Design the Modules using the State Machine of each module.

Providing FLUTE interface to the Flute Module.

Implementation of DASH client requests handling.

Android Java service design and Modem Interface design.

Modem Interface implmentation for Interacting with QCT Jar libraries.

Desing Java SDK for accessing the eMBMS Middleware.

UPnP is a set of networking protocols for primarily residential networks. This stack helps in mobile devices or in any PCs or Home appliances to identify home network devices which are UPnP enabled. This stack is implemented using SSDP protocol, SOAP protocol, GENA protocol and HTTP Protocol. SSDP protocol uses UDP multicast messages to discover devices in the network. SOAP protocol is used to invoke actions on the remote devices. GENA protocol used to subscribe events and receive events from remote devices. GENA events are used to know the current status of the device. These all protocols are encapsulated with HTTP protocol and added with XML tags. This stack is implemented with multi-threaded and event based. Open source libxml2 parser used to parse XML body. This stack provides easy APIs for implementing any UPnP based applications. This UPnP Stack is certified by the UPnP

In Telecom domain for carrying Internet data from one country to another country OPTICAL data transmission medium is used. For supporting Optical data transmission Erricson uses MultiHaul equipment which can handle upto 40gb per second data transmission. This hardware can hold different types of card types. This plugin provides capability to user for configuring the route from one place to another place to transmit the data. By using this plug in we can create PATH,LINKs between NEs. This plug-in have the capability to raise alarms if any connection losses, Link broken, Heavy data transmission, any power problems. This plug-in works as interface between NE and EMS. It carries data from NE to EMS in the form of socket messages. NE generates messages in the form of QX messages, this messages are analyzed by plug in and then translated into I36 messages which is understand by EMS or NMS. This Plug in also provides performance measures in thresholds of 15minutes and 24 hours. These measures are used to calculate how accurate NE is utilizing.

In 3G technology, all phones have support for Video Calls Handling. For the operators, providing services to 3G subscribers it is difficult to handle with Normal Media servers. This Mediaserver have the capability to handle all Video Calls and Video RBT mechanism. This Media server have the capability to detect the call whether the call is Video or Audio Call. If it is SIP Call Depends on SDP it will detect type of calls. If operator is providing regular TDM model connectivity then that call will be converted into SIP using any standard Gateway. This Mediaserver having the features of Playing Video RBT, Play Video Call flows ,Collecting RFC2833 DTMF, INFO DTMF and Merging Video Calls between two Called Parties. Supports H.263 codec, can be easily Incorporate other Video & audio Codecs with third party Plug-ins.

Mediaserver applications need Boards and E1 interfaces. This MediaServer works on pure IP base, which uses SIP rfc 3261 and RTP protocols. This Application is developed using PJSIP open source stack for sip and rtp. This application will registers with openser proxy server. It can register with 32 user registrations. 1000 simultaneous calls are supported with call duration of 60 seconds without any problem, this was tested with simulator to check the stability of the application. For Identifying online calls status have used mysql database to update the calls status and maintaining CDRs for internal usage. Used Threads to handle all calls messages and to process all Message Queues. This supports Call Transfer and Early Media i.e. 183. If the Application is registered with Proxy then only it will receive the calls and place the calls to SIP users. This Same application have the ability to place outbound calls to SIP users. It supports 32 Party Conference calls simultaneous 5 rooms. It can increase depends on CPU & RAM configuration.

This Project provides as native library for java programing. Main aim is providing sip media , rtp transmission between two sessions. This project also includes conference server and call recording.

This project aim is to merge two different calls. One In coming call is received from the network, It will ask for DTMF and after completing the collection of DTMF, there is some other module which will check for the balance of the A-Party, and B-Party status. It will place the call to B-Party, after call is connected it waits for the B-Party confirmation to accept A-Party call or not. There is some other module in between these two calls to handle the Business logic about Call Charges. This project is implemented in Malavi, Here we have two options for charging the calls, one is sharing between two parties, other is call rate will be charged from the B-party.

This Application provides common Call Control Mechanism for R2MF , ISDN and ISUP protocols. This application can support FAX also. For Implementing the Channel wise status is Oracle database is used. From the application it will call the stored procedure with certain parameters to maintain the call report and updated status of each Channel. This application contains two threads for handling multiple call messages and one server socket. Here server socket is used to intimate the other application about to process the call. Two Queues are used to handle incoming and outgoing messages of the socket , each queue is processed by each thread. One Client Socket is maintained to generate the alarms about application or Link down problems.

Customer will call to a short code followed by destination party address. For all users it maintains a default music background for the conference. If any wants to change their music at that time they need to register before that usage of service. This project will handle more than 10 parties in a single conference. Simultaneously number of conferences can be handled depends on capacity of board.

This project is used to convert PSTN calls to SP Calls and vice versa. This project is developing using Natural Access and Fusion API on NMS Boards. It can support 4 E1s and one IP port. One IP Number is assigned to Board, so that it will registers in SIP Proxy server with username and password. when ever it receives a call it checks from database and picks corresponding registered sip user, that call is transferred to that SIP user.

Version 1.0 of the VoiceXML Platform support for creating telephony applications. With a broad base of support, VoiceXML enables platform-independent development and deployment of all types of telephony applications. Developers can use web-oriented programming techniques to control and automate inbound or outbound calls over all types of telephone lines (ISUP, ISDN, SIP,H323 etc.). Enhancements offered by the VoiceXML 2.0 specification enable a reduction in the time spent to create data-driven applications, and further enhancements to call control.

In3G technology, all phones have support for Video Calls Handling. For the operators, providing services to 3G subscribers it is difficult to handle with Normal Media servers. This Mediaserver have the capability to handle all Video Calls and Video RBT mechanism. This Media server have the capability to detect the call whether the call is Video or Audio Call. If it is SIP Call Depends on SDP it will detect type of calls. If operator is providing regular TDM model connectivity then that call will be converted into SIP using any standard Gateway. This Mediaserver having the features of Playing Video RBT, Play Video Call flows ,Collecting RFC2833 DTMF, INFO DTMF and Merging Video Calls between two Called Parties. Supports H.263 codec, can be easily Incorporate other Video & audio Codecs with third party Plug-ins.

This Application is developed only for handling SS7 Call Control part. This application is having two server sockets. When ever call is received that MSG will be send to erlang application to control the recharge operations. Depends on response Call will be further processed. For each Call IAM is received , and processed for the CGPN and CDPN details to process necessary operations.

This Application is developed for maintaining all hospital records. In-Patient records, Out-patient records and Pharmacy stock and billing maintenance. This software shows all daily, monthly, yearly reports. Payroll system of the hospital also included in this application.

This project is very unique project. Basic idea of this project to provide telephony features to non-telephony devices (TV, Tablet, Fridge etc.). UPnP based call control & media transfer handling is implemented. UPnP telephony supports call management, message management, and media management services. Some other services like input configuration, phone management are in design phase. This project facilitates to access the mobile from any devices with in the home network. Each device will be in the home network, and acts as telephony control point or telephony client, based on the application configuration. From any device we can access voice call or video call and can transferred to any device with in the home network.