainees coming from all over the world

Together with the trainees AIESEC arranged a lot of great activities. Country presentations, sumptuous dinners, the weirdest parties, cinema evenings and other activities were held almost on a daily basis. In three weekends touristy places were visited and more of the countries richness was enjoyed. Obliged working on Saturday on the other hand prevented us most of the time from traveling in the weekends. Sometimes the AIESEC’ers showed up everyday, this according to all of the trainees really was too much. Another thing that was surprising was the goal of getting as many trainees to Baroda as possible and the mindset of willing to be rewarded by hard numbers seemed sometimes more important than the state of happiness of the trainees or even of their own team.

The time with the trainees however was a nice one; people from all over the world living in the same house was the basis for a lot of good activities, hot discussions, a lot of hilarious moments and of course a lot of fun! Most of the trainees have become good friends and learning from their approach to several things was excellent. We can come to each others places all over the world whenever around. One thing that was unavoidable and nice on one side but not so nice on the other was that I was not the only Dutch trainee around. This made us stick to each other too much sometimes and prevented us from interacting with the others.

Quite disturbing from the AIESEC’ers was that they sometimes prevented the trainees from getting the right information and told that things were not as bad as they really were, especially during the riots. Of course it is understandable that they wanted to give all the trainees the time of their lives and that was appreciated, telling the truth however, especially in this riot case would have been better by any means. Also their notion of time seemed to be hardly developed. Arriving late, sometimes up to an hour, did result in painful situations. Many activities were cancelled or announced far and far too late. This by the end really harmed the atmosphere amongst trainees and the AIESEC’ers.

But due to the many great AIESEC activities however the trainees got closer to the Indian culture. All the AIESEC’ers were very nice people, they brought us to the right places and restaurants and explained about the culture and the religion. In the last week the lecture from



a university professor and the village trip through some tribal villages were absolute superb experiences and they were very educative.

One other thing to mention is that there is hardly anything to do in an Indian city like Baroda. Apart from pool parlors or cinemas, it was tough to find some other excitement in the city. (No discos, no bars, no alcohol) Of course a lot of time was spent indoors chatting with the other trainees in the evenings, but sitting at home everyday could get quit boring after sometime. This was however also the charm of the country. The Hindi movies were very nice to watch and they provided the right mechanisms to make live survivable in the Baroda. The party atmosphere was most of the time really a weird one but taking it all together: exposure to a lot new idea’s and people as well as learning a lot made it a blast of a time!



12. Conclusion

After three month of work the project is in a state that the implementation just started. The labs are being set up and products are being bought. The decision phase has ended by deciding to build three top notch VOIP products all focusing on their own big market niche in India.

An IP Phone will and should be developed around the ATMEL AT75C220 chipset based on a 40 MHz ARM7 processing core and a 60 MHz OAK DSP form DSP groups. The IP Phone will have all the specifications that evaluated during the development process as defined in APPENDIX F

An 8 port IP Gateway should be developed around the TI VNET2020 gateway solution from Texas Instruments containing a 100 MHz MIPS R4000 processor and two 133 MHz TMSC5441 DSP’s The IP Gateway should have the specifications that evaluated during the development process as defined in APPENDIX E. Implementation is dependant on funding issues.

An IPPBX gateway will be developed in a later stadium based on the specifications that evaluated during the de development process. The major functions will be billing, call lookup, IVR and voicemail.

All the decisions made are well considered options based on rational arguments with input coming from Madhubhai and Parag, as well as from the research performed by Ekta and Me and the students from Nirma.

It was a good project to work on. The people and the environment gave it a special flavor that would have been unimaginable in The Netherlands. Furthermore I sense that I definitely contributed in a relevant and sometimes essential way to the research, the debates and the final decisions taken. Besides that, there was a lot that I learnt in India and it really opened my eyes to the world and the working field of IT.



13. RecommendationsHere are recommendations listed related to the project, the company and AIESEC.

The project Looking at the complexity of the project itself, from the company side, it might have

been better to have some experienced people involved who knew more about the topic in advance rather than having everyone to start researching from scratch.

Planning and work with deadlines to force certain decisions and prevent pending of other important issues would have assured the progress of the project also in the early stages. It is advisable to take a more schematic approach to achieve a faster result.

To prevent meetings between Madhubhai and, Parag and his employees from getting to chaotic and unstructured I would advice to make a thorough preparation in advance of the meetings to make sure everyone talks on the same level.

A clearer formulation of tasks and addressing possible goals and achievements should be given to the students to motivate them, to let them work harder and improve their achievements. Make sure all the students are working on the project and not a small group. Make more specific tasks and issue them accordingly.

Also a thorough preparation before the Saturday meetings at Nirma is advisable to keep the students sharp and willing to work in order to guarantee progress.

Prevent the absence of the leaders during meetings, or address clearly a substitute, it will delay the progress otherwise.

The company

Increase of salary and adjusting to strict working hours as well as written regulations and contracts, larger workspace and more collective breaks might result into higher motivated employees and increased productivity.

Assuring progress will keep the employees from getting distracted by means of chatting or useless browsing.



Define tasks clearly and do not alter them too often or too suddenly in order to prevent confusion. People working on the same project should sit together and have progress discussions on a daily bases.

The employees should have a more relaxed attitude towards Parag to smooth uncertainties. Complaining about problems sooner will result in solving the problem quicker.

Assure social interaction by meetings or activities to make them one big team with a common goal.

AIESEC

Prepare the trainee better on the traineeship and the associated conditions to prevent misunderstandings or misperceptions.

Organize more background or in depth activities like seminars or debates rather than just parties to give the trainee more cultural exposure

Do more different things instead of the same things too often. Things might get quite boring.

Be on time! It is very annoying for all the trainees to promise to be on time the next time and (again) show up one hour late even taken into account that it is a cultural difference.

Do not announce activities too short in advance and sudden cancellation of activities should be avoided to guarantee a relaxed and friendly atmosphere amongst the trainees and the AIESEC’ers.

Always talk the truth to the trainees; it is really disturbing for trainees to find out that what you told them was far from the truth.

There is no need of meeting trainees everyday, they don’t need to be amused every hour they are free. However, if there is an activity scheduled inform them at least three days in advance to avoid a low attendance number.

The goal of AIESEC Baroda should not be to exchange the highest number of trainees; the goal should be to attain a mental state of satisfaction and happiness



even without watching the numbers. Show more openness and interest towards the trainees, this will result in more fun and satisfaction!



14. Literature en references*

[@01] The CIA world fact book http://www.cia.gov/cia/publications/factbook/

[@02] The Hindustan Times, main site http://www.hindustantimes.com

[@03] Atmel’s site for VOIP products and the Atmel AT75c220 chipsethttp://www.atmel.com/atmel/products/prod33.htm

[@04] Intel, main site http://www.intel.com

[@05] Texas Instruments, main site http://www.TI.com

[@06] The Internet Engineering consortium tutorialshttp://www.IEC.org/online/tutorials/

[@07] The site for Fax and voice over IP news and market researchhttp://www.IPtelephony.org/

[@08] University of Souther Carolina, RSVP overview,http://www.isi.edu/div7/rsvp/overview.html

[@09] The open H323 project, main sitehttp://www.openh323.org

[@10] Nirma Institute, main sitehttp://www.nit.edu

[@11] AIESEC International, main sitehttp://www.aiesec.org

[@12] Microcontroller manufacturer ARM, main sitehttp://www.arm.com

[@13] Microcontroller manufacturer MIPS, main site


http://www.arm.com/

http://www.aiesec.org/

http://www.nit.edu/

http://www.openh323.org/

http://www.isi.edu/div7/rsvp/overview.html

http://www.IPtelephony.org/

http://www.IEC.org/online/tutorials/

http://www.TI.com/

http://www.intel.com/

http://www.atmel.com/atmel/products/prod33.htm

http://www.hindustantimes.com/

http://www.cia.gov/cia/publications/factbook/


http://www.mips.com


http://www.mips.com/


[@14] DSP groups, main sitehttp://www.dspg.com

[@15] Internet Engineering Task Force, main sitehttp://www.ietf.org

[@16] Surya Infotech, main sitehttp://www.suryagroup.com

[BO00] Y.Boger (Radcom Ltd), White paper: ‘Fine-tuning voice over packet services’, p2

[IN01] K. Shetty (Intel Corp.), White paper: ‘Embedded Intel architecture in the carrier class Voice over IP Gateway’, Part Number: 200011-002, June 2001, p8.

[CR00] S.Christensen (Juniper Networks), Whitepaper: ‘Voice over IP Solutions’, p 5-9.

[MO00] S.Mourad & H.Eladb (Realtech communications Inc.), ‘Communication system on a chip’, p1-10.

[MC00] B. Mc Connell (HelloDirect.com) ‘A comparative analysis of next generation phone systems’.

[SH00] T.Shughart (Analogic corporation inc. ), ‘White Paper : Voice over IP networks’ ,p6

[WIT98] William E. Witowski (Telogy networks Inc.), ‘IP telephony and implementation Issues’, Version 2.1, July 1998, pp 2-12.

*) During the research conducted many more papers and websites were visited. Only the most relevant ones are listed here.


http://www.suryagroup.com/

http://www.ietf.org/

http://www.dspg.com/


APPENDIX A “ A comparison between SIP and H323”

IntroductionThis research has been done in order to distinguish amongst the H323 and the SIP protocol. Both of them are used for implementing Voice over IP (VOIP). The focus differs from reliability on one side until the PSTN (normal telephone network) integration on the other side. After reading this document you will be able to make a proper decision in choosing the proper protocol for your VOIP implementation.

Introduction to H323This Packet based multimedia communication system recommended by the ITU-T is a version of the H.320 Multimedia-over-ISDN standard optimized for packet-based networks such as TCP/IP. Although H.323 is not specific to IP (it could also be used with IPX or AppleTalk), it relies on some IETF technologies--most notably RTP (Real Time Protocol) and RTCP (Real Time Control Protocol).

Introduction to SIPAnother VOIP standard in the works is SIP (Session Initialization Protocol). SIP offers many of the same architectural features of H.323, but relies on IP-specific technologies, such as DNS, as well. It also incorporates the concept of fixed port numbers for all devices and allows the use of proxy servers, both of which ease firewall implementation concerns. The Proxy server itself is able after connecting the location server to assure the called party will respond to the caller, this will relief the caller. Another point is that it is quite a simple protocol that can be implemented in two ways, with and without a proxy.

BackgroundH.323 was designed with a good understanding of the requirements for multimedia communication over IP networks, including audio, video, and data conferencing. It defines an entire, unified system for performing these functions, leveraging the strengths of the IETF and ITU-T protocols. On the other side there is not much need for the rich video features of H323 in VOIP.SIP was designed to setup a "session" between two points and has a loose concept of a call. There is no conference control function here.

ReliabilityThere are many features in H323 that handle failure of network entities. If the gatekeeper fails, the protocol will search for another. Besides that it has a re-router call, re-routing the call if one router fails so that the call is not disrupted.



These features are not implemented in SIP. This means that if a proxy fails, the SIP user agent has no means to detect this.

ExtensibilitySIP can be extended with new header lines; this is risky on interoperability issues in a means of each building its own headers. H323 can only be extended with non standard features, running a lower risk.New revisions of H323 are published now and then having new features and updates. These however will always take care of backward compatibility. This policy might considerably slow down the system after a few years. SIP updates on the contrary are not backwards compatible; this makes it sometimes mandatory to update the system and is therefore vulnerable for interoperability problems.

ScalabilityLooking at load balancing, the H323 protocol has a feature to load balance endpoints across a number of different gatekeepers, in order to scale a local point of presence. SIP does nothing with direct load balancing, it merely requires on DNS or HTTP load balancing. This is risky incase a gateway is completely utilized. Calls might fail in this way.For call signaling H323 uses RAS for address resolutions. In large networks the direct call model might be used to setup a connection directly. Looking at address resolution in SIP protocol, at least three message exchanges have to be exchanged by the proxy to set up a call. The proxy is free to use any protocol in order to discover the address of the one being called (TRIP, ENUM, DNS…). This also goes for the gatekeeper in H323, so the endpoints have no concerns of how the address is found. The difference is again the three mentioned messages that have to be exchanged between the endpoint and the proxy in SIP and the single message that has to be exchanged in the H323 protocol between the endpoint and the gatekeeper. H323 allows the endpoint to do his own address resolution as well. It is therefore able to send the address as an alias to the gateway.Focusing on addressing, H323 is able to address in many more ways than the SIP protocol, instead of using only URL addresses like SIP, the H323 protocol is able to alias e.g. email, UIM mobile and transport addresses as well as E 164 dialed digits.

CapabilityFocusing on capability, H323 entities are able to negotiate on which channels to open for a connection. Individual channels might be opened or closed without loosing the connection or disrupting the other channels. SIP entities can only propose a set of channels towards another. The receiver of this proposal is limited to choose one of these.



Message definitionThe ASN1 message definition is a structural and easy to understand precise notation and is used by many other systems. The coder and the decoder can be generated automatically at changing specifications they only need to rerun the compiler in order to get the new coder and decoder.SIP uses a form of ABNF, or Augmented Backus-Naur Form, which is not standardized. Besides that the codec can not be generated automatically. When the ABNF specification is changed the codec has to be rewritten.

Message encodingThe messages in H323 are encoded in a binary so that it is compact and easy to use on both broadband as narrow band connections. SIP encodes in an ASCII way making it easy to read by humans, but harder to transfer over narrower bandwidths. H323 is likely to be faster because of the ASCII encoding of SIP.

BillingThe ability to bill correctly in H323 is certified because of the endpoint. It reports to the gatekeeper the beginning and end time of the call via the RAS protocol. If the SIP proxy wants to collect billing information, it has no choice but to stay in the call signaling path as long as the call lasts. In this way is it able to detect when the call completes. However, the call signaling may have been delayed here due to congestion. There is no exact time reported in SIP. Besides that the BYE massage is not explicitly required. In H323 the call is aborted in a well defined manner.

Call setupThe call setup for H323 and SIP can both be performed in 1.5 roundtrips. When there is an audio cut through in such a way that there is intermediate response, the setup in H323 takes five messages and the cut through only one. For SIP, the cut through takes two messages and the setup will last for six. H323 is using TCP as a reliable channel for signaling, this might evolve in several more messages being send. H323 Annex E however allows the use of the unreliable UDP channel. Sip however is able to prioritize among calls, this is not possible in H323

PSTN IntegrationFor integration with the traditional PSTN lines The H323 protocol uses the PSTN Q931 protocol in a way that integration is relatively easy. H323 however not employs the circuit switch technology. SIP on the contrary is able to do this but has further no commonality with PSTN and signaling must completely ‘translated’ into SIP. SIP has no architecture that describes the decomposition of a gateway into the media gateway controller and the media



gateways. H323 on the other hand defines very well how Media Gateway Controllers fit in its architecture.

ConclusionIt is certain that the H323 protocol is far more sophisticated than the SIP protocol. However, it must be kept in mind that regarding implementation, decisions have to be made according to meet the desires of the project. For a simple office VOIP network with a low budget SIP will certainly suffice. On more complex projects involving a 1000 telephones or more H323 will be stronger regarding for example reliability and billing



APPENDIX B “A comparison amongst MEGACO gateways”

Supplier Version type

Protocol support

Nr of links supported

Extra component

needed

Functions of the gateway

Manag-ement

Media conversion

Supported codecs Extra

Audiocodes Mp 100 MGCPH323MEGACO

8 trunk lines(analog)

To be integrated with a IP/PBX gatekeeper or call manager

connectinglegacy telephone systems with new IP-based PBXarchitecture

SNMP,HTTP

From analog phone or PBX to IP

G711,G723.1G726,G729ANetCoder

Audiocodes Mp 104,108,124

MGCPH323MEGACOSIP

2-24 trunk lines(analog)

To be integrated with a IP/PBX gatekeeper or call manager


SNMP,HTTP

From analog phone or PBX to IP

G711,G723.1G726,G727,G729ANetCoder

Audiocodes MP 200 MGCPMEGACO

4 E1/T1 linesup to 120 ports

To be integrated with a MGCP Call manager

Bridge between a corporate PBX and an IP network for interoperability with other control protocols

SNMP,HTTP

Transformation amongst control protocol types

G711,G723.1,G726,G727,G729A,NetCoder




Protocol support

Nr of links supported

Extra component

needed


Manag-ement

Media conversion

Supported codecs

Extra

Blue Lava TPC 300 MGCP MEGACO H.323 v2

VOIP API Library

2-24 trunk lines(analog)

NO, but to be integrated with a IP/PBX gatekeeper or call manager


SNMP, HTTP

From analog phone to IP

G711, G.723.1, G.726, G.729A, Net Coder

Selectable coders per channel:Analog to Digital converter

Blue Lava TPC 500 MGCP MEGACO H.323 v2 ·

VOIP API Library

4 E1/T1 upto 120 ports

NO, but to be used with a call agent and a corporate switch

Bridge between a corporate PBX and an IP network for interoperability with other control protocols

SNMPHTTP

Trasformation amongst control protocol types

G.711, G.723.1, G.726, G.727, G.729A, Net Coder

Comm Works

Total Control 2000

MegacoSIPH323

14 DS3 linecard interfaces

Part ofCommworks

3 tier architecture

Integration of diverse traffic types across multiple access networks

CLI, SNMP, element-manager

From PSTN to packet based networks

G.711, G.729aT.38 (fax)

Up to 1 million busy call attempts, for a nationwide network

Comm Match

Duet 6001 H323V2 16 E1 upto480 ports

Class 5 switch for processing,

To provide IP end users with the legacy services of

CLI, SNMP, element-

G.711, G.729A, G.723.1, G.728,G.726 and

ISDN over IP support



billing and admini-stration

the PSTN network manager G.727


Protocol support

Nr of Links supported

Extra component

needed


Management

Media conversion

Supported codecs

Extra

Comm Match

Duet 6002 MGCP 16 E1 up to480 ports

Class 5 switch for processing, billing and administration

Providing the IP end phones with the advanced legacy features of PSTN

CLI, SNMP element manager

G.711, G.729A, G.723.1, G.728,G.726 and G.727

Especially for voice over cable TV. ISDN over IP support


APPENDIX C “A comparison amongst IP Phones”

PHONE Chip DSP OS Libraries Stand-Alone?

Remoteconfiguration

Extra

Applio IP PhoneARM7TDMI based 32 bit RISC CPU up to 20 MIPS

G723.1, G729, G711, Silence compression and echo cancellation

Linux TCP/IP, RTP/RTCP,DHCP, NAT, Ethernet Interface, H323 V2, H 450, MGCP, SIP

Yes DHCP 2 dsp’s enabling two simultaneous phone calls

Freeride 200G.711, G.729 (G.723.1 optional) echo cancellation, configurable jitter buffer, silence compression

H.323v2, (802.11b, SCCP, MGCP, SIP optional)

DHCPFTP

FreeRide was the first standards compliant IP Phone on the market.2x 16 LCD

netergy Audacity-T2 processor IC- Unified RISC/DSP

G.711, G.723.1, G.729A/B- G.165/G.168, echo cancellation, voice activity detection

Microelectronics' POSIXcompliant OS(linux based)

H.323v2 OR MGCP, OR SIP callcontrol protocols

HTTPH.323v2 - requires an H.323v2 Gatekeeper, development platform included

Nortelsi2050

Pentium pro200 with 64 mb(win98) or 128 mb ram(win2k)

G.711,G.723,audio interface ,jitter buffer

Windows98,98se and 2000

MGCP, SIP,h323, PSTN devices.

NODHCP

This is an software phone for in a computer

Tnet v1001 Dual CPU processorARM7TDMIRISC integrated with the TMS320C54x

100MHz, low power,G711, G.729AB or G.723.1A . echo cancellation, configurable jitter buffer, silence compression, lost packet compensation

H323 or MGCP or Simple basic call support. API’s to acess DSP

Possible Tnet v1001

Freeride 210 G.711, G.729 (G.723.1 optional)echo cancellation, configurable jitter buffer, silence compression

H.323v2, 802.11b(SCCP, MGCP, SIP optional)

DHCPFTP

FreeRide was the first standards compliant IP Phone on the market.2x 16 LCD


APPENDIX D “ATMEL and TI IP Phone chipsets compared”

Product ATMEL AT75C220 TI TMS C5471

DSP One 16-bit Fixed-point OakDSPCore® Core Running at 60 MIPS

TMS320C54xDSP core running on 100 MHZ

MicrocontrollerARM7TDMI® ARM® Thumb® Processor Core Running on 40 MIPS/MHZ

ARM7TDMI CPU core (32/16-bit RISC processor) 47.5 MHZ with extended emulation capabilities

TimersThree 16-bit Timers/CountersAdditional Watchdog Timer

Three timers (two general-purpose, one watchdog)

Ethernet connectionDual Ethernet 10/100 Mbps MAC Interface with Voice Priority

Ethernet 10/100Base-T interface

Memory256 x 32-bit Boot ROM88K Bytes of Integrated Fast RAM

16K x 16 General purpose RAM24K x 16 loadable program RAM2K x 16 X RAM2K x 16 Y RAM

72K x 16-bit internal SRAM (100 MHZ) organized as 32K x 16-bit of data SRAM and 40K x 16-bit of program SRAM. (16KB)

On-chip 16K-byte (4K x 32) zero wait-state SRAM on ARM.

Shared memory between microcontroller and DSP

256 x 16 Dual port mailbox ARM port interface (API) to provide access by the MCU to 8K x 16 of the DSP’s data SRAM

ControllersOn-chip SDRAM Controller for Embedded ARM7TDMI and OakDSPCore

6 channel Direct memory access (DMA) controller.

Interfaces 2 USARTs with FIFO and Modem Control Lines1 JTAG Debug Interface1 SPI

2 UART’s (1IrDA)1 SPI1 I2c (master only)2 McBSP1 MII port1 JTAG boundary scan

Used Voltage2.5V Core and the PLL Pins3.3V for Other I/O Pins

1,8 Volt supply for the core3,3 Volt Supply for the I/O

I/O pins Up to 24 General-purpose I/O Pins MCU general-purpose I/Os ( 36 General Purpose I/Os),including support for an 8 x 8 keyboard.

Software availabillity Software Development Suites Available

AT75C220DK-SMEC costs 1250 USD

Supported by a Wide Range of Ready-to-use Application Software, Including Multitasking Operating System, Networking and Voice Processing

Software development suite available

Spectrum digital’s ‘development Suite 2999 USD

Supported by a Wide Range of Ready-to-use Application Software, Including Multitasking Operating System, Networking and Voice Processing


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Package 208-lead PQFP Package or in 256 ball PBGA

257-BALL Microstar BGA (GHK) package

Extra featuresMulti-layer AMBA™ ArchitectureFlexible External Bus Interface with a Programmable Chip Selects, a Codec Interface and a Multi-level Priority, Individually Maskable, Vectored Interrupt Controller

Programmable wait-state generatorProgrammable Hardware timerExternal memory interfaceMCU memory interface for external SRAM, Flash, ROM, and SDRAM.MII portPLL Clock generator and controlInterrupt HandlerAPI Boot mode

Price >5ku 14.7 USD>10ku 13.2 USD>50ku 11 USD

> 10ku 17.57 USD


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APPENDIX E “The Gateway Specifications”

General Features8 simultaneous voice / fax channelsITU- H323 compliantGateKeeper Mode / Peer-to-Peer mode settingsE.164 dial plan (user configurable)User Programmable Call Progress tones settingsUser Programmable Gain / Attenuation settingsTFTP/FTP software upgradeCDR (Call Detail Record) supportRemote / Local ConfigurationSupport for Fix IP and DHCPSupport QOS by setting Type Of Service parameters

for VOIP packets (RSVP, IntServ)LED indication for system statusFall Back to PSTN supported

Interface8 RJ-11 FXO/FXS Loop Start Telephone InterfaceStandard 10/100 Base TX RJ 45 Network InterfaceRS 232 DB9 Terminal Interface

Audio featuresRTP/RTCP Voice StreamG.711, G.729AB, G.723.1 Voice Compression CodecsG.729 Annex B, G.723.1 Annex-A based Voice Activity Detection (VAD)CNG (Comfort Noise Generation)G.168/G.165-compliant echo cancellationDynamic Jitter Buffer MechanismDTMF Tone Detection / GenerationCaller ID Detection / Generation Call Progress Tones Detection / Generation Supports H.450 supplementary services (Transfer, Hold, Forward)

FAX

SupportAutomatically FAX DetectionT.38 / Group3 FAX support at various speedVarious Modulation Formats supported

Voice ProtocolsIPv4 based H323 stack which includes H.225 Call Signaling


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Management FeaturesCom port managementSNMP supportTelnet supportWeb based Interface

LED Indicators for system statusReady :Power On/Off IndicationLink :Ethernet Line StatusData :Data Transfer Status8 LEDs :Individual Channel Status


H.245 Control SignalingH.225 RASFast Connect- H.245 tunneling


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APPENDIX F “The IP Phone Specifications”

User Interface

Display :4 line 16/20 character backlit LCD

- Support for upgrades up to TFT- Contrast Control supported (menu)

Ringer- Audible indicator for incoming calls- Volume (menu)- 5 basic ringtones - Support for downloading ringtones from the net- User configurable ringtones ( allowed by the mixing of existing ones)

Input Interface- Basic keys for dialing (0-9)- Additional Keys: Mute,Redial,Tranfer,Conference,Speakerphone,Flash

Memory dial, and Dial- Menu driven environment

- Menu key + 4 arrow keys- Features in menu:

Date/TimePassword/Authentication (3* levels)Call LogsCurrent Call status & Cost Billing Support Billing Information Display Directory Search (Using LDAP)Upgrade via InternetContrastRinger Volume

Handset with an RJ8 jack inVolume SliderVoice interface: Digitization of analog signals - G.712 (16 bit PCM)Gain Control (-10 dB to +10 dB) Speech compression

- G.723.1 (5.3kbps and 6.3kbps) [DSP]- G.729 AB (8 kbps) [DSP]- Voice Activity Detection (silence supression) [DSP]- Silence Insertion Description support [DSP]

FDSP module- Acoustic Echo Cancellation (G.167) [DSP]

-Noise Burst Estimation of Environment [DSP]- Doubletalk Detection [DSP]- Noise Floor Elimination [DSP]- Participant Noise Balancing [DSP]


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Packet Playout Unit- Adaptive/Configurable Jitter Buffer [DSP]- Lost Packet Compensation [DSP]- Comfort Noise Generation [DSP]

Real Time Diagnostics- QOS measurement- Codec Switching- Jitter analysis

DTMF generation and Detection [DSP]

Voice Encryption Issues *** for support in future versions

Network Interface:Interface to the IP: Ethernet MAC (RJ45 Port)Interface to the PSTN: RJ11

The following protocols fulfill all our requirements- TCP - IP, UDP- RTP + RTCP- DHCP (IP allocation both manual as well automatic)- TFTP- LDAP

Call Setup and supplementary services are handled by all of these protocol stacks:- H.323 v 3- H.450- H.225- H.245- H.235- Q.931- T.38

Miscellaneous:

USB 1.0 interface (printers, web cams, PC)


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