Hamsa AI operates a cloud-native voice platform that connects telecom carriers worldwide to real-time AI voice agents. Our infrastructure spans multiple regions, integrates with enterprise carriers via secure VPN tunnels, and bridges traditional SIP telephony with modern WebRTC-based media processing — all running on Kubernetes across AWS.
Role Overview
We are seeking a skilled and detail-oriented VoIP Engineer to join our growing team. You will design, implement, maintain, and optimize our cloud-native VoIP infrastructure — SIP proxies, media relay, carrier integrations, and WebRTC bridging — to ensure high availability, performance, and security across all communication systems.
Key Responsibilities
SIP & Media Infrastructure
- Design, deploy, and manage SIP proxy infrastructure using Kamailio and/or OpenSIPS, including routing scripts, dialog handling, and SDP manipulation
- Operate and tune RTPEngine for media relay, SDP rewriting, NAT traversal, and ICE handling
- Bridge SIP telephony with WebRTC platforms (LiveKit or similar), including SIP-over-WebSocket and media interworking
Carrier & Network Integration
- Integrate with carrier SIP trunks over IPsec site-to-site VPNs (strongSwan, AWS Site-to-Site VPN) and public internet — including number manipulation, codec negotiation, and authentication
- Maintain VoIP infrastructure on AWS (EC2, VPC, VGW, Site-to-Site VPN, security groups, route tables) and Kubernetes/EKS with Docker
Operations & Reliability
- Monitor call quality and troubleshoot issues such as latency, jitter, packet loss, one-way audio, and call setup failures
- Perform SIP-level packet analysis and debugging using Wireshark, sngrep, sipgrep, and Homer SIP Capture
- Automate operational tasks and monitoring with Python and Bash
- Provide technical support, incident response, and root-cause analysis for production issues
Security & Documentation
- Ensure system security through proper firewall configuration, SBC best practices, and encryption (TLS, SRTP)
- Maintain clear documentation of SIP flows, carrier integrations, tunnel configurations, and runbooks
- Collaborate with platform, infrastructure, and AI teams to optimize scalability and reliability
Qualifications
Required
- Bachelor's degree in Computer Engineering, Telecommunications, or related field
- 3+ years of hands-on experience in VoIP engineering or telecom environments
- Strong knowledge of SIP, SDP, RTP/RTCP, TCP/IP, and networking fundamentals
- Hands-on experience with Kamailio or OpenSIPS (writing routing scripts, working with modules like drouting, dialog, rtpengine, tm, rr)
- Experience with media proxies — RTPEngine preferred (or rtpproxy/MediaProxy)
- Solid Linux administration (iptables, routing, networking, systemd)
- Experience with carrier SIP trunk integration and NAT traversal in real-world conditions
- Proficiency with SIP troubleshooting tools: Wireshark, sngrep, sipgrep, sipsak, SIPp
- Scripting skills in Python and Bash for automation
Preferred
- AWS networking: VPC, VGW, Site-to-Site VPN, Transit Gateway, security groups, route tables
- Docker and Kubernetes/EKS experience
- WebRTC and SIP <> WebRTC bridging (LiveKit, Asterisk/FreeSWITCH as B2BUA, Janus, or similar)
- IPsec/VPN experience (strongSwan, AWS managed VPN, Palo Alto, Fortigate interoperability)
- Session Border Controller (SBC) concepts and hardening
- Observability for SIP: Homer SIP Capture, CDR pipelines, Grafana/Prometheus
- Understanding of QoS, VLANs, and network security best practices
- Familiarity with Asterisk or FreeSWITCH (a plus, but not a substitute for SIP proxy experience)
- Certifications such as CCNA/CCNP Voice or equivalent
