FreeSWITCH Developer

CC33 Global Limited; Bengaluru; On-site; INR 1,500,000 - 2,100,000; Full time

Job summary

CC33 Global Limited is seeking a FreeSWITCH Developer to design and scale real-time communication platforms with a focus on WebRTC and WebSocket-based calling. You will build XML dialplans, integrate via ESL, and troubleshoot SIP/media across browsers and endpoints.

Responsibilities include scaling telephony, assisting frontend teams, and implementing AI/voice integrations. Expected to own multi-node deployments and production incident handling.

Qualifications

• 3+ years hands-on FreeSWITCH in production.
• XML dialplan, Sofia SIP profile, and gateway configuration experience.
• Production experience with ESL for call control and event handling.
• Working knowledge of the module ecosystem (mod_sofia, mod_conference, mod_verto, mod_curl, mod_dptools).

Responsibilities

• Design, configure, develop, and maintain FreeSWITCH platforms.
• Build and maintain XML dialplans, gateways, SIP profiles, extensions, and call-routing logic.
• Implement and scale WebRTC calling — SIP over WebSocket (WSS), DTLS-SRTP, ICE/STUN/TURN, Opus/VP8, browser-to-PSTN and browser-to-browser flows.
• Work with JsSIP/SIP.js and other WebRTC/SIP clients; support the frontend team on client-side call integration.
• Develop integrations using ESL, mod_xml_curl, REST APIs, and WebSockets.
• Configure and troubleshoot SIP trunks, endpoints, DID/DDI routing, caller ID/CLI, and number translation.
• Diagnose SIP signalling and media issues — one-way audio, no audio, call drops, registration and NAT failures, ICE failures, codec and transcoding problems — using Sofia logs, PCAPs, Wireshark, and tcpdump.
• Integrate FreeSWITCH with CRMs, databases, third-party APIs, and AI/voice platforms via mod_audio_stream / WebSocket media streaming.
• Implement call recording, CDR, billing hooks, call-detail processing, and reporting.
• Own scaling of the telephony layer: multi-node FreeSWITCH deployments, media/signalling separation, TURN server capacity, failover, and safe rolling deploys.
• Work with the infrastructure team on sizing, networking, firewall and media port configuration, and deployment topology.
• Handle production incidents and lead root-cause analysis.
• Write and maintain technical documentation for call flows, configurations, and troubleshooting runbooks.

Skills

FreeSWITCH (production)

XML dialplan

Sofia SIP & gateways

ESL for call control

Tools

ESL

mod_sofia

mod_curl

WebSocket signaling

Job description

Job Overview

We're looking for a FreeSWITCH Developer to build and scale real-time communication platforms, with a strong focus on WebRTC and WebSocket-based calling. You'll design XML dialplans and call flows, build integrations over ESL, debug SIP and media problems across browser and SIP endpoints, and take ownership of how the telephony layer scales in production.

Key Responsibilities

• Design, configure, develop, and maintain FreeSWITCH platforms.
• Build and maintain XML dialplans, gateways, SIP profiles, extensions, and call-routing logic.
• Implement and scaleWebRTCcalling — SIP over WebSocket (WSS), DTLS-SRTP, ICE/STUN/TURN, Opus/VP8, browser-to-PSTN and browser-to-browser flows.
• Work with JsSIP/SIP.js and other WebRTC/SIP clients; support the frontend team on client-side call integration.
• Develop integrations usingESL(inbound and outbound event socket), mod_xml_curl, REST APIs, and WebSockets.
• Configure and troubleshoot SIP trunks, endpoints, DID/DDI routing, caller ID/CLI, and number translation.
• Diagnose SIP signalling and media issues — one-way audio, no audio, call drops, registration and NAT failures, ICE failures, codec and transcoding problems — using Sofia logs, PCAPs, Wireshark, and tcpdump.
• Integrate FreeSWITCH with CRMs, databases, third-party APIs, and AI/voice platforms (ASR, TTS, LLM-driven voice agents) via mod_audio_stream / WebSocket media streaming.
• Implement call recording, CDR, billing hooks, call-detail processing, and reporting.
• Own scaling of the telephony layer: multi-node FreeSWITCH deployments, media/signalling separation, TURN server capacity, failover, and safe rolling deploys.
• Work with the infrastructure team on sizing, networking, firewall and media port configuration, and deployment topology.
• Handle production incidents and lead root-cause analysis.
• Write and maintain technical documentation for call flows, configurations, and troubleshooting runbooks.

Required Skills

FreeSWITCH (deep)

• 3+ years hands-on FreeSWITCH in production.
• Strong XML dialplan, Sofia SIP profile, and gateway configuration experience.
• Production experience withESLfor call control and event handling.
• Working knowledge of the module ecosystem (mod_sofia, mod_conference, mod_verto, mod_curl, mod_dptools).

WebRTC & WebSockets

• Proven production experience deliveringWebRTCcalling through FreeSWITCH.
• Solid understanding of SIP over WSS, ICE/STUN/TURN, DTLS-SRTP, and WebRTC codec/transcoding trade-offs.
• Hands-on with WebSocket-based signalling and media streaming.

SIP & Media

• Excellent working knowledge of SIP, SDP, RTP/RTCP, DTMF, codecs, NAT traversal, and SIP response codes.
• Proven ability to root-cause complex signalling and media issues from PCAPs and SIP logs.
• Experience with SIP trunking, carrier integrations, DID/DDI, CLI, and number translation.

Engineering & Infrastructure

• Strong scripting/development skills in Python (or PHP/Go/Node) and Bash.
• PostgreSQL or MySQL/MariaDB, and Redis.
• Solid Linux administration, networking, firewalls, DNS, and load balancers.
• Experience scaling telephony infrastructure — multi-node deployments, HA, TURN/media capacity planning.
• Comfortable in cloud environments; containers/Kubernetes is a plus.

Good to Have

• Experience integrating telephony with AI voice systems — real-time audio streaming to ASR, TTS playback, barge-in handling, latency tuning for conversational voice bots.
• Exposure to Kamailio/OpenSIPS as an edge SIP proxy.
• Monitoring and observability for VoIP (Prometheus/Grafana, Homer/sipcapture).