3. 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
4. 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.
5. 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
6. 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
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7. 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.
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8. 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.
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9. 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.
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10. 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 .
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11. 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
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12. 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-
13. 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-
14. 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.
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15. 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
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16. 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-
17. 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.
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18. 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-
19. 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.
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20. 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.
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21. 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
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22. 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.
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23. 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
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24. 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.
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25. 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.
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26. 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 -
27. 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.
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