VoIP Architecture and Cost Optimization in UTM

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1 VoIP Architecture and Cost Optimization in UTM HAMIDREZA MOHAJERI, MAZDAK ZAMANI, WARDAH ZAINAL-ABIDIN Advanced Informatics School Universiti Teknologi Malaysia Kuala Lumpur MALAYSIA Abstract: - Telephone systems are crucial elements to businesses in allowing for internal and external communication to users. Previous call logs and other related information in Universiti Teknologi Malaysia shows Making call on legacy telephone system is costly. Academic institutes and universities are looking for decreasing overall costs in order to focus on educational activities. VoIP is a technology that not only transmits voice but also conveys fax, video, data at low cost in an IP network, such as the, virtual phone, virtual speech, fax mailbox, unified news, TV conference and etc. These days with financial crisis, cost solutions are highly attractive among different sectors. Therefore to solve this issue, VoIP architecture for UTM campus to reduce operational costs in telecommunication area has been proposed and components of proposed architecture have been discussed. Then cost reduction in UTM telephone system has been calculated and amount of cost reduction has been defined. Key-Words: - VoIP Architecture; Cost reduction; UTM; Background of VoIP 1 Introduction Communication is of vital essence to the operations of any enterprise office, with the telephone being its most typical mode. In today's world, two various architectures exist for voice communication: the traditional Public Switched Telephone Network (PSTN) and the newer Voice over Internet Protocol, which are both widely, practiced methods based on the telephony network in the transfer of voice communications. Both have displayed benefits and limitations specific to their own. It is of note that PSTN is the underlying architecture of the traditional telephony network [1, 2]. With the advent of Information Technology as a whole, and specifically telecommunications and IP technologies, enterprise communication systems have undergone a series of transformations. Great changes have come about in this sense, which has led the industry far past traditional calling systems. Since its start in 1876, telephony has drastically evolved, dubbing it a necessity in human communication. The main technology propelling it through this period of reign has been a circuit switching by time division multiplexing (TDM) system. This has brought about the current PSTN system in order to manage call switching between users. In recent years though, widespread access to computer systems has fueled users' need to communicate in ways not limited to basic speech. The scales are tipping in favor of the use of IP services against existing PSTN traffic. This ultimately means new communication methods for users and opportunities in creating an IP-oriented service that can function as traditional telephony was meant to; giving way to the appearance of the Voice over Internet Protocol (VoIP) Model [3]. Widespread IP telephony has proved valuable to early adopter companies large and small. Integration of voice, video, and data into a single IP network has brought about reduced communication costs for organizations, and led them to utilization of underused network resources and a foundation for unified communications. As of September 2010, 37% of Japanese telephone lines were IP based. Just as such, the 36% of French telephone lines were based upon IP networks, and IP telephony alone accounted for 62% of all voice traffic in the country [4]. New enterprises can no longer choose timedivision multiplexing (TDM) systems as their telephony services due to the favoring of IP systems by all telephony service providers in comparison to TDM systems. As such, corporations must fixate a plan of timely rollout of their enterprise-wide IP based systems based on their capabilities, cost, and benefit [5]. Telephone systems limited to only voice call functionality does not satisfy today s requirements any more. ISBN:

2 Single uniform medium which focuses on real-time communications channels is highly sought after by organizations looking for systems which can enhance their organizational, workgroup, and even individual productivity. This is possible by enabling integration of all communication channels into the unified medium. The real-time communications channels might include web, instant messaging, tele-presence, voice, video conferencing, mobile telephony, and file transfer. As M. J. Bakhashwain [6] has pointed out, all these modes of communication are enjoying increased acceptance by many enterprises. Hence the challenge faced by many enterprises is the path towards a strategic and economic implementation of this unified architecture [6]. 2 PSTN and VoIP Historical Background The history of public switched telephone network is based on the history AT &T and American Bell companies. According to [7] in American Bell was formed by Alexander Graham Bell. In 1876 there was an improved in Alexander Bell s telephone, in 1878 earliest telephone exchange was built in new Haven, Connecticut, within this year first switchboard was installed and the earliest telephone was set up in Whitehouse. In 1882 controlling interest was obtained in Western Electric Power Company by American Bell. First long distance connection was created from Boston to New York City in 1884 by Bell. In 1968 Microwave Communication of America (MCI) was incorporated. MCI received government acceptance to compete with AT&T in long distance market. In 1986 MCI went international.us telecom and Sprint were merged in 1986 moreover Sprint completed deployment of fiber optic Network. First successful voice over IP was demonstrated in August 1974 on the ARPANET and internet protocol (IP) was emerged basically as a consequence of that result, employing continuously variable slope delta (CVSD) modulation at 16 kb/s in real time between the Information Science Institute and Lincoln Lab the network voice Protocol was experiencing success but the quality of speech was poor [8]. A small company called Vocaltec, Inc in February of 1995 launched VoIP. Internet phone was their product which allowed one user to communicate and call another user through their PCs, a microphone and a set of speakers; furthermore caller and receiver could communicate if both receiver and caller both used the same application or software setup [9]. In 1996 the telecommunication standardization sector (ITU-T) started the standardization of VoIP to begin with the H.323 standard. In 1998 some industrialists began to market phoneto- phone and PC -to - phone Voice over IP solution.by 1998 there was also another improvement which was the hardware s ravage such as several IP Switch manufactures that introduced switching application as a standard in their routing equipment. In 1998, VoIP traffic had grown to represent almost 1% of 9 all voice traffic in the United States. By 1999 the Session Initiation Protocol (SIP) was released and afterward, Mark Spencer created Asterisk which was the first open source SIP PBX. Cisco and Lucent, pioneer of the networking and hardware manufacturers, came out soon with applications that could switch and route VoIP traffic. By 2000, VoIP calls had total 3% of all voice calls and by the year 2003 VoIP traffic had jumped up to 25%. [9]. Knight [10] stated that Marketers can expect VoIP penetration to reach 79% by 2013 that is almost a 50% increase over 2009 numbers. 3 Overview of current system in UTM Figure 1 shows the current telephone system architecture of UTM University which is collected from Pejabat Harta Bina (PHB) department. According to the Figure1, UTM campus in Kuala Lumpur consists of three PABXs which are connected to them. Pejabat Harta Bina Unit takes care and manages these PABXs. These PABXs via ISDN line through PSTN are connected to UTM JB (Johor Bahru) PABX system. The PABXs are: PABX UTM KL with 1000 extensions PABX HEAMC building with 1200 extensions PABX MJIT building with 500 extensions At CICT department there is Cisco router which is used for data connection and they often employed it for voice communication. Figure 2 represents before implementing Voice over IP in Campus, UTM University requires managing separate networks for data and Voice. It is obvious that Migration to IP allows the organization to converge its data and voice networks. IP PBX can be connected to the PSTN network for circuit switched calls using specialized gateway device. ISBN:

3 4 Proposed Architecture In this section first components of the proposed architecture will be discussed, and then architecture for UTM campus will be proposed to reduce operational costs in telecommunication area System Components System components include two parts: hardware and software Hardware Components Cisco IOS ISR running CUBE 1.2 (IOS image version 12.4(20) T3 or later) Cisco MCS 7800 Series server (Cisco Unified Communications Manager) Cisco Unified Border Element Cisco IOS Gateway Cisco IP Phones Software Requirements Cisco GW IOS Release: 12.4 or later CUBE version 1.2 (IOS version 12.4(20) T3 or later) Cisco Unified CM Additional versions may be supported, Cisco Integrated Services Router (ISRs) provides the flexibility you need to evolve your network in ways that most benefit your organization. Unlike other vendors Cisco ISRs provide the multi service, single platform for both flexible migration and deployment choice, this allows you to take full advantage of what you already have as you transition your network to SIP Trunking and beyond. To begin realizing the benefits of SIP Trunking, organizations need to deploy a session border controller (SBC) in order to efficiently and securely connects to service providers while preserving voice quality and features. Session border controllers connect IP networks and provide session control, security, demarcation for better troubleshooting and interworking to help overcome differences in the deployment of the SIP standard (such as CODEC or signaling). It is one of the significant elements of SIP Trunking solution.one of the main roles of the SBC is ability to terminate different IP addresses. A second vital feature of SBC is protecting and securing the Service Provider networks. In addition SBC guarantees that no harmful and damaging traffic from SP can affect or infect the enterprise and one enterprise traffic cannot impact another. One of the services enable the Cisco ISR is session border control. Cisco session border controller is called Cisco unified border element (CUBE). The CUBE performs critical services for SIP Trunking to allow secure, flexible and reliable network interconnection. It offers you the most efficient way to manage and migrate to your end to end SIP communications network. Cisco CUBE offers solutions for every size business and location. It supports both TDM gateways and cube on the same platform. This allowed you phasing your transitions at the paste that's right for your business. Cisco Unified Call Manager or Cisco Call Manager, recently Cisco unified communication Manager (CCUM) has been software -based call- processing system developed by Cisco System. Cisco Unified Communication Manager routes all active voice over IP network elements, these consist of gateways, Voice mailboxes, phones, Conference Bridge and trance coding resources. CCUM often employs Skinny Client Control Protocol (SCCP) as a connection protocol for signaling the Hardware system such as IP phone. Furthermore SIP or H.323 is utilized to pass signaling of call to the gateway. Figure 3 which is shown blow, demonstrates proposed architecture for UTM. 5 Findings of current Telephone Cost in UTM According to the Figure 4 which is shown below depicts UTM campus operational telephone cost in eleven months that is from January to November From the diagram more than RM30,000 was spent on the telephone in January, in February the expenditures decreased to over RM25,000, there was a slight increase in March to RM30,000, it kept rising in April which has the highest cost at over RM35,000. There was a steady decrease in the following months of May, June, July and August which had below RM35K as a cost on the telephone. In July and August the cost of the telephone was same at RM30K. In September there was a huge drop in the cost to RM20K which is the beginning of the session and it also recorded the lowest cost. There was a sharp rise in October to RM28Kand in November it was at almost RM25K. ISBN:

4 6 Cost Estimation using Ratio Formula A successful installation of VoIP can be employed as a cost reduction method to aid pay for implementation of other module for Unified communication solution. Therefore organization can cut operational expenditure as well as capital expenditures (such as maintenance cost and etc). In this section cost reduction in UTM telephone system will be calculated and amount of cost reduction will be defined. To estimate reduced cost using two different VoIP approaches, ratio formula is employed. To calculate the Y2 rate, X1 and Y1 need to be entered from the table 4.3. Y2 = X1= Old system rate X 2= New system rate Y1= Current UTM Cost Y2= Estimated cost Two different approaches of VoIP solution in academic environment employed for calculating cost reduction which are following: Approach 1 is at Sevastopol National Technical University. The monthly expenses of the University on a fixed telephony before adoption of VoIP was about 8000 UAH. Migrating to an IP based system allows the university to save up to 1,500 UAH per month it means 18 percent cost reduction in operational cost [11]. Approach 2 is at Public educational sector in Croatia. The Monthly expenses before implementing VoIP was more than 5 million Kunas. Using VoopIX around 1.2 million canals can be saved monthly reflects a 24% cost reduction [12]. For example: Using approach 2 (VoopIX) [12]: for Academic, Scientific and Educational Institutions in Croatia entirely based on Open Source software. Public educational sector in Croatia spends more than 5 million canss (approx million ) each month on fixed (land line) telephony costs. By combining all possible savings when using VoopIX in the area of international, mobile and intra VoopIX calls, it is estimated that around 1.2 million Kunas can be saved monthly at country level. So it can be solved for Y2 as follows: X1= 5, X2= Y1= Y2=? Cost reduction ratio (CRR) = Y2 = CRR * Y1 Y2 = Y2 = Table 1 shows the information about operating cost by comparing costs using 2 different approaches. Operating cost divided into 3 categories current cost and percentage of cost reduction using 2 different approaches. Approach 2 seemed to and is predicted to be more economical than approach1. However both approaches showed gradual decrease in telecom cost during the given period. The percent described the percentage of cost reduction using approach 1 and 24 percent described the percentage of cost reduction using approach 2. The given line graph illustrates information on the percentage of cost reduction using 2 different Voice over IP approach in UTM international campus Telephony system over eleven months between January and November in From the graph more than RM30K was spent on the telephone in January in current system while using approach 1 estimated at RM in January says almost RM 6K cost reduction during this month. Using approach 2 estimated more than 7K cost reduction in telephony system in February the expenditures decreased to over RM25K while using approach 1 and 2 respectively estimated at 21, there was a slight increase in March to RM30K using approach 1 and 2 respectively is estimated RM5, 770 and RM7, 385 cost reduction it kept rising in April which has the highest cost at over RM35K there is an estimation of RM6, 575 and RM8, 416 cost reduction during this month. There were a steady decrease in the following months of May June July and August which had below RM35K as a cost on the telephone. With cost reduction of almost RM 6K using approach1 and more than 7K using approach 2. In September there was a huge drop in the cost to RM20K which is recorded the lowest cost. There was a sharp rise in October to RM28K and in November it was at almost RM25K. In these month expenditures estimated to almost RM23K and RM20K using approach 1 as well RM21K and RM19K using approach 2. 7 Conclusion VoIP is a technology that not only transmits voice but also conveys fax video data at low cost in IP network such as the virtual phone virtual speech fax mailbox unified news TV conference and etc. These days with financial crisis cost reduction solutions are highly attractive among different sectors. Academic institutes and universities are not exceptions [13-22]. In this paper we proposed architecture based on VoIP to reduce operational cost and in the next step cost reduction using ratio formula was estimated. ISBN:

5 Acknowledgment This work is part of research done with support from Universiti Teknologi Malaysia, Malaysia. References: [1] Venkatesha R. Prasad " Comparison of voice activity detection algorithms for VoIP " in in Computers and Communications Proceedings. ISCC 2002 Seventh International Symposium on pp [2] A. Sinaeepourfard and H. M. Hussain "Comparison of VoIP and PSTN services by statistical analysis" in Research and Development (SCOReD) 2011 IEEE Student Conference on 2011 pp [3] B. M. Raymundo et al. "Implementation of voip at the University of Colima utilizing H.323 protocol" in rd International Conference on Electrical and Electronics Engineering September September Veracruz Mexico [4] L. Tian et al. "Study of SIP protocol through VoIP solution of Asterisk " presented at the Mobile Congress (GMC) 2011 Global2011. [5] Lassman Jay and D. O'Connell. ( 2007). "IPT vs. TDM Life Cycle Purchase and Operations Costs. [6] M. J. Bakhashwain "A case study on the road to IP telephony: A company's strategic decision to move toward the Next-Generation Network" in Technology Management Conference (ITMC) 2011 IEEE International 2011 pp [7] InetDaemon. ( Oct 2012). History of the Public Switched Telephone Network (PSTN). Available: pstn/history.shtml [8] R. M. Gray "The 1974 origins of VoIP" Signal Processing Magazine IEEE vol. 22 pp [9] J. Hallock "A Brief History of VoIP" Evolution [10] K. Knight. (2010 April 2012). Forecast: VoIP penetration. Available: p_penetration_to_reach _79.html [11] A. A. Savochkin et al. "Adaptation of university fixed telephony to IP technologies" in Microwave and Telecommunication Technology (CriMiCo) th International Crimean Conference 2011 pp [12] B. Radojevic and D. Milic "voopix: Building free VoIP infrastructure for academic science and educational institutions in Croatia entirely based on Open Source software" in MIPRO 2011 Proceedings of the 34th International Convention 2011 pp [13] Shima Beigzadeh, Mazdak Zamani, Suhaimi Ibrahim, and Maslin Masrom. Design and Implementation of a Web-Based Database- Centric Management Information System for a Social Community International Conference on Information Systems and Computational Intelligence (ICISCI 2011). January 18, Harbin, Northeastern China. [14] Shima Beigzadeh, Mazdak Zamani, Suhaimi Ibrahim. Development of a Web-Based Community Management Information System. The Fourth International Conference on Information and Computing (ICIC2011) April Phuket, Thailand. [15] Farnaz Arab, Harihodin Selamat, and Mazdak Zamani. An Overview of Success Factors for CRM nd IEEE International Conference on Information and Financial Engineering September Chongqing, China. [16] Farnaz Arab, Harihodin Selamat, Suhaimi Ibrahim, and Mazdak Zamani. A Survey of Success Factors for CRM. International Conference on Computer Science and Applications (ICCSA'10) October San Francisco, USA. [17] Farhang Jaryani, Shamsul Sahibudin, Maslin Masrom, Babak Zandi, Mazdak Zamani and Samaneh Salehy. Framework of a Reflective E- portfolio Supported by Outcome Based Education and Problem Based Learning International Conference on Computer Research and Development (ICCRD 2010). May 7-9, Kuala Lumpur, Malaysia. [18] Farnaz Arab, Harihodin Selamat, Suhaimi Ibrahim, and Mazdak Zamani. A Survey of Success Factors for CRM. Lecture Notes in Engineering and Computer Science. ISSN EISSN Volume Issue 1. Pages [19] Hamed Taherdoost, Mazdak Zamani, and Meysam Namayandeh. Study of Smart Card Technology and Probe User Awareness about It: A Case Study of Middle Eastern Students. The 2009 International Conference on Management Technology and Applications. ISBN: Volume 5. Pages August Beijing, China. [20] Neda Jalaliyoon, Hamed Taherdoost, & Mazdak Zamani. Utilizing the BSC and EFQM as a Combination Framework; Scrutinizing the Possibility by TOPSIS Method. International ISBN:

6 Journal of Business Research and Management, (IJBRM), ISSN: Volume (2): Issue (1) [21] Hamed Taherdoost, Meysam Namayandeh, Neda Jalaliyoon, Kaveh Ahmadi, Mazdak Zamani, and Mohammad Zargar. Study of Internet Protocol Television in Iran rd IEEE International Conference on Computer Science and Information Technology July 2010 Chengdu, China. [22] Hamed Taherdoost, Arash Forghani, Meysam Namayandeh, Mazdak Zamani and Neda Jalaliyoon. Adoption Framework Expansion based on the Computer Ethics' Related Research Models and Ethical Scenarios Analysis. International Proceedings of Economics Development and Research. Vol. 2 (2011) IAC S IT Press. ISSN: DOI: /IPEDR. Figure 1. Architecture of current system Figure 2. Proposed Architecture ISBN:

7 Figure 3. Before implementation of VoIP Figure 4. Telephone system operational cost for UTM Figure 5. Cost Estimation Using two different approaches TABLE 1. COST ESTIMATION USING TWO DIFFERENT APPROACHES Jan Feb Mar Apr May Jun July Aug Sep Oct Nov Total UTM International Campus Current Bill VoIP Aproach1 (18.75%) VoIP Aproach2 (24%) ISBN:

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