Voice Engineer

Job Summary

We are looking for a Voice Engineer with 1–2 years of hands-on experience in voice communication technologies and real-time media protocols. The ideal candidate should have a strong understanding of SIP, RTP, and media servers, along with the ability to troubleshoot, develop, and support voice communication solutions.

Experience with WebRTC is an added advantage.

Key Responsibilities

• Develop, maintain, and support voice communication and telephony-based software solutions.
• Work with SIP (Session Initiation Protocol) for call signaling, session establishment, and call management.
• Implement and troubleshoot RTP (Real-time Transport Protocol) for real-time audio and media transmission.
• Work with media servers to support voice processing, call routing, and media streaming.
• Troubleshoot voice quality issues, call connectivity, signaling failures, and media transmission problems.
• Analyze SIP messages, RTP packets, and call flows to identify and resolve technical issues.
• Collaborate with software engineering and infrastructure teams to improve voice application performance.
• Support integration and testing of voice communication components.
• Ensure reliability, scalability, and performance of voice-based applications.

Required Technical Skills

Required Qualifications

• 1–2 years of experience in voice engineering, telecommunications, or software engineering.
• Hands-on knowledge of SIP and RTP protocols.
• Practical experience working with media servers.
• Understanding of voice signaling, call flows, and real-time media communication.
• Strong troubleshooting and analytical skills.
• Good communication and collaboration skills.

Preferred Qualifications

• Exposure to WebRTC technologies.
• Understanding of real-time communication architectures.
• Familiarity with voice quality monitoring and troubleshooting tools.

Work Location

Austin, TX – Onsite (Mandatory)