SIP Developer
Jobskey Consultancy
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Job summary
Jobskey Consultancy is seeking a SIP Developer on a remote 6-month contract to join our VoIP-focused team. You will design telephony integrations and implement SIP-based solutions for robust communications.
The ideal candidate has 5–8 years of hands-on VoIP development, deep SIP/SIPREC expertise, and practical experience with SIP servers such as FreeSWITCH, Asterisk, Kamailio or OpenSIPS. Strong WebRTC and RTP knowledge is required.
Qualifications
- 5–8 years of hands-on experience in VoIP development.
- In-depth knowledge of SIP and SIPREC.
- Practical expertise with SIP Servers (FreeSWITCH, Asterisk, Kamailio, OpenSIPS).
- Strong understanding of WebRTC, RTP, and VoIP media handling.
Responsibilities
- Design and implement telephony integrations using SIP and SIPREC
- Develop and maintain call recording solutions utilizing SIPREC protocols
- Work with industry-standard SIP servers such as FreeSWITCH, Asterisk, Kamailio, or OpenSIPS
- Handle WebRTC, RTP streams, and VoIP media processing for robust communication solutions
Skills
Tools
Job description
Job Opening: SIP Developer (Remote, Contract) Location : Remote Duration : 6-Month Contract Work Hours : 12 PM to 10 PM IST Experience : 5-8 Years Compensation : Best in the Industry
About the Role : We are seeking a skilled SIP Developer to join our team on a contract basis. This is a great opportunity for professionals with a strong background in VoIP technologies and telephony integrations to work remotely in a flexible, fast-paced environment.
- Design and implement telephony integrations using SIP and SIPREC
- Develop and maintain call recording solutions utilizing SIPREC protocols
- Work with industry-standard SIP servers such as FreeSWITCH, Asterisk, Kamailio, or OpenSIPS
- Handle WebRTC, RTP streams, and VoIP media processing for robust communication solutions
- 5-8 years of hands‑on experience in VoIP development
- In‑depth knowledge of SIP and SIPREC
- Practical expertise with SIP Servers (FreeSWITCH, Asterisk, Kamailio, OpenSIPS)
- Strong understanding of WebRTC, RTP, and VoIP media handling
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