Telephony Infrastructure Engineer

Salient•San Francisco, CA

2d•$200,000 - $300,000•Onsite

About The Position

Salient builds AI agents for regulated financial services. Our agents automate loan servicing, compliance, collections, recovery, insurance claims, and disputes for banks, captives, and specialty lenders. We are backed by a16z and Y Combinator, with $75M raised in Series A funding, cash flow positive and profitable, with mid eight figure ARR achieved in under two years. We are already serving more than 20% of the auto lending industry, processing millions of real customer calls and transactions every day, fully deployed in production with major financial institutions, and actively expanding into new financial services segments. We are building together in person in San Francisco. We are deeply integrated with our customers, own the full technology stack, and move quickly to bring modern AI into regulated industries where precision, reliability, and performance matter. We're hiring a Telephony Infrastructure Engineer to own the voice platform behind Salient's AI agents: the SIP edge, Asterisk, the voice audio and dialing layers, and the call servers that connect borrowers to our agents. You will work directly with our AI agent engineers, telephony specialist, customer success, and carriers. This is a hands-on ownership role. You will finish work in flight, make the platform reliable and safe to change, and prove it can handle the next order of magnitude of call volume. You will also design and implement customer telephony integrations and make them increasingly self-serve.

Requirements

You've run production voice infrastructure at meaningful scale and have been on call for it.
You know SIP, SDP, and RTP deeply and debug from traces and pcaps.
You can debug EC2/K8s networking issues that result in deadair—and develop monitoring and testing plans to minimize deadair.
You understand real-time media, WebRTC, and latency budgets.
You design for backpressure, graceful degradation, and safe deploys of stateful services.
You've built load tests that found real bottlenecks before customers did.
You write clear design docs, runbooks, and postmortems.
You've integrated SIP trunks with customer contact centers and carriers, including transfers and header passing.

Nice To Haves

Experience with Kamailio, Asterisk, OpenSIPS, FreeSWITCH, or rtpengine.
Experience developing regional failover for SIP
Experience with latency-aware multi-homing
Experience with contact-center, dialer, CpaaS, or carrier infrastructure.
Experience with AI voice agents, speech pipelines, or regulated industries such as lending.
Experience with contact-center platforms such as Genesys, Five9, NICE, or Amazon Connect.

Responsibilities

Finishing the Asterisk migration: cutting over remaining customers, trunks, and call flows with no regression, then retiring the legacy path.
Reliability of the voice audio and dialing layers: SLOs, monitoring, incident response, and eliminating recurring failure modes.
Zero-downtime call-server deployments: draining, graceful shutdown, safe rollout, and rollback without dropping live calls.
Building tools to manage Kamailio and Asterisk reconfiguration without server restarts inside Kubernetes
Backpressure, load shedding, and self-recovery when carriers, media, or AI dependencies degrade.
Load testing and capacity planning that validates thousands of concurrent calls and CPS at the SIP edge.
Voice observability: call-quality and platform-health metrics, alerting, runbooks, and postmortems.
Security of the voice edge against scanning, toll fraud, and abuse.
Customer telephony integrations: custom SIP trunks (e.g., Genesys, Bandwidth), call transfers, SIP header passing, and API prototypes with new contact-center platforms.
Self-serve setup: productized trunk configurations, validation tooling, and docs that shorten time to go live.