Build low-latency Vision AI applications using our new open-source Vision AI SDK. ⭐️ on GitHub
Stream handles video complexity, maintains modern SDKs, and provides the native Chat integration, so your team can ship video calling and livestreaming in hours, not months.
From a globally distributed edge network to adaptive media optimization, every layer of Stream's infrastructure is engineered for low-latency, high-reliability performance at scale.
With SFUs distributed across every major region, every participant connects through the closest available server, ensuring low latency and minimal packet loss even for calls that span multiple continents.
Edge InfrastructureProven at scale with 100,000 concurrent participants. Zero crashes, zero API failures, rock-solid frame rates.
Real-time codec switching, redundant audio encoding, and multi-datacenter failover ensure every call stays sharp and stable, regardless of network conditions.
Dynamically adjusts resolutions up to 1080p and switches codecs like AV1, VP9, and H.264 based on network and device.
Redundant audio encoding handles packet loss while DTX skips silence to save bandwidth. Reliable audio quality on any network.
Multi-datacenter failover and auto-scaling handle spikes to 100,000+ participants at 225 Gbps peak with zero API failures and 30 FPS stability.

Components that give you full control over UI, flows, and performance—no opinionated constraints.

Globally distributed SFUs keep media close to users for low-latency, predictable performance.

Architecture designed to grow from 1:1 calls to large-scale sessions without architectural changes.

Real-time voice agents with low-latency audio, streaming inference, and event-driven control.

An open-source framework powered by Stream Video that allows agents to see, hear, and remember.

Build low-latency video calling APIs with full control over UI, flows, and scaling.

Build 1:1 calls or a Telegram-clone with crystal-clear audio, unmatched reliability, and scale.

Low-latency broadcasting from your phone or RTMP and scale reliably to millions.

Scalable, low-latency audio rooms built on edge-based real-time infrastructure.

Power high-quality direct calls or meetings with multiple participants.

Broadcast to over 100k participants with high quality and low latency.

Optimize call quality and costs with adaptive resolution that adjusts to network and client conditions.

Improve the user experience by eliminating background noises, voices, and echo.

Safely record calls to Stream or external storage systems. Multiple resolutions are available.

Transcribe your calls to multiple languages.
Yes. Stream’s video infrastructure is distributed across global edge locations to deliver ultra-low latency and higher reliability, even at scale.
We offer client SDKs for Python AI, React, React Native, iOS, Android, Flutter, and more. You can build with or without our prebuilt UI kits.
Yes. Stream supports livestreams with 100k+ viewers using RTMP ingest and HLS playback, with WebRTC for low-latency interactions.
You can integrate Stream's AI Moderation API to gain greater control over live video content.
Yes. Stream is SOC2, ISO 27001, HIPAA, and GDPR compliant . We also support SAML/SSO, audit logs, and multi-tenant architectures for enterprise use.
Yes. Stream provides both prebuilt components for speed and low-level primitives for full customization.
Yes. Stream’s APIs are designed to work together seamlessly. You can combine Video with Chat for in-call messaging, Feeds for activity updates, and Moderation for real-time content safety—all within the same app experience.
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