Role Overview We are looking for a Senior Real-Time Voice Engineer to design, build, and optimize high-performance voice infrastructure that powers millions of automated, interactive AI voice experiences. In this role, you will own the end-to-end media pipeline—handling everything from edge routing and WebRTC streaming to real-time audio synchronization. You will work closely with our engineering teams to ensure that real-time voice interactions across phone lines and web/mobile apps are seamless, ultra-low-latency, and highly resilient under heavy enterprise scale. What You’ll Do
• Build Streaming Media Pipelines: Design and implement highly concurrent, real-time backend systems that ingest, buffer, and process live audio streams with sub-second latency.
• Optimize Interactive Voice Systems: Implement core voice-handling features, including real-time speech detection, interruption handling, and streaming audio playback systems.
• Manage Communications Infrastructure: Configure and scale session border controllers (SBCs), media gateways, and WebRTC media servers to route multi-channel audio traffic reliably.
• Integrate Applied AI Frameworks: Develop secure, performant integrations with industry-leading Speech-to-Text (STT) and Text-to-Speech (TTS) streaming APIs.
• Own Platform Telemetry: Establish deep observability across the voice stack to track packet health, system latency, and audio quality, ensuring a high-fidelity experience for every user. What We’re Looking For WebRTC & Digital Media Streaming
• WebRTC Protocol Internals: Strong understanding of the WebRTC stack, including SDP negotiation, ICE, STUN/TURN, and DTLS/SRTP security.




• Media Server Deployments: Hands-on experience working with open-source WebRTC media servers and SFU/MCU frameworks (e.g., Janus, Mediasoup, LiveKit, or Pion).
• Audio Optimization: Experience tuning network congestion algorithms, managing
• Telephony Infrastructure & Protocolsnetwork jitter buffers, and handling packet loss concealment.
• SBC & Gateway Management: Practical experience deploying and managing carrier-grade SBCs (e.g., AudioCodes, Ribbon) or open-source SIP proxies (Kamailio, OpenSIPS).
• Telephony Protocol Mastery: Deep knowledge of SIP, RTP/RTCP, and SDP state machines, including experience writing header manipulation rules.
• Media Capture & Transcoding: Experience with media duplication techniques (SIPREC / media forking) and audio codec mechanics (Opus, G.711) to normalize streaming data formats.
• Packet-Level Troubleshooting: Fluency using packet analyzers like Wireshark to diagnose signaling drops, media stream anomalies, and network delays. Modern Software & AI Infrastructure Engineering
• Production Coding: Fluency in either Go (Golang) or Java for building resilient backend middleware.
• Streaming Media Architectures: Hands-on experience with WebSockets and WebRTC for real-time bi-directional data and audio streaming.
• AI & Speech Models: Experience integrating streaming API endpoints with contemporary STT (Speech-to-Text) frameworks and TTS (Text-to-Speech) providers like ElevenLabs.
• Engineering Best Practices: Strong background in writing clean code, designing clean SDK/API abstractions, and managing code via Git-driven CI/CD pipelines.

📌 Principal Voice Platform Engineer (Gurugram)
🏢 Birdeye
📍 Gurugram