*** ******** ******, ***.***, Lake Mary, Florida 32746
Experienced working as Technical Consultant from USA for Altran primarily in Telecom and Networking domain.
12+ years experience in Development, Testing and maintenance of Communication Software (VOIP based applications) following the standards of SIP (Session Initiation Protocol).
Majorly involved in gathering functional requirements, implementation, system & functional testing and maintenance of SIP Terminals.
Experience in enhancing and testing of SIP User Agents and SIP-PSTN Gateway and SIP-ISDN Gateway and provided support of various Supplementary services.
Experience in enhancing and supporting call related features for Back-To-Back User Agent (maintaining call state for both Client as well as Server side) supporting Standard RFCs.
Working knowledge of developing Finite State Machine for handling of Events of Call related scenarios in various Call States.
Experience in working in Agile methodology and working experience on tools like JIRA and Confluence.
Experience in Management of technical team working for client.
Experience working as Client facing role in discussing requirements and building understanding for team to work on implementation part.
OS: Working knowledge on following OS: Linux and Vxworks
Experience in Network Protocols: UDP, TCP/IP, TLS/SSL, SCTP, DNS
Working knowledge of following tools: IPNetfusion (Network Simulator), Ethereal (Network Packet Analyser), Purify, WinCVS, Netscape Communicator, Cygwin, GDB (Debugger on Linux), clear case tool, Wireshark (Network Packet Analyzer), SVN, SIPP (SIP Message Simulator)
Exposure to OpenSips and Asterisk (SIP Proxy) and Asterisk PBX
DNS Server Configuration for FQDN testing
ISDN Analyzer for Capturing of ISDN Q.931 Packets
Experience of Development Languages: C & Basics of C++
Basic Knowledge of SQL and Python.
Work Experience (Projects Undertaken in Aricent) (Nov-2006-Present)
Keymile AG (VOIP Gateway)- (October-2010 to Sept-2016 & March-2017 to Present)
IPSS is a media gateway for both PSTN (Public Switched Telephone Network) and ISDN (Integrated Services Digital Network) users which performs conversion from PSTN/ISDN signalling to IP Network. It supports various SIP based supplementary services.
-Majorly involved in Maintenance/bug fixing and testing of various SIP Features.
-Worked on implementation and Integration Testing of feature: Advice of Charge which provides information to ISDN user related to Call Charges like Currency ID, Currency Amount, and Scale.
-Worked on a feature of Applied Codec for Ongoing SIP Calls.
-Resolved various client side critical issues and implemented Call Diversion Supplementary Service Activation and Deactivation for ISDN users.
-Worked on Asterisk scripts to modify it for testing Partial rerouting and Call Diversion supplementary services.
-Asterisk PBX setup for testing various ISDN scenarios.
-Implementation of Partial Rerouting feature for ISDN users.
-Worked on handling of Race-Condition Scenarios in SIP.
-Worked on Offline Configuration Feature for generation of simulation files of product.
-Worked on enhancing the HuntGroup functionality as per customer requirements.
-Worked as Consultant and implemented Multiple IP feature for VOIP GW
-Fixed critical field issues in ISDN area.
-Worked on various performance issues reported by client.
Ditech Networks(PVP: B2BUA)- (July-2008 to October-2010)
PVP is a packet voice processor which enhances the voice quality using algorithms like echo cancellation and also acts as B2BUA..
-Worked on Utility Integration which is basically used with this product to get the call related information, resource manager information and dsp drivers. Also worked in developing and Testing a CLI command which can provide us the all active calls (max 13K) related information like callid, to address, from address, call duration, call leg information with media ip and ports.
-Worked on various performance and functional signalling related bugs raised by testing team.
-Worked on a TCP related enhancement for PVP, From Header Transparency of PVP and also Routing related changes for “maddr” parameter enhancement with documentation appreciated by Customer.
-Worked for enhancement of a Call Transfer Service for PVP that called up “Enhanced REFER” Service which helps the user to transfer the call to third party by giving address of third party in header of SIP Message.
-Worked on various features like: Dynamic VQA Profile feature to change VQA Profile dynamically, DTMF-INFO Handling and End-to-End INFO Transparency for B2BUA.
-Handled Release management in team and weekly status calls with customer.
Customer Onsite Experience: Having customer onsite experience in California, US for 3 months in 2009 to support various customer related issues and testing of various SIP terminals.
Aricent In-house Product (IPTK) (November-2006 to July-2008)
The Project is based on SIP Protocol as per RFC 3261 for Session Initiation Protocol.
IP Phone Toolkit provides control plane and media plane infrastructure to handle the SIP signaling and media procedures. In addition, it provides the interfaces for user management and device management, thereby allowing the developer to focus on the application scenario. IP Phone Toolkit can be used in SIP phone client and edge gateways to implement various VOIP applications.
-Involved in development, testing and maintenance of SIP UA along with Unit Testing and Integration/Regression Testing.
-Worked on Session-Expires Feature in SIP from Requirement analysis to System Integration Testing following SDLC (Software Development Life Cycle)
-Worked on implementation of P-Headers for SIP UA.
-Involved in solving various SIP Stack related bugs.
-Porting: Ported IPTK on Windows XP, Windows Vista and Windows 7.
Bachelor of Engineering (June-2006) from University of Rajasthan in Electronics and Communication Engineering
Authorized to work in USA for any employer