Integration of Voice over Internet Protocol Experiment in Computer Engineering Technology Curriculum

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1 Integration of Voice over Internet Protocol Experiment in Computer Engineering Technology Curriculum V. Rajaravivarma and Farid Farahmand Computer Electronics and Graphics Technology School of Technology, Central Connecticut State University 1615 Stanley Street, New Britain, CT Cajetan M. Akujuobi Center of Excellence for Communication Systems Technology Research (CECSTR) Electrical Engineering Department Prairie View A&M University P. O. Box 4078 Prairie View, TX Session Topic: Engineering Technology/Community College Abstract The Internet in its simplest form is a worldwide communication network consisting of a vast connection of computers. Since Internet data traffic is growing much faster than voice traffic, the idea of transporting voice over data networks has gone farther than the more traditional data over voice networks. These networks of computers, which have revolutionized practically every sector of modern world, are ideal for transmitting data. When the internet protocol supports voice communications, it is called Voice over IP or simply VoIP. VoIP can be defined as the ability to place telephone calls and to send facsimiles over IP-based data networks with suitable quality and better rate. The most common forms of VoIP can be from inter-office intercom to multi-point teleconferencing. This method of communication has become very attractive and advantageous because of its low-cost, flat-rate pricing, and can be integrated in the existing Intranet LAN infrastructure. So, many organizations, including corporations and universities, are migrating to VoIP by integrating their IT infrastructure. Consequently, learning and understanding VoIP technology in classroom can be considered as a valuable academic investment. The first part of this paper will focus on the basic concept of VoIP communications. Then the interaction of the Internet Telephony and the public telephone system technologies and the technical issues will be discussed in detail. The second part of the paper will focus towards the classroom experiment of VoIP. This is done by using minimal hardware and available open-source software required to demonstrate VoIP communications in action. As a result, students will learn (1) how to set up a voice server; (2) how telephone numbering scheme is implemented and how phone calls are routed; (3) how different protocols are used to convert voice into packets and send them through the Internet between multiple users. Possible expansions to this experiment are also briefly discussed in this paper.

2 Basic Concept of VoIP Communications Telephony has traditionally been used in voice carrier transportation. Telephone carriers used circuit switching method design to carry voice only. Circuit switching is not capable of effectively supporting data oriented applications such as instant messaging, video, and the World Wide Web. The main advantage of VoIP is to integrate voice and data applications. In using VoIP, your personal computer can become your personal voice line. The telephone, PC, fax machine, etc. are all implemented into one. Therefore access lines are dramatically decreased. VoIP supports nearly all the features of a traditional phone system 911 Support Call ID Voic Music on hold Three way calling The level of acceptable voice call quality is in which participants can communicate effectively without difficulty. Speech frames using packet switching design are normally transported with IP packets in VoIP. Multiple speech frames are carried in a single packet. Users normally run into some type of delay on international calls. The telecommunications companies test the quality of the VoIP s regularly. Test labs such as AT&T Labs, COMSTAT and Lucent perform such evaluations. Users also may run into something called jitter. It is a variation of delay. A packet in a voice conversation can experience longer queuing times. Jitter is not an issue with some networks. Jitter buffer and voice suppressors have been developed to resolve such issues. Jitter buffers simply make sure that all packets are delivered to its network in a steadily fashion. In using jitter buffers, speech packets are buffered in order to receive audio as well as video in a steady state. Silence suppressors are also a new service and requirement for VoIP. Bandwidth must be able to support any voice traffic in the network. If there are two users on the same network and the bandwidth support is not the same, there may be some problems with delay. They have been developed in order to reduce bandwidth usage. During a conversation, there are moments of silence. Coding techniques have been developed to utilize suppression such that traffic is passed over the network only when something is being said. In traditional systems, silence was transported thus wasting bandwidth. On the other hand, silence suppressors detect moments of silence and the bandwidth is not used. This allows the bandwidth span to be cut dramatically. Now standards such as H.323 and G.711 have created silence suppressor to filter out the silence. VoIP networks must accept signaling and media from circuit switched networks. In doing so, signaling must pass through gatekeepers. For many years this idea was held up under the Signaling System 7 (SS7) architecture. Now the current standard used is the H.323. It mainly deals with description of audio/video and media streaming networks. It involves the usage of gateways. H.323 terminal network supports Fast-Connect

3 procedures. The network gateway function allows the user to make a call using either an IP phone, PC, or fax to make a call. The call is then sent through the Internet telephony server and across the Internet. Through transmission the IP address is located and taps into the Local Area Network and the call is connected. The good news is the call is considered to be a local call no matter the location. VoIP Classifications and Benefits VoIP can take place between various types of end-to-end systems. Table 1 highlights each method and a brief explanation. Table 1.Types of End-to-End Systems Type of Endpoints PC to PC Telephone to Telephone Telephone to PC PC to Telephone Explanation Benefits include cost savings for long-distance calls Software from MSN, Yahoo messenger can be used for this method of VoIP. All you need is an Internet connection, sound card, microphone, speakers, and software. Using gateways, users can connect with any telephone in the world. The user has to call the gateway first and then dial the number they wish to connect to. The rates are much lower than standard long distance call. Companies provide calling cards or special numbers that allow users using standard telephone to call a user with PC. The PC user must have the vendor software installed and running. This is the more popular option. Software is typically available freely ( e.g. Net2Phone ) The main attraction of VoIP technology is utilizing the existing data communication network (the Internet connection) within an organization in order to support voice services. Hence, while providing reliable and high-quality service, VoIP can potentially be cost-effective. In fact, today VoIP is being widely deployed by many carriers, particularly for international telephone calls. Such services are completely transparent to customers and users are unaware that their telephone call are being routed over IP infrastructure for most of its distance, instead of entirely over the circuit switched public switched telephone network (PSTN). [1] VoIP is also attracting many companies in order to eliminate call charges between their offices. Using VoIP they can utilize their data network to carry inter-office voice calls. Furthermore, by implementing VoIP, companies can potentially reduce the costs of calls outside the company premises. In such cases, the calls are carried to the nearest points on the data network before being handed off to the PSTN and eventually to the end-user.

4 The major benefits of VoIP can be summarized as follows: 1. Reducing the cost of long distance and international calls. 2. Decreasing hardware expenditures as well as the support and administrative costs of managing both a voice and a data infrastructure. 3. Allowing remote office workers and mobile employees to take advantage of all corporate telephone capabilities, including the ability to make and receive phone calls at their assigned extension. Facing such opportunities, we believe, IT and technology students, who have been introduced to the basics of VoIP and understand its advantages and limitations can be very valuable in the future competitive job market. Figure 1. Connecting LAN to the PSTN using

5 Classroom Experiment Figure 1 shows a generic VoIP network which is interfaced with the PSTN. Motivated by this setup, in our experiment, we limited the network to a few clients (PC terminal) interconnected to a voice server. The client computers where loaded with Windows 2003 operating system. On the voice server we installed Linux-based operating system, which can be downloaded for no charge. essentially provides similar core functionality as a Caller Manager in a VoIP network. They process all calls and provide a database of the IP telephone /device MAC address, which are mapped to IP addresses of the machines in the network. In order to provide call monitoring, we installed Stat-2.0. This add-on software perform a variety of call monitoring functionalities, such as call duration, call destinations, caller ID information, and call time information. Using this tool, students can experiment with basic call management techniques, and call analysis. Furthermore, the students can examine the voice quality of each call as they disturb the communication line characteristics. Error! Reference source not found. provides a snap shot of features available in Stat-2.0. Motivated students can download other call monitoring utilities with mode sophisticated features. A number of VoIP analysis tools are listed in [4]. In the second part of the experiment, we asked our students to perform cost analysis for an imaginary organization investigating the possibility of migrating to VoIP. Using an on-line tool provided by [5], each student was asked to consider different scenarios, with various parameters, including the number of users in the organization, phone usage, geographical disparity of the users, etc., and investigate equipment cost as well as the potential return-on-investment. VoIP Implementation Details In this section we describe the implementation details of our experiment. The setup included 3 x86 based PC s one running the OS and the other 2 running the MS Windows 2003 Server OS. Then following steps were performed. 1. Insert the Operating System CD and start the computer. 2. When prompted hit the enter key to begin OS installation. While waiting for the installation screen begin installing MS Windows Server 2003 on the other two PC s. 3. After installations is complete, reboot and select the CentOS option from the boot menu. 4. The first time the OS runs it will compile itself on the machine this may take awhile so take some time to install the X-Lite IP Softphone software on the Windows machines. 5. Once the OS has compiled, login to the server and type netconfig at the prompt to setup the server s network interface. 6. Perform IP configuration for server.

6 Figure 4. Asterisk Monitor 7. At this point, the Asterisk Server has been configured and you should configure the client machines with IP addresses. 8. Connect the server and client machines to a network switch. 9. Use the ping command at the command prompt on the client machines to verify connectivity with the Asterisk server. 10. From the client machine, use Internet Explorer and type in the IP of the Asterisk server in the Address bar to access the web based management interface. 11. The Asterisk Management Portal (AMP) is where you will setup most of the features offered by 12. Fill in the extension information and password and specify the phone protocol being used. 13. All extensions registered to the server, along with their names are shown in the Management Portal

7 14. Now, configure your Softphones with their respective extension information and fill in the server information. Laboratory Expansion Possibilities An interesting extension to this experiment can be done by purchasing and installing a Cisco VIC-2FXO-M2 (approximately $200) on the voice server. This expansion card allows the voice server to be interfaced with the PSTN. Furthermore, more motivated student can explore the conference calling (MeetMe) feature of the Asterisk OS. Conclusion In this paper we presented a practical demonstration of VoIP as a classroom experiment. We introduced a simple and easy-to-use open source Linux based graphical software toolkit, called As a result, students will learn (1) how to set up voice server; (2) how telephone numbering scheme is implemented and how phone calls are routed; (3) how different protocols are used to convert voice into packets and send them through the Internet between multiple users. These test cases can easily be extended to examine more complicated and advanced scenarios. Possible expansions to this experiment are also briefly discussed in this paper. Acknowledgement The authors would like to thank Mr. Steven Pollock, Mr. Steven Fonda, and Mr. Michael Pieciuk for their efforts in setting up and testing the experiment at the Advanced Internet Technology in the Interest of the Society (AITIS) laboratory at Central Connecticut State University. References [1] Wikipedia Dictionary Voice over IP ( 19 April 2004 ) [2] Image taken from: [3] [4] [5]

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