Principal Software Engineer - Real-Time Voice
Remote • Berlin Office
Posted 5d ago
About the job
Parloa is shaping the foundation of a new era in customer experience where customer support is no longer transactions, but meaningful exchanges. As Principal Software Engineer for Voice/VoIP, you will own the technical direction of our real-time voice infrastructure. This is a hands-on technical leadership role where you will set architectural direction, solve complex problems, and elevate the engineering organization's approach to voice systems. You will collaborate with infrastructure, AI/ML, and product teams to ensure our voice stack is fast, reliable, and globally scalable.
Responsibilities
- Own the architecture and technical roadmap for Parloa's Voice/VoIP infrastructure, including SIP trunking, media handling, and real-time audio pipelines.
- Design for low-latency, high-availability voice at enterprise scale across global telephony carriers and providers.
- Drive technical decisions on protocols and standards (SIP, SDP, RTP/SRTP, RTCP, WebRTC, ICE, STUN/TURN) and real-time media transport.
- Partner with AI/ML teams to optimize the integration between voice infrastructure and conversational AI models (ASR, TTS, turn-taking, barge-in).
- Diagnose and resolve deep, systemic issues in call quality, latency, jitter, and packet loss across distributed, multi-region deployments.
- Define and uphold engineering standards for reliability, observability, and security across the voice stack.
- Mentor senior and staff engineers, acting as a technical multiplier across the organization.
- Evaluate and integrate with telephony providers, carriers, and SIP trunk vendors as Parloa expands into new markets.
- Represent Parloa's technical voice expertise in conversations with enterprise customers and partners.
Requirements
- 10+ years of engineering experience, including 5+ years hands-on building, operating, or debugging production Voice/VoIP, SIP, WebRTC, or carrier-grade voice infrastructure.
- Strong knowledge of SIP, SDP, RTP/SRTP, RTCP, WebRTC, ICE, and real-time media transport protocols.
- Hands-on understanding of audio codecs and media behavior, including Opus, G.711/G.722, codec negotiation/transcoding, DTMF/RFC2833, jitter buffers, packetization, and codec-related quality/latency trade-offs.
- Experience building or operating large-scale, distributed, real-time systems with strict latency and availability requirements.
- A track record of owning architecture for complex systems and driving technical decisions with broad organizational impact.
- Experience working with telephony carriers, SIP trunk providers, or PSTN integrations at scale.
- Deep, hands-on ownership of production SIP-based voice infrastructure at enterprise scale (e.g. Kamailio, FreeSWITCH, Asterisk, pjsip).
- Strong systems programming skills in one or more of Go, C, C++ or Rust, with the ability to work at the media/transport layer.
- Comfort operating in cloud environments (AWS, GCP, or Azure) and with container orchestration (Kubernetes).
- Excellent communication skills — able to explain complex technical trade-offs to engineers and non-engineers alike.
- A genuine interest in AI and conversational technology, and how voice infrastructure underpins it.
- Experience instrumenting and operating voice-quality observability — call-quality metrics (e.g. MOS), per-leg call tracing, and diagnosing jitter, latency, and packet loss from telemetry.