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Parloa is building agentic AI for enterprise customer experience, having powered over one billion interactions for companies like Booking.com, HealthEquity, Allianz, and SAP. The company is backed by General Catalyst, EQT Ventures, and Altimeter Capital.
As Principal Software Engineer for Real-Time Voice, you will own the technical direction and architecture of Parloa's voice/VoIP infrastructure—the systems enabling AI agents to conduct natural, low-latency phone conversations at enterprise scale across global telephony providers and carriers. This is a hands-on technical leadership role where you will set architectural direction, solve complex problems directly, and raise engineering standards across the organization.
Key responsibilities include:
- Owning the architecture and technical roadmap for Voice/VoIP infrastructure, including SIP trunking, media handling, and real-time audio pipelines
- Designing for low-latency, high-availability voice at enterprise scale across global carriers
- Driving technical decisions on protocols (SIP, SDP, RTP/SRTP, RTCP, WebRTC, ICE, STUN/TURN) and real-time media transport
- Partnering with AI/ML teams to optimize integration between voice infrastructure and conversational AI (ASR, TTS, turn-taking, barge-in)
- Diagnosing and resolving systemic issues in call quality, latency, jitter, and packet loss across multi-region deployments
- Defining engineering standards for reliability, observability, and security across the voice stack
- Mentoring senior and staff engineers as a technical multiplier
- Evaluating and integrating with telephony providers, carriers, and SIP trunk vendors
- Representing Parloa's voice expertise 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 (Opus, G.711/G.722, codec negotiation/transcoding, DTMF/RFC2833, jitter buffers, packetization)
- Experience building or operating large-scale, distributed, real-time systems with strict latency and availability requirements
- 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 Go, C, C++, or Rust, with ability to work at media/transport layer
- Comfort operating in cloud environments (AWS, GCP, Azure) and container orchestration (Kubernetes)
- Excellent communication skills for explaining complex technical trade-offs to engineers and non-engineers
- Genuine interest in AI and conversational technology
- Experience instrumenting and operating voice-quality observability (call-quality metrics like MOS, per-leg call tracing, diagnosing jitter/latency/packet loss from telemetry)
