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Apply now: https://talynce.com/jobs/59cb2f80-efd1-414e-81ac-79b86bd28496 Job Title - Telephony Engineer Salary - 25 to 40 LPA Mode โ WFO Notice Period โ immediate to 30 days Exp โ 3-10 yrs Mandatory - 1-5 yrs in Freeswitch Development exp mandatory Work Location - Gurgaon [Candidates to be based out of Delhi NCR Only] Job Summary We are looking for a FreeSWITCH Implementation Engineer who will be responsible for designing, deploying, configuring, integrating, and maintaining FreeSWITCH-based VoIP platforms . The role involves working closely with SIP, RTP, WebRTC, IVR, conferencing, call routing, and external integrations in a production environment. ๐ Key Responsibilities ๐น Free SWITCH Core Responsibilities Install, configure, and maintain FreeSWITCH in production environments from the scratch. Design and implement call flows, dialplans, IVRs, conferencing, and call routing with complete ownership. Configure SIP profiles (internal/external) , gateways, trunks, and extensions Implement multi-tenant and high-availability FreeSWITCH architectures ๐น SIP, RTP & Networking Strong understanding of SIP signaling , RTP media , and SDP negotiation Troubleshoot one-way audio, no-audio, call drops, registration failures Configure NAT traversal , firewall rules, and RTP port ranges Analyze SIP logs and traces (siptrace, sngrep, tcpdump) ๐น Dialplan & Scripting Develop dialplans using: XML Lua / JavaScript / Python Implement logic for: Call queues for inbound and outbound calls Time-based routing Skill-based routing Failover and fallback handling ๐น Integrations & APIs Integrate FreeSWITCH with: CRM / ERP systems Billing platforms AI voice bots (ASR/TTS) External WebSocket or REST services Use Event Socket Library (ESL) : Inbound & Outbound ESL Real-time call control and monitoring ๐น WebRTC & Media Configure WebRTC support (DTLS-SRTP, ICE, OPUS) Work with mod_verto , SIP over WSS Implement browser-based calling solutions ๐น Monitoring & Debugging Monitor system performance and call quality Analyze logs and CDRs Optimize codec usage and RTP performance Handle live production incidents and RCA ๐น Deployment & Operations Deploy FreeSWITCH on: Linux (CentOS / Ubuntu) Cloud platforms (AWS / GCP / Azure โ preferred) Configure HA, load balancing, and scaling Work with DevOps teams for CI/CD, backups, and automation ๐ Required Skills โ Technical Skills Strong experience with FreeSWITCH Solid knowledge of VoIP protocols (SIP, RTP, RTCP, WebRTC) Linux system administration Dialplan development (XML/Lua/JS/Python) ESL & event-driven call control Debugging tools (fs_cli, sngrep, tcpdump) SQL / MySQL / PostgreSQL (CDR, reporting) Apply now: https://talynce.com/jobs/59cb2f80-efd1-414e-81ac-79b86bd28496
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Preferred
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Apply now: https://talynce.com/jobs/59cb2f80-efd1-414e-81ac-79b86bd28496 Job Title - Telephony Engineer Salary - 25 to 40 LPA Mode โ WFO Notice Period โ immediate to 30 days Exp โ 3-10 yrs Mandatory - 1-5 yrs in Freeswitch Development exp mandatory Work Location - Gurgaon [Candidates to be based out of Delhi NCR Only] Job Summary We are looking for a FreeSWITCH Implementation Engineer who will be responsible for designing, deploying, configuring, integrating, and maintaining FreeSWITCH-based VoIP platforms . The role involves working closely with SIP, RTP, WebRTC, IVR, conferencing, call routing, and external integrations in a production environment. ๐ Key Responsibilities ๐น Free SWITCH Core Responsibilities Install, configure, and maintain FreeSWITCH in production environments from the scratch. Design and implement call flows, dialplans, IVRs, conferencing, and call routing with complete ownership. Configure SIP profiles (internal/external) , gateways, trunks, and extensions Implement multi-tenant and high-availability FreeSWITCH architectures ๐น SIP, RTP & Networking Strong understanding of SIP signaling , RTP media , and SDP negotiation Troubleshoot one-way audio, no-audio, call drops, registration failures Configure NAT traversal , firewall rules, and RTP port ranges Analyze SIP logs and traces (siptrace, sngrep, tcpdump) ๐น Dialplan & Scripting Develop dialplans using: XML Lua / JavaScript / Python Implement logic for: Call queues for inbound and outbound calls Time-based routing Skill-based routing Failover and fallback handling ๐น Integrations & APIs Integrate FreeSWITCH with: CRM / ERP systems Billing platforms AI voice bots (ASR/TTS) External WebSocket or REST services Use Event Socket Library (ESL) : Inbound & Outbound ESL Real-time call control and monitoring ๐น WebRTC & Media Configure WebRTC support (DTLS-SRTP, ICE, OPUS) Work with mod_verto , SIP over WSS Implement browser-based calling solutions ๐น Monitoring & Debugging Monitor system performance and call quality Analyze logs and CDRs Optimize codec usage and RTP performance Handle live production incidents and RCA ๐น Deployment & Operations Deploy FreeSWITCH on: Linux (CentOS / Ubuntu) Cloud platforms (AWS / GCP / Azure โ preferred) Configure HA, load balancing, and scaling Work with DevOps teams for CI/CD, backups, and automation ๐ Required Skills โ Technical Skills Strong experience with FreeSWITCH Solid knowledge of VoIP protocols (SIP, RTP, RTCP, WebRTC) Linux system administration Dialplan development (XML/Lua/JS/Python) ESL & event-driven call control Debugging tools (fs_cli, sngrep, tcpdump) SQL / MySQL / PostgreSQL (CDR, reporting) Apply now: https://talynce.com/jobs/59cb2f80-efd1-414e-81ac-79b86bd28496
Required
Preferred