Real-time streaming. Under 500ms, where the moment needs it.
Sub-500ms end-to-end streaming for use cases where interactive response matters: esports, auctions, trading broadcasts, live Q&A. SRT and WebRTC protocols with MCR-grade encoding.
- <200ms
- <500ms
- Live
- Any
What we deliver on every engagement.
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SRT protocol for reliable sub-500ms delivery over lossy networks
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WebRTC for sub-200ms peer-to-peer or two-way interactive streaming
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Hardware encoding with minimal buffering for fastest glass-to-glass time
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Network monitoring with real-time latency visualisation per destination
Some broadcasts live or die by the delay.
A standard streaming delay of 8-15 seconds is fine for a keynote. It is catastrophic for an auction where bidders clash over the last second, for a trading floor where price movements are the story, for a live Q&A where the delay turns dialogue into confusion, or for an esports tournament where viewers see the result before they see the play.
Low-latency streaming is a different engineering problem from broadcast streaming. Buffers are minimised. Encoders are hardware. Delivery protocol is SRT or WebRTC, not HLS. CDN configuration is tuned for freshness over resilience. Every millisecond of latency that can be saved is saved.
We run low-latency productions to spec, with network telemetry visible to the production team in real time. If a link starts adding latency, we see it and switch before it affects the audience.
Where under-500ms matters.
- Competitive gaming depends on sub-second broadcast for live commentary, viewer reaction, and anti-spoiler timing. Sub-500ms SRT delivery keeps the cast in sync with the play.
- High-value auctions where bidders compete across geographies need sub-second delivery so all bidders see the current bid at the same time. Bidding is only fair when latency is equal.
- Analyst broadcasts, earnings commentary, and financial news where the delay between the speaker and the audience determines whether the insight is actionable.
- Where the conversation between audience and speaker is the point. 8-second buffering turns dialogue into awkward silence. Sub-500ms keeps it live.
- Remote guests joining a broadcast need under-200ms round-trip latency to look natural on camera. WebRTC handles the guest leg; SRT handles the audience leg.
Specification.
- Secure Reliable Transport protocol with packet loss recovery, 1-2 second latency target glass-to-glass. Multi-stream with adaptive bitrate. Resilient to lossy networks.
- Sub-200ms latency for interactive and two-way applications. Browser-based viewer (no app install). Limited to moderate concurrent audiences (typically under 10,000 per session) before scaling complexity increases.
- Hardware H.264 or H.265 encoding with minimal buffering. B-frames disabled on lowest-latency configurations. Adaptive bitrate output for viewer-side variability.
- SRT or WebRTC ingest with redundancy. Backup encoder runs hot in parallel; failover is in-stream.
- SRT-aware delivery with edge regions tuned for GCC viewer geography. WebRTC delivery via our own WebRTC gateway.
- Real-time latency telemetry per destination. Production team sees each platform's live delay in milliseconds. If latency drifts, switch before audience notices.
- LL-HLS (Low-Latency HLS) is an option for audiences already on HLS-native players (YouTube Live supports it). Achievable latency: 3-5 seconds, better than standard HLS but not under 500ms.
Esports broadcast production
Esports broadcast production and video game tournament production for leagues and competitions. Multi-camera, instant replay, low-latency SRT streaming.
Learn moreAudience engagement
Live polling, Q&A moderation, interactive overlays, and gamification for live events and webinars. Keep audiences actively participating throughout broadcasts.
Learn moreWebinar production
Webinar production in Dubai. Broadcast cameras, up to eight speakers, live polls and questions answered on air, and a ranked attendee list your sales team can act on.
Learn moreWhat buyers ask before they book.
What actually determines end-to-end latency?
Encoder buffering + network transit + CDN caching + player buffering. SRT optimises all four. WebRTC optimises the first three but limits player compatibility to browsers. Regular HLS prioritises resilience over latency and cannot go under about 6 seconds.
Can viewers watch in a normal browser without installing anything?
Yes. WebRTC delivery runs in any modern browser. SRT delivery usually goes through a custom web player that wraps the SRT stream. No app install required on either protocol.
Can we do two-way interaction, not just one-way broadcast?
Yes. WebRTC is built for two-way. Guests join from a browser, appear on the broadcast, and can hear the audience questions in real time. Used on our webinar and hybrid productions for remote guest integration.
How many concurrent viewers can low-latency streaming handle?
SRT scales like HLS and can handle hundreds of thousands of concurrent viewers via CDN. WebRTC scales differently; practical upper bound on a single session is in the low tens of thousands without multi-region mesh architecture.
Can we combine low-latency delivery with standard HLS for broader reach?
Yes. Dual-protocol delivery is standard: SRT or WebRTC for the latency-critical audience (bidders, competitors, interactive participants) and standard HLS for everyone else. One production, two audiences, two experiences.
Tell us who needs to be in real time, and why.
We scope latency against the use, not a package.