50 Freeswitch Jobs - Page 2

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

About the Role We're seeking a skilled and proactive VoIP Engineer with at least 2 years of hands-on experience in SIP-based systems, Linux server management, and VoIP troubleshooting. If you've ever found beauty in a perfect SIP handshake or debugged NAT hell like a champ, youll feel right at home here. Youll be responsible for maintaining, optimizing, and expanding our IP telephony and unified communication infrastructureworking across Asterisk, FreeSWITCH, Kamailio, and other open-source tools. Key Responsibilities Deploy, configure, and manage VoIP infrastructure (Asterisk, FreePBX, Kamailio, OpenSIPS, or similar). Monitor and troubleshoot call quality issues, SIP signaling, RTP media streams, and NAT/firewall behavior. Manage Linux servers for VoIP applications service tuning, logs, security, cron jobs. Handle integrations with CRM, call recording, SIP gateways, SBCs, and media servers. Write and optimize dialplans, IVRs, and call routing logic using Asterisk or Lua/JSON for FreeSWITCH. Analyze PCAPs and SIP traces using Wireshark, sngrep, Homer, or similar tools. Work with networking and DevOps teams to ensure QoS, bandwidth, and latency optimization. Maintain documentation for configurations, deployments, and internal processes. Required Skills & Experience Minimum 2 years of VoIP experience in a production environment. Strong knowledge of SIP, RTP, SRTP, SIP registration, and NAT traversal techniques. Proficiency in Linux system administration (Ubuntu/RHEL) and bash scripting. Practical experience with VoIP monitoring and debugging tools (e.g., sngrep, tcpdump, Wireshark). Understanding of IP networking (TCP/UDP, DNS, DHCP, VLANs, routing). Ability to interpret SIP INVITE, 200 OK, BYE, REGISTER flows like poetry. Good-to-Have (Not Mandatory, but Gold) Familiarity with Kamailio, OpenSIPS, or other SIP proxy/registrar solutions. Experience with VoIP security measures (TLS, SRTP, fail2ban, SIP authentication). Exposure to SIPREC, TTS/STT, or AI-based call analytics. Understanding of number masking, multi-tenant PBX, or contact center integrations. Knowledge of monitoring stacks (Grafana/Prometheus) or VoIP-aware dashboards. DevOps-friendly skills: Docker, Git, CI/CD pipelines. Education Bachelors degree in Computer Science, Information Technology, Electronics, or relevant field.

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 3 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.

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.

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.

Role & responsibilities Asterisk Call Flow VOIP Telephony Linux Shell Scripting with Asterisk IP PBX Asterisk Gateway Interface (AGI) SIP MySQL Database Preferred candidate profile

We are looking for a MERN Stack developer with hands on experience in developing VOIP Applications. The candidate should have sufficient knowledge of integration of Freeswitch into Node JS applications.Ability to use various AI Agents for coding. Performance bonus Annual bonus

- Develop, implement, and maintain VoIP solutions using FreeSwitch and Asterisk. - Integrate VoIP systems with existing applications and infrastructure. - Troubleshoot and resolve issues related to VoIP systems and protocols. - Collaborate with cross-functional teams to design and implement new features and enhancements. - Stay updated with emerging technologies and industry trends in VoIP.

Hiring Software Engineer - 3 (Telephony & VoIP) | 3+ yrs exp in SIP, RTP, WebRTC, Asterisk/Freeswitch, Golang, AWS, Kubernetes, DevOps, Microservices | Build scalable, real-time VoIP infra | Bangalore | Full-time | Apply: ssankala@toppersedge.com

What You ll Do: Implement core voice capabilities using FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine. Build and optimize SIP call routing logic, RTP media relays , failover mechanisms, and NAT traversal. Develop and manage configurations for scalability, codec negotiation, SIP trunk registration . Implement and test features like call recording, IVR, voicemail, DTMF detection. Monitor live traffic and participate in 24x7 on-call rotation for critical escalations. Collaborate with QA on stress/load testing and with Backend teams on provisioning APIs. Document design decisions, configurations, and troubleshooting runbooks. What Makes You Qualified 5 to 9 years of experience building and operating VoIP systems or CPaaS platforms . Solid expertise with SIP signaling, RTP, and media relay techniques . Strong hands-on with FreeSWITCH, Kamailio/OpenSIPs, RTPEngine . Hands-on experience with Session Border Controller (SBC), Media Servers and WebRTC . Experience in managing telephony infrastructure for uptime, latency, and call quality optimization. Strong systems programming and debugging skills in C/C++ . Good scripting/debugging skills ( Bash, Python, or Lua for FreeSWITCH modules ). Proficiency with diagnostic tools ( Wireshark, tcpdump etc ). Experience working with geographically distributed infrastructure or HA deployments.

What You ll Do: Implement core voice capabilities using FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine. Build and optimize SIP call routing logic, RTP media relays , failover mechanisms, and NAT traversal. Develop and manage configurations for scalability, codec negotiation, SIP trunk registration . Implement and test features like call recording, IVR, voicemail, DTMF detection. Monitor live traffic and participate in 24x7 on-call rotation for critical escalations. Collaborate with QA on stress/load testing and with Backend teams on provisioning APIs. Document design decisions, configurations, and troubleshooting runbooks. What Makes You Qualified 5 to 9 years of experience building and operating VoIP systems or CPaaS platforms . Solid expertise with SIP signaling, RTP, and media relay techniques . Strong hands-on with FreeSWITCH, Kamailio/OpenSIPs, RTPEngine . Hands-on experience with Session Border Controller (SBC), Media Servers and WebRTC . Experience in managing telephony infrastructure for uptime, latency, and call quality optimization. Strong systems programming and debugging skills in C/C++ . Good scripting/debugging skills ( Bash, Python, or Lua for FreeSWITCH modules ). Proficiency with diagnostic tools ( Wireshark, tcpdump etc ). Experience working with geographically distributed infrastructure or HA deployments.

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 trac 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 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

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. Required Candidate profile Interested candidate can share their resume at divya.garg@webviotechnologies.com or contact at 9045186615.

YES SECURITIES is looking for Network In-Charge to join our dynamic team and embark on a rewarding career journey. Network Infrastructure Management : Design, implement, and maintain the organization's network infrastructure, including local area networks (LANs), wide area networks (WANs), wireless networks, and network hardware (routers, switches, firewalls, etc. ). Network Security : Develop and implement security measures to safeguard the organization's network from unauthorized access, data breaches, and cyber threats. This includes setting up firewalls, intrusion detection/prevention systems, and encryption protocols. Network Monitoring and Troubleshooting : Continuously monitor the network for performance issues, downtime, and anomalies. Identify and resolve network-related problems promptly to minimize disruptions. Capacity Planning : Analyze network traffic patterns and usage to forecast future network capacity needs. Scale the network infrastructure as necessary to accommodate growth. Vendor Management : Collaborate with vendors and service providers to ensure the organization has access to reliable and cost-effective network services, hardware, and software solutions. Network Policies and Procedures : Develop and enforce network-related policies, procedures, and best practices to ensure consistency, compliance, and efficient network usage. Team Leadership : Lead and manage a team of network administrators and technicians. Assign tasks, provide guidance, and facilitate professional development. Network Documentation : Maintain accurate and up-to-date documentation of the network architecture, configurations, diagrams, and procedures. Disaster Recovery and Business Continuity : Develop and implement plans to ensure network availability during unforeseen events or disasters. Regularly test and update these plans to minimize downtime. Collaboration : Work closely with other IT teams, such as systems administrators, cybersecurity professionals, and application developers, to ensure seamless integration of network services with other IT components.

As an AI Voice LLM Integration Developer, you will play a pivotal role in building and optimizing Your work will focus on: Integrating GPT's ASR (speech-to-text) and TTS (text-to-speech) APIs into our platform. Handling real-time voice streaming from VoIP/SIP systems. Optimizing low-latency communication between AI models and telephony platforms (e.g., FreeSWITCH, WebRTC). Supporting PBX call routing handoWs with AI agents. Ensuring high-quality, natural-sounding AI-driven voice interactions. Key Responsibilities Design and develop real-time voice processing pipelines for AI-driven conversations. Integrate OpenAI's Whisper (ASR) and TTS APIs with VoIP and WebRTC systems. Develop low-latency audio streaming middleware between telephony providers(Telnyx, Bandwidth) and GPT. Work with SIP and WebRTC to ensure seamless audio transmission. Implement PBX call routing and AI handoW mechanisms (e.g., transferring AI-handled calls to human agents). Optimize AI voice latency and response times for a real-time, natural experience. Collaborate with VoIP engineers to integrate AI capabilities into SIP-based systems. Design APIs to manage AI-driven IVR workflows and customer interactions. Ensure compliance with telephony regulations (e.g., STIR/SHAKEN, call recording laws). Monitor system performance and implement scalability strategies for voice interactions. Required Qualifications AI LLM Integration o Experience integrating GPT APIs (ChatGPT, Whisper, TTS engines) for voice applications. o Understanding of real-time AI voice processing and conversational AI workflows. Voice Audio Streaming o Experience working with low-latency, real-time audio streaming. o Familiarity with WebRTC, SIP/RTP, and VoIP streaming. Programming Middleware Development o Proficiency in Python, Node.js, or Go for API and middleware development. o Experience developing RESTful APIs and real-time streaming solutions. VoIP Telephony Knowledge o Familiarity with SIP, FreeSWITCH, Asterisk, or other PBX platforms. o Experience working with SIP trunking providers (Telnyx, Bandwidth, Flowroute, etc.). Performance Optimization o Ability to reduce latency in AI-driven conversations. o Experience with audio codec optimization (G.711, Opus, etc.). One or more of the following additional qualifications: o Experience with STT/ASR models beyond GPT (Google Speech-to- Text, Deepgram, Kaldi). o Background in signal processing or audio engineering. o Familiarity with cloud-based voice solutions (AWS Connect, Twilio Voice, Dialogflow CX). o Knowledge of containerization (Docker, Kubernetes) for scaling AI voice workloads. o Experience with multi-tenant architectures for voice AI.

Job Summary: We are looking for a Kamailio & FreeSWITCH Engineer with expertise in VoIP protocols, Linux administration, web servers, and MySQL databases to maintain and optimize our telecommunications infrastructure. Key Responsibilities: Kamailio SIP Server Management Configure, maintain, and troubleshoot SIP routing. FreeSWITCH Administration – Install, optimize, and debug VoIP functionalities. Linux OS & Web Server – Manage Ubuntu/CentOS , NGINX, and Apache for high performance. Database Management – Optimize MySQL databases for scalability. Collaboration & Support – Work with teams to deploy features and troubleshoot issues. Required Skills: Proven experience with Kamailio & FreeSWITCH Strong knowledge of Linux OS & VoIP protocols (SIP, RTP) Expertise in NGINX, Apache, and MySQL Hands-on experience with scripting (Shell, Python) Problem-solving mindset & ability to work in a fast-paced environment Preferred Qualifications: Familiarity with AWS/Azure cloud platforms Knowledge of virtualization tools (VMware, Proxmox) Experience in high-availability and load-balancing configurations Location: Hyderabad Employment Type: Full-Time Experience: Minimum 1 year

We are looking for an experienced Asterisk and FreeSWITCH Developer to design, develop, and maintain robust VoIP communication systems. You will be instrumental in creating scalable PBX systems, call center solutions, and custom telephony applications using open-source VoIP platforms. Key Responsibilities: Design, implement, and maintain VoIP solutions using Asterisk and FreeSWITCH . Develop and manage PBX systems , IVRs, and SIP-based telephony features. Integrate VoIP systems with CRM, databases, and third-party APIs. Monitor and troubleshoot VoIP network performance, ensuring high availability and quality of service. Customize dial plans, call flows, codecs, and other telephony features. Maintain system documentation and provide post-deployment support. Required Skills: Hands-on experience (3-5 years) with Asterisk and FreeSWITCH . Strong understanding of SIP, RTP, and VoIP protocols . Proficiency in Linux system administration and scripting (e.g., Bash, Python). Familiarity with MySQL/PostgreSQL for backend data handling. Experience with tools like Wireshark , SIPp , or other VoIP testing utilities. Knowledge of call routing , media transcoding , and NAT traversal . Good to Have: Experience integrating VoIP platforms with CRM systems . Exposure to Kamailio/OpenSIPS , WebRTC, or telephony billing systems. Certification in VoIP technologies or Linux administration.

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

The Associate Software Team Lead is pivotal in steering a group of talented software engineers towards the successful execution of R&D projects. This role involves a blend of technical expertise and leadership skills tmanage the development lifecycle, mentor team members, and ensure that software deliverables are innovative, robust, and align with customer expectations. The Associate Team Lead acts as a bridge between the engineering team and senior management, translating business objectives inttechnical strategies, fostering a culture of continuous improvement, and maintaining a focus on both short-term milestones and long-term goals. Qualification: Relevant industry certifications (Azure/AWS certified professional) Bachelor s degree in computer science / engineering, or equivalent work experience. Software Engineer level of experience with exceptional Real-Time skills and enthusiasm Proven ability tself-manage and structure work, this must be demonstrated through clear examples in your application Product / Technical : Degree in Computer Science or Engineering or Equivalent with 10+ years of relevant experience working with C/C++, C#, PHP. Must have 6+ years of Linux C++ / C developer. Must have 6+ years of Windows C/ C#/.NET, Dependency Injection Experience with Service Bus, Test Driven development Must have strong background in muti-process / multi-threaded application design. Must be proficient in Linux (currently using EL9) - Development, Bash shell. Must have strong background using and/or implementing SIP, RTP, or other voice protocols. Working knowledge of Asterisk/FreeSwitch Experience with Machine Learning technologies, NLP, Python libraries (Pandas, Keras, TensorFlow etc.) Good understanding of Python libraries for machine learning, Computer vision, Speech Analytics and Deep Learning tools & techniques Working experience of Cloud (preferably AWS) is an added advantage. Working experience of Cloud (OKD / OpenShift preferred) development Working knowledge of Cloud tools such as Kubernetes, and CI/CD tools such as Harness and/or Jenkins. Working knowledge of Monitoring Tools such as Datadog and/or OpsGenie. Experience working JIRA and in an Agile team. Knowledge of front end technologies (React Js , Node Js, Java script) Working knowledge on Application Security/Vulnerability tools like Black Duck, Coverity / App Scan etc. Experience with API / RESTful data services Experience using Postgres and SQL Server database technologies. Knowledge of VXML & IVR technologies/solutions. Experience of voice & viderecording platforms is advantageous Good understanding of Computer Vision, Speech Analytics and Deep Learning tools & techniques Core Tasks: Lead and support the VASR and Fonolproduct development and maintenance, ensuring global customer success. Initial ramp up is expected tbe based on small product issue resolution building tnew feature development. Once team established and product knowledge at required level, lead the development and implementation of software projects from conception tdeployment. Provide technical expertise and guidance in software design, coding standards, and system integration. Participate in technical requirements though tdelivery Estimates take intconsideration all aspects of solution and are relatively accurate. Tasks and Defects are addressed proactively. Quality gates are met for deliverables. Champion agile development methodology within the development organization. Ensure customer success when called upon tassist in complex issues. Mentor Associate and Graduate Engineers. Ensure the quality and reliability of software through rigorous testing and code reviews. Encourage innovation and the exploration of new technologies tenhance product capabilities. Troubleshoot and resolve complex technical issues that arise during the development process. Manage the allocation of resources, including personnel and technology, toptimize productivity. Establish and monitor performance metrics tevaluate the success of software projects

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 worlds 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 trac 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? 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

Expertise in C, SIP and RTP Expertise and customize Freeswitch for audio/video conferencing 2-4 years of experience in telecom protocols like SIP, RTP, and SMPP. (FreeSwitch, WebRTC). Familiarity with Opensip, Lua, PBX and Kamailio Required Candidate profile 5 days working but rotational off Note- Interested candidate can directly contact at 9045186615.

Software Engr(Backend+VoIP, 3–5 yrs), SIP/SDP/RTP/RTCP, Asterisk/Freeswitch, Kamailio/Opensips,WebRTC,REST APIs,Golang/Ruby/C++, AWS/GCP/Azure, MySQL/Postgres,DevOps (K8s/Terraform/Ansible). C2H via TE Infotech(Exotel), BLR. @ssankala@toppersedge.com

We are seeking a highly skilled and motivated VoIP Engineer with deep expertise in the FreeSWITCH open-source communication platform. The ideal candidate will be responsible for designing, developing, deploying, and maintaining our Voice over IP (VoIP) infrastructure. You will be a key player in ensuring the high availability, scalability, and performance of our telephony systems, and will work on a variety of projects from building custom call flows to integrating with external APIs and services. Key Responsibilities Design, implement, and manage VoIP solutions using FreeSWITCH, ensuring high availability, reliability, and security. Configure and optimize FreeSWITCH components, including dial plans, call queues, Interactive Voice Response (IVR) systems, and other call-handling logic. Troubleshoot and resolve complex VoIP issues, including call quality problems, routing failures, and signaling issues, using tools like Wireshark, sngrep, and tcpdump. Develop custom modules and applications for FreeSWITCH using scripting and programming languages such as Lua, Python, C/C++, or JavaScript. Integrate FreeSWITCH with third-party systems, including Customer Relationship Management (CRM) software, databases (e.g., MySQL, PostgreSQL), and RESTful APIs. Monitor system performance and proactively identify and address potential issues using monitoring and logging tools (e.g., Prometheus, Grafana, syslog). Ensure security best practices are implemented, including firewall configuration, TLS encryption, and protection against toll fraud. Collaborate with cross-functional teams, including software developers, network engineers, and support staff, to support the companys communication infrastructure strategy. Maintain comprehensive documentation for all configurations, procedures, and troubleshooting steps. Stay up-to-date with the latest trends and technologies in VoIP and real-time communications. Required Qualifications & Skills Bachelor's degree in Computer Science, Information Technology, Telecommunications, or a related field (or equivalent practical experience). Proven experience as a VoIP Engineer, Telecommunications Engineer, or a similar role. Deep and proven expertise with FreeSWITCH, including its core components, dial plan syntax (XML and Lua), and modules. Strong understanding of VoIP protocols, including SIP, RTP, and WebRTC. Proficiency in Linux system administration (e.g., Debian, Ubuntu, CentOS) and comfortable with the command-line interface. Experience with scripting languages like Lua, Python, or Perl for call flow logic and automation. Solid understanding of networking fundamentals (TCP/IP, UDP, DNS, NAT traversal, QoS). Experience with network troubleshooting tools like Wireshark, sngrep, and tcpdump. Familiarity with database technologies (e.g., MySQL, PostgreSQL) and how to integrate them with FreeSWITCH. Excellent problem-solving, analytical, and critical-thinking skills. Strong communication and interpersonal skills, with the ability to explain complex technical concepts to non-technical stakeholders. Preferred Skills (Nice to Have) Experience with other open-source PBX platforms like Asterisk or Kamailio. Knowledge of cloud infrastructure (e.g., AWS, Azure, GCP) and experience deploying and managing FreeSWITCH in a cloud environment. Familiarity with containerization technologies like Docker and orchestration platforms like Kubernetes. Experience with other related technologies such as FusionPBX, Kamailio, or OpenSIPS. Experience in a call center or a service provider environment. Knowledge of VoIP security frameworks like STIR/SHAKEN. Experience with automated testing for VoIP systems (e.g., using SIP).

Hiring a Telephony & CCaaS Integration Engineer to lead on-prem integrations with Genesys, RingCentral & Ozonetel. Must have SIP/VoIP, APIs, SBCs & Asterisk/FreeSWITCH expertise. Work with AI & backend teams to deliver end-to-end voice solutions.

Role & responsibilities Junior System Administrator (VoIP/Telephony) / Telephony Engineer (Telecommunications) No. of Positions - 1 Night Shift) 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