32 Kamailio Jobs - Page 2

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.