Manager - Customer Interaction Suite

We are looking for a highly experienced VoIP Developer and Subject Matter Expert (SME) with deep hands-on expertise in FreeSWITCH and Kamailio. The ideal candidate will design, develop, debug, and optimize carrier-grade real-time communication platforms supporting UCaaS and CPaaS use cases, including audio/video conferencing, WebRTC, and SIP-based systems.

This role requires strong systems-level programming skills in Go, C, and C++, deep understanding of VoIP protocols, and experience building scalable, asynchronous, API-driven platforms.

Key Responsibilities

Core VoIP & RTC Development

• Design, develop, and maintain FreeSWITCH and Kamailio modules

• Debug and optimize call signaling, media flows, and RTP streams

• Implement advanced SIP routing, load balancing, NAT traversal, and failover

• Work on audio/video conferencing systems (mixing, recording, SFU/MCU concepts)

• Handle high concurrency and low-latency real-time traffic

Protocols & Media

• Deep expertise in SIP, SDP, RTP/RTCP, SRTP, DTLS

• Strong experience with WebRTC internals (ICE, STUN, TURN, DTLS-SRTP)

• Troubleshoot interoperability issues across softphones, browsers, SBCs, and carriers

• Optimize codecs (Opus, G.711, G.729, VP8/VP9, H.264)

Backend & Systems Programming

• Develop high-performance components in Go, C, and C++

• Implement asynchronous and event-driven programming models

• Build and integrate RESTful APIs for call control, provisioning, analytics, and billing

• Work with message queues, state machines, and distributed systems

UCaaS / CPaaS Platforms

• Design and enhance cloud-native UCaaS and CPaaS solutions

• Implement features like:

  • Call routing and IVR

  • Call recording and transcription

  • Conferencing and collaboration

  • Messaging and notifications

• Integrate with third-party carriers, SBCs, and CPaaS providers

Debugging & Performance

• Analyze logs, SIP traces, RTP dumps, and core dumps

• Use tools like sngrep, Wireshark, tcpdump, gdb

• Identify and resolve memory leaks, race conditions, and latency issues

• Tune systems for high availability, scalability, and fault tolerance

Required Skills & Qualifications

Technical Skills

• FreeSWITCH (core, modules, dialplan, ESL, mod_sofia, mod_conference)

• Kamailio (routing logic, modules, scripting, dispatcher, registrar)

• Programming Languages:

  • Go (preferred for backend services)

  • C / C++ (VoIP core and module development)

• Protocols: SIP, SDP, RTP, RTCP, SRTP, DTLS, WebRTC

• Networking: TCP/IP, UDP, NAT, firewalls, load balancers

• APIs: REST, JSON, async APIs, webhook-based systems

Platform & Infrastructure

• Linux system internals

• Docker / Kubernetes (nice to have)

• Cloud platforms (AWS, GCP, Azure)

• Monitoring and observability (Prometheus, Grafana, ELK)

Nice to Have

• Experience with Janus, Asterisk, OpenSIPS

• Experience building SFU/MCU architectures

• Familiarity with CI/CD pipelines

• Knowledge of telecom security and compliance

• Contributions to open-source VoIP projects

Soft Skills

• Strong troubleshooting and analytical skills

• Ability to act as a technical authority and mentor

• Excellent communication and documentation skills

• Experience working with globally distributed teams