SIP Engineer

Weave•Lehi, UT

3d•Hybrid

About The Position

Weave is looking for a VoIP Engineer with a strong background in modern communication tools and cloud technologies to help maintain and optimize our distributed, cloud-based voice platform. In this role, you’ll ensure our communication systems are robust, efficient, scalable, and observable—working hands-on with SIP/RTP infrastructure and core components like Kamailio, RTPengine, FreeSWITCH, and Homer/HEPIC. You’ll collaborate cross-functionally with engineering and operations teams to improve reliability, performance, and customer call quality while shipping improvements through DevOps best practices. This position can be remote, in office, or hybrid Reports to: Sean Landis

Requirements

Previous experience working as a VoIP network engineer
Experience with VoIP applications such as Kamailio, OpenSIPS, FreeSWITCH, RTPengine, and Asterisk
Experience with troubleshooting tools such as Wireshark, sngrep, and Homer
Deep understanding of TCP/IP networks, firewalls, and related protocols (SIP, RTP, RTCP, TLS, ICE, STUN, TURN, WebRTC)
Programming experience with a major language such as Go, Java, or C
Experience with scripting languages such as Python, Bash, and LUA
Experience with DevOps tools for cloud infrastructure management and automation such as Terraform and Ansible
Flexibility and adaptability in learning new technologies and tools
Strong written, oral, and interpersonal communication skills
Highly self-motivated and directed
Keen attention to detail
Experience operating production services in a major cloud environment (Google Cloud preferred), including networking concepts such as VPCs, load balancing, service discovery, and IAM/security fundamentals
Comfort participating in on-call rotations and driving incident remediation through postmortems and follow-up engineering work

Nice To Haves

5+ years of experience designing and supporting large scale VoIP networks
Certifications in Kamailio or OpenSIPS
Hands-on experience with Kubernetes operations (deployments, services/ingress, Helm/Kustomize, autoscaling, and debugging)
Experience designing or improving call-quality tooling (MOS/RTCP analytics, jitter/loss monitoring, SBC/edge routing strategies, or capacity planning)

Responsibilities

Grow and manage a distributed cloud-based VoIP platform
Optimize and troubleshoot FreeSWITCH and Kamailio configurations for performance, resiliency, and reliability
Design, build, test, deploy, and maintain services that support the voice platform
Monitor and analyze VoIP network performance, identify areas for improvement, and implement solutions to enhance quality and efficiency
Troubleshoot and resolve VoIP-related issues including call quality problems, signaling/connectivity issues, and system outages
Develop Golang-based microservices and containers that support the voice ecosystem
Deploy and operate services in Kubernetes environments
Maintain and configure Kamailio and FreeSWITCH instances
Implement and improve observability for SIP/RTP flows using HEP/Homer and telemetry (metrics/logs/traces) to accelerate incident response and root-cause analysis
Partner with SRE/Platform teams to improve CI/CD, infrastructure-as-code, and reliability practices (e.g., automated rollouts, canarying, and repeatable environments)