Job Description
VoIP Developer (SIP/WebRTC) – 5 to 8 Years Experience
Location: [Remote] - Bangalore , Pune , Mumbai , Gurgaon , Mohali
Experience Required: 5 to 8 years
Employment Type: Full-Time
About the Role:
We are seeking a skilled VoIP Developer with deep expertise in SIP protocols, WebRTC, and media handling to join our dynamic engineering team. This role requires hands-on experience with telephony systems, RTP streams, and backend service development using Node.js.
Key Responsibilities:
Develop and maintain real-time communication solutions using SIP and WebRTC.
Work with SIP Servers like FreeSWITCH, Asterisk, Kamailio, or OpenSIPS.
Implement SIP-based features (e.g., INVITE, ACK, BYE, REGISTER, REFER, OPTIONS).
Use SIPREC for recording VoIP calls and manage RTP media streams.
Troubleshoot SIP call flows and analyze SIP traces using Wireshark, sngrep, etc.
Build and integrate RESTful APIs to support VoIP features.
Collaborate with cross-functional teams to deliver high-quality, scalable solutions.
Required Skills & Experience:
5–8 years of experience in VoIP development.
Strong knowledge of SIP protocol and related tools.
Proficient in JavaScript and Node.js.
Hands-on experience with SIPREC, RTP, and WebRTC.
Familiarity with networking concepts including UDP, TCP, NAT traversal, STUN/TURN.
Ability to debug complex telephony and media issues in production environments.
Preferred Tools:
Wireshark, sngrep, FreeSWITCH, Asterisk, Kamailio, OpenSIPS, Postman, Git.
How to Apply:
Please submit your resume along with a brief cover letter to [[email protected]] or apply directly via Indeed.
Job Types: Full-time, Permanent
Pay: ₹5+2,683.83 - ₹1,18+,837.17 per year
Schedule:
Monday to Friday
Work Location: Remote
Location: [Remote] - Bangalore , Pune , Mumbai , Gurgaon , Mohali
Experience Required: 5 to 8 years
Employment Type: Full-Time
About the Role:
We are seeking a skilled VoIP Developer with deep expertise in SIP protocols, WebRTC, and media handling to join our dynamic engineering team. This role requires hands-on experience with telephony systems, RTP streams, and backend service development using Node.js.
Key Responsibilities:
Develop and maintain real-time communication solutions using SIP and WebRTC.
Work with SIP Servers like FreeSWITCH, Asterisk, Kamailio, or OpenSIPS.
Implement SIP-based features (e.g., INVITE, ACK, BYE, REGISTER, REFER, OPTIONS).
Use SIPREC for recording VoIP calls and manage RTP media streams.
Troubleshoot SIP call flows and analyze SIP traces using Wireshark, sngrep, etc.
Build and integrate RESTful APIs to support VoIP features.
Collaborate with cross-functional teams to deliver high-quality, scalable solutions.
Required Skills & Experience:
5–8 years of experience in VoIP development.
Strong knowledge of SIP protocol and related tools.
Proficient in JavaScript and Node.js.
Hands-on experience with SIPREC, RTP, and WebRTC.
Familiarity with networking concepts including UDP, TCP, NAT traversal, STUN/TURN.
Ability to debug complex telephony and media issues in production environments.
Preferred Tools:
Wireshark, sngrep, FreeSWITCH, Asterisk, Kamailio, OpenSIPS, Postman, Git.
How to Apply:
Please submit your resume along with a brief cover letter to [[email protected]] or apply directly via Indeed.
Job Types: Full-time, Permanent
Pay: ₹5+2,683.83 - ₹1,18+,837.17 per year
Schedule:
Monday to Friday
Work Location: Remote
💡 Quick Summary
Seeking a career-building opportunity? The SIP Developer position is now open for candidates interested in the Work from home Jobs sector. This role in Bangalore offers a professional environment and growth potential.
Requirement Snapshot: Candidates should possess basic communication skills, a proactive attitude, and the ability to work in a team. Experience in Work from home Jobs is a plus.
