FreeSWITCH WebRTC C++ Developer
1 week ago
Bengaluru, KA, India
Cerebra
Full-time
Free with email or Google
Save this job and keep your search organized
Create a free account to save jobs, create alerts and return to this listing from your dashboard.
Free with email or Google
By continuing, you agree to our Terms & Privacy Policy.
Role & responsibilities
Job Description
FreeSWITCH / WebRTC C++ Developer
Experience:
6–8 Years Must-Have
Skills:
C, C++, SIP, FreeSWITCH, WebRTC Good-to-Have: Asterisk or similar telephony platforms
Key Responsibilities
- Develop and customize FreeSWITCH using C/C++, including dialplans, modules, and ESL/API integrations.
- Implement and troubleshoot WebRTC components, including ICE, STUN/TURN, DTLS-SRTP, SDP, mod\_verto, and WSS.
- Configure and troubleshoot SIP trunks, NAT, codecs, RTP, SRTP, and TLS.
- Integrate FreeSWITCH with Contact Center / CCaaS platforms.
- Configure ACD, skill-based routing, and queue management using mod\_callcenter or FIFO queues.
- Enable call recording, call monitoring, whisper, barge-in, and supervisor features.
- Monitor system health, analyze logs, handle incidents, and manage changes.
- Troubleshoot gateways, WSS, TURN, ICE/DTLS, and codec-related issues.
- Prepare runbooks and SOPs and collaborate with cross-functional teams. Required Qualifications
- B.
E./B.
Tech in Computer Science, Electronics, Telecommunications, or equivalent.
- 6–8 years of relevant experience in telecom, VoIP, or real-time communication systems.
- Strong hands-on experience with FreeSWITCH, C/C++ programming, SIP, and WebRTC.
- Asterisk or similar telephony platform experience is an advantage.
Job Description
FreeSWITCH / WebRTC C++ Developer
Experience:
6–8 Years Must-Have
Skills:
C, C++, SIP, FreeSWITCH, WebRTC Good-to-Have: Asterisk or similar telephony platforms
Key Responsibilities
- Develop and customize FreeSWITCH using C/C++, including dialplans, modules, and ESL/API integrations.
- Implement and troubleshoot WebRTC components, including ICE, STUN/TURN, DTLS-SRTP, SDP, mod\_verto, and WSS.
- Configure and troubleshoot SIP trunks, NAT, codecs, RTP, SRTP, and TLS.
- Integrate FreeSWITCH with Contact Center / CCaaS platforms.
- Configure ACD, skill-based routing, and queue management using mod\_callcenter or FIFO queues.
- Enable call recording, call monitoring, whisper, barge-in, and supervisor features.
- Monitor system health, analyze logs, handle incidents, and manage changes.
- Troubleshoot gateways, WSS, TURN, ICE/DTLS, and codec-related issues.
- Prepare runbooks and SOPs and collaborate with cross-functional teams. Required Qualifications
- B.
E./B.
Tech in Computer Science, Electronics, Telecommunications, or equivalent.
- 6–8 years of relevant experience in telecom, VoIP, or real-time communication systems.
- Strong hands-on experience with FreeSWITCH, C/C++ programming, SIP, and WebRTC.
- Asterisk or similar telephony platform experience is an advantage.