96 Kamailio Jobs - Page 2

Role & responsibilities Must-Haves Strong VoIP Protocol Knowledge - SIP, RTP, WebRTC Hands-on VoIP Platforms - Asterisk or Freeswitch (Kamailio/OpenSIPS is a bonus) Programming in Golang (preferred) OR Ruby / C++ with willingness to learn Golang Networking Fundamentals for VoIP - NAT traversal, ICE/STUN, L2/L3 routing Cloud Experience - AWS (preferred) or GCP/Azure Linux Proficiency (deployment, debugging, and scripting) Nice-to-Haves Kamailio/OpenSIPS expertise SIP testing tools Elasticsearch + Grafana/Kibana DevOps tools (Terraform, Kubernetes, Ansible, Jenkins) Android/iOS VoIP client development

Must have experience in Freeswitch SIP and RTP Protocols Must have experience in coding language C, Python or Lua Kamailio and Opensip is a plus point. Call Hr 8016099788

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

Location Name: Pune Corporate Office - Mantri Job Purpose VoIP Engineer (Asterisk, Kamailio) - 5-7 Years Experience Duties And Responsibilities A- Minimum required Accountabilities for this role VoIP Infrastructure Design & Maintenance: o Configure, deploy, and maintain FreeSWITCH and Kamailio-based systems. o Understanding of VoIP architectures to support business needs. o Implement call routing, DID management, and trunk configurations. SIP & Call Routing: o Develop and manage SIP-based call routing for internal and external communication. o Troubleshoot SIP signaling issues using tools like Wireshark or sngrep. o Optimize routing rules for least-cost routing (LCR) and high availability. Monitoring & Performance Optimization: o Monitor VoIP systems for performance, security, and uptime. o Conduct capacity planning and optimize system resources. o Implement call quality monitoring tools (e.g., RTCP, QoS metrics). Collaboration & Support: o Work with cross-functional teams to integrate VoIP systems with CRM and other platforms. o Provide Level 2/3 support for VoIP-related issues. o Train team members on VoIP best practices and system usage. B- Additional Accountabilities Pertaining To The Role Security & Compliance: o Ensure VoIP security measures to prevent fraud and mitigate risks. o Ensure compliance with industry standards and regulations (e.g., GDPR, HIPAA). o Configure firewalls and SBCs for secure SIP trunking. Database & Messaging Integration: o Integrate VoIP systems with databases like MongoDB and PostgreSQL. o Leverage Redis for caching and RabbitMQ for messaging queues. o Ensure efficient data storage and retrieval mechanisms to support VoIP services. Programming & Scripting: o Develop custom VoIP features and modules using languages like Golang, Lua, Python, C, and C++. o Automate repetitive tasks and processes through scripting. o Understanding of WebRTC solutions for real-time communication. Key Decisions / Dimensions Soft Skills: o Strong analytical and problem-solving skills. o Excellent communication and documentation abilities. o Ability to work collaboratively in a team environment Major Challenges VoIP Expertise: o Strong hands-on experience with FreeSWITCH and Kamailio. o In-depth knowledge of SIP protocols, RTP, and VoIP troubleshooting. o Familiarity with codecs like G.711, G.729, Opus, etc. Networking Proficiency: o Solid understanding of networking concepts such as NAT, RTP, and STUN/TURN. o Experience with firewall configurations and SBCs. Development Skills: o Proficiency in programming languages like Golang, Lua, Python, C, and C++. o Experience with WebRTC for real-time communication. o Familiarity with database systems like MongoDB and PostgreSQL. o Experience with Redis for caching and RabbitMQ for asynchronous messaging. Tools & Platforms: o Familiarity with monitoring tools like Homer, Grafana, or Nagios. o Hands-on experience with cloud platforms like AWS, Azure, or Google Cloud. o Proficiency in containerization tools like Docker and orchestration with Kubernetes. Required Qualifications And Experience Qualifications Graduates with relevant voip experience of 4-5 Years Experience Work Experience VoIP Infrastructure Design & Maintenance: o Configure, deploy, and maintain FreeSWITCH and Kamailio-based systems. o Design scalable and reliable VoIP architectures to support business needs. o Implement call routing, DID management, and trunk configurations. SIP & Call Routing: o Develop and manage SIP-based call routing for internal and external communication. o Troubleshoot SIP signaling issues using tools like Wireshark or sngrep. o Optimize routing rules for least-cost routing (LCR) and high availability. Monitoring & Performance Optimization: o Monitor VoIP systems for performance, security, and uptime. o Conduct capacity planning and optimize system resources. o Implement call quality monitoring tools (e.g., RTCP, QoS metrics). Collaboration & Support: o Work with cross-functional teams to integrate VoIP systems with CRM and other platforms. o Provide Level 2/3 support for VoIP-related issues. o Train team members on VoIP best practices and system usage. Security & Compliance: o Implement VoIP security measures to prevent fraud and mitigate risks. o Ensure compliance with industry standards and regulations (e.g., GDPR, HIPAA). o Configure firewalls and SBCs for secure SIP trunking.

ITPune Corporate Office - Mantri Posted On 18 Aug 2025 End Date 18 Aug 2026 Required Experience 4 - 5 Years BASIC SECTION Job Level GB05 Job Title Associate Tech Solutionist - Contact Centre - IT, Emerging Tech, Contact Centre Job Location Country India State MAHARASHTRA Region West City Pune Location Name Pune Corporate Office - Mantri Tier Tier 1 Skills SKILL SKILLS AS PER JD Minimum Qualification OTHERS JOB DESCRIPTION Job Purpose VoIP Engineer (Asterisk, Kamailio) - 5-7 Years Experience Duties and Responsibilities A- Minimum required Accountabilities for this role VoIP Infrastructure Design & Maintenance: o Configure, deploy, and maintain FreeSWITCH and Kamailio-based systems. o Understanding of VoIP architectures to support business needs. o Implement call routing, DID management, and trunk configurations. SIP & Call Routing: o Develop and manage SIP-based call routing for internal and external communication. o Troubleshoot SIP signaling issues using tools like Wireshark or sngrep. o Optimize routing rules for least-cost routing (LCR) and high availability. Monitoring & Performance Optimization: o Monitor VoIP systems for performance, security, and uptime. o Conduct capacity planning and optimize system resources. o Implement call quality monitoring tools (e.g., RTCP, QoS metrics). Collaboration & Support: o Work with cross-functional teams to integrate VoIP systems with CRM and other platforms. o Provide Level 2/3 support for VoIP-related issues. o Train team members on VoIP best practices and system usage. B- Additional Accountabilities pertaining to the role Security & Compliance: o Ensure VoIP security measures to prevent fraud and mitigate risks. o Ensure compliance with industry standards and regulations (e.g., GDPR, HIPAA). o Configure firewalls and SBCs for secure SIP trunking. Database & Messaging Integration: o Integrate VoIP systems with databases like MongoDB and PostgreSQL. o Leverage Redis for caching and RabbitMQ for messaging queues. o Ensure efficient data storage and retrieval mechanisms to support VoIP services. Programming & Scripting: o Develop custom VoIP features and modules using languages like Golang, Lua, Python, C, and C++. o Automate repetitive tasks and processes through scripting. o Understanding of WebRTC solutions for real-time communication. Key Decisions / Dimensions Soft Skills: o Strong analytical and problem-solving skills. o Excellent communication and documentation abilities. o Ability to work collaboratively in a team environment Major Challenges VoIP Expertise: o Strong hands-on experience with FreeSWITCH and Kamailio. o In-depth knowledge of SIP protocols, RTP, and VoIP troubleshooting. o Familiarity with codecs like G.711, G.729, Opus, etc. Networking Proficiency: o Solid understanding of networking concepts such as NAT, RTP, and STUN/TURN. o Experience with firewall configurations and SBCs. Development Skills: o Proficiency in programming languages like Golang, Lua, Python, C, and C++. o Experience with WebRTC for real-time communication. o Familiarity with database systems like MongoDB and PostgreSQL. o Experience with Redis for caching and RabbitMQ for asynchronous messaging. Tools & Platforms: o Familiarity with monitoring tools like Homer, Grafana, or Nagios. o Hands-on experience with cloud platforms like AWS, Azure, or Google Cloud. o Proficiency in containerization tools like Docker and orchestration with Kubernetes. Required Qualifications and Experience a) Qualifications Graduates with relevant voip experience of 4-5 Years Experience b) Work Experience VoIP Infrastructure Design & Maintenance: o Configure, deploy, and maintain FreeSWITCH and Kamailio-based systems. o Design scalable and reliable VoIP architectures to support business needs. o Implement call routing, DID management, and trunk configurations. SIP & Call Routing: o Develop and manage SIP-based call routing for internal and external communication. o Troubleshoot SIP signaling issues using tools like Wireshark or sngrep. o Optimize routing rules for least-cost routing (LCR) and high availability. Monitoring & Performance Optimization: o Monitor VoIP systems for performance, security, and uptime. o Conduct capacity planning and optimize system resources. o Implement call quality monitoring tools (e.g., RTCP, QoS metrics). Collaboration & Support: o Work with cross-functional teams to integrate VoIP systems with CRM and other platforms. o Provide Level 2/3 support for VoIP-related issues. o Train team members on VoIP best practices and system usage. Security & Compliance: o Implement VoIP security measures to prevent fraud and mitigate risks. o Ensure compliance with industry standards and regulations (e.g., GDPR, HIPAA). o Configure firewalls and SBCs for secure SIP trunking.

We're on the hunt for a technically-minded, carrier-savvy Voice/Telecommunications Support Engineer to join our team. You’ll be the frontline detective, problem solver, and system tuner — ensuring smooth operation of our core voice systems and integrations across our global footprint. This is a hands-on role supporting both internal operations and external customers via our Atomic Solution Centre, with a focus on VoIP, SIP routing, carrier interconnects, softswitch functionality, and end-to-end service delivery. Key Responsibilities  Provide Level 2/3 support for voice and UCaaS services including SIP trunking, DID provisioning, number porting, Teams Direct Routing, and Hosted PBX.  Manage and troubleshoot call routing logic, carrier interconnects, and softswitch configurations.  Work closely with global voice and SMS carriers to manage interconnects, test routing paths, and resolve downstream quality issues.  Configure and monitor routing policies, codecs, call detail records (CDRs), and fraud mitigation parameters.  Maintain and update system-wide infrastructure components (e.g. SBCs, SIP proxies, load balancers).  Assist with monthly billing reconciliations, call rating issues, and reporting anomalies.  Support infrastructure updates and software upgrades in production and staging environments.  Contribute to the customer knowledge base and internal documentation repository (e.g. Atomic Solution Centre).  Participate in a rotating on-call schedule for incident response and after-hours escalations. What You’ll Bring  3+ years in a technical telco, VoIP, or UCaaS support role.  Strong understanding of SIP, RTP, DNS, NAT traversal, and voice codec behaviour.  Experience with softswitch environments (e.g. SippySoft, FreeSWITCH, Kamailio, OpenSIPS).  Familiarity with billing systems (e.g. rate plans, mediation, CDR reconciliation).  Previous involvement with carrier provisioning, global numbering plans, and LCR.  Proficiency using Wireshark or similar tools for SIP diagnostics.  Comfortable with Linux-based systems and basic scripting for automation/troubleshooting.  Excellent communication and customer engagement skills. Nice to Have  Exposure to Microsoft Teams Direct Routing and Operator Connect configuration.  SMS gateway knowledge and Sender ID registration practices.  Experience with fraud detection platforms and IP reputation scoring tools.  Familiarity with REST APIs for service integration and automation. Training & Certifications We’re looking for candidates who’ve invested in their technical growth and understand the intricacies of modern voice and telecom networks. Preferred certifications include:  SIP/VoIP Technologies:  - Ribbon, Oracle, or Audiocodes SBC certifications  - SIP School Certification (SSCA)  Networking & Infrastructure:  - Cisco CCNA/CCNP  - Juniper JNCIA  - CompTIA Network+ or equivalent  Cloud/Unified Comms:  - Microsoft Teams/365 Administrator certifications (e.g., MS-700)  - AWS Certified Cloud Practitioner or Solutions Architect (nice to have)  Security/Fraud Mitigation (bonus):  - CompTIA Security+  - Experience with STIR/SHAKEN, IP filtering, or voice firewall systems If you haven’t ticked every box — but you're sharp, resourceful, and eager to learn — don’t let that stop you. We invest in upskilling the right people. Monitoring & Tools Experience with the following tools and platforms will be highly regarded:  VoIPmonitor for SIP call tracing, MOS scoring, and packet analysis.  Grafana for real-time system monitoring and dashboard visualisation.  AWS services including CloudWatch, EC2, S3, and networking components for infrastructure management and scalability.  Basic scripting for automation (Bash, Python) and log parsing.

Position Overview: We are looking for a talented VoIP Engineer to join our dynamic team. The ideal candidate will possess solid expertise in VoIP technologies, including Kamailio, FreeSWITCH, SIP, and WebRTC, along with practical experience in networking, AWS cloud services, and Linux system administration. This role is perfect for someone who enjoys problem-solving, collaborative teamwork, and working in a fast-paced technology-driven environment. Key Responsibilities Design, deploy, and maintain VoIP infrastructure using Kamailio, FreeSWITCH and related tools. Configure and troubleshoot SIP and WebRTC protocols. Administer and optimize Linux-based servers for performance and reliability. Monitor network performance, manage network infrastructure, and troubleshoot networking issues. Deploy and manage infrastructure on AWS, leveraging relevant cloud technologies. Document infrastructure setup, configuration, and procedures. Collaborate closely with other teams (Backend, QA, Operations) to support the overall infrastructure strategy. Participate in on-call rotations and respond effectively to infrastructure incidents. Qualifications & Requirements Minimum 2+ years of experience with VoIP applications. Hands-on experience with Kamailio and FreeSWITCH. Solid understanding of SIP, RTP, WebRTC and Websockets. Proven Linux system administration experience (Ubuntu/Debian preferred) Familiarity with networking concepts (TCP/IP, TLS, VPN, IP Addressing, NAT Traversal) Practical experience with AWS services such as EC2, S3, VPC, and RDS. Understanding of Docker and scalable deployment strategies. Desired Soft Skills Strong analytical and problem-solving skills. Excellent communication and interpersonal skills. Self-motivated, proactive, and able to work effectively in a team. Organized with the ability to manage multiple priorities simultaneously. Adaptable, eager to learn new technologies, and committed to continuous improvement.

The company is looking for a motivated and technically inclined NOC Engineer (Entry-Level) to join the Network Operations team in New Town, Kolkata. This is an excellent opportunity for recent graduates or individuals with relevant certifications/projects in Asterisk, VoIP, Networking, or Linux to kickstart their career in the telecom industry. As a part of the team, you will collaborate with senior engineers, receive on-the-job training, and contribute to building, maintaining, and supporting telephony infrastructure and services. Your role will require a basic understanding of VoIP and telephony protocols like SIP, familiarity with Linux-based systems such as Ubuntu/CentOS, basic networking concepts like TCP/IP, NAT, and Firewall rules, exposure to scripting languages like Bash or Python, and an understanding of databases like MySQL or MongoDB. Additionally, you should be willing to learn about Asterisk, IPPBX, and EPABX systems. Preferred skills include academic projects or certifications in VoIP or Asterisk, exposure to OpenSIPS/Kamailio or other SIP proxies, familiarity with network monitoring tools like Nagios or Zabbix, and basic knowledge of telecom billing and call routing. Your key responsibilities will involve assisting in monitoring VoIP traffic and network health, supporting senior engineers in troubleshooting voice/network issues, learning and contributing to the deployment and configuration of PBX systems, participating in system maintenance and upgrade activities, and maintaining documentation and incident reports. Soft skills such as eagerness to learn and grow in the telecom domain, good verbal and written communication, and being a team player with a proactive and analytical approach are highly valued. If you are enthusiastic about telecom technologies and keen to establish a career in VoIP and networking, this role is ideal for you. Freshers with strong foundational skills or relevant coursework are encouraged to apply. This is a full-time position with the benefit of working from home and the opportunity to work in person at the specified location.,

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.