Essay: Health Care For Cattles Using Telecommunication

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CHAPTER-1
INTRODUCTION
1.1 Introduction of Amul
The rapidly growing health care industry of India is one of country’s largest sectors in both terms of revenue and employment. It has been estimated that the health care industry of India for human will grow 40 billion by 2015.The reason behind growth in healthcare industry of India is increase population of India. There are 60% India reside in the rural area and depends on framing. The dairy business is one of the largest businesses in India which depends on animals. Amul is well-known rank dairy of world and rank one dairy of Asia, which the coverage of more than 30 million rural family employment. Amul is covering more than 1030 villages for animal care and around 3.38 mulch animals. People live in rural area are not aware of needful primary care information about their cattle like symptoms of disease.
1.2 Classifications of diseases
Major problems of the cattle:
1. Bacterial Diseases:
Anthrax is an infectious disease caused by a bacterium called Bacillus anthracis, It is carried by wild and domestic animals in Asia, Africa and parts of Europe.
2. Urinary System Diseases:
Most infectious diseases of the urinary system in small animals are aerobic bacterial infections. Mycoplasma is an uncommon urinary tract infection and is usually found as a co-infection with bacteria.
3. Surgical:
Veterinary surgery is surgery performed on animals by veterinarians. Advanced surgical procedures such as joint replacement, fracture repair, cancer surgery etc. is performed by Veterinary Surgeons.[4]
1.3 Why health care is require for cattle’s
Sometimes the care taker of the cattle are not aware of primary treatment their cattle, how to take care of cattle in different season, visit booking of veterinary doctor.
Due to this problem now we are develop new concept of health care for animals.
This services are to be Implemented to reduce the Emergence congestion, Cattle travel, working towards prevention and cure rather than treatment and sickness and overall healthcare costs.
The aim to decongest to reduce costs, for both Companies and farmers, and making healthcare available to all who need it is met.
The infrastructure is an all-inclusive term that encompasses a number of high-tech applications that involve providing remote care for cattle.
The action expected from the valerians as a result of such monitoring is to make a clinical decision on whether the patient requires:
Immediate treatments
Urgent doctor visit
Continued monitoring
The judgment on which patients can be put into a continuous remote monitoring mode Instead of frequents visits by veterinary doctors.
CHAPTER 2
LITERATURE SURVEY
2.1 REMOTE MONITORING USING WIRELESS CELLULAR NETWORKS
Author:Dr. Akram Aburas ,Prof. Khalid Al-Mashouq ,Dr. Zeyad Al-Hokail
Publication (with year) : IJDIWC Journal,2013
Conclusion:
In this paper, presented focused on the study on remote monitoring system in greenhouse which has capability of sending alert notification messages to farmers using GSM and SMS technology and is aimed to be a reliable and cost effective. There are a lot of technologies that have been created to perform the operation; however, many of the existing technologies would still require a great deal of human intervention. All the current technologies for remote monitoring detect and alert using wired technologies, but the proposed design not only alert and detect the change in thresholds it removes the distance criteria and labor to deploy the system and its maintenance. The proposed system can store and forward the information to the end user for forecasting and analysing the historical data. Apart from general remote monitoring, mobile operators are very much in need to monitor the call quality as perceived by the end-user. The mobile equipment with an application installed can be used as sensor to collect and monitor the call quality.
2.2 DIABETES MONITORING SYSTEM USING MOBILE COMPUTING TECHNOLOGIES
Author: Mashael Saud Bin-Sabbar,Mznah Abdullah Al-Rodhaan
Publication (with year) : (IJACSA) Journal,2012
Conclusion:
In this paper, we discuss our implementation of a diabetes monitoring system for managing and monitoring diabetes patients. The specification of our implemented project is based on Android platform. The architecture of the system is depends on three units: the patient unit, the endocrinologist unit, and the general physician unit. These three units are working together forming an integrated diabetes monitoring system.
2.3 NOVEL COMPONENT-BASED DEVELOPMENTMODEL FOR SIP-BASED MOBILE APPLICATION
Author: Ahmed Barnawi, Abdulrahman H. Altalhi, M. Rizwan Jameel Qureshi and Asif Irshad Khan
Publication (with year) : (IJSEA), January 2012
Conclusion:
In this paper, explain need to expedite the process of examination in order to meet the increasing enrolment of students at the universities and institutes. Sip Based Mass Mobile Examination System (SiBMMES) expedites the examination process by automating various activities in an examination such as exam paper setting, Scheduling and allocating examination time and evaluation etc. The SiBMMES will assess to students by conducting online/mobile based objective exam. This will be highly customizable for any university who acquired to adopt similar IMS based examination system and Faculties to create their own dashboard (create set of questions, creates groups, adds related students into the groups, schedule exams, etc). Further the exams will be associated with specific groups so that only associated students can appear for the test, result will be notified to the student either through SMS/email.
2.4 LOCATION-BASED SERVICES USING SIP
Author: Abdullah Azfar, Md. Sakhawat Hossen, Mar??a Jos?? Peroza Marval, Razib Hayat Khan.
Publication (with year): (IJEST), 2010.
Conclusion: from this paper I get idea about how location based service help with use of SIP protocol.
CHAPTER 3
TECHNOLOGY REQUIREMENT
1. Inbound Toll Free Number
2. Outbound Calling Number
3. EPBAX System
4. Servers
5. Workstations
6. Networking
7. Routers
8. Dialer
9. Database
10. Call Management Software
3.1 TOLL FREE NUMBER
A number on which a customer can call up, where they will not be charges for the calls. (I.e. Customer Care)This number would be used by the customer (From the villagers who need veterinary doctor) to the Call Centre for Veterinary Help; the charges would be applicable to the call Centre on per minute basis. This number will be common, for all the zones that would be decided by AMUL LTD.
3.2 OUTBOUND CALLING NUMBER
A number on which would be used to call Veterinary Doctor by the Call Centre.
This would be a Hot Transfer (i.e. Two Way Calling) i.e when the customer is calling for Veterinary help he/she would be directly in conference with the doctor, so that the understanding of the location for Doctor would become easier.
3.3 EPBAX SYSTEM
EPBAX is basically used for landing the incoming & outgoing calls. This is also used to manage the Inbound & Outbound lines of the calls. These systems transfer the calls to the server via Dialogic Card & so on & so forth.
3.4 SERVER
This will be used to Manage:
Internal Domain Network
Fall Back Domains Server
Dialer Software
Call Management Software (Ticket)
Doctor’s Database (Area wise)
Antivirus
3.5 WORKSTATION
The computer systems which will be used by the Agent (Person attending the calls).
These are normal desktop which are used by all of us in day to day life, but this systems will be equipped with:
1. Dialer Software
2. Call Record Management Software
3. User Access Permissions (Restrictions)
4. Head Phone
5. Dial Pad
6. Soft Dialer
3.6 NETWORKING & ROUTERS
Networking & Routers are used to connect & manage the Application Server, Database Server, Dialer & users & also they are used in routing the inbound/outbound calls in a proper fashion.
3.7 DATABASE
This is complete data of
1. Detailed List of doctors ‘ area wise
2. Details of all the incoming calls
3. Emergency Contact details of AMUL Authority
3.8 CALL MANAGEMENT SOFTWARE DETAILED DESCRIPTION
This software manages the details of the incoming calls
The primary work of the software is to keep the records of the incoming calls i.e
1. Name
2. Address of the Customer (Calling for Veterinary Help)
3. Contact Details
4. Generate Case Number (For Reference)
Secondarily, it will automatically search for the nearest available Veterinary doctor to attends the case on immediate, in case the doctor is busy it will immediately give the name & number of the nearest available doctor.
The contact details will enable the call Centre to call up the doctor & take him in the conference immediately, which can help the Doctor reach the destination immediately. This software will also SMS the doctor the name & address of the customer with the Case Number. Once the Doctor reaches the destination & completed the treatment, they have to call back & close the case.
If the call from the doctor do not turns up after 1 Hrs / 2 Hrs, the call Centre will call up the doctor to verify the status of the calls. This will also help Call Centre to submit the EOD (End of Day) call reports to AMUL & for internal verification to reduce any human errors that are caused.
CHAPTER 4
SOFTWARE DETAIL
4.1 Introduction of Asterisk Software
Asterisk is a telephone private branch exchange (PBX) software implementation; Its name comes from the asterisk symbol’*’. It was created in 1999 by Mark Spencer of Digium., it allows attached telephones to make calls to one another, and to connect to other telephone services, such as the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services.
Asterisk is released under a dual license model, one is using the GNU General Public License (GPL) as a free software license and second one as a proprietary software license to permit licensees to distribute proprietary, unpublished system components.
Originally designed for Linux, Asterisk also runs on a variety of different operating systems including NetBSD, OpenBSD, FreeBSD, Mac OS X, and Solaris. Asterisk is small enough to run in an embedded environment like Customer-premises equipment-hardware running OpenWrt.
4.2 Features of Asterisk
The many features includes in Asterisk software in available proprietary PBX systems: voice mail, conference calling, interactive voice response (phone menus), and automatic call distribution. Users can create new functionality by writing dial plan scripts in several of Asterisk’s own extensions languages, by addition of custom loadable modules written in C, or by implementing Asterisk Gateway Interface (AGI) programs with use of any programming language capable of communicating via the standard streams system (std in and std out) or by network TCP sockets.
A wide range of Voice over IP protocols supports by Asterisk, including the Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), and H.323. Asterisk can interoperate with most SIP telephones, acting both as registrar and as a gateway between IP phones and the PSTN. The Inter-Asterisk eXchange (IAX2) protocol, RFC 5456, native to Asterisk, provides efficient trunking of calls among Asterisk PBXes, in addition to distributed configuration logic, and call completion to VoIP service providers who support it.
Asterisk allows deployers to build new telephone systems, or gradually migrate existing systems to new technologies by supporting a mix of traditional and VoIP telephony services,. Some sites are using Asterisk servers to replace proprietary PBXes; others to provide additional features or to reduce costs by carrying long-distance calls over the Internet. Asterisk was one of the first open source PBX software packages.
In addition to VoIP protocols, Asterisk supports many traditional circuit-switching protocols such as ISDN and SS7. This requires appropriate hardware interface cards supporting such protocols, marketed by third-party vendors. Each protocol requires the installation of software modules. With these features, Asterisk provides a wide spectrum of communications option [10]
4.3 Asterisk software system Architecture
4.3.1 Asterisk software system Architecture
4.4 Asterisk Capabilities
Telephony gateway (TDM channels – PRI,POTS)
VoIP Gateway (IP channels)
IVR system (Interactive Voice Response)
Voicemail System
Scriptable telephony-to-anything (Perl, C, etc.)
4.5 Advantages of asterisk software
The Asterisk software includes features like: voice mail conference calling, interactive voice response and automatic call distribution.
Users can create new functionality by writing dial plan scripts in several of Asterisk’s own extensions languages, by adding custom loadable modules written in C, or by implementing Asterisk Gateway Interface (AGI) programs using any programming language capable of communicating by network TCP sockets.
Asterisk can interoperate with most SIP telephones, acting both as registrar and as a gateway between IP phones and the PSTN.[1]
4.6 Dialler Software
GOautodial is a CentOS base open source web based call center system.
GOautodial automatically installs the GO autodial applications (GO admin, GO reports and GO agent), Vicidial, Mysql, PHP, Asterisk, Lime survey and other open source software to have a fully functional call center system.
Major Features
Inbound, Outbound and Blended call handling
Broadcast and survey dialing
Web-based agent application
Web-based system administration
Wizard based configurations
Ability to have agents operate remotely (home based)
Full call recording
GOautodial is use for call transfer, call hang up, call pause, call resume.[3]
4.7 Database server
Database server is the term used to refer to the back-end system of a database application using client/server architecture.
Database server performs tasks such as data analysis, storage, data manipulation, archiving, and other non-user specific tasks.[6]
4.7.1 Database server
4.8 ASP.NET Software
We should use Windows Server when comparing ASP.net with another application server because In the ASP.NET world, all of the functionality that makes up an application server, such as a web server (i.e., IIS), an application framework (.NET Framework, ASP.NET) a database engine (e.g., MSDE or SQL Server Express) and transaction services (and related services, e.g., COM+ or Indigo) are built into the OS. So, with all of these services combined as the comparison platform. Incidentally, there’s no additional cost for these services running on the OS as there is with other app servers.
4.8.1 Why Asp?
Active Server Pages (ASP) is Microsoft’s technology for building interactive web pages, using HTML similar techniques authoring. It has been described as “the tool that will allow you to build an almost infinite number of web pages with only a few files”. It is intended to be usable by relative beginners, but has the potential to be used for advanced developments by professionals.
ASP.net is still best supported by Visual Studio .net, but the days when it was confined to JScript and VBScript are long gone, since “classic” ASP was replaced by ASP.net. Microsoft said it now supports more than 25 languages. Microsoft has also set up the Web Matrix for ASP.net, its first venture into Linux-like community-support, development and maintenance
4.8.2 What’s it for?
Building and deploying applications and pages with the minimum of effort and error, using “an HTML-like style of declarative programming”. ASP.net enables interactive web pages to be built using just a text editor, but the tools and techniques available are a great deal richer with Visual Studio .net, including code support, integrated debugging and easier deployment. ASP.net mobile controls automatically generate the appropriate code for phones and PDAs. Web Matrix has built-in support for Microsoft’s Access and SQL Server databases, and for XML web services.
4.8.3 Where is it used?
From local web design services to advanced finance applications, such as hedge funds. It is no longer restricted to Microsoft-only installations, with ports to other web servers and operating systems.
CHAPTER 5
HARDWARE DETAIL
5.1 Introduction of PRI line
A PRI line is end to end digital circuit .There is only one line physically terminating on the customer PBX but still a PRI line can receive/send 30 calls simultaneously.
A PRI (Primary Rate Interface) line is a form of ISDN (Integrated Services Digital Network) line which is a telecommunication standard that enables traditional phone lines.It is use for to carry voice, data and video traffic, among others.
A PRI circuit consists of two pairs of copper lines terminating on a modem from a service provider premises to the customer premises. It can multiplexing/de-multiplexing techniques to carry more than one channel in a single circuit. There are two common part of PRI lines ‘ E1 (which carry 30 channels) and T1 (which carry 23/24 channels).
Each channel in it provides 64 Kbps for data transmission.
A PRI line can connect to both Analog and Mixed EPABX systems and also the newer Internet Protocol PBX systems. A PRI Card terminate the PRI circuit on the PBX.
A PRI line can also be used to connect two PBX systems just providing 30 channels for interoperability.
5.1.1PRI Line Architecture
If thirty different analog trunks are taken in place of one PRI line, the cost of terminating all the thirty analog trunk lines becomes higher than terminating one PRI line.
Terminating 30 analog trunks in a PBX also requires more free slots and cards than the one slot sometimes it occupied only one or two PRI trunk cards.
PRI lines can be used for voice communication, data communication, video conferencing, faxing. Also PBI can do simultaneously on different different channel.
This lines are end-to-end digital lines so voice clarity is much better than analog trunk lines.
Since they are digital lines, PRI lines are more reliable and trouble shooting is also easier with them. They are mostly on a fiber core ring and hence there is some redundancy.
It is harder to tap into digital lines and listen to the conversations.
There are flexible billing options available with most of the PRI service providers. The billing can be centralized or distributed (department wise, etc).
PRI lines take lesser time to establish calls then analog trunk lines.
It is possible to offer both voice and data in the PRI line. Some service providers have dynamic offerings where data is transmitted in all the channels that are free (not occupied by voice) at that given point of time.
5.2 PRI Features
Direct Inward Dialing: For each PRI line, the service provider would provide more around 100-500 numbers which can be used by outsiders to call the extension directly, instead of having to go through the PBX Auto-attendant.
Caller ID: Since all the extensions have their own number, this unique number will be displayed in the phones that they are calling to. Some call centre applications are based on the unique caller ID number for differentiation of services.
It is possible to offer both voice and data in the PRI line. Some service providers have dynamic offerings where data is transmitted in all the channels that are free (not occupied by voice) at that given point of time.
Call hunting: (Where the call lands in any channel that is free, instead of the called number specifically ‘ For example, if there is one board number but a number of people are calling in at the same time and still a channel is allocated to them .With analog lines, if one number is busy, they need to call in another number manually) is possible by default with a PRI connection, but for the analog trunks this facility needs to be extended by the service provider and enabled on the PBX, involving additional cost at times.
PRI lines are end-to-end digital lines and hence the clarity is much better than analog trunk
Since they are digital lines, PRI lines are more reliable and trouble shooting is also easier with them. They are mostly on a fiber core ring and hence there is some redundancy.
It is harder to tap into digital lines and listen to the conversations.
There are flexible billing options available with most of the PRI service providers. The billing can be centralized or distributed (department wise, etc).
PRI lines take lesser time to establish calls then analog trunk lines.
Some service providers offer flexible plans where instead of the full 30 channels, they provide and charge for only 20 channels etc. This makes PRI lines more economical for smaller companies.
5.3 Dis-advantages of PRI lines
A PRI line is economical only if the minimum rental charged by the service provider for a PRI line is more than the average value of calls with analog trunk lines every month in an organization. Otherwise, the usage may not even cross the free call value provided by the service provider for a PRI line.
A PRI line is not so economical for long distance/ international calling. An ITSP or SIP trunk service provider who takes the calls over the internet might charge much lesser for international long distance calls.
Inter branch communication between the branches is not free of cost with PRI lines (Some PRI service providers provide this facility, but all your branches may need to have PRI lines from the same service provider and there also might be a minimum revenue commitment for the same). With VOIP systems, inter-branch communication can be done over internet/ leased lines hence reducing the cost drastically.
The cost of a single PRI card to connect to your EPABX/ IP PBX is still very high. Most of these cards are proprietary, meaning you can buy them only from your EPABX vendor.[5]
CHAPTER 6
VOICE OVER INTERNET PROTOCOL
6.1 Introduction Of VoIP
Voice over Internet Protocol (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol(IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, IP communications, and broadband phone service.
The term Internet telephony specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN). The steps and principals involved in originating VoIP telephone calls are similar to traditional digital telephony, and involve signaling, channel setup, digitization of the analog voice signals, and encoding.
Instead of being transmitted over a circuit-switched network, however, the digital information is packetized and transmission occurs as Internet Protocol (IP) packets over a packet-switched network Such transmission entails careful considerations about resource management different from time-division multiplexing (TDM) networks.
Early providers of voice over IP services offered business models and technical solutions that mirrored the architecture of the legacy telephone network. Second generation providers, such as Skype, have built closed networks for private user bases, offering the benefit of free calls and convenience, while potentially charging for access to other communication networks, such as the PSTN. This has limited the freedom of users to mix-and-match third-party hardware and software.
Third generation providers, such as Google Talk have adopted the concept of federated VoIP ‘ which is a departure from the architecture of the legacy networks. These solutions typically allow dynamic interconnection between users on any two domains on the Internet when a user wishes to place a call.
VoIP systems employ session control and signaling protocols to control the signaling, set-up, and tear-down of calls. They transport audio streams over IP networks using special media delivery protocols that encode voice, audio, video with audio codecs and video codecs as Digital audio by streaming media. Various codecs exist that optimize the media stream based on application requirements and network bandwidth; some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs. Some popular codecs include ??-law and a-law versions of G.711, G.722 which is a high-fidelity codec marketed as HD Voice by Polycom, a popular open source voice codec known as iLBC, a codec that only uses 8 kbit/s each way called G.729, and many others. VoIP is available on many smartphones, personal computers, and on Internet access devices. Calls and SMS text messages may be sent over 3G or Wi-Fi.
6.2 Protocols of VoIP
Voice over IP has been implemented in various ways using both proprietary protocols, as well as protocols based on open standards. Examples of the VoIP protocols are:
H.323
Media Gateway Control Protocol (MGCP)
Session Initiation Protocol(SIP)
Media Gateway Control or H.248 (Megaco)
Real-time Transport Protocol (RTP)
Real-time Transport Control Protocol (RTCP)
Secure Real-time Transport Protocol (SRTP)
Session Description Protocol(SDP)
Inter-Asterisk exchange (IAX)
Jingle XMPP VoIP extensions
Skype protocol
The H.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic, as well as local area network services. However, since the development of newer, less complex protocols such as MGCP and SIP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic. In particular, the Session Initiation Protocol (SIP) has gained widespread VoIP market penetration.
6.3 Use of VoIP
6.3.1 Example of residential network including VoIP
A major development that started in 2004 was the introduction of mass-market VoIP services that utilize existing broadband Internet access, by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network (PSTN). Full-service VoIP phone companies provide inbound and outbound service with direct inbound dialing. Many offer unlimited domestic calling for a flat monthly subscription fee. This sometimes includes international calls to certain countries. Phone calls between subscribers of the same provider are usually free when flat-fee service is not available. A VoIP phone is necessary to connect to a VoIP service provider. This can be implemented in several ways:
Dedicated VoIP phones connect directly to the IP network using technologies such as wired Ethernet or wireless Wi-Fi. They are typically designed in the style of traditional digital business telephones.
An analog telephone adapter is a device that connects to the network and implements the electronics and firmware to operate a conventional analog telephone attached through a modular phone jack. Some residential Internet gateways and cable modems have this function built in.
A softphone is application software installed on a networked computer that is equipped with a microphone and speaker, or headset. The application typically presents a dial pad and display field to the user to operate the application by mouse clicks or keyboard input.
It is becoming increasingly common for telecommunications providers to use VoIP telephony over dedicated and public IP networks to connect switching centers and to interconnect with other telephony network providers.
Smartphones and Wi-Fi-enabled mobile phones may have SIP clients built into the firmware or available as an application download.[12]
CHAPTER 7
SIP PROTOCOL
7.1 Introduction Of SIP
SIP, the session initiation protocol, is the IETF protocol for VOIP and other text and multimedia sessions, like instant messaging, video, online games and other services.
Abstract from the RFC 3261 (formatted_and_explained version) – SIP: Session Initiation Protocol
This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.
SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user’s current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.
SIP is very much like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for phone calls are defined in SDP -the session description protocol.
SIP is a text-based protocol that uses UTF-8 encoding
SIP uses port 5060 both for UDP and TCP. SIP may use other transports
SIP offers all potentialities of the common Internet Telephony features like:
call or media transfer
call conference
call hold
Since SIP is a flexible protocol, it is possible to add more features and keep downward interoperability.
SIP also does suffer from NAT or firewall restrictions. (Refer to NAT and VOIP). SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The following features of SIP play a major role in the enablement of IP telephony and VoIP:
Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported ””’? recognizing that not all the parties can support the same level of features. For example video may or may not be supported; as any form of MIME type is supported by SIP, there is plenty of scope for negotiation.
Call Participant Management: During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
Call feature changes: A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as ‘voice-only’, but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call
Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.[11]
7.2 Compare Sip Vs H.323
SIP Protocol H.323 Protocol
SIP encoded in ASCII text formate. H.323 encoded in binary formate.
Use Tcp. Use Udp.
Not complex Complex.
CHAPTER 8
VOICE LOGGER
8.1 Circuit Diagram of Voice logger
Detail Description Of Voice Logger’s ICS
8.1 STC
STC11/10xx Series are a single ‘chip microcontroller based on high performance IT architecture 80C51 CPU, which is produced by STC MCU Limited.With the enhanced kernel, STC11/10xx series execute instruction in 1~6 clock cycle9about6~7 times the rate of standard 8051 device0, and have fully compatible instruction set with industrial-standard 80C51 series microcontroller. In ‘system-Programming(ISP) and In-Application- Programming(IAP) support the users to upgrade the program and data in system.ISP allows the user to download new code without removing the microcontroller from actual end product; IAP means that the device can write non volatile data in Flash memory while the application program is running. The STC11/10xx series retain all features of standard 80C51. In addition, the STC11/10xx series have a extra I/O port (P4), a 6-sources,2-priority-level interrupt structure, on-chip crystal oscillator, and a one-time enable watchdog timer.
8.1.1 Pin Diagram of STC
FEATURES OF PCM 1801
Enhanced 80C51 Central Processing Unit, 1T per machine cycle, faster 6-7 times than the rate of a standard 80C51.
‘ Operating voltage range;5.5V~4.1V/3.7V or 2.1V/2.4V~3.6V (STC11/10xx series)
‘ Operating frequency range; 0-35MHz, is equivalent to standard 801;0~420MHz
‘ STC11/10xx series Flash program memory; 2/4/6/8/10/12/14K
‘ Two 16-it timer/counter, as the same as timer0/Timer 1of standard 80C51
8.2 PCM 1801U
The DEM-PCM1801 is a basic evaluation fixture for the PCM1801 16-bit stereo audio A/D converter. It may be used as part of the custmer’s prototype system design, or in conjunction with the DEM-DAI mother board to provide a complete evaluation platform for the PCM1801.
8.2.1 Pin Diagram of PCM 1801U
FEATURES OF PCM 1801
‘ Easy Configuration Using The On-Board Dip Switch
‘ Compatible With The Dem-Dai Mother Board
‘ Power Supply, Digital I/O, And Analog Input Connectors Requires A Single +5v Power Supply
8.3 CH 372
CH372 is a USB bus universal device interface chip.It is upgrade production of CH371,and function predigested edition of CH375.At location,CH372 has 8-bit data bus and read strobe input,write strobe input and interrupt output.In computer system, the equipped software of CH372 supplies operation interface which is handy and wieldy.When CH372 communicates with local MCU is just like read/write file.
8.3.1 Pin Diagram of CH 372
FEATURES OF CH 372
‘ Full-speed USB device interface,complies with Universal Serial Bus Specification Revision 2.0,plug-and-play,only needs crystal and capacitance as peripheral components.
‘ Universal Windows drive program supplies device layer interface,and supplies API application layer interface via DLL.
‘ Universal local 8-bit data bus,4-wire to control:read strobe,write strobe,chip select input and interrupt output.
‘ Operation at either +5v or +3.3v power supply input,support low-power mode.

CHAPTER 9
RESULT
Result For Call Booking
In Amul call center there are 13 operators,all agent in call center have own agent Id on their computer,when they log on their computer and open Call Management Software and if any call come,on display it see live call when you select call park option you can receive that call after and go with Call Booking option.
In Call Booking all above option you can see on your desktop after then whatever information you need from customer you fill in this section.
Doctor’s Duty & Route
When every doctor assign for their case after that they have to fill up the following document by selecting Doctor’s Duty & Route option.
View Case Paper
By selecting View Case Paper option its doctor job to fill up the doctor code,gender and all other things related to farmers cattles.
Call Report
Here in this option we get all the information related to customer call and which doctor should be assign for which case how much time should taken for it to complete the case.
Doctor Wise Visit
By selecting this option we get all the information related to doctor which handle the case also we got information about that how many doctor have attend the case and which are still unattended.
Center Report
Get information related to center and attended and unattended also about cancel case.
Vehicle Report
Get information related to vehicle which assign for all doctor those are on duty.
Report & Analytical
By this we get information related to call duration total time, average time, waiting time total no of call disconnected and reminder.
Call Conferencing
By selecting barge option we can do conference call in Go Auto Dialler Software.
Listen The Talk Of Operation
By this Listen option we can listen the talk done by operator to customer.
Reports & Analytics
By this type of analytics method we get exact information related on which date how many calls come and how many time duration they taken for it also we get information related about Inbound and Outbound calls.
Graphical representation
By this graphical method we get information related to how many normal case, emergency case, call disconnect, reminder call 1,2,3,4,5, or 6 are coming.
Simulation of ST064C EZ2K833
ST064C simulation result u1 and u2 are variables.u1 is work as interrupt pin and u2 is a output at different points.
Here it show that how value should be change from u1 to u2.At some point u1 is high and u2 is low or same result give as a u1 on same point of u1.
In this figure it show above part of simulation result of ST064C and how it should be work which is explain above.
Simulation of STC 35I-LQFP32G
It is result of STC microcontroller.here u2 is probe .There are upper and lower addressing buses so total 16 address lines are available..8 data buses.u2 work as XTAL, ALE,WR and RD in STC Simulation result.
Simulation of PCM 1801U
It simply show how serial clock and serial data send parallel in PCM 1801U Ic.Both work parallel so there is no any error should be present in it.
CHAPTER 10
CONCLUISION AND WORK PLAN
10.1 Conclusion
The Cattle monitoring services are to be Implemented to reduce the Emergence congestion, Cattle travel, working towards prevention and cure rather than treatment and sickness and overall healthcare costs.
10.2 Work plan
 
CHAPTER 11
REFERENCES
References
11.1 Papers
1. Dr. Akram A., Prof. Al-Mashouq k., Dr. Al-Hokail z, Remote Monitoring Using Wireless Cellular Networks; (IJACSA,Vol. 4, No. 2, 2012
2. Mashael S.; Mznah A; Diabetes Monitoring System Using Mobile Computing Technologies; IJDIWC Vol. 4, No 2,79-83 The Society of Digital Information and Wireless Communications, 2013
3. Ahmed B., Abdulrahman H.; Rizwan M.; Qureshi J. and Khan A; Novel Component-Based Development model For Sip-Based Mobile Application; IJSEA, Vol.3, No.1, January 2012
4. Abdullah A, Md. Sakhawat H, Mar??a Jos?? P, Razib H; Location-Based Services Using Sip; IJEST,2010
11.2 Website
1 www.convergencetechnologycenter.org
2 http://www.ibef.org/industry/healthcare-india.aspx
3 http://goautodial.org/
4 http:a//www.nda.agric.za/docs/Infopaks/diseases.htm
5 http://www.excitingip.com/687/what-is-a-pri-line-what-are-the-advantages-and-limitations-of-pri-circuits/
6 http://www.webopedia.com/TERM/D/database_server.html
7 http://www.keithjbrown.co.uk/vworks/images/mysql.png
8 www.cs.tau.ac.il
9 http://www.baytalkitee.com/voice_loggers.html
10 en.wikipedia.org/wiki/Asterisk_(PBX)
11 www.voip-info.org/wiki/view/SIP
12 en.wikipedia.org/wiki/voice_over_ip

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