37 Kamailio Jobs - Page 2

QA Automation & VoIP Engineer Location Experience : 3+ Years Joining : Immediate joiners or max 1-week notice preferred Job Summary: We are hiring a QA Automation Engineer with experience in VoIP systems and automation testing . This role involves testing backend systems, APIs, and VoIP protocols using tools and frameworks. The ideal candidate should be comfortable with both testing and development tasks and have hands-on experience with cloud telephony , VoIP components , and backend APIs . Key Responsibilities: Design and execute automation tests for VoIP and backend systems. Validate SIP, RTP , and HTTP-based communications . Work on VoIP tools like SBC, Kamailio, RTP Engine, FreeSWITCH, or Asterisk. Perform testing of REST APIs and microservices . Analyze SIP logs and troubleshoot VoIP call flow issues. Write automation scripts in Python or similar. Work in Unix/Linux environments and use databases like PostgreSQL, MongoDB, or Cassandra . Collaborate with developers and DevOps for end-to-end testing. Requirements: 3+ years in QA/Automation Testing . 2+ years working with VoIP protocols (SIP, RTP, IMS). Experience with HTTP, REST APIs, and microservices . Hands-on with Python automation or other scripting languages. Familiar with Unix/Linux systems . Basic knowledge of cloud telephony components. Good communication and team collaboration skills. Bonus Skills: Experience with Twilio . CI/CD exposure (e.g., Jenkins). Knowledge of call quality metrics. Show more Show less

Sprinklr is a leading enterprise software company for all customer-facing functions. With advanced AI, Sprinklr's unified customer experience management (Unified-CXM) platform helps companies deliver human experiences to every customer, every time, across any modern channel. Headquartered in New York City with employees around the world, Sprinklr works with more than 1,000 of the world’s most valuable enterprises - global brands like Microsoft, P&G, Samsung and more than 50% of the Fortune 100. What Does Success Look Like? We are seeking a Lead VoIP Engineer to design and build high-performance modules within our Voice platform. You’ll work on the core telephony stack involving signaling, media processing, NAT traversal, and RTP relaying. This is a hands-on execution role ideal for engineers who love building, debugging, and optimizing real-time communication systems. Seniority Level: Lead Individual Contributor 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? 7+ 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. Show more Show less

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.

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. Show more Show less

Job Summary: We are seeking an experienced NOC Manager with strong expertise in VoIP, Asterisk, Linux, and networking to oversee and optimize our telephony and network operations. The ideal candidate will have hands-on experience with Kamailio/OpenSIPS, SIP, TCP/IP, and scripting (Bash, Python, PHP, Perl). Knowledge of AI/ML and open-source technologies will be an added advantage. Key Responsibilities: Technical Operations & Maintenance: Operate, maintain, and expand SIP proxies (Kamailio/OpenSIPS). Manage Asterisk-based systems (AGI, AMI, ARI, IPPBX, EPBAX, custom PBX). Troubleshoot VoIP gateways, SBCs, and telephony routing issues. Ensure high availability, security, and performance of Linux servers (Ubuntu, CentOS, Debian). Handle Level 3 escalations for critical telephony and network issues. Monitoring & Optimization: Design and maintain monitoring and alerting systems for VoIP services. Analyze telephony traffic patterns to detect anomalies and optimize performance. Resolve NAT, PAT, and firewall-related VoIP issues. Automation & Development: Develop scripts in Bash, Python, Perl, PHP for automation. Work with MySQL/MongoDB databases for call routing and billing systems. Implement new telephony functionalities and integrations. Team & Process Management: Lead and mentor NOC engineers in troubleshooting and best practices. Coordinate with cross-functional teams for seamless operations. Maintain documentation and SOPs for network and VoIP services. Required Skills & Qualifications: 3-5 years of hands-on experience in Asterisk, VoIP, and Networking. Strong knowledge of Linux (Ubuntu, CentOS, Debian), Bash, TCP/IP, SIP. Experience with Kamailio/OpenSIPS, VoIP gateways, and SBCs. Proficiency in scripting (Python, Bash, Perl, PHP). Understanding of NAT, PAT, firewalls, and VoIP security. Experience with MySQL/MongoDB and call billing systems. Excellent communication, problem-solving, and teamwork skills. Show more Show less

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

Location - South Delhi Exp - Min 3+ years (5 Days working in roster) Responsibilities:  Design, develop, deploy, troubleshoot, and maintain tools and services supporting our cloud telephony network.  Expertise in SIP, RTP, RTCP, TURN, STUN, NAT, and TLS.  Customize FreeSWITCH for audio/video conferencing, ensuring it handles 1000-1500 concurrent calls.  Proficiency in RTP Proxy and routed audio conferences.  Understanding of SDP Protocol offer/answer.  Work with load testing tools for FreeSWITCH audio conferences.  Deploy multiple FreeSWITCH instances using load balancers.  3-4 years of experience in telecom protocols like SIP, RTP, and SMPP.  Knowledge of codecs (PCMU, PCMA, G729, Opus) and open-source telephony technologies (FreeSWITCH, WebRTC).  Debugging using packet captures.  Familiarity with SIP/RTP, H323 a plus.  Collaborate with the mobile team on APIs and support.  Knowledge of SBC, FreeSWITCH, and SIPX is a plus.  Bachelor's degree in Engineering (B.Tech or MCA).  3+ years of FreeSWITCH development or Ring Group Module or call centre module.  Proficient in Linux environments.  Basic to intermediate SQL skills.  Familiarity with network tracing tools (Wireshark/ SNgrep).  Strong problem-solving skills.  Strong understanding of VoIP protocols (SIP, RTP), codecs, and related technologies.  Solid knowledge of Linux operating systems and command-line tools.  Proficiency in coding language like C and Lua.  Understanding of networking principles, TCP/IP, DNS, DHCP, and routing protocols.  Knowledge of FusionPBX, Kamailio and Opensips. Job Type: Full-time Pay: ₹50,000.00 - ₹100,000.00 per month Benefits: Cell phone reimbursement Health insurance Life insurance Paid sick time Provident Fund Schedule: Day shift Work Location: In person

Sprinklr is a leading enterprise software company for all customer-facing functions. With advanced AI, Sprinklr's unified customer experience management (Unified-CXM) platform helps companies deliver human experiences to every customer, every time, across any modern channel. Headquartered in New York City with employees around the world, Sprinklr works with more than 1,000 of the world’s most valuable enterprises - global brands like Microsoft, P&G, Samsung and more than 50% of the Fortune 100. What Does Success Look Like? We are seeking a Lead VoIP Engineer to design and build high-performance modules within our Voice platform. You’ll work on the core telephony stack involving signaling, media processing, NAT traversal, and RTP relaying. This is a hands-on execution role ideal for engineers who love building, debugging, and optimizing real-time communication systems. Seniority Level: Lead Individual Contributor 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? 7+ 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. Show more Show less

● Strong experience with SIP protocol (INVITE, ACK, BYE, REGISTER, REFER,OPTIONS, etc.) ● Practical experience with SIPREC for recording VoIP calls. ● Solid development skills in JavaScript (Node.js). ● Experience working with SIP Servers (e.g., FreeSWITCH, Asterisk, Kamailio,OpenSIPS). ● Hands-on knowledge of WebRTC, RTP streams, and VoIP media handling. ● Experience building and consuming RESTful APIs. ● Familiarity with call flows, SIP traces analysis (using Wireshark, sngrep, or similar). ● Strong understanding of networking basics (UDP, TCP, NAT traversal, STUN/TURN). ● Ability to troubleshoot and debug complex telephony and media issues Show more Show less

Design and implement telephony integrations using SIP and SIPREC Practical experience with SIPREC for recording VoIP calls. Experience working with SIP Servers (e.g., FreeSWITCH, Asterisk, Kamailio, OpenSIPS). Hands-on knowledge of WebRTC , RTP streams, and VoIP media handling We are looking for a SIP Developer only. Show more Show less

This role is for one of the Weekday's clients Min Experience: 5 years JobType: full-time We are looking for an experienced SIP Developer with a strong foundation in VoIP technologies and telephony integrations. This role offers the opportunity to work remotely in a dynamic and flexible environment, contributing to cutting-edge communication solutions. Requirements Key Responsibilities Design and implement telephony integrations using SIP and SIPREC protocols. Develop and maintain call recording solutions leveraging SIPREC. Work with industry-standard SIP servers such as FreeSWITCH, Asterisk, Kamailio, or OpenSIPS. Manage WebRTC, RTP streams, and VoIP media processing to build reliable and scalable communication systems. Required Skills & Experience 5-8 years of hands-on experience in VoIP development. Deep knowledge of SIP and SIPREC protocols. Proficiency in working with SIP servers like FreeSWITCH, Asterisk, Kamailio, and OpenSIPS. Solid understanding of WebRTC, RTP, and VoIP media handling. Technical Skills SIP, SIPREC FreeSWITCH, Asterisk Kamailio, OpenSIPS WebRTC, RTP VoIP protocols Show more Show less

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.