Job Title SIP Developer Associate Architect Location Mumbai Bangalore - Hybrid 2 days in a week from office Only Immediate Joiners please Experience 6 Years We are seeking a highly skilled Telephony Integration Developer with deep expertise in SIP Session Initiation Protocol and SIPREC SIP Recording to join our growing team You will be responsible for designing developing and integrating telephony systems with a strong emphasis on VoIP communication call recording and SIP signaling Responsibilities Design and implement telephony integrations using SIP and SIPREC Develop APIs and backend services to handle call control call recording and session management Work with PBX systems SIP Servers and Media Servers for SIP call flows and media capture Integrate third-party VoIP systems with internal applications and platforms Analyze and troubleshoot SIP signaling and RTP media flows Collaborate with cross-functional teams including DevOps Product and QA to deliver scalable solutions Create technical documentation diagrams and support material Ensure systems are secure resilient and scalable Must-Have Skills 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 Good to Have Skills Experience with Media Servers e g Janus Kurento Mediasoup Knowledge of Call Recording Systems architecture and compliance standards PCI-DSS GDPR Experience with Cloud Telephony Platforms Twilio Genesys Cloud Amazon Chime SDK etc Familiarity with Session Border Controllers SBCs Prior experience with SIP trunking and carrier integrations Exposure to Protocol Buffers or gRPC for real-time messaging Understanding of security practices in VoIP TLS SRTP SIP over WebSockets Knowledge of Docker and Kubernetes for deploying SIP services at scale Sound knowledge of telecom protocols like SIP ICE STUN TURN SRTP DTLS H323 Diameter Radius Shall be thoroughly analytical and fix issues for SBC Portfolio of Products Shall be thorough with Linux RTOS internals and product Architecture is preferred Strong Knowledge of TCP UDP IP and networking concepts is a must Knowledge of IP telephony SIP Call Routing Techniques of ARS AAR on Trunk config workplace Prior Experience on working with FreeSwitch Kamailio RTP Proxy etc Strong understanding of Audio streaming websockets and their application in real-time communication systems In-depth knowledge of audio codecs and their impact on voice quality and bandwidth utilization Experience with gRPC and Protobuf for building efficient and scalable communication interfaces Extensive experience in large scale product development in Enterprise webRTC VoIP VoLTE based products Base Language Framework Primary Language JavaScript Node js backend Frameworks Tools Express js Socket io for signaling if needed Wireshark for debugging Sngrep Apply Contact WhatsApp 91 (phone hidden) Email Maqsood Jobskey com