Synechron seeks a proficient Real-Time Voice/Media Engineer with expertise in Twilio, WebRTC, and low-latency media streams to support telephony and voice applications. This role is essential for building and maintaining real-time communication pipelines that deliver high-quality audio with minimal latency, enabling seamless customer interactions and reliable voice solutions. The engineer will focus on optimizing media handling, telemetry, and orchestration, ensuring the delivery of efficient, scalable, and robust real-time voice features aligned with organizational goals.
Software Requirements
Required Skills:
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Twilio PSTN & WebRTC: Expertise in integrating Twilio voice services including PSTN connectivity, WebRTC signaling, and media streams.
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Media Stream Orchestration (WebSocket): Experience in orchestrating real-time media and signaling using WebSocket protocols for low-latency communication.
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Real-Time Audio Pipelines: Design and implementation of high-performance audio pathways, including barge-in and turn-taking mechanisms.
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Telemetry & Latency Monitoring: Implementing and analyzing telemetry for each stage of voice processing (ASR, LLM, TTS) to monitor and optimize latency.
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Multilingual Speech Configurations: Experience configuring speech systems with support for Spanish and English, including handling speech recognition and synthesis.
Preferred Skills:
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Familiarity with additional telephony platforms or media services beyond Twilio.
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Knowledge of media encoding standards, codecs, and jitter buffer management.
Overall Responsibilities
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Develop and optimize real-time voice and media streaming solutions using Twilio and WebRTC technologies.
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Implement telemetry and monitoring tools to measure and reduce latency at various stages of voice processing, including speech recognition and synthesis.
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Manage media orchestration and signaling flow using WebSocket channels to ensure smooth interactions, barge-in, and turn-taking functionality.
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Collaborate with software engineering teams to integrate voice features into broader customer engagement solutions.
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Maintain high-quality voice streams by troubleshooting issues related to latency, jitter, packet loss, and network disruptions.
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Configure and tune telephony components to meet reliability, scalability, and security standards.
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Continuously monitor system performance, providing insights and improvements for latency reduction and user experience enhancement.
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Document configurations, protocols, and procedures for current and future reference.
Technical Skills (By Category)
Programming Languages:
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Required: JavaScript/TypeScript (WebRTC signaling, front-end integrations), familiarity with scripting languages (e.g., Python, Bash) for automation.
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Preferred: Experience with low-level media processing languages (C/C++).
Databases / Data Management:
Cloud Technologies:
Frameworks and Libraries:
Development Tools & Methodologies:
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Version control: Git, GitHub or Azure DevOps.
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Agile development practices, continuous integration/deployment processes.
Security Protocols:
Monitoring & Telemetry:
Experience Requirements
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At least 3-5 years of experience working with real-time voice communications, media streams, or telephony systems.
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Proven experience configuring and managing Twilio Voice, WebRTC implementations, or similar platforms.
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Experience monitoring, analyzing, and optimizing low-latency communication systems, including telemetry for latency measurement.
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Familiarity with multilingual speech recognition and synthesis in commercial or custom environments is advantageous.
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Prior work in contact center, customer engagement, or voice-enabled applications preferred.
Day-to-Day Activities
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Developing and refining real-time media pipelines to enable low-latency voice communication.
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Configuring and managing media signaling via WebSocket connections, ensuring efficient session management.
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Building telemetry and monitoring dashboards to track latency, jitter, packet loss, and other performance metrics.
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Troubleshooting audio quality issues, latency spikes, and media disruptions, implementing solutions promptly.
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Collaborating with product teams and developers to integrate voice features and ensure quality standards.
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Performing regular system tests, tuning buffer sizes, and adjusting configurations to optimize performance.
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Documenting system architectures, protocols, and operational procedures.
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Participating in backlog grooming, sprint planning, and review meetings to align technical development with business needs.
Qualifications
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Educational: Bachelors degree or higher in Computer Science, Electrical Engineering, Telecommunications, or related fields. Equivalent practical experience is acceptable.
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Certifications: Relevant telephony, VoIP, or WebRTC certifications (e.g., Twilio Certified, WebRTC Developer) are a plus.
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Training & Development: Commitment to ongoing learning in voice technology, latency optimization, and network security.
Professional Competencies
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Analytical mindset with strong troubleshooting and problem-solving skills.
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Ability to interpret telemetry data and translate insights into actionable improvements.
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Excellent communication skills for technical collaboration and documentation.
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Team-oriented approach with experience working across cross-functional teams.
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Adaptability to evolving voice and conferencing technologies, with a focus on security and performance.
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Ability to manage multiple priorities and deliver solutions within specified timelines.
