Project Role : Application Support Engineer
Project Role Description : Act as software detectives, provide a dynamic service identifying and solving issues within multiple components of critical business systems.
Must have skills : Cloud Contact Center Operations
Good to have skills : NA
Minimum 3 year(s) of experience is required
Educational Qualification : 15 years full time education

Summary: We are seeking an experienced Voice Engineer with 3 to 5 years of experience in Asterisk, Kamailio, SIP, WebRTC, and VoiceXML, along with strong scripting skills in Java. The ideal candidate will be responsible for designing, deploying, and maintaining real-time communication systems, IVR applications, and contact center solutions. A solid foundation in networking protocols and routing, combined with hands-on experience in voice platforms, is essential for this role. Roles & Responsibilities: - Design, configure, and optimize SIP-based communication systems using Asterisk and Kamailio/OpenSIPS. - Deep knowledge of SIP Signaling Protocols and standards. - Implement SIP trunks, call routing policies, and carrier interconnects. - Develop advanced IVR solutions using VoiceXML and Asterisk dial-plans. - Integrate IVR workflows with databases, APIs, and third-party applications. - Implement call flows for self-service, transfers, and intelligent routing in Contact Center environments. - Develop automation scripts and tools in Python for call flow testing, monitoring, and provisioning. - Implement APIs and microservices for system integration and real-time reporting. - Automate deployment and scaling of voice infrastructure in production environments. - Analyze SIP messages, RTP streams, and QoS metrics for performance tuning. - Use protocol analysis tools (e.g., Wireshark, sngrep) for issue resolution. - Collaborate with DevOps/Network teams to ensure system reliability and scalability. - Apply knowledge of networking protocols (TCP/IP, UDP, DNS, DHCP, TLS, SRTP) and routing fundamentals. - Work with firewalls, SBCs, and NAT traversal for secure and reliable call delivery. - Ensure adherence to VoIP security best practices and compliance standards. Professional & Technical Skills: - Hands-on expertise with Asterisk and Kamailio/OpenSIPS in enterprise deployments. - Strong understanding of SIP, RTP, and VoIP call flows. - Proficiency in VoiceXML for IVR development and Asterisk dial-plan scripting. - Practical experience with WebRTC signaling, codecs (G.711, G.729, Opus), and NAT traversal techniques. - Strong Python scripting skills for automation, monitoring, and integrations. - Experience working in Contact Center environments (routing, IVR, CTI integrations). - Familiarity with Linux/Unix environments and system administration. - Solid knowledge of networking protocols and routing basics. - Familiarity with FreeSWITCH, OpenSIPS, and SBCs. - Experience integrating with CRM/Contact Center platforms (Genesys, Amazon Connect, Cisco, etc.). - Knowledge of VoIP security (TLS, SRTP, encryption, SBCs). - Exposure to cloud-based telephony platforms (Twilio, SignalWire, Plivo, Amazon Chime). - Certifications such as CCNA/CCNP Voice, Kamailio Developer Certification, or SIP School Certification (SSCA). - Nice to Have: Experience with Voice AI, TTS/STT engines (Google Dialogflow, Amazon Polly, Azure Cognitive). - Nice to Have: Familiarity with event-driven frameworks (Socket.io, asyncio, Kafka, RabbitMQ). - Nice to Have: Knowledge of observability and monitoring tools (Grafana, Prometheus, ELK stack). - Nice to Have: Exposure to CI/CD and DevOps pipelines for voice application deployment. Additional Information: - Bachelor’s degree in Computer Science, Electronics, or equivalent field. - Strong communication and documentation skills. - Ability to work under pressure and in 24x7 environments and weekends. - Experience in global operations or managed service environments is a plus. - Relevant professional certifications are preferred.