28 Freeswitch Jobs

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

Roll-Freeswitch/Asterisk Developer Exp- 2+ years Type-FTE Budget- As per company standard Location- Pune Mode- WFO Notice Period- Immediate to 90 days Max Mandatory skill set- FreeSWITCH, Kamilio, Programming & Scripting: Develop custom VoIP features and modules using any one languages like Golang, Lua, Python, C, and C++,AWS.

As a Senior Java Developer/Technical Lead for Voice & Email Platform at our growing team, you will be responsible for developing, optimizing, and maintaining Java-based applications. Your role involves working with RESTful APIs, Webhooks, and WebSocket protocols to facilitate real-time communication. Building secure and scalable backend systems to efficiently handle high-volume traffic is a key aspect of your responsibilities. Collaboration with cross-functional teams is essential to enhance performance and reliability. Additionally, you will troubleshoot issues related to support tickets and third-party integrations and optimize system performance using multithreading, caching, and database indexing techniques. To excel in this role, you should have 3 to 7 years of experience as a Senior Developer, with expertise in Java, Spring Boot, Hibernate, and Microservices. Hands-on experience with Asterisk or Freeswitch, along with a solid understanding of RESTful APIs, Webhooks, and JSON, is required. Proficiency in MySQL, PostgreSQL, or NoSQL databases (MongoDB, Redis), exposure to AWS and scalable application architecture, and familiarity with message brokers like RabbitMQ and Kafka are necessary. Knowledge of security protocols such as OAuth, JWT, and experience with Docker, Kubernetes, CI/CD pipelines, and Git (GitHub, GitLab, Bitbucket) is expected. Unit testing using JUnit, Mockito, experience in CPaaS, and understanding of chatbots, NLP, and AI-driven messaging are essential skills. For the Tech Lead role, in addition to the above requirements, you need to have a strong grasp of design patterns and best practices, proven leadership and team mentorship abilities, and experience in managing sprints and tasks via tools like JIRA. Exposure to building Voice Bots using AI/ML is considered a significant advantage. The ideal candidate should hold a Bachelors or Masters degree in Computer Science, Information Technology, or a related field. If you are passionate about building high-performance communication systems and enjoy solving real-world challenges, we encourage you to apply for this exciting opportunity with us. Contact us directly via messaging or apply now to jobs@pinnacle.in or dhiraj.rokade@pinnacle.in. Join us in revolutionizing communication platforms and making a difference in the tech industry! #JavaJobs #Hiring #TechLead #SeniorDeveloper #SpringBoot #VoicePlatform #MessagingPlatform #CPaaS #AI #Freeswitch #Asterisk #Microservices #LiveChat,

Job Title: Engineering Manager Location: Gurugram, Haryana Experience Required: 12-15 years About the Role: We are seeking an experienced Engineering Manager to lead a talented team of FreeSWITCH engineers, Python developers, and UI/UX designers in building and scaling our click-to-call marketing SaaS analytics platform for advertisers. You will drive the development of a robust, scalable, and secure platform that empowers advertisers with real-time call tracking, analytics, and campaign optimization tools. Key Responsibilities Lead and mentor a cross-functional team of FreeSWITCH engineers, Python developers, and UI/UX designers to deliver high-quality, scalable features for our SaaS platform. Oversee the end-to-end development lifecycle, from planning and architecture to deployment and maintenance, using Agile/Scrum methodologies. Ensure seamless integration of telephony systems (e.g., FreeSWITCH, VoIP) with analytics pipelines and advertiser-facing dashboards. Collaborate with product managers and stakeholders to translate advertiser needs (e.g., call attribution, ROI tracking) into technical solutions. Implement and maintain CI/CD pipelines, containerized environments, and API-driven integrations to support rapid iteration and scalability. - Uphold SaaS security best practices (e.g., SOC2, GDPR) to protect sensitive call and advertiser data. Foster a collaborative, inclusive team culture, motivating engineers to achieve technical and professional excellence. Qualifications Bachelors degree in Computer Science, Engineering, or a related field (Masters preferred). 1215 years of software engineering experience, with expertise in both front-end (e.g., React, Angular) and back-end development (e.g., Python, Node.js). Proven experience building and scaling B2B SaaS platforms, ideally in adtech, martech, or analytics. Hands-on experience with VoIP and telephony systems (e.g., FreeSWITCH, Asterisk) and real-time communication protocols (e.g., SIP, WebRTC). Strong understanding of data analytics pipelines for processing and visualizing marketing metrics (e.g., call attribution, conversion tracking). Demonstrated leadership in managing cross-functional engineering teams, with a track record of delivering complex projects on time. Proficiency in containerization (e.g., Docker, Kubernetes), microservices, and API-first architectures. Experience with CI/CD tools (e.g., Jenkins, GitLab CI) and cloud platforms (e.g., AWS, GCP, Azure). Knowledge of SaaS security, including access control, data privacy, and compliance (SOC2, GDPR). Excellent problem-solving and decision-making skills in fast-paced environments. - Strong communication skills to convey technical concepts to non-technical stakeholders, such as advertisers or marketing teams. (Optional) Experience working in dynamic markets like Dubai or the Middle East, with an understanding of regional business practices. Preferred Skills Familiarity with adtech/martech ecosystems (e.g., CRM integrations, Google Ads, or call tracking platforms). Experience mentoring remote or distributed teams to maintain productivity and collaboration. Passion for building user-centric analytics platforms with intuitive UI/UX. Why Join Us? Be part of a fast-growing SaaS company revolutionizing click-to-call marketing analytics. Lead a talented team, work on cutting-edge telephony and analytics solutions, and make a direct impact on advertisers success.

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.,

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.