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VOIP Developer

Pulse Telesystems

4 - 6 years

Chennai

Posted: 26/05/2026

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Job Description

We are looking for a skilled VoIP Developer with 34 years of hands-on experience in designing, developing, and operating production-grade VoIP and telecom systems. The ideal candidate should have strong expertise in FreeSWITCH, OpenSIPS, SIP protocols, media handling, IVR development, and call flow management, along with the ability to troubleshoot real-time voice quality and signaling issues in live environments.


What Youll Do
  • Design, configure, deploy, and manage FreeSWITCH and OpenSIPS infrastructure
  • Own the SIP/RTP/media stack and ensure reliable telephony performance across all communication flows
  • Implement and maintain SIP trunking, call routing logic, and carrier integrations
  • Develop, test, and optimize IVR systems and dialplans using XML, Lua, and JavaScript
  • Lead client integrations end-to-end, including SIP trunks, enterprise PBXs, contact center platforms, and cloud telephony providers
  • Integrate VoIP systems with backend services and third-party APIs
  • Troubleshoot live VoIP and media issues using PCAPs, SIP traces, logs, tcpdump, sngrep, and Wireshark
  • Monitor and debug CDRs, SIP signaling, codec negotiation, NAT traversal, and RTP flows
  • Optimize voice quality by improving latency, jitter, packet loss, codec handling, and call success rates
  • Work on WebRTC, browser-based calling, softphone integrations, and SIP WebRTC media workflows
  • Build scripts and services using Python or Bash for provisioning, monitoring, automation, and incident response
  • Ensure high availability, scalability, stability, and performance of VoIP infrastructure
  • Collaborate with backend, DevOps, AI/platform, and product teams to deliver robust communication solutions
  • Act as a technical point of contact during client discussions, troubleshooting sessions, and production deployments


Required Skills
  • 34 years of experience building or operating production VoIP/telecom systems
  • Strong hands-on experience with FreeSWITCH , OpenSIPS, Kamailio, Asterisk
  • Deep understanding of SIP, SDP, RTP, SRTP, and VoIP protocols
  • Strong experience creating and managing dialplans and IVR systems
  • Proficiency in Python, Bash, Lua, JavaScript, or XML scripting
  • Experience with RTPEngine or equivalent media proxy solutions
  • Familiarity with Docker, Kubernetes, and Linux server administration
  • Strong troubleshooting skills for voice quality, signaling, NAT, and media handling issues
  • Experience using tools such as sngrep, tcpdump, Wireshark, and PCAP analysis
  • Knowledge of WebRTC and softphone technologies
  • Understanding of CDRs, log analysis, monitoring tools, and scalable telephony infrastructure
  • Strong communication and collaboration skills for working with internal teams and client engineering teams


Preferred Qualifications
  • Experience with multi-tenant VoIP platforms
  • Exposure to cloud deployment and scalable telecom infrastructure
  • Experience supporting enterprise-grade voice or AI communication platforms



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