Senior SIP, Asterisk & DevOps Engineer for High-Concurrency AI Voice Infrastructure

# Senior SIP, Asterisk & DevOps Engineer for High-Concurrency AI Voice Infrastructure ## About Voicing AI Voicing AI builds enterprise-grade AI Voice Agents that handle real-time inbound and outbound phone conversations at scale. Our platform operates across complex telephony environments and integrates with SIP trunks, carriers, contact center platforms, cloud infrastructure, and enterprise customer environments. We manage high-concurrency voice workloads where telephony reliability, call quality, latency, scalability, and infrastructure performance are critical. We are looking for a highly experienced **SIP, Asterisk, and DevOps Engineer** who can own telephony integrations and infrastructure optimization end-to-end. This is not a basic Asterisk administration role. We need someone who deeply understands SIP signaling, RTP/media flows, carrier integrations, Asterisk performance, network troubleshooting, and high-scale production infrastructure. --- ## What You Will Work On ### SIP & Telephony Engineering You will be responsible for designing, implementing, troubleshooting, and optimizing SIP-based telephony integrations, including: * Deploying and configuring Asterisk SIP servers in cloud environments * Integrating SIP trunks with telecom carriers and enterprise telephony systems * Troubleshooting SIP signaling and RTP/media issues * Analyzing call failures, disconnects, jitter, packet loss, latency, one-way audio, and call quality issues * Optimizing Asterisk for high-concurrency call workloads * Designing reliable telephony architectures for large-scale AI voice agent deployments * Configuring and troubleshooting SIP over UDP, TCP, and TLS * Managing secure SIP and SRTP configurations * Analyzing SIP traces and packet captures using tools such as: * sngrep * Wireshark * tcpdump * Asterisk CLI and logs * Troubleshooting SIP response codes and complex call flows * Working with NAT, firewalls, SBCs, private networks, VPNs, and IP whitelisting * Managing DTMF configurations and troubleshooting RFC2833, SIP INFO, and in-band DTMF * Analyzing codec negotiation and media compatibility * Understanding SIP registration-based and IP-authenticated trunking * Designing failover, redundancy, and high-availability strategies for telephony infrastructure --- ## Asterisk Expertise Required The ideal engineer should have strong hands-on experience with: * Asterisk * PJSIP * SIP trunk configuration * Dialplans * RTP configuration * Call routing * SIP headers * Codec negotiation * DTMF handling * TLS certificates and secure SIP * Asterisk performance tuning * Concurrent call capacity planning * Linux system tuning for high call volumes * Asterisk process monitoring * CPU, memory, file descriptor, socket, and network optimization * Troubleshooting production telephony incidents Experience operating Asterisk with hundreds or thousands of concurrent calls is highly preferred. --- ## High-Concurrency Telephony & Performance Engineering Our infrastructure handles real-time voice workloads at scale. You should be comfortable working on: * High concurrent call architectures * Capacity planning * Load testing * SIP server benchmarking * Horizontal scaling strategies * Traffic distribution across multiple SIP servers * Active-active and active-passive architectures * SIP load balancing * Failure recovery * Bottleneck identification * Network bandwidth analysis * RTP performance optimization * Linux kernel and network tuning * Connection and socket limits * Monitoring telephony infrastructure under peak load You should be able to investigate questions such as: * Why is jitter occurring only under high concurrency? * Is the issue caused by Asterisk, the SIP carrier, network bandwidth, RTP handling, or application infrastructure? * How many concurrent calls can a specific server safely handle? * Should a workload be vertically scaled or distributed across multiple SIP servers? * Why are calls dropping after a specific duration? * Why is one-way audio occurring? * Why are SIP messages being retransmitted? * Why are we seeing 4xx, 5xx, or 6xx SIP errors? * Why is call quality degrading during traffic spikes? --- ## Enterprise Telephony Integration Experience Experience integrating with one or more of the following is strongly preferred: * Genesys * Genesys Cloud * Avaya * Amazon Connect * Cisco Unified Communications * Twilio * 3CX * RingCentral * Contact center platforms * Session Border Controllers * Enterprise PBX systems * Telecom carriers and SIP providers You should understand enterprise SIP trunking architectures and be able to collaborate with customer networking, telecom, infrastructure, and security teams. --- ## DevOps & Cloud Infrastructure Responsibilities This role also requires strong DevOps experience. You will work with our engineering team on infrastructure supporting high-scale real-time AI voice workloads. Required experience includes: * Kubernetes * Docker * Linux * AWS, Azure, or GCP * Infrastructure monitoring * Autoscaling * Load balancing * High availability * Production incident troubleshooting * Networking * DNS * TLS certificates * Firewalls * Security groups * VPC/VNet networking * Logging and observability Strong experience with the following is preferred: * Kubernetes HPA and autoscaling * Helm * Terraform * CI/CD * Prometheus * Grafana * CloudWatch * ELK/OpenSearch * Loki * Infrastructure as Code * Kubernetes networking * Ingress controllers * Service discovery * Pod scaling and resource optimization --- ## What We Are Looking For We are looking for someone who: * Has deep hands-on experience with SIP and Asterisk * Can independently debug complex telephony incidents * Understands both SIP signaling and RTP/media * Has operated production telephony infrastructure * Understands high-concurrency systems * Has strong Linux and networking fundamentals * Has strong DevOps and cloud infrastructure experience * Can analyze problems using logs, packet captures, metrics, and system-level diagnostics * Can work directly with telecom carriers and enterprise customer teams * Can take ownership of production issues until root cause is identified * Thinks systematically about performance, reliability, scalability, and observability --- ## Must-Have Skills * SIP * Asterisk * PJSIP * RTP * SIP Trunking * Linux * Networking * Wireshark * sngrep * tcpdump * UDP, TCP, and TLS * VoIP troubleshooting * High-concurrency infrastructure * Kubernetes * Docker * Cloud infrastructure * DevOps * Production troubleshooting --- ## Preferred Skills * Genesys or Genesys Cloud * Avaya * Amazon Connect * SBC experience * Kamailio * OpenSIPS * FreeSWITCH * SIP load balancing * Terraform * Helm * Prometheus and Grafana * AWS * Azure * GCP * Telecom carrier integrations * Enterprise contact center integrations --- ## Example Projects You May Work On * Deploying a production Asterisk SIP server on AWS or Azure * Integrating Voicing AI with a customer's enterprise SIP trunk * Troubleshooting jitter affecting production calls * Investigating intermittent SIP 5xx errors * Optimizing an Asterisk server for hundreds of concurrent calls * Designing a multi-server telephony architecture for high availability * Troubleshooting one-way audio or RTP routing issues * Setting up SIP over TLS * Integrating with Genesys, Avaya, or another enterprise contact center * Building monitoring and alerting for telephony infrastructure * Performing load testing and determining safe concurrent call capacity * Scaling Kubernetes workloads supporting real-time voice traffic * Performing root cause analysis for production telephony incidents --- ## Screening Questions Please answer the following when applying: 1. How many years of hands-on experience do you have with SIP and Asterisk? 2. What is the highest number of concurrent calls you have supported in a production environment? 3. Describe the most complex SIP or Asterisk production issue you have personally diagnosed and resolved. 4. Which tools do you normally use to troubleshoot SIP signaling and RTP/media issues? 5. Have you worked with SIP over TLS? Please briefly describe your experience. 6. Have you integrated Asterisk or another SIP platform with Genesys, Avaya, Amazon Connect, Cisco, Twilio, or similar enterprise telephony platforms? Please provide examples. 7. What experience do you have with Kubernetes and cloud infrastructure? 8. Have you designed or operated highly available or horizontally scalable telephony infrastructure? 9. Are you comfortable participating in production incident troubleshooting and working directly with customers, carriers, and enterprise infrastructure teams? 10. Please explain the difference between a SIP signaling issue and an RTP/media issue, and how you would approach troubleshooting each. --- ## Important Please do not apply if your experience is primarily limited to basic PBX setup, extension configuration, or small office VoIP systems. We are specifically looking for an engineer with strong experience in production-grade SIP infrastructure, Asterisk, troubleshooting, performance optimization, high-concurrency telephony, and DevOps. This can become a long-term engagement for the right engineer.