I
Innvocon Learning Solutions
VoIP Engineer
Skills
DockerGrafanaData AnalysisAIPrometheusLinuxCRMSystem & Network Administration
10 days ago
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Important information
Contract type:
Permanent
Location:
Jaipur, Rajasthan, India
Work mode:
Onsite
Published on:
15 October 2026
What they need
About the Role
- We're seeking a skilled and proactive VoIP Engineer with at least 2 years of hands-on experience in SIP-based systems, Linux server management, and VoIP troubleshooting.
- If you've ever found beauty in a perfect SIP handshake or debugged NAT hell like a champ, youll feel right at home here.
- Youll be responsible for maintaining, optimizing, and expanding our IP telephony and unified communication infrastructureworking across Asterisk, FreeSWITCH, Kamailio, and other open-source tools.
Key Responsibilities
- Deploy, configure, and manage VoIP infrastructure (Asterisk, FreePBX, Kamailio, OpenSIPS, or similar)
- Monitor and troubleshoot call quality issues, SIP signaling, RTP media streams, and NAT/firewall behavior
- Manage Linux servers for VoIP applications service tuning, logs, security, cron jobs
- Handle integrations with CRM, call recording, SIP gateways, SBCs, and media servers
- Write and optimize dialplans, IVRs, and call routing logic using Asterisk or Lua/JSON for FreeSWITCH
- Analyze PCAPs and SIP traces using Wireshark, sngrep, Homer, or similar tools
- Work with networking and DevOps teams to ensure QoS, bandwidth, and latency optimization
- Maintain documentation for configurations, deployments, and internal processes
- Required Skills & ExperienceMinimum 2 years of VoIP experience in a production environment
- Strong knowledge of SIP, RTP, SRTP, SIP registration, and NAT traversal techniques
- Proficiency in Linux system administration (Ubuntu/RHEL) and bash scripting
- Practical experience with VoIP monitoring and debugging tools (e g , sngrep, tcpdump, Wireshark)
- Understanding of IP networking (TCP/UDP, DNS, DHCP, VLANs, routing)
- Ability to interpret SIP INVITE, 200 OK, BYE, REGISTER flows like poetry
- Good-to-Have (Not Mandatory, but Gold)Familiarity with Kamailio, OpenSIPS, or other SIP proxy/registrar solutions
- Experience with VoIP security measures (TLS, SRTP, fail2ban, SIP authentication)
- Exposure to SIPREC, TTS/STT, or AI-based call analytics
- Understanding of number masking, multi-tenant PBX, or contact center integrations
- Knowledge of monitoring stacks (Grafana/Prometheus) or VoIP-aware dashboards
- DevOps-friendly skills: Docker, Git, CI/CD pipelines
Education
- Bachelors degree in Computer Science, Information Technology, Electronics, or relevant field
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