Most approaches to WebRTC video conferencing use a limiting approach called SFU.
SFU devours bandwidth, device CPU, and battery. SFU delivers individual video and audio streams of each participant to every other participant, leaving the work of decoding and rendering the media to the device CPUs. So with 5 participants, every participant receives and decodes 4 video streams. 16 participants? 15 streams. You see where this is going?
The SignalWire Video Conferencing API uses an MCU to mix all video and audio in the cloud and distribute a single stream back to every participant. 50-80% less bandwidth and compute consumption on the client-side + <50ms round-trip latency + none of the video and audio artifacts seen with SFU = unified, high production value experiences for large audiences.
Instead of 20 active participants and a few hundred passive spectators, support 300 active video and audio streams in every room...and run hundreds of rooms at the same time.
3-4 prebuilt video layouts may work for basic video calls, simple webinars, and small virtual classrooms. When your audience is large and your production values need to be premium, you'll want more granular options. We offer 12.
Run your video apps in real-time through natural language and machine vision models and deliver utterly original experiences
Our unique cloud multiplexing empowers you to integrate video conferencing rooms for up to 300 people per room (and run hundreds of those at once)...adding audio effects and XR overlays or running real-time NLP and machine vision models...all without consuming your customers' bandwidth and CPU.
Build video conferencing features that scale, traverse firewalls, work across browsers, smartphones, and (eventually) XR devices. Integrate with your existing apps and phone systems. Maintain <50ms latency while minimizing bandwidth and CPU consumption on your customers' devices. Do it all without worrying about any of that yourself.
With the Chat API, add basic 1:1 and group chat features to video rooms without adding another API to the stack.
Native SIP and PSTN integration means your video conferencing rooms can extend an existing telecom stack without changing vendors or adding new equipment, conference in voice callers via SIP, dial out to old-school phone numbers, and send SMS invites with links to join the room.
Customize everything precisely as you wish with our SDKs or ship a basic video conferencing feature with Embeddable Rooms.
100% of features available with no upfront fees or subscriptions required.
Rates are per-minute, per-participant.
|MONTHLY MINUTES||STANDARD (720p)||FULL HD (1080p)|
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