Open assignment

About the Role

Our Cloud Telephony team operates a high-scale, production-grade SIP infrastructure serving thousands of concurrent calls across multiple regions. We're looking for an experienced SIP Consultant to own the day-to-day maintenance, optimization, and evolution of this infrastructure.

This is a hands-on technical role requiring deep expertise in SIP/VoIP architecture - someone equally comfortable debugging a PJSIP registration failure, tuning Kamailio routing logic, and advising on codec trade-offs for a real-time voice AI pipeline.

Infrastructure Overview

• Two Asterisk clusters - handling inbound/outbound call termination and voice application logic
• Kamailio load balancer - fronting both clusters, managing SIP routing, failover, and dispatcher logic
• Multiple SIP providers across multiple geographic regions - requiring robust provider selection, failover, and quality monitoring
• Thousands of concurrent calls - demanding deep expertise in performance tuning, resource limits, and horizontal scaling

Responsibilities

• Maintain and optimize the Asterisk clusters and Kamailio LB configuration in a high-availability, high-concurrency environment
• Manage SIP trunk relationships with multiple providers across regions, including failover logic, routing policies, and quality monitoring
• Troubleshoot complex SIP signaling, RTP media, and call quality issues in production
• Develop and maintain PJSIP endpoint configurations, trunks, and dialplan logic (ARI/AGI/AMI)
• Tune system-level and application-level parameters to sustain target concurrency and call quality under load
• Own codec configuration and negotiation strategy (G.711, G.729, Opus, etc.) across providers and endpoints
• Collaborate with the backend engineering team on real-time voice AI integrations, audio streaming pipelines.
• Define and track SIP infrastructure KPIs (ASR, MOS) and proactively surface degradation
• Maintain infrastructure documentation: topology diagrams, runbooks, provider SLAs, and escalation paths
• Advise on architecture decisions for capacity growth, new provider onboarding, and multi-region expansion

Required Skills Experience

• 5+ years of hands-on SIP/VoIP infrastructure experience in production environments
• Asterisk - deep expertise in configuration, dialplan, clustering, and performance tuning
• Kamailio - dispatcher module, LB logic, SIP routing rules, failover scripting
• PJSIP - endpoint configuration, transport layers, registration, authentication
• ARI (Asterisk REST Interface) - programmatic call control and application integration
• High-scale SIP operations - proven experience with thousands of concurrent sessions
• SIP signaling - RFC 3261, dialog management, SDP negotiation, re-INVITEs, BYE handling, 183/180 flows
• RTP/SRTP codecs - media plane troubleshooting, codec negotiation (G.711a/u, G.729, Opus, G.722)
• Linux system administration - systemd, network stack
• VoIP monitoring tools - Homer/SipCapture, sngrep, tcpdump

Nice to Have

• Node.js - building or maintaining services that interface with Asterisk (ARI), SIP providers, or the broader telephony stack
• Git - version control for configurations, dialplans, IaC, and application code
• Infrastructure as Code (IaC) - Terraform or equivalent, for provisioning telephony infrastructure
• NATS - pub/sub messaging and real-time audio/event streaming pipelines
• Docker - containerized service deployment in cloud-native environments (Cloud Run or equivalent)
• Prometheus + Grafana - defining telephony-specific metrics, alerts, and dashboards within an existing observability stack
• Real-time audio streaming - integrating Asterisk/Kamailio with voice AI services or streaming pipelines
• Cloud Run / GCP - containerized VoIP deployments in a cloud-native environment
• CPaaS / carrier experience - Twilio, Vonage, Bandwidth, or wholesale carrier interconnects

This assignment is managed by FILL that has the exclusive end client relationship. Consultants presented to FILL through Commended will be priced at their indicated rate +4%.