WebRTC and low-latency streaming, tuned for your use case.
Low latency is a trade-off, not a setting. WebRTC gets you closest to real time but is harder to scale; LL-HLS scales through ordinary CDNs with a few seconds of delay; standard HLS is the cheapest and slowest. We help you choose, then build it properly.
Book a callThe right delay for the product
Interactive shows, betting and auctions need near real time; a concert can take a few seconds. We size latency to the use case.
Less bandwidth per viewer
Codec and ABR ladder choices that keep quality up while cutting delivery cost.
Resilient on bad networks
SRT error recovery on ingest and adaptive bitrate on playback, so a weak link doesn't stall the stream.
- WebRTC with WHIP ingest and WHEP playback
- SRT contribution links with ARQ recovery
- LL-HLS and CMAF packaging
- H.264, HEVC and AV1 encoding
- ABR ladder design and testing
- End-to-end latency measurement
- 01
Measure today's delay
Glass-to-glass latency on your current setup, stage by stage, so we know where the seconds go.
- 02
Choose the protocol mix
Often WebRTC for the few who need real time and LL-HLS for everyone else.
- 03
Tune encoding
GOP length, segment size and the ABR ladder, tested on real network profiles.
- 04
Prove it under load
Latency and quality checked at the viewer counts you expect.
Where we've done this.
Is WebRTC always the best choice for low latency?+
No. It gives the lowest delay, but it costs more to scale to large audiences. Many products mix WebRTC for interactive users with LL-HLS for everyone else.
Can you lower latency without changing our player?+
Often, yes. Shorter segments, GOP tuning and LL-HLS support in common players can cut delay without a rewrite.
Building something live?
Tell us what you're building. We'll tell you how we'd stream it, and what it takes to ship.
Book a free infra call