5 quick facts about the project

Client

FCC-licensed Video Relay Service (VRS) provider

Use case

Users access live ASL interpretation in any real-time video call without coordination steps or extra hardware.

Team

Media technology engineers, IT architect, and delivery managers

Solution

FCC-certified VRS platform bridging WebRTC and SIP PSTN infrastructures with real-time media processing and call recording

Messaging and integrations

Supports Zoom, Google Meet

What success looks like

< 300 ms

Audio latency across WebRTC to SIP bridge for natural conversation flow

5+

VRS carrier and PSTN integrations through a single platform

FCC

Certified and compliant from first production release

The use case that started it all

The company’s Product Manager reflects:
“We needed a software solution to connect deaf and hearing participants with sign-language interpreters in real time.
A key complexity in this process stems from the fact that participants often connect through entirely different communication systems, including modern video platforms, traditional phones, and VRS infrastructure.One of the biggest challenges was that the communication systems our users relied on operate in fundamentally different ways. Bringing them together into a single, seamless experience was far more complex than it appeared for us.

We decided to delegate solving this technical gap to subject-matter experts.”

Defining the system requirements

“We saw the problem and basically said: “Alright, challenge accepted.”

Our job was to deliver video calls with low enough latency for real-time interpreting, even though each provider behaves slightly differently. We also had to account for the fact that calls take place in a tightly regulated, privacy-sensitive environment.

From day one, we brought in senior WebRTC and media technology experts to streamline delivery and avoid a learning curve on the client’s side.”

— Oxagile’s Delivery Manager

Easy access
One-click joining for video calls with an interpreter
No extra hardware
Works with existing VRS and phone infrastructure
Compliant by design
Built to meet FCC requirements
Low latency
Real-time, no-lag communication for live sign-language interpreting

The VRS solution in a nutshell

Real-time WebRTC – SIP/PSTN connector

Connects online video conferencing apps with interpreters on traditional phone, PSTN, and VRS networks
Enables calls to smoothly operate in both ecosystems

VRS workflow support

Relay-specific functionality missing from mainstream platforms
Includes interpreter relay handoff and Voice Carryover
Supports DTMF menu navigation and compliant call recording workflows

VRS workflow support

High-performance media layer built with Go and GStreamer
Real-time transcoding and mixing of audio/video streams
Supports Opus, G.722, G.711, VP8, and H.264 codecs

Cloud-native infrastructure

Deployed on AWS ECS/Fargate
Infrastructure provisioned as code using OpenTofu
Built-in secrets management, multi-AZ resilience, and HIPAA-grade security controls

Tech obstacles we went through

Codec fragmentation across endpoints

A real-time transcoding pipeline decodes, resamples, and re-encodes audio and video per session so every participant receives a compatible format with no perceptible delay.

Inconsistent SIP behavior across VRS carriers

A carrier-aware SIP layer with domain whitelisting and structured interoperability testing isolates provider-specific quirks so one carrier cannot destabilize the platform.

Cross-protocol orchestration

A central orchestration service coordinates WebRTC and SIP components as a unified state machine to provide deterministic call lifecycle management and eliminate race conditions.