Asiful Hoque Khan Id#042191056

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Asiful Hoque Khan Id#042191056

  1. 1. IP Telephony NORTH SOUTH UNIVERSITY PREPARED BY : Md. Asiful Hoque Khan. ID # 042 191 056
  2. 2. 15th April, 2008 Subject: Submission letter of individual Assignment. Dear Sir, I’m a student of your ETE605 course; section 2. As a requisite of this course I’ve done my assignment on, “VoIP with wireless mobile Com munication”, which you have approved so far. I would like to thank you for letting me work on the given topic, as it had helped us gather more idea about the subject matter of this course through analysis and reporting. It was a real pleasure working on s uch project and we hope it will be enjoyable to you as well. Most obediently, Md. Asiful Hoque Khan ID # 042 191 056
  3. 3. ACKNOWLEDGEMENTS This is my individual assignment of IP Telephony in MS program and the assignment was a new experience for me. The assignment has been completed with maximum efforts possible throughout the semester within the time limits. I could gather information and support from those who co-operated us wholeheartedly. Since the list of contributors is not long, we want to mention about all of them. At first I would like to thank our hon orable faculty Dr. Mashiur Rahman. From the very beginning, he has always been helpful to us regarding our problems. When I fell in difficulty in solving certain problems it was him who got us out of the situation. He made my works easy too. I thank him for giving me his valuable hours whenever required. My gratitude also goes for the writers and editors whose published materials on VoIP and IP telephony helped us to analyze my assignment in detail. Last of all we would like to thank The Almighty Allah for keeping me fit and fine to perform our due works on time.
  4. 4. TABLE OF CONTENT 1. BACKGROUND ................................................................................................................................. 1 1.1 A BRIEF HISTORY OF TELECOMMUNICATION INDUSTRY IN BANGLADESH ..................................... 1 1.2 VOICE COMMUNICATIONS .............................................................................................................. 1 1.3 DATA COMMUNICATIONS ............................................................................................................... 2 2. COMBINING WIRELESS AND VOIP ........................................................................................... 3 2.1 VOIP............................................................................................................................................... 3 2.2 VOIP BASICS .................................................................................................................................. 4 3. INFRASTRUCTURE ......................................................................................................................... 4 3.1 INFRASTRUCTURE OF VOIP SYSTEM ............................................................................................... 5 3.2 SIGNALING IN VOIP NETWORKS ..................................................................................................... 6 3.3 PROTOCOLS .................................................................................................................................... 7 3.4 NETWORK COMPONENTS ................................................................................................................ 8 4. CONCLUSION ................................................................................................................................. 10 5. REFERENCES ................................................................................................................................. 10 APPENDIX ......................................................................................................................................... 1
  5. 5. 1. Background Good communication services and universal access are necessary for a higher standard of living and economic growth. However the high cost of legacy PSTN equipment may not be affordable to some developing nations, especially in rur al areas which have a much lower subscriber density, or areas with geographic challenges such as large bodies of water, jungles, mountainous terrain etc. 1.1 A Brief History of Telecommunication Industry in Bangladesh After independence of The Peoples Republ ic of Bangladesh in 1971, Bangladesh Telegraph & Telephone (T&T) Department was created under the Ministry of Posts & Telecommunications with a view to run the telecommunication services on commercial basis. The Bangladesh T & T Department was converted in to a corporate body in 1976. In pursuance of Ordinance No. XII promulgated by the President of the Peoples Republic of Bangladesh on 24 th February, 1979 The Bangladesh Telegraph & Telephone Board (BTTB) came into existence. Growth of Telephone in Bangladesh The growth of telephone exchange capacity in Bangladesh in the last five years was on average only 40,000 lines per year. The recorded pending demand of telephone has been increasing at a faster rate than the telephone expansion rate. The actual demand is really much more than the numbers expressed here . For cast: Table XI. Projected Demand : Expected number of telephone to be in use in Bangladesh By Year 2002 800,000 units By Year 2005 1600,000 units (16lakhs ) By Year 2010 3200,000 units Source : BTTB 1999 1.2 Voice Communications There are several paradigm shifts happening in today’s telephony markets which are driving costs down by orders of magnitude. First legacy telephony systems are based on Circuit -1-
  6. 6. Switched Networks or (CSNs) this means a telephone call is allocated a dedicated circuit from end to end. In the old days this meant a physical pair of wires for the audio to travel over. Today this typically means two 64Kbps channels one in each direction which are dedicated to that call even if no one is talking, and since usually only one person i s talking at a time about half of the bandwidth is wasted. For example, a typical small PSTN trunk can carry 24 or 30 simultaneous calls. If the bandwidth were used more effectively the circuit could carry much more if not almost twice as many calls. On th e positive side CSN technology is very robust and predictable which made it easier to build reliable telephone networks in the early years of the industry. Because these PSTN switching systems were very big and centralized due to the state of the art at th at time, they were very expensive and relatively few were sold to big companies like AT&T. So the market never developed to a point where the prices could drop significantly. Limitation of the system One limitation of this technology that may slow down the complete conversion to an audio over data network is that there needs to be power at the subscribers’ site for the terminal equipment. Legacy telephones are powered only by the PSTN so they will still work if there is a power failure, and this is often when it's needed the most. The PSTN is able to provide this by having a battery bank and generator at each switching si te. To provide a reliable VoIP system it is usually necessary to have battery backup at each subscriber site. 1.3 Data Communications Data rates on wired networks have been increasing by powers of ten over the years, and more recently wireless rates have been catching up. This is due to many factors. Among them are the commercialization of spread spectrum technology, improvements in IC manufacturing processes to fit these radios on small cards, and the allocation of radio spectrum in the Gigahertz range for licensed and unlicensed use of these devices. -2-
  7. 7. 2. Combining Wireless and VoIP Wireless telephony is nothing new, there are microwave links for the trunk lines and Wireless Local Loop (WLL) for the subscriber terminal equipment. But it's mostly CSN based technology and is therefore quite expensive . If one combines a network built out of commodity wireless cards with Voice over IP equipment it is a low cost delivery infrastructure that makes efficient use of the bandwidth it provides. Additionally one gets a high speed data network that can also pro vide Internet access. 2.1 VoIP Voice over IP (VoIP) exploits the ability of IP to deliver multiple services over a single access link. Increased computing power and available bandwidth in endpoints, reduced cost and size of electronics, as well as sensor and positioning technologies allow us to develop new interactive multimedia applications that are able to adapt to the communication context of the end-user. This is particularly important as wireless access to the Internet and connectivity between mobile artifacts can leverage these possibilities even further to bring us new ways of communication. Our work shows that, with respect to VoIP over wireless networks, bandwidth is not the problem and that Quos can match that of voice in today’s cellular networks. We therefore propose to run IP directly over wireless links to bring multimedia services to mobile users. Even more importantly, this leads to a significant simplification, and consequently a cost reduction, of the wireless infrastructure. Electronics is dropping while, at the same time, there has been a tremendous increase in computational power. As far as personal communication and mob ility is concerned, we are in the position to create new applications and services that go far beyond what telephony systems have been concerned with and able to accomplish. One of the main contributing factors is the Internet Protocol, which allows these new applications to benefit from the fact that end-user devices are now able to use multiple services over a single link. The result is that we are now able to build new interactive services, which can combine both voice and data simultaneously. Fig. 1 illustrates the change. -3-
  8. 8. 2.2 VoIP Basics VoIP attempts to transmit human audible voice through IP packets using the Internet. Deploying accelerating hardware or using a regular PC can either use it. 3. Infrastructure To understand the network a rchitecture, we must have to understand the how GSM (Global System for Mobile Communication) network works. In GSM when the MS (mobile station) is assigned to a channel after requesting the network for a free channel. The Network then establishes a channel for the user after authentication with routing process. Connection established when the destination is idle or available. Then communication starts between the MS and destination. The network needs a gateway namely GMSC (Gateway for Mobile Switching Cente r) which communicates with GSM network and PSTN network. Here, there is temporary database, VLR (Visitor Location Register) which keeps the information about the user in a cell. HLR (Home Location Register) is the pe rmanent register to keep information for the users permanently locating at NSS (Network Sub -system). Main function of the AUC (Authentication center) to authenticate the user trying to access the network. -4-
  9. 9. 3.1 Infrastructure of VOIP system In order to understand VOIP it is essential to have a complete understanding of what the difference between circuit switching and packet switching. A normal telephone uses circuit switching for phone calls, which involves routing of your call through the switch at your l ocal carrier to the person you are calling. The connection of two points in both directions is known as circuit. Packet switching on the other hand is more efficient in transmitting data since small amount of data which is called a packet, is sent from one system to another. Data networks do not use circuit switching because there is huge data loss . -5-
  10. 10. In a wireless system, GSM or CDMA, the system also uses circuit switched network. But VOIP uses packet switched network for higher e fficient and so some technical and physical difference can be observed in VOIP design. The main difference is that VOIP must go through IP and Routing process. SIP plays one of the basic role in VOIP, which is located in the service provider’s network and provides call logic and call control functions. There is a gateway which passes all the data after routing. A modem is needed which acts as a DAC ( Digital to Analog) to convert digital data to analog data. Analog data is sent with a satellite dish. 3.2 Signaling in VoIP Networks VoIP networks carry SS7–over–IP using protocols defined by Signaling Transport (sigtran) Working Group of the Internet Engineering Task Force (IETF), the international organization responsible for recommending Internet standards. Th e sigtran protocols support the stringent requirements for SS7/C7 signaling as defined by International Telecommunication Union (ITU) Telecommunication Standardization Sector. In IP–telephony networks, signaling information is exchanged between the follow ing key functional elements: Media Gateway: A media gateway terminates voice calls on inter -switch trunks from the PSTN, compresses and packetizes the voice data, and delivers compressed voice packets to the IP network. For voice calls originating in an IP ne twork, the media gateway performs these functions in reverse order. For ISDN calls from the PSTN, Q.931 signaling information is transported from the media gateway to the media gateway controller (described below) for call processing. Media Gateway Controller: A media gateway controller handles the registration and management of resources at the media gateway(s). A media gateway controller exchanges ISUP messages with central -office switches via a signaling gateway (described below). Because vendors of med ia gateway controllers often use off -the- shelf computer platforms, a media gateway controller is sometimes called a softswitch. Signaling Gateway: A signaling gateway provides transparent interworking of signaling between switched -circuit and IP networks. The signaling gateway may -6-
  11. 11. terminate SS7 signaling or translate and relay messages over an IP network to a media gateway controller or another signaling gateway. Because of its critical role in integrated voice networks, signaling gateways are often deploy ed in groups of two or more to ensure high availability. A media gateway, signaling gateway, or media gateway controller (softswitch) may be separate physical devices or integrated in any combination. Figure : Example of a VoIP network configuration 3.3 Protocols The sigtran protocols specify the means by which SS7 messages can be reliably transported over IP networks. The architecture ident ifies two components: a common transport protocol for the SS7 protocol layer being carried and an adaptation module to emulate lower layers of the protocol. For example, if the native protocol is message transport layer (MTP) Level 3, the sigtran protocols provide the equivalent functionality of MTP Level 2. If the native protocol is ISUP or SCCP, the sigtran protocols provide the same functionality as MTP Levels 2 and 3. If the native protocol is TCAP, the sigtran protocols provide the functionality of SCCP (connectionless classes) and MTP Levels 2 and 3. The sigtran protocols provide all the functionality needed to support SS7 signaling over IP networks, including: Flow control In-sequence delivery of signaling messages within a single control stream Identification of the originating and terminating signaling points -7-
  12. 12. Identification of voice circuits Error detection, retransmission and other error -correcting procedures Recovery from outages of components in the transit path Controls to avoid congestion on the Internet Detection of the status of peer entities (e.g., in service, out -of-service, etc.) Support for security mechanisms to protect the integrity of the signaling information Extensions to support security and future requirements Restrictions imposed by narrowband SS7 networks, such as the need to segment and reassemble messages greater than 272 bytes, are not applicable to IP networks and therefore not supported by the sigtran protocols. 3.4 Network Components This section describes the function of the network components needed to build up a VOIP system. Depending upon the particular network architecture some of these net work components may be combined into a single solution. Call Agent/SIP Server/SIP Client The Call Agent/SIP Server/SIP Client is located in the service provider’s network and provides call logic and call control functions, typically maintaining call state for every call in the network. Many call agents include service logic for supplementary services, e.g. Caller ID, Call Waiting, and also interact with application servers to supply services that are not directly hosted on call agent. The Call Agent will p articipate in signaling and device control flows originating, terminating or forwarding messages. There are numerous relevant protocols depending upon the network architecture including SIP, SIP -T, H.323, BICC, H.248, MGCP/NCS, SS7, AIN, ISDN, etc. Call Ag ents also produce details of each call to support billing and reconciliation. A SIP Server provides equivalent function to a Call Agent in a SIP signaling network, its primary roles are to route and forward SIP requests, enforce policy (for example call admission control) and maintain call details records. For example the SIP Server in Service Provider 1’s network will route and forward SIP requests from SIP Phones belonging to customers. -8-
  13. 13. A SIP Client provides similar function to a SIP Server, but originat es or terminates SIP signaling rather than forwarding it to a SIP Phone or other CPE device. Call Agents are also known as Media Gateway Controllers, Softswitches and Call Controllers. All these terms convey a slightly different emphasis but maintaining ca ll state is the common function. Application Server The Application Server is located in the service provider’s network and provides the service logic and execution for one or more applications or services that are not directly hosted on the Call Agent. Typically the Call Agent will route calls to the appropriate application server when a service is invoked that the Call Agent cannot itself support. Media Server This Media Server is located in the service provider’s network. It is also referred to as an announcement server. For voice services, it uses a control protocol, such as H.248 (Megaco) or MGCP, under the control of the call agent or application server. Some of the functions the Media Server can provide are- Playing announcements ixing – providing support for 3-way calling etc odec transcoding and voice activity detection one detection and generation nteractive voice response (IVR) processing Fax processing. Signaling Gateway The Signaling Gateway is located in the service provider’s network and acts as a gateway between the call agent signaling and the SS7 -based PSTN. It can also be used as a signaling gateway between different packet -based carrier domains. It may provide signaling tran slation, for example between SIP and SS7 or simply signaling transport conversion e.g. SS7 over IP to SS7 over TDM. -9-
  14. 14. 4. Conclusion This project was successfully deployed and commissioned, and should be tested further by providing Internet access to some schools and communities. Much has been learned and a new generation of equipment is already in the design stages which will correct most of the known shortcomings of the current generation. Most notably the issues being addressed are scalability, configuration management, better monitoring capabilities, lower power consumption, and high speed backbones. This will all add up to a lower Total Cost of Ownership or TCO. 5. References 1. http://www.bhutan-notes.com/clif/ 2. http://www.thefeature.com/article?articleid=100667&ref=2446671 3. http://www.nextgendc.com/ 4. http://ps.verkstad.net/Papers/Conferences/PCC/99/VWMMA_PCC.PDF 5. http://wireless.newsfactor.com/perl/story/22103.html ---------------------------------------------------------------------- - 10 -
  15. 15. APPENDIX VoIP Advantages Multifunction and multi communication : When using a PSTN line, one typically pays for the time used to a PSTN line managing company according to the duration of time a person uses the connection. The PSTN line connection also has the limitation of communicating with one person: more time you stay at phone and more you'll pay. In addition you couldn't talk with other that one per son at a time. On the other hand, the VoIP mechanism allows one to communicate all the time and multi tasking with many individuals at the same time with the caveat that the other person is logged on to the Internet at the same time. Therefore, using VoIP, while exchanging data with multiple people, one can communicate vocally and also send images, GIF files, graphs, videos, etc. Phone bills too cheap to meter: There are a number of benefits to making phone calls over the Internet, but the number one reason people use VoIP is because it dramatically reduces phone bills. For example, through a VoIP company, we pay a flat fee for unlimited local calling, and just pennies per minute to call other countries. The traditional phone companies, which for decades have been able to get away with charging several dollars a minute for an overseas call, are trying to compete with VoIP startups, but they just can’t keep their rates that low. Naturally, they’re doing everything they can to kill VoIP companies by lawyering them to death, but cool technologies have always been able to mutate their way out of any impediment. (Look at what happened when the record industry shut down Napster, and as a result, help spawn umpteen all-but-unstoppable peer-to-peer networks.) The joys of VoIP have been restricted to landline phone use. This has made wirel ess carriers very happy. If we want to make an international or out-of-state call with our mobile, we are stuck with our wireless carrier’s typically exorbita nt toll charges. For example, the carrier, Cingular, charges $1.49 a minute to call the United Kingdom, which is ridiculous. How can Cingular get away with it? Simple, it has locked its competitors out. It’s using th e -1-
  16. 16. old movie theater concession stand tactic. The candy bar at the theater concession is worth 79 cents on the open market, but if we want to buy one at the movies, we ’ll have to fork over $3.50. Sure, it’s robbery, but they can get away with it because you ’ve got no other choice, other than sneaking a store -bought candy bar into the theater. VoIP Limitations Although the technology described above of communicating with multiple people appears exciting, VoIP has certain limitations. The problem stems from t he integration of the VoIP architecture with the Internet architecture. As is discernible, voice and data communication must be in a real time environment (streaming data) since waiting to hear the return communication is expected to be immediate by the hu man sensory nerves. This real time environment is in contrast with the Internet’s heterogeneous architecture comprising possibly of 20-30 routers, the equipment that route packets. The large number of routers can employ a very high round trip time (RTT) an d therefore necessitating modification of the architecture or the VoIP protocol used to communicate through the Internet. It is important to realize that it is very difficult to guarantee bandwidth on the Internet for VoIP application. -2-
  17. 17. Specific Features in VoIP Specific features that are important in a VoIP system . We need to consider them as much as possible. Security o Because there are many CPEs which the customers have physical access to it's best to have a cryptographically strong security system with unique keys for each CPE. Authentication o Each Device in the VoIP net work should be able to authenticate itself using these keys. IP or MAC addresses can't be considered as forms of authentication. Privacy o Since the network is wireless the traffic can be easily monitored with software readily available on the Internet. So it's advisable for each call to be encrypted. This uses only a minimal amount of additional processing power. CDRs o Call Data Records should be in a form easily usable by many third party billing systems. Often a vendor will list the third party products that they interoperate with. Least Cost Routing (LCR) o This is probably not as important for a small system but it can be useful if one has several choices for routing either within the system or to multiple long distance carriers. Billing o The billing system should probably be considered separately from the VoIP system. They come in many shapes and sizes, find one that fits the project's needs. Again check what third party vendors a package will work with. Call flow monitoring o The Gatekeeper or network monitor should collect useful statistics on system usage and state. An operator console should be available to keep an eye on things and make simple changes as needed. Call tracing -3-
  18. 18. o Some real time and logged information on how calls are routed can be us eful for growth planning and troubleshooting. System testing o Test call facility There should be a way to call "test numbers" on any unit, in fact probably an arbitrary number of simultaneous test calls should be possible on a GW (CPE or otherwise). In a ddition to this, one should be able to specify several parameters of the test call: Codec, audio file to play, (ie. test tone), or echo audio data back, etc. The idea here is to have a GW automatically answer a test call without ringing a customer's phone. If it reports, or logs statistics for these test calls that's a plus. As mentioned above SNMP retrieval of stats is highly desirable. o FTP Again, being able to send and receive arbitrary data to a network device is very handy for simple throughput testin g. Of course it's a convenient way to update the units firmware and configuration also. Specific features that are important in a wireless system We need to consider them as well. Security o Having Link level or end to end encryption of data is nice for pri vacies sake. Monitoring o SNMP It's important for all Wireless (and most wired) devices to allow for the collection of statistics via Simple Network Management Protocol (SNMP). The items below should all be accessible this way. o Signal strength / Signal to Noise Ratio (SNR), Retries, Holdoffs, Receive errors. Over time these stats will tell one if there is a LOS path problem, interference from other sources, or simply the useful capacity of this link. o Network Interface Throughput This will help track where the bandwidth is being used. -4-
  19. 19. o System Load Gives an indication of how well a unit is keeping up with it's tasks. System testing o FTP: Being able to transfer data into or out of a device is a great diagnostic tool. Pricing policy Cost Reduction: One KEY ingredient can be lower Cost of making telephone calls, and to be able to deploy such Service faster than any other Carrier to any location worldwide. Billing system: With all our equipment including billing being Web Based we can change features, functions and technical interface conditions remotely. Management Experienced manpower: For efficiency in applying the technology, there has to be sufficient manpower experienced in manufacturing, R&D, LD Service Providers, National and International Telecom Projects and past employment experience with the likes of Cable & Wireless, just to mention. Strategic Planning Forming of Partnership In order to provide maximum Call Quality and R eliability partnerships have to be implemented utilizing various Data Networks formulated and streamlined to meet the needs. -5-
  20. 20. RECOMMENDATIONS Software General guidelines When choosing software or hardware/firmware packages for a system we need to keep in mind these general guidelines. Conforms to Open Standards o The system should be built around open standards. Ide ally this allows many to contribute to a standard and puts manufacturers on equal footing when building products based on it. o Thus there will tend to be more product choices. De facto standards o De facto standards are less desirable because fewer companie s were involved in setting them. Interoperability o Check for interoperability between different vendors. This shows that the standard is being followed faithfully, and gives one many more options when looking for solutions. Reliability o Many things contribute to overall system reliability: Redundancy, such as RAID arrays, multiple network connections, multiple servers, to name a few. Graceful fail over to other redundant components when one fails. Early warning of problems that might predict a failure and just good system design. All of these are desirable. Affordability o When making comparisons it's useful to evaluate the Total Cost of Ownership (TCO). For example one would tally up the expected cost of customer support, replacement hardware, softwa re upgrades, hardware upgrades, maintenance personal, etc. Ease of use o Setup -6-
  21. 21. Ease of setup usually means one needs less support, less time, and that the configuration process is less error prone. If the system is continually growing then this is an ongo ing issue. o Administration Ease of administration will be a large factor in TCO. Ease and cost of maintenance o The cost of maintenance depends on the rate of failures, cost of the replacement hardware, number of people it takes to do the work, etc. o Monitoring Allows Simple Network Management Protocol (SNMP) monitoring of all units. A good monitoring system will filter and present the relevant problems. All events should be logged and searchable. There are many general purpose monitoring systems based o n SNMP. Also simple reachability tests using ping are good way to monitor general system health. Check out refs. Allows logging of notable events to a central logging server. o Upgrading The firmware should be easily remotely upgradeable. Beyond this it's desirable to have a firmware and configuration management system to make it less time consuming to track and update many CPEs at once. This all adds up to low TCO. -7-
  22. 22. BILLING/TARIFF MSC VOIP Mediation Billing System CDR Generation: The CDR is generated in the switch when subscribers make the call. The Mediation server brings the CDR from switch and process it to some format and send it to billing system so that it can process billing for the subscriber. Billing Process Overview Zone Definition UDR File Operator table Rating Plan Air Part VOIP Part -8-
  23. 23. The mediation server makes the CDR into a special format file called UDR files. This file contains call information such as source and destination number, zon e code, call type, call date time etc. The billing system will look at zone definition part and rate plan part and will do billing separately for air and VOIP part. The total bill for ISD call will be - ISD call bill= Air amount + Land Amount Zone Definition We can divide the all the countri es in the world into several zone parts. For example we can divide the world into six parts. USA and related region: USA zone Middle-east countries: Middle-east zone African Countries: African Zone European Countries : European Zone Asian Countries: Asian Zone Australian and related region: Australian Zone This is just an example. For simplicity we just shown six parts but we can divide more than that according to management decision and marketing plan. Operator Table This table contain list of different operators and there rate per minutes. The table can be divided into two parts; operator definition contains operator id, name and digit. Another part will be operator rateplan which will contain operator id and peak and offpeak rate. Rating Plan This table will contain subscriber rateplan information like he may get reduce amount or may get free unit etc. Rating process Logic Let say we are following the number plan as follows - +1010-01-033-9887772 VOIP route number-country code-operator digit-destination number. -9-
  24. 24. Now when rating process look at the UDR file it peaks up the called par ty number and according to following algorithm it process rating 1. Checks whether it is VOIP routed call (get +1010) 2. If it is VOIP routed call then get the country code 3. From database get the country id and name by the country code 4. From zone table get the zone id and zone name by the country id 5. Get the operator id from zone -operator table by operator digit. 6. Get the operator name and rateplan from operator rating by operator id 7. Check call time from UDR 8. Select the peak or offpeak rateplan according to call time . 9. Get call duration from UDR table 10. Calculate rating according to call duration. - 10 -
  25. 25. UDR VoIP route? Country Code Get country id and name Get Zone Get operator list Get rate plan Rate Plan VOIP Amount Table Definition Country Table Country ID Country Name Digit Zone definition - 11 -
  26. 26. Zone ID Zone Name Zone Operators Zone ID Operator Digit Operator ID Operator Rate plan Operator Operator Peak Peak Off Peak Off Peak ID Name Time Value Time Value Tuning Then contents of desired table can be extracted and kept as flat file. This way the system can read the flat file for required data, and the ratin g procedure will be faster. Other Rating As the user is making a call using the mobile operator network, the operator also will charge the subscriber as normal air network usage rateplan according to there existing system. - 12 -

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