78 Freeswitch Jobs - Page 3

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 supp...

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 trunkin...

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 y...

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 st...

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,...

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 err...

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 operat...

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 operat...

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...

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 net...

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 ...

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 ...

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. ...

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 desig...

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/...

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 carri...

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