6 Opensips Jobs

Sr. Software Engr–VoIP (3+ yrs), SIP, RTP, Asterisk/Freeswitch, Kamailio, WebRTC, AWS, Golang, REST APIs, DevOps, MySQL, Linux. Prod-grade VoIP dev exp a must. C2H via TE Infotech (Exotel), Convertible to Permanent, Loc:BLR @ ssankala@toppersedge.com

We are seeking a skilled and dedicated FreeSWITCH Engineer with hands-on experience in VoIP systems to join our team. As a FreeSWITCH Engineer, you will be instrumental in the development, configuration, and maintenance of scalable and reliable FreeSWITCH-based voice infrastructures. Your responsibilities will include designing, deploying, and maintaining FreeSWITCH servers and related VoIP infrastructure. You will troubleshoot and resolve FreeSWITCH and VoIP-related issues, develop custom dial plans, modules, and call routing logic, and work with SIP, RTP, and related VoIP protocols. Monitoring system performance, ensuring high availability, collaborating with development, network, and support teams, and documenting configurations and system changes will also be part of your role. To be successful in this position, you should have a minimum of 2 years of hands-on experience with FreeSWITCH in a production environment, a strong understanding of VoIP technologies and SIP protocol, experience with Linux system administration, and familiarity with scripting languages such as Bash, Python, and Lua. The ability to work independently in a remote setup, strong problem-solving and analytical skills are also essential. Preferred skills include experience with other VoIP platforms like Asterisk, Kamailio, OpenSIPS, knowledge of WebRTC, RTP engines, or media servers, exposure to monitoring tools like Grafana and Prometheus, familiarity with APIs and backend integration. Join us for a collaborative and supportive team environment where you will have the opportunity to work on innovative VoIP solutions at scale.,

You should have a minimum of 6 months to 5 years of experience in the field. Your qualifications must include a good understanding of VoIP and SIP, a strong grasp of Linux OS (CentOS/Redhat), and practical experience with at least one open-source VoIP solution such as Freeswitch, Asterisk, OpenSIPS, or Kamailio. Proficiency in troubleshooting using tools like Wireshark, tcpdump, sipp, fail2ban, and IPtables is essential. It is important to have a solid grasp of networking concepts and good troubleshooting skills. In addition, knowledge of MySQL Database and experience with Database Replication & Heartbeat configuration would be beneficial. Your communication skills should be excellent, and you must possess a problem-solving attitude. Previous experience in client support is preferred, and you should be willing to work in shifts.,

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.

Are you passionate about cloud telephony, VoIP systems, and building next-gen PBX solutions? Were looking for a skilled FreeSWITCH Developer to help us develop a robust and scalable PBX panel for enterprise-level deployments. Key Responsibilities: • Design, develop, and maintain tools supporting a scalable cloud telephony infrastructure. • Customize FreeSWITCH for audio/video conferencing, capable of 1000 to 1500 concurrent calls. • Deep understanding of SIP, RTP, RTCP, NAT traversal (TURN/STUN), and TLS encryption. • Build and deploy multiple FreeSWITCH instances behind load balancers for high availability. • Analyze media stream issues using tools like Wireshark and debug SIP signaling errors. • Work closely with frontend/mobile teams to develop API-driven PBX solutions. • Integrate codecs like PCMU, PCMA, G729, Opus and support SDP offer/answer model. • Collaborate on RTP Proxy, routed audio conferencing, and NAT traversal setups. • Familiarity with SIP proxy servers (e.g. Kamailio/OpenSER) and Session Border Controllers (SBCs). • Bonus: Knowledge of H.323, SIPX, WebRTC, and SMPP. You Should Have: • Strong hands-on experience with FreeSWITCH or other open-source telephony platforms. • Deep understanding of telecom signaling protocols and VoIP architecture. • Practical experience developing or supporting PBX platforms. • Experience with SIP debugging and high-concurrency media handling.

Position Summary : The candidate suitable for this role of Senior Software Engineer will be responsible for leading the development and implementation of complex software solutions. This role involves a high level of technical expertise and the ability to guide and mentor junior team members. The Senior Software Engineer will collaborate with cross-functional teams to define, design, and ship new features while maintaining high standards of software quality. Key Responsibilities : - Design and develop high-volume, low-latency applications for mission-critical systems, delivering high availability and performance. - Contribute to all phases of the development lifecycle, from concept and design to testing. - Write well-designed, testable, and efficient code. - Ensure designs comply with specifications. - Prepare and produce releases of software components. - Support continuous improvement by investigating alternatives and technologies and presenting these for architectural review. Skills : - 4-6 years of experience in software development. - Solid background in GoLang. - Strong data structures and algorithms concepts. - Designing and problem-solving skills, with a strong bias for architecting for performance and scalability. - Sound knowledge of cloud services and Kubernetes. Good to have Skills : - Good Understanding SIP/RTP protocols - Hands-on experience with any of FreeSWITCH/Asterisk/OpenSIPS/Kamailio open source VoIP softwares. Qualifications : - B. Tech/M. Tech/MCA in Computer Science Benefits : - Flexible Working Hours. - Hybrid Working Style. - Personal Accidental Insurance. - Health Insurance to Self, Spouse and two kids. - 5 days working week.