VoIP Engineer

Department Icon IT / Software Development & Related
Posted: 11 months ago
2-4 years
Jaipur
IT / Software Development & Related
work from office
Full Time

Posted: 11 months ago
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Posted over 15 days ago, so the role may have closed. Confirm on the employer's page when you apply, or see similar jobs.

Job Description

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

Skills

VoIPDevOpsVoIP Applications ServiceFreeSWITCHSIPSRTPRTPApplicationMediaOptimization

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