60 Kamailio Jobs

Junior System Administrator (VoIP/Telephony) / Telephony Engineer (Telecommunications) No. of Positions - 1 We are looking for a Junior System Administrator (VoIP) / Telephony Engineer (Telecommunications) to join our Telebu's Communications engineering Team. The Telebuin will develop, implement and support IP Telephony related technologies including and not limited to IP Telephony, IVR platforms, Conferencing solutions, Voice engineering integration, Voice over IP (VoIP), Session Border Controllers (SBC), Session Initiation Protocol (SIP), WebRTC, and Public Switched Telephone Network (PSTN) gateways. Responsibilities Install & maintain Freeswtich and other SIP servers. Administration of SIP and Media Servers, Network/Protocol level debugging and testing, Contact center solutions, Troubleshoots and resolves complex problems. Provide IP Telephony and VoIP Subject Matter Expertise for Company and Company's managed service providers, manages 3rd party telecom carriers and providers. Requirements 3 years of hands-on industry experience in telecommunications Strong conceptualize knowledge and experience with telephony protocols like SIP, SDP, RTP, SRTP and audio/video codecs. In-depth working experience with Kamailio, Freeswitch, Any of the SIP stack (Sofia, reSIProcate, PJSIP, etc.), and Linux Experience in using the VoIP testing tools like Wireshark, VoIPMonitor, SIPp, SIPCapture, Homer etc. Strong understanding of implementing various network setups (Private VPNs, multi-zone secure connectivity etc) Nice To Have Experience with virtualization/container related technologies (Xen, VMware vSphere / ESXi, Docker, Kubernetes) Hands on writing production quality code using any of the scripting languages like Python, Go, Erlang etc. Working knowledge in any NoSQL databases like MongoDB, Redis, Cassandra etc. Passionate about knowing everything about VoIP Protocol standards & related RFCs Skills:- FreeSWITCH, Voice Over IP (VoIP) and Telephony

Join Vonage and help us innovate cloud communications for businesses worldwide! Vonage has built its successful global Support teams on individuals with technical savviness, superior customer relationship skills, and a passion for learning. We challenge our Support Engineers to provide a customer experience that leaves our users impressed, loyal and true advocates of our company. As a Senior Support Engineer, you will provide first-class technical support to our rapidly growing strategic customer base, who rely on our real-time communication APIs and SDKs. You will be responsible for driving and managing customer-related projects, initiatives and tasks for our strategic accounts, collaborating heavily with Sales, Engineering, and the rest of the Vonage organization. What will you do Investigate, troubleshoot, diagnose and resolve technical issues related to customer API and SDK implementations Communicate effectively (both verbal and written) with our customers and internal stakeholders Be a problem solver, have a natural curiosity, and demonstrate the ability to learn rapidly Contribute to internal and external knowledge bases Collaborate with your team to identify bugs and escalate to Product/Engineering teams Communicate well with different audiences (developers, technical and non-technical users) What You Must Have 3+ years as a Support Engineer in the telecommunications or SaaS sectors Messaging technologies: SMPP, GSM, SMS Strong knowledge of RESTful APIs and the ability to understand and troubleshoot issues with cloud solutions English and Japanese language proficiency Any Of The Following Is a Plus Experience with Voice technologies: SIP, VoiceXML, CCXML, WebRTC Supporting APIs or SDKs Excellent understanding of networking: TCP/IP, UDP, most common protocols Voice software: Asterisk, Freeswitch, Kamailio, Voxeo Prophecy Theres no perfect candidate. You don&apost need all the preferred qualifications to make a valuable impact on our team. Our employees and customers come from diverse backgrounds, so if you&aposre passionate about what you could achieve at Vonage, we&aposd love to hear from you. Who We Are Vonage is a global cloud communications leader. And your talent will further help brands - such as Airbnb, Viber, WhatsApp, and Snapchat - accelerate their digital transformation through our fully programmable-based unified communications, contact center solutions, and communications APIs. Ready to innovate Then join us today. Note: The purpose of this profile is to provide a general summary of essential responsibilities for the position and is not meant as an exhaustive list. Assignments may differ for individuals within the same role based on business conditions, departmental need or geographic location. Show more Show less

Role Description We’re looking for a Senior Frontend Developer who loves to tackle challenging problems with a firm grasp on browser technologies having more than 5 years of experience. You will take on a central role in developing our products using ReactJS, Ant Design, and other libraries with input from product management. Our teams are spread across several locations & serve customers in the US, Europe, and India (Pune, Bangalore, & NCR). Our team is at the forefront of technology, and loves working with others via Meetups and Hackathons. We are one of a couple of hundred companies who applied for the TiE Pune Nurture Accelerator Program for 2019/20 and 1 of 12 that actually graduated. We were also 1 of 4 accepted companies out of 170 that applied, for the 2021 Brigade REAP Accelerator Program. Our Technologies Include Python ElasticSearch ReactJS React Native / Flutter / iOS / Android Apache Cassandra VoIP and related technologies (Freeswitch, Kamailio, etc) Docker/K8/Puppet AWS/GCP/Azure Responsibilities Develop user interface components that are robust and easy to maintain Build, test, document, and deploy at scale Implement and integrate RESTful APIs in ReactJS Work in a team-oriented environment, providing software development technical expertise and guidance to key stakeholders on variety of enterprise-scale applications and projects Provide technical direction and guidance, as well as draft specifications, architect solutions, define timelines, advise on industry best practices and problems to be solved Work closely with Customers, Product Managers, and Architects to develop effective, high-quality enterprise software solutions Understand and apply a variety of project life-cycles, methods, and software development techniques Write code and review other people’s code. Ensure the technical feasibility of UI/UX designs. Optimize application for maximum speed and scalability. About You 5+ years of overall software development experience Proficient understanding of modern web tech stack including HTML, Less, JQuery, and ES6. Strong proficiency in JavaScript, including DOM manipulation and the JavaScript object model Good understanding of React.js and its core principles Experience with popular React.js workflows (such as Flux or Redux) Familiarity with integrating RESTful APIs and browser nuances Experience with front-end development tools such as Babel, Webpack, NPM, Yarn Attention to detail and a strong sense of ownership. The mindset to take up project individually and meet the deadline BS/MS in Computer Science or related stream is a must Bonus: Experience with unit testing using jest or react-testing- library. Perks A great team culture Challenging work environment Open door policy Liberal work from home Conference and training support Amazing referral program PF & Health Insurance Team outings (Regular & Annual offsite) About Us We Are Engineers. We Are Innovators. We Are Creators. Inspired by real problems, driving real results, MetroGuild, a global B2B SaaS company, developed MetroLeads – marketing, sales, and communications management platform. Rooted in the science of selling, MetroGuild evolved to offer a range of products and services to your Sales team. MetroGuild empowers organizations globally to own and grow their Marketing and Sales Teams and drive growth. MetroGuild provides CRM, digital asset building, and support to help organizations reach their true growth potential. Desired Position * Applicant Name * Email Address * Phone Number Qualification * Associate DegreeBachelor's DegreeCollegePostgraduateOther Resume * The file can be in PDF/TXT format.(upload limit upto 6MB) Remarks Fields with * are required. Be assured that your information will not be sold or distributed and will only be used to respond to your query. Thanks for your interest! Δ

Subject: Walk-in Opportunity – NOC Engineer (Fresher) | 28th July 2025 | Kolkata Dear Candidate, We’re excited to share a fantastic career opportunity for freshers or entry-level candidates looking to launch their careers in telecom and networking . We are currently hiring for the position of NOC Engineer at our New Town, Kolkata location. This role is ideal for recent graduates or candidates who have academic exposure to VoIP, Networking, Linux , or related technologies and are eager to gain real-time industry experience. Position Details: Job Title: NOC Engineer (Fresher / Entry-Level) Location: New Town, Kolkata (Chinar Park – Near Akanksha More) Experience: 0 to 1 year Industry: Telecom Services Department: Network Operations Center (NOC) Job Type: Full-time Working Days: Monday to Saturday Work Hours: 9:30 AM – 6:30 PM | Rotational Shift About the Role: As a NOC Engineer, you will work closely with experienced telecom professionals, gaining hands-on experience in VoIP infrastructure, Asterisk systems, and network operations . This is a training-intensive role , ideal for candidates eager to learn and grow in a real-time telecom environment. Key Responsibilities: Monitor VoIP traffic and overall network health Assist in troubleshooting and incident resolution Support configuration and maintenance of PBX systems Document incidents and participate in upgrade tasks Collaborate with senior engineers on live technical issues Required Skills (Academic or Training Exposure): Basic understanding of VoIP and SIP protocols Familiarity with Linux systems (Ubuntu/CentOS) Basic networking concepts: TCP/IP, NAT, Firewalls Exposure to scripting languages (Bash or Python) Project-level understanding of MySQL or MongoDB Interest in learning Asterisk, IPPBX, and EPABX systems Preferred / Bonus Skills: Academic project or certification in VoIP or Asterisk Exposure to OpenSIPS/Kamailio or SIP proxies Familiarity with monitoring tools (Nagios, Zabbix) Understanding of telecom billing and call routing Soft Skills: Eagerness to learn and upskill in telecom/VoIP Strong communication and documentation skills Proactive, analytical, and team-oriented mindset Walk-In Interview Details: Date: Monday, 28th July 2025 Time: 12:00 PM – 2:00 PM Reporting Time: 11:45 AM Venue: ProHR Strategies Private Limited PS Abacus Building, Unit No. 329, 3rd Floor New Town, Chinar Park, Kolkata – 700157 Landmark: Near Akanksha More Bus Stop Contact Person: Sujoy What to Bring: Updated resume (hard copy) Valid government-issued photo ID (for verification) Business formal attire is mandatory If you're passionate about telecom technologies and ready to kickstart your career, we encourage you to attend the walk-in interview. To confirm your participation, please share your updated CV at: anindita.goswami@prohrstrategies.com We look forward to meeting you in person! Best regards, HR Team ProHR Strategies Private Limited Job Types: Full-time, Fresher Pay: Up to ₹15,000.00 per month Benefits: Provident Fund Work Location: In person

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 will be part of Telebu's Communications engineering team as a Senior System Administrator (VoIP) / Telephony Engineer (Telecommunications). Your primary responsibility will be to develop, implement, and support IP Telephony related technologies such as IP Telephony, IVR platforms, Conferencing solutions, Voice engineering integration, Voice over IP (VoIP), Session Border Controllers (SBC), Session Initiation Protocol (SIP), WebRTC, and Public Switched Telephone Network (PSTN) gateways. Your key responsibilities will include developing and implementing telephony networks with various components like SIP proxies, registrar, media-servers, billing systems, and deploying SIP VOIP/PRI trunking solutions that are highly scalable, robust, high-availability (HA), and fault-tolerant. You will also be responsible for the administration of SIP and Media Servers, network/protocol level debugging and testing, contact center solutions, and troubleshooting and resolving complex problems. Additionally, you will provide IP Telephony and VoIP Subject Matter Expertise for the company and its managed service providers, as well as manage 3rd party telecom carriers and providers. To be successful in this role, you should have at least 5 years of hands-on industry experience in telecommunications. You should possess strong conceptual knowledge and experience with telephony protocols like SIP, SDP, RTP, SRTP, WebRTC, and audio/video codecs. In-depth working experience with Kamailio, Freeswitch, any of the SIP stack (Sofia, reSIProcate, PJSIP, etc.), ICE Framework (STUN/TURN), and Linux is required. Hands-on experience in writing production quality code using scripting languages like Python, Go, Erlang, etc., is essential. Experience in using VoIP testing tools like Wireshark, VoIPMonitor, SIPp, SIPCapture, Homer, etc., will be beneficial. Nice to have skills include working knowledge of NoSQL databases like MongoDB, Redis, Cassandra, a passion for knowing everything about VoIP Protocol standards & related RFCs, and experience with virtualization/container-related technologies such as Xen, VMware vSphere/ESXi, Docker, Kubernetes.,

About the Role: We are looking for a skilled and dedicated FreeSWITCH Engineer with hands-on experience in VoIP systems. You will play a key role in developing, configuring, and maintaining scalable and reliable FreeSWITCH-based voice infrastructures. Key Responsibilities: • Design, deploy, and maintain FreeSWITCH servers and related VoIP infrastructure. • Troubleshoot and resolve FreeSWITCH and VoIP-related issues. • Develop custom dial plans, modules, and call routing logic. • Work with SIP, RTP, and related VoIP protocols. • Monitor system performance and ensure high availability. • Collaborate with development, network, and support teams to optimize voice systems. • Document configurations, workflows, and system changes. Requirements: • Minimum 2 years of hands-on experience with FreeSWITCH in a production environment. • Strong understanding of VoIP technologies and SIP protocol. • Experience with Linux system administration. • Familiarity with scripting languages (e.g., Bash, Python, Lua). • Ability to work independently in a remote setup. • Strong problem-solving and analytical skills. Preferred Skills: • Experience with other VoIP platforms (e.g., Asterisk, Kamailio, OpenSIPS). • Knowledge of WebRTC, RTP engines, or media servers. • Exposure to monitoring tools (Grafana, Prometheus, etc.). • Familiarity with APIs and backend integration. Why Join Us? • Collaborative and supportive team environment • Opportunity to work on innovative VoIP solutions at scale

Job Summary Looking for a tech-savvy engineer with 2–3 years’ experience in VoIP, Linux server administration, web server management, and MySQL. You’ll support and optimize our telecom infrastructure using Kamailio, Freeswitch, and related tools. Responsibilities Configure and maintain Kamailio and Freeswitch for SIP routing and telephony. Manage Linux servers (Ubuntu, CentOS): setup, patching, scripting. Handle NGINX/Apache web servers and secure with SSL. Support MySQL databases: tuning, backups, indexing. Collaborate with dev/network teams for integration and support. Requirements 2–3 years in Kamailio/Freeswitch setup and Linux administration. Hands-on with web servers and MySQL optimization. Knowledge of SIP/RTP, Wireshark, basic scripting (Shell/Python). Experience with monitoring tools (Nagios/Prometheus).

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.

Job Description We are looking for a skilled and passionate FreeSWITCH & Kamailio Developer to join our on-site team in Noida Extension . The ideal candidate will help us build a scalable, secure, and high-performance PBX platform tailored for enterprise-level VoIP deployments. You will be responsible for designing, customizing, and maintaining cloud telephony systems, optimizing call flows, and handling SIP signaling at scale. This role is perfect for someone with deep technical knowledge of VoIP architecture and a hands-on approach to real-time communications. Responsibilities Design and maintain scalable cloud telephony infrastructure Customize FreeSWITCH for audio/video conferencing (4000–5000 concurrent calls) Deploy FreeSWITCH behind load balancers for high availability Debug SIP signaling and analyze RTP/media streams (Wireshark, sngrep) Integrate codecs (PCMU, PCMA, G729, Opus) and support SDP offer/answer models Develop API-integrated PBX systems in coordination with frontend/backend teams Configure RTP Proxy and handle NAT traversal (TURN/STUN) Set up and manage SIP proxy servers (Kamailio/OpenSIPS) Work on SBCs to ensure secure and reliable call routing Bonus: Experience with WebRTC, SIPX, SMPP, and H.323 Required Skills Hands-on experience with FreeSWITCH, Asterisk, or similar VoIP platforms Strong knowledge of SIP, RTP, RTCP, NAT traversal, and TLS Experience in Kamailio/OpenSIPS configuration and routing logic Proficiency in Linux system administration and shell scripting Familiarity with tools like sngrep, tcpdump, and Wireshark Ability to debug and optimize VoIP call flows and media streams Excellent troubleshooting and problem-solving skills Bachelor's degree in Computer Science, IT, or related field Experience in telecom or VoIP support environment is a plus Job Types: Full-time, Permanent Pay: ₹13,650.56 - ₹100,000.00 per month Benefits: Commuter assistance Flexible schedule Ability to commute/relocate: Noida, Uttar Pradesh: Reliably commute or planning to relocate before starting work (Preferred) Work Location: In person

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.

Job Description We are looking for a skilled and passionate FreeSWITCH & Kamailio Developer to join our on-site team in Noida Extension . The ideal candidate will help us build a scalable, secure, and high-performance PBX platform tailored for enterprise-level VoIP deployments. You will be responsible for designing, customizing, and maintaining cloud telephony systems, optimizing call flows, and handling SIP signaling at scale. This role is perfect for someone with deep technical knowledge of VoIP architecture and a hands-on approach to real-time communications. Responsibilities Design and maintain scalable cloud telephony infrastructure Customize FreeSWITCH for audio/video conferencing (4000–5000 concurrent calls) Deploy FreeSWITCH behind load balancers for high availability Debug SIP signaling and analyze RTP/media streams (Wireshark, sngrep) Integrate codecs (PCMU, PCMA, G729, Opus) and support SDP offer/answer models Develop API-integrated PBX systems in coordination with frontend/backend teams Configure RTP Proxy and handle NAT traversal (TURN/STUN) Set up and manage SIP proxy servers (Kamailio/OpenSIPS) Work on SBCs to ensure secure and reliable call routing Bonus: Experience with WebRTC, SIPX, SMPP, and H.323 Required Skills Hands-on experience with FreeSWITCH, Asterisk, or similar VoIP platforms Strong knowledge of SIP, RTP, RTCP, NAT traversal, and TLS Experience in Kamailio/OpenSIPS configuration and routing logic Proficiency in Linux system administration and shell scripting Familiarity with tools like sngrep, tcpdump, and Wireshark Ability to debug and optimize VoIP call flows and media streams Excellent troubleshooting and problem-solving skills Bachelor's degree in Computer Science, IT, or related field Experience in telecom or VoIP support environment is a plus Job Types: Full-time, Permanent Pay: ₹13,650.56 - ₹100,000.00 per month Benefits: Commuter assistance Flexible schedule Ability to commute/relocate: Noida, Uttar Pradesh: Reliably commute or planning to relocate before starting work (Preferred) Work Location: In person

Design, develop & maintain scalable FreeSWITCH-based cloud telephony. Handle SIP, RTP, NAT, TLS, 5000+ calls, conferencing, SBCs, Kamailio, APIs. Debug via Wireshark. Integrate codecs, build HA clusters. Bonus: WebRTC, SMPP, H.323, SIPX knowledge.

We are a fun-loving, energetic and fast growing company that breathes innovation. We strive to give an unparalleled experience to our customers and win them for life. One in every 24 people on this planet is served by Airtel. Here, we put our customers at the heart of everything we do. We encourage our people to push boundaries and evolve from skilled professionals of today to risk-taking entrepreneurs of tomorrow. We hire people from every realm and offer them opportunities that encourage individual and professional growth. We are always looking for people who are thinkers & doers; people with passion, curiosity & conviction; people who are eager to break away from conventional roles and do 'jobs never done before. Job Summary: Seeking an experienced VoIP Engineer with 2+ years of expertise in open-source technologies like Asterisk, Kamailio, and RTPengine. Responsible for designing, implementing, and maintaining VoIPsystems for optimal performance and reliability. Collaborate with cross-functional teams to deliver high-quality voice communication solutions. Join our team if you are passionate about open-source technologies and possess a deep understanding of VoIP protocols. Responsibilities: Own all aspects of VoIP systems (Asterisk, Kamailio, RTPengine) from design of new features, to the implementation, QA, deployment to production, troubleshooting and maintenance. Be the subject matter expert for any VoIP related question coming from different parts of the company. Integrate VoIP systems into new and existing applications. Identify, optimize and resolve issues related to latency, scalability and performance. Monitor system performance, analyse traffic patterns, and suggest improvements. Automate processes that allow for faster deployment cycles and capacity scaling. Stay updated with VoIP and open-source software advancements. Participate in on-call rotations and respond to system emergencies. Qualifications (Candidate Profile): The candidate must have: Bachelor's degree in Computer Science, Engineering, or related field (or equivalent work experience). 2+ years of experience with VoIP, SIP/RTP and with open-source technologies. 2+ years of professional software development experience with C/C++/Golang building multi-threaded and highly performant client/server applications. Familiarity/Experience with open-source VoIP platform Asterisk/FreeSWITCH, Kamailio/OpenSIPS, RTPEngine in Linux environment. Experience with IP telephony and Networking protocols (SIP, RTP, RTCP, T.38, ISUP, TLS, STUN, TURN, WebRTC, T38). Full-stack troubleshooting skills across network, application, hardware and any distributed service stack. Proficiency in Linux and shell scripting. Excellent problem-solving and communication skills. Self-motivated and proactive learner.

Job Title: Senior VoIP Developer (Full-Time) Confidential Location: Remote / Jaipur (Optional Onsite) Address: Sitapura, Jaipur Experience Required: 3+ Years in VoIP Development Job Summary: We are looking for an experienced and passionate VoIP Developer to join our growing tech team. You will be responsible for designing, building, and maintaining robust and scalable VoIP systems including softswitches, dialers, PBX GUIs, billing solutions, and real-time integrations. You must have expertise in platforms like Asterisk, FreeSWITCH, Kamailio, OpenSIPS, VICIdial, FreePBX, FusionPBX, and billing systems such as MagnusBilling and ASTPP. Experience with GUI development, API integrations, and multi-tenant VoIP systems is essential. Key Responsibilities: ● Build and manage VoIP platforms using: ○ Asterisk, FreeSWITCH, Kamailio, OpenSIPS ○ GUI frameworks: FreePBX, FusionPBX ○ Dialers: VICIdial, GoAutoDial ○ Billing platforms: MagnusBilling, ASTPP ● Design and develop custom IVRs, call routing logic, and SIP trunk integrations. ● Develop web-based VoIP control panels, dashboards, and custom admin/client GUIs. ● Integrate with 3rd-party platforms like: ○ WhatsApp Business API ○ CRMs and SMS Gateways ○ Payment and Email Systems ● Perform SIP debugging, audio troubleshooting, NAT traversal handling, and codec optimization. ● Work on multi-tenant architecture, high availability, failover, and security setup. ● Automate and script deployments using Bash, Python, or PHP. ● Manage VoIP instances on Proxmox, VMware, or cloud providers like AWS/DigitalOcean. Confidential Required Skills & Experience: ● Minimum 3 years of experience in VoIP system development. ● Strong knowledge of SIP/RTP/NAT, VoIP codecs, and signaling. ● Hands-on experience with: ○ Linux environments (Ubuntu, Debian, CentOS) ○ MySQL/PostgreSQL databases ○ Scripting in Bash/Python/PHP ● Tools: Wireshark, sngrep, tcpdump, Asterisk CLI, FS-CLI Preferred Skills (Bonus): ● Familiarity with WebRTC, STIR/SHAKEN, and telecom compliance. ● Experience in multi-tenant VoIP platform scaling. ● API development and third-party app integrations. ● Prior work in a telecom SaaS, startup, or CPaaS provider. What We Offer: ● Competitive salary based on skills and experience. ● Remote work flexibility and performance-based growth. ● Exposure to advanced telecom projects and platforms. ● Fast-paced, collaborative, and innovation-driven culture. Job Types: Full-time, Permanent Pay: ₹3.50 - ₹7.00 per year Application Question(s): Do you have expertise in platforms like Asterisk, FreeSWITCH, Kamailio, OpenSIPS, VICIdial, FreePBX, FusionPBX, and billing systems ? Do you have minimum 3 years of experience in VoIP system development ? Work Location: Remote

About Client :- Our client is a French multinational information technology (IT) services and consulting company, headquartered in Paris, France. Founded in 1967, It has been a leader in business transformation for over 50 years, leveraging technology to address a wide range of business needs, from strategy and design to managing operations. The company is committed to unleashing human energy through technology for an inclusive and sustainable future, helping organizations accelerate their transition to a digital and sustainable world. They provide a variety of services, including consulting, technology, professional, and outsourcing services. Job Details :- Position: VOIP Engineer Experience Required: 4-8yrs Notice: immediate Work Location: PAN India Mode Of Work: Hybrid Type of Hiring: Contract to Hire Job Description:- The colleague who has comprehensive VoIP Solutions: Extensive experience in deploying and managing open-source telephony platforms, including Kamailio for SIP routing, Asterisk for PBX functionalities, and PJSIP for advanced SIP handling. Skilled in integrating these technologies to create scalable, high-performance communication systems tailored to business needs.

Tech @ Exotel Exotel engineering solves some really cool infrastructure level problems with the goal of ensuring no one misses a call or an SMS. ● Our focus is on building a very fault-tolerant, loosely coupled, scalable and real-time distributed system ● We are generally agnostic of language, technology or tools. Currently, our tech stack is built on Golang, Node.js, Ruby, Java and PHP. We use Aerospike, MySQL as data stores, ElasticSearch for search and Beanstalkd for queuing ● We emphasize a lot on clean abstractions of code, loosely coupled services and good coding practices ● We are very strong believers in "you built it, you own it!". And running a distributed system is very different from just building one! ● We are crazy about the high availability What you will do? ● Lead projects related to Exotel's telephony and VoIP stack. Responsible for driving projects throughout its lifecycle. ● Work with a team of engineers to explore, design, develop, test, deploy, and operationalize a product's features & improvements. ● Build fault-tolerant, scalable and real-time distributed voip system ● Effectively work in a collaborative and agile team environment ● Support team with timely analysis and debugging of operational issues. ● Be on rotational on-call roster to handle operational issues ● Emphasis on automation and scripting. ● Mentor junior engineers in the team. What we look for? Must Haves ● Bachelors or Masters degree in Computer Science or Communications Engineering ● Solid knowledge on VoIP domain technologies including SIP, SDP, RTP, RTCP. ● Solid knowledge on IP networking concepts and L2/L3 routing including Subnets, VLAN, NAT Traversal (ICE/STUN). ● Experience leading project team to deliver critical software solutions ● 3+ years experience in high-availability, scalable and fault tolerant voip infrastructure. ● 3+ years experience in working with server-side voip infrastructure components like Asterisk/Freeswitch, Kamailio/Opensips. including understanding of stack internals ● 3+ years experience in working with a major cloud platform like AWS (preferred), GCP, Azure. ● 2+ years experience in design and development of REST-based microservices. ● 2+ Experience with WebRTC ● Experience in programming with one of Golang (preferred), Ruby, C/C++ ● Experience in scripting with shell, python ● Experience with datastores such as MySQL, Postgres ● Experience with DevOps tools like Ansible, Jenkins, Terraform, kubernetes, Git ● Good understanding of data structures, multi-threading and concurrency concepts. ● Proficiency in working in Linux environment ● Experience working in Agile SDLC process ● Strong analytical, problem solving and troubleshooting skills ● Excellent written and verbal communication skills ● Team-player, flexible and able to work in a fast-paced environment ● A "devops" mindset. You own what you will develop. Job Type: Full-time Pay: ₹150,000.00 - ₹200,000.00 per year Location Type: In-person Schedule: Day shift Work Location: In person Speak with the employer +91 6393722524

Job description Job Description: We are seeking an experienced Asterisk Developer with a strong background in Lua scripting to join our dynamic team. The ideal candidate will have in-depth knowledge of Asterisk Developer , telecommunication protocols, and extensive experience in developing and maintaining robust communication solutions. Responsibilities: Install, configure and deploy Hosted VoIP systems, ensuring seamless functionality. Design, develop, and maintain sophisticated IVR and Call Center applications using Asterisk Developer , with a focus on Lua scripting. Integrate Asterisk with PSTN, demonstrating proficiency in bridging VoIP technology with traditional telephony. Implement Asterisk integration with databases, including MySQL/PostgreSQL/MongoDB. Utilize Lua scripting, as well as other scripting languages such as Shell and Python, for automation and customization Solid understanding of Linux servers, including Debian, CentOS, Nginx,Flask , Apache, MySQL,Python and Lua scripting. Install and configure Kamailio sip proxy and RTP Engine Conduct rigorous testing and troubleshooting of Astersik applications Develop and maintain a comprehensive understanding of PBX, SIP, RTP, and related telecommunication protocols. Collaborate with cross-functional teams to design and implement communication solutions. Stay up-to-date with industry trends and contribute to research and development efforts. Qualifications: Bachelor's degree in Computer Science, Telecommunications, or related field. 4 years of hands-on experience with Asterisk and some exposure to Asterisk. Strong understanding of PBX, SIP, RTP, and related telecommunication protocols. Proficiency in Asterisk Developer Dial-plan, Event Socket Library (ESL), and APIs. Experience in developing applications related to Queues, IVR, and Voicemail. Hands on Python and Lua scripting along with Solid understanding of LAMP servers (i.e CentOS, Apache, MySQL, PHP). Exposure to DevOPS and CI/CD systems and processes would be added advantage Job Types: Full-time, Permanent Pay: ₹30,000.00 - ₹60,000.00 per month Benefits: Paid sick time Paid time off Location Type: In-person Schedule: Day shift Monday to Friday Education: Bachelor's (Preferred) Contact Person - HR NIDHI THAKUR CONTACT NUMEBR - 9999753291/ 8929340639 Email -ID - nidhi.thakur@spinonweb.biz Job Types: Full-time, Permanent Pay: ₹30,000.00 - ₹60,000.00 per month Benefits: Leave encashment Paid sick time Paid time off Work Location: In person

Role Description This is a full-time on-site role located in Ahmedabad for a Kamailio (VOIP) Developer. The Kamailio (VOIP) Developer will be responsible for the development, configuration, and maintenance of Kamailio, Freeswitch and related VOIP systems. Day-to-day tasks include scripting, debugging, and testing VOIP solutions, collaborating with team members on system design, and ensuring system security and reliability. The developer will also need to troubleshoot system issues and provide technical support as required. Qualifications Experience with Kamailio, OpenSIPS, or similar VOIP servers Skills in scripting languages such as Lua, Python, or Perl Knowledge of SIP protocol, RTP, and other VOIP technologies Understanding of network protocols and network security Experience with Linux/Unix system administration Excellent problem-solving and troubleshooting skills Ability to work independently and collaboratively in a team environment Bachelor’s degree in Computer Science, Engineering, or a related field Experience in telecommunications is a must Job Type: Full-time Pay: ₹50,000.00 - ₹400,000.00 per month Benefits: Flexible schedule Paid sick time Paid time off Work from home Schedule: Night shift Rotational shift Work Location: In person

What Does Success Look Like? We are looking for a Principal VOIP Engineer to lead the architecture and technical direc8on of our nextgen voice infrastructure. You’ll be responsible for building carrier- grade systems with high availability, low latency, and global scalability — powering mission-cri8cal voice communica8on in our CCaaS plaIorm. This is a hands-on leadership role where you will influence architecture, establish best prac8ces, and work cross-func8onally across Engineering, DevOps, Product, and QA teams. Seniority Level: Principal / Individual Contributor with technical leadership scope What You’ll Do • Design and implement VOIP (signaling and media) infrastructure using FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine • Architect session border controllers (SBC), NAT traversal, load balancing, and failover strategies • Define standards for call rou8ng and audio quality op8miza8on (codecs, jieer, etc.) • Lead ini8a8ves for scalability, observability, security, and resiliency of our voice infrastructure • Troubleshoot live traffic and provide technical leadership during major incidents • Collaborate with Backend and API teams to design provisioning, billing, and call analy8cs APIs • Evaluate and onboard open-source tools or commercial carriers as needed • Coach and mentor junior/lead engineers in VoIP best prac8ces What Makes You Qualified? • 12+ years of hands-on experience in the Telephony / VoIP / CPaaS domain • Strong knowledge of VoIP Protocols (SIP/SDP, RTP/RTCP), Networking fundamentals (UDP/TCP/IP, DNS, MPLS), QoS (latency, jieer, packet loss mi8ga8on). • Hands-on experience with Session Border Controller (SBC), Media Servers and WebRTC. • Expert-level understanding of SIP, RTP, NAT traversal (ICE/STUN/TURN), and VoIP security (TLS, SRTP, fraud preven8on) • Hands-on development experience with FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine • Experience in designing carrier-grade telephony plaXorms serving millions of calls • Strong systems programming and debugging skills in C/C++ • Strong troubleshoo8ng skills, with experience using network monitoring and debugging tools. • Familiarity with distributed systems and cloud-based deployments (AWS, GCP, Azure)

Sprinklr is a leading enterprise software company for all customer-facing functions. With advanced AI, Sprinklr's unified customer experience management (Unified-CXM) platform helps companies deliver human experiences to every customer, every time, across any modern channel. Headquartered in New York City with employees around the world, Sprinklr works with more than 1,000 of the world’s most valuable enterprises - global brands like Microsoft, P&G, Samsung and more than 50% of the Fortune 100. What Does Success Look Like? We are looking for a Principal VOIP Engineer to lead the architecture and technical direction of our next-gen voice infrastructure. You’ll be responsible for building carrier- grade systems with high availability, low latency, and global scalability- powering mission-critical voice communication in our CCaaS platform. This is a hands-on leadership role where you will influence architecture, establish best practices, and work cross-functionally across Engineering, DevOps, Product, and QA teams. Seniority Level: Principal / Individual Contributor with technical leadership scope. What You’ll Do: Design and implement VOIP (signaling and media) infrastructure using FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine. Architect session border controllers (SBC), NAT traversal, load balancing, and failover strategies. Define standards for call routing and audio quality optimization (codecs, jitter, etc.) Lead initiatives for scalability, observability, security, and resiliency of our voice infrastructure. Troubleshoot live traffic and provide technical leadership during major incidents. Collaborate with Backend and API teams to design provisioning, billing, and call analytics APIs. Evaluate and onboard open-source tools or commercial carriers as needed. Coach and mentor junior/lead engineers in VoIP best practices. What Makes You Qualified? 9 to 12 years of hands-on experience in the Telephony / VoIP / CPaaS domain. Strong knowledge of VoIP Protocols (SIP/SDP, RTP/RTCP), Networking fundamentals (UDP/TCP/IP, DNS, MPLS), QoS (latency, jitter, packet loss mitigation). Hands-on experience with Session Border Controller (SBC), Media Servers and WebRTC . Expert-level understanding of SIP, RTP, NAT traversal (ICE/STUN/TURN) , and VoIP security (TLS, SRTP, fraud prevention). Hands-on development experience with FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine. Experience in designing carrier-grade telephony plaforms serving millions of calls. Strong systems programming and debugging skills in C/C++ Strong troubleshooting skills, with experience using network monitoring and debugging tools. Familiarity with distributed systems and cloud-based deployments (AWS, GCP, Azure) Excellent problem-solving, debugging, and performance tuning skills

We’re Hiring: NOC (Network Operation Center) Support Engineer! (VOIP Process) Looking to take your telephony and network skills to the next level? Join a team working on exciting, advanced projects in the heart of Bhubaneswar! Location: Bhubaneswar (immediate joiners preferred) Experience: Minimum 3–5 years We’re looking for an immediate joiner with 3–5 years of experience in NOC support. What you’ll work on: VOIP, SIP, FreeSWITCH, Vicidial, FusionPBX, FreePBX Linux (Ubuntu, CentOS, Debian) TCP/IP, NAT & PAT firewall configurations Bash, Perl, PHP, Python scripting MySQL, MongoDB OpenSIPS, Kamailio Telephony troubleshooting & monitoring Apply now: careers@thecorporatematchmakers.com Call/WhatsApp: +91 63725 46110 #Bhubaneswar #BhubaneswarJobs #OdishaJobs #HiringInBhubaneswar #ImmediateJoiner #NOC #NOCSupport #NetworkEngineer #NetworkOperations #TelecomJobs #Telephony #VOIP #SIP #FreeSWITCH #Vicidial #FusionPBX #FreePBX #Linux #Ubuntu #CentOS #Debian #TCPIP #Firewall #Networking #ITJobs #ITCareers #TechJobsIndia #PythonJobs #Bash #Perl #PHP #MongoDB #MySQL #OpenSIPS #Kamailio #ITSupport #NetworkSecurity #SystemEngineer #TelecomEngineer #TechHiring #EngineeringJobs #BhubaneswarHiring #OdishaTechJobs #LinuxJobs #SoftwareJobsIndia #CloudJobsIndia #TechCommunity #NetworkingCareers #TechTalentIndia #TheCorporateMatchmakers #CareerOpportunities #JoinOurTeam #Bhubaneswar #BhubaneswarJobs #OdishaJobs #HiringInBhubaneswar #ImmediateJoiner #NOC #NOCSupport #NetworkEngineer #NetworkOperations #VOIP #VOIPProcess #TelecomJobs #Telephony #SIP #FreeSWITCH #Vicidial #FusionPBX #FreePBX #Linux #Ubuntu #CentOS #Debian #TCPIP #NAT #Firewall #Networking #ITJobs #ITCareers #TechJobsIndia #PythonJobs #Bash #Perl #PHP #MySQL #MongoDB #OpenSIPS #Kamailio #TelecomEngineer #TelephonySupport #SystemEngineer #NetworkSupport #TechHiring #EngineeringJobs #BhubaneswarHiring #OdishaTechJobs #LinuxJobs #SoftwareJobsIndia #CloudJobsIndia #TechCommunity #NetworkingCareers #TechTalentIndia #TheCorporateMatchmakers #CareerOpportunities #JoinOurTeam #VOIPJobs

JD:Job Description. We are hiring a QA Automation Engineer with experience in VoIP systems and automation testing . Candidates should have strong experience in VoIP engineering, including SIP messaging, call hold processes, and network/server-side VoIP development. Exposure to automation frameworks, test case integration with Jenkins, and scripting for SIP message generation is expected. Familiarity with Docker, CI/CD pipelines, and building automation from scratch is preferred. Candidates should also demonstrate knowledge in network testing, call quality analysis, and handling real-world technical challenges. Key Responsibilities: Design and execute automation tests for VoIP and backend systems. Validate SIP, RTP , and HTTP-based communications . Work on VoIP tools like SBC, Kamailio, RTP Engine, FreeSWITCH, or Asterisk. Perform testing of REST APIs and microservices . Analyze SIP logs and troubleshoot VoIP call flow issues. Write automation scripts in Python or similar. Work in Unix/Linux environments and use databases like PostgreSQL, MongoDB, or Cassandra . Collaborate with developers and DevOps for end-to-end testing. Requirements: 3+ years in QA/Automation Testing . 2+ years working with VoIP protocols (SIP, RTP, IMS). Experience with HTTP, REST APIs, and microservices . Hands-on with Python automation or other scripting languages. Familiar with Unix/Linux systems . Basic knowledge of cloud telephony components. Good communication and team collaboration skills. Bonus Skills: Experience with Twilio . CI/CD exposure (e.g., Jenkins). Knowledge of call quality metrics. Looking forward to your response.

Experience Required: 3+ Years in VoIP/SIP QA, Telecom, or CPaaS/UCaaS Testing About Zudu AI Zudu AI is a next-generation voice automation platform that leverages AI-driven voice agents, programmable SIP/VoIP infrastructure, and real-time call automation to help enterprises transform their customer communications. Our scalable CPaaS/UCaaS platform enables millions of calls for leading enterprises across industries. Role Overview As a Senior Quality Analyst , you will take ownership of QA strategy, automation, and regression testing for our SIP/VoIP, AI-powered voice platform. You will design, automate, and execute end-to-end tests for SIP trunking, call flows, failover, media, and AI agent integrations. This is a hands-on technical role working directly with backend, DevOps, and AI teams. Key Responsibilities Lead the planning and execution of comprehensive QA strategies for VoIP, SIP, and programmable call flow components Build and maintain automated test frameworks using SIPp, JMeter, Python/Bash, and cloud-based test rigs Design and execute load, performance, failover, and regression tests on CPaaS/UCaaS platform features Debug and analyze SIP signaling, RTP/media issues using Wireshark, SIPp, and protocol analyzers Own bug lifecycle: document, triage, and work with engineering to resolve critical issues Validate TTS/STT, AI bot integrations, DTMF, call transfers, SIP trunk registration, and trunk failover logic Collaborate with DevOps for CI/CD QA pipelines, test data management, and test environment setup Prepare and maintain clear test plans, reports, and best practice documentation for ongoing QA Requirements 3+ years of hands-on QA/automation experience in VoIP/SIP, telecom, or CPaaS/UCaaS Strong knowledge of SIP, RTP, SDP, media negotiation, and protocol debugging Proficient with SIPp, Wireshark, JMeter, and scripting (Python, Bash) Experience with at least one open-source SIP stack (OpenSIPS, Kamailio, Asterisk, FreeSWITCH) Familiarity with cloud infrastructure (AWS, Azure) and CI/CD tools (Git, Jenkins, Docker) Demonstrated experience automating regression and performance testing for high-concurrency call flows Excellent written and verbal communication, clear bug reporting, and attention to detail (Bonus) QA for programmable voice APIs, voicebots, or AI integrations What You’ll Gain Define and lead QA for a rapidly growing AI voice/CPaaS platform Exposure to next-gen programmable SIP, AI voice, and multi-tenant call automation Opportunity to influence platform quality for enterprise and global scale Potential to build and mentor a future QA team