Voice Engineer
Indexed description
Our flagship products include Tringy (IP-PBX), CPaaS white-label platforms, and eSIM provisioning systems, serving top telecom brands across India, the Middle East, Africa, Europe, and North America. We operate with an Agile development approach, strong technical leadership, and a 24/7 global support framework.
About The Role
We are seeking a Voice Engineer (Mid-level to Senior) with strong expertise in C/C++ programming, VoIP technologies, and Linux systems. You will play a key role in building and optimizing our next-generation Voice Engine that powers Voxvalley’s CPaaS and Tringy platforms. This role involves end-to-end development, debugging, optimization, and deployment of real-time voice services with a focus on scalability, low-latency performance, reliability, and operational excellence. Ideal candidates will demonstrate strong systems-level skills (POSIX APIs, multithreading, socket programming), hands-on protocol knowledge (SIP, RTP/RTCP, SDP, SDP negotiation), and experience with voice codecs and real-time audio processing.
Key Responsibilities
- Design, develop, and maintain VoIP-based voice applications and services with a focus on reliability and performance.
- Develop high-performance components in C/C++ on Linux for low-latency voice processing, including multithreaded and real-time code.
- Implement and troubleshoot SIP, RTP/RTCP, and voice codec flows (G.711, G.729, Opus, etc.), including SDP and NAT traversal handling.
- Debug and resolve complex voice, signaling, and media issues across multiple layers using tools like Wireshark, tcpdump, gdb, strace, perf, and valgrind.
- Integrate and optimize back-end components such as MySQL and Redis for state, call metadata and real-time lookups.
- Manage deployments and observability using AWS, Docker, container orchestration (Kubernetes preferred), and CI/CD pipelines.
- Collaborate closely with Product, QA, DevOps, and Support teams to ensure stable releases, runbooks, and 24/7 operational readiness.
- For Senior-level candidates: lead design reviews, mentor junior engineers, drive technical decisions, and take ownership of major modules and performance goals.
- (Optional) Configure and manage Kamailio, OpenSIPS and RtpProxy for SIP routing and media handling; tune SBCs and NAT traversal mechanisms.
- Minimum 4 years of experience in VoIP or Voice Engineering; 4-6 years typical for Mid-level, 6+ years preferred for Senior.
- Proficiency in C/C++ and Linux systems programming (POSIX APIs, pthreads, real-time scheduling).
- Strong understanding of SIP, RTP/RTCP, SDP, and VoIP call flows, including SIP trunking and session negotiation.
- Experience with voice codecs such as G.711, G.729, Opus and real-time audio processing.
- Hands-on experience with MySQL, Redis and service design for low-latency access.
- Familiarity with cloud and container tooling: AWS, Docker, Kubernetes, and Git.
- Excellent debugging, profiling, performance tuning, and problem-solving skills; comfortable with observability and load-testing.
- Experience with network fundamentals (TCP/UDP, NAT, STUN/TURN/ICE) and troubleshooting tools (Wireshark, tcpdump).
- Hands-on experience with Kamailio, OpenSIPS, PJSIP, or RtpProxy.
- Knowledge of SBCs, NAT traversal, and STUN/TURN/ICE mechanisms.
- Exposure to WebRTC, SRTP/TLS, GStreamer or media frameworks.
- Experience with microservices architecture, Kubernetes, and CI/CD pipelines for production deployments.
- Familiarity with observability stacks (Prometheus, Grafana), and logging/alerting best practices.
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