Name: Jobpe
Address: T-Hub, Plot No 1/C, Sy No 83/1, Raidurgam panmaktha, Knowledge City Rd, Hyderabad, Telangana, 500081, IN
Telephone: +91-83339-09630
Price range: Free

FreeSWITCH Developer (with OpenSIPS/Kamailio Experience) RioPath Technologies

3.0 - 7.0 years

0 Lacs

delhi

On-site

As a skilled FreeSWITCH Developer with experience in OpenSIPS and/or Kamailio, you will join our core VoIP engineering team. In this hands-on role, you will be responsible for architecting, building, and maintaining robust SIP-based communication systems. If you are excited about solving complex signaling challenges and scaling VoIP infrastructure, this is the right opportunity for you. Your responsibilities will include building and maintaining scalable VoIP services using FreeSWITCH, configuring and managing SIP proxies (OpenSIPS, Kamailio) for routing, load balancing, and NAT traversal. You will also develop and fine-tune dialplans, call flows, and scripts in Lua, Python, or JavaScript. Troubleshooting signaling/media issues using tools like Wireshark, sngrep, and FreeSWITCH logs will be a part of your routine. Additionally, you will integrate VoIP systems with external APIs, databases, CRMs, and billing platforms, ensuring security, uptime, and scalability across the voice infrastructure. Documenting systems and contributing to architectural decisions will also be crucial aspects of your role. To be successful in this role, you should have at least 3 years of experience working with FreeSWITCH in production environments, a strong grasp of the SIP protocol and related RFCs, and practical experience with OpenSIPS and/or Kamailio configuration. Deep troubleshooting skills using tools like Wireshark, sngrep, and HOMER/HEPIC, as well as scripting experience (Lua, Bash, Python, JS) are required. A solid understanding of VoIP architecture, NAT traversal, and RTP handling, along with Linux systems admin experience (Ubuntu, Debian, CentOS), are also essential. Bonus points if you have experience integrating WebRTC, knowledge of rtpproxy, rtpengine, monitoring with Grafana, Prometheus, or HEPIC, working in containerized environments using Docker/Kubernetes, or experience in billing or CRM system integration. In return, you will receive a competitive salary, flexible hours with remote work options, the opportunity to have a direct impact on production systems and architecture, a chance to work with advanced VoIP technologies, and the support of smart teammates who are passionate about what they do. If you are passionate about VoIP and building high-performance SIP systems, we encourage you to apply by sending your resume and a brief cover letter to info@riopath.com.,

Posted 2 weeks ago

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VoIP Engineer Innvocon Learning Solutions

2.0 - 4.0 years

6 - 8 Lacs

Jaipur

Work from Office

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 & Experience Minimum 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.

Posted 1 month ago

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