ByteLeapsBYTELEAPSEngineering StudioContact Us →
Service

WebRTC Real-Time Video & Audio Solutions

WebRTC (Web Real-Time Communication) enables browsers and mobile apps to stream audio, video, and data directly between peers with sub-second latency, via SFUs, media servers, signaling controllers, and camera stream decoders.

Start a project →
What is WebRTC solutions development?

WebRTC (Web Real-Time Communication) is a technology enabling browsers and mobile apps to stream audio, video, and data directly between peers with sub-second latency. WebRTC development involves engineering selective forwarding units (SFUs), media servers, signaling controllers, and camera stream decoders.

Sub-Second Latency WebRTC Engineering

Traditional video streaming (HLS, RTMP) suffers from 5-30 second lag times. We engineer custom WebRTC infrastructure supporting sub-second video calling, screen sharing, vector coordinates exchange, and RTSP IP camera decoding directly to browsers.

Key benefits
Sub-Second Lag: Media streaming under 400ms globally, ideal for intercoms, robotics, calls, and live bids.
SFU Scalability: Deploying media relays (Mediasoup, Janus) to handle high-volume streams without crushing client bandwidth.
RTSP to WebRTC Conversion: Converting surveillance IP camera feeds in real-time to play on modern browsers without plugins.
Data Channel Sync: Secure data channels to transfer drawing coordinates, chat messages, or game states in real-time.
Technologies
MediasoupJanusCoturn (STUN/TURN)WebRTC APIWHIP/WHEPNode.jsGStreamerFFmpeg
Our process
01Topology Selection

Choosing Mesh (simple calls), SFU (group conferences), or MCU (video mixing) architectures.

02Signaling & TURN Setup

Building WebSocket signaling servers and deploying TURN relays to bypass firewalls.

03Media Server Integration

Setting up SFU pipelines, routing audio tracks, and configuring packet re-transmissions.

04Client Tuning

Handling browser permissions, resolving device shifts, and managing connection drops.

FAQ

Frequently asked questions

WebRTC latency is sub-second, typically between 150ms to 400ms. This is significantly faster than standard HLS or RTMP streaming.
We deploy SFU (Selective Forwarding Unit) architectures like Mediasoup or Janus. This distributes bandwidth loads and supports hundreds of concurrent viewers.
Ready to get started?

Let's talk about your webrtc solutions company project.

Start a project →